Asterisk sending INVITE on behalf of my softphone (endpoint) and put own private IP in "From" header

Asterisk Asterisk SIP
1

I have an Asterisk behind the NAT with ip 172.17.0.2 and 2 endpoints connected to him 192.168.12.101 (softphone) and 192.168.12.100 (mobile). When I calling mobile from softphone I wondered because I see incoming call on mobile is:
softphone@172.17.0.2
What is my Asterisk private IP. I enabled “pjsip set logger on” and write down who actually doing what:

Softphone                     ASTERISK                        Mobile
192.168.12.101        192.168.12.101-nat-172.17.0.2       192.168.12.100      
+-------------------------------+------------------------------+
|                               |                              |
|INVITE mobile                  |                              |
+------------------------------>+                              |
|                               |                              |
| 401 Unathorized               |                              |
<-------------------------------+                              |
|                               |                              |
| ACK                           |                              |
+------------------------------>+                              |
|                               |                              |
| INVITE mobile                 |                              |
+------------------------------>+                              |
|                               |                              |
| 100 Trying                    |                              |
<-------------------------------+                              |
|                               |                              |
|                               | INVITE mobile                |
|                               +------------------------------>
|                               | FROM: softphone@172.17.0.2   |

There was another messages after, but I am intrested in last. Why Asterisk substitute the “From” header. Actually softphone has another ip 192.168.12.101, but Asterisk put in header his IP. Why? And how I can specify there at least Asterisk’s external IP, which can be seen by softphone and mobile (192.168.12.101)

Thank you for answer or pointing to the link what I can read to better understand this and answer by myself on this question.

Full log of pjsip:

INVITE sip:mobile@192.168.12.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:63480;rport;branch=z9hG4bKPjTQ-HJ9lyTBUYgOFHXKZtR3Qs90s9wDVp
Max-Forwards: 70
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>
Contact: "softphone" <sip:softphone@192.168.12.101:63480;ob>
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 32175 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.4
Content-Type: application/sdp
Content-Length:   304

v=0
o=- 3788197877 3788197877 IN IP4 192.168.12.101
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 0 101
c=IN IP4 192.168.12.101
b=TIAS:64000
a=rtcp:4023 IN IP4 192.168.12.101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<--- Transmitting SIP response (572 bytes) to UDP:172.17.0.1:56465 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.12.101:63480;rport=56465;received=172.17.0.1;branch=z9hG4bKPjTQ-HJ9lyTBUYgOFHXKZtR3Qs90s9wDVp
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=z9hG4bKPjTQ-HJ9lyTBUYgOFHXKZtR3Qs90s9wDVp
CSeq: 32175 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1579209077/ceda0f57d821d9621abb351e96b18498",opaque="5f76ab853b396ba2",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.7.0
Content-Length:  0


<--- Received SIP request (399 bytes) from UDP:172.17.0.1:56465 --->
ACK sip:mobile@192.168.12.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:63480;rport;branch=z9hG4bKPjTQ-HJ9lyTBUYgOFHXKZtR3Qs90s9wDVp
Max-Forwards: 70
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=z9hG4bKPjTQ-HJ9lyTBUYgOFHXKZtR3Qs90s9wDVp
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 32175 ACK
Content-Length:  0


<--- Received SIP request (1226 bytes) from UDP:172.17.0.1:56465 --->
INVITE sip:mobile@192.168.12.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:63480;rport;branch=z9hG4bKPjwkNRMrj3MyXbSlYVG6rGHoOPKonnMMlj
Max-Forwards: 70
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>
Contact: "softphone" <sip:softphone@192.168.12.101:63480;ob>
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 32176 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
User-Agent: Telephone 1.4
Authorization: Digest username="softphone", realm="asterisk", nonce="1579209077/ceda0f57d821d9621abb351e96b18498", uri="sip:mobile@192.168.12.101", response="6060304c6b6f8c10bb6aa3230b06ad99", algorithm=md5, cnonce="BNd-NwVaEt0NFlSZfK-T2-nK85wxmLxg", opaque="5f76ab853b396ba2", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   304

v=0
o=- 3788197877 3788197877 IN IP4 192.168.12.101
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 0 101
c=IN IP4 192.168.12.101
b=TIAS:64000
a=rtcp:4023 IN IP4 192.168.12.101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

