Hi Andre, That's interesting info.
I will try the suggestion and play around with it to see if it will do the trick. Elliott > -----Original Message----- > From: Andre Courchesne - Consultant [mailto:[email protected]] > Sent: Thursday, June 18, 2009 9:16 PM > To: Elliott Jeyaseelan > Cc: [email protected] > Subject: Re: [on-asterisk] Paging question > > Hi, > > For sure you can not do that all within freePBX, some > custom coding is required. > > For the cyberdata hangup when the line is picked-up, Here > is the sequence I would try to do. > > 1. Someone presses on the intercom button > 2. The call goes in a custom code that: > A. Generates a call file to context intercom-speaker > B. Dials a bumch of extension (similar to a > ring group), but uses the > M(x) parameter. Something like M(hangup-intercom-speaker. > > The intercom-speaker context which would be called by the > callfile would first make an astdb entry of the channel it is > using, then play either a wav file (Playback) or a tone > (PlayTone) for a very long time. > > The macro-hangup-intercom-speaker context which would be > called when someone answers the phone on the previously > calles list of extension would fetch the astdb entry made in > the intercom-speaker context (let's name it mychannel) and do > something like a System(asterisk -rx "soff hangup > ${mychannel}) which should hangup the cyberdata channel. > > This should in theory work for one intercom and 1 > cyberdata. If you have multiple intercom, you need to find a > way to identify the mychannel... > > > ---- > Andre Courchesne - Consultant > http://www.net-forces.com > Phone: (514) 667-8448 > MSN: [email protected] > Skype: VoipForces > > L'information contenue dans le présent document est la propriété de > Andre Courchesne. Et est divulguée en toute confidentialité. Cette > information ne doit pas être utilisée, divulguée à d'autres > personnes ou > reproduite sans le consentement écrit explicite de Andre Courchesne. > > The information contained in this document is confidential > and property > of Andre Courchesne. It shall not be used, disclosed to others or > reproduced without the express written consent of Andre Courchesne. > > Elliott Jeyaseelan wrote: > > Andre, > > > > That sounds like what I am tring to achieve. > > > > Not sure where to even start in Freepbx, but I am going to > give it a try > > etc. > > > > Thanks for that idea... > > > > Elliott > > > >> -----Original Message----- > >> From: Andre Courchesne - Consultant > [mailto:[email protected]] > >> Sent: Thursday, June 18, 2009 2:24 PM > >> To: Elliott Jeyaseelan > >> Cc: [email protected] > >> Subject: Re: [on-asterisk] Paging question > >> > >> I did something similar for doing a page group on a failover > >> for a queue where a page would be sent out and the caller be > >> put in an overflow queue. > >> > >> The idea is to make a script that would create a callfile to > >> do the paging and at the "same time" send the caller to > the ringgroup. > >> > >> The callfile would go in a context that would do a page on > >> your cyberdata playing your tone. > >> > >> > >> Elliott Jeyaseelan wrote: > >>> Need to pick your brain folks, I am sure someone has tackled this > >>> before. > >>> > >>> I have a cyberdata voip paging amplifier. Its registered > >> with Asterisk > >>> as a SIP extension. > >>> > >>> The voip amp automatically answers any incoming call and > >> plays it to the > >>> horn. > >>> > >>> I have a scenario where there is an Intercom call box, > >> someone pushes > >>> the Call button, it rings a ring-group, I want the paging > >> amp to play a > >>> tone and other phones in the ring group to ring. > >>> > >>> When one of the phones answers, I would like the paging amp > >> to hang up. > >>> Problem is, the paging amp being factory set to > auto-answer, ends up > >>> getting the call and none of the other phones in the ring > group ever > >>> ring, obviously nature of the ring group and the paging amp > >> pickup the > >>> call. > >>> > >>> Any suggestions on how to go about implementing, I am > using Freepbx, > >>> hoping your suggestion is can be implemented on Freepbx. > >>> > >>> Thanks. > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >>> To unsubscribe, e-mail: [email protected] > >>> For additional commands, e-mail: [email protected] > >>> > >> -- > >> ---- > >> Andre Courchesne - Consultant > >> http://www.net-forces.com > >> Phone: (514) 667-8448 > >> MSN: [email protected] > >> Skype: VoipForces > >> > >> L'information contenue dans le pre'sent document est la > >> proprie'te' de > >> Andre Courchesne. Et est divulgue'e en toute > confidentialite'. Cette > >> information ne doit pas e^tre utilise'e, divulgue'e a` > >> d'autres personnes ou > >> reproduite sans le consentement e'crit explicite de Andre > Courchesne. > >> > >> The information contained in this document is confidential > >> and property > >> of Andre Courchesne. It shall not be used, disclosed to others or > >> reproduced without the express written consent of Andre Courchesne. > >> > >> > > --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
