[asterisk-bugs] [JIRA] (ASTERISK-23791) PJSIP use UUID in Contact header when Dial

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23791?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-23791:
------------------------------------

    Assignee: Matt Jordan  (was: Dmitry Kamovsky)
      Status: Triage  (was: Waiting for Feedback)

> PJSIP use UUID in Contact header when Dial
> ------------------------------------------
>
>                 Key: ASTERISK-23791
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23791
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 12.2.0
>         Environment: CentOS 6.5
>            Reporter: Dmitry Kamovsky
>            Assignee: Matt Jordan
>         Attachments: ASTERISK-23791_full_log, ASTERISK-23791_full_log.txt, ps_endpoint_id_ips.txt, ps_endpoint_id_ips.txt, view_ps_aors.txt, view_ps_aors.txt, view_ps_endpoints.txt, view_ps_endpoints.txt
>
>
> pjsip use UUID in contact header of INVITE message. This leads to core crashing when I try to redirect call, because softphone sfl-phone doesn't understand contact header like {{<sip:uuid at server>}}. Could you tell me how can I transform Contact header to {{<sip:number at server>}} instead {{<sip:uuid at server>}}?
> {code}
> INVITE sip:2302033 at 172.16.99.2 SIP/2.0
> Record-Route: <sip:172.16.99.2;lr;ftag=2394a326f2575a8co0;did=2b3.3238d645>
> Via: SIP/2.0/UDP 172.16.99.2:5060;branch=z9hG4bK282f.b2f7cb54.0
> Via: SIP/2.0/UDP 172.16.99.101:5064;branch=z9hG4bK-9113c249
> From: "sip123456" <sip:sip123456 at 172.16.99.2>;tag=2394a326f2575a8co0
> To: <sip:2302033 at 172.16.99.2>
> Call-ID: a8a2baae-de70e8a4 at 172.16.99.101
> CSeq: 102 INVITE
> Max-Forwards: 69
> *Contact: "sip123456" <sip:sip123456 at 172.16.99.101:5064>*
> Expires: 240
> User-Agent: Linksys/SPA921-5.1.8
> Content-Length: 399
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
> Content-Type: application/sdp
> v=0
> o=- 1958139 1958139 IN IP4 172.16.99.101
> s=-
> c=IN IP4 172.16.99.101
> t=0 0
> m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
> <--- Transmitting SIP response (458 bytes) to UDP:172.16.99.2:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.16.99.2:5060;rport;received=172.16.99.2;branch=z9hG4bK282f.b2f7cb54.0
> Via: SIP/2.0/UDP 172.16.99.101:5064;branch=z9hG4bK-9113c249
> Record-Route: <sip:172.16.99.2;lr;ftag=2394a326f2575a8co0;did=2b3.3238d645>
> Call-ID: a8a2baae-de70e8a4 at 172.16.99.101
> From: "sip123456" <sip:sip123456 at 172.16.99.2>;tag=2394a326f2575a8co0
> To: <sip:2302033 at 172.16.99.2>
> CSeq: 102 INVITE
> Server: Asterisk PBX 12
> Content-Length:  0
> ..........
>     -- Executing [2302033 at outgoing:7] Set("PJSIP/sip123456-00000008", "CALLERID(num)=4232050001") in new stack
>     -- Executing [2302033 at outgoing:14] Dial("PJSIP/sip123456-00000008", "PJSIP/4232302033 at opensips,60,TM(utm5^start^sip123456^7ca2bee8-a884-4cb1-a8ca-66a7c4136b3f^4232050001^4232302033)") in new stack
> <--- Transmitting SIP request (965 bytes) to UDP:172.16.99.2:5060 --->
> INVITE sip:4232302033 at 172.16.99.2:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.16.99.9:5060;rport;branch=z9hG4bKPj65c4ed8a-35cd-4ee5-a9d2-ca5f6f7bf6eb
> From: "sip123456" <sip:4232050001 at 172.16.99.9>;tag=e8843033-1e78-44e9-bd3a-ec97c7a4dd25
> To: <sip:4232302033 at 172.16.99.2>
> *Contact: <sip:310fc49d-b1a9-448b-b8bc-92530f07dd26 at 172.16.99.9:5060>*
> Call-ID: 012a4451-d3ba-45a7-96a1-5aa17c7cb89c
> CSeq: 9163 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Max-Forwards: 70
> User-Agent: Asterisk PBX 12
> Content-Type: application/sdp
> Content-Length:   267
> v=0
> o=- 1570975488 1570975488 IN IP4 localhost.localdomain
> s=Asterisk
> c=IN IP4 172.16.99.9
> t=0 0
> m=audio 17522 RTP/AVP 8 101
> c=IN IP4 172.16.99.9
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> {code}



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