[asterisk-bugs] [Asterisk 0014244]: No Audio on Call Transfer (Invite not being forwarded to Provider via Asterisk)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jun 2 13:50:12 CDT 2010
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=14244
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Reported By: mbnwa
Assigned To:
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Project: Asterisk
Issue ID: 14244
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: SVN
JIRA: SWP-19
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-01-14 18:13 CST
Last Modified: 2010-06-02 13:50 CDT
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Summary: No Audio on Call Transfer (Invite not being
forwarded to Provider via Asterisk)
Description:
Notes:
Asterisk 1.4.18 & Asterisk 1.6.x effected
Running directrtpsetup=yes
OS Debian
Kernel Version 2.6.26-1-amd64
Called number 13605551212
Caller's number 13605551211
Extension to get transfer: 13605551210
Caller z.z.z.z
Asterisk Server x.x.x.x
Carrier y.y.y.y
Call Flow
13605551211 calls 13605551212 makes the transfer to 13605551210 at this
point call is direct between 13605551212 and 13605551210 but no audio
Issue:
Phone 1 makes an outbound call then transfers to another extension invite
is sent from phone 1 to asterisk, Asterisk ack's however it never sends
the invite to the carrier to update the audio path resulting in no audio
SIP Trace
(lmadsen: I have removed the inline call trace, and placed it in a text
file as 'original-call-trace.txt' and attached it to this issue. Long
inline traces make open issues difficult to work with.)
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----------------------------------------------------------------------
(0122827) pabelanger (manager) - 2010-06-02 13:50
https://issues.asterisk.org/view.php?id=14244#c122827
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Please upgrade to the 1.4 or 1.6.2 branch and retest your issue. If the
problem still exists attach a debug log (see below) and sample
extensions.conf to reproduce the issue.
---
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:
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
Issue History
Date Modified Username Field Change
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2010-06-02 13:50 pabelanger Note Added: 0122827
2010-06-02 13:50 pabelanger Status acknowledged =>
feedback
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