|Summary:||ASTERISK-20230: Attended transfer using SIP Refer to a queue fails to play MOH|
|Date Opened:||2012-08-14 12:45:52||Date Closed:||2012-08-15 14:36:53|
|Environment:||Fedora 10, asterisk 188.8.131.52||Attachments:|
|Description:||Caller A calls through the PSTN to a queue Q1.|
SIP Peer B is logged in the queue Q1.
Call from A reaches peer B.
B answers the call from A.
B presses Xfer button in the Linksys Harphone (Transfer with SIP Refer)
A is on hold and listening to MOH.
B dials the number of another queue Q2, and starts to hear Q2`s MOH.
B presses Xfer button.
A is transfered to queue Q2 but stops to hear MOH. A should hear MOH but doesn´t.
|Comments:||By: David Woolley (davidw) 2012-08-15 05:24:00.670-0500|
Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. After testing with Asterisk 1.8, if you find this problem has not been resolved, please open a new issue against Asterisk 1.8.
Your sub-version of 1.6.2 was many bug fixes away from the final version, as well.
By: agustina (agustina) 2012-08-15 09:04:57.479-0500
Ok, can you validate if there is any chance this solve the problem:
"I solved changing chan_sip.c, moving the "ast_quiet_chan(peerb);" (line
14086) to go inside the "if(peerd)" if, right below, dont forget to put the
Thanks a a lot!!!
By: agustina (agustina) 2012-08-15 10:23:02.030-0500
I have also tested in asterisk 184.108.40.206 and happens the same problem!
By: Matt Jordan (mjordan) 2012-08-15 14:36:53.468-0500
As David said, support for 1.4 and 1.6.2 has ended.