similar to: RE: RE: dialing out

Thanks to Greg Hill for pointing me to the 'sip debug on' cmd that helped me resolve the sip connection problem! The other issue I'm trying to resolve is configuring outgoing calls. I need to configure outgoing calls to use the FXO card in the PBX (zaptel device) via sip connected ip phones when a user dials 9. I need to support local and long distance dialing. Below is an excerpt of

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

I currently have two fax machines on my system. Both of them seem to send and receive very slowly. My end users are complaining; saying it was faster before we moved to * (Straight Analog Lines) Any help would be great. PS: I already have the d option on the Dial line. Both fax machines are in their own context: [faxes] exten => _9NXXXXXX,1,StripMSD,1 exten =>

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

hi when two calls are active on controller 2, chan_capi won't use controller 1. this is with AVM C2 roy -- Executing Goto("SIP/torgeir-b476", "capiring|BYEXTENSION|1") in new stack -- Goto (capiring,90044875,1) -- Executing Dial("SIP/torgeir-b476", "CAPI/22545066:bBYEXTENSION|120|Ttr") in new stack == data = 22545066:b90044875 ==

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

If this handles the case where 10 digits are required: exten => _9NXXXXXXXXX,1,StripMSD,1 exten => _NXXXXXXXXX,2,Dial,Zap/4/BYEXTENSION How do you create a handler which works for either this or the case with a leading '1' plus 10 digits? tnx -kim -- w8hdkim@gmail.com

Hi, Having Asterisk-0.4.0 with 2FXO port and Samsung SKP 816H PBX in 2 offices. I am able to make call between two offices. But the problem is that call dosen't hangup. Office A [Asterisk+2FXO+SamsungPBX] <------------- I A X ------------> Office B [Asterisk+2FXO+SamsungPBX] Configuration files are given here.............. ------------------------ zapata.conf

Hi, I can't make outgoing calls with CAPI (passive ISDN Fritz card). See Asterisk error below. Incoming calls and SIP to SIP calls do work. It looks like a msn mismatch in extensions.conf and capi.conf, but I can't find it. Can anyone help me find the problem? Thanks, Rob *CLI> -- Executing Dial("SIP/8112-1be9", "CAPI/356666666:BYEXTENSION") in new stack

Hello all, taking examples from various pointers, I am attempting to put together an outbound dialing example using SIP (Cisco 7960) with 2 X100P. Everything seems to be working without generating errors, but the problem is the phone hangs up (102/Bye). Any pointers/advice are much appreciated Here is the section in extensions.conf: extensions.conf ; From CISCO at work ; exten =>

I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack NOTICE[245776]: File app_dial.c, Line 698

Hello List! I finally got asterisk with capi working, and its already answering my call as well! :) Now i would like to call a number from my shoft phone (kphone). This is my extentions.conf: --- [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password

hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1!

I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve any in-dialing nor I can dial out; this is what I have from "pri intense debug span 1" command ---------- *CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 -- Executing Playback("SIP/201-a862", "tt-weasels") in new stack -- Playing 'tt-weasels' (language

Hello, I try to place a call between CISCO IP phone 7905 (h323) and an analogue telephone on DLINK DG-104S (mgcp). Call setup is handled ok - both way, but after connection of the call I get both site no-audio, even codec are Set to alow-64 both. The call is terminated with this debug output - Didn't get a frame from channel: MGCP/aaln/1@dlink-1 and is Hanged up. Jan 18 20:32:55

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1

Hi, I am trying to make an outbound call from phone attached to FXS port. My telephone (VoIP) line is connected to FXO port (Zap/4) Default context for channel # 4 is 'directdial' here is part of my extension.conf [directdial] ignorepat => 9 exten => 9,1,Dial,Zap/4/ exten => 9,2,Congestion include => international [international] ignorepat => 9 exten =>