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

Tue Jul 20 11:10:59 CDT 2010

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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>>>> <http://www.api-digital.com>   --
>>>>
>>>>
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>>>> http://www.aste...
>>>>
>>>>
>>>> -- 
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>>          
>>      
>
>    

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