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
>
> --
> _____________________________________________________________________
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