[asterisk-users] Problem with libpri / asterisk

Andres andres at telesip.net
Mon Feb 13 09:52:43 CST 2012


On 2/13/2012 10:49 AM, Andres wrote:
>
>>
>>    -- Accepting call from '418nxxxxx2' to '418nxxxxx1' on channel 
>> 0/1, span 1
>>    -- Executing [418nxxxxx1 at ael-default:1] 
>> Answer("DAHDI/i1/418nxxxxx2-b", "") in new stack
>>    -- Executing [418nxxxxx1 at ael-default:2] 
>> Wait("DAHDI/i1/418nxxxxx2-b", "2") in new stack
>>    -- Executing [418nxxxxx1 at ael-default:3] 
>> Playback("DAHDI/i1/418nxxxxx2-b", "demo-thanks") in new stack
>>    -- <DAHDI/i1/418nxxxxx2-b> Playing 'demo-thanks.ulaw' (language 'fr')
>>    -- Executing [418nxxxxx1 at ael-default:4] 
>> Dial("DAHDI/i1/418nxxxxx2-b", "DAHDI/G1/418nxxxxx2") in new stack
>>    -- Requested transfer capability: 0x00 - SPEECH
>>    -- Called DAHDI/G1/418nxxxxx2
>>    -- DAHDI/i1/418nxxxxx2-c is proceeding passing it to 
>> DAHDI/i1/418nxxxxx2-b
>>    -- DAHDI/i1/418nxxxxx2-c is ringing
>>    -- DAHDI/i1/418nxxxxx2-c is making progress passing it to 
>> DAHDI/i1/418nxxxxx2-b
>>    -- DAHDI/i1/418nxxxxx2-c answered DAHDI/i1/418nxxxxx2-b
>>    -- Native bridging DAHDI/i1/418nxxxxx2-b and DAHDI/i1/418nxxxxx2-c
>>    -- Span 1: Channel 0/1 got hangup request, cause 16
>>    -- Hungup 'DAHDI/i1/418nxxxxx2-c'
>>  == Spawn extension (ael-default, 418nxxxxx1, 4) exited non-zero on 
>> 'DAHDI/i1/418nxxxxx2-b'
>>    -- Hungup 'DAHDI/i1/418nxxxxx2-b'
>>
>> BUT, if I originate the call from my curent PRI, it goes in and out 
>> and all
>> is well. I noticed that if the calls go trough correctly and hangup 
>> manually, it also stats the exact same thing (cause 16). So the above 
>> console output might not be that much usefull...
>>
>> I've had a case open with Sangoma for this issue, and they suggested 
>> I go the libpri/asterisk for more help debuging this issue, since on 
>> their end, the disconnect comes from the telco...
> My guess is your new setup is trying to do a PRI 2B Transfer (meaning 
> that Asterisk is trying to handoff two B channels of a PRI to the 
> upstream switch).  It is probably being rejected and the call is 
> hanging up.  You will need to dig into the PRI debug of both scenarios 
> and compare.   I was not even aware that Asterisk could do that so it 
> may be some new feature being worked on.
I just found this:  http://wiki.sangoma.com/Asterisk-FAQ#TBCT

Maybe you should check and see if it is enabled.
>>
>> They suggested I try a different version of asterisk, wich I did to 
>> no avail, or try there NBE product instead of libpri...
>>
>> So, did anybody ever encontered something like that ? What steps 
>> should I take to diagnose the problem furhter ?
>>
>> Thanks for any help.
>>
>>
>>
>
>


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