From axg at syntec.co.uk  Wed Nov  1 02:06:09 2006
From: axg at syntec.co.uk (Andy)
Date: Wed, 1 Nov 2006 08:06:09 -0000
Subject: [VoIP] Power backup questions
In-Reply-To: <000501c6fd05$95524b20$0600a8c0@bluegrasspals.com>
Message-ID: <001101c6fd8c$97380ec0$8658b281@andy>

How about running them on a battery fed supply? They are probably both DC
machines anyway (if the input is ac take the cover off and you will probable
find a bridge recifier in there making it DC), so get a battery of the
correct voltage, and size to cover the length outage you think and have a
mains charger charging the battery.

The battery size could be reduced if you have the charger plugged in on your
ups.

Andy G
 

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Jayson Smith
Sent: 31 October 2006 16:00
To: Voice Over IP Tandem for Analog Switches
Subject: [VoIP] Power backup questions

Hi guys,
This is mostly off-topic, but this morning we had an approx. two-hour power
outage.  Yeah Doug, I know that's nothing next to what you just went
through!  Anyway, while the inexpensive UPS's kept all our computers going,
the router and DSL modem had problems.  I guess either the DSL modem or the
router just can't handle much in the way of power glitches.  They were both
on a UPS, but when the power died so did our DSL connection.  This has
happened before when we would have tiny power glitches during storms and the
UPS would kick in.  I'm considering the purchase of a more expensive UPS for
my computers, probably one of the full ones that runs on battery even when
the power's on so there's no switchover time when the power actually does
die.  However, I don't really want to spend several hundred dollars for
another such UPS just for our router and DSL modem.  That seems a bit
overkill to me.  Both these devices use the typical wall wart type power
transformers, so they can't be using much power.  I'm just wondering if
there's any type of inverter/something out there that could help eliminate
the switchover effects?  E.G. a very tiny full, always on battery, UPS that
would only actually run these things for, say, ten seconds would do the
trick, since the switchover happens well within one second of the power
going out and the regular UPS would then be powering it again.  Any
thoughts?
Thanks,
Jayson.


_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/

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From jnovack at stromberg-carlson.org  Wed Nov  1 07:43:34 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 01 Nov 2006 08:43:34 -0500
Subject: [VoIP] Power backup questions
In-Reply-To: <002201c6fd3b$dbc11600$0600a8c0@bluegrasspals.com>
References: <5.1.0.14.0.20061031171638.00b4eae0@incoming.verizon.net>
	<002201c6fd3b$dbc11600$0600a8c0@bluegrasspals.com>
Message-ID: <4548A486.1030308@stromberg-carlson.org>



Jayson Smith wrote:
<snip>
> I am using Minuteman UPS's bought at Radio Shack for about $40 each or so.
>   
Tripplite strikes once again
The APC 200 VA UPS are available for around the same price. Don't know 
if you have a micro center near you, but the BK350 sells for 42 bucks 
there and some other locations.

John Novack



From dfroula at sbcglobal.net  Thu Nov  2 13:11:11 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 2 Nov 2006 11:11:11 -0800 (PST)
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
In-Reply-To: <5.1.0.14.0.20061031190126.02cdb008@incoming.verizon.net>
Message-ID: <20061102191111.94647.qmail@web83206.mail.mud.yahoo.com>

I received my Smart-1 the other day and, after some
head scratching over the manual and odd programming,
got it working (sort of) as a dial pulse to tone
converter in fromt of my ATA (a Linksys PAP2).

The problem is that the Smart-1 picks up the outgoing
phone line as soon as the phone goes off-hook. By the
time I rotary dial all the digits and the Smart-1 cuts
through, the PAP2 ATA has already timed out to
reorder.

I can adjust interdigit timers in the PAP2 to wait
long enough after the first digit, but there is no
timer I can find to control the off-hook to reorder
timing.

Ideally, I'd like the Smart-1 not to pick up and dial
the outgoing line until it detects a pattern match on
the incoming digits. I've been through the manual a
few times, but can't see how this would be done.

Any help is appreciated.

Regards,

Don


From stfkerman at jps.net  Thu Nov  2 13:40:37 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Thu, 02 Nov 2006 14:40:37 -0500
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
In-Reply-To: <20061102191111.94647.qmail@web83206.mail.mud.yahoo.com>
References: <20061102191111.94647.qmail@web83206.mail.mud.yahoo.com>
Message-ID: <454A49B5.3030803@jps.net>



Donald Froula wrote:
> The problem is that the Smart-1 picks up the outgoing phone line as 
> soon as the phone goes off-hook. By the time I rotary dial all the 
> digits and the Smart-1 cuts through, the PAP2 ATA has already timed 
> out to reorder.
That is to prevent an incoming call on the associated CO line during 
digit acquisition by the Smart-1, which would make the call impossible 
to complete.
> I can adjust interdigit timers in the PAP2 to wait long enough after 
> the first digit, but there is no timer I can find to control the 
> off-hook to reorder timing.
>
> Ideally, I'd like the Smart-1 not to pick up and dial the outgoing 
> line until it detects a pattern match on the incoming digits. I've 
> been through the manual a
> few times, but can't see how this would be done.
I'm not familiar with the programming of the Smart-1.

A hardware kludge could be installed in the Smart-1 to open the loop, 
releasing the ATA until the last DP digit is received.  You would have 
to count digits and re-close the loop during the last digit so that the 
ATA would be ready to accept DTMF by the time the Smart-1 starts 
outpulsing.  Pretty kludgy but short of disassembling the Smart-1 
firmware or replacing the ATA with one that allows control over the 
permanent signal timing, no other solution comes to mind.

I have schematics for some Smart-1 versions, which would facilitate 
modifications should you choose this solution.  Perhaps the Sandman 
device, with an applique to prevent the ATA from detecting the dial 
pulses, is a simpler and better solution after all!

Steph


From dfroula at sbcglobal.net  Thu Nov  2 13:59:58 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 2 Nov 2006 11:59:58 -0800 (PST)
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
In-Reply-To: <454A49B5.3030803@jps.net>
Message-ID: <20061102195959.93668.qmail@web83214.mail.mud.yahoo.com>

Your explanation of why it picks up the outgoing line
makes perfect sense. Thanks for the suggestions.

I think I may have found a solution on the PAP2 side.
>From DSLReports archives:

"Look in the "Regional" tab of the voice admin pages.
The dial tone textbox has a default value of this:

350 at -19,440 at -19;10(*/0/1+2)

The 350 and 440 are the frequencies of the two-toned
dial tone. (Change them to hear your dialtone change,
if you like.  ) The -19 after the @ symbol controls
the volume of each tone. The 10 after the semi-colon
controls the duration of the dialtone. Increase this
number to 30, if you like--that should be PLENTY of
time.  The */0/1+2 sequence means that the tone is
always on with both frequencies. If you change the *
to a numeric value and change the 0 to a non-zero
number, you can get a repeating tone (check out the
busy signal entry or the MWI dialtone (stutter
dialtone) entry)."

Don

--- Steph Kerman <stfkerman at jps.net> wrote:

> 
> 
> Donald Froula wrote:
> > The problem is that the Smart-1 picks up the
> outgoing phone line as 
> > soon as the phone goes off-hook. By the time I
> rotary dial all the 
> > digits and the Smart-1 cuts through, the PAP2 ATA
> has already timed 
> > out to reorder.
> That is to prevent an incoming call on the
> associated CO line during 
> digit acquisition by the Smart-1, which would make
> the call impossible 
> to complete.
> > I can adjust interdigit timers in the PAP2 to wait
> long enough after 
> > the first digit, but there is no timer I can find
> to control the 
> > off-hook to reorder timing.
> >
> > Ideally, I'd like the Smart-1 not to pick up and
> dial the outgoing 
> > line until it detects a pattern match on the
> incoming digits. I've 
> > been through the manual a
> > few times, but can't see how this would be done.
> I'm not familiar with the programming of the
> Smart-1.
> 
> A hardware kludge could be installed in the Smart-1
> to open the loop, 
> releasing the ATA until the last DP digit is
> received.  You would have 
> to count digits and re-close the loop during the
> last digit so that the 
> ATA would be ready to accept DTMF by the time the
> Smart-1 starts 
> outpulsing.  Pretty kludgy but short of
> disassembling the Smart-1 
> firmware or replacing the ATA with one that allows
> control over the 
> permanent signal timing, no other solution comes to
> mind.
> 
> I have schematics for some Smart-1 versions, which
> would facilitate 
> modifications should you choose this solution. 
> Perhaps the Sandman 
> device, with an applique to prevent the ATA from
> detecting the dial 
> pulses, is a simpler and better solution after all!
> 
> Steph
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 



From ian at uax.org.uk  Thu Nov  2 14:09:25 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Thu, 2 Nov 2006 20:09:25 -0000
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
References: <20061102191111.94647.qmail@web83206.mail.mud.yahoo.com>
Message-ID: <02be01c6feba$ce328f70$0a01a8c0@acer1dd0bbc6d0>

I too have been playing with a SMarT-1  - however mine is a nifty little 
version which doesn't appear to have made it to the 'colonies'  (They are 
made here in Wales or rather were made!)

It is a SMarT-1 Compact  - about 3in wide, 4in long and about 1in thick. 
What's more it can be line powered if you only need to go from LD to DTMF ! 
All I needed to do was to reset it to the Factory Defaults.  It the 
automatically detects where the telephone is LD or DTMF  and also if the 
Line/Switch is LD or DTMF.

I'm using it with one of the SX100-FX   IAX ATA's and it seems fine. When 
you lift the telephone the dial tone is received from the ATA.  You then 
dial the digits in LD.into the 'SMarT-1 Compact'  - the only downside side 
is that it waits for about 5 seconds after the telephone has dialled the 
last digit before it decides that the dialling has finished (dialling tone 
is still present during this period). Then the digits are released on one 
train of DTMF tones.  I'm not sure if it releases the digits through as LD 
to the ATA but they don't seem to affect the ATA.  All goes well from then 
on.

The ATA has proved to be OK working both locally and as an 'extension' off a 
distant Asterisk box.  Programming took about two minutes  - only three 
boxes to fill in and another couple of boxes to tick.  I found the 
manufacturers support OK. I could not program the first unit that arrived - 
I contacted customer support by email. They 'talked' me by email, through 
whilst they worked out if the unit was faulty or whether I was!  Took couple 
of days due to the time difference. They emailed me on a Friday morning to 
say that the unit was faulty and they were despatching a new one. It arrived 
the following Monday and was in use within minutes!
The unit is the S100-FX http://www.x100p.com/products/FXS.php   They only 
seem to sell them through their eBay shop http://stores.ebay.co.uk/x100p  at 
the moment. They have just updated their website www.x100p.com  plus 
incorporated a couple of changes I suggested last week!  You can now read 
the page without scrolling left to right!   It looks as though they are 
setting up to sell direct via their website.  They've also just brought 
another 'box' that links to the S100FX to convert/add an IAX FXO port as 
well.

Ian Jolly
+44 (0)352 85 26 (via a 1929 GPO Rural Automatic eXchange - UAX5)
CNET - the Heritage Telephone Network


----- Original Message ----- 
From: "Donald Froula" <dfroula at sbcglobal.net>
To: <voip at ckts.info>
Sent: Thursday, November 02, 2006 7:11 PM
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter


>I received my Smart-1 the other day and, after some
> head scratching over the manual and odd programming,
> got it working (sort of) as a dial pulse to tone
> converter in fromt of my ATA (a Linksys PAP2).
>
> The problem is that the Smart-1 picks up the outgoing
> phone line as soon as the phone goes off-hook. By the
> time I rotary dial all the digits and the Smart-1 cuts
> through, the PAP2 ATA has already timed out to
> reorder.
>
> I can adjust interdigit timers in the PAP2 to wait
> long enough after the first digit, but there is no
> timer I can find to control the off-hook to reorder
> timing.
>
> Ideally, I'd like the Smart-1 not to pick up and dial
> the outgoing line until it detects a pattern match on
> the incoming digits. I've been through the manual a
> few times, but can't see how this would be done.
>
> Any help is appreciated.
>
> Regards,
>
> Don
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> --
> This email has been verified as Virus free
> Virus Protection and more available at http://www.plus.net
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.409 / Virus Database: 268.13.23/513 - Release Date: 
> 02/11/2006
> 



From dfroula at sbcglobal.net  Thu Nov  2 16:53:42 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 2 Nov 2006 14:53:42 -0800 (PST)
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
In-Reply-To: <454A49B5.3030803@jps.net>
Message-ID: <20061102225342.91898.qmail@web83209.mail.mud.yahoo.com>

Steph, I understand there was a configurator program
to make the setup of the unit easier. Would you have
access to that program? I saw references to it on the
DSLReports web site.

Thanks for the suggestions.

Don

--- Steph Kerman <stfkerman at jps.net> wrote:

> 
> 
> Donald Froula wrote:
> > The problem is that the Smart-1 picks up the
> outgoing phone line as 
> > soon as the phone goes off-hook. By the time I
> rotary dial all the 
> > digits and the Smart-1 cuts through, the PAP2 ATA
> has already timed 
> > out to reorder.
> That is to prevent an incoming call on the
> associated CO line during 
> digit acquisition by the Smart-1, which would make
> the call impossible 
> to complete.
> > I can adjust interdigit timers in the PAP2 to wait
> long enough after 
> > the first digit, but there is no timer I can find
> to control the 
> > off-hook to reorder timing.
> >
> > Ideally, I'd like the Smart-1 not to pick up and
> dial the outgoing 
> > line until it detects a pattern match on the
> incoming digits. I've 
> > been through the manual a
> > few times, but can't see how this would be done.
> I'm not familiar with the programming of the
> Smart-1.
> 
> A hardware kludge could be installed in the Smart-1
> to open the loop, 
> releasing the ATA until the last DP digit is
> received.  You would have 
> to count digits and re-close the loop during the
> last digit so that the 
> ATA would be ready to accept DTMF by the time the
> Smart-1 starts 
> outpulsing.  Pretty kludgy but short of
> disassembling the Smart-1 
> firmware or replacing the ATA with one that allows
> control over the 
> permanent signal timing, no other solution comes to
> mind.
> 
> I have schematics for some Smart-1 versions, which
> would facilitate 
> modifications should you choose this solution. 
> Perhaps the Sandman 
> device, with an applique to prevent the ATA from
> detecting the dial 
> pulses, is a simpler and better solution after all!
> 
> Steph
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 



From greg at vyger.net  Thu Nov  2 21:07:56 2006
From: greg at vyger.net (Greg Blakely)
Date: Thu, 2 Nov 2006 21:07:56 -0600
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
Message-ID: <A691C3685A152E4AAC9608B8D8E113163EB050@godzilla.hickory.vyger.net>

The default route in a SMART-1 is run by this command:

#80327##

The first three digits translate to "wait for precise dial tone", and
the rest of it amounts to "dial the number and cut through."

