[asterisk-bugs] [JIRA] (ASTERISK-29955) chan_sip: SIP route header is missing on UPDATE

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29955?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259004#comment-259004 ] 

Joshua C. Colp commented on ASTERISK-29955:
-------------------------------------------

[~asterisk.org at zombie.dk] The tests accompanying these appear to be failing when run in the testsuite. Are you going to look into this? If not then we'll need to revert the changes.

>From the PJSIP test in Jenkins (https://jenkins2.asterisk.org/job/Asterisk%20Gates/job/19/447/artifact/tests/CI/output/pjsip2/logs/channels/pjsip/transfers/attended_transfer/nominal/callee_local_semi_attended_transfer_record_route/run_1/ast1/var/log/asterisk/full.txt):

{noformat}
[Apr 27 16:11:15] ERROR[1604] pjproject: 	       sip_transport.c Error processing 649 bytes packet from UDP 127.0.0.1:5068 : PJSIP syntax error exception when parsing 'Record-Route' header on line 3 col 15:
INVITE sip:alice at 127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5068;branch=z9hG4bK-1676-1-1
Record-Route: <sip:127.0.0.1:5068;transport=UDP;lr>
From: sipp <sip:alice at 127.0.0.1:5068>;tag=1676SIPpTag001
To: sut <sip:alice at 127.0.0.1:5060>
Call-ID: 1-1676 at 127.0.0.1
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Contact: sip:alice at 127.0.0.1:5068
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length:   130

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 43836 RTP/AVP 0
a=rtpmap:0 PCMU/8000

-- end of packet.
{noformat}

The chan_sip one is along the same lines. It could be a difference in SIPp version. Which were you using?

> chan_sip: SIP route header is missing on UPDATE
> -----------------------------------------------
>
>                 Key: ASTERISK-29955
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29955
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/Transfers
>    Affects Versions: 16.24.1, 18.10.1
>            Reporter: Mark Petersen
>            Assignee: Mark Petersen
>         Attachments: lunken.pcap
>
>
> if Asterisk need to send an UPDATE before answer on a channel that uses Record-Route: it will not include a Route header
> how to reproduce:
> A call B
> B put A on HOLD
> B call C
> C send 180 Ringing with a Record-Route header
> B refer A to C (while C is still ringing)
> note only newer phone will send the refer with the unanswered channel of C
> bug introduced with ASTERISK-24628
> that is trying to prevent set_destination from being set,
> but inadvertently also prevents add_route from being set



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