Summary: | ASTERISK-20257: chan_motif: no audio to Google Talk when SIP phone uses G722 | ||
Reporter: | Artem Makhutov (artem) | Labels: | |
Date Opened: | 2012-08-19 05:03:09 | Date Closed: | 2012-09-19 20:34:35 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_motif |
Versions: | 11.0.0-beta1 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Attachments: | ||
Description: | motif.conf: [default](!) disallow=all allow=alaw allow=ulaw allow=h264 context=incoming-motif [google](default) transport=google connection=google sip.conf: [xxxxx] defaultuser=xxxxx directmedia=no disallow=all allow=g722 allow=alaw allow=ulaw allow=h264 When the SIP Phone negotiates G722 with asterisk then there is no audio in both directions between the SIP Phone and the Google Talk client. It looks like asterisk is not performing audio transcoding. Enabling G722 in motif.conf also results in having no audio. Only disabling G722 in sip.conf helps here. | ||
Comments: | By: Michael L. Young (elguero) 2012-08-19 13:33:40.528-0500 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Rusty Newton (rnewton) 2012-09-19 20:34:26.694-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |