[VoIP] Registered but will not work !

Sat Nov 25 15:41:29 CST 2006

Just a stab, since I can't see your total configuration, but you issue 
the dial command with a g1, and if you only have one FXO, is it possible 
that one circuit has a problem, is busy or ??
g is for a group, from low to high channel numbers, G is for the reverse.
When you start Linux, what does ztcfg -vvv produce at the console?

Is it ONLY for incoming sipgate calls?

John Novack

Ian Jolly wrote:
> 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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>
>
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