similar to: Followme() and variables

Hi In asterisk channel ,I so number of variable regarding the Codec ,Can anyone explain what are those variable variable means.Below are the variables 1. chan->readformat 2. chan->writeformat 3. chan ->rawreadformat 4. chan ->rawwriteformat 5. chan->nativeformats Thanks Nikhil

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

On 26 May 2005 20:08:03 +0100 Karl Heyes <karl@xiph.org> wrote: > On Thu, 2005-05-26 at 19:35, Thomas D.Simes wrote: > > I'm running darkice-0.15 to stream live mp3 encoded audio to > > icecast2-2.2.0_1,1 on a FreeBSD 4.10 machine and everything is > > working great. Now that I have my icecast stream server set up, I > > would like to register it in the

On Tue, 2005-05-31 at 13:38 -0400, Jared Mauch wrote: > On Tue, May 31, 2005 at 12:30:07PM -0500, John Lange wrote: > > I am not a Cisco person; so the question is, is it possible to have one > > of the following: > > > > 1) Have the Cisco authenticate (register) as a SIP client to the > > Asterisk server. This allows me to place the Cisco in its own context. >

Hello there I have been struggling with queues, because i think this is the right module for our business. My main goal, is when we receive external calls, the receptionist should be able to transfer the call to us Technicians, and I am trying to add 2 extensions to a queue name [teknisk] Extension 301 and 302. I have a test setup now which I thought should look like this: When a external call

Afternoon All, Is anyone aware of a way to generate ringing as opposed to starting music on hold for the party originating a call with followme? I'm assuming its doable as it looks like FreePBX users get the option (Not to say that FreePBX haven't got their own followme implementation though). Cheers Nick.

Hellos, I have using asterisk 1.2 and freepbx 2.3. I need users to disable and enable followme from there phones. I can't see any support for it. Is this possible/available.? I have googled and I can't get information on it -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer

Hello everybody, I am trying to set up followme. As far as I understood from followme.conf multiple digits are allowed to be used for taking call. Currently call is accepted only if one digit is configured for takecall, e,g takecall=>5 Don't you know are there any real limitations for using multi-digit combinations for accepring calls? Thanks, Kseniia

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next

I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which

Hello, I am writing an AGI script to achieve the following - Users can Disable/Enable followme from their extension. They can also change the followme details from their extensions. I have looked at the follow me table for freepbx. I can't see the field for the values enabling/disable followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems

I have a question related to FollowMe on Asterisk 1.4. Is there a way to force Asterisk to always leave VM on the forwarded extension's cell phone, as opposed to pulling the call back from the forward to cell and depositing in Asterisk voicemail? Thanks in Advance! -- *Cory J Andrews* 725 Powell Lane Lewiston, NY 14092 voice - 716.579.6331 email - ipcbcory at gmail.com --------------

Trying to do a FollowMe test. When the extension is dialed, it dials my cellphone and my cell phone rings. But when I answer my cell phone it's just silence. When I press '1' on my cell phone, nothing happens. extensions.conf: exten => 140,1,FollowMe(mleonetti) followme.conf [general] featuredigittimeout=>5000 takecall=>1 declinecall=>2

For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the

Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for example in the queue with the setinterfacevar field. I just need to pass a variable from the channel placing the call to the followme to the channel

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. Does someone know what "order field" means in followme.conf? The Doku says: number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ] So an example would be: number=> 123&124&125,10,? It would be nice if someone could enlighten me. cheers t.

Is there a way to strip the normal features out of FollowMe (call acceptance, etc), and just set followme up to to blind transfer any call to an extension's associated cell number if it is not answered on the extension after 4 rings? Users want followme calls to wind up in their cellphone voicemail and I'm having some issues with ring/answer timing and Asterisk wants to pull the call

Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES