rendered paste body -- outgoing agentcall, to agent '1001', on 'Local/337@agent-handler-0f90,1'
-- Called Agent/1001
-- Executing [337@agent-handler:1] NoOp("Local/337@agent-handler-0f90,2", "-- Got 337 --") in new stack
-- Executing [337@agent-handler:2] Dial("Local/337@agent-handler-0f90,2", "SIP/337|25|t") in new stack
Audio is at 10.2.2.1 port 12024
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.2.2.218:2088:
INVITE sip:337@10.2.2.218:2088;line=xjtqfh5j SIP/2.0
Via: SIP/2.0/UDP 10.2.2.1:5060;branch=z9hG4bK1df15bb0;rport
From: "0184628106" <sip:0184628106@10.2.2.1>;tag=as205751f8
To: <sip:337@10.2.2.218:2088;line=xjtqfh5j>
Contact: <sip:0184628106@10.2.2.1>
Call-ID: 61d9b2501d0626aa62252820117da20b@10.2.2.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Mar 2007 10:36:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 28915 28915 IN IP4 10.2.2.1
s=session
c=IN IP4 10.2.2.1
t=0 0
m=audio 12024 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 337
sawvoice*CLI>
<--- SIP read from 10.2.2.218:2088 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.2.1:5060;branch=z9hG4bK1df15bb0;rport=5060
From: "0184628106" <sip:0184628106@10.2.2.1>;tag=as205751f8
To: <sip:337@10.2.2.218:2088;line=xjtqfh5j>;tag=64exieayyk
Call-ID: 61d9b2501d0626aa62252820117da20b@10.2.2.1
CSeq: 102 INVITE
Contact: <sip:337@10.2.2.218:2088;line=xjtqfh5j>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/337-09b1ae68 is ringing
-- Agent/1001 is ringing
-- Nobody picked up in 10000 ms
-- Notifying manager of RTQ on interface 'Agent/1001' in queue 'claims-301'
Scheduling destruction of SIP dialog '61d9b2501d0626aa62252820117da20b@10.2.2.1' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.2.2.218:2088:
CANCEL sip:337@10.2.2.218:2088;line=xjtqfh5j SIP/2.0
Via: SIP/2.0/UDP 10.2.2.1:5060;branch=z9hG4bK1df15bb0;rport
From: "0184628106" <sip:0184628106@10.2.2.1>;tag=as205751f8
To: <sip:337@10.2.2.218:2088;line=xjtqfh5j>
Call-ID: 61d9b2501d0626aa62252820117da20b@10.2.2.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '61d9b2501d0626aa62252820117da20b@10.2.2.1' in 6400 ms (Method: INVITE)
== Spawn extension (agent-handler, 337, 2) exited non-zero on 'Local/337@agent-handler-0f90,2'
sawvoice*CLI>
<--- SIP read from 10.2.2.218:2088 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.2.1:5060;branch=z9hG4bK1df15bb0;rport=5060
From: "0184628106" <sip:0184628106@10.2.2.1>;tag=as205751f8
To: <sip:337@10.2.2.218:2088;line=xjtqfh5j>;tag=64exieayyk
Call-ID: 61d9b2501d0626aa62252820117da20b@10.2.2.1
CSeq: 102 CANCEL
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
sawvoice*CLI>
<--- SIP read from 10.2.2.218:2088 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.2.1:5060;branch=z9hG4bK1df15bb0;rport=5060
From: "0184628106" <sip:0184628106@10.2.2.1>;tag=as205751f8
To: <sip:337@10.2.2.218:2088;line=xjtqfh5j>;tag=64exieayyk
Call-ID: 61d9b2501d0626aa62252820117da20b@10.2.2.1
CSeq: 102 INVITE
Contact: <sip:337@10.2.2.218:2088;line=xjtqfh5j>;flow-id=1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.2.2.218:2088:
ACK sip:337@10.2.2.218:2088;line=xjtqfh5j SIP/2.0
Via: SIP/2.0/UDP 10.2.2.1:5060;branch=z9hG4bK1df15bb0;rport
From: "0184628106" <sip:0184628106@10.2.2.1>;tag=as205751f8
To: <sip:337@10.2.2.218:2088;line=xjtqfh5j>;tag=64exieayyk
Contact: <sip:0184628106@10.2.2.1>
Call-ID: 61d9b2501d0626aa62252820117da20b@10.2.2.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---