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

[Dmitry Kamovsky (JIRA)]

Dmitry Kamovsky created ASTERISK-23791:
------------------------------------------

             Summary: 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 don't understand contact header like <sip:uuid at server>. Could you tell me how can I transform Contact header to <sip:number at server> instad <sip:uuid at server>?

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





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