[asterisk-bugs] [JIRA] (ASTERISK-27963) res_pjsip: ignore Path header on INVITE

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27963?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-27963:
-----------------------------------

    Assignee: Slava Bendersky
      Status: Waiting for Feedback  (was: Triage)

Are you using PJSIP_DIAL_CONTACTS? What is the complete console output?

> res_pjsip: ignore Path header on INVITE
> ---------------------------------------
>
>                 Key: ASTERISK-27963
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27963
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 15.3.0
>         Environment: fedora server 27
>            Reporter: Slava Bendersky
>            Assignee: 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.
> {noformat}
> <--- 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
> {noformat}
> The Route header is missing in the outgoing INVITE:
> {noformat}
> <--- 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
> {noformat}



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