[asterisk-bugs] [Asterisk 0018936]: When transferring a call using the REFER command with replaces enabled, Asterisk drops the call.
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Mar 8 07:16:18 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18936
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Reported By: ltc
Assigned To:
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Project: Asterisk
Issue ID: 18936
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!): 309808
Request Review:
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Date Submitted: 2011-03-07 14:10 CST
Last Modified: 2011-03-08 07:16 CST
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Summary: When transferring a call using the REFER command
with replaces enabled, Asterisk drops the call.
Description:
When transferring a call using the REFER command with replaces enabled,
Asterisk drops the call with "SIP/2.0 481 Call leg/transaction does not
exist". I have tried performing this on 1.4 SVN, 1.8.3, Trunk SVN 309404,
and 1.8 SVN 309808. All have produced the same result. I have attached the
debug logs with core and sip debugging on. Unfortunately I do not have a
way to perform a successful transfer with REFER to compare against. Line
432 is where the REFER starts. Chan_sip.c starts to process the REFER but
then later drops it on line 538 with the "SIP/2.0 481 Call leg/transaction
does not exist".
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(0132718) ltc (reporter) - 2011-03-08 07:16
https://issues.asterisk.org/view.php?id=18936#c132718
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The only part of the trace that was excluded was the initial phone call.
Debugging was turned on after the call was established. I can post the
entire log if requested.
In this trace, the call was initiated by an extension on Asterisk (ext
1820) to extension 1173 on another IP-PBX. Once the call is established,
extension 1173 then attempts to transfer the call to another extension
(1265) on the same PBX. The REFER message is notifying Asterisk and 1820
that the transfer is to take place and that 1820 should now contact 1265.
The problem also occurs if extension 1173 initiates the phone call and then
transfers.
In the debug I see the transferor sending the Replaces header field in the
Refer-to header but it appears that Asterisk is trying to bridge channels
together instead of sending the invite to the target.
Issue History
Date Modified Username Field Change
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2011-03-08 07:16 ltc Note Added: 0132718
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