[VoIP] Strange interaction with Asterisk 1.4.13 and Linksys PAP2T

Sat Nov 17 13:32:13 CST 2007

Lee

There are many different routers and many different forms of NAT.  If you are
using a router with what is known as 'consistent NAT' it will allow you to
allocate the same port  (eg 4569 or 5060) to a number of devices.  The router
itself will then allocate a random port to a particular device which it then
remembers when packets come back intended for that device.  The port number
allocated can change with different sessions.

Peter

On Sat, 17 Nov 2007 13:01:19 -0500, "Lee Spenadel" <lee at spenadel.com> wrote:

>Brian,
>
>Don't get this.  At my friends house, we connect on port 3878.  At his
>office, where he plugged the ATA in to his network, we connect on port
>50478.  I checked the port designation in the ATA: 5060.  I set the
>port=5060 in the sip peer definition.  So I'm not sure where the random
>ports are coming from.  What's even more interesting is that the firewalls
>are not set to pass traffic on these ports.
>
>Lee
>
>-----Original Message-----
>From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>windmill
>Sent: Saturday, November 17, 2007 10:14 AM
>To: Voice Over IP Tandem for Analog Switches
>Subject: Re: [VoIP] Strange interaction with Asterisk 1.4.13 and Linksys
>PAP2T
>
>Lee,
>
>Althjough port 3763 is unusual I have seen similar strange ports before. 
>On one of my Windows PCs I am running three softphones , two are 
>registered with one experimental *box and the third with my regular 
>*box. So the third softphone uses port 5060. registered to my 
>experimental *box I have a Zoiper with two sip accounts but I also have 
>an Xlite. I cannot set the port on the Xlite and it automatically 
>allocates itself to a port which does not conflict with the other ports 
>in use by the other softphones. At times I have seen it set itself to 
>port 37763 for instance but it seems to be a different port at every 
>boot. It appears that even though ports can be set in sip.conf entries 
>the remote registering device can register and connect with any port it 
>desires, I suspect this is because  the bindport in the general entry of 
>sip.conf is usually defaulted to all ports.
>
>Brian
>
>Lee Spenadel wrote:
>> As it turns out, the qualify statement on the sip peer definition is what
>> fixed it.  For some reason in 1.2 it wasn't defined, nor needed.  Unless I
>> did have it defined and removed it in a "senior" moment.....
>Nonetheless,
>> I defined it and the firewall is happy, the peer comes up as OK and I can
>> call my friend again.
>>
>> I define the port as 5060, per the norm.  I'm still not sure why it's
>going
>> out on port 3763.  That one I'll have to look in to.
>>
>> Lee
>>
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> windmill
>> Sent: Friday, November 16, 2007 9:27 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Strange interaction with Asterisk 1.4.13 and Linksys
>> PAP2T
>>
>> As long as the ATA is recognised by your *box it will always be able to 
>> make calls through it. If it was working before the upgrade I can't see 
>> a reason why it would not work now unless something in the conf files 
>> has changed.
>>
>> Port 3763 seems odd for a sip port, do you have that port set in the 
>> sip.conf entry? If not then * will be looking on port 5060 or some other 
>> port perhaps depending how your entries are arranged. Changing some 
>> other entry in sip.conf might have caused the problem.
>>
>> Because the port is unmonitored the CLI information for sip show peers 
>> is irrelevant except when registration occurs. I think this would be 
>> proved if you put qualify = yes in the sip.conf entry but be prepared 
>> for acres of CLI screen to disappear as the ATA is reported 
>> 'unreachable' continuously!
>>
>> Are you sure the IP address of the ATA is stable?
>>
>> Brian
>>
>> Lee Spenadel wrote:
>>   
>>> I've had this ATA at a friend's house for months working without a hitch.
>>>     
>> I
>>   
>>> don't know if this is related to my upgrade to 1.4 from 1.2, but when I
>>>     
>> call
>>   
>>> his extension, I get no ring progress tone and the phone fails to ring on
>>> his end.  He, however, can call me or any other CNET user at any time.
>>> Nothing has changed in the ATA configuration.  Nothing has changed on my
>>> sip.conf related to his definition.  The ATA registers normally on my
>>> switch, with an interval of 3600.  A sip show peers shows a healthy ATA
>>> registered out there in the ether.  
>>>
>>>  
>>>
>>> 3494358/3494358            24.60.244.56     D   N      3763
>>>     
>> Unmonitored
>>   
>>>  
>>>
>>> When I reboot the ATA I can call it, which is probably due to it
>>> re-registering.  Nothing has changed re: it's registration status on the
>>> CLI.  However, my ability to continue calling the extension passes with
>>> about 2-3 minutes of inactivity.  The only way I can call him is to
>reboot
>>> the ATA again. 
>>>
>>>  
>>>
>>> Anyone seen this one?
>>>
>>>  
>>>
>>> Here's what the CLI spits out when I can him when it's working:
>>>
>>>  
>>>
>>>  
>>>
>>>     -- Executing [3494358 at extensions:1] Dial("SIP/6000-09ec91d8",
>>> "SIP/3494358|30") in new stack
>>>
>>>     -- Called 3494358
>>>
>>>     -- SIP/3494358-09ecc9c0 is ringing
>>>
>>>  
>>>
>>>  
>>>
>>> When it's not working:
>>>
>>>  
>>>
>>>  
>>>
>>>     -- Executing [3494358 at extensions:1] Dial("SIP/6000-09e5ce40",
>>> "SIP/3494358|30") in new stack
>>>
>>>     -- Called 3494358
>>>
>>> CAPECODNET*CLI>
>>>
>>>  
>>>
>>> Lee
>>>
>>>  
>>>
>>>  
>>>
>>>  
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
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>>> Project Web Page: http://www.ckts.info/
>>>
>>>
>>>
>>>   
>>>     
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