similar to: SIP Gerenal settings conufsion

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

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,

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:

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

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

How can i activate "ChanSpy" to spy on a dedicated extension? I see the following in "/etc/asterisk/extensions_additional.conf" [chanspy] include => chanspy-custom exten => 501**,1,Chanspy(801) exten => 501**,n,Hangup exten => 502**,1,Chanspy(802) exten => 502**,n,Hangup But when i try to call "501**", it doesn't give any response. Thanks.

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

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

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 have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

Hi Daniel, Am 27.03.20 um 09:24 schrieb Administrator: > Hangup is h extension. your macro will never be executed. Solution: > > same = n,Dial(whatever) > same = n,[...]) > same = n,Hangup > > exten  = h,1,1,DumpChan() >  same = n,System(/home/asterisk/bash_test) I don't really understand your code… I think I don't have to edit the first part of the conf file

Hello, I have two Samba 3.0.22 PDCs and each trust each other. When I add an user of each domain to the permissions of a file on a windows machine (W2k, WXP), it shows for them DOMAIN\USERNAME. Everything is fine. But when i close the permission window and reopen it, then the user out of the trusted domain is only shown as SID. The one of the own domain is resolved fine. This happens on clients

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.

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Estimados compañeros tengo el siguiente problema: quiero poner etiqueta a los valores de una variable. He creado una lista que se llama geren que tiene 194 valores distintos. Creo la siguiente data.frame: dg<-data.frame(c_autónoma=names(tca)[sb7$ca],prov=names(cprov)[sb7$prov],geren=names(geren)[sb7$gerencia]) sb7$gerencia tiene 107 códigos distintos incluidos dentro de los códigos de

I know I should probably post this to the Trixbox forums, but am hoping someone might have a quick answer here for me. Client with Trixbox 2.6.2 with a recompiled asterisk 1.4.23 and various patches for RPID. A meetme conference with several people was in progress when the UPS died and the machine suddenly lost power. It is back up now, of course, but when you call into the conference room

I've added an introduction article about the ARA on my web site http://www.voip-forum.com/ The same text is now also added to CVS head as README.realtime. On the same site, you will also find the news item about how we used Asterisk for a call from an airline jet above Greenland to Stockholm, Sweden. The world is getting smaller and more connected every day! /Olle

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"

subject: ssh non-intuitive logging setting (priority names). I installed openssh 2.3.0p1 on Solaris 7 for x86 box and sshd worked fine. However, somehow the logging of connection and disconnection to sshd was not recorded as I wished. Time to investigate. On a host where sshd from data-fellows once ran, the log was recorded with auth.info level. After trying to modify sshd_config, I found that