If the precise dial tone has timed out, the SMART-1 does not detect it,
and **should** hang up and then re-seize the line, detect the dial tone,
and then dial and cut through.

If this is not happening, it may be that the #803 part of the sequence
is missing...  

Something to check, anyway.

Greg.

PS.  I'm fairly good at programming SMART-1 dialers, since I installed
them for a 0+ provider for about three years, back in the late 1980s and
early 1990s.  I'm not an expert, by any means, but I can usually get
them running.

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
Of
> Donald Froula
> Sent: Thursday, November 02, 2006 1:11 PM
> To: voip at ckts.info
> Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
> 
> I received my Smart-1 the other day and, after some
> head scratching over the manual and odd programming,
> got it working (sort of) as a dial pulse to tone
> converter in fromt of my ATA (a Linksys PAP2).
> 
> The problem is that the Smart-1 picks up the outgoing
> phone line as soon as the phone goes off-hook. By the
> time I rotary dial all the digits and the Smart-1 cuts
> through, the PAP2 ATA has already timed out to
> reorder.
> 
> I can adjust interdigit timers in the PAP2 to wait
> long enough after the first digit, but there is no
> timer I can find to control the off-hook to reorder
> timing.
> 
> Ideally, I'd like the Smart-1 not to pick up and dial
> the outgoing line until it detects a pattern match on
> the incoming digits. I've been through the manual a
> few times, but can't see how this would be done.
> 
> Any help is appreciated.
> 
> Regards,
> 
> Don
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 




From dfroula at sbcglobal.net  Fri Nov  3 08:21:52 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 3 Nov 2006 06:21:52 -0800 (PST)
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter
In-Reply-To: <A691C3685A152E4AAC9608B8D8E113163EB050@godzilla.hickory.vyger.net>
Message-ID: <20061103142152.35251.qmail@web83210.mail.mud.yahoo.com>

Perfect! This works with no dependency of the dial
tone settings in the PAP2. I see in the manual that it
will retry the connection if no dial tone detected,
but failed to see the utility in my case.

Don

--- Greg Blakely <greg at vyger.net> wrote:

> The default route in a SMART-1 is run by this
> command:
> 
> #80327##
> 
> The first three digits translate to "wait for
> precise dial tone", and
> the rest of it amounts to "dial the number and cut
> through."
> 
> If the precise dial tone has timed out, the SMART-1
> does not detect it,
> and **should** hang up and then re-seize the line,
> detect the dial tone,
> and then dial and cut through.
> 
> If this is not happening, it may be that the #803
> part of the sequence
> is missing...  
> 
> Something to check, anyway.
> 
> Greg.
> 
> PS.  I'm fairly good at programming SMART-1 dialers,
> since I installed
> them for a 0+ provider for about three years, back
> in the late 1980s and
> early 1990s.  I'm not an expert, by any means, but I
> can usually get
> them running.
> 
> > -----Original Message-----
> > From: voip-bounces at ckts.info
> [mailto:voip-bounces at ckts.info] On Behalf
> Of
> > Donald Froula
> > Sent: Thursday, November 02, 2006 1:11 PM
> > To: voip at ckts.info
> > Subject: [VoIP] Mitel Smart-1 as Pulse to Tone
> Converter
> > 
> > I received my Smart-1 the other day and, after
> some
> > head scratching over the manual and odd
> programming,
> > got it working (sort of) as a dial pulse to tone
> > converter in fromt of my ATA (a Linksys PAP2).
> > 
> > The problem is that the Smart-1 picks up the
> outgoing
> > phone line as soon as the phone goes off-hook. By
> the
> > time I rotary dial all the digits and the Smart-1
> cuts
> > through, the PAP2 ATA has already timed out to
> > reorder.
> > 
> > I can adjust interdigit timers in the PAP2 to wait
> > long enough after the first digit, but there is no
> > timer I can find to control the off-hook to
> reorder
> > timing.
> > 
> > Ideally, I'd like the Smart-1 not to pick up and
> dial
> > the outgoing line until it detects a pattern match
> on
> > the incoming digits. I've been through the manual
> a
> > few times, but can't see how this would be done.
> > 
> > Any help is appreciated.
> > 
> > Regards,
> > 
> > Don
> > 
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> > 
> 
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 



From jnovack at stromberg-carlson.org  Fri Nov  3 08:46:12 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 03 Nov 2006 09:46:12 -0500
Subject: [VoIP] 4 port T1 card
Message-ID: <454B5634.3010403@stromberg-carlson.org>


Anyone else received and got their fire sale 4 port T1 card working/

I finally received mine, got it working fairly quickly, but port one 
would go into a yellow alarm after a period. move the same channel  bank 
and cable to 2,3 or 4 and all seems OK with minimal testing.
Contacting support at govarion.com yielded no response, so I sent a message 
to sales. They promised to send an advance replacement after I supplied 
a CC number, which I did, then promised a tracking number and shipment 
the next day.
Since then no response to numerous E-mails, no replacement card, no 
tracking number.

Anyone else have better results?

John Novack


From voiptandem at shaneyoung.com  Fri Nov  3 10:27:12 2006
From: voiptandem at shaneyoung.com (Shane Young)
Date: Fri, 03 Nov 2006 10:27:12 -0600
Subject: [VoIP] 4 port T1 card
In-Reply-To: <454B5634.3010403@stromberg-carlson.org>
References: <454B5634.3010403@stromberg-carlson.org>
Message-ID: <20061103102712.qinznheo00wss8o4@mail.shaneyoung.com>

Quoting John Novack <jnovack at stromberg-carlson.org>:

>
> Anyone else received and got their fire sale 4 port T1 card working/
>

The first one never made it to me.  After a paypal case was opened,  
for about a week or two, one showed up in the mail.  It's been ok, but  
I haven't beat it up yet.  I need to put channel banks on all 4 ports  
for a week and see how it goes.

Jim Dixon, who designed the board, reccomended Varion because of their  
high-quality parts and uses the boards made by Varion.  I'm not sure  
why they are being such stinkers.

I'd like to order another one or two.  Phil McCarter wants one for  
sure and there may be others.

--Shane




From dfroula at sbcglobal.net  Fri Nov  3 11:38:03 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Fri, 3 Nov 2006 09:38:03 -0800 (PST)
Subject: [VoIP] A valuable lesson
In-Reply-To: <20061103102712.qinznheo00wss8o4@mail.shaneyoung.com>
Message-ID: <20061103173803.96383.qmail@web83208.mail.mud.yahoo.com>

Well, I learned a lesson about proper use of contexts
in one's extensions.conf Asterisk file!

My wife called yesterday to say the police showed up
at the door, claiming a 911 call had been received
from my address. After reassurance, they departed.

I browsed the CDR file and spotted what happened. Some
poor CNETer (you may remain anonymous) was
experimenting with ny voice dialer number and hit a
seven digit number that turned out to be the access
code for my FXO card, followed by 9-1111 (XX9-1111). 

He found himself connected to our local 911 center!
Poor guy....

Needless to say, I fixed that. Need to take my wife to
dinner this weekend....

Don


From jnovack at stromberg-carlson.org  Fri Nov  3 12:06:44 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 03 Nov 2006 13:06:44 -0500
Subject: [VoIP] A valuable lesson
In-Reply-To: <20061103173803.96383.qmail@web83208.mail.mud.yahoo.com>
References: <20061103173803.96383.qmail@web83208.mail.mud.yahoo.com>
Message-ID: <454B8534.80502@stromberg-carlson.org>

You probably want to reconfigure your dial plan to prevent that 
happening again.
Look at better use of contexts and context includes.
My plan prevents any CNET access to either my voice mail checking, or my 
outbound services that cost me.
I also have no direct access to my PSTN connections.
If you have too many 911 calls to or from your house, you may not like 
the results. Some jurisdictions take a really dim view of false 911 calls.

John Novack

Donald Froula wrote:
> Well, I learned a lesson about proper use of contexts
> in one's extensions.conf Asterisk file!
>
> My wife called yesterday to say the police showed up
> at the door, claiming a 911 call had been received
> from my address. After reassurance, they departed.
>
> I browsed the CDR file and spotted what happened. Some
> poor CNETer (you may remain anonymous) was
> experimenting with ny voice dialer number and hit a
> seven digit number that turned out to be the access
> code for my FXO card, followed by 9-1111 (XX9-1111). 
>
> He found himself connected to our local 911 center!
> Poor guy....
>
> Needless to say, I fixed that. Need to take my wife to
> dinner this weekend....
>
> Don
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>
>   

From david at josephson.com  Fri Nov  3 14:10:42 2006
From: david at josephson.com (David Josephson)
Date: Fri, 03 Nov 2006 12:10:42 -0800
Subject: [VoIP] 4 port T1 card
In-Reply-To: <454B5634.3010403@stromberg-carlson.org>
References: <454B5634.3010403@stromberg-carlson.org>
Message-ID: <454BA242.4070908@josephson.com>

John Novack wrote:
> Anyone else received and got their fire sale 4 port T1 card working/
>   
I got mine (also after opening a Paypal case), but there is a big bulge 
and crack in the input transformer chip (usually caused by putting a 
board into a reflow oven without baking out the components that haven't 
been stored in dry air). I can't imagine it will work very well, and Ben 
Bawkon has assured me he'll replace it pronto. We'll see.

David Josephson




From jnovack at stromberg-carlson.org  Fri Nov  3 14:23:03 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 03 Nov 2006 15:23:03 -0500
Subject: [VoIP] 4 port T1 card
In-Reply-To: <454BA242.4070908@josephson.com>
References: <454B5634.3010403@stromberg-carlson.org>
	<454BA242.4070908@josephson.com>
Message-ID: <454BA527.5090204@stromberg-carlson.org>



David Josephson wrote:
> John Novack wrote:
>> Anyone else received and got their fire sale 4 port T1 card working/
>>   
> I got mine (also after opening a Paypal case), but there is a big 
> bulge and crack in the input transformer chip (usually caused by 
> putting a board into a reflow oven without baking out the components 
> that haven't been stored in dry air). I can't imagine it will work 
> very well, and Ben Bawkon has assured me he'll replace it pronto. 
> We'll see.
>
> David Josephson
That's what Ben told me on 10/23
No word or replacement yet, no response to any e-mails since 10/23
I have no idea where these guys even are in the world.

Wish us both luck
At least mine seems to work OK on 3 of the 4 ports.


JN



From jnovack at stromberg-carlson.org  Sat Nov  4 12:19:54 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sat, 04 Nov 2006 13:19:54 -0500
Subject: [VoIP] Varion 4 port card update, sort of
Message-ID: <454CD9CA.8080001@stromberg-carlson.org>


An E-mail dated 3 AM this morning now gives me a tracking number for the 
USPS shipment of a replacement card, though the USPS claims only to have 
received notification that an item will be shipped. dated  11/1
Ben claimed that the item would be "out the door" on 10/24

I guess the fire sale price of this card is somewhat reflective of the 
general health of the company. Their shipping schedule sure is.
Anyone having one of these cards better put it through it's paces, 
replacement of defective cards in a timely manner doesn't look promising

John Novack


From chad at maine.maine.edu  Sun Nov  5 16:34:28 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Sun, 05 Nov 2006 17:34:28 -0500
Subject: [VoIP] Power backup questions
In-Reply-To: <002201c6fd3b$dbc11600$0600a8c0@bluegrasspals.com>
Message-ID: <454E20A4.26428.F2C83AC@localhost>

> Hi,
[snip]
> One thing that annoys me is that while the power's out they
> go "Beep, beep!  Beep, beep!  Beep, beep!" all the time.  When the
> battery gets low they change to going more like, "Beep beep, beep
> beep, beepbeep, beepbeep, beebeep, beebeep!" or something, you get the
> idea.  Do more expensive UPS's either not beep as often or have an
> option to turn beeping off, at least until the battery is low? 
[big snip]

Professionally I use APC SmartUPS, while some would argue they are not Telco 
grade, they don't carry the telco price tag.

Two features with respect to alarming that many using SmartUPS are not aware of, 
is not only can the alarm be disabled, it can be set to only go off at "low battery", or 
"power fail + 30".  

A SmartUPS-700 runs one PC and a little network equipment fairly well.

Chad


From chad at maine.maine.edu  Sun Nov  5 16:51:14 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Sun, 05 Nov 2006 17:51:14 -0500
Subject: [VoIP] Power backup questions
In-Reply-To: <5.1.0.14.0.20061031190126.02cdb008@incoming.verizon.net>
References: <002201c6fd3b$dbc11600$0600a8c0@bluegrasspals.com>
Message-ID: <454E2492.10582.F3BDD70@localhost>

Speaking of wall warts, I was pleasantly surprised to find the wall wart for my new 
Netgear FS108P is 48 VDC.

The FS108P is Netgears Power Over Ethernet (POE) version of their eight port fast 
Ethernet switch (the FS108) and I think is their smallest POE offering.  Now one wall 
wart on my UPS powers not only the LAN here at the house, but my POE IP phones 
as well.

By the way, if anyone cares the FS108P which is an IEEE 802.af standard PSE 
device, will power pre-802.af Cisco 7900 IP phones via a FULL and PROPER 
Ethernet crossover cable (that fixes the 802.af standard problem being on the right 
pins, and the Netgear automatically handles the crossover by autocrossing itself).

Chad

> At 05:28 PM 10/31/2006 -0500, Jayson Smith wrote:
> >Do more expensive UPS's either not beep as often or have an option to
> >turn beeping off, at least until the battery is low?
> 
> As Steph noted, the ones with which I have had contact have an alarm
> silencing switch, to acknowledge the condition.  Mine, after being
> silenced for the initial outage, will then beep when the battery level
> is approaching cutoff.
> 
> >I think the Dlink router uses five volts, probably DC.  No idea about
> >the DSL modem.  Guess we'd have to take a voltmeter to those
> >transformers if we wanted to cobble together a charger.
> 
> I just looked at the labels on the wall worts for my DSL modem, router
> and small 8-port switch.  The DSL modem and router's wall worts have
> AC output, the switch's is DC.  One is 12VAC, another 9VAC and the DC
> one is 9V.  Most likely the router and modem's power inputs go right
> to a bridge rectifier, filtering and DC regulation, but without
> actually opening them up and attempting to trace out the circuit, one
> can't be certain they aren't using the AC for something, which makes
> the direct battery power option not as easy as it might be at first
> glance.  I had thought of the direct power route myself during our
> recent extended outage here but when I saw the AC inputs on two of the
> devices I decided against exploring that further at the time.
> 
> Doug.
> 
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 
> 
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.409 / Virus Database: 268.13.23/513 - Release Date:
> 11/2/06
> 




From keelan at mail.grenander.com  Mon Nov  6 23:31:13 2006
From: keelan at mail.grenander.com (Keelan Lightfoot)
Date: Mon, 6 Nov 2006 21:31:13 -0800
Subject: [VoIP] Nortel/Ambit ATA
In-Reply-To: <454E20A4.26428.F2C83AC@localhost>
References: <454E20A4.26428.F2C83AC@localhost>
Message-ID: <193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>

I thought I'd take a minute to share some information on what seems  
to be a unknown brand of ATA.

I happened across a couple Nortel single port ATAs (re-badged Ambit  
ATAs) a while back. Like Vonage ATAs, they were locked up. The only  
access I had was to a useless client-side interface where I could  
change their network port settings. I popped the case open, and there  
was a well-labeled serial port header on the board. With a RS-232- 
 >3.3v level convertor I was able to access the serial console  
interface, but it unfortunately was also limited without the correct  
password.

The good news is that if I rebooted it and hit ctrl-c during the boot  
sequence, it would dump to a very simple boot monitor (commands:  
help, exit, boot, memdump). I searched the resulting dump for the  
password for the area that I had access to. About 2 lines of garbage  
above my password was the supervisor password that gave me access to  
the entire system.

The interesting thing is that my two ATAs both had the same 6  
character password -- It makes me think twice about the security of  
some VOIP providers.

Anyway, if you have access to one of these things, finding the  
password is not too difficult given time and the motivation to do it.  
They support pulse dialing, and have a fairly competent web based  
management interface, and they play well with Asterisk.

- Keelan


From stfkerman at jps.net  Tue Nov  7 00:53:54 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Tue, 07 Nov 2006 01:53:54 -0500
Subject: [VoIP] Nortel/Ambit ATA
In-Reply-To: <193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>
References: <454E20A4.26428.F2C83AC@localhost>
	<193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>
Message-ID: <45502D82.5040502@jps.net>

Nice work, Keelan!

Steph

Keelan Lightfoot wrote:
> I thought I'd take a minute to share some information on what seems  
> to be a unknown brand of ATA.
>
> I happened across a couple Nortel single port ATAs (re-badged Ambit  
> ATAs) a while back. Like Vonage ATAs, they were locked up. The only  
> access I had was to a useless client-side interface where I could  
> change their network port settings. I popped the case open, and there  
> was a well-labeled serial port header on the board. With a RS-232- 
>  >3.3v level convertor I was able to access the serial console  
> interface, but it unfortunately was also limited without the correct  
> password.
>
> The good news is that if I rebooted it and hit ctrl-c during the boot  
> sequence, it would dump to a very simple boot monitor (commands:  
> help, exit, boot, memdump). I searched the resulting dump for the  
> password for the area that I had access to. About 2 lines of garbage  
> above my password was the supervisor password that gave me access to  
> the entire system.
>
> The interesting thing is that my two ATAs both had the same 6  
> character password -- It makes me think twice about the security of  
> some VOIP providers.
>
> Anyway, if you have access to one of these things, finding the  
> password is not too difficult given time and the motivation to do it.  
> They support pulse dialing, and have a fairly competent web based  
> management interface, and they play well with Asterisk.
>
> - Keelan
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>   

-- 
General request: Please *DO NOT* send messages with attachments larger than 300K total to this JPS email address without my prior agreement.  Send those messages to me at stfkerman at gmail.com.  I have a small mailbox at JPS. Large attachments will interfere with my email and cause other messages to bounce.  If you don't know how big the attachment is, please send it to stfkerman at gmail.com.  

Thanks.



From g4vft at btinternet.com  Tue Nov  7 13:51:59 2006
From: g4vft at btinternet.com (Jonathan Kay)
Date: Tue, 07 Nov 2006 19:51:59 +0000
Subject: [VoIP] 44-202
Message-ID: <4550E3DF.2060908@btinternet.com>

Alan is having some problems posting to the group.
He'd like it known that Dewlands Common is back on the air. After some 
problems with his wireless network.

Jon K


From dfroula at sbcglobal.net  Tue Nov  7 14:19:02 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 7 Nov 2006 12:19:02 -0800 (PST)
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter - Converts
	digit-by-digit!
In-Reply-To: <20061103142152.35251.qmail@web83210.mail.mud.yahoo.com>
Message-ID: <20061107201902.82165.qmail@web83205.mail.mud.yahoo.com>

My impression was the Smart-1 was strictly a store and
forward device that enforced one fixed route for each
length string of dialed digits.

However, I found a route combination that translates
dial pulses to touch tones as they are dialed,
variable length.

I set the default route (0) to "#8032#007##"

I then set search table 801 to "#9#8"

The search table will mach any first rotary digit and
go to route 0. Route 0 waits for a valid dialtone
(retrying if not found) and sends the first digit,
translated to DTMF. It will then wait for the
inter-digit timeout period, send a DTMF #, and cut
through. The interesting thing is that one can rotary
dial any number of digits after the first and they
will be translated to DTMF as they are dialed. After
the inter-digit time pause, the DTMF "#" is sent to
terminate the entry to the ATA. This allows any length
number of any number to be dialed. 

I'm not sure if this behavior is specific to the
firmware in my Smart-1, but it works like this with
the route encoding mentioned above. Very convenient!
However, the user is still blocked from hearing any
sounds from the line side until the dialing is
completed and the call cuts through the Smart-1.

Don

> 
> --- Greg Blakely <greg at vyger.net> wrote:
> 
> > The default route in a SMART-1 is run by this
> > command:
> > 
> > #80327##
> > 
> > The first three digits translate to "wait for
> > precise dial tone", and
> > the rest of it amounts to "dial the number and cut
> > through."
> > 
> > If the precise dial tone has timed out, the
> SMART-1
> > does not detect it,
> > and **should** hang up and then re-seize the line,
> > detect the dial tone,
> > and then dial and cut through.
> > 
> > If this is not happening, it may be that the #803
> > part of the sequence
> > is missing...  
> > 
> > Something to check, anyway.
> > 
> > Greg.
> > 
> > PS.  I'm fairly good at programming SMART-1
> dialers,
> > since I installed
> > them for a 0+ provider for about three years, back
> > in the late 1980s and
> > early 1990s.  I'm not an expert, by any means, but
> I
> > can usually get
> > them running.
> > 
> > > -----Original Message-----
> > > From: voip-bounces at ckts.info
> > [mailto:voip-bounces at ckts.info] On Behalf
> > Of
> > > Donald Froula
> > > Sent: Thursday, November 02, 2006 1:11 PM
> > > To: voip at ckts.info
> > > Subject: [VoIP] Mitel Smart-1 as Pulse to Tone
> > Converter
> > > 
> > > I received my Smart-1 the other day and, after
> > some
> > > head scratching over the manual and odd
> > programming,
> > > got it working (sort of) as a dial pulse to tone
> > > converter in fromt of my ATA (a Linksys PAP2).
> > > 
> > > The problem is that the Smart-1 picks up the
> > outgoing
> > > phone line as soon as the phone goes off-hook.
> By
> > the
> > > time I rotary dial all the digits and the
> Smart-1
> > cuts
> > > through, the PAP2 ATA has already timed out to
> > > reorder.
> > > 
> > > I can adjust interdigit timers in the PAP2 to
> wait
> > > long enough after the first digit, but there is
> no
> > > timer I can find to control the off-hook to
> > reorder
> > > timing.
> > > 
> > > Ideally, I'd like the Smart-1 not to pick up and
> > dial
> > > the outgoing line until it detects a pattern
> match
> > on
> > > the incoming digits. I've been through the
> manual
> > a
> > > few times, but can't see how this would be done.
> > > 
> > > Any help is appreciated.
> > > 
> > > Regards,
> > > 
> > > Don
> > > 
> > > _______________________________________________
> > > VoIP mailing list
> > > VoIP at ckts.info
> > > http://lists.ckts.info/mailman/listinfo/voip
> > > Project Web Page: http://www.ckts.info/
> > > 
> > 
> > 
> > 
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> > 
> > 
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 



From stfkerman at jps.net  Tue Nov  7 14:35:17 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Tue, 07 Nov 2006 15:35:17 -0500
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter -
	Converts	digit-by-digit!
In-Reply-To: <20061107201902.82165.qmail@web83205.mail.mud.yahoo.com>
References: <20061107201902.82165.qmail@web83205.mail.mud.yahoo.com>
Message-ID: <4550EE05.3090000@jps.net>

Don,

That is very interesting.  While it is true that the existing hardware 
inherently splits the line, a hardware modification might be designed 
now that there is known to be firmware support for this on-the-fly mode. 

I assume that once timeout occurs and # is sent, it no longer recognizes 
DP and will not split the line if DP is received.

Steph

Donald Froula wrote:
> My impression was the Smart-1 was strictly a store and
> forward device that enforced one fixed route for each
> length string of dialed digits.
>
> However, I found a route combination that translates
> dial pulses to touch tones as they are dialed,
> variable length.
>
> I set the default route (0) to "#8032#007##"
>
> I then set search table 801 to "#9#8"
>
> The search table will mach any first rotary digit and
> go to route 0. Route 0 waits for a valid dialtone
> (retrying if not found) and sends the first digit,
> translated to DTMF. It will then wait for the
> inter-digit timeout period, send a DTMF #, and cut
> through. The interesting thing is that one can rotary
> dial any number of digits after the first and they
> will be translated to DTMF as they are dialed. After
> the inter-digit time pause, the DTMF "#" is sent to
> terminate the entry to the ATA. This allows any length
> number of any number to be dialed. 
>
> I'm not sure if this behavior is specific to the
> firmware in my Smart-1, but it works like this with
> the route encoding mentioned above. Very convenient!
> However, the user is still blocked from hearing any
> sounds from the line side until the dialing is
> completed and the call cuts through the Smart-1.
>
> Don
>
>   
>> --- Greg Blakely <greg at vyger.net> wrote:
>>
>>     
>>> The default route in a SMART-1 is run by this
>>> command:
>>>
>>> #80327##
>>>
>>> The first three digits translate to "wait for
>>> precise dial tone", and
>>> the rest of it amounts to "dial the number and cut
>>> through."
>>>
>>> If the precise dial tone has timed out, the
>>>       
>> SMART-1
>>     
>>> does not detect it,
>>> and **should** hang up and then re-seize the line,
>>> detect the dial tone,
>>> and then dial and cut through.
>>>
>>> If this is not happening, it may be that the #803
>>> part of the sequence
>>> is missing...  
>>>
>>> Something to check, anyway.
>>>
>>> Greg.
>>>
>>> PS.  I'm fairly good at programming SMART-1
>>>       
>> dialers,
>>     
>>> since I installed
>>> them for a 0+ provider for about three years, back
>>> in the late 1980s and
>>> early 1990s.  I'm not an expert, by any means, but
>>>       
>> I
>>     
>>> can usually get
>>> them running.
>>>
>>>       
>>>> -----Original Message-----
>>>> From: voip-bounces at ckts.info
>>>>         
>>> [mailto:voip-bounces at ckts.info] On Behalf
>>> Of
>>>       
>>>> Donald Froula
>>>> Sent: Thursday, November 02, 2006 1:11 PM
>>>> To: voip at ckts.info
>>>> Subject: [VoIP] Mitel Smart-1 as Pulse to Tone
>>>>         
>>> Converter
>>>       
>>>> I received my Smart-1 the other day and, after
>>>>         
>>> some
>>>       
>>>> head scratching over the manual and odd
>>>>         
>>> programming,
>>>       
>>>> got it working (sort of) as a dial pulse to tone
>>>> converter in fromt of my ATA (a Linksys PAP2).
>>>>
>>>> The problem is that the Smart-1 picks up the
>>>>         
>>> outgoing
>>>       
>>>> phone line as soon as the phone goes off-hook.
>>>>         
>> By
>>     
>>> the
>>>       
>>>> time I rotary dial all the digits and the
>>>>         
>> Smart-1
>>     
>>> cuts
>>>       
>>>> through, the PAP2 ATA has already timed out to
>>>> reorder.
>>>>
>>>> I can adjust interdigit timers in the PAP2 to
>>>>         
>> wait
>>     
>>>> long enough after the first digit, but there is
>>>>         
>> no
>>     
>>>> timer I can find to control the off-hook to
>>>>         
>>> reorder
>>>       
>>>> timing.
>>>>
>>>> Ideally, I'd like the Smart-1 not to pick up and
>>>>         
>>> dial
>>>       
>>>> the outgoing line until it detects a pattern
>>>>         
>> match
>>     
>>> on
>>>       
>>>> the incoming digits. I've been through the
>>>>         
>> manual
>>     
>>> a
>>>       
>>>> few times, but can't see how this would be done.
>>>>
>>>> Any help is appreciated.
>>>>
>>>> Regards,
>>>>
>>>> Don
>>>>
>>>> _______________________________________________
>>>> VoIP mailing list
>>>> VoIP at ckts.info
>>>> http://lists.ckts.info/mailman/listinfo/voip
>>>> Project Web Page: http://www.ckts.info/
>>>>
>>>>         
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>
>>>       
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>     
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>   

-- 
General request: Please *DO NOT* send messages with attachments larger than 300K total to this JPS email address without my prior agreement.  Send those messages to me at stfkerman at gmail.com.  I have a small mailbox at JPS. Large attachments will interfere with my email and cause other messages to bounce.  If you don't know how big the attachment is, please send it to stfkerman at gmail.com.  

Thanks.



From dfroula at sbcglobal.net  Tue Nov  7 14:41:54 2006
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 7 Nov 2006 12:41:54 -0800 (PST)
Subject: [VoIP] Mitel Smart-1 as Pulse to Tone Converter - Converts
	digit-by-digit!
In-Reply-To: <4550EE05.3090000@jps.net>
Message-ID: <20061107204154.65453.qmail@web83206.mail.mud.yahoo.com>

Yes, after the inter-digit timeout expires, the "#" is
sent, and it cuts through, the dialpulses get passed
through to the line with no further translation.

I have not experimented to see if the "#00" in the
route (which send the final "#") is needed to force
this behavior. It does seem that it is necessary to
have it wait for a valid dialtone in the route with
"#8XX" for it to work.

I verified this by listening on the line side as I
dialed through a butt set.

