similar to: Guidance need for trunking