[asterisk-bugs] [JIRA] (ASTERISK-25741) res_pjsip: "Contact" contains UUID for user portion

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25741?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-25741:
-----------------------------------

    Assignee: Oliver Nauliv
      Status: Waiting for Feedback  (was: Triage)

According to the SIP RFC:

The Contact header field provides a SIP or SIPS URI that can be used
   to contact that specific instance of the UA for subsequent requests.
   The Contact header field MUST be present and contain exactly one SIP
   or SIPS URI in any request that can result in the establishment of a
   dialog.  For the methods defined in this specification, that includes
   only the INVITE request.  For these requests, the scope of the
   Contact is global.  That is, the Contact header field value contains
   the URI at which the UA would like to receive requests, and this URI
   MUST be valid even if used in subsequent requests outside of any
   dialogs.

Within Asterisk this holds true, and the provider in question shouldn't derive any meaning from the Contact header itself. The fact that we use a UUID for the Contact user portion shouldn't matter. Have they provided any reasoning?

As for your second thing that's not a bug, and has been answered on http://community.asterisk.org/

> res_pjsip: "Contact" contains UUID for user portion
> ---------------------------------------------------
>
>                 Key: ASTERISK-25741
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25741
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 13.1.0
>         Environment: CentOS 2.6.32-504.23.4.el6.i686
>            Reporter: Oliver Nauliv
>            Assignee: Oliver Nauliv
>            Severity: Minor
>
> 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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