|Summary:||ASTERISK-11186: Attended transfer completes but keeps the zap channel with music on hold|
|Reporter:||Matheus Rossato (matheusrossato)||Labels:|
|Date Opened:||2008-01-09 12:01:29.000-0600||Date Closed:||2011-06-07 14:02:57|
|Environment:||Attachments:||( 0) sipdebug_1_4_4.txt|
( 1) sipdebug.txt
|Description:||When doing an attended transfer in Asterisk the zap channel with the customer is placed on hold, the operator talks with the closer. However when the operator sends the call to the closer, the call is brigded but the customer in the zap channel keeps in the music on hold. Attached is a SIP Debug from asterisk where you can see that the music on hold only stops on zap channel after the call is hangup.|
|Comments:||By: Joshua C. Colp (jcolp) 2008-01-09 12:05:35.000-0600|
A step by step list of what is being done and by whom would be extremely useful.
By: Matheus Rossato (matheusrossato) 2008-01-09 12:40:03.000-0600
Asterisk(172.25.1.226) is the gateway for genesys sip server(172.20.0.38). The 106 is an operator in genesys that placed a call to a customer 00111111004884195169008441311. When the call is answered by the customer the operator transfers the call to the closer, however the transfer is mainly managed by genesys sip server. This operation works fine with Asterisk 1.4.5 and earliers, from Asterisk 1.4.13 to 1.4.17 it has this behavior. Also I attached a sip debug with asterisk 1.4.4 for the same operation.
Let me know if you need more info.
By: Matheus Rossato (matheusrossato) 2008-01-23 12:06:59.000-0600
Hi all, any estimative when this issue can be addressed?
By: jmls (jmls) 2008-05-03 14:18:34
matheusrossato is this still a problem with the latest 1.4 svn release ?
By: Russell Bryant (russell) 2008-05-12 15:02:06
I am suspending this issue for now. The debug output shows that the version of Asterisk is 1.4.4, which is _VERY_ out of date. Please try with the latest version, and if you still have a problem, feel free to reopen this issue.