Use the external_media_address and external_signalling_address in PJSIP in general section.
These variables will explicitly put the required IP in SIP messages, so that other SIP phone/VoIP server know where to reply. Thanks & Regards Manikanta On Tue, Oct 30, 2018 at 10:08 AM KoltogyanU2 SergeyU2 <u...@amintegrator.com> wrote: > > PJSIP . How to force use FQDN in the "Contact" field ( INVITE) ? > > In the INVITE the "Contact" field looks like this: > Contact: <sip:XXYYZZ@11.22.33.44:5061;transport=TLS> > > How to reconfigure Asterisk, or where in the source code to make a change, > so that the "Contact" always use FQDN =ast.firma.org and looked like > this: > Contact: <sip:xxy...@ast.firma.org:5061;transport=TLS> > > ? > > Description of the problem: > Asterisk 16 (use PJSIP. asterisk build with: > ./configure --with-pjproject-bundled -sysconfdir=/etc --libdir=/usr/lib64 > ) > > Asterisk sends a INVITE to the sip.pstnhub.microsoft.com in this form: > <--- Transmitting SIP request (806 bytes) to TLS:52.114.75.24:5061 ---> > INVITE sip:+380770...@sip.pstnhub.microsoft.com:5061 SIP/2.0 > Via: SIP/2.0/TLS 11.22.33.44:5061 > ;rport;branch=z9hG4bKPjd2417f6c-8788-4d40-b666-3244b903d886;alias > From: <sip:6...@ast.firma.org>;tag=9912223a-ff74-4ba6-8a0f-c3225e70eaba > To: <sip:+380770...@sip.pstnhub.microsoft.com> > Contact: <sip:XXYYZZ@11.22.33.44:5061;transport=TLS> > Call-ID: ee581ee7-e624-41cb-a486-b06cf233c63c > CSeq: 19204 INVITE > Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, > CANCEL, UPDATE, PRACK, MESSAGE, REFER > Supported: 100rel, timer, replaces, norefersub > Session-Expires: 1800 > Min-SE: 90 > Max-Forwards: 70 > User-Agent: Asterisk PBX 16.0.0 > Content-Type: application/sdp > Content-Length: 92 > > v=0 > o=- 233177990 233177990 IN IP4 11.22.33.44 > s=Asterisk > c=IN IP4 40.127.205.7 > t=0 0 > > > Where 11.22.33.44 - Asterisk public IP Address ( Asterisk over NAT ): > Asterisk(172.18.1.16)--->NAT(11.22.33.44)---->ISP > > In the INVITE the "Contact" field looks like this: > Contact: <sip:XXYYZZ@11.22.33.44:5061;transport=TLS> > > How to reconfigure Asterisk, or where in the source code to make a change, > so that the "Contact" always use FQDN =ast.firma.org and looked like > this: > Contact: <sip:xxy...@ast.firma.org:5061;transport=TLS> > > Serg > ? > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
