[asterisk-bugs] [Asterisk 0019337]: Call shows on hold after attended transfer with a Polycom phone
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed May 25 10:05:37 CDT 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=19337
======================================================================
Reported By: remiq
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 19337
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA: SWP-3492
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.8
SVN Revision (number only!): 319938
Request Review:
======================================================================
Date Submitted: 2011-05-20 10:33 CDT
Last Modified: 2011-05-25 10:05 CDT
======================================================================
Summary: Call shows on hold after attended transfer with a
Polycom phone
Description:
When I do an attended transfer from a Polycom IP650 the call is
transferring successfully, but the call is not releasing properly on the
phone that is initiating the transfer. Instead the call shows that it is
on hold.
======================================================================
----------------------------------------------------------------------
(0135381) remiq (reporter) - 2011-05-25 10:05
https://issues.asterisk.org/view.php?id=19337#c135381
----------------------------------------------------------------------
After further investigation it looks like the SIP proxy is not liking the
BYE message after the transfer, it's returning with a '400 SIP Parser
Error':
BYE sip:322-eng at 209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0^M
Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229eab44^M
Max-Forwards: 70^M
From: <sip:312 at 64.19.145.13;user=phone>;tag=as0868ad46^M
To: "Poly_test ENG"<sip:322-eng at 64.19.145.13>;tag=E7EA8417-AA13A95A^M
Call-ID: dd352991-ef95b5a4-7585dccf at 10.0.15.105^M
CSeq: 102 BYE^M
User-Agent: Asterisk PBX SVN-branch-1.8-r319997^M
Proxy-Authorization: Digest username="322-eng", realm="asterisk",
algorithm=MD5, uri="64.19.145.13", nonce="",
response="eac3218b89666699bb97133fa8966982"^M
X-Asterisk-HangupCause: Normal Clearing^M
X-Asterisk-HangupCauseCode: 16^M
Content-Length: 0^M
<--- SIP read from UDP:209.191.39.117:5060 --->
SIP/2.0 400 SIP Parser Error : Unexpected '\"', line 9, column 99
From: <sip:312 at 64.19.145.13;user=phone>;tag=as0868ad46
To: "Poly_test ENG"<sip:322-eng at 64.19.145.13>;tag=E7EA8417-AA13A95A
Call-ID: dd352991-ef95b5a4-7585dccf at 10.0.15.105
CSeq: 102 BYE
Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK229eab44
Max-Forwards: 70
User-Agent: Asterisk PBX SVN-branch-1.8-r319997
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Proxy-Authorization: Digest username="322-eng" realm="asterisk"
algorithm=MD5 uri="64.19.145.13", nonce="",
response="eac3218b89666699bb97133fa8966982"
Content-Length: 0
<------------->
Any clue why?
Issue History
Date Modified Username Field Change
======================================================================
2011-05-25 10:05 remiq Note Added: 0135381
======================================================================
More information about the asterisk-bugs
mailing list