Hi all,
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
On 4/12/06, Abhimanyu Rapria <Abhimanyu@synotek.com>
wrote:>
> 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/78600051@defau 78600051@default:1 Up MeetMe(8600051|q)
>
> Local/78600051@defau 8309@default:3 Up Wait(3600)
>
> SIP/primus-8f43 (None) Ringing AppDial((Outgoing Line))
>
> Local/761394353177@d 761394353177@default Ring Dial(
> SIP/61394353177@primus||t
>
> Local/761394353177@d s@default:1 Down (None)
>
> Local/78600053@defau 78600053@default:1 Up MeetMe(8600053|q)
>
> Local/78600053@defau 8309@default:3 Up Wait(3600)
>
> SIP/primus-00fe (None) Ringing AppDial((Outgoing Line))
>
> Local/761394357078@d 761394357078@default Ring Dial(
> SIP/61394357078@primus||t
>
> Local/761394357078@d s@default:1 Down (None)
>
> Local/78600054@defau 78600054@default:1 Up MeetMe(8600054|q)
>
> Local/78600054@defau 8309@default:3 Up Wait(3600)
>
> SIP/primus-95db 8600051@default:1 Up MeetMe(8600051)
>
> Zap/pseudo-122590356 s@default:1 Rsrvd (None)
>
> SIP/agent7-44fa 8600055@default:1 Up MeetMe(8600055)
>
> SIP/primus-0a7c 8600053@default:1 Up MeetMe(8600053)
>
> SIP/primus-7c73 8600054@default:1 Up MeetMe(8600054)
>
> Local/78600052@defau 78600052@default:1 Up MeetMe(8600052|q)
>
> Local/78600052@defau 8309@default:3 Up Wait(3600)
>
> SIP/primus-2ed8 8600052@default:1 Up MeetMe(8600052)
>
> Zap/pseudo-104079549 s@default:1 Rsrvd (None)
>
> SIP/agent1-32b5 8600054@default:1 Up MeetMe(8600054)
>
> Zap/pseudo-204709889 s@default:1 Rsrvd (None)
>
> SIP/agent8-d3ab 8600056@default:1 Up MeetMe(8600056)
>
> SIP/agent5-ec77 8600051@default:1 Up MeetMe(8600051)
>
> Zap/pseudo-926666046 s@default:1 Rsrvd (None)
>
> SIP/agent3-2df5 8600053@default:1 Up MeetMe(8600053)
>
> Zap/pseudo-204290210 s@default:1 Rsrvd (None)
>
> SIP/agent2-4ff6 8600052@default:1 Up MeetMe(8600052)
>
> SIP/primus-fc90 8600051@default:1 Up MeetMe(8600051)
>
> Zap/pseudo-170346238 s@default:1 Rsrvd (None)
>
> 31 active channels
>
> After agents have logged out
>
>
> vicidial2*CLI> show channels
> Channel Location State Application(Data)
> SIP/primus-fc90 8600051@default:1 Up MeetMe(8600051)
> Zap/pseudo-170346238 s@default: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: 5420f80176e3e56679ce4e537ffdbd3f@203.196.128.56
>
> 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:61356251162@216.181.122.44:5060
>
> Need Destroy: 0
>
> Last Message: Tx: ACK
>
> Promiscuous Redir: No
>
> Route: sip:61356251162@203.196.128.56
> ;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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