[asterisk-users] Originate from the Dialplan
Hi all,
I an using the Originate() dialplan command but I cant get it to save cdr's.
Here is the line I am using:
exten =>
_61X,53,Originate(SIP/${TRUNK}/${PREFIX}${PHONE},exten,${DESTCONTEXT},${PHONE},1);
The call goes out fine, but CDR's get inserted into the DB.
Any ideas on why this is happening? Is it a bug or a feature?
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Re: [asterisk-users] Multiple Channel Variables with AMI Originate
Thanks Jim. We're using the 1.6.2 rc's. I'll try it on 1.6.0 and see if it works. If it does then I guess its a bug in the new releases. Problem is we need features of the 1.6.2. On Fri, Dec 4, 2009 at 11:03 AM, Jim Dickenson wrote: > I do this: > > Action: Originate > Channel: Local/dial_num...@cfmc_cdi_private > Exten: queue_answer > Context: cfmc_cdi_private > Priority: 1 > Variable: CfMC_ActionID=CallAndQueue > Variable: CfMC_QueueToUse=tqe > Variable: CfMC_AgentToUse=1001 > Variable: CfMC_DialInfo=SIP/GXP280_18 > Variable: CfMC_RingTimeout=30 > Variable: CfMC_DoAMD=No > ActionID: CallAndQueue > Async: true > > > and things work as expected. This is in recent versions of 1.4 and 1.6.0. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > On Dec 3, 2009, at 4:22 PM, Matthew Edmondson wrote: > > > Hi guys I seem to be having a problem, I don't know if it's a bug or > whether I'm just doing it incorrectly. > > I want to set about 3 channel variables when I originate a call via AMI. > > All the documentation I have found says to do it like this: > > Variable: variable1=value|variable2=value|variable3=value > > > However when I do this it runs them all together and I end up with: > > variable1 = "value|variable2=value|variable3=value" > > > Instead of: > > variable1 = "value" > variable2 = "value" > variable3 = "value" > > > So I think the delimiters are not working. > > I tried just adding multiple "Variable:" lines one for each, but it only > sets the first one and ignores the rest. > > Any help with this would be greatly appreciated. > > Thanks! --Matt > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Channel Variables with AMI Originate
Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with: variable1 = "value|variable2=value|variable3=value" Instead of: variable1 = "value" variable2 = "value" variable3 = "value" So I think the delimiters are not working. I tried just adding multiple "Variable:" lines one for each, but it only sets the first one and ignores the rest. Any help with this would be greatly appreciated. Thanks! --Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play audio file within an active call
I didn't think of Benny's solution. That would be the way to go as it is a
core asterisk command.
On Tue, Sep 29, 2009 at 6:47 PM, Vieri wrote:
>
> --- On Tue, 9/29/09, Matthew Edmondson wrote:
>
> > If you redirect the channel, the
> > person they're talking to is likely
> > to be dropped.
>
> Thanks for pointing that out. So it sounds like RedirectChannel() is
> similar to Transfer().
>
> > The only way I know of doing this is with a conference
> > bridge like
> > meetme. You would have to have both parties in the
> > conference and then
> > call the 3rd party (your msg) into it.
>
> This may be trivial but how can I "force" both parties to enter a
> conference (eg. meetme)?
>
> Also, once they're in the conference and I've "called" a third party
> playing a sound file, how can I "force" them to exit the conference and
> revert to their bridged call as before the conference? Or, if I have to keep
> them within the conference/meetme, then I'd have to make sure that the "3rd
> party" can :
> 1- play a msg such as "on the phone too long; consider hanging up"
> 2- wait N minutes
> 3- play "you have Z minutes of conversation left. Call will be hung up
> automatically"
> 4- hang up both parties in conference
>
> Does this make sense?
>
> Thanks,
>
> Vieri
>
> > On Tue, Sep 29, 2009 at 6:05 PM, Vieri
> > wrote:
> > > Hi,
> > >
> > > I'm wondering if someone can share their thoughts on
> > how to implement a system that periodically checks active
> > channels which have been up for more than X minutes and
> > plays/injects a sound file. The idea is to simply warn users
> > that they've been on the phone for quite a while and maybe
> > they should consider hanging up. If the call stays up for
> > more than Y minutes, it is dropped automatically
> > (softhangup).
> > >
> > > What's the simplest approach to playing a sound file
> > within an active channel?
> > >
> > > I thought of writing a cron agi script that scans
> > active channels, retrieves their duration and if it's > X
> > minutes then "RedirectChannel" to a context which executes a
> > Playback(file); if it's > Y minutes then
> > "RedirectChannel" to a context which executes both a
> > Playback("forcing hang up now...") and HangUp.
> > >
> > > Any thoughts?
> > >
>
>
>
>
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Re: [asterisk-users] play audio file within an active call
If you redirect the channel, the person they're talking to is likely
to be dropped.
The only way I know of doing this is with a conference bridge like
meetme. You would have to have both parties in the conference and then
call the 3rd party (your msg) into it.
On Tue, Sep 29, 2009 at 6:05 PM, Vieri wrote:
> Hi,
>
> I'm wondering if someone can share their thoughts on how to implement a
> system that periodically checks active channels which have been up for more
> than X minutes and plays/injects a sound file. The idea is to simply warn
> users that they've been on the phone for quite a while and maybe they should
> consider hanging up. If the call stays up for more than Y minutes, it is
> dropped automatically (softhangup).
>
> What's the simplest approach to playing a sound file within an active channel?
>
> I thought of writing a cron agi script that scans active channels, retrieves
> their duration and if it's > X minutes then "RedirectChannel" to a context
> which executes a Playback(file); if it's > Y minutes then "RedirectChannel"
> to a context which executes both a Playback("forcing hang up now...") and
> HangUp.
>
> Any thoughts?
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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