  == Setting global variable 'SIPDOMAIN' to '192.168.12.101'
<--- Transmitting SIP response (374 bytes) to UDP:172.17.0.1:56465 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.12.101:63480;rport=56465;received=172.17.0.1;branch=z9hG4bKPjwkNRMrj3MyXbSlYVG6rGHoOPKonnMMlj
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>
CSeq: 32176 INVITE
Server: Asterisk PBX 16.7.0
Content-Length:  0


    -- Executing [mobile@internal:1] Dial("PJSIP/softphone-00000006", "PJSIP/mobile") in new stack
    -- Called PJSIP/mobile
<--- Transmitting SIP request (908 bytes) to UDP:172.17.0.1:57846 --->
INVITE sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPjddc57832-bcf5-4811-9185-8758e0a1ba2c
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>
Contact: <sip:asterisk@192.168.12.101:5060>
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1165 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 16.7.0
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 719225989 719225989 IN IP4 192.168.12.101
s=Asterisk
c=IN IP4 192.168.12.101
t=0 0
m=audio 11904 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (365 bytes) from UDP:172.17.0.1:57846 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.12.101:5060;rport=5060;branch=z9hG4bKPjddc57832-bcf5-4811-9185-8758e0a1ba2c;received=192.168.12.101
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1165 INVITE
Content-Length: 0


    -- PJSIP/mobile-00000007 is ringing
    -- PJSIP/mobile-00000007 is ringing
<--- Transmitting SIP response (563 bytes) to UDP:172.17.0.1:56465 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.12.101:63480;rport=56465;received=172.17.0.1;branch=z9hG4bKPjwkNRMrj3MyXbSlYVG6rGHoOPKonnMMlj
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
CSeq: 32176 INVITE
Server: Asterisk PBX 16.7.0
Contact: <sip:192.168.12.101:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (656 bytes) from UDP:172.17.0.1:57846 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:5060;rport=5060;branch=z9hG4bKPjddc57832-bcf5-4811-9185-8758e0a1ba2c;received=192.168.12.101
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1165 INVITE
Contact: "mobile" <sip:mobile@172.17.0.1:57846;transport=udp>
Content-Type: application/sdp
Content-Length: 200

v=0
o=- 1579209076347 1579209081946 IN IP4 192.168.12.100
s=-
c=IN IP4 192.168.12.100
t=0 0
m=audio 41194 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

       > 0x7f96940233b0 -- Strict RTP learning after remote address set to: 192.168.12.100:41194
    -- PJSIP/mobile-00000007 answered PJSIP/softphone-00000006
<--- Transmitting SIP request (405 bytes) to UDP:172.17.0.1:57846 --->
ACK sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj42158528-3a57-4d44-b99e-d33c59a4d9c5
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1165 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.7.0
Content-Length:  0


       > 0x7f9694014d80 -- Strict RTP learning after remote address set to: 192.168.12.101:4022
<--- Transmitting SIP response (881 bytes) to UDP:172.17.0.1:56465 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:63480;rport=56465;received=172.17.0.1;branch=z9hG4bKPjwkNRMrj3MyXbSlYVG6rGHoOPKonnMMlj
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
CSeq: 32176 INVITE
Server: Asterisk PBX 16.7.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.12.101:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 3788197877 3788197879 IN IP4 192.168.12.101
s=Asterisk
c=IN IP4 192.168.12.101
t=0 0
m=audio 30336 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

    -- Channel PJSIP/mobile-00000007 joined 'simple_bridge' basic-bridge <115cd5c3-aaf1-43d7-9db3-df6480ef9864>
    -- Channel PJSIP/softphone-00000006 joined 'simple_bridge' basic-bridge <115cd5c3-aaf1-43d7-9db3-df6480ef9864>
       > Bridge 115cd5c3-aaf1-43d7-9db3-df6480ef9864: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'PJSIP/softphone-00000006' and 'PJSIP/mobile-00000007' - media will flow directly between them
<--- Transmitting SIP request (922 bytes) to UDP:172.17.0.1:57846 --->
INVITE sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj991bbcd9-dcb8-424b-b364-48e70e8bafad
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Contact: <sip:asterisk@192.168.12.101:5060>
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1166 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 16.7.0
Content-Type: application/sdp
Content-Length:   238

v=0
o=- 719225989 719225990 IN IP4 192.168.12.101
s=Asterisk
c=IN IP4 192.168.12.101
t=0 0
m=audio 4022 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (392 bytes) from UDP:172.17.0.1:56465 --->
ACK sip:192.168.12.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:63480;rport;branch=z9hG4bKPjbOSKVpBc8aFeK6.JigxSSJ6.DMakVkVp
Max-Forwards: 70
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 32176 ACK
Content-Length:  0


