ABBAS SHAKEEL
2011-Jan-23 19:04 UTC
[asterisk-users] end a call after a specific time period
Hello all, I am trying to end a call after a specific time period for that reason i have tried various options like using S, L in the dial command. But in vain. Now i am thinking to end the call using the AMI... but i am unable to get the current active channel. . i.e SIP/NT000 when i ask for getchannel it return some thing like this Channel=Local/99449070380109 at default-f08b,2 As i have to use this channel name for hangup... Could some one let me know how to get channel name.... or any other way to end call.. Thanks in advance -- Best Regards Shakeel Abbas -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110123/6d81ce78/attachment.htm>
Sherwood McGowan
2011-Jan-23 19:14 UTC
[asterisk-users] end a call after a specific time period
On Sun, Jan 23, 2011 at 1:04 PM, ABBAS SHAKEEL <shakeel.abbas.qau at gmail.com> wrote:> Hello all, > I am trying to end a call after a specific time period for that reason i > have tried various options like using S, L in the dial command. But in vain. > Now i am thinking to end the call using the AMI... but i am unable to get > the current active channel. > . i.e SIP/NT000 when i ask for getchannel it return some thing like > this?Channel=Local/99449070380109 at default-f08b,2 > As i have to use this channel name for hangup... Could some one let me know > how to get channel name.... or any other way to end call.. > Thanks in advance > > -- > Best Regards > Shakeel Abbas > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ? ? ? ? ? ? ? http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >Easy solution is to store the channel name in a variable for later use, use ${CHANNEL} to get the channel name
ABBAS SHAKEEL
2011-Jan-23 19:19 UTC
[asterisk-users] end a call after a specific time period
Thank you very much for the prompt response.
I also tried this i.e in beginning i used Noop(${CHANNEL}) to get channel
name but it display some thing like ("Local/99449051010033 at
default-8307,2"
But i was expecting some thing like SIP/NT000
I spend alot of time on it but still in vain
On Sun, Jan 23, 2011 at 10:14 PM, Sherwood McGowan <
sherwood.mcgowan at gmail.com> wrote:
> On Sun, Jan 23, 2011 at 1:04 PM, ABBAS SHAKEEL
> <shakeel.abbas.qau at gmail.com> wrote:
> > Hello all,
> > I am trying to end a call after a specific time period for that reason
i
> > have tried various options like using S, L in the dial command. But in
> vain.
> > Now i am thinking to end the call using the AMI... but i am unable to
get
> > the current active channel.
> > . i.e SIP/NT000 when i ask for getchannel it return some thing like
> > this Channel=Local/99449070380109 at default-f08b,2
> > As i have to use this channel name for hangup... Could some one let me
> know
> > how to get channel name.... or any other way to end call..
> > Thanks in advance
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> Easy solution is to store the channel name in a variable for later
> use, use ${CHANNEL} to get the channel name
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Best Regards
Shakeel Abbas
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I believe absolute timeout will do that. http://www.voip-info.org/wiki/view/Asterisk+func+timeout On Sun, Jan 23, 2011 at 2:04 PM, ABBAS SHAKEEL <shakeel.abbas.qau at gmail.com> wrote:> Hello all, > I am trying to end a call after a specific time period for that reason i > have tried various options like using S, L in the dial command. But in vain. > Now i am thinking to end the call using the AMI... but i am unable to get > the current active channel. > . i.e SIP/NT000 when i ask for getchannel it return some thing like > this?Channel=Local/99449070380109 at default-f08b,2 > As i have to use this channel name for hangup... Could some one let me know > how to get channel name.... or any other way to end call.. > Thanks in advance > > -- > Best Regards > Shakeel Abbas > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ? ? ? ? ? ? ? http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >
ABBAS SHAKEEL
2011-Jan-31 19:34 UTC
[asterisk-users] end a call after a specific time period
Thanks you @ Godson Gera , @Sherwood McGowan , @ CF Thank you for mentioning. I have tried all the options (excluding AMI) but in vain. Let me show you what happens.... When the call starts core show channels shows me Channel Location State Application(Data) SIP/NTT00-00000000 99449046902115 at vicid Down AppDial((Outgoing Line)) Local/99449046902115 99449046902115 at defau Up Dial(SIP/NTT00/449046902115||o Local/99449046902115 8302 at default:2 Up Playback(conf) After a few seconds it shows 0*CLI> core show channels Channel Location State Application(Data) SIP/NTT00-00000000 8302 at default:2 Up Playback(conf) 1 active channel 1 active call Now the conf file is not that long that it keeps on playing. I dont know what else to use to end the call. On Mon, Jan 24, 2011 at 1:35 AM, C F <shmaltz at gmail.com> wrote:> I believe absolute timeout will do that. > http://www.voip-info.org/wiki/view/Asterisk+func+timeout > > > > On Sun, Jan 23, 2011 at 2:04 PM, ABBAS SHAKEEL > <shakeel.abbas.qau at gmail.com> wrote: >> Hello all, >> I am trying to end a call after a specific time period for that reason i >> have tried various options like using S, L in the dial command. But in vain. >> Now i am thinking to end the call using the AMI... but i am unable to get >> the current active channel. >> . i.e SIP/NT000 when i ask for getchannel it return some thing like >> this?Channel=Local/99449070380109 at default-f08b,2 >> As i have to use this channel name for hangup... Could some one let me know >> how to get channel name.... or any other way to end call.. >> Thanks in advance >> >> -- >> Best Regards >> Shakeel Abbas >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> ? ? ? ? ? ? ? http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> ? http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ? ? ? ? ? ? ? http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >-- Best Regards Shakeel Abbas
> > Channel ? ? ? ? ? ? ?Location ? ? ? ? ? ? State ? Application(Data) > SIP/NTT00-00000000 ? 99449046902115 at vicid Down ? ?AppDial((Outgoing Line)) > Local/99449046902115 99449046902115 at defau Up ? ? ?Dial(SIP/NTT00/449046902115||o > Local/99449046902115 8302 at default:2 ? ? ? Up ? ? ?Playback(conf) > > > After a few seconds it shows > > 0*CLI> core show channels > Channel ? ? ? ? ? ? ?Location ? ? ? ? ? ? State ? Application(Data) > SIP/NTT00-00000000 ? 8302 at default:2 ? ? ? Up ? ? ?Playback(conf) > 1 active channel > 1 active call > > > Now the conf file is not that long that it keeps on playing. > > I dont know what else to use to end the call.Post your dialplan please
ABBAS SHAKEEL
2011-Feb-01 14:29 UTC
[asterisk-users] end a call after a specific time period
exten => _9944NXXXXXXXXX,1,Answer()
exten => _9944NXXXXXXXXX,2,Noop(GOING FOR THE AGI)
exten => _9944NXXXXXXXXX,3,Noop(XXXXXXXXXXXXXXXXXX)
exten => _9944NXXXXXXXXX,4,Noop(XXXXXXXXXXXXXXXX)
exten => _9944NXXXXXXXXX,5,AGI(//Some script here it works perfectly fine)
exten => _9944NXXXXXXXXX,6,Noop(AGI ENDED)
exten => _9944NXXXXXXXXX,7,Set(TIMEOUT(absolute)=${TIME_OF_CALL_SECONDS})
exten =>
_9944NXXXXXXXXX,8,Dial(${VICITRUNK2}/${EXTEN:2},35,goL(${TIME_OF_CALL_SECONDS}))
exten => _9944NXXXXXXXXX,9,Hangup
I have tried many options but in vain. Thanks in advance
On Tue, Feb 1, 2011 at 3:40 AM, C F <shmaltz at gmail.com>
wrote:>>
>> Channel ? ? ? ? ? ? ?Location ? ? ? ? ? ? State ? Application(Data)
>> SIP/NTT00-00000000 ? 99449046902115 at vicid Down ? ?AppDial((Outgoing
Line))
>> Local/99449046902115 99449046902115 at defau Up ? ?
?Dial(SIP/NTT00/449046902115||o
>> Local/99449046902115 8302 at default:2 ? ? ? Up ? ? ?Playback(conf)
>>
>>
>> After a few seconds it shows
>>
>> 0*CLI> core show channels
>> Channel ? ? ? ? ? ? ?Location ? ? ? ? ? ? State ? Application(Data)
>> SIP/NTT00-00000000 ? 8302 at default:2 ? ? ? Up ? ? ?Playback(conf)
>> 1 active channel
>> 1 active call
>>
>>
>> Now the conf file is not that long that it keeps on playing.
>>
>> I dont know what else to use to end the call.
>
>
> Post your dialplan please
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> ? ? ? ? ? ? ? http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ? http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Best Regards
Shakeel Abbas