asterisk users - Jul 2011 - No Audio after attended tranfer

I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an
attended transfer. The transfer is going to an outbound number (normally AA on
another IP PBX). the audio on the first transfer is fine. But if the user
requests a transfer from AA to another department, I loose audio from Asterisk
to the 2nd transfer. Based on the review of SIP packets, the second transfer
issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not
being handled properly by asterisk. I searched thru JIRA cases, but did not find
anything like that. Any help would be appreciated.

Am 18.07.11 16:15, schrieb Alex Vishnev:
> I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing
an attended transfer. The transfer is going to an outbound number (normally AA
on another IP PBX). the audio on the first transfer is fine. But if the user
requests a transfer from AA to another department, I loose audio from Asterisk
to the 2nd transfer. Based on the review of SIP packets, the second transfer
issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not
being handled properly by asterisk. I searched thru JIRA cases, but did not find
anything like that. Any help would be appreciated.
> --
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Hello,

maybe this is the problem you have:

https://issues.asterisk.org/jira/browse/ASTERISK-18136

best regards

Stefan

No, that looks like a separate issue. Mine is a 100% repeatable and the asterisk
does not lock up. SIP and RTP on other sessions are still going. in my cases
this is the exchange I see

Asterisk                                                                        
Service Provider
INVITE (initial Invite to Service Provider with Outbound number) ------->
<----------------------200 OK
<---------------------INVITE (put session on hold)
---------------------->200OK 
<----------------------ACK
<--------------------->RTP
<<---------------------INVITE (no SDP) -- First transfer complete
---------------------->200OK (SDP)
<----------------------ACK
<--------------------->RTP
<<---------------------INVITE (no SDP) -- Second Transfer
---------------------->200OK (SDP)
<----------------------ACK (SDP)
<----------------------RTP

On Jul 19, 2011, at 3:41 AM, Stefan Schmidt wrote:
> Am 18.07.11 16:15, schrieb Alex Vishnev:
>> I am wondering if anyone hit this case yet. I am using 1.6.2.19 and
doing an attended transfer. The transfer is going to an outbound number
(normally AA on another IP PBX). the audio on the first transfer is fine. But if
the user requests a transfer from AA to another department, I loose audio from
Asterisk to the 2nd transfer. Based on the review of SIP packets, the second
transfer issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP
is not being handled properly by asterisk. I searched thru JIRA cases, but did
not find anything like that. Any help would be appreciated.
>> --
>> _____________________________________________________________________
>> -- 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
> Hello,
> 
> maybe this is the problem you have:
> 
> https://issues.asterisk.org/jira/browse/ASTERISK-18136
> 
> best regards
> 
> Stefan
> 
> --
> _____________________________________________________________________
> -- 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