[asterisk-users] Problem with libpri / asterisk
Hi all !
We currently have an asterisk box that is rather old (runs Asterisk
1.4.21.2), and it's connected to the PSTN with a sangoma A104d card.
Now we have a new PRI at another location, and I use that occasion to
build 2 new servers, one to replace our aging one and a new one for this new
pri.
So I downloaded the lastest libpri / asterisk / wanpipe driver, but the
previous version of dahdi (2.5), since the latest wanpipe isn't compatible
with dahdi 2.6. All is built from source
Now, all seems to be working OK. I can connect a SIP phone to my new box,
make calls to the outside, receive calls etc.
But, I can't seem to bridge a call. So on my new server, with the new PRI, I
got a Sangoma a104 card (no echo-canceler on this one).
In my extensions.ael, I got this :
418nx1 = {
Answer();
Wait (2);
Playback(demo-thanks);
Dial(${TRUNK}/418nx2);
};
TRUNK is DAHDI/G1
Where 418nx1 is a DID on my new PRI and 418nx2 is my cellphone
number.
When I do a call from my home phone or cell phone to my new PRI to
418nx1, I hear the demo-thanks file, and then it dials out. My cellphone
rings, but as soon as I pick up the call, the calls hangs up :
-- Accepting call from '418nx2' to '418nx1' on channel 0/1, span
1
-- Executing [418nx1@ael-default:1] Answer(DAHDI/i1/418nx2-b,
) in new stack
-- Executing [418nx1@ael-default:2] Wait(DAHDI/i1/418nx2-b,
2) in new stack
-- Executing [418nx1@ael-default:3]
Playback(DAHDI/i1/418nx2-b, demo-thanks) in new stack
-- DAHDI/i1/418nx2-b Playing 'demo-thanks.ulaw' (language 'fr')
-- Executing [418nx1@ael-default:4] Dial(DAHDI/i1/418nx2-b,
DAHDI/G1/418nx2) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called DAHDI/G1/418nx2
-- DAHDI/i1/418nx2-c is proceeding passing it to
DAHDI/i1/418nx2-b
-- DAHDI/i1/418nx2-c is ringing
-- DAHDI/i1/418nx2-c is making progress passing it to
DAHDI/i1/418nx2-b
-- DAHDI/i1/418nx2-c answered DAHDI/i1/418nx2-b
-- Native bridging DAHDI/i1/418nx2-b and DAHDI/i1/418nx2-c
-- Span 1: Channel 0/1 got hangup request, cause 16
-- Hungup 'DAHDI/i1/418nx2-c'
== Spawn extension (ael-default, 418nx1, 4) exited non-zero on
'DAHDI/i1/418nx2-b'
-- Hungup 'DAHDI/i1/418nx2-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...
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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Re: [asterisk-users] Problem with libpri / asterisk
-- Accepting call from '418nx2' to '418nx1' on channel 0/1, span 1 -- Executing [418nx1@ael-default:1] Answer(DAHDI/i1/418nx2-b, ) in new stack -- Executing [418nx1@ael-default:2] Wait(DAHDI/i1/418nx2-b, 2) in new stack -- Executing [418nx1@ael-default:3] Playback(DAHDI/i1/418nx2-b, demo-thanks) in new stack -- DAHDI/i1/418nx2-b Playing 'demo-thanks.ulaw' (language 'fr') -- Executing [418nx1@ael-default:4] Dial(DAHDI/i1/418nx2-b, DAHDI/G1/418nx2) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/418nx2 -- DAHDI/i1/418nx2-c is proceeding passing it to DAHDI/i1/418nx2-b -- DAHDI/i1/418nx2-c is ringing -- DAHDI/i1/418nx2-c is making progress passing it to DAHDI/i1/418nx2-b -- DAHDI/i1/418nx2-c answered DAHDI/i1/418nx2-b -- Native bridging DAHDI/i1/418nx2-b and DAHDI/i1/418nx2-c -- Span 1: Channel 0/1 got hangup request, cause 16 -- Hungup 'DAHDI/i1/418nx2-c' == Spawn extension (ael-default, 418nx1, 4) exited non-zero on 'DAHDI/i1/418nx2-b' -- Hungup 'DAHDI/i1/418nx2-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. 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. -- Technical Support http://www.telesip.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with libpri / asterisk
On 2/13/2012 10:49 AM, Andres wrote: -- Accepting call from '418nx2' to '418nx1' on channel 0/1, span 1 -- Executing [418nx1@ael-default:1] Answer(DAHDI/i1/418nx2-b, ) in new stack -- Executing [418nx1@ael-default:2] Wait(DAHDI/i1/418nx2-b, 2) in new stack -- Executing [418nx1@ael-default:3] Playback(DAHDI/i1/418nx2-b, demo-thanks) in new stack -- DAHDI/i1/418nx2-b Playing 'demo-thanks.ulaw' (language 'fr') -- Executing [418nx1@ael-default:4] Dial(DAHDI/i1/418nx2-b, DAHDI/G1/418nx2) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/418nx2 -- DAHDI/i1/418nx2-c is proceeding passing it to DAHDI/i1/418nx2-b -- DAHDI/i1/418nx2-c is ringing -- DAHDI/i1/418nx2-c is making progress passing it to DAHDI/i1/418nx2-b -- DAHDI/i1/418nx2-c answered DAHDI/i1/418nx2-b -- Native bridging DAHDI/i1/418nx2-b and DAHDI/i1/418nx2-c -- Span 1: Channel 0/1 got hangup request, cause 16 -- Hungup 'DAHDI/i1/418nx2-c' == Spawn extension (ael-default, 418nx1, 4) exited non-zero on 'DAHDI/i1/418nx2-b' -- Hungup 'DAHDI/i1/418nx2-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. -- Technical Support http://www.telesip.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with libpri / asterisk
(...) 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. My god ! That was it ! It was enabled, I had transfer = yes, but there was no mention of facilityenable. I disabled it, restarted asterisk, and voilà ! Thanks for pointing that out ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
