similar to: Context fails so falling back to extension " s" ?

I realize it's bad form, but I'd really appreciate some hand holding here. AMP is making me pull my hair out and the mountain of configuration data in extensions.conf is starting to get to me... I have the first channel configured in zapata.conf to "take" incoming contexts to "from-pstn". I'm using AMP neat "Incoming Calls" configuration page and

Folks, First off, this is messy, and I hope someone will be kind enough to help me clean this up (the part added to extensions_additional.conf). You've been warned! For those of your using AMP or A@H, there has been a lot of talk about how to route incoming calls to different places based on which trunk is ringing. The standard answer is that you can only do this by using DIDs,

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

Figuring I knew what I was doing (I didn't - surprise) I added a totally unnecessary line in /etc/asterisk/extensions_additional.conf a couple of days ago. Troubleshooting a dialing rule issue, I'm now realizing that FreePBX is updating its database with the new settings but is not rewriting/updating extensions_additional.conf with the changes I'm making. I've tried renaming the

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

do you have selinux enabled? It should not be. p p.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything. From: "Lachek Butalek" <lachek@gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Date:

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

It has been narrowed down to the CONTEXT the call was originally located in. I need to figure a way within the parking application to set the extension I want the call to ring back to. Specifically I want to set it to always ring back to the phone that parked it. Example, below .. When the call rings only a specific phone http://pastebin.ca/2452 the context it originates from aa_1,2204,1

Hi, I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message. My Zaptel.conf is as

Greetings *`s, I am having what appears to be a small problem, but the frustration is erally getting to me, what am I doing wrong here ? I used AMP to set up a custom menu, so if caller presses 1 it goes to ext200, if caller presses 2 it goes to ext201 etc etc... Now I have created a third option that when the caller presses 3 it must play a sound and hang up. No rocket science yet. When

Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the

Hi all, I'm trying to setup a custom extension in AMP (yes i can code it by hand but the on-site admin that does moves & changes cannot). I've tried the following > add cutom extension 600 in the dial box i have Dial(IAX2/username:password@host/$EXTEN@from-internal) this doesnt work as these lines are added to extensions_additional.conf exten =>

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

Hello every one. This is a question done by me, not yet answered. Please, help. I: 1. Run install-pdf from linux to support faxes on my asterisk. 2. Made the configurations throuhg AMP in a. Setup->Inbound Routing->(the only route I have)->fax extension->System b. Setup->Inbound Routing->(the only route I have)->fax email->(my email) c.

Thanks Paradise, this seems to have worked a treat!!! I commented out the: exten => 110,hint,SIP/110 lines which were in extensions_additional.conf for each sip extension I had. This seems to have stopped the crashes which were previously 3-5 times a day, now: System uptime: 1 day, 18 hours, 10 minutes, 3 seconds Interestingly it had the knock on effect of fixing another problem I had

I've finally got @Home .7 up and working. I have a Digium TDM400P with 2 fxo and 2 fxs. All I've done so far after loading is to run genzaptelconf -s -d ; verified that ztcfg -v shows all 4 ports and logged into the AMP app and added 1 extension (200) to one of my fxs ports. When I pick up either of the phones plugged into the fxs ports and press the number 2, it immediately rings

Hello all, A new version of the Asterisk Management Portal is available for download. Please visit the AMP homepage at http://amp.coalescentsystems.ca Upgrade instructions are at http://amp.coalescentsystems.ca/UPGRADE Use our Sourceforge mailing list and forum for discussions about AMP. 1.10.006 ChangeLog: - Use extensions_custom.conf for customizations. Sample included. - Added option

Ok I have read every line in the wiki for call parking. HELP PLEASE I need to perform a simple call parking that upon timeout will ALWAYS rings back to the extension that actually parked the call. Am I just crazy or is this not possible. >It has been narrowed down to the CONTEXT the call was originally located in. >I need to figure a way within the parking application to set the

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP/303 extensions_additional.conf:exten => 304,hint,SIP/304

I know almost nothing linux, and don't really know that much about Asterisk (proper).. but I was 'pulled' by this subject and grabbed an <mailto:Asterisk@home> Asterisk@home installation CD (version 1.3) and just went with it. Newbie doesn't quite describe it, I'm a banker.. this simply goes to show how easy Asterisk is becoming (I certainly hope this direction was meant