[asterisk-users] No Audio after attended tranfer

[Alex Vishnev]

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.
>> --
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> 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
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