similar to: How to grab CallerId information

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

Andi - I have experienced the same issue you mention and gotten no reply as to a way to fix it. I finally implemented "blacklist" into my Asterisk and added "Anonymous", "anonymous", "unknown", "Unknown", etc., into my blacklist file. When those come in with an IP address instead of a phone number but have no real name, they get the

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

> -----Original Message----- > From: Michiel van Baak [mailto:michiel@vanbaak.info] > Sent: 26 May 2005 20:22 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) > > > Anton, > > My script is not connecting to the manager interface. > The php script is run as agi script as first when a call > comes in. The

I am running asterisk and I want to achieve the following scenario My goal in the end is to achieve the scenario (example using extension A and Extension B) 1. Extension A has a line apperance of 4(4 calls can ring on it). 2. Extension B calls extension A(which is busy on one of the lines). 3. Extension A sees the second light blinking and hears the beeps (currently working). 4. Extension B is

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in

> > > >| -----Original Message----- > >| From: asterisk-users-bounces@lists.digium.com > >| [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > >| Michiel van Baak > >| Sent: Martes, 24 de Mayo de 2005 04:45 p.m. > >| To: asterisk-users@lists.digium.com > >| Subject: Re: [Asterisk-Users] CallerID > >| > >| On 23:44, Tue 24

Hello, We use astGUIclient, it does have server side apps that have to be installed on your Asterisk server, but it does have callerID popups that allow you to search a customizable web page when a call comes in. We are also releasing a new version of the astGUIclient app next week that is entirely web-based and easier to configure the client side. http://astguiclient.sf.net/ MATT---

FYI - We have a solution here provided by Lucent that allows us to play / review voicemails left on the Octel attached to the 5ESS switch... While this is a simple webpage - doing a refresh every 3 to 4 seconds, it does actually work. The only loss of course would be if someone hung up on first ring - the browser might not catch it in time. -----Original Message----- From:

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

hello everyone, I just had a thought on this subject why not create a daemon process on the Client PC That registers its self and What phone the user is connected. An AGI script could monitor the progress and when answered could send a push to the registered daemon which would push a link to the registered daemon on the telephone operators on the desk top. this would not waist resource as much as

I am also very interested in CRM integration. Anything I can do to help? One thing I don't understand is how is the browser being launched on the person's PC. Or is it not launched automatically? Anyone know of a simple app running on the desktop to do this? I looked into IPSwithcBoard and it appears like it should be able to do the job, but: a) It is pretty heavy - it does a lot of

Hello, In my 1.6.1.18-powered system, I've got the following dialplan (in extensions.ael) : Dial(SIP/foo,15); if (${DIALSTATUS}="NOANSWER") Dial(SIP/bar,15); When SIP/baz dials peer SIP/foo which do not answer, I've got a single CDR entry like this: SIP/baz SIP/bar time_when_foo_started_to_ring time_when_bar_ended_talking ANSWERED How can I get two CDR entries :

I set up a 4 digit extension for parking calls. 5400 All works fine from Asterisk. If I park a call from a legacy PBX, it parks OK and can be picked up, but if unanswered, the call parkingtime timer tries to send the call back to the PRI Zaptel trunk that it was parked with. That circuit is no longer in use. I can add a t extension for context 'park-dial' (which I assume is hardcoded

Guys. This is a good one... Is anybody doing callerid on the PC? What are you using besides yac or things like that? And are you using some CRM like Goldmine with it? Good huh?