<--- Transmitting SIP request (948 bytes) to UDP:172.17.0.1:56465 --->
INVITE sip:softphone@172.17.0.1:56465;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj7f7e3c7f-53f4-4f54-a349-64359052d220
From: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
To: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
Contact: <sip:192.168.12.101:5060>
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 4991 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 16.7.0
Content-Type: application/sdp
Content-Length:   241

v=0
o=- 3788197877 3788197880 IN IP4 192.168.12.101
s=Asterisk
c=IN IP4 192.168.12.101
t=0 0
m=audio 41194 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (656 bytes) from UDP:172.17.0.1:57846 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:5060;rport=5060;branch=z9hG4bKPj991bbcd9-dcb8-424b-b364-48e70e8bafad;received=192.168.12.101
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1166 INVITE
Contact: "mobile" <sip:mobile@172.17.0.1:57846;transport=udp>
Content-Type: application/sdp
Content-Length: 200

v=0
o=- 1579209076347 1579209081981 IN IP4 192.168.12.100
s=-
c=IN IP4 192.168.12.100
t=0 0
m=audio 41194 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

       > 0x7f96940233b0 -- Strict RTP learning after remote address set to: 192.168.12.100:41194
<--- Transmitting SIP request (405 bytes) to UDP:172.17.0.1:57846 --->
ACK sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPja259c592-0598-4720-93c2-7b4527557491
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1166 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.7.0
Content-Length:  0


<--- Received SIP response (909 bytes) from UDP:172.17.0.1:56465 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:5060;rport=5060;received=192.168.12.101;branch=z9hG4bKPj7f7e3c7f-53f4-4f54-a349-64359052d220
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
From: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
To: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
CSeq: 4991 INVITE
Contact: "softphone" <sip:softphone@192.168.12.101:63480;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length:   280

v=0
o=- 3788197877 3788197878 IN IP4 192.168.12.101
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4022 RTP/AVP 8 101
c=IN IP4 192.168.12.101
b=TIAS:64000
a=rtcp:4023 IN IP4 192.168.12.101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

       > 0x7f9694014d80 -- Strict RTP learning after remote address set to: 192.168.12.101:4022
<--- Transmitting SIP request (437 bytes) to UDP:172.17.0.1:56465 --->
ACK sip:softphone@172.17.0.1:56465;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj0fc730fc-9798-4379-b6d1-bf75dba00d15
From: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
To: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 4991 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.7.0
Content-Length:  0


<--- Received SIP request (379 bytes) from UDP:172.17.0.1:57846 --->
OPTIONS sip:192.168.12.101 SIP/2.0
Call-ID: 539f24109e0d1b6953c48fae214e28ed@192.168.12.100
CSeq: 1581 OPTIONS
From: "mobile" <sip:mobile@192.168.12.101>;tag=153936636
To: "mobile" <sip:mobile@192.168.12.101>
Via: SIP/2.0/UDP 192.168.12.100:48151;branch=z9hG4bKe1107dde8e7363491d155b39a6e98228363937;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0


<--- Transmitting SIP response (575 bytes) to UDP:172.17.0.1:57846 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.12.100:48151;rport=57846;received=172.17.0.1;branch=z9hG4bKe1107dde8e7363491d155b39a6e98228363937
Call-ID: 539f24109e0d1b6953c48fae214e28ed@192.168.12.100
From: "mobile" <sip:mobile@192.168.12.101>;tag=153936636
To: "mobile" <sip:mobile@192.168.12.101>;tag=z9hG4bKe1107dde8e7363491d155b39a6e98228363937
CSeq: 1581 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1579209085/6d794c530790d165345bad6c77e6e19c",opaque="1aeb99b701896855",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.7.0
Content-Length:  0


