similar to: No ringing on outbound PRI calls

Hi Adam, I appreciate your feedback, and understand totally where you're coming from as far as the database perspective goes. For the first "draft" of the app, I think I'm going to let it default to using a conf file for two reasons. First, my setup currently does not utilize a database. I would like to move to that type of a setup in the future however. Secondly, seeing as

I ordered 10 7960's with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, that's not really applicable in a SIP setup? I'm guessing it's going to be a more XML

Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office

Hello, In setting up * for my company's office and remote employees, I have a question about how to log one username into * as either a SIP account, or IAX account. For example, we will be using SIP phones in the office locally to the * server, however some employees travel, and want to use IAX (as it's much friendlier with firewall/proxy setups than SIP) clients on their notebooks.

I have been attempting to download, compile and configure spandsp to function with * without much luck. I am guessing that some assumptions were made regarding the users knowledge level of Linux. Sadly, I must not live up to those assumptions. My problem begins when after compiling spandsp I look for the app_rxfax.c, app_txfax.c, app_dtmftotext.c and makefile.patch files to place in the

Set(CALLERID(number)=XXXXXXXXXX) works here. Also check with your VoIP provider what format they want for the number. (I believe) most accept a 10-digit number, but I seem to remember reading about the odd provider that wanted a leading "1". On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Thu, 18 Jun 2015 13:45:10 EDT > kenner at

Hello all, I'm going to tackle learning C this week, and start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

I have an issue with ringing. Some users who call my switch hear ringing and others don't. I have researched this and understand the issue of firewalling and RTP. My switch has UDP ports 10000 to 20000 open. In any case I think that blocked RTP would block all ringing, not just some. I have one origination provider. As far as I can tell the issue is related to the remote user's

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

Hey everyone ! I have a problem in making outbound calls in PRI connection. I have E1 PRI airtel connection [ India ] [ asterisk-1.2.12.1 on CentOS 4.4 ] zaptel.conf ---------- [channels] language=en usecallerid = yes hidecallerid = no callwaiting=yes threewaycalling = yes usecallingpres=yes transfer = yes echocancel = yes echotraining = yes immediate = no ;group=0 ;context = from-pstn

On Thu, 18 Jun 2015 11:27:08 -0600 Greg Woods <greg at gregandeva.net> wrote: > [macro-callout] > exten => s,1,Set(CALLERID(num)=##########) > exten => s,2,Set(CALLERID(name)=##########) > exten => s,3,Set(CALLERID(all)="##########" <##########>) > exten => s,4,Dial(${PROVIDER}/"1${ARG1}",30,tr) > exten => s,n,Hangup() CALLERID is a

On Thu, 18 Jun 2015 13:45:10 EDT kenner at gnat.com (Richard Kenner) wrote: > > CALLERID is a read only variable. > > That's not correct. I set it all over the place in my dialplan. Then someone needs to fix this page. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/

Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

D'Arcy J.M. Cain If the device is registering and then dropping there are several usual items. The router may be closing the ports on the device. The router may have a AGL SIP helper that is causing issues. Make sure that the device is sending out keep alive packets. Shut down any AGL helpers on the router. Make sure that the site is not double NATing Try using a stun

Hello all, One of my remote employees is using a 7960 we sent him, on a public IP address at his home office. I've run pings and traceroutes both from the server to his phone, and from the cable modem to our server, there's never a high ping time, or a dropped packet, however about every 30 minutes to an hour into his calls (not all of them, it's random) he can no longer hear the

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

If I remove the password, how can anyone access the mailbox if the 'mailbox' prompt is not played? Nabeel On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" <darcy at vex.net> wrote: > On Sat, 30 Jul 2016 06:43:47 +0100 > Nabeel <nabeelshikder at gmail.com> wrote: > > I am using Asterisk voicemail on a CentOS 7 server. I would like to > > be able to