| Summary: | ASTERISK-03617: MOH stopped working properly in CVS Head | ||
| Reporter: | Trevor Hammonds (trevmeister) | Labels: | |
| Date Opened: | 2005-03-02 05:39:31.000-0600 | Date Closed: | 2008-01-15 15:26:41.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Resources/res_musiconhold |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) mydiff.txt | |
| Description: | When using either the hold, transfer, or conference buttons on my SIP phones, the party on the other half of the call does not hear MOH as expected, only silence. When using the # transfer function, the caller DOES hear MOH as expected. I have set up an extension with MusicOnHold(default). When calling that extension, I hear MOH as expected. I am experiencing the same symptoms as described in the following message, except I am using hard SIP phones: http://lists.digium.com/pipermail/asterisk-users/2005-February/091850.html ****** ADDITIONAL INFORMATION ****** I am running both CVS Head and Stable releases of Asterisk. The problem appeared after updating (CVS checkout and recompile) my servers running Asterisk CVS Head on 2/28/05, and persists after having done so again today. MOH was working normally on the CVS Head servers up until the update. The servers running Stable releases are not affected. As there were no major changes made immediately before or after the update, I suspect this to be a bug in CVS Head. | ||
| Comments: | By: paradise (paradise) 2005-03-02 07:15:57.000-0600 the same problem with Stable branch of CVS! the problem appeared after CVS stable 2/26 By: paradise (paradise) 2005-03-02 07:34:21.000-0600 found the problem! it's due to change of chan_sip.c according to bug ASTERISK-3580 1.0.6 is affected too! edited on: 03-02-05 08:38 By: Mark Spencer (markster) 2005-03-02 10:05:49.000-0600 Fixed in CVS By: Kevin P. Fleming (kpfleming) 2005-03-02 10:05:56.000-0600 Hmm... It's not clear to me how that could make any difference, since it only affects incoming INVITEs from the SIP phones, but I'll trust that your debugging confirms that change is causing the problem :-) By: Russell Bryant (russell) 2005-03-02 11:45:42.000-0600 fixed in 1.0 ... By: Digium Subversion (svnbot) 2008-01-15 15:26:39.000-0600 Repository: asterisk Revision: 5118 U trunk/channels/chan_sip.c ------------------------------------------------------------------------ r5118 | markster | 2008-01-15 15:26:38 -0600 (Tue, 15 Jan 2008) | 2 lines Be sure to process SDP if we already have an owner (bug ASTERISK-3617) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=5118 By: Digium Subversion (svnbot) 2008-01-15 15:26:41.000-0600 Repository: asterisk Revision: 5120 U branches/v1-0/channels/chan_sip.c ------------------------------------------------------------------------ r5120 | russell | 2008-01-15 15:26:40 -0600 (Tue, 15 Jan 2008) | 2 lines Be sure to process SDP if we already have an owner (bug ASTERISK-3617) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=5120 | ||