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

Oliver Nauliv (JIRA) noreply at issues.asterisk.org
Tue Feb 9 16:54:33 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25741?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229352#comment-229352 ] 

Oliver Nauliv commented on ASTERISK-25741:
------------------------------------------

Hello Joshua,

Thanks again for responding to the message. Here are 2 short questions, taking aside the SIP RFC for just a moment. 

1) Why do all the other Asterisk PBX (not using PJSIP) don't garble the Contact field ? And can we just have an option to keep the Contact field intact the way it has always been ?

2) Why is the P-Asserted-Identity not working? I did everything suggested on https://community.asterisk.org/t/asterisk-13-pjsip-contact-field-in-sip-is-randomized/65044 and packaging it into a pre-dialer, but the field is still not showing up.

Please advise. The carrier and I have been working on this issue for 20+ combined hours... we are both willing to work with the Asterisk team to help get this resolved and help the community, as apparently other people are reporting similar issues on the forums.

Is there an e-mail I can send the full unedited SIP trace, as it contains private information ?

Thanks!

> 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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