[asterisk-dev] Asterisk 16. PJSIP. INVITE. "Contact" field and FQDN

[Mani Kanta Gadde]

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 <u2 at 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 at 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:XXYYZZ at 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:+380770081 at 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:6001 at ast.firma.org>;tag=9912223a-ff74-4ba6-8a0f-c3225e70eaba
> To: <sip:+380770081 at sip.pstnhub.microsoft.com>
> Contact: <sip:XXYYZZ at 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 at 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:XXYYZZ at ast.firma.org:5061;transport=TLS>
>
> Serg
> ?
>
>
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