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

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23791?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=218487#comment-218487 ] 

Matt Jordan commented on ASTERISK-23791:
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We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

I'd like to make sure we have a complete understanding of the generation of the Contact header in this case. Please make sure you have {{pjsip set logger on}} in the log file. Please also attach your {{pjsip.conf}}. Thanks!

> 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
>
> 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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