[VoIP] Registered but will not work !
Ian Jolly
ian at uax.org.uk
Sat Nov 25 14:44:01 CST 2006
I'm trying to get an incoming call from a sipgate.co.uk number 1558081 (also
accessed by a 'landline' PSTN number 01330 55 8081) to route through my
Asterisk box and out on an FXO card (TDM11B) to my digital PABX extension
(extension 8081) which has the Speaking Clock connected to it.
I have this in my extensions.conf
[incoming_sipgate] ; 0133 055 8081
SPEAKING CLOCK
exten => 1558081,1,NoOp(--- ${CALLERID} calling on Sipgate (${EXTEN}) ---)
exten => 1558081,2,Wait(1)
exten => 1558081,3,Dial,(ZAP/g1/8081)
exten => 1558081,4,Hangup
I'm mystified why on the first occasion I tried, I at least saw the call
arrive at the Asterisk but get no further. Now they don't even arrive at the
Asterisk.
The sip.conf is shown below in my previous message. It also appears in the
'unreachable/reachables' list at the CLI> prompt as below -
Nov 25 21:23:50 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
'sipgate' is now UNREACHABLE! Last qualify: 51
Nov 25 21:24:00 NOTICE[2491]: chan_iax2.c:7772 iax2_poke_noanswer: Peer
'056456922' is now UNREACHABLE! Time: 37
Nov 25 21:24:08 NOTICE[2491]: chan_iax2.c:7772 iax2_poke_noanswer: Peer
'pwalker' is now UNREACHABLE! Time: 65
Nov 25 21:24:36 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
'jgriffiths' is now UNREACHABLE! Last qualify: 103
Nov 25 21:25:01 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
'sipgate' is now REACHABLE! (1050ms / 2000ms)
Nov 25 21:25:01 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
'jgriffiths' is now REACHABLE! (1140ms / 2000ms)
-- Registered IAX2 '056456922' (AUTHENTICATED) at 84.69.67.29:4569
Nov 25 21:25:03 NOTICE[2491]: chan_iax2.c:7102 socket_read: Peer '056456922'
is now REACHABLE! Time: 50
Nov 25 21:25:10 NOTICE[2485]: chan_sip.c:9689 handle_response_peerpoke: Peer
'056456921' is now REACHABLE! (92ms / 2000ms)
Nov 25 21:25:32 NOTICE[2485]: chan_sip.c:5267 sip_reg_timeout: --
Registration for '1558081 at sipgate.co.uk' timed out, trying again (Attempt
#1)
Nov 25 21:26:41 NOTICE[2491]: chan_iax2.c:5121 register_verify: Host
84.69.67.29 failed MD5 authentication for '056456922'
(b98234e1698e8da55b2b7f39c4d2c8cb != fc60f31f5998392a5465e6b1b542eca3)
Nov 25 21:30:58 NOTICE[2491]: chan_iax2.c:7108 socket_read: Peer 'AT323' is
now TOO LAGGED (-4 ms)!
Nov 25 21:31:08 NOTICE[2491]: chan_iax2.c:7102 socket_read: Peer 'AT323' is
now REACHABLE! Time: 3
Ian Jolly
----- Original Message -----
From: "Andrew Green" <andrew.e.green at gmail.com>
To: "Ian Jolly" <ian at uax.org.uk>; "Voice Over IP Tandem for Analog Switches"
<voip at ckts.info>
Sent: Saturday, November 25, 2006 6:53 PM
Subject: Re: [VoIP] Registered but will not work !
> It seems you are trying to dial out to a Zaptel line group but it's
> not working because you do not have the Zaptel asterisk modules
> installed ("Unable to create channel of type '(ZAP' (cause 66 -
> Channel not implemented)"). Are you using any type of PCI FXS or FXO
> card in your machine to dial out and how are you wanting to dial out?
>
> -Andrew
>
> On 11/25/06, Ian Jolly <ian at uax.org.uk> wrote:
>> I have a 'stand alone' Speaking Clock working on my Asterisk box accessed
>> by dialling my area code followed by the old national UK code for the
>> Speaking Clock - 8081.
>>
>> I've managed to get a VoIP UK 'geographic' phone number from an ITSP
>> (sipgate.co.uk) which ends in 8081 which I'm trying to route to the
>> Clock on my Asterisk box. With the assistance of the UK's Asterisk Guru
>> (a.k.a. Jon Kay) I've added it to my sip.conf and extensions.conf.
>>
>> On my first attempt to dial the number - I got the following at the CLI>
>>
>> -- Executing NoOp("SIP/1558081-d469", "--- "01352700484" <01352700484>
>> calling on Sipgate (1558081) ---") in new stack
>> -- Executing Wait("SIP/1558081-d469", "1") in new stack
>> -- Executing Dial("SIP/1558081-d469", "(ZAP/g1/8081)") in new stack
>> Nov 24 15:38:20 WARNING[17787]: channel.c:2530 ast_request: No channel
>> type registered for '(ZAP'
>> Nov 24 15:38:20 NOTICE[17787]: app_dial.c:1010 dial_exec_full: Unable to
>> create channel of type '(ZAP' (cause 66 - Channel not implemented)
>> == Everyone is busy/congested at this time (1:0/0/1)
>> -- Executing Hangup("SIP/1558081-d469", "") in new stack
>> == Spawn extension (incoming_sipgate, 1558081, 4) exited non-zero on
>> 'SIP/1558081-d469'
>>
>> I didn't get through to the Speaking Clock but at least I did get through
>> to my Asterisk box as it saw the CLI of the landline I was calling from
>> (01352 700484) !!
>>
>> However all subsequent calls from a landline receive 'Number Unobtainable
>> Tone' from the landline provider and nothing is seen reaching the
>> Asterisk box.
>>
>> I added the following set in my sip.conf -
>>
>> in [general]
>> register => 1558081:mypassword at sipgate.co.uk/1558081
>>
>> in [authentication]
>>
>> [sipgate] ; 0133 055 8081 SPEAKING CLOCK
>> type=peer
>> context=incoming_sipgate
>> fromuser=1558081
>> username=1558081
>> authuser=1558081
>> secret=mypassword ; (Whatever your account password is)
>> host=sipgate.co.uk
>> fromdomain=sipgate.co.uk
>> dtmfmode=info
>> insecure=very
>> qualify=yes
>>
>> sip show registry gives -
>> asterisk*CLI> sip show registry
>> Host Username Refresh State
>> sipgate.co.uk:5060 1558081 105 Registered
>>
>> sip show peers gives -
>> asterisk*CLI> sip show peers
>> Name/username Host Dyn Nat ACL Port Status
>> sipgate/1558081 217.10.79.23 N 5060 OK (81
>> ms)
>>
>> It appears to be registered but I still get Number Unobtainable Tone when
>> I dial 0133 055 8081 from a landline and still get nothing happening at
>> the CLI>
>>
>> The 0133 055 8081 number works OK if I set it up on one of the voice
>> ports of my Draytek router.
>>
>> Anyone any ideas please?
>>
>> Ian Jolly
>>
>>
>> +44 (0)352 85 26 (via a 1929 GPO Rural Automatic eXchange!)
>>
>> _______________________________________________
>> 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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