similar to: strange asterisk issue

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

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

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

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

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

Hey guys, I'm looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, I'm not sure the extent of the coding as far as technical terms go at all. Currently we have a "call center" with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use

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

>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 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

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

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

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've put a t100p in our * server and I'm having trouble configuring it. It is directly connected to an Adtran TA 750 channel bank with two FXO cards (8 analog incoming lines total). I'm able to insmod and modprobe both zaptel and wct1xxp with no trouble, but when I start * with /usb/sbin/asterisk -c I get the following output: [root@rosella root]# /usr/sbin/asterisk -c Asterisk

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Here is an example of a couple of macros that help me where I have a SOHO with a home phone line and a work phone line. If I pick up line 2 my work line I would prefer the call I make to go out my office phone line same with if I pick up line 1 my home phone line I would prefer it go out my home line but want it to roll if needed. So with this little macro it is possible for that to happen.

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

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

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)

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've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,