rendered paste body[Sep 28 17:23:44] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
INVITE sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
Contact: <sip:320@192.168.15.189>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 278
v=0
o=- 1362 1362 IN IP4 192.168.15.189
s=SIP Session
c=IN IP4 192.168.15.189
t=0 0
m=audio 10004 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
[Sep 28 17:23:44] VERBOSE[20826] logger.c: --- (10 headers 13 lines)
[Sep 28 17:23:44] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Sending to 192.168.15.189 : 5060 (no NAT)
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Using INVITE request as basis request - 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found user '320'
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found RTP audio format 0
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found RTP audio format 8
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found RTP audio format 18
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found RTP audio format 4
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found RTP audio format 101
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Peer audio RTP is at port 192.168.15.189:10004
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found description format PCMU for ID 0
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found description format PCMA for ID 8
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found description format G729 for ID 18
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found description format G723 for ID 4
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Found description format telephone-event for ID 101
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Peer audio RTP is at port 192.168.15.189:10004
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Looking for 119 in mx37 (domain 192.168.15.222)
[Sep 28 17:23:44] VERBOSE[20826] logger.c: list_route: hop: <sip:320@192.168.15.189>
[Sep 28 17:23:44] VERBOSE[20826] logger.c: Transmitting (no NAT) to 192.168.15.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Length: 0
---
[Sep 28 17:23:44] VERBOSE[20881] logger.c: -- Executing [119@mx37:1] Dial("SIP/320-08207870", "Zap/g2/119") in new stack
[Sep 28 17:23:44] VERBOSE[20881] logger.c: -- Requested transfer capability: 0x00 - SPEECH
[Sep 28 17:23:44] DEBUG[20815] channel.c: Avoiding initial deadlock for channel '0x820bba0'
[Sep 28 17:23:44] VERBOSE[20881] logger.c: -- Called g2/119
[Sep 28 17:23:44] VERBOSE[20881] logger.c: Audio is at 192.168.15.222 port 56312
[Sep 28 17:23:44] VERBOSE[20881] logger.c: Adding codec 0x4 (ulaw) to SDP
[Sep 28 17:23:44] VERBOSE[20881] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 28 17:23:44] VERBOSE[20881] logger.c: Transmitting (no NAT) to 192.168.15.189:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20811 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:44] DEBUG[20830] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 2
[Sep 28 17:23:44] VERBOSE[20881] logger.c: -- Zap/32-1 is proceeding passing it to SIP/320-08207870
[Sep 28 17:23:44] DEBUG[20830] chan_zap.c: No echo cancellation requested
[Sep 28 17:23:44] VERBOSE[20881] logger.c: -- Zap/32-1 is ringing
[Sep 28 17:23:44] VERBOSE[20881] logger.c: Transmitting (no NAT) to 192.168.15.189:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Length: 0
---
[Sep 28 17:23:44] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:44] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:44] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:45] DEBUG[20830] chan_zap.c: No echo cancellation requested
[Sep 28 17:23:45] VERBOSE[20881] logger.c: -- Zap/32-1 answered SIP/320-08207870
[Sep 28 17:23:45] VERBOSE[20881] logger.c: Audio is at 192.168.15.222 port 56312
[Sep 28 17:23:45] VERBOSE[20881] logger.c: Adding codec 0x4 (ulaw) to SDP
[Sep 28 17:23:45] VERBOSE[20881] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 28 17:23:45] VERBOSE[20881] logger.c: Reliably Transmitting (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:46] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:46] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:46] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:46] VERBOSE[20826] logger.c: Retransmitting #1 (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:47] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:47] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:47] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:47] VERBOSE[20826] logger.c: Retransmitting #2 (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:48] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:48] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:48] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:49] VERBOSE[20826] logger.c: Retransmitting #3 (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:50] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:50] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:50] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:53] VERBOSE[20826] logger.c: Retransmitting #4 (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:54] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:54] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:54] VERBOSE[20826] logger.c: ---
[Sep 28 17:23:57] VERBOSE[20826] logger.c: Retransmitting #5 (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:23:58] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:23:58] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:23:58] VERBOSE[20826] logger.c: ---
[Sep 28 17:24:01] VERBOSE[20826] logger.c: Retransmitting #6 (no NAT) to 192.168.15.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK8d5c03a5e6eabc8dcbfe8d9ce4d3540f;received=192.168.15.189
From: 320 <sip:320@192.168.15.222>;tag=2020a833
To: 119 <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:119@192.168.15.222>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20811 20812 IN IP4 192.168.15.222
s=session
c=IN IP4 192.168.15.222
t=0 0
m=audio 56312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
[Sep 28 17:24:02] VERBOSE[20826] logger.c:
<-- SIP read from 192.168.15.189:5060:
ACK sip:119@192.168.15.222 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.189:5060;branch=z9hG4bK526a79cba6014e7059c08e7f363b022b
From: <sip:320@192.168.15.222>;tag=2020a833
To: <sip:119@192.168.15.222>;tag=as52d29457
Call-ID: 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
[Sep 28 17:24:02] VERBOSE[20826] logger.c: --- (8 headers 0 lines)[Sep 28 17:24:02] VERBOSE[20826] logger.c: ---
[Sep 28 17:24:05] WARNING[20826] chan_sip.c: Maximum retries exceeded on transmission 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189 for seqno 1 (Critical Response)
[Sep 28 17:24:05] WARNING[20826] chan_sip.c: Hanging up call 757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189 - no reply to our critical packet.
[Sep 28 17:24:05] DEBUG[20881] channel.c: Didn't get a frame from channel: SIP/320-08207870
[Sep 28 17:24:05] DEBUG[20881] channel.c: Bridge stops bridging channels SIP/320-08207870 and Zap/32-1
[Sep 28 17:24:05] DEBUG[20881] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/32-1
[Sep 28 17:24:05] DEBUG[20881] chan_zap.c: Hangup: channel: 32 index = 0, normal = 41, callwait = -1, thirdcall = -1
[Sep 28 17:24:05] DEBUG[20881] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call
[Sep 28 17:24:05] DEBUG[20881] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/32-1
[Sep 28 17:24:05] DEBUG[20881] chan_zap.c: Updated conferencing on 32, with 0 conference users
[Sep 28 17:24:05] DEBUG[20881] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/32-1
[Sep 28 17:24:05] VERBOSE[20881] logger.c: -- Hungup 'Zap/32-1'
[Sep 28 17:24:05] VERBOSE[20881] logger.c: == Spawn extension (mx37, 119, 1) exited non-zero on 'SIP/320-08207870'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '"V.Pelih" <320>'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '320'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '119'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'mx37'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'SIP/320-08207870'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'Zap/32-1'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'Dial'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'Zap/g2/119'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '2006-09-28 17:23:44'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '2006-09-28 17:23:45'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '2006-09-28 17:24:05'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '21'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '20'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'ANSWERED'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is 'DOCUMENTATION'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is ''
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is '1159424624.4'
[Sep 28 17:24:05] DEBUG[20881] pbx.c: Function result is ''
[Sep 28 17:24:06] VERBOSE[20826] logger.c: Really destroying SIP dialog '757bf1bd08a05b0318dfd07f134cb43f@192.168.15.189' Method: ACK