rendered paste bodyCapabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|
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 192.168.1.105:8000
Looking for 123 in mycontext (domain 192.168.1.102)
list_route: hop: <sip:me@192.168.1.105:5060>
<--- Transmitting (no NAT) to 192.168.1.105:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bKDB7566B661F04741ACC7734C12A4D7
8B;received=192.168.1.105;rport=5060
From: me <sip:me@192.168.1.102>;tag=2509996786
To: <sip:123@192.168.1.102>
Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
CSeq: 15527 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:123@192.168.1.102>
Content-Length: 0
<------------>
-- Executing [123@mycontext:1] Dial("SIP/me-0988e160", "SIP/123@home|50") in
new stack
Audio is at 88.149.162.157 port 17416
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 81.174.8.170:5060:
INVITE sip:123@81.174.8.170 SIP/2.0
Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport
From: "me" <sip:marco@88.149.162.157>;tag=as14f08839
To: <sip:123@81.174.8.170>
Contact: <sip:marco@88.149.162.157>
Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
ate: Wed, 31 Jan 2007 15:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 9339 9339 IN IP4 88.149.162.157
s=session
c=IN IP4 88.149.162.157
t=0 0
m=audio 17416 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 123@home
Retransmitting #1 (NAT) to 81.174.8.170:5060:
INVITE sip:123@81.174.8.170 SIP/2.0
Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport
From: "me" <sip:marco@88.149.162.157>;tag=as14f08839
To: <sip:123@81.174.8.170>
Contact: <sip:marco@88.149.162.157>
Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 Jan 2007 15:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 9339 9339 IN IP4 88.149.162.157
s=session
c=IN IP4 88.149.162.157
t=0 0
m=audio 17416 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (NAT) to 81.174.8.170:5060:
INVITE sip:123@81.174.8.170 SIP/2.0
Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport
From: "me" <sip:marco@88.149.162.157>;tag=as14f08839
To: <sip:123@81.174.8.170>
Contact: <sip:marco@88.149.162.157>
Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 Jan 2007 15:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 9339 9339 IN IP4 88.149.162.157
s=session
c=IN IP4 88.149.162.157
t=0 0
m=audio 17416 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (NAT) to 81.174.8.170:5060:
INVITE sip:123@81.174.8.170 SIP/2.0
Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport
From: "me" <sip:marco@88.149.162.157>;tag=as14f08839
To: <sip:123@81.174.8.170>
Contact: <sip:marco@88.149.162.157>
Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 Jan 2007 15:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 9339 9339 IN IP4 88.149.162.157
s=session
c=IN IP4 88.149.162.157
t=0 0
m=audio 17416 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (NAT) to 81.174.8.170:5060:
INVITE sip:123@81.174.8.170 SIP/2.0
Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport
From: "me" <sip:marco@88.149.162.157>;tag=as14f08839
To: <sip:123@81.174.8.170>
Contact: <sip:marco@88.149.162.157>
Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 Jan 2007 15:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 9339 9339 IN IP4 88.149.162.157
s=session
c=IN IP4 88.149.162.157
t=0 0
m=audio 17416 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (NAT) to 81.174.8.170:5060:
INVITE sip:123@81.174.8.170 SIP/2.0
Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport
From: "me" <sip:marco@88.149.162.157>;tag=as14f08839
To: <sip:123@81.174.8.170>
Contact: <sip:marco@88.149.162.157>
Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 Jan 2007 15:51:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 241
v=0
o=root 9339 9339 IN IP4 88.149.162.157
s=session
c=IN IP4 88.149.162.157
t=0 0
m=audio 17416 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
rfc-1918*CLI>
<--- SIP read from 192.168.1.105:5060 --->
CANCEL sip:123@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;rport;branch=z9hG4bKDB7566B661F04741ACC7734C
12A4D78B
From: me <sip:me@192.168.1.102>;tag=2509996786
To: <sip:123@192.168.1.102>
Contact: <sip:me@192.168.1.105:5060>
Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
CSeq: 15527 CANCEL
Proxy-Authorization: Digest username="me",realm="asterisk",nonce="374691c0",resp
onse="71cb3b9b2e18a88c0641b44709329d77",uri="sip:123@192.168.1.102",algorithm=MD
5
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.105 : 5060 (NAT)
<--- Reliably Transmitting (NAT) to 192.168.1.105:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bKDB7566B661F04741ACC7734C12A4D7
8B;received=192.168.1.105;rport=5060
From: me <sip:me@192.168.1.102>;tag=2509996786
To: <sip:123@192.168.1.102>;tag=as49d906ae
Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
CSeq: 15527 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 192.168.1.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bKDB7566B661F04741ACC7734C12A4D7
8B;received=192.168.1.105;rport=5060
From: me <sip:me@192.168.1.102>;tag=2509996786
To: <sip:123@192.168.1.102>;tag=as49d906ae
Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
CSeq: 15527 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:123@192.168.1.102>
Content-Length: 0
<------------>
== Spawn extension (mycontext, 123, 1) exited non-zero on 'SIP/me-0988e160'
rfc-1918*CLI>
<--- SIP read from 192.168.1.105:5060 --->
ACK sip:123@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;rport;branch=z9hG4bKDB7566B661F04741ACC7734C
12A4D78B
From: me <sip:me@192.168.1.102>;tag=2509996786
To: <sip:123@192.168.1.102>;tag=as49d906ae
Contact: <sip:me@192.168.1.105:5060>
Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
CSeq: 15527 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
' Method: ACK
<--- SIP read from 192.168.1.105:5060 --->
ACK sip:123@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;rport;branch=z9hG4bKDB7566B661F04741ACC7734C
12A4D78B
From: me <sip:me@192.168.1.102>;tag=2509996786
To: <sip:123@192.168.1.102>;tag=as49d906ae
Contact: <sip:me@192.168.1.105:5060>
Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105
CSeq: 15527 ACK
Max-Forwards: 70
Content-Length: 0
rfc-1918*CLI>
<------------->
--- (9 headers 0 lines) ---
rfc-1918*CLI>