Don

--- Steph Kerman <stfkerman at jps.net> wrote:

> Don,
> 
> That is very interesting.  While it is true that the
> existing hardware 
> inherently splits the line, a hardware modification
> might be designed 
> now that there is known to be firmware support for
> this on-the-fly mode. 
> 
> I assume that once timeout occurs and # is sent, it
> no longer recognizes 
> DP and will not split the line if DP is received.
> 
> Steph
> 
> Donald Froula wrote:
> > My impression was the Smart-1 was strictly a store
> and
> > forward device that enforced one fixed route for
> each
> > length string of dialed digits.
> >
> > However, I found a route combination that
> translates
> > dial pulses to touch tones as they are dialed,
> > variable length.
> >
> > I set the default route (0) to "#8032#007##"
> >
> > I then set search table 801 to "#9#8"
> >
> > The search table will mach any first rotary digit
> and
> > go to route 0. Route 0 waits for a valid dialtone
> > (retrying if not found) and sends the first digit,
> > translated to DTMF. It will then wait for the
> > inter-digit timeout period, send a DTMF #, and cut
> > through. The interesting thing is that one can
> rotary
> > dial any number of digits after the first and they
> > will be translated to DTMF as they are dialed.
> After
> > the inter-digit time pause, the DTMF "#" is sent
> to
> > terminate the entry to the ATA. This allows any
> length
> > number of any number to be dialed. 
> >
> > I'm not sure if this behavior is specific to the
> > firmware in my Smart-1, but it works like this
> with
> > the route encoding mentioned above. Very
> convenient!
> > However, the user is still blocked from hearing
> any
> > sounds from the line side until the dialing is
> > completed and the call cuts through the Smart-1.
> >
> > Don
> >
> >   
> >> --- Greg Blakely <greg at vyger.net> wrote:
> >>
> >>     
> >>> The default route in a SMART-1 is run by this
> >>> command:
> >>>
> >>> #80327##
> >>>
> >>> The first three digits translate to "wait for
> >>> precise dial tone", and
> >>> the rest of it amounts to "dial the number and
> cut
> >>> through."
> >>>
> >>> If the precise dial tone has timed out, the
> >>>       
> >> SMART-1
> >>     
> >>> does not detect it,
> >>> and **should** hang up and then re-seize the
> line,
> >>> detect the dial tone,
> >>> and then dial and cut through.
> >>>
> >>> If this is not happening, it may be that the
> #803
> >>> part of the sequence
> >>> is missing...  
> >>>
> >>> Something to check, anyway.
> >>>
> >>> Greg.
> >>>
> >>> PS.  I'm fairly good at programming SMART-1
> >>>       
> >> dialers,
> >>     
> >>> since I installed
> >>> them for a 0+ provider for about three years,
> back
> >>> in the late 1980s and
> >>> early 1990s.  I'm not an expert, by any means,
> but
> >>>       
> >> I
> >>     
> >>> can usually get
> >>> them running.
> >>>
> >>>       
> >>>> -----Original Message-----
> >>>> From: voip-bounces at ckts.info
> >>>>         
> >>> [mailto:voip-bounces at ckts.info] On Behalf
> >>> Of
> >>>       
> >>>> Donald Froula
> >>>> Sent: Thursday, November 02, 2006 1:11 PM
> >>>> To: voip at ckts.info
> >>>> Subject: [VoIP] Mitel Smart-1 as Pulse to Tone
> >>>>         
> >>> Converter
> >>>       
> >>>> I received my Smart-1 the other day and, after
> >>>>         
> >>> some
> >>>       
> >>>> head scratching over the manual and odd
> >>>>         
> >>> programming,
> >>>       
> >>>> got it working (sort of) as a dial pulse to
> tone
> >>>> converter in fromt of my ATA (a Linksys PAP2).
> >>>>
> >>>> The problem is that the Smart-1 picks up the
> >>>>         
> >>> outgoing
> >>>       
> >>>> phone line as soon as the phone goes off-hook.
> >>>>         
> >> By
> >>     
> >>> the
> >>>       
> >>>> time I rotary dial all the digits and the
> >>>>         
> >> Smart-1
> >>     
> >>> cuts
> >>>       
> >>>> through, the PAP2 ATA has already timed out to
> >>>> reorder.
> >>>>
> >>>> I can adjust interdigit timers in the PAP2 to
> >>>>         
> >> wait
> >>     
> >>>> long enough after the first digit, but there is
> >>>>         
> >> no
> >>     
> >>>> timer I can find to control the off-hook to
> >>>>         
> >>> reorder
> >>>       
> >>>> timing.
> >>>>
> >>>> Ideally, I'd like the Smart-1 not to pick up
> and
> >>>>         
> >>> dial
> >>>       
> >>>> the outgoing line until it detects a pattern
> >>>>         
> >> match
> >>     
> >>> on
> >>>       
> >>>> the incoming digits. I've been through the
> >>>>         
> >> manual
> >>     
> >>> a
> >>>       
> >>>> few times, but can't see how this would be
> done.
> >>>>
> >>>> Any help is appreciated.
> >>>>
> >>>> Regards,
> >>>>
> >>>> Don
> >>>>
> >>>> _______________________________________________
> >>>> VoIP mailing list
> >>>> VoIP at ckts.info
> >>>> http://lists.ckts.info/mailman/listinfo/voip
> >>>> Project Web Page: http://www.ckts.info/
> >>>>
> 
=== message truncated ===



From jnovack at stromberg-carlson.org  Wed Nov  8 18:30:32 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 08 Nov 2006 19:30:32 -0500
Subject: [VoIP] Reserve an office code
Message-ID: <455276A8.50207@stromberg-carlson.org>

Someone asked me to reserve an office code, but that site seems to be down
Anyone else have that problem??

John Novack


From wepbx at sbcglobal.net  Wed Nov  8 19:14:28 2006
From: wepbx at sbcglobal.net (Richard Walsh)
Date: Wed, 8 Nov 2006 17:14:28 -0800 (PST)
Subject: [VoIP] Reserve an office code
In-Reply-To: <455276A8.50207@stromberg-carlson.org>
Message-ID: <20061109011428.31634.qmail@web82010.mail.mud.yahoo.com>

Yup, Can't get to the CNET Directory of Member Listings either.
  Rick Walsh

John Novack <jnovack at stromberg-carlson.org> wrote:
  Someone asked me to reserve an office code, but that site seems to be down
Anyone else have that problem??

John Novack

_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/



From ka2wft at arrl.net  Wed Nov  8 19:48:00 2006
From: ka2wft at arrl.net (Doug Alderdice)
Date: Wed, 08 Nov 2006 20:48:00 -0500
Subject: [VoIP] Reserve an office code
In-Reply-To: <20061109011428.31634.qmail@web82010.mail.mud.yahoo.com>
References: <455276A8.50207@stromberg-carlson.org>
Message-ID: <5.1.0.14.0.20061108204624.02a925e8@incoming.verizon.net>

At 05:14 PM 11/8/2006 -0800, Rick Walsh wrote:
>Yup, Can't get to the CNET Directory of Member Listings either.
>   Rick Walsh
>
>John Novack <jnovack at stromberg-carlson.org> wrote:
>   Someone asked me to reserve an office code, but that site seems to be down
>Anyone else have that problem??


I was just in the member directory looking up #s to test dial just now 
(0146 GMT, 2046 EST) with no problems.  Even did a "refresh" just to be sure.

Doug.




From david at josephson.com  Thu Nov  9 02:20:59 2006
From: david at josephson.com (David Josephson)
Date: Thu, 09 Nov 2006 00:20:59 -0800
Subject: [VoIP] Reserve an office code
In-Reply-To: <455276A8.50207@stromberg-carlson.org>
References: <455276A8.50207@stromberg-carlson.org>
Message-ID: <4552E4EB.5010504@josephson.com>

Argh, why is it that servers fail only when you are thousands of miles 
away? The machine hosting the office code wiki is down, due to a 
filesystem failure. This is causing all sorts of havoc as one might 
imagine. I am at a small airplane factory in Germany doing some 
acoustical engineering work and will be back in a week, on the 16th. 
It's an ordinary Linux box in my office but there are no Linux gurus at 
the shop, and I think it would be dicey to try to run fsck over the 
phone. Back online in about a week ...

David Josephson
> Someone asked me to reserve an office code, but that site seems to be down
> Anyone else have that problem??
>
> John Novack
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>   



From hockd at dteenergy.com  Thu Nov  9 03:50:33 2006
From: hockd at dteenergy.com (Dennis D Hock)
Date: Thu, 9 Nov 2006 04:50:33 -0500
Subject: [VoIP] D Link Cable Router Info
Message-ID: <OF7A38627B.47E2CE0B-ON85257221.00361116-85257221.0036111C@dteenergy.com>



John,

I have the info on our D-Link router.  It is:

    D-Link  DI 604, v:3.20, Jul 2003

Also I was able to access the CNET docs this morning.  13 more people weree
tapped on the shoulder and escorted off the property yesterday.  Radio says
we are looking to raise 1 Billion dollars by selling off deregs and gas
fields as well as soime generation.  Everyone figures something is way up.
Stock is up to 46 strange going ons.

Take care,

Dennis

From jnovack at stromberg-carlson.org  Thu Nov  9 08:33:06 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Thu, 09 Nov 2006 09:33:06 -0500
Subject: [VoIP] Reserve an office code
In-Reply-To: <4552E4EB.5010504@josephson.com>
References: <455276A8.50207@stromberg-carlson.org>
	<4552E4EB.5010504@josephson.com>
Message-ID: <45533C22.50502@stromberg-carlson.org>



David Josephson wrote:
> Argh, why is it that servers fail only when you are thousands of miles 
> away? 
They know! The machines are alive and ready to take over the world!
> The machine hosting the office code wiki is down, due to a filesystem 
> failure. This is causing all sorts of havoc as one might imagine. I am 
> at a small airplane factory in Germany doing some acoustical 
> engineering work and will be back in a week, on the 16th. It's an 
> ordinary Linux box in my office but there are no Linux gurus at the 
> shop, and I think it would be dicey to try to run fsck over the phone. 
> Back online in about a week ...
>
> David Josephson
I use a hosting company, have plenty of space and can probably provide 
everything needed, except the brainpower, to host this.
With the low cost of hosting these days, that seems a smarter way to go 
than some machine off in the corner of an office with no one to feed and 
care for it.

Enjoy your time in Germany

John Novack

>> Someone asked me to reserve an office code, but that site seems to be 
>> down
>> Anyone else have that problem??
>>
>> John Novack
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>   
>
>
>


From chad at maine.maine.edu  Fri Nov 10 20:58:47 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Fri, 10 Nov 2006 21:58:47 -0500
Subject: [VoIP] SIP/firewall problem - the saga
Message-ID: <4554F617.15903.CE6253@localhost>

I hope someone can shed some light on a problem I'm having.  I recently deployed a 
couple two line SIP ATAs (Sipura SPA-2002 & Linksys PAP2).  They are enroute to a 
couple Telephone Museum members (to whom I have assigned individual CNET 
numbers off the tandem here on Line 2 of the ATA).  The ATAs went on the road with 
me Monday and installed at a test site across town.  They tested perfectly.  Home 
free.  Wrong.

The test site (my work) has three broadband drops with three different firewall 
routers.  After testing on one I relocated to my workbench which has it's own drop 
verified basic connectivity then I had to leave.  After getting home Monday I find 
errors on the Asterisk console.

NOTICE[98310]: chan_sip.c:7641 handle_request: Registration from  
'<sip:ATA2L2 at agstmexx700.homeip.net>' failed for '76.179.29.137'

some time later (much longer than 30 seconds though, however Register_Expires: is 
set to 30 seconds in the ATAs):
    -- Registered SIP 'ATA2L2' at 76.179.29.137 port 5061 expires 30

I have confirmed that registration is failing on the CNET lines from time to time (as far 
as Asterisk is concerned).  
AGSTMESEPS0*CLI> sip show peers
ATA3L2                  (Unspecified)    D   N      255.255.255.255  0        Unmonitored
ATA2L2/ATA2L2    (Unspecified)    D   N      255.255.255.255  0        Unmonitored
ATA1L2/ATA1L2    198.182.163.2    D   N      255.255.255.255  32845    Unmonitored

Line 1 is subscribed to Stanaphone on one and BroadVoice on the other; they 
appear to be fine.  I didn't know but having two units connecting back to the same ip 
(for Asterisk) on the same port (5060) might be causing a conflict on the 
nat/router/firewall; so I moved one ATA to the third drop Tuesday (the first drop is not 
mine to play with). 

Tuesday night I get home and find the errors continue.  I am starting to wonder if I 
am having port 5060 conflicts between the Line 1 and Line 2, so I set the port to 
5061 in sip.conf and change Line 2 (back) to 5061 in the ATAs Wednesday.  Sip 
show peers as of Wednesday night follows.

AGSTMESEPS0*CLI> sip show peers
ATA3L2                    (Unspecified)    D   N      255.255.255.255  0        Unmonitored
ATA2L2/ATA2L2    76.179.29.137    D   N      255.255.255.255  5061     Unmonitored
ATA1L2/ATA1L2    198.182.163.2    D   N      255.255.255.255  32845    Unmonitored
[snip]

Problem continues.  Thursday I discover things are broken in the audio path and 
calls are NOT connecting properly (even when registered)!  I continue to think about 
NAT, etc. so I enable STUN.  No dice; this had worked for my Grandstream a year or 
so ago.

Today I routed one of the ATAs through a test ethernet switch in the lab that has 6 
LEDs per port so I could see what was going on a little better.  What I found out is 
that the audio path is one way (transmitting); it confirms that I hear nothing because 
there is nothing in the way of RTP making it to the ATA.

Okay so I'm starting to loose my mind.  I break down the test network and recable 
the ATA via the test switch to the Linksys router on Broadband 1.  Presto bingo, 
switch lights up and I have two-way audio!  I can't leave it there so I don't know 
whether the registration problem returns.

So I know this is a Firewall/NAT problem of sorts.  I am little puzzled as to why I have 
this problem and how to fix it; the VoIP provider on Line 1 is always fine.  One 
obvious difference is I am also NATed; they are not. I have UDP 5060-5063 and 
10000-20000 port forwarded to Asterisk, but that doesn't totally eliminate the effects 
of NAT on SIP.  I am confused.

I am at a loss why is works on the Linksys but not on the Netgear (or the 
Smoothwall).  I am really not looking forward to SIP debug and packet captures, 
though I am equipped.  This end is Asterisk 1.0 via standard 3Mb Verizon ADSL and 
the Westell VersaLink 327W firewall/router/four port switch/wireless access point.

Chad
+1 955-9924 
(US EST)


[ATA3L2]
type=friend
secret=PAP2           
callerid="Unassigned - L2" <  17007272>
host=dynamic
port=5061                       ; Line 2 port 11-9-2006
nat=yes                         ; behind a NAT router, 11-7-2006
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
context=cnet
outgoinglimit=1
;incominglimit=1
mailbox=7007272





From greg at vyger.net  Fri Nov 10 21:20:48 2006
From: greg at vyger.net (Greg Blakely)
Date: Fri, 10 Nov 2006 21:20:48 -0600
Subject: [VoIP] SIP/firewall problem - the saga
Message-ID: <A691C3685A152E4AAC9608B8D8E113163EB0A9@godzilla.hickory.vyger.net>

I hate SIP.

