|Summary:||ASTERISK-19576: DTMF Passed Unreliably from DAHDI Analog to GTalk|
|Reporter:||Vladimir Mikhelson (vmikhelson)||Labels:|
|Date Opened:||2012-03-22 20:22:47||Date Closed:||2012-04-24 14:40:20|
|Environment:||CentOS 5.7, FreePBX 2.10||Attachments:|
|Description:||If a call is originated on an analog phone connected via TDM400/FXS and is routed via a GTalk cahnnel DTMF shows reliably in Debug mode on the console but a remote side does not register all tones passed.|
If a call is placed from a SIP extension or via an ATA and is routed via GTalk all DTMF tones are processed as they should.
If a call is originated on an analog phone connected via TDM400/FXS and is routed via a SIP, IAX or a POTS line all DTMF tones are processed as they should.
|Comments:||By: Matt Jordan (mjordan) 2012-03-23 09:03:27.002-0500|
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: Matt Jordan (mjordan) 2012-04-24 14:40:12.669-0500
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 http://www.asterisk.org/developers/bug-guidelines