similar to: Can't access outside voicemail services through asterisk

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

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'm wondering if I can do the following: Caller activates the Directory application Caller enters the first 3 digits of a person's last name ===== Normally here, Asterisk will say the extension number of a person found. Is there a way to get Asterisk to say the name as well? (perhaps using the same sound file that is used for their name in the voicemail application) Can this be

Besides email notification, is there another way to get asterisk notify the user that they have a message? Example: Some analog phones have a blinking light that lets the user know that they have a voicemail message. Is asterisk capable of doing this?

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) --

Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before?

Ok. I just got a TDM10B and it is in with my X100P. So as it says in the provided instructions, I used the command modprobe tor2 I get an error message saying that there is no such device. My zaptel.conf looks like this: fxsks=1 fxoks=2 So I load the X100P first. (modprobe wcfxo) Then I load the TDM10B (modprobe tor2) Then I'm told that the device doesn't exist. Please help

My setup: X100P and Quicknet PhoneJack. I can't seem to properly set up a Zap channel for my X100P. Here are some of my configurations: [zaptel.conf] fxsks=1 #X100P fxoks=2 #Quicknet PhoneJack defaultzone=us loadzone=us [zapata.conf] [channels] context=local signalling=fxs_ks channel->1 ;X100P [extensions.conf] ... [local] exten=>_NXXNXXXXXX,1,Dial,Zap/1 ;I'm pretty sure the

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

I just wanted to try out Festival, but I can't get it patched. I'm thinking that there is something missing from the steps listed at http://www.marko.net/asterisk/archives/0209/0389.html. >>tar xvzf festival-1.4.2-release.tar.gz >>patch -p0 </usr/src/asterisk-ng/festival-1.4.2-diff >> (or wherever the patch is located) When I run the patch command, I get the

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

I am trying to run the make command to install Asterisk, but I get the following error: make ... ... checking for tgetent in -ltermcap ... no checking for tgetent in -ltinfo ... no checking for tgetent in -lcurses ... no checking for tgetent in -lncurses ... no configure: error: termcap support not found I am running Mandrake 9.1 on a Pentium II 200MHz. Could this be a hardware issue? I

I have a Pitney Bowes (USPS Postage) machine that connects via a USB modem to fill it. It connects but soon disconnects. It works fine through a standard analog phone line not connected to asterisk. I also have the 'd' option on the Dial command. exten => _1NXXNXXXXXX,1,Dial,Zap/47/BYEXTENSION||d Any ideas? John

Is there a way to record your own voice messages ("Welcome to my PBX, Press 1 for ...") using asterisk and an analog phone, or do they need to be recorded using traditional voice recording software? Also, I am confused as to why my replies to the message board are never indented. Call this the ultimate newbie question, but how should the reply be worded so that I don't screw up

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 =>

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

This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for

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 =>

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

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial