Summary:	ASTERISK-16046: MOH not playing when phone transfer softkey pressed
Reporter:	Sergio Charrua (scharrua)	Labels:
Date Opened:	2010-05-03 18:23:43	Date Closed:	2010-05-10 11:19:22
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Transfers
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Hi! the client's setup is as follows: digium 1FX0 1FXs card on dahdi 2.2.0, Asterisk 1.4.27, 1 Yealink T28P amd 5 Yealink T22P, 1 GSM Gateway 1SIM card the problem is: caller A (from mobile or fixed line phone) calls client. A RingGroup is set and correctly passes call to the extensions. Operator (T28P) extension B answers the call and puts caller on hold and transfers call to extension C. While on hold, caller A should hear MOH, but, despite being correctly set, no music is started, nothing is heard. On CLI, when operator presses the Transfer softkey on the phone, nothing appears, suggesting that there is no SIP request for transfer, whatsoever. Strangely, the same setup but using a OpenVox B200P (2BRI card) and mISDN drivers does everything correctly. This is the only diference between the 2 clients. i've checked musiconhold.conf, features.conf, extensions.conf, everything seems correctly set!
Comments:	By: Paul Belanger (pabelanger) 2010-05-04 10:36:06 While your description is detailed, we require debug logs, dialplans and .config file to reproduce the issue. http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We\'d be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. the specific steps or actions you took that caused you to encounter the problem, 2. the behavior you expected, and 3. the behavior you actually encountered (in as much detail as possible). This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks! By: Paul Belanger (pabelanger) 2010-05-04 10:36:58 You also referenced 1.4.27 in your description, but set 1.4.30 which one is correct? By: Leif Madsen (lmadsen) 2010-05-07 10:37:12 Console is also required in order to see what is going on. If this is SIP, you need the SIP trace which shows the full call flow. By: Sergio Charrua (scharrua) 2010-05-07 10:44:49 sorry for the late answer. i erroneously referenced the 1.4.30, it was in fact 1.4.27. i followed your instructions, enabled Logger and Sip debug, etc... and indeed, the log file doesn't show any request for Attended Transfer or Hold (unless operator uses *8). I did solved the problem by updating to 1.4.31. I only updated Asterisk code, haven't touched ANY of the .conf files, and everything is now working perfectly. So i suspect it might be something with chan_sip in 1.4.27 (just a wild guess...) I will attach the requested files ASAP. Please note that i'm using Asterisk GUI lastest SVN version with many mods coded by me, but none is relevant to MOH or to this particular case. Thanks once again. Sergio Charrua www.voip.pt - Voice Over IP Portugal By: Leif Madsen (lmadsen) 2010-05-10 11:19:22 Closing this as fixed in 1.4.31 per the reporter. Thanks for the feedback!