@ david551 I only listed a small fragment of the SIP session because the forum wouldn’t let me enter all the text that I intended to list. When I was trying to save the post it kept sayng “Sorry, new users can only put 2 links in a post.” although I didn’t enter any link except ‘localphone.com’ which is just mentioned as a domain. That’s why I tried to rewrite what the forum considered links by entering spaces around dots. Anyway, the SIP session part before the lines listed above is the following:
<— Received SIP request (1175 bytes) from UDP:140.153.72.56:5060 —>
INVITE sip:136.46.324.75:5060 SIP/2.0
Record-Route: < sip:140.153.72.56;lr=on;ftag=gK086bb5a7 >
Record-Route: < sip:126.219.52.41;lr;ftag=gK086bb5a7 >
Via: SIP/2.0/UDP 140.153.72.56;branch=z9hG4bK356c.5360dbb7.0
Via: SIP/2.0/UDP 126.219.52.41:5060;rport=5060;branch=z9hG4bK356c.10463da.0
From: < sip: 10782172281 @ 126.219.52.41 >;tag=gK086bb5a7
To: < sip: 13051079265 @ localphone . com >;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
Call-ID: 793253110_109248450 @ 199.199.12.56
CSeq: 63984 INVITE
Max-Forwards: 12
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: < sip: 126.219.52.41;vbdid=941 . 29056c13 >
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Length: 239
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 13229 650067 IN IP4 199.199.12.56
s=SIP Media Capabilities
c=IN IP4 199.199.12.54
t=0 0
m=audio 40808 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:20
0x7fddf001b170 – Strict RTP learning after remote address set to: 199.199.12.54:40808
<— Transmitting SIP response (1062 bytes) to UDP: 140.153.72.56:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 140.153.72.56;rport=5060;received=140.153.72.56;branch=z9hG4bK356c . 5360dbb7 . 0
Via: SIP/2.0/UDP 126.219.52.41:5060;rport=5060;branch=z9hG4bK356c .10463da . 0
Record-Route: < sip: 140.153.72.56;lr;ftag=gK086bb5a7 >
Record-Route: < sip: 126.219.52.41;lr;ftag=gK086bb5a7 >
Call-ID: 793253110_109248450 @ 199.199.12.56
From: < sip: 10782172281 @ 126.219.52.41 >;tag=gK086bb5a7
To: < sip: 13051079265 @ localphone . com >;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
CSeq: 63984 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: < sip: 136.46.324.75 : 5060 >
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 15.6.1
Content-Type: application/sdp
Content-Length: 234
v=0
o=- 13229 650069 IN IP4 136.46.324.75
s=Asterisk
c=IN IP4 136.46.324.75
t=0 0
m=audio 10010 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
0x7fddf001b170 – Strict RTP switching to RTP target address 199.199.12.54:40808 as source
<— Received SIP request (411 bytes) from UDP:140.153.72.56:5060 —>
ACK sip:136.46.324.75:5060 SIP/2.0
Via: SIP/2.0/UDP 140.153.72.56;branch=z9hG4bK356c . 5360dbb7.2
Via: SIP/2.0/UDP 126.219.52.41:5060;rport=5060;branch=z9hG4bK356c.10463da.2
From: < sip: 10782172281 @ 126.219.52.41 >;tag=gK086bb5a7
To: < sip: 13051079265 @ localphone . com >;tag=514dc3ba-68ed-42ff-9733-bd65b8ebb7c7
Call-ID: 793253110_109248450 @ 199.199.12.56
CSeq: 63984 ACK
Max-Forwards: 12
Content-Length: 0