I don't know if this will help, but, when I do a "sip show peers," I get
some really odd ports for Dennis and for Jim:

voipgw*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status

10/10                      (Unspecified)    D   N      0
Unmonitored
952.949.6767/952.949.6767  (Unspecified)    D   N      0
Unmonitored
2697/2697                  (Unspecified)    D          0        UNKNOWN

gblakely2/gblakely2        172.26.0.3       D   N      5060     OK (56
ms)
gblakely/gblakely          172.26.0.3       D   N      5060     OK (57
ms)
jday/jday                  69.37.44.178     D   N      60010    OK (109
ms)
dhock/dhock                68.61.110.28     D   N      61152    OK (75
ms)
iconnect/12345678          213.137.73.140       N      5060
Unmonitored
guest                      (Unspecified)        N      5060
Unmonitored
9 sip peers [8 online , 1 offline]

The first five entries above are for my own soft phones and for a Cisco
7940 telephone.

You can see that Jim is using port 60010, and Dennis' old connection
(which is still live) uses port 61152.

I'm not sure whether those odd numbers are on their end or on mine, but
(knock on wood) they appear to be working.  

My asterisk box is NATted behind an IP-COP linux firewall.  I have port
5060 forwarded to my asterisk box, but the RTP ports are just opened up
-- not forwarded anywhere.

And I'm starting to lose my mind, too.  But that's a different story.

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
Of
> Chad Perkins
> Sent: Friday, November 10, 2006 8:59 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: [VoIP] SIP/firewall problem - the saga
> 
> I hope someone can shed some light on a problem I'm having.  I
recently
> deployed a
> couple two line SIP ATAs (Sipura SPA-2002 & Linksys PAP2).  They are
> enroute to a
> couple Telephone Museum members (to whom I have assigned individual
CNET
> numbers off the tandem here on Line 2 of the ATA).  The ATAs went on
the
> road with
> me Monday and installed at a test site across town.  They tested
> perfectly.  Home
> free.  Wrong.
> 
> The test site (my work) has three broadband drops with three different
> firewall
> routers.  After testing on one I relocated to my workbench which has
it's
> own drop
> verified basic connectivity then I had to leave.  After getting home
> Monday I find
> errors on the Asterisk console.
> 
> NOTICE[98310]: chan_sip.c:7641 handle_request: Registration from
> '<sip:ATA2L2 at agstmexx700.homeip.net>' failed for '76.179.29.137'
> 
> some time later (much longer than 30 seconds though, however
> Register_Expires: is
> set to 30 seconds in the ATAs):
>     -- Registered SIP 'ATA2L2' at 76.179.29.137 port 5061 expires 30
> 
> I have confirmed that registration is failing on the CNET lines from
time
> to time (as far
> as Asterisk is concerned).
> AGSTMESEPS0*CLI> sip show peers
> ATA3L2                  (Unspecified)    D   N      255.255.255.255  0
> Unmonitored
> ATA2L2/ATA2L2    (Unspecified)    D   N      255.255.255.255  0
> Unmonitored
> ATA1L2/ATA1L2    198.182.163.2    D   N      255.255.255.255  32845
> Unmonitored
> 
> Line 1 is subscribed to Stanaphone on one and BroadVoice on the other;
> they
> appear to be fine.  I didn't know but having two units connecting back
to
> the same ip
> (for Asterisk) on the same port (5060) might be causing a conflict on
the
> nat/router/firewall; so I moved one ATA to the third drop Tuesday (the
> first drop is not
> mine to play with).
> 
> Tuesday night I get home and find the errors continue.  I am starting
to
> wonder if I
> am having port 5060 conflicts between the Line 1 and Line 2, so I set
the
> port to
> 5061 in sip.conf and change Line 2 (back) to 5061 in the ATAs
Wednesday.
> Sip
> show peers as of Wednesday night follows.
> 
> AGSTMESEPS0*CLI> sip show peers
> ATA3L2                    (Unspecified)    D   N      255.255.255.255
0
> Unmonitored
> ATA2L2/ATA2L2    76.179.29.137    D   N      255.255.255.255  5061
> Unmonitored
> ATA1L2/ATA1L2    198.182.163.2    D   N      255.255.255.255  32845
> Unmonitored
> [snip]
> 
> Problem continues.  Thursday I discover things are broken in the audio
> path and
> calls are NOT connecting properly (even when registered)!  I continue
to
> think about
> NAT, etc. so I enable STUN.  No dice; this had worked for my
Grandstream a
> year or
> so ago.
> 
> Today I routed one of the ATAs through a test ethernet switch in the
lab
> that has 6
> LEDs per port so I could see what was going on a little better.  What
I
> found out is
> that the audio path is one way (transmitting); it confirms that I hear
> nothing because
> there is nothing in the way of RTP making it to the ATA.
> 
> Okay so I'm starting to loose my mind.  I break down the test network
and
> recable
> the ATA via the test switch to the Linksys router on Broadband 1.
Presto
> bingo,
> switch lights up and I have two-way audio!  I can't leave it there so
I
> don't know
> whether the registration problem returns.
> 
> So I know this is a Firewall/NAT problem of sorts.  I am little
puzzled as
> to why I have
> this problem and how to fix it; the VoIP provider on Line 1 is always
> fine.  One
> obvious difference is I am also NATed; they are not. I have UDP
5060-5063
> and
> 10000-20000 port forwarded to Asterisk, but that doesn't totally
eliminate
> the effects
> of NAT on SIP.  I am confused.
> 
> I am at a loss why is works on the Linksys but not on the Netgear (or
the
> Smoothwall).  I am really not looking forward to SIP debug and packet
> captures,
> though I am equipped.  This end is Asterisk 1.0 via standard 3Mb
Verizon
> ADSL and
> the Westell VersaLink 327W firewall/router/four port switch/wireless
> access point.
> 
> Chad
> +1 955-9924
> (US EST)
> 
> 
> [ATA3L2]
> type=friend
> secret=PAP2
> callerid="Unassigned - L2" <  17007272>
> host=dynamic
> port=5061                       ; Line 2 port 11-9-2006
> nat=yes                         ; behind a NAT router, 11-7-2006
> canreinvite=no
> disallow=all
> allow=alaw
> allow=ulaw
> context=cnet
> outgoinglimit=1
> ;incominglimit=1
> mailbox=7007272
> 
> 
> 



From chad at maine.maine.edu  Sat Nov 11 11:33:58 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Sat, 11 Nov 2006 12:33:58 -0500
Subject: [VoIP] SIP/firewall problem - the saga
In-Reply-To: <A691C3685A152E4AAC9608B8D8E113163EB0A9@godzilla.hickory.vyger.net>
Message-ID: <4555C336.19514.A7E6B6@localhost>

> I hate SIP.
> I don't know if this will help, but, when I do a "sip show peers," I
> get some really odd ports for Dennis and for Jim:
> 
> voipgw*CLI> sip show peers
> Name/username              Host            Dyn Nat ACL Port     Status
> 10/10                                      (Unspecified)    D   N      0 Unmonitored
> 952.949.6767/952.949.6767  (Unspecified)    D   N      0 Unmonitored
> 2697/2697                              (Unspecified)    D          0       UNKNOWN
> gblakely2/gblakely2        172.26.0.3       D   N      5060     OK (56 ms) 
> gblakely/gblakely          172.26.0.3       D   N      5060     OK (57 ms) 
> jday/jday                  69.37.44.178     D   N      60010    OK (109 ms) 
> dhock/dhock                68.61.110.28     D   N     61152    OK (75 ms) 
> iconnect/12345678          213.137.73.140       N     5060 Unmonitored 
> guest                      (Unspecified)        N      5060 Unmonitored
> 9 sip peers [8 online , 1 offline]
> 
> The first five entries above are for my own soft phones and for a
> Cisco 7940 telephone.
> You can see that Jim is using port 60010, and Dennis' old connection
> (which is still live) uses port 61152.
> I'm not sure whether those odd numbers are on their end or on mine,
> but (knock on wood) they appear to be working.  

>From what I can tell, those port numbers are the NA(P)T port numbers assigned to 
the connection by their router/firewall on egress during "registration"; so that is the 
port number Asterisk sees them coming FROM (in other words their src port - dst 
port should still be 5060).  

69.37.44.178:60010 ---> 209.98.47.194:5060

This behavior varies by vendor/model as there are 4 "types" of NAT.  High/odd port 
number *do* signify NAT, however the absence of high port number does *not* 
signify the absence of NAT (as some types of NAT don't play that game).  

This totally raises hell with SIP, H.323, RTP, etc.  This is where the STUN protocol 
comes in handy as STUN can figure out if, and which type of, NAT is present and 
pass that information to a UDP application (i.e. SIP/RTP).
 
> My asterisk box is NATted behind an IP-COP linux firewall.  I have
> port 5060 forwarded to my asterisk box, but the RTP ports are just
> opened up -- not forwarded anywhere.

I'm confused as to how the RTP knows where to go if not forwarded.  Is Asterisk on 
an "Orange" DMZ port?  Atleast IP-Cop should have some logs that help with this 
sort of thing.  Too many COTS firewall/routers have zipo.  :(

> And I'm starting to lose my mind, too.  But that's a different story.
Ah as one gets older there so many good reasons! 
> > -----Original Message-----
> > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On
> > Behalf
> Of
> > Chad Perkins
> > Sent: Friday, November 10, 2006 8:59 PM
> > To: Voice Over IP Tandem for Analog Switches
> > Subject: [VoIP] SIP/firewall problem - the saga
> > 
> > I hope someone can shed some light on a problem I'm having.  I
> recently
> > deployed a
> > couple two line SIP ATAs (Sipura SPA-2002 & Linksys PAP2).  They are
> > enroute to a couple Telephone Museum members (to whom I have
> > assigned individual
> CNET
> > numbers off the tandem here on Line 2 of the ATA).  The ATAs went on
> the
> > road with
> > me Monday and installed at a test site across town.  They tested
> > perfectly.  Home free.  Wrong.
> > 
> > The test site (my work) has three broadband drops with three
> > different firewall routers.  After testing on one I relocated to my
> > workbench which has
> it's
> > own drop
> > verified basic connectivity then I had to leave.  After getting home
> > Monday I find errors on the Asterisk console.
> > 
> > NOTICE[98310]: chan_sip.c:7641 handle_request: Registration from
> > '<sip:ATA2L2 at agstmexx700.homeip.net>' failed for '76.179.29.137'
> > 
> > some time later (much longer than 30 seconds though, however
> > Register_Expires: is
> > set to 30 seconds in the ATAs):
> >     -- Registered SIP 'ATA2L2' at 76.179.29.137 port 5061 expires 30
> > 
> > I have confirmed that registration is failing on the CNET lines from
> time
> > to time (as far
> > as Asterisk is concerned).
> > AGSTMESEPS0*CLI> sip show peers
> > ATA3L2                  (Unspecified)    D   N      255.255.255.255 
> > 0 Unmonitored ATA2L2/ATA2L2    (Unspecified)    D   N     
> > 255.255.255.255  0 Unmonitored ATA1L2/ATA1L2    198.182.163.2    D  
> > N      255.255.255.255  32845 Unmonitored
> > 
> > Line 1 is subscribed to Stanaphone on one and BroadVoice on the
> > other; they appear to be fine.  I didn't know but having two units
> > connecting back
> to
> > the same ip
> > (for Asterisk) on the same port (5060) might be causing a conflict
> > on
> the
> > nat/router/firewall; so I moved one ATA to the third drop Tuesday
> > (the first drop is not mine to play with).
> > 
> > Tuesday night I get home and find the errors continue.  I am
> > starting
> to
> > wonder if I
> > am having port 5060 conflicts between the Line 1 and Line 2, so I
> > set
> the
> > port to
> > 5061 in sip.conf and change Line 2 (back) to 5061 in the ATAs
> Wednesday.
> > Sip
> > show peers as of Wednesday night follows.
> > 
> > AGSTMESEPS0*CLI> sip show peers
> > ATA3L2                    (Unspecified)    D   N     
> > 255.255.255.255
> 0
> > Unmonitored
> > ATA2L2/ATA2L2    76.179.29.137    D   N      255.255.255.255  5061
> > Unmonitored ATA1L2/ATA1L2    198.182.163.2    D   N     
> > 255.255.255.255  32845 Unmonitored [snip]
> > 
> > Problem continues.  Thursday I discover things are broken in the
> > audio path and calls are NOT connecting properly (even when
> > registered)!  I continue
> to
> > think about
> > NAT, etc. so I enable STUN.  No dice; this had worked for my
> Grandstream a
> > year or
> > so ago.
> > 
> > Today I routed one of the ATAs through a test ethernet switch in the
> lab
> > that has 6
> > LEDs per port so I could see what was going on a little better. 
> > What
> I
> > found out is
> > that the audio path is one way (transmitting); it confirms that I
> > hear nothing because there is nothing in the way of RTP making it to
> > the ATA.
> > 
> > Okay so I'm starting to loose my mind.  I break down the test
> > network
> and
> > recable
> > the ATA via the test switch to the Linksys router on Broadband 1.
> Presto
> > bingo,
> > switch lights up and I have two-way audio!  I can't leave it there
> > so
> I
> > don't know
> > whether the registration problem returns.
> > 
> > So I know this is a Firewall/NAT problem of sorts.  I am little
> puzzled as
> > to why I have
> > this problem and how to fix it; the VoIP provider on Line 1 is
> > always fine.  One obvious difference is I am also NATed; they are
> > not. I have UDP
> 5060-5063
> > and
> > 10000-20000 port forwarded to Asterisk, but that doesn't totally
> eliminate
> > the effects
> > of NAT on SIP.  I am confused.
> > 
> > I am at a loss why is works on the Linksys but not on the Netgear
> > (or
> the
> > Smoothwall).  I am really not looking forward to SIP debug and
> > packet captures, though I am equipped.  This end is Asterisk 1.0 via
> > standard 3Mb
> Verizon
> > ADSL and
> > the Westell VersaLink 327W firewall/router/four port switch/wireless
> > access point.
> > 
> > Chad
> > +1 955-9924
> > (US EST)
> > 
> > 
> > [ATA3L2]
> > type=friend
> > secret=PAP2
> > callerid="Unassigned - L2" <  17007272>
> > host=dynamic
> > port=5061                       ; Line 2 port 11-9-2006
> > nat=yes                         ; behind a NAT router, 11-7-2006
> > canreinvite=no disallow=all allow=alaw allow=ulaw context=cnet
> > outgoinglimit=1 ;incominglimit=1 mailbox=7007272
> 



From greg at vyger.net  Sat Nov 11 11:47:52 2006
From: greg at vyger.net (Greg Blakely)
Date: Sat, 11 Nov 2006 11:47:52 -0600
Subject: [VoIP] SIP/firewall problem - the saga
Message-ID: <A691C3685A152E4AAC9608B8D8E113163EB0AB@godzilla.hickory.vyger.net>

> 
> > My asterisk box is NATted behind an IP-COP linux firewall.  I have
> > port 5060 forwarded to my asterisk box, but the RTP ports are just
> > opened up -- not forwarded anywhere.
> 
> I'm confused as to how the RTP knows where to go if not forwarded.  Is
> Asterisk on
> an "Orange" DMZ port?  Atleast IP-Cop should have some logs that help
with
> this
> sort of thing.  Too many COTS firewall/routers have zipo.  :(
> 
Yes, it is.  My desktop PCs are in the Green Zone, and my servers are in
the DMZ (orange interface).

Biggest thing, though, is that I double checked my port forwarding, and
see that I have ports 10000-20000 UDP forwarded to the Asterisk box.
So, I stand corrected.  

And this will show a certain amount of network ignorance on my part,
but, is RTP UDP? Or is it its own animal?




From jnovack at stromberg-carlson.org  Sat Nov 11 12:09:09 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sat, 11 Nov 2006 13:09:09 -0500
Subject: [VoIP] SIP/firewall problem - the saga (continues)
In-Reply-To: <A691C3685A152E4AAC9608B8D8E113163EB0AB@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E113163EB0AB@godzilla.hickory.vyger.net>
Message-ID: <455611C5.2050606@stromberg-carlson.org>


> Yes, it is.  My desktop PCs are in the Green Zone, and my servers are in
> the DMZ (orange interface).
>
> Biggest thing, though, is that I double checked my port forwarding, and
> see that I have ports 10000-20000 UDP forwarded to the Asterisk box.
> So, I stand corrected.  
I had a problem appear with my Stanaphone SIP, then discovered that I 
had put my Asterisk box into the DMZ AND did port forwarding, and it 
didn't seem to like that at all
Once I left port forwarding for SIP and IAX alone, and removed the 
Asterisk box from the DMZ, Stanaphone started to work properly once again.
Since I mostly blunder into and out of all of this stuff, and really 
don't have much of a clue, I can't say why it worked. Or why some time 
before I had enabled the DMZ
This is with a cheep Linksys 8 port router. I had problems earlier with 
their 4 port version and SIP, which would die after 18 minutes. I was 
not aware that Richard Nixon did router software.
Don't mean to add to the confusion. Just another SIP horror story
Unfortunately there are few IAX devices available for use where a full 
blown Asterisk box isn't warranted.
Has anyone been brave enough yet to attempt a minimal Asterisk 
installation into one of the Linksys routers. I have read about this 
being done with the proper router version, but as with most things 
Linux, the "how to do it for Dummies" isn't available.

John Novack



From ratguy at bellsouth.net  Sat Nov 11 12:33:05 2006
From: ratguy at bellsouth.net (Jayson Smith)
Date: Sat, 11 Nov 2006 13:33:05 -0500
Subject: [VoIP] SIP/firewall problem - the saga
References: <4555C336.19514.A7E6B6@localhost>
Message-ID: <000801c705bf$d3d51cc0$0600a8c0@bluegrasspals.com>

Hi,
HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
accepting IAX2 connections on a random high port rather than 4569 (is that
the right port or am I losing my mind too?).  Yeah, some NAT systems do
really strange things!
Jayson.

----- Original Message ----- 
From: "Chad Perkins" <chad at maine.maine.edu>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Saturday, November 11, 2006 12:33 PM
Subject: Re: [VoIP] SIP/firewall problem - the saga


> > I hate SIP.
> > I don't know if this will help, but, when I do a "sip show peers," I
> > get some really odd ports for Dennis and for Jim:
> >
> > voipgw*CLI> sip show peers
> > Name/username              Host            Dyn Nat ACL Port     Status
> > 10/10                                      (Unspecified)    D   N      0
Unmonitored
> > 952.949.6767/952.949.6767  (Unspecified)    D   N      0 Unmonitored
> > 2697/2697                              (Unspecified)    D          0
UNKNOWN
> > gblakely2/gblakely2        172.26.0.3       D   N      5060     OK (56
ms)
> > gblakely/gblakely          172.26.0.3       D   N      5060     OK (57
ms)
> > jday/jday                  69.37.44.178     D   N      60010    OK (109
ms)
> > dhock/dhock                68.61.110.28     D   N     61152    OK (75
ms)
> > iconnect/12345678          213.137.73.140       N     5060 Unmonitored
> > guest                      (Unspecified)        N      5060 Unmonitored
> > 9 sip peers [8 online , 1 offline]
> >
> > The first five entries above are for my own soft phones and for a
> > Cisco 7940 telephone.
> > You can see that Jim is using port 60010, and Dennis' old connection
> > (which is still live) uses port 61152.
> > I'm not sure whether those odd numbers are on their end or on mine,
> > but (knock on wood) they appear to be working.
>
> >From what I can tell, those port numbers are the NA(P)T port numbers
assigned to
> the connection by their router/firewall on egress during "registration";
so that is the
> port number Asterisk sees them coming FROM (in other words their src
port - dst
> port should still be 5060).
>
> 69.37.44.178:60010 ---> 209.98.47.194:5060
>
> This behavior varies by vendor/model as there are 4 "types" of NAT.
High/odd port
> number *do* signify NAT, however the absence of high port number does
*not*
> signify the absence of NAT (as some types of NAT don't play that game).
>
> This totally raises hell with SIP, H.323, RTP, etc.  This is where the
STUN protocol
> comes in handy as STUN can figure out if, and which type of, NAT is
present and
> pass that information to a UDP application (i.e. SIP/RTP).
>
> > My asterisk box is NATted behind an IP-COP linux firewall.  I have
> > port 5060 forwarded to my asterisk box, but the RTP ports are just
> > opened up -- not forwarded anywhere.
>
> I'm confused as to how the RTP knows where to go if not forwarded.  Is
Asterisk on
> an "Orange" DMZ port?  Atleast IP-Cop should have some logs that help with
this
> sort of thing.  Too many COTS firewall/routers have zipo.  :(
>
> > And I'm starting to lose my mind, too.  But that's a different story.
> Ah as one gets older there so many good reasons!
> > > -----Original Message-----
> > > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On
> > > Behalf
> > Of
> > > Chad Perkins
> > > Sent: Friday, November 10, 2006 8:59 PM
> > > To: Voice Over IP Tandem for Analog Switches
> > > Subject: [VoIP] SIP/firewall problem - the saga
> > >
> > > I hope someone can shed some light on a problem I'm having.  I
> > recently
> > > deployed a
> > > couple two line SIP ATAs (Sipura SPA-2002 & Linksys PAP2).  They are
> > > enroute to a couple Telephone Museum members (to whom I have
> > > assigned individual
> > CNET
> > > numbers off the tandem here on Line 2 of the ATA).  The ATAs went on
> > the
> > > road with
> > > me Monday and installed at a test site across town.  They tested
> > > perfectly.  Home free.  Wrong.
> > >
> > > The test site (my work) has three broadband drops with three
> > > different firewall routers.  After testing on one I relocated to my
> > > workbench which has
> > it's
> > > own drop
> > > verified basic connectivity then I had to leave.  After getting home
> > > Monday I find errors on the Asterisk console.
> > >
> > > NOTICE[98310]: chan_sip.c:7641 handle_request: Registration from
> > > '<sip:ATA2L2 at agstmexx700.homeip.net>' failed for '76.179.29.137'
> > >
> > > some time later (much longer than 30 seconds though, however
> > > Register_Expires: is
> > > set to 30 seconds in the ATAs):
> > >     -- Registered SIP 'ATA2L2' at 76.179.29.137 port 5061 expires 30
> > >
> > > I have confirmed that registration is failing on the CNET lines from
> > time
> > > to time (as far
> > > as Asterisk is concerned).
> > > AGSTMESEPS0*CLI> sip show peers
> > > ATA3L2                  (Unspecified)    D   N      255.255.255.255
> > > 0 Unmonitored ATA2L2/ATA2L2    (Unspecified)    D   N
> > > 255.255.255.255  0 Unmonitored ATA1L2/ATA1L2    198.182.163.2    D
> > > N      255.255.255.255  32845 Unmonitored
> > >
> > > Line 1 is subscribed to Stanaphone on one and BroadVoice on the
> > > other; they appear to be fine.  I didn't know but having two units
> > > connecting back
> > to
> > > the same ip
> > > (for Asterisk) on the same port (5060) might be causing a conflict
> > > on
> > the
> > > nat/router/firewall; so I moved one ATA to the third drop Tuesday
> > > (the first drop is not mine to play with).
> > >
> > > Tuesday night I get home and find the errors continue.  I am
> > > starting
> > to
> > > wonder if I
> > > am having port 5060 conflicts between the Line 1 and Line 2, so I
> > > set
> > the
> > > port to
> > > 5061 in sip.conf and change Line 2 (back) to 5061 in the ATAs
> > Wednesday.
> > > Sip
> > > show peers as of Wednesday night follows.
> > >
> > > AGSTMESEPS0*CLI> sip show peers
> > > ATA3L2                    (Unspecified)    D   N
> > > 255.255.255.255
> > 0
> > > Unmonitored
> > > ATA2L2/ATA2L2    76.179.29.137    D   N      255.255.255.255  5061
> > > Unmonitored ATA1L2/ATA1L2    198.182.163.2    D   N
> > > 255.255.255.255  32845 Unmonitored [snip]
> > >
> > > Problem continues.  Thursday I discover things are broken in the
> > > audio path and calls are NOT connecting properly (even when
> > > registered)!  I continue
> > to
> > > think about
> > > NAT, etc. so I enable STUN.  No dice; this had worked for my
> > Grandstream a
> > > year or
> > > so ago.
> > >
> > > Today I routed one of the ATAs through a test ethernet switch in the
> > lab
> > > that has 6
> > > LEDs per port so I could see what was going on a little better.
> > > What
> > I
> > > found out is
> > > that the audio path is one way (transmitting); it confirms that I
> > > hear nothing because there is nothing in the way of RTP making it to
> > > the ATA.
> > >
> > > Okay so I'm starting to loose my mind.  I break down the test
> > > network
> > and
> > > recable
> > > the ATA via the test switch to the Linksys router on Broadband 1.
> > Presto
> > > bingo,
> > > switch lights up and I have two-way audio!  I can't leave it there
> > > so
> > I
> > > don't know
> > > whether the registration problem returns.
> > >
> > > So I know this is a Firewall/NAT problem of sorts.  I am little
> > puzzled as
> > > to why I have
> > > this problem and how to fix it; the VoIP provider on Line 1 is
> > > always fine.  One obvious difference is I am also NATed; they are
> > > not. I have UDP
> > 5060-5063
> > > and
> > > 10000-20000 port forwarded to Asterisk, but that doesn't totally
> > eliminate
> > > the effects
> > > of NAT on SIP.  I am confused.
> > >
> > > I am at a loss why is works on the Linksys but not on the Netgear
> > > (or
> > the
> > > Smoothwall).  I am really not looking forward to SIP debug and
> > > packet captures, though I am equipped.  This end is Asterisk 1.0 via
> > > standard 3Mb
> > Verizon
> > > ADSL and
> > > the Westell VersaLink 327W firewall/router/four port switch/wireless
> > > access point.
> > >
> > > Chad
> > > +1 955-9924
> > > (US EST)
> > >
> > >
> > > [ATA3L2]
> > > type=friend
> > > secret=PAP2
> > > callerid="Unassigned - L2" <  17007272>
> > > host=dynamic
> > > port=5061                       ; Line 2 port 11-9-2006
> > > nat=yes                         ; behind a NAT router, 11-7-2006
> > > canreinvite=no disallow=all allow=alaw allow=ulaw context=cnet
> > > outgoinglimit=1 ;incominglimit=1 mailbox=7007272
> >
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>



From ian at uax.org.uk  Sun Nov 12 14:11:16 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Sun, 12 Nov 2006 20:11:16 -0000
Subject: [VoIP] SIP/firewall problem - the saga
References: <4555C336.19514.A7E6B6@localhost>
	<000801c705bf$d3d51cc0$0600a8c0@bluegrasspals.com>
Message-ID: <007f01c70696$b87b0700$0b01a8c0@acer1dd0bbc6d0>

Hi All
from the other side of the Pond!

Glad to know that we aren't the only ones having problems with SIP.

When I do a 'sip show peers' mine are all 5060.  However when I do an 'iax2 
show peers' I've got one  on Port 37657, it is on my own network -
AT323/AT323      81.174.170.48   (D)  255.255.255.255  37657         OK (833 
ms)


We've had success with one single CNET number off my *box working with SIP 
to a Fritz AVM router with two VoIP ports on it.  Works OK bothways 
(although we haven't sorted out the dyndns set yet).

However if we transfer exactly the same settings to a Sipura 2000 at the 
same location - it will not register and I can't see it when I do a 'sip 
show peers'.

Anyone any ideas?  Anyone got a working configuration for a Sipura 2000 end?

I also get this message every few seconds for a period of about 30 second 
occasionally - could this be why the above link became 'unable to register' 
for 24 hours then just came back on without us doing anything at either end?

Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:35:57 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:36:01 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:36:05 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:36:09 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:36:13 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:36:17 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info
Nov 12 19:36:21 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to 
parse contact info



I am hosting another ATA - a Draytek 2600V with two VoIP ports one of which 
is on CNET. We've got the SIP working OK bothways but every so often  I get 
the following message at the CLI>-

Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer 
'jgriffiths' is now UNREACHABLE!  Last qualify: 226
Nov 12 19:35:45 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer 
'jgriffiths' is now REACHABLE! (214ms / 2000ms)

The Draytek becomes 'unreachable for ten seconds at a time every few 
minutes. During this period you can't dial into the number.  However it 
doesn't interrupt an existing call.  Anyone any idea what is causing this 
'unreachable/reachable' situation?  Incidentally the 'chan_sip.c:11328' and 
'sip.c:9689'  are always the same in the error messages - are they SIP 
ports?

Ian Jolly

----- Original Message ----- 
From: "Jayson Smith" <ratguy at bellsouth.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Saturday, November 11, 2006 6:33 PM
Subject: Re: [VoIP] SIP/firewall problem - the saga


> Hi,
> HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
> accepting IAX2 connections on a random high port rather than 4569 (is that
> the right port or am I losing my mind too?).  Yeah, some NAT systems do
> really strange things!
> Jayson.
>
> ----- Original Message ----- 
> From: "Chad Perkins" <chad at maine.maine.edu>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Saturday, November 11, 2006 12:33 PM
> Subject: Re: [VoIP] SIP/firewall problem - the saga
>
>
>> > I hate SIP.
>> > I don't know if this will help, but, when I do a "sip show peers," I
>> > get some really odd ports for Dennis and for Jim:
>> >



From andrew.e.green at gmail.com  Sun Nov 12 14:24:36 2006
From: andrew.e.green at gmail.com (Andrew Green)
Date: Sun, 12 Nov 2006 16:54:36 -0330
Subject: [VoIP] SIP/firewall problem - the saga
In-Reply-To: <007f01c70696$b87b0700$0b01a8c0@acer1dd0bbc6d0>
References: <4555C336.19514.A7E6B6@localhost>
	<000801c705bf$d3d51cc0$0600a8c0@bluegrasspals.com>
	<007f01c70696$b87b0700$0b01a8c0@acer1dd0bbc6d0>
Message-ID: <c30e256a0611121224o1aa1772bs711e9bde1a9e971e@mail.gmail.com>

In a Sipura the only configuration options you should need to set are,
"Display Name" (what you want the caller-id name to come up as),
"Proxy" (the ip/dns of the asterisk box), "Register = Yes", "User ID"
(the sip user in asterisk), and the Password (the secret in asterisk).
I never had to set any other options to get mine to work.

Here is my asterisk sip context for my sipura-1001 (should also work
for a 2000):

[spa-1001] ;you should set this to the same as the username
type=friend
host=dynamic
username=spa-1001 ;this is the User ID
secret=spa ;this is the password
context=default

Also about the Draytek issue, try changing the registration timout to
about a minute so it will re-register every minute. That has helped a
friend's spa-1001 that was having the same problem.


Hope that helps a bit,
Andrew



On 11/12/06, Ian Jolly <ian at uax.org.uk> wrote:
> Hi All
> from the other side of the Pond!
>
> Glad to know that we aren't the only ones having problems with SIP.
>
> When I do a 'sip show peers' mine are all 5060.  However when I do an 'iax2
> show peers' I've got one  on Port 37657, it is on my own network -
> AT323/AT323      81.174.170.48   (D)  255.255.255.255  37657         OK (833
> ms)
>
>
> We've had success with one single CNET number off my *box working with SIP
> to a Fritz AVM router with two VoIP ports on it.  Works OK bothways
> (although we haven't sorted out the dyndns set yet).
>
> However if we transfer exactly the same settings to a Sipura 2000 at the
> same location - it will not register and I can't see it when I do a 'sip
> show peers'.
>
> Anyone any ideas?  Anyone got a working configuration for a Sipura 2000 end?
>
> I also get this message every few seconds for a period of about 30 second
> occasionally - could this be why the above link became 'unable to register'
> for 24 hours then just came back on without us doing anything at either end?
>
> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:35:57 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:36:01 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:36:05 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:36:09 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:36:13 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:36:17 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
> Nov 12 19:36:21 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
> parse contact info
>
>
>
> I am hosting another ATA - a Draytek 2600V with two VoIP ports one of which
> is on CNET. We've got the SIP working OK bothways but every so often  I get
> the following message at the CLI>-
>
> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226
> Nov 12 19:35:45 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
>
> The Draytek becomes 'unreachable for ten seconds at a time every few
> minutes. During this period you can't dial into the number.  However it
> doesn't interrupt an existing call.  Anyone any idea what is causing this
> 'unreachable/reachable' situation?  Incidentally the 'chan_sip.c:11328' and
> 'sip.c:9689'  are always the same in the error messages - are they SIP
> ports?
>
> Ian Jolly
>
> ----- Original Message -----
> From: "Jayson Smith" <ratguy at bellsouth.net>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Saturday, November 11, 2006 6:33 PM
> Subject: Re: [VoIP] SIP/firewall problem - the saga
>
>
> > Hi,
> > HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
> > accepting IAX2 connections on a random high port rather than 4569 (is that
> > the right port or am I losing my mind too?).  Yeah, some NAT systems do
> > really strange things!
> > Jayson.
> >
> > ----- Original Message -----
> > From: "Chad Perkins" <chad at maine.maine.edu>
> > To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> > Sent: Saturday, November 11, 2006 12:33 PM
> > Subject: Re: [VoIP] SIP/firewall problem - the saga
> >
> >
> >> > I hate SIP.
> >> > I don't know if this will help, but, when I do a "sip show peers," I
> >> > get some really odd ports for Dennis and for Jim:
> >> >
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>


From jnovack at stromberg-carlson.org  Sun Nov 12 14:31:07 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 12 Nov 2006 15:31:07 -0500
Subject: [VoIP] SIP/firewall problem - the saga
In-Reply-To: <c30e256a0611121224o1aa1772bs711e9bde1a9e971e@mail.gmail.com>
References: <4555C336.19514.A7E6B6@localhost>	<000801c705bf$d3d51cc0$0600a8c0@bluegrasspals.com>	<007f01c70696$b87b0700$0b01a8c0@acer1dd0bbc6d0>
	<c30e256a0611121224o1aa1772bs711e9bde1a9e971e@mail.gmail.com>
Message-ID: <4557848B.4000800@stromberg-carlson.org>

In a situation where both ends are through routers with NAT, it is more 
problematical., and many have problems. One issue seems to be that NAT 
is available in several flavors, none documented with the equipment.

For a SIP phone or ATA within one's own LAN it mostly just works.


John Novack


Andrew Green wrote:
> In a Sipura the only configuration options you should need to set are,
> "Display Name" (what you want the caller-id name to come up as),
> "Proxy" (the ip/dns of the asterisk box), "Register = Yes", "User ID"
> (the sip user in asterisk), and the Password (the secret in asterisk).
> I never had to set any other options to get mine to work.
>
> Here is my asterisk sip context for my sipura-1001 (should also work
> for a 2000):
>
> [spa-1001] ;you should set this to the same as the username
> type=friend
> host=dynamic
> username=spa-1001 ;this is the User ID
> secret=spa ;this is the password
> context=default
>
> Also about the Draytek issue, try changing the registration timout to
> about a minute so it will re-register every minute. That has helped a
> friend's spa-1001 that was having the same problem.
>
>
> Hope that helps a bit,
> Andrew
>
>
>
> On 11/12/06, Ian Jolly <ian at uax.org.uk> wrote:
>   
>> Hi All
>> from the other side of the Pond!
>>
>> Glad to know that we aren't the only ones having problems with SIP.
>>
>> When I do a 'sip show peers' mine are all 5060.  However when I do an 'iax2
>> show peers' I've got one  on Port 37657, it is on my own network -
>> AT323/AT323      81.174.170.48   (D)  255.255.255.255  37657         OK (833
>> ms)
>>
>>
>> We've had success with one single CNET number off my *box working with SIP
>> to a Fritz AVM router with two VoIP ports on it.  Works OK bothways
>> (although we haven't sorted out the dyndns set yet).
>>
>> However if we transfer exactly the same settings to a Sipura 2000 at the
>> same location - it will not register and I can't see it when I do a 'sip
>> show peers'.
>>
>> Anyone any ideas?  Anyone got a working configuration for a Sipura 2000 end?
>>
>> I also get this message every few seconds for a period of about 30 second
>> occasionally - could this be why the above link became 'unable to register'
>> for 24 hours then just came back on without us doing anything at either end?
>>
>> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:35:57 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:01 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:05 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:09 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:13 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:17 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:21 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>>
>>
>>
>> I am hosting another ATA - a Draytek 2600V with two VoIP ports one of which
>> is on CNET. We've got the SIP working OK bothways but every so often  I get
>> the following message at the CLI>-
>>
>> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
>> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226
>> Nov 12 19:35:45 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
>> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
>>
>> The Draytek becomes 'unreachable for ten seconds at a time every few
>> minutes. During this period you can't dial into the number.  However it
>> doesn't interrupt an existing call.  Anyone any idea what is causing this
>> 'unreachable/reachable' situation?  Incidentally the 'chan_sip.c:11328' and
>> 'sip.c:9689'  are always the same in the error messages - are they SIP
>> ports?
>>
>> Ian Jolly
>>
>> ----- Original Message -----
>> From: "Jayson Smith" <ratguy at bellsouth.net>
>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>> Sent: Saturday, November 11, 2006 6:33 PM
>> Subject: Re: [VoIP] SIP/firewall problem - the saga
>>
>>
>>     
>>> Hi,
>>> HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
>>> accepting IAX2 connections on a random high port rather than 4569 (is that
>>> the right port or am I losing my mind too?).  Yeah, some NAT systems do
>>> really strange things!
>>> Jayson.
>>>
>>> ----- Original Message -----
>>> From: "Chad Perkins" <chad at maine.maine.edu>
>>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>>> Sent: Saturday, November 11, 2006 12:33 PM
>>> Subject: Re: [VoIP] SIP/firewall problem - the saga
>>>
>>>
>>>       
>>>>> I hate SIP.
>>>>> I don't know if this will help, but, when I do a "sip show peers," I
>>>>> get some really odd ports for Dennis and for Jim:
>>>>>
>>>>>           
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>     
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>
>   

From jjones3601 at yahoo.com  Sun Nov 12 16:25:30 2006
From: jjones3601 at yahoo.com (john jones)
Date: Sun, 12 Nov 2006 14:25:30 -0800 (PST)
Subject: [VoIP] SIP/firewall problem - the saga
Message-ID: <20061112222530.41197.qmail@web34311.mail.mud.yahoo.com>

Are you using static NAT or dynamic NAT?  Typically, dynamic NAT is the default and if the device on the inside does not continue to use the connection, the translation is torn down after a relatively short time.  Also, the NAT translation is usually specific to an individual peer and can't be used by another external partner.

John

----- Original Message ----
From: John Novack <jnovack at stromberg-carlson.org>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Sunday, November 12, 2006 3:31:07 PM
Subject: Re: [VoIP] SIP/firewall problem - the saga

In a situation where both ends are through routers with NAT, it is more 
problematical., and many have problems. One issue seems to be that NAT 
is available in several flavors, none documented with the equipment.

For a SIP phone or ATA within one's own LAN it mostly just works.


John Novack


Andrew Green wrote:
> In a Sipura the only configuration options you should need to set are,
> "Display Name" (what you want the caller-id name to come up as),
> "Proxy" (the ip/dns of the asterisk box), "Register = Yes", "User ID"
> (the sip user in asterisk), and the Password (the secret in asterisk).
> I never had to set any other options to get mine to work.
>
> Here is my asterisk sip context for my sipura-1001 (should also work
> for a 2000):
>
> [spa-1001] ;you should set this to the same as the username
> type=friend
> host=dynamic
> username=spa-1001 ;this is the User ID
> secret=spa ;this is the password
> context=default
>
> Also about the Draytek issue, try changing the registration timout to
> about a minute so it will re-register every minute. That has helped a
> friend's spa-1001 that was having the same problem.
>
>
> Hope that helps a bit,
> Andrew
>
>
>
> On 11/12/06, Ian Jolly <ian at uax.org.uk> wrote:
>   
>> Hi All
>> from the other side of the Pond!
>>
>> Glad to know that we aren't the only ones having problems with SIP.
>>
>> When I do a 'sip show peers' mine are all 5060.  However when I do an 'iax2
>> show peers' I've got one  on Port 37657, it is on my own network -
>> AT323/AT323      81.174.170.48   (D)  255.255.255.255  37657         OK (833
>> ms)
>>
>>
>> We've had success with one single CNET number off my *box working with SIP
>> to a Fritz AVM router with two VoIP ports on it.  Works OK bothways
>> (although we haven't sorted out the dyndns set yet).
>>
>> However if we transfer exactly the same settings to a Sipura 2000 at the
>> same location - it will not register and I can't see it when I do a 'sip
>> show peers'.
>>
>> Anyone any ideas?  Anyone got a working configuration for a Sipura 2000 end?
>>
>> I also get this message every few seconds for a period of about 30 second
>> occasionally - could this be why the above link became 'unable to register'
>> for 24 hours then just came back on without us doing anything at either end?
>>
>> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:35:57 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:01 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:05 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:09 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:13 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:17 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>> Nov 12 19:36:21 WARNING[2485]: chan_sip.c:6393 register_verify: Failed to
>> parse contact info
>>
>>
>>
>> I am hosting another ATA - a Draytek 2600V with two VoIP ports one of which
>> is on CNET. We've got the SIP working OK bothways but every so often  I get
>> the following message at the CLI>-
>>
>> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
>> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226
>> Nov 12 19:35:45 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
>> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
>>
>> The Draytek becomes 'unreachable for ten seconds at a time every few
>> minutes. During this period you can't dial into the number.  However it
>> doesn't interrupt an existing call.  Anyone any idea what is causing this
>> 'unreachable/reachable' situation?  Incidentally the 'chan_sip.c:11328' and
>> 'sip.c:9689'  are always the same in the error messages - are they SIP
>> ports?
>>
>> Ian Jolly
>>
>> ----- Original Message -----
>> From: "Jayson Smith" <ratguy at bellsouth.net>
>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>> Sent: Saturday, November 11, 2006 6:33 PM
>> Subject: Re: [VoIP] SIP/firewall problem - the saga
>>
>>
>>     
>>> Hi,
>>> HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
>>> accepting IAX2 connections on a random high port rather than 4569 (is that
>>> the right port or am I losing my mind too?).  Yeah, some NAT systems do
>>> really strange things!
>>> Jayson.
>>>
>>> ----- Original Message -----
>>> From: "Chad Perkins" <chad at maine.maine.edu>
>>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>>> Sent: Saturday, November 11, 2006 12:33 PM
>>> Subject: Re: [VoIP] SIP/firewall problem - the saga
>>>
>>>
>>>       
>>>>> I hate SIP.
>>>>> I don't know if this will help, but, when I do a "sip show peers," I
>>>>> get some really odd ports for Dennis and for Jim:
>>>>>
>>>>>           
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>     
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>
>   
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/






From jnovack at stromberg-carlson.org  Sun Nov 12 16:37:12 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 12 Nov 2006 17:37:12 -0500
Subject: [VoIP] SIP/firewall problem - the saga
In-Reply-To: <20061112222530.41197.qmail@web34311.mail.mud.yahoo.com>
References: <20061112222530.41197.qmail@web34311.mail.mud.yahoo.com>
Message-ID: <4557A218.3090600@stromberg-carlson.org>

The scenario here is, behind one router, an Asterisk box with a static 
private IP on the lan servicing an ATA pr SIP phone behind another 
router, also with a static private IP on another LAN
Many folks on the Asterisk users list have problems, especially one way 
audio.
Some time back Kirt and I attempted to get such a connection working, 
and found no solution, though I am sure there must be one.
Most of the examples don't fit, either with one device or another not 
behind a router, or in a router DMZ..

