rendered paste body
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 10.10.1.23: 5061; transport UDP (non-NAT)
Transmitting (no NAT) to 10.10.1.23:5061:UDP
with fd 16
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.1.23:5061;branch=z9hG4bK7e5ec2b9;received=10.10.1.23
From: sip:112@10.10.1.16
To: sip:112@10.10.1.16
Call-ID: 000bfd5d-ef200003-18084653-346c6c9d@10.10.1.23
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:112@10.10.1.16;transport=UDP>
Content-Length: 0
---
UDP write: fd 16
-- Contact header: transport
-- Contact header: q
[Kasterisk1*CLI>
Transmitting (no NAT) to 10.10.1.23:5061:UDP
with fd 16
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.1.23:5061;branch=z9hG4bK7e5ec2b9;received=10.10.1.23
From: sip:112@10.10.1.16
To: sip:112@10.10.1.16;tag=as76222803
Call-ID: 000bfd5d-ef200003-18084653-346c6c9d@10.10.1.23
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:112@10.10.1.23:5061>;expires=120
Date: Sun, 01 Oct 2006 20:10:35 GMT
Content-Length: 0
---
[Kasterisk1*CLI>
UDP write: fd 16
Scheduling destruction of call '000bfd5d-ef200003-18084653-346c6c9d@10.10.1.23' in 15000 ms
[Kasterisk1*CLI>
<-- Sip read from 207.174.202.4:5060:UDP
INVITE sip:s@10.10.1.16;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK4d97e4fd;rport
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as132f5b62
To: <sip:s@10.10.1.16;transport=UDP>
Contact: <sip:5555555555@207.174.202.4>
Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4
CSeq: 102 INVITE
User-Agent: "TELIAX GEAR v2.1.0x"
Max-Forwards: 70
Date: Sun, 01 Oct 2006 20:14:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 315
v=0
o=root 724 724 IN IP4 207.174.202.4
s=session
c=IN IP4 207.174.202.4
t=0 0
m=audio 10088 RTP/AVP 0 18 111 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (13 headers 14 lines)---
Using INVITE request as basis request - 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4
Sending to 207.174.202.4: 5060; transport UDP (NAT)
Found peer 'teliax'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 207.174.202.4:10088
Found description format PCMU
Found description format G729
Found description format G726-32
Found description format GSM
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x116 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in default (domain 10.10.1.16;transport=UDP)
list_route: hop: <sip:5555555555@207.174.202.4>
Transmitting (NAT) to 207.174.202.4:5060:UDP
with fd 16
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK4d97e4fd;received=207.174.202.4;rport=5060
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as132f5b62
To: <sip:s@10.10.1.16;transport=UDP>
Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.1.16;transport=UDP>
Content-Length: 0
---
UDP write: fd 16
-- Executing Playback("SIP/mybiz-08d48280", "vm-goodbye") in new stack
We're at 10.10.1.16 port 17248
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 207.174.202.4:5060:UDP
with fd 16
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK4d97e4fd;received=207.174.202.4;rport=5060
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as132f5b62
To: <sip:s@10.10.1.16;transport=UDP>;tag=as6d0a49e8
Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@10.10.1.16;transport=UDP>
Content-Type: application/sdp
Content-Length: 233
v=0
o=root 5675 5675 IN IP4 10.10.1.16
s=session
c=IN IP4 10.10.1.16
t=0 0
m=audio 17248 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
UDP write: fd 16
-- Playing 'vm-goodbye' (language 'en')
[Kasterisk1*CLI>
<-- Sip read from 207.174.202.4:5060:UDP
ACK sip:s@10.10.1.16;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 207.174.202.4:5060;branch=z9hG4bK11f016d9;rport
From: "Cell Phone CA" <sip:5555555555@207.174.202.4>;tag=as132f5b62
To: <sip:s@10.10.1.16;transport=UDP>;tag=as6d0a49e8
Contact: <sip:5555555555@207.174.202.4>
Call-ID: 6ae342da6e3a2d082d62a0c12627f62e@207.174.202.4
CSeq: 102 ACK
User-Agent: "TELIAX GEAR v2.1.0x"
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines)---
[Kasterisk1*CLI>
-- Executing Macro("SIP/mybiz-08d48280", "hangupcall") in new stack
-- Executing ResetCDR("SIP/mybiz-08d48280", "w") in new stack
[Kasterisk1*CLI>
-- Executing NoCDR("SIP/mybiz-08d48280", "") in new stack
-- Executing Wait("SIP/mybiz-08d48280", "5") in new stack
[Kasterisk1*CLI>
<-- Sip read from 10.10.1.3:37810:UDP
--- (0 headers 1 lines)---
[Kasterisk1*CLI>
12 headers, 3 lines
Reliably Transmitting (no NAT) to 10.10.1.23:5061:UDP
with fd 16
NOTIFY sip:112@10.10.1.23:5061 SIP/2.0
Via: SIP/2.0/UDP 10.10.1.16:5060;branch=z9hG4bK5826f50b
From: "mybiz" <sip:mybiz@10.10.1.16>;tag=as230f95ca
To: <sip:112@10.10.1.23:5061>
Contact: <sip:mybiz@10.10.1.16;transport=UDP>
Call-ID: 7265553c6e240b0f5b6378b2482ad6d4@10.10.1.16
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.10.1.16
Voice-Message: 0/0 (0/0)