AGC shows call still connected after hangup from SIP channel
Posted: Mon Feb 09, 2009 2:53 am
We have successfully installed and configured ViciDialNOW and run a number of outbound calls in manual and automatic dial modes. Our outbound calls are using SIP channels but when the far end hangs up it is not being detected by ViciDial.
We receive a BYE:
U 147.202.001.001:5060 -> 124.198.001.001:5060
BYE sip:6411223344@124.198.001.001 SIP/2.0
Record-Route: <sip:147.202.001.001;lr=on;ftag=as15a4b3ea>
Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK7cfa.057f8bd7.0
Via: SIP/2.0/UDP 202.180.001.001:5060;branch=z9hG4bK52400902;rport=5060
From: <sip:094466548@Domain.co.nz>;tag=as15a4b3ea
To: "M0209204132000000047" <sip:6411223344@Domain.co.nz>;tag=as1ba4a5b1
Call-ID: 09f788a30fd682611aded6aa6dcc2351@Domain.co.nz
CSeq: 102 BYE
User-Agent: Domain SIP proxy
Max-Forwards: 69
Content-Length: 0
The Asterisk CLI shows the call being hungup (note that I have added the NoOps for clarification):
-- Executing NoOp("Local/8600051@default-b733,1", "in h extension") in new stack
-- Executing Hangup("Local/8600051@default-b733,1", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'Local/8600051@default-b733,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-b733,2'
-- Executing DeadAGI("Local/8600051@default-b733,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing NoOp("Local/8600051@default-b733,2", "in h extension") in new stack
-- Executing Hangup("Local/8600051@default-b733,2", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'Local/8600051@default-b733,2'
But the web page shows that the call is still connected. The call is not disconnected and the disposition screen doesn't pop until the user clicks the Hangup customer button.
Any suggestions on what we have done wrong would be appreciated.
Regards
Cameron
We receive a BYE:
U 147.202.001.001:5060 -> 124.198.001.001:5060
BYE sip:6411223344@124.198.001.001 SIP/2.0
Record-Route: <sip:147.202.001.001;lr=on;ftag=as15a4b3ea>
Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK7cfa.057f8bd7.0
Via: SIP/2.0/UDP 202.180.001.001:5060;branch=z9hG4bK52400902;rport=5060
From: <sip:094466548@Domain.co.nz>;tag=as15a4b3ea
To: "M0209204132000000047" <sip:6411223344@Domain.co.nz>;tag=as1ba4a5b1
Call-ID: 09f788a30fd682611aded6aa6dcc2351@Domain.co.nz
CSeq: 102 BYE
User-Agent: Domain SIP proxy
Max-Forwards: 69
Content-Length: 0
The Asterisk CLI shows the call being hungup (note that I have added the NoOps for clarification):
-- Executing NoOp("Local/8600051@default-b733,1", "in h extension") in new stack
-- Executing Hangup("Local/8600051@default-b733,1", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'Local/8600051@default-b733,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-b733,2'
-- Executing DeadAGI("Local/8600051@default-b733,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing NoOp("Local/8600051@default-b733,2", "in h extension") in new stack
-- Executing Hangup("Local/8600051@default-b733,2", "") in new stack
== Spawn extension (default, h, 3) exited non-zero on 'Local/8600051@default-b733,2'
But the web page shows that the call is still connected. The call is not disconnected and the disposition screen doesn't pop until the user clicks the Hangup customer button.
Any suggestions on what we have done wrong would be appreciated.
Regards
Cameron