Re: [Asterisk-Users] Automatic 3 Way Call
Hello, On 4/11/06, Shad Mortazavi [EMAIL PROTECTED] wrote: I'm working on a call recording solution and would like to have the ability to initiate a 3 way call based on an incoming call. One party will be an AGI that I have other will be an outbound call via a second T1 interface. Does anyone have a working configuration for an Asterisk initiated 3 way call? See http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro Hope this helps Regards, Alexander Chemeris ___ --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
Re: [Asterisk-Users] MeetMe/Asterisk Timer
Hello, On 4/4/06, Kelvin Williams [EMAIL PROTECTED] wrote: In short, does anyone have any other advice to get the MeetMe application working on a potentially virtualized server (although the box said dedicated), without kernel sources, and a box that has no apparent USB? This may sound stupid, but did you tried app_conference? It does not require any hardware timers. Regards, Alexander Chemeris. ___ --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
Re: [Asterisk-Users] Big Traffic anyway?
Hello, On 3/22/06, Dovid Bender [EMAIL PROTECTED] wrote: I have a rule in my outlook to delete any email that in the subject it says asterisk users mailing list traffic. Get my drift ? This topic has been around for a long time as others have mentioned and people keep replying. This useless topic alone adds several messages to my inbox daily. Heh, I changed subject, and this you'll receive this mail. I don't want to create thousands filters in my mail client - one for each topic I dont't want to read. Subject filters is very bad idea too. Isn't it simpler not to receive mails instead of deleting? Regards, Alexander Chemeris. ___ --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
Re: [Asterisk-Users] meetme application
Miguel, On 2/11/06, Miguel [EMAIL PROTECTED] wrote: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension i did a normal make, make install, did i miss something? You need zaptel headers installed to build MeetMe application. And you need zaptel devices (or ztdummy) to run MeetMe. See voip-info.org for more information. Regards, Alexander Chemeris ___ --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
Re: [Asterisk-Users] Re: Configuring Meeting Room from Asterisk Manager API
Somesh, On 2/3/06, Somesh S Shanbhag [EMAIL PROTECTED] wrote: I want to do a three-party conferencing using manager api. But I found out from the asterisk-users list that I *MUST* use the meeting room concept. I wanted to know wheather meeting room can be configured dynamically? on the fly? Otherwise, configuring meeting room statically is not scalable. First search for 'dynamic conferences' on voip-info.org. There you'll find macro to create dynamic conferences on the fly. Main idea is to enable dynamic creation of meetme rooms and create them according to user phone number. See also Originate command in manager actions reference. You may use command similar to this: Action: Originate Channel: SIP/4 Application: MeetMe Data: 41|adEpq ActionID: MeetMe-id CallerID: MeetMe-caller-id Use 'Channel' to specify user you want to add, and you may use 'CallerID' to track following events. ___ --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
Re: Re[2]: [Asterisk-Users] Re: MeetMe Dialplan question
On 1/23/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote: On Saturday, January 21, 2006 8:02 PM Alexander Chemeris wrote: What is the problem with step 3? See this example as basis for modifications: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro Unless I have terribly misunderstood that macro, that is basically the same thing I am doing now, is it not? Simply transfer the customer to a conference room (I might have a look into the automatically determined conf room number), then transfer all collegues in there as well and finally jump in myself. It is however not quite what I described in step 3. Yes, that's so. I tested this macro with SIP-softphones and it works. May be this is the simplest way to do what you want. And this is a good start point for modifications. -- Good luck, Alexander Chemeris ___ --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
Re[2]: [Asterisk-Users] Re: MeetMe Dialplan question
Hello,
Saturday, January 21, 2006, 7:34:50 PM, you wrote:
3. I transfer the call to my personal MeetMe room. In this step I
would like not only the customer but also me to be connected to the
MeetMe room automatically. Basically I can continue to chat with the
customer without him noticing anything.
4. I now put the call on hold and call the collegue. If he wants to join
I simply transfer him to the room as well and can continue to do so with
other collegues. In order to return to the conference myself I now do
not need to call the conference number myself but simply return to the
call created in step 3.
With the exception of step 3 everything seems easy. How can I solve this
with the G-option?
What is the problem with step 3?
See this example as basis for modifications:
http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro
Use blind transfer and macro as above and you'll get this.
With step 4 we have tricky thing:
exten = _#901!,1,Dial(${EXTEN:4}, 10, G(meet^s-ENTER^1)
[meet]
exten = s-ENTER,1,Goto(s-HOLDER,1) -- Here we get one channel
exten = s-ENTER,2,Goto(s-COLLEGUE) -- And here we get other channel
exten = s-HOLDER,1,Hangup() -- Leave this call, your collegue is here
exten = s-COLLEGUE,1,MeetMe(${CONFHOLDER},dwx) -- Retrieve CONFHOLDER from
somewhere
exten = s-COLLEGUE,2,Hangup()
There may be minor mistakes, but I tried such dialplan and it works.
Hope this helps you.
--
Best regards,
Alexander Chemeris
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[Asterisk-Users] Dynamic conference - add participants
Hello all, I need to create dynamic conferences with variable number of patricipants. My users use custom SIP softphone and I want to implement fast conference creation/moderation. I search through voip-info.org examples and found some useful information in this page: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro But this solution uses ugly way to add new participants. Any ideas how can I add several participants at a heat? I think this may be done though Asterisk manager interface, but I think there must be a way to achieve utilizing Dialplan features. ___ --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
