[asterisk-users] Call not going through and failing because "never answered"

Tue Jul 20 11:24:00 CDT 2010

Posting a sip debug will probably be helpfull aswell as you can see 
exactly where the traffic is being sent and what the response was.

Andy Beak wrote:
> Hi,
> 
> Thanks, I added that.  I'll ask my network provider if they received 
> these message tomorrow morning.  That will narrow things down to either 
> an Asterisk configuration or a network routing issue.
> 
> There is not really a caller, I'm trying to use Asterisk as an Automated 
> Voice Message server to dial phone numbers and play an mp3.
> 
> I'm using my mobile phone to test on and it doesn't ring.  Asterisk 
> gives the following message immediately after reading the .call file 
> from the spool directory:
> 
> -- Attempting call on SIP/MTN-NEW/mynumber for application 
> MP3Player(/myfile) (Retry 1)
>   == Using SIP RTP CoS mark 5
>  > Channel SIP/MTN-NEW-00000001 was never answered.
> [Jul 20 18:07:37] NOTICE[22259]: pbx_spool.c:339 attempt_thread: Call 
> failed to go through, reason (8) Congestion (circuits busy)
> 
> Because the phone doesn't ring and the error message appears immediately 
> I don't think it's a timeout issue.
> 
> Will reading the source for pbx_spool.c at line 339 give any clues as to 
> what's happening or will that be a waste of time?
> 
> Cheers,
>  Andy
> 
> 
> On 20/07/2010 05:42 PM, Gareth Blades wrote:
>> If you add qualify=yes to the setting in sip.conf it will send a sip
>> message to the peer every 60 seconds to check if it is alive.
>> If you try to make a call while the peer is not alive it will fail
>> immediatly rather than the caller hearing silence while your box waits
>> for a reply timeout.
>>
>> Andy Beak wrote:
>>   
>>> Hi,
>>>
>>> No that is the correct address.  I know it is an internal IP.
>>>
>>> We have our machine hosted in racks at our SIP providers data center.
>>>
>>> They've patched a new port to our cabinet and linked that to a gateway
>>> (172.28.20.105).
>>>
>>> As long as we use that gateway (and the IP address they assigned to us)
>>> our traffic will reach their SBC.
>>>
>>> I've confirmed that traceroute follows the path it is supposed to:
>>>
>>> traceroute to 192.168.34.1 (192.168.34.1), 30 hops max, 60 byte packets
>>>   1  192.168.0.1 (192.168.0.1)  0.656 ms  0.562 ms  0.501 ms
>>>   2  172.28.20.105 (172.28.20.105)  1.211 ms  1.209 ms  1.196 ms
>>>   3  192.168.34.5 (192.168.34.5)  23.270 ms  23.269 ms  23.328 ms
>>>   4  * * *
>>>   5  * * *
>>>   6  * * *^C
>>>
>>> Is there a way to test in Asterisk if it is able to reach a particular
>>> IP address?  Do you think that there is a routing problem here?
>>>
>>> Thanks,
>>>   Andy
>>>
>>>
>>>
>>>
>>> On 20/07/2010 04:58 PM, Zeeshan Zakaria wrote:
>>>     
>>>> This "host=192.168.34.1" is where you'll put your provider's IP
>>>> address. Currently you are using some local address which is not your
>>>> provider's IP address. Where did you get it from? Call your providrr
>>>> and ask them the IP address of the server where you'll be sending your
>>>> calls.
>>>>
>>>> Zeeshan A Zakaria
>>>>
>>>> -- 
>>>> www.ilovetovoip.com<http://www.ilovetovoip.com>
>>>>
>>>>       
>>>>> On 2010-07-20 10:27 AM, "Andy Beak"<andrewb at cellsmart.co.za
>>>>> <mailto:andrewb at cellsmart.co.za>>  wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> I set my list to subscribe to digest and I can't see how to reply to
>>>>> your reply without starting a new thread.
>>>>>
>>>>> There is no need for SIP username and password because the provider
>>>>> authenticates me on my IP address.
>>>>>
>>>>> I thought that "host=192.168.34.1" would be the sip provider IP 
>>>>> address.
>>>>>
>>>>> At this point I don't need to accept incoming calls or place
>>>>> VOIP-to-VOIP.  All I need to do is connect to their PBX to place a
>>>>> call to a cellphone.
>>>>>
>>>>> I reread all the documentation I could find and couldn't see where
>>>>> else in sip.conf I should set the provider IP.
>>>>>
>>>>> Thanks for your reply,
>>>>>   Andy
>>>>>
>>>>>
>>>>>
>>>>>         
>>>>>> In your sip.conf, there is no mention of your sip provider's IP
>>>>>>            
>>>>> address, username and secret (pa...
>>>>>
>>>>> www.ilovetovoip.com<http://www.ilovetovoip.com>
>>>>> <http://www.ilovetovoip.com>
>>>>>
>>>>>
>>>>>
>>>>>         
>>>>>> On 2010-07-20 5:09 AM, "Andy Beak"<andrewb at xxxxxxxxxxxxxxx
>>>>>>            
>>>>> <mailto:andrewb at xxxxxxxxxxxxxxx<mailto:andrewb at xxxxxxxxxxxxxxx>>>  
>>>>> wr...
>>>>>
>>>>> -- 
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>>>      
>>
>>    
> 