<--- Received SIP request (419 bytes) from UDP:172.17.0.1:56465 --->
BYE sip:192.168.12.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:63480;rport;branch=z9hG4bKPjxVpl60YlpYpCkicq6OcD-cTz7uQOY8Sw
Max-Forwards: 70
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
CSeq: 32177 BYE
User-Agent: Telephone 1.4
Content-Length:  0


<--- Transmitting SIP response (408 bytes) to UDP:172.17.0.1:56465 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:63480;rport=56465;received=172.17.0.1;branch=z9hG4bKPjxVpl60YlpYpCkicq6OcD-cTz7uQOY8Sw
Call-ID: irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y
From: "softphone" <sip:softphone@192.168.12.101>;tag=jPPIoHYF1MqQQgiFVlt74NmzvuXgv.bU
To: <sip:mobile@192.168.12.101>;tag=9fc2f597-3f0d-4713-bbbe-c96790a326ea
CSeq: 32177 BYE
Server: Asterisk PBX 16.7.0
Content-Length:  0


    -- Channel PJSIP/softphone-00000006 left 'native_rtp' basic-bridge <115cd5c3-aaf1-43d7-9db3-df6480ef9864>
    -- Channel PJSIP/mobile-00000007 left 'native_rtp' basic-bridge <115cd5c3-aaf1-43d7-9db3-df6480ef9864>
  == Spawn extension (internal, mobile, 1) exited non-zero on 'PJSIP/softphone-00000006'
    -- Executing [h@internal:1] Dial("PJSIP/softphone-00000006", "PJSIP/h") in new stack
    -- Caller hung up before dial.
  == Spawn extension (internal, h, 1) exited non-zero on 'PJSIP/softphone-00000006'
<--- Transmitting SIP request (923 bytes) to UDP:172.17.0.1:57846 --->
INVITE sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj85fa4e37-525b-45fb-9fc5-d4be9a17f784
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Contact: <sip:asterisk@192.168.12.101:5060>
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1167 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 16.7.0
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 719225989 719225991 IN IP4 192.168.12.101
s=Asterisk
c=IN IP4 192.168.12.101
t=0 0
m=audio 11904 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (656 bytes) from UDP:172.17.0.1:57846 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:5060;rport=5060;branch=z9hG4bKPj85fa4e37-525b-45fb-9fc5-d4be9a17f784;received=192.168.12.101
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1167 INVITE
Contact: "mobile" <sip:mobile@172.17.0.1:57846;transport=udp>
Content-Type: application/sdp
Content-Length: 200

v=0
o=- 1579209076347 1579209084901 IN IP4 192.168.12.100
s=-
c=IN IP4 192.168.12.100
t=0 0
m=audio 41194 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<--- Transmitting SIP request (405 bytes) to UDP:172.17.0.1:57846 --->
ACK sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj77ed8977-129e-4e3c-8b66-fe8b1fdc9602
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1167 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.7.0
Content-Length:  0


<--- Transmitting SIP request (405 bytes) to UDP:172.17.0.1:57846 --->
BYE sip:mobile@172.17.0.1:57846 SIP/2.0
Via: SIP/2.0/UDP 192.168.12.101:5060;rport;branch=z9hG4bKPj58791f06-fdfb-469a-aadb-f0834a5f92d2
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1168 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 16.7.0
Content-Length:  0


<--- Received SIP response (357 bytes) from UDP:172.17.0.1:57846 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.12.101:5060;rport=5060;branch=z9hG4bKPj58791f06-fdfb-469a-aadb-f0834a5f92d2;received=192.168.12.101
From: "softphone" <sip:softphone@172.17.0.2>;tag=7c53b704-62f9-48af-aaaa-68d4a297b88c
To: <sip:mobile@172.17.0.1>;tag=2718536061
Call-ID: 51f6004b-0918-4d40-adc0-69a37683eaac
CSeq: 1168 BYE
Content-Length: 0```
2

Asterisk is a back to back user agent, not a sip proxy.

There are two calls in progress, one between your softphone and asterisk with a call ID of irLJORPBUK1Chp5RLqFY9SK5f5HFfX0Y , and second between asterisk and your mobile with a call ID of 51f6004b-0918-4d40-adc0-69a37683eaac

3

Is there in Asterisk parameters for edit his “From” header when he making a call to another sip endpoint which was initially called?

I want that header “From” in INVITE sending from Asterisk to mobile will have not private IP of Asterisk “172.17.0.2” but hist external address “192.168.12.101”.

4

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