Beyond that I can't say.
That is your area of expertise

John Novack


john jones wrote:
> Are you using static NAT or dynamic NAT?  Typically, dynamic NAT is 
> the default and if the device on the inside does not continue to use 
> the connection, the translation is torn down after a relatively short 
> time.  Also, the NAT translation is usually specific to an individual 
> peer and can't be used by another external partner.
>
> John
>
> ----- Original Message ----
> From: John Novack <jnovack at stromberg-carlson.org>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Sunday, November 12, 2006 3:31:07 PM
> Subject: Re: [VoIP] SIP/firewall problem - the saga
>
> In a situation where both ends are through routers with NAT, it is more
> problematical., and many have problems. One issue seems to be that NAT
> is available in several flavors, none documented with the equipment.
>
> For a SIP phone or ATA within one's own LAN it mostly just works.
>
>
> John Novack
>
>
> Andrew Green wrote:
> > In a Sipura the only configuration options you should need to set are,
> > "Display Name" (what you want the caller-id name to come up as),
> > "Proxy" (the ip/dns of the asterisk box), "Register = Yes", "User ID"
> > (the sip user in asterisk), and the Password (the secret in asterisk).
> > I never had to set any other options to get mine to work.
> >
> > Here is my asterisk sip context for my sipura-1001 (should also work
> > for a 2000):
> >
> > [spa-1001] ;you should set this to the same as the username
> > type=friend
> > host=dynamic
> > username=spa-1001 ;this is the User ID
> > secret=spa ;this is the password
> > context=default
> >
> > Also about the Draytek issue, try changing the registration timout to
> > about a minute so it will re-register every minute. That has helped a
> > friend's spa-1001 that was having the same problem.
> >
> >
> > Hope that helps a bit,
> > Andrew
> >
> >
> >
> > On 11/12/06, Ian Jolly <ian at uax.org.uk> wrote:
> >  
> >> Hi All
> >> from the other side of the Pond!
> >>
> >> Glad to know that we aren't the only ones having problems with SIP.
> >>
> >> When I do a 'sip show peers' mine are all 5060.  However when I do 
> an 'iax2
> >> show peers' I've got one  on Port 37657, it is on my own network -
> >> AT323/AT323      81.174.170.48   
> (D)  255.255.255.255  37657         OK (833
> >> ms)
> >>
> >>
> >> We've had success with one single CNET number off my *box working 
> with SIP
> >> to a Fritz AVM router with two VoIP ports on it.  Works OK bothways
> >> (although we haven't sorted out the dyndns set yet).
> >>
> >> However if we transfer exactly the same settings to a Sipura 2000 
> at the
> >> same location - it will not register and I can't see it when I do a 
> 'sip
> >> show peers'.
> >>
> >> Anyone any ideas?  Anyone got a working configuration for a Sipura 
> 2000 end?
> >>
> >> I also get this message every few seconds for a period of about 30 
> second
> >> occasionally - could this be why the above link became 'unable to 
> register'
> >> for 24 hours then just came back on without us doing anything at 
> either end?
> >>
> >> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:35:57 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:01 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:05 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:09 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:13 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:17 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:21 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >>
> >>
> >>
> >> I am hosting another ATA - a Draytek 2600V with two VoIP ports one 
> of which
> >> is on CNET. We've got the SIP working OK bothways but every so 
> often  I get
> >> the following message at the CLI>-
> >>
> >> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
> >> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226
> >> Nov 12 19:35:45 NOTICE[2485]: chan_sip.c:9689 
> handle_response_peerpoke: Peer
> >> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
> >>
> >> The Draytek becomes 'unreachable for ten seconds at a time every few
> >> minutes. During this period you can't dial into the number.  However it
> >> doesn't interrupt an existing call.  Anyone any idea what is 
> causing this
> >> 'unreachable/reachable' situation?  Incidentally the 
> 'chan_sip.c:11328' and
> >> 'sip.c:9689'  are always the same in the error messages - are they SIP
> >> ports?
> >>
> >> Ian Jolly
> >>
> >> ----- Original Message -----
> >> From: "Jayson Smith" <ratguy at bellsouth.net>
> >> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> >> Sent: Saturday, November 11, 2006 6:33 PM
> >> Subject: Re: [VoIP] SIP/firewall problem - the saga
> >>
> >>
> >>    
> >>> Hi,
> >>> HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
> >>> accepting IAX2 connections on a random high port rather than 4569 
> (is that
> >>> the right port or am I losing my mind too?).  Yeah, some NAT 
> systems do
> >>> really strange things!
> >>> Jayson.
> >>>
> >>> ----- Original Message -----
> >>> From: "Chad Perkins" <chad at maine.maine.edu>
> >>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> >>> Sent: Saturday, November 11, 2006 12:33 PM
> >>> Subject: Re: [VoIP] SIP/firewall problem - the saga
> >>>
> >>>
> >>>      
> >>>>> I hate SIP.
> >>>>> I don't know if this will help, but, when I do a "sip show peers," I
> >>>>> get some really odd ports for Dennis and for Jim:
> >>>>>
> >>>>>          
> >> _______________________________________________
> >> VoIP mailing list
> >> VoIP at ckts.info
> >> http://lists.ckts.info/mailman/listinfo/voip
> >> Project Web Page: http://www.ckts.info/
> >>
> >>
> >>    
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> >
> >  
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>

From ian at uax.org.uk  Sun Nov 12 16:43:31 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Sun, 12 Nov 2006 22:43:31 -0000
Subject: [VoIP] IAX/SIP ATA  -  was  Re:  SIP/firewall problem - the saga
References: <20061112222530.41197.qmail@web34311.mail.mud.yahoo.com>
Message-ID: <013701c706ab$fc1de3a0$0b01a8c0@acer1dd0bbc6d0>

Tnx for the info, folks. We'll give that a try but not tonight - supper and 
bed I think!

Anyone know anything about the ATcom AG-188 
http://www.atcom.cn/En_products_AG188.html  - Single FXS which seems to work 
with both SIP and IAX2  at the same time!!

Ian J



----- Original Message ----- 
From: "john jones" <jjones3601 at yahoo.com>
To: <jnovack at stromberg-carlson.org>; "Voice Over IP Tandem for Analog 
Switches" <voip at ckts.info>
Sent: Sunday, November 12, 2006 10:25 PM
Subject: Re: [VoIP] SIP/firewall problem - the saga


> Are you using static NAT or dynamic NAT?  Typically, dynamic NAT is the 
> default and if the device on the inside does not continue to use the 
> connection, the translation is torn down after a relatively short time. 
> Also, the NAT translation is usually specific to an individual peer and 
> can't be used by another external partner.
>
> John
>
> ----- Original Message ----
> From: John Novack <jnovack at stromberg-carlson.org>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Sunday, November 12, 2006 3:31:07 PM
> Subject: Re: [VoIP] SIP/firewall problem - the saga
>
> In a situation where both ends are through routers with NAT, it is more
> problematical., and many have problems. One issue seems to be that NAT
> is available in several flavors, none documented with the equipment.
>
> For a SIP phone or ATA within one's own LAN it mostly just works.
>
>
> John Novack
>
>
> Andrew Green wrote:
>> In a Sipura the only configuration options you should need to set are,
>> "Display Name" (what you want the caller-id name to come up as),
>> "Proxy" (the ip/dns of the asterisk box), "Register = Yes", "User ID"
>> (the sip user in asterisk), and the Password (the secret in asterisk).
>> I never had to set any other options to get mine to work.
>>
>> Here is my asterisk sip context for my sipura-1001 (should also work
>> for a 2000):
>>
>> [spa-1001] ;you should set this to the same as the username
>> type=friend
>> host=dynamic
>> username=spa-1001 ;this is the User ID
>> secret=spa ;this is the password
>> context=default
>>
>> Also about the Draytek issue, try changing the registration timout to
>> about a minute so it will re-register every minute. That has helped a
>> friend's spa-1001 that was having the same problem.
>>
>>
>> Hope that helps a bit,
>> Andrew



From chad at maine.maine.edu  Sun Nov 12 22:24:47 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Sun, 12 Nov 2006 23:24:47 -0500
Subject: [VoIP] SIP/firewall problem - the saga
In-Reply-To: <A691C3685A152E4AAC9608B8D8E113163EB0AB@godzilla.hickory.vyger.net>
Message-ID: <4557AD3F.9150.822642B@localhost>

> > > My asterisk box is NATted behind an IP-COP linux firewall.  I have
> > > port 5060 forwarded to my asterisk box, but the RTP ports are just
> > > opened up -- not forwarded anywhere.
> > 
> > I'm confused as to how the RTP knows where to go if not forwarded. 
> > Is Asterisk on an "Orange" DMZ port?  Atleast IP-Cop should have
> > some logs that help with this
> > sort of thing.  Too many COTS firewall/routers have zipo.  :(
> 
> Yes, it is.  My desktop PCs are in the Green Zone, and my servers are
> in the DMZ (orange interface).
> 
> Biggest thing, though, is that I double checked my port forwarding,
> and see that I have ports 10000-20000 UDP forwarded to the Asterisk
> box. So, I stand corrected.  

That makes sense.

> And this will show a certain amount of network ignorance on my part,
> but, is RTP UDP? Or is it its own animal?

Yes, check out http://en.wikipedia.org/wiki/Real-time_Transport_Protocol
short and sweet (cough, cough)...

Chad


From chad at maine.maine.edu  Sun Nov 12 22:37:17 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Sun, 12 Nov 2006 23:37:17 -0500
Subject: [VoIP] SIP/firewall problem - the saga
In-Reply-To: <007f01c70696$b87b0700$0b01a8c0@acer1dd0bbc6d0>
Message-ID: <4557B02D.5440.82DD70C@localhost>

> Hi All
> from the other side of the Pond!
> Glad to know that we aren't the only ones having problems with SIP.
> 
> When I do a 'sip show peers' mine are all 5060.  However when I do an
> 'iax2 show peers' I've got one  on Port 37657, it is on my own network
> - AT323/AT323      81.174.170.48   (D)  255.255.255.255  37657 OK (833 ms)
> 
> We've had success with one single CNET number off my *box working with
> SIP to a Fritz AVM router with two VoIP ports on it.  Works OK
> bothways (although we haven't sorted out the dyndns set yet).
> 
> However if we transfer exactly the same settings to a Sipura 2000 at
> the same location - it will not register and I can't see it when I do
> a 'sip show peers'.
> 
> Anyone any ideas?  Anyone got a working configuration for a Sipura
> 2000 end?

Is the SPA nated?  Do you have 'nat=yes' in sip.conf for that device (that killed my 
first one).  I wouldn't think the 2000 would be all that different from the 2002 I posted 
the other day.

[snip] 
> I am hosting another ATA - a Draytek 2600V with two VoIP ports one of
> which is on CNET. We've got the SIP working OK bothways but every so
> often  I get the following message at the CLI>-
> 
> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226 Nov 12 19:35:45
> NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
> 
> The Draytek becomes 'unreachable for ten seconds at a time every few
> minutes. During this period you can't dial into the number.  However
> it doesn't interrupt an existing call.  Anyone any idea what is
> causing this 'unreachable/reachable' situation?  Incidentally the
> 'chan_sip.c:11328' and 'sip.c:9689'  are always the same in the error
> messages - are they SIP ports?
> Ian Jolly

No.  I think they are code line numbers.  This message is usually a sign that the 
network is "down" somewhere and they can't communicate; but if you able to talk as 
this problem comes and goes then that is weird.

chad
 
> ----- Original Message ----- 
> From: "Jayson Smith" <ratguy at bellsouth.net>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Saturday, November 11, 2006 6:33 PM
> Subject: Re: [VoIP] SIP/firewall problem - the saga
> 
> 
> > Hi,
> > HMMM, sound familiar?  Of course, in my situation, I'm appearing to
> > be accepting IAX2 connections on a random high port rather than 4569
> > (is that the right port or am I losing my mind too?).  Yeah, some
> > NAT systems do really strange things! Jayson.
> >
> > ----- Original Message ----- 
> > From: "Chad Perkins" <chad at maine.maine.edu>
> > To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> > Sent: Saturday, November 11, 2006 12:33 PM Subject: Re: [VoIP]
> > SIP/firewall problem - the saga
> >
> >
> >> > I hate SIP.
> >> > I don't know if this will help, but, when I do a "sip show
> >> > peers," I get some really odd ports for Dennis and for Jim:
> >> >
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 
> 
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.409 / Virus Database: 268.14.3/530 - Release Date:
> 11/11/06
> 




From ian at uax.org.uk  Mon Nov 13 03:47:45 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Mon, 13 Nov 2006 09:47:45 -0000
Subject: [VoIP] SIP/firewall problem - the saga
References: <4557B02D.5440.82DD70C@localhost>
Message-ID: <00a301c70708$c5894930$0b01a8c0@acer1dd0bbc6d0>

Chad asked if I'd got NAT=yes  - I have!

The Fritz AVM router works OK through one of its two VoIP ports but does 
that work through the NAT ? Or is it on the 'dirty' side of the NAT?

That is a point about the SPA end being NAT'ed - I'm sure it will be.   I'll 
try changing the 'nat=   ' setting and see if we have any success!  My 
sip.conf settings (excluding the individual settings)

[general]
context=default           ; Default context for incoming calls
bindport=5060           ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes           ; Enable DNS SRV lookups on outbound calls
qualify=yes
externip = 81.174.170.48  ; Address that we're going to put in outbound SIP 
messages
                                  ; if we're behind a NAT - My fixed IP
localnet=192.168.1.0/255.255.255.0; All RFC 1918 addresses are local 
networks
nat=yes                         ;Global NAT settings  (Affects all peers and 
users)

I'll let you know how we get on!

Tnx

Ian


----- Original Message ----- 
From: "Chad Perkins" <chad at maine.maine.edu>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Monday, November 13, 2006 4:37 AM
Subject: Re: [VoIP] SIP/firewall problem - the saga


>> Hi All
>> from the other side of the Pond!
>> Glad to know that we aren't the only ones having problems with SIP.
>>
>> When I do a 'sip show peers' mine are all 5060.  However when I do an
>> 'iax2 show peers' I've got one  on Port 37657, it is on my own network
>> - AT323/AT323      81.174.170.48   (D)  255.255.255.255  37657 OK (833 
>> ms)
>>
>> We've had success with one single CNET number off my *box working with
>> SIP to a Fritz AVM router with two VoIP ports on it.  Works OK
>> bothways (although we haven't sorted out the dyndns set yet).
>>
>> However if we transfer exactly the same settings to a Sipura 2000 at
>> the same location - it will not register and I can't see it when I do
>> a 'sip show peers'.
>>
>> Anyone any ideas?  Anyone got a working configuration for a Sipura
>> 2000 end?
>
> Is the SPA nated?  Do you have 'nat=yes' in sip.conf for that device (that 
> killed my
> first one).  I wouldn't think the 2000 would be all that different from 
> the 2002 I posted
> the other day.
>
> [snip]
>> I am hosting another ATA - a Draytek 2600V with two VoIP ports one of
>> which is on CNET. We've got the SIP working OK bothways but every so
>> often  I get the following message at the CLI>-
>>
>> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
>> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226 Nov 12 19:35:45
>> NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
>> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
>>
>> The Draytek becomes 'unreachable for ten seconds at a time every few
>> minutes. During this period you can't dial into the number.  However
>> it doesn't interrupt an existing call.  Anyone any idea what is
>> causing this 'unreachable/reachable' situation?  Incidentally the
>> 'chan_sip.c:11328' and 'sip.c:9689'  are