My architecture is:
PSTN-----E1----OldPBX----E1-----Asterisk
I've a similar problem, SIP user agents using X-Lite:
Sip User Agent "A" calls to PSTN user "B"
"B" user hangs the call
"A" user starts listening busy indications on the phone, and if he doesn't hangup correctly on Xlite
The calls seems to be alive.... Only solved it with soft hangup, and that is not an acceptable solution.
I have on user that seems to have turned off the pc ( at least he reports me that) and the call (at least on Asterisk CDR) remained alive....didn't disconnect....
It is working fine only if SIP user agents dials to an extension in the Old PBX, that case if the called party Hangs, the Old Pbx immediately sends a DISCONNECT message to Asterisk and the call hangs...
I hope someone could help US.
Best regards,
Marco Mouta
Hi,
We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel.
When agents are dialing, channels doesn't show calls
vicidial2*CLI> show channels
Channel Location State Application(Data)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600051|q)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600)
SIP/primus-8f43 (None) Ringing AppDial((Outgoing Line))
Local/[EMAIL PROTECTED] [EMAIL PROTECTED] Ring Dial(SIP/[EMAIL PROTECTED]||t
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Down (None)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600053|q)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600)
SIP/primus-00fe (None) Ringing AppDial((Outgoing Line))
Local/[EMAIL PROTECTED] [EMAIL PROTECTED] Ring Dial(SIP/[EMAIL PROTECTED]||t
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Down (None)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600054|q)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600)
SIP/primus-95db [EMAIL PROTECTED]:1 Up MeetMe(8600051)
Zap/pseudo-122590356 [EMAIL PROTECTED]:1 Rsrvd (None)
SIP/agent7-44fa [EMAIL PROTECTED]:1 Up MeetMe(8600055)
SIP/primus-0a7c [EMAIL PROTECTED]:1 Up MeetMe(8600053)
SIP/primus-7c73 [EMAIL PROTECTED]:1 Up MeetMe(8600054)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1 Up MeetMe(8600052|q)
Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:3 Up Wait(3600)
SIP/primus-2ed8 [EMAIL PROTECTED]:1 Up MeetMe(8600052)
Zap/pseudo-104079549 [EMAIL PROTECTED]:1 Rsrvd (None)
SIP/agent1-32b5 [EMAIL PROTECTED]:1 Up MeetMe(8600054)
Zap/pseudo-204709889 [EMAIL PROTECTED]:1 Rsrvd (None)
SIP/agent8-d3ab [EMAIL PROTECTED]:1 Up MeetMe(8600056)
SIP/agent5-ec77 [EMAIL PROTECTED]:1 Up MeetMe(8600051)
Zap/pseudo-926666046 [EMAIL PROTECTED]:1 Rsrvd (None)
SIP/agent3-2df5 [EMAIL PROTECTED]:1 Up MeetMe(8600053)
Zap/pseudo-204290210 [EMAIL PROTECTED]:1 Rsrvd (None)
SIP/agent2-4ff6 [EMAIL PROTECTED]:1 Up MeetMe(8600052)
SIP/primus-fc90 [EMAIL PROTECTED]:1 Up MeetMe(8600051)
Zap/pseudo-170346238 [EMAIL PROTECTED]:1 Rsrvd (None)
31 active channels
After agents have logged out
vicidial2*CLI> show channels
Channel Location State Application(Data)
SIP/primus-fc90 [EMAIL PROTECTED]:1 Up MeetMe(8600051)
Zap/pseudo-170346238 [EMAIL PROTECTED]:1 Rsrvd (None)
Calls doesn't show channels
vicidial2*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
203.63.248.197 122001 20a58a2e251 00651/00000 unkn No
203.196.128.56 6135625116 5420f80176e 00102/00000 g729 No Tx: ACK
calls doesn't show channel
CLI>sip show channel 5420f80176e* SIP Call
Direction: Outgoing
Call-ID: [EMAIL PROTECTED]
Our Codec Capability: 256
Non-Codec Capability: 1
Their Codec Capability: 256
Joint Codec Capability: 256
Format g729
Theoretical Address: 203.196.128.56:5060
Received Address: 203.196.128.56:5060
NAT Support: RFC3581
Audio IP: 220.227.174.4 (local)
Our Tag: as7a55ac7a
Their Tag: 29258
SIP User agent:
Username: 61356251162
Peername: 90340
Original uri: sip:[EMAIL PROTECTED]:5060
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:[EMAIL PROTECTED];ftag=as7a55ac7a;lr=on
DTMF Mode: rfc2833
SIP Options: (none)
BUT ONE THING IS COMMON IS THAT OLDEST SIP CALL WILL COME IN THE BOTTOM OF THE LIST of COMMAND sip show channels (agents will be above it) so it is hung and needs to be destroyed manually. Also channel corresponding to this call will also come in the bottom of SHOW Channels command for same technology i.e. it will be last SIP/XYZ entry so to destroy this call lets try destroy last SIP channel entry.vicidial2*CLI> soft hangup SIP/primus-fc90
Requested Hangup on channel 'SIP/primus-fc90'
-- Hungup 'Zap/pseudo-1703462386'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/primus-fc90'
-- Executing DeadAGI("SIP/primus-fc90", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("SIP/primus-fc90", "VD_hangup.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
vicidial2*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
0 active SIP channels
IT WORKS!! A crude way but very important to save 100 of dollars of hung call while agent are dialing. You can always do stop now but then whole operations will stop.
Dont know why this happens in first place but atleast I have seen it coming twice and now keep a vigil that no call is below the agents in sip show channels, it there is any it means its a hung call costing you money
Abhimanyu
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