similar to: Special Dialplan

I'm trying to to to "dial1" if caller id match: but dial plan execute 220,n(dial1) regardless exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1) exten => 220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w) exten => 220,n, I was under impression that if condition is met it will jump to

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

I'm learning asterisk, and decided to make myself an answering machine out of it. Seems pretty straightforward to use an agi (perl) to do what I want. What I want is: Answer the phone. check for time of the day If TOD is during the time I sleep I announce i'm sleeping & prompt caller to dial1 (or whatever) to connect to my extension & then go to voicemail if busy/una,

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia

Hi, I am trying to understand why some of my call transfers fail. My scenario is as follows: Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2 Step1: PBX1 extension 101 calls PBX2 extension 102 Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension 103 Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension 104 Step3 fails and extension 103

Rafa³ Dutko has just discovered a potentially serious bug in version 1.3.0 and 1.3.1. In both versions, where an interface option appears on multiple interfaces, the option may only be applied to the first interface on which it appears. A corrected firewall script for 1.3.1 is available at: http://www.shorewall.net/pub/shorewall/errata/1.3.1/firewall and

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway

Deal All Asterisk Expert If this possible to Create Voice Recording System Beside Main Asterisk PBX?, so Call be handle by 1 Server and Recording by other server. 1. How to accomplish. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090325/d57465cd/attachment.htm

Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0<:1010 at GW1>). I want to forward at 1020 too. I tested (S0<:1010|1020 at GW1>) and doesn't work. Did you have any other ideea? Thank you.

Just built myself a little test server with an Atom 230 processor in it and am quite impressed with it so-far. Wondering is anyones used one in anger for a VoIP platform? I'm after something with a bit more oomph than the VIAs I'm currently using that I can use in a small box (mini ATX size) without going full-blown Xeons, etc. Cheers, Gordon

Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk supports SMS over GSM modem. I know chan_mobile had SMS in the future at one point but have not revisited the project since. "America Movil's MVNO TracFone Wireless quietly unveiled a prepaid, nationwide unlimited offering for $45 per month that includes unlimited text messaging and 30 MB of data."

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

Dear All I installed a Digium card TE405P with zaptel and its running successfully with no alarms, but asterisk is not running . Any one have a cure or advice 03:09.0 Communication controller: Digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02) Nov 14 07:56:58 localhost kernel: wct4xxp: Clearing yellow alarm on span 2 Nov 14 07:56:58 localhost kernel: wct4xxp: Clearing

Hi, Quick question to the real world. Approx what specs would I need on server to handle 95 ZAP or Dahdi -> SIP gateway using G729 on the SIP to carrier side (nothing else, just media conversion)? Does the latest Asterisk/DAHDI significantly improve these numbers over say, Asterisk 1.2.X? Sure, there is plenty to read but nothing I could find quickly on my exact needs that was clear and I

Hello, What is the maximum number of simultaneous calls supported by asterisk. thks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090703/0794c554/attachment.htm