similar to: Bug in Ex-Girlfriend logic?

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Hey all, Someone's posted one of my 800#'s on a poster in California for free concert tickets, so I'm getting calls from California AC's at all times of the day asking for tickets. I'm just using the 800# for friends and family, and don't know anyone in these area codes, so I'd like to just give these callers either congestion or a prerecorded message. Works fine

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

Hi, Some background: I would like to call my girlfriend over the internet. We are both behind a nat router and I want to avoid portmapping. I've heard that you can call someone behind a firewall (nat router) with the IAX protocol, but I'm not sure. The questions: Do I have to set up my own PBX asterisk server or are there any other (free) servers where I can register on and connect

Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router: http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie.

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is

Hello, I get some problem installing asterisk + ekiga on my debian etch: ii asterisk 1.2.13~dfsg-2 Open Source Private Branch Exchange (PBX) ii ekiga 2.0.3-4 H.323 and SIP compatible VOIP client $: asterisk -U asterisk -vgc give me some WARNING like : ,---- | WARNING[21806]: res_musiconhold.c:852 moh_register: Unable to open | pseudo channel for timing... Sound

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Hello, I don't really understand how channel Local works. I need that asterisk initiate a call and get some data (DTMF). So to do that I've created this dialplan : ; extensions.conf - the Asterisk dial plan ; [general] static=yes writeprotect=no clearglobalvars=no [dtmf] exten => 1,1,Verbose(Get User ID) exten => 1,n,Dial(dahdi/1/99999999,120,G(read^1^1)) exten

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

Hi All, I am running asterisk on Linux machine and trying to use confbridge application. Please have a look at Conf files. sip.conf ====== [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow = all allow=ulaw allow=alaw defaultexpiry=100 [5001] type=friend nat=yes host=dynamic canreinvite=no context= conferences disallow = all

hello list, I am trying to arm an ivr for schedule of office and outside of office [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include =>outsideofschedule|*|sat-sun|*|* [scheduleofservice] exten

the schedule of my server this configured with -6:00, and this correct one with the normal hour of my country, I made the change but I don't work me [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can