rendered paste body<-- SIP read from 192.168.0.107:52805:
INVITE sip:0095@192.168.0.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK69431df7
From: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
To: <sip:0095@192.168.0.50>
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 101 INVITE
User-Agent: CSCO/7
Contact: <sip:0095@192.168.0.107:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 249
Accept: application/sdp
Remote-Party-ID: "0095" <sip:0095@192.168.0.107>;party=calling;id-type=subscriber;privacy=off;screen=no
v=0
o=Cisco-SIPUA 13516 18140 IN IP4 192.168.0.107
s=SIP Call
c=IN IP4 192.168.0.107
t=0 0
m=audio 20984 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (13 headers 11 lines)---
Using INVITE request as basis request - 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
Sending to 192.168.0.107 : 5060 (non-NAT)
Found user '0095'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.107:20984
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 0095 in testing (domain 192.168.0.50)
list_route: hop: <sip:0095@192.168.0.107:5060>
Transmitting (no NAT) to 192.168.0.107:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK69431df7;received=192.168.0.107
From: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
To: <sip:0095@192.168.0.50>
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:0095@192.168.0.106:5061>
Content-Length: 0
---
-- Executing Macro("SIP/0095-6ae1", "stdexten|0095|SIP/0095") in new stack
-- Executing Dial("SIP/0095-6ae1", "SIP/0095|20") in new stack
We're at 192.168.0.106 port 12690
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 192.168.0.107:5060:
INVITE sip:0095@192.168.0.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>
Contact: <sip:0095@192.168.0.106:5061>
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Dec 2005 23:24:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 3585 3585 IN IP4 192.168.0.106
s=session
c=IN IP4 192.168.0.106
t=0 0
m=audio 12690 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 0095
<-- SIP read from 192.168.0.107:52925:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:0095@192.168.0.107:5060>
Content-Length: 0
--- (9 headers 0 lines)---
<-- SIP read from 192.168.0.107:52926:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>;tag=000ab747a225753768d896cd-4ebc9ba0
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:0095@192.168.0.107:5060>
Content-Length: 0
--- (9 headers 0 lines)---
-- SIP/0095-125f is ringing
Transmitting (no NAT) to 192.168.0.107:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK69431df7;received=192.168.0.107
From: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
To: <sip:0095@192.168.0.50>;tag=as130e56a7
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:0095@192.168.0.106:5061>
Content-Length: 0
---
-- Nobody picked up in 20000 ms
Reliably Transmitting (no NAT) to 192.168.0.107:5060:
CANCEL sip:0095@192.168.0.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>
Contact: <sip:0095@192.168.0.106:5061>
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call '5f66276e0c1721707972276818015f9a@192.168.0.106' in 15000 ms
-- Executing Goto("SIP/0095-6ae1", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail("SIP/0095-6ae1", "u0095") in new stack
We're at 192.168.0.106 port 19190
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.107:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK69431df7;received=192.168.0.107
From: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
To: <sip:0095@192.168.0.50>;tag=as130e56a7
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:0095@192.168.0.106:5061>
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 3585 3585 IN IP4 192.168.0.106
s=session
c=IN IP4 192.168.0.106
t=0 0
m=audio 19190 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'vm-theperson' (language 'en')
<-- SIP read from 192.168.0.107:52928:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>;tag=000ab747a225753768d896cd-4ebc9ba0
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 CANCEL
Server: CSCO/7
Content-Length: 0
--- (8 headers 0 lines)---
<-- SIP read from 192.168.0.107:52929:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>;tag=000ab747a225753768d896cd-4ebc9ba0
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 INVITE
Server: CSCO/7
Contact: <sip:0095@192.168.0.107:5060>
Content-Length: 0
--- (9 headers 0 lines)---
Transmitting (no NAT) to 192.168.0.107:5060:
ACK sip:0095@192.168.0.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK642fe0bf;rport
From: "0095" <sip:0095@192.168.0.106:5061>;tag=as5b878ddb
To: <sip:0095@192.168.0.107:5060>;tag=000ab747a225753768d896cd-4ebc9ba0
Contact: <sip:0095@192.168.0.106:5061>
Call-ID: 5f66276e0c1721707972276818015f9a@192.168.0.106
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Destroying call '5f66276e0c1721707972276818015f9a@192.168.0.106'
<-- SIP read from 192.168.0.107:52930:
ACK sip:0095@192.168.0.106:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK7d259208
From: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
To: <sip:0095@192.168.0.50>;tag=as130e56a7
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 101 ACK
User-Agent: CSCO/7
Content-Length: 0
--- (8 headers 0 lines)---
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/9' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/0095/INBOX/msg0000 format: g729, 0x95b5aa8
Dec 28 15:24:43 WARNING[3747]: channel.c:2318 set_format: Unable to find a codec translation path from g729 to slin
Dec 28 15:24:43 WARNING[3747]: app.c:630 ast_play_and_record: Unable to set to linear mode, giving up
-- Recording was 0 seconds long but needs to be at least 3 - abandoning
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/0095-6ae1' in macro 'stdexten'
== Spawn extension (testing, 0095, 1) exited non-zero on 'SIP/0095-6ae1'
set_destination: Parsing <sip:0095@192.168.0.107:5060> for address/port to send to
set_destination: set destination to 192.168.0.107, port 5060
Reliably Transmitting (no NAT) to 192.168.0.107:5060:
BYE sip:0095@192.168.0.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK1f4527b5;rport
From: <sip:0095@192.168.0.50>;tag=as130e56a7
To: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
Contact: <sip:0095@192.168.0.106:5061>
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<-- SIP read from 192.168.0.107:52933:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.106:5061;branch=z9hG4bK1f4527b5;rport
From: <sip:0095@192.168.0.50>;tag=as130e56a7
To: "0095" <sip:0095@192.168.0.50>;tag=000ab747a22575363e3526aa-2433f63d
Call-ID: 000ab747-a22500db-1a429321-1090eb0a@192.168.0.107
CSeq: 102 BYE
Server: CSCO/7
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '000ab747-a22500db-1a429321-1090eb0a@192.168.0.107'