similar to: Custom Development

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

Hi, all I have couple of SIP phones and they are in [from-sip] context. I also have an IAX2 phone. I have put this one in [iax-user] context. I want to make calls between SIP and IAX2 phones. If I put them all in same context all is fine, however when they are in different contexts they will not call each other and I will get message (in * CLI) that particular extension does not exist in a

Dear all, I've been taking my baby-steps toward setting up an Asterisk phone system in my office, as also between my home and office (connected by DSL). I'm have a rough time getting two * boxes talk IAX over a LAN. I don't know what I am doing wrong, but am attaching my iax.conf and extensions.conf on both the boxes. Does anyone see it? ------config files start------ site-0

Hey, how can I implement a dial plan like the following: incoming call: 1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no answer after 15 sec also ring phones 4 and 5 2. ring phone 1 monday to friday between 0:00-9:00 and 20:00-24:00; if no answer after 20 sec also ring phones 2 and 3 3. ring phone 1 saturday and sunday all day I do not need a in detail answer for each of the

>From this list I found that I could use SetGroup and CheckGroup to do what I wanted. But I'm not quite sure how I do it. The case is that I have 3 user groups, and one main group. The main group is for all the incoming calls from external phones. The main group should be allowed to have 3 calls at the time. The 3 user groups are internal groups that I want to limit by ony having one

Hello. I have another issue. When I call in, everything is processed correctly, including voicemail, but I don't hear any ringing/ringback. exten => s,1,Zapateller(answer|nocallerid) exten => s,2,NoOp exten => s,3,Playback(pls-wait-connect-call) exten => s,4,Dial(${PHONE1}&${PHONE2}&${PHONE3}&${PHONE4},15,Ttm) exten => s,5,Answer exten => s,6,Wait(1) exten

I'm trying to get two linux machines with kphone-3.11 two communicate with each other over asterisk. I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing

I'm having trouble getting festival to work with asterisk. We are running debian (sarge) and got asterisk from CVS. Here's what I'm using as far as festival goes. Debian (Sarge) gcc version 3.3.4 (Debian 1:3.3.4-3) Connected to Asterisk CVS-HEAD-07/28/04-21:08:19 festival-1.4.3-release.tar.gz speech-tools_1.2.3.orig.tar.gz I got patches for both of these. Speech tools

My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? [root at robin asterisk]# cat chan_dahdi.conf [trunkgroups] [channels] [phone](!) usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres = yes

I'm new in hire so Hello to everyone! I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration (used make samples). I would like to make phone connections between X-Lite (SIP) installed on computers in LAN. How to make this? I was reading manual, and tried to make changes in sip.conf but this all

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

Hello All, This could be more a routing/linux problem, but I'm wondering if can be handled somehow within *. I have a client that has a contract with a VoIP provider. This contract limits number of phones registered using same IP to 3. There are 3 ways to handle this: 1. Few ATA connected to 3 phones + network 2. Linux router that lies between * box and network. Virual interfaces, iptables

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

Hi, I have an FXO card connected to my phone line which works in Asterisk as Zap/1. Is there any way of detecting whether something else is on the line before picking up on this channel? For example, I dont want to pick up and dial out on the line if someone is on it using another phone (which is connected directly to the line, rather than through Asterisk). Also, when an incoming call comes

Hello! I've upgraded from 1.2.9 to 1.2.14 recently but experience an unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was playing continuous on sequential extensions after a timeout, it is restarted for every extension in 1.2.14: ;music starts exten => 902,1,Dial(SIP/phone1@proxy.com|5|m(mymusic)) ;music starts again exten =>

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

I set up a call forwarding script in extensions.conf which uses the AstDB but I'm wondering if I reboot the server, will the entry in AstDB still reside? What the script does is when a call comes in, it check to see if there is a null value or a call forward number. If null, it will call the local office connections. If there is a number, it calls that. Now I just need to know if I reboot

Hello everybody and merry xmas. I have a problem with sip phones calling each other inside the same network (no nat, no firewall). You can make and receive calls and pick them up, but you cannot hear anything on any side of the call. But if you press hold/unhold or you transfer the call, then everything works as expected. Ths SIP phones I've tried are Swissvoice IP10s and kphone, it

I'm having this problem where if the phone is ringing from IncomingCall #1, IC#2 will be immediately sent to VM. Is there somethign wrong with my dial plan? I currently have 4 incoming lines going into a TDM400 with the group set to g0. Could it be that the way I've set this up, if any of the phones are busy, it goes immediately to VM? exten => s,1,Answer() exten => s,2,Wait(1)

Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions... This will hopefully be my