rendered paste bodyhostname*CLI>
<--- SIP read from Asterisk:5060 --->
INVITE sip:some_ext@Asterisk:5070 SIP/2.0
Record-Route: <sip:Asterisk;lr=on;ftag=2127514061631>
Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0
Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564
Content-Length: 355
Contact: <sip:9492666205@118.110.94.145:5060>
Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126
Content-Type: application/sdp
CSeq: 2 INVITE
From: "unknown"<sip:9492666205@sip.demo.com:5060>;tag=2127514061631
Max-Forwards: 69
To: <sip:*721111111111@sip.demo.com:5060>
v=0
o=- 3388311921 3388311921 IN IP4 192.168.0.126
s=SJphone
c=IN IP4 Asterisk
t=0 0
a=direction:active
m=audio 37924 RTP/AVP 3 97 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
a=nortpproxy:yes
<------------->
--- (12 headers 16 lines) ---
Sending to Asterisk : 5060 (no NAT)
Using INVITE request as basis request - 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126
Found no matching peer or user for 'Asterisk:5060'
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port Asterisk:37924
Found description format GSM for ID 3
Found description format iLBC for ID 97
Found description format iLBC for ID 98
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port Asterisk:37924
Looking for some_ext in default (domain Asterisk)
list_route: hop: <sip:Asterisk;lr=on;ftag=2127514061631>
hostname*CLI>
<--- Transmitting (no NAT) to Asterisk:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk
Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564
From: "unknown"<sip:9492666205@sip.demo.com:5060>;tag=2127514061631
To: <sip:*721111111111@sip.demo.com:5060>
Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:some_ext@Asterisk:5070>
Content-Length: 0
<------------>
-- Executing [some_ext@default:1] Answer("SIP/5060-00663300", "") in new stack
Audio is at Asterisk port 11176
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to Asterisk:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk
Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564
Record-Route: <sip:Asterisk;lr=on;ftag=2127514061631>
From: "unknown"<sip:9492666205@sip.demo.com:5060>;tag=2127514061631
To: <sip:*721111111111@sip.demo.com:5060>;tag=as082880aa
Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:some_ext@Asterisk:5070>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 15109 15109 IN IP4 Asterisk
s=session
c=IN IP4 Asterisk
t=0 0
m=audio 11176 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Executing [some_ext@default:2] Playback("SIP/5060-00663300", "tt-weasels") in new stack
-- <SIP/5060-00663300> Playing 'tt-weasels' (language 'en')
Retransmitting #1 (no NAT) to Asterisk:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk
Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564
Record-Route: <sip:Asterisk;lr=on;ftag=2127514061631>
From: "unknown"<sip:9492666205@sip.demo.com:5060>;tag=2127514061631
To: <sip:*721111111111@sip.demo.com:5060>;tag=as082880aa
Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:some_ext@Asterisk:5070>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 15109 15109 IN IP4 Asterisk
s=session
c=IN IP4 Asterisk
t=0 0
m=audio 11176 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (no NAT) to Asterisk:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP Asterisk;branch=z9hG4bKaa96.eaa0fbd7.0;received=Asterisk
Via: SIP/2.0/UDP 192.168.0.126;received=118.110.94.145;rport=5060;branch=z9hG4bKc0a8007e00000096464b0af1000017fa00000564
Record-Route: <sip:Asterisk;lr=on;ftag=2127514061631>
From: "unknown"<sip:9492666205@sip.demo.com:5060>;tag=2127514061631
To: <sip:*721111111111@sip.demo.com:5060>;tag=as082880aa
Call-ID: 32489568-AC55-4D66-9E61-C04A332C9391@192.168.0.126
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:some_ext@Asterisk:5070>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 15109 15109 IN IP4 Asterisk
s=session
c=IN IP4 Asterisk
t=0 0
m=audio 11176 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv