Hi,
I have problem with Dialing Out using the trunk. Unfortunatelly, have no support from operator in case of SIP conversation.
When I trying to call outside, I receive unknown phone number.
In My opinion, my header does not contain number in INVITE command.
How can I put ii? Communication looks like:
[2025-04-10 15:04:26] VERBOSE[160848] res_pjsip_logger.c: <— Transmitting SIP request (1222 bytes) to UDP:195.xxx.xxx.xxx:5060 —>
INVITE sip:195.xxx.xxx.xxx SIP/2.0^M
Via: SIP/2.0/UDP 212.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPjc4d6a5a0-984f-4da8-8665-1636f518cee5^M
From: sip:[email protected];tag=f2661dbb-700d-479c-a353-73c7dfee3178^M
To: sip:[email protected]^M
Contact: sip:[email protected]:5060^M
Call-ID: 4a6bb29a-4860-4c11-85f6-eb5f379471df^M
CSeq: 7310 INVITE^M
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INFO, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub, histinfo^M
Session-Expires: 1800^M
Min-SE: 90^M
Max-Forwards: 70^M
User-Agent: FPBX-17.0.19.24(22.1.0)^M
Proxy-Authorization: Digest username=“motorstest”, realm=“195.xxx.xxx.xxx”, nonce=“67f7c1f8ebb02eb134d18ff408f2ff6fbbed07b0”, uri=“sip:195.xxx.xxx.xxx”, response=“cba5c86aa76f83b40d52df8f5dee2c7d”^M
Route: sip:[email protected]:5060^M
Content-Type: application/sdp^M
Content-Length: 288^M
^M
v=0^M
o=- 343881126 343881126 IN IP4 212.xxx.xxx.xxx^M
s=Asterisk^M
c=IN IP4 212.xxx.xxx.xxx^M
t=0 0^M
m=audio 10812 RTP/AVP 9 18 101^M
a=rtpmap:9 G722/8000^M
a=rtpmap:18 G729/8000^M
a=fmtp:18 annexb=no^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=maxptime:140^M
a=sendrecv^M
[2025-04-10 15:04:26] VERBOSE[1746] res_pjsip_logger.c: <— Received SIP response (401 bytes) from UDP:195.xxx.xxx.xxx:5060 —>
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 212.xxx.xxx.xxx:5060;rport=5060;branch=z9hG4bKPjc4d6a5a0-984f-4da8-8665-1636f518cee5;received=212.xxx.xxx.xxx^M
From: sip:[email protected];tag=f2661dbb-700d-479c-a353-73c7dfee3178^M
To: sip:[email protected]^M
Call-ID: 4a6bb29a-4860-4c11-85f6-eb5f379471df^M
CSeq: 7310 INVITE^M
Server: Kamailio (1.4.4-notls (i386/linux))^M
Content-Length: 0^M
^M
[2025-04-10 15:04:26] VERBOSE[1746] res_pjsip_logger.c: <— Received SIP response (479 bytes) from UDP:195.xxx.xxx.xxx:5060 —>
SIP/2.0 404 Not Found^M
Via: SIP/2.0/UDP 212.xxx.xxx.xxx:5060;rport=5060;branch=z9hG4bKPjc4d6a5a0-984f-4da8-8665-1636f518cee5^M
From: sip:[email protected];tag=f2661dbb-700d-479c-a353-73c7dfee3178^M
To: sip:[email protected];tag=as62d9f9da^M
Call-ID: 4a6bb29a-4860-4c11-85f6-eb5f379471df^M
CSeq: 7310 INVITE^M
Server: HiperusC5^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M
Supported: replaces, timer^M
Content-Length: 0^M