[asterisk-bugs] [JIRA] (ASTERISK-25741) "Contact" in sip INVITES is garbled

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25741?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229259#comment-229259 ] 

Asterisk Team commented on ASTERISK-25741:
------------------------------------------

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].

> "Contact"  in sip INVITES is garbled
> ------------------------------------
>
>                 Key: ASTERISK-25741
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25741
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 13.1.0
>         Environment: CentOS 2.6.32-504.23.4.el6.i686
>            Reporter: Oliver Nauliv
>
> Hello,
> Using Asterisk 13.1-cert2 and PJSIP, with Nextiva as the SIP Trunk provider. The carrier says that there are 2 reasons why dialing out is not working :
> 1) If you look at the SIP trace below, the carrier is indicating that the reason for not working is because the "Contact:" line has a cryptic-randomized number in there, instead of the actual contact.
> It sends:
> Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 107.207.90.90:5060>
> Instead of sending:
> Contact: <sip:7143861212 at 107.207.90.90:5060>
> 2) We are unable to send the X-P-Asserted-Identity, because while the dialplan runs PJSIP_HEADER without error, the option doesn't make it into the SIP packet.
> ******** SIP TRACE FOR OUTBOUND CALL **********
> <--- Transmitting SIP request (891 bytes) to UDP:208.73.140.70:5060 --->
> INVITE sip:9492345566 at 208.73.140.70:5060 SIP/2.0
> Via: SIP/2.0/UDP 107.207.90.90:5060;rport;branch=z9hG4bKPja-5qMYTknEobutnWKzOTPpSrCg3swCz
> From: "Manager Office"
> <sip:7143861212 at 107.207.90.90>;tag=slIrplpaQV6S.tpB64u4BEbK-.7HBEis
> To: <sip:9492345566 at 208.73.140.70>
> Contact: <sip:ec89dc3e-90df-4fbd-99bd-f5b225f8da3d at 107.207.90.90:5060>
> Call-ID: QmF69h0tpNPtZhE6IrwYqtoQ67kcJ5pn
> CSeq: 9147 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE,
> PRACK, REFER, REGISTER, MESSAGE
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 237
> v=0
> o=- 455864548 455864548 IN IP4 107.207.90.90
> s=Asterisk
> c=IN IP4 107.207.90.90
> t=0 0
> m=audio 19628 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
> ******** DIALPLAN FOR OUTBOUND CALL *********************
> [dial-out]
> exten => _X.,1,NoOp()
> same => n,Set(PJSIP_HEADER(add,X-P-Asserted-Identity)=sip:7147890101)
> same => n,Set(CALLERID(num)=7143861212)
> same => n,Dial(PJSIP/${EXTEN}@nextiva1,45,r)
> same => n,Hangup()
> ******** PJSIP CONFIGURATION FOR SIP TRUNK ***********
> [nextiva1]
> type=endpoint
> transport=transport-udp
> context=from-nextiva
> disallow=all
> allow=ulaw
> aors=nextiva1-aor
> [nextiva1-auth]
> type=auth
> auth_type=userpass
> username=7143861212
> password=NotMyRealPassword
> [nextiva1-reg]
> type=registration
> outbound_auth=nextiva1-auth
> server_uri=sip:bt.voipdnsservers.com
> client_uri=sip:7143861212 at bt.voipdnsservers.com
> [nextiva1]
> type=identify
> endpoint=nextiva1
> match=208.73.140.70
> [nextiva1-aor]
> type=aor
> contact=sip:bt.voipdnsservers.com:5060



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