similar to: dialing 2 channels at the same timewithdifferentcaller ID number?

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 1:48 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the > sametimewithdifferentcaller ID number? > >

*** If anyone has a better way of doing this, please post to the list. I hadn't seen anything on this list or in channel.c/chan_local.c - which prompted this email *** I'm not sure how many VoIP providers out there are using Asterisk as a service platform like we do, but I thought I'd share an experience with call looping that was racking my brain with the list. One of the

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Damon Estep > Sent: Tuesday, January 31, 2006 8:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels > atthesametimewithdifferentcaller ID number? > > > >

> > Yes you must prefix a variabel with __ that's (2) _ underscores so that > it cross channels. > Aah, the magic formula - documented where? :) Thanks a million, have a great day. Damon

Don't feal bad about not reading. I yell at my 10 y.o. about it all the time. READ, NO more TV, READ!!! > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 10:23 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

Hi, Yet another question, now that I have callerid working correctly, I'm trying to work out how to utilise the different numbers I have. I have a 100 number range allocated to my E1/PRI/OnRamp service. My incoming calls are handled like this: Advertised/published number is an analogue line terminating on a X101P. If the analog line is busy, it has a call diversion to the PRI on a TE405P

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

Thanks to Greg Hill for pointing me to the 'sip debug on' cmd that helped me resolve the sip connection problem! The other issue I'm trying to resolve is configuring outgoing calls. I need to configure outgoing calls to use the FXO card in the PBX (zaptel device) via sip connected ip phones when a user dials 9. I need to support local and long distance dialing. Below is an excerpt of

Nevermind. Figured this out. I needed the following in extensions.conf to enable outbound dial. exten => _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt) Thanks -----Original Message----- From: Info [mailto:info@psgsite.com] Sent: Tuesday, August 17, 2004 9:27 AM To: 'asterisk-users@lists.digium.com' Subject: RE: dialing out Thanks to Greg Hill for pointing me to the 'sip debug on'

I'm looking for a way to detect when a VOIP provider is unable to complete a call and thus try another VOIP provider (failover/backup type situation). using qualify is NOT sufficient, since the provider could very well be reachable but not be able to complete the call for other reasons. A perfect example: setting my caller ID number to my real number and calling a local number causes the

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

I'm trying to copy the functionality of something like this in extensions.conf [extension matching on callerid]: exten => _NXXNXXXXXX/7065557230,1,NoCDR exten => _NXXNXXXXXX/7065557230,2,Dial(Zap/g1/${EXTEN}) exten => _NXXNXXXXXX/7065557230,3,Busy in realtime: | 16 | fb | _NXXNXXXXXX/7065557230 | 1 | NoCDR | | | 17 | fb | _NXXNXXXXXX/7065557230 | 2 |

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

Hello All, How can I SET EXTENSION from context? This is my context: - [docall-usa] exten => _NXXNXXXXXX,1,Answer exten => _NXXNXXXXXX,n,Set() ; <<What do I need to set here>> exten => _NXXNXXXXXX,n,DeadAGI(dousacall.php|1) exten => _NXXNXXXXXX,n,Hangup I need to add 1 in front of ${EXTEN} and then send the call to dousa.php. Set(CALLERID(number)=1${EXTEN}) will set