similar to: capi dial in/out configuration

Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.

Hello, I have *@Home 1.5 installed and all is working fine for incoming calls and sometimes outgoing calls. Installed in the box is a digium TDM04B (4xFXO Ports) setup as ZAP1 to ZAP4. I have incoming calls coming in on lines 1-4 in that order and outgoing calls prefering ZAP4 then ZAP3 then ZAP2. When i try to dial out to the PSTN from a SIP phone it sometimes works (normally after a reboot)

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

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

I have a problem calling a certain number from our PRI line. Calling the number from a separate PSTN phone works fine. The remote number seems to have some funny call redivert setup, when you call it, it answers immediately, makes some kind of beep and then starts to ring. Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing calls work without a problem. The server is

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

I am posting this in case someone need help.. ========================================================= YOU THA MAN!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! No sure how I will repay you, but anything you need, just let me know!!!!!!!!!!!!!!!!!!!!!!!!!!!!! Thank you, thank you, thank you!!!!!!!! -- Executing GotoIf("SIP/2007-12c7", "0?4") in new stack

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

Hi, I'm a newbie on * trying to setup an HFC card. I'm locked for many days getting the all-circuits-busy. And no idea what else to look for/how to diagnose. I'm in Spain, I've tried changing many parameters on zapata/zaptelcong with no luck, also NT & TE modes (honsetly, I've no idea what is). Any clue will be very much appreciated! I've installed *@home on my RH9,

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

Hello everybody, and first of all sorry for my poor English. I'm having trouble with Quadbri installed on Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e. Everything is working fine, except calling to switched off or "out of coverage" cell phones. In this case I have to wait 40 seconds until Asterisk tell me that "all circuits are busy now" instead of receive cell phone

Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an asterisk@home setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP

I just had a user report that they called out to someone on a cell phone this morning, and after a short conversation, the call was dropped/lost. The person on the cell phone says that this is very rare, and would not suspect the dropped/lost call to be on their side. I have looked at the asterisk/full log as thoroughly as I can, and have pasted the lines which seem relevant to that call below.