[Asterisk-Users] SIP call hangup from asterisk CLI

Marco Mouta marco.mouta at gmail.com
Wed Apr 12 04:04:19 MST 2006


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 at 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 at defau 78600051 at default:1   Up      MeetMe(8600051|q)
>
> Local/78600051 at defau 8309 at default:3       Up      Wait(3600)
>
> SIP/primus-8f43      (None)               Ringing AppDial((Outgoing Line))
>
> Local/761394353177 at d 761394353177 at default Ring    Dial(
> SIP/61394353177 at primus||t
>
> Local/761394353177 at d s at default:1          Down    (None)
>
> Local/78600053 at defau 78600053 at default:1   Up      MeetMe(8600053|q)
>
> Local/78600053 at defau 8309 at default:3       Up      Wait(3600)
>
> SIP/primus-00fe      (None)               Ringing AppDial((Outgoing Line))
>
> Local/761394357078 at d 761394357078 at default Ring    Dial(
> SIP/61394357078 at primus||t
>
> Local/761394357078 at d s at default:1          Down    (None)
>
> Local/78600054 at defau 78600054 at default:1   Up      MeetMe(8600054|q)
>
> Local/78600054 at defau 8309 at default:3       Up      Wait(3600)
>
> SIP/primus-95db      8600051 at default:1    Up      MeetMe(8600051)
>
> Zap/pseudo-122590356 s at default:1          Rsrvd   (None)
>
> SIP/agent7-44fa      8600055 at default:1    Up      MeetMe(8600055)
>
> SIP/primus-0a7c      8600053 at default:1    Up      MeetMe(8600053)
>
> SIP/primus-7c73      8600054 at default:1    Up      MeetMe(8600054)
>
> Local/78600052 at defau 78600052 at default:1   Up      MeetMe(8600052|q)
>
> Local/78600052 at defau 8309 at default:3       Up      Wait(3600)
>
> SIP/primus-2ed8      8600052 at default:1    Up      MeetMe(8600052)
>
> Zap/pseudo-104079549 s at default:1          Rsrvd   (None)
>
> SIP/agent1-32b5      8600054 at default:1    Up      MeetMe(8600054)
>
> Zap/pseudo-204709889 s at default:1          Rsrvd   (None)
>
> SIP/agent8-d3ab      8600056 at default:1    Up      MeetMe(8600056)
>
> SIP/agent5-ec77      8600051 at default:1    Up      MeetMe(8600051)
>
> Zap/pseudo-926666046 s at default:1          Rsrvd   (None)
>
> SIP/agent3-2df5      8600053 at default:1    Up      MeetMe(8600053)
>
> Zap/pseudo-204290210 s at default:1          Rsrvd   (None)
>
> SIP/agent2-4ff6      8600052 at default:1    Up      MeetMe(8600052)
>
> SIP/primus-fc90      8600051 at default:1    Up      MeetMe(8600051)
>
> Zap/pseudo-170346238 s at 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 at default:1    Up      MeetMe(8600051)
> Zap/pseudo-170346238 s at 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 at 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 at 216.181.122.44:5060
>
>   Need Destroy:           0
>
>   Last Message:           Tx: ACK
>
>   Promiscuous Redir:      No
>
>   Route:                  sip:61356251162 at 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
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060412/79088afe/attachment.htm


More information about the asterisk-users mailing list