MITOL*CLI> sip set debug peer sim1
SIP Debugging Enabled for IP: 192.168.0.100

<--- SIP read from UDP:192.168.0.100:5064 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5064;rport;branch=z9hG4bK8444cd044f
From: "+79250287897" <sip:[email protected]:5064>;tag=446a7210
To: <sip:[email protected]>
Call-ID: [email protected]
Contact: <sip:[email protected]:5064>
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Supported: replaces
Content-Type: application/sdp
User-Agent: Mv-37x (904290)
Content-Length: 323

v=0
o=CMI-SIPUA 723 0 IN IP4 192.168.0.100
s=SIP CALL
c=IN IP4 192.168.0.100
t=0 0
m=audio 20004 RTP/AVP 0 8 4 18 23 22 2 21 101
a=rtpmap:23 G726-16/8000
a=rtpmap:22 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:21 G726-40/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=sendrecv
<------------->
--- (13 headers 14 lines) ---
Sending to 192.168.0.100:5064 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'sim1' for 'sim1' from 192.168.0.100:5064
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 23
Found RTP audio format 22
Found RTP audio format 2
Found RTP audio format 21
Found RTP audio format 101
Found unknown media description format G726-16 for ID 23
Found unknown media description format G726-24 for ID 22
Found audio description format G726-32 for ID 2
Found unknown media description format G726-40 for ID 21
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008090e (gsm|ulaw|alaw|g726|g729|h263|testlaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90c (ulaw|alaw|g726|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.100:20004
Looking for receiver in sip (domain 192.168.0.25)

<--- Reliably Transmitting (NAT) to 192.168.0.100:5064 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.100:5064;branch=z9hG4bK8444cd044f;received=192.168.0.100;rport=5064
From: "+79250287897" <sip:[email protected]:5064>;tag=446a7210
To: <sip:[email protected]>;tag=as36414709
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.8.2~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Mar 24 19:25:10] NOTICE[27874]: chan_sip.c:22147 handle_request_invite: Call from 'sim1' (192.168.0.100:5064) to extension 'receiver' rejected because extension not found in context 'sip'.
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.100:5064 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5064;branch=z9hG4bK8444cd044f
From: "+79250287897" <sip:[email protected]:5064>;tag=446a7210
To: <sip:[email protected]>;tag=as36414709
Call-ID: [email protected]
Contact: <sip:[email protected]:5064>
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK