[asterisk-bugs] [JIRA] (ASTERISK-27963) PJSIP ignore Path header

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27963?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=244115#comment-244115 ] 

Asterisk Team commented on ASTERISK-27963:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> PJSIP ignore Path header 
> -------------------------
>
>                 Key: ASTERISK-27963
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27963
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 15.3.0
>         Environment: fedora server 27
>            Reporter: Slava Bendersky
>              Labels: pjsip
>
> PJSIP stack  INVITE not honoring Path header and not adding Route header to it. That cause send call to wrong directions. That quite critical issue
> Tested on asterisk version 15.3.
> ````
> <--- Transmitting SIP request (560 bytes) to UDP:10.30.100.41:5060 --->
> OPTIONS sip:101-1033 at 192.168.1.150:54642;transport=tls;rinstance=13DAEF9D SIP/2.0
> Via: SIP/2.0/UDP 10.30.100.27:5080;rport;branch=z9hG4bKPj99e2c53c-e091-46b1-80bc-894e989cf727
> From: <sip:101-1033 at 10.30.100.27>;tag=31e196f7-7997-4bc1-ab3b-1013c5f33811
> To: <sip:101-1033 at 192.168.1.150;rinstance=13DAEF9D>
> Contact: <sip:101-1033 at 10.30.100.27:5080>
> Call-ID: 4d4146b5-a0ce-4057-9d52-dc3ef3d4a526
> CSeq: 1826 OPTIONS
> Supported: path
> Route: <sip:101-1033 at 10.30.100.41;transport=udp;lr> ---> Follow PATH header ( correct )
> Max-Forwards: 70
> User-Agent: Asterisk PBX 15.3.0
> Content-Length: 0
> <--- Transmitting SIP request (967 bytes) to UDP:192.168.1.150:55089 --->
> INVITE sip:101-1033 at 192.168.1.150:55089;transport=tls;rinstance=13DAEF9D SIP/2.0
> Via: SIP/2.0/UDP 10.30.100.27:5080;rport;branch=z9hG4bKPj90c1d0eb-a580-407d-add1-fe167790b687
> From: "4039143" <sip:4039143 at 10.30.100.27>;tag=da646822-7ac1-48de-88a9-8efb44ab5a60
> To: <sip:101-1033 at 192.168.1.150;rinstance=13DAEF9D>
> Contact: <sip:asterisk at 10.30.100.27:5080>
> Call-ID: f1be92ad-f51d-43ec-ada3-7064cb1bf513
> CSeq: 30947 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub, path
> Session-Expires: 1800
> Min-SE: 90
> Max-Forwards: 70
> User-Agent: Asterisk PBX 15.3.0
> Content-Type: application/sdp
> Content-Length: 235
> v=0
> o=- 779031891 779031891 IN IP4 10.30.100.27
> s=Asterisk
> c=IN IP4 10.30.100.27
> t=0 0
> m=audio 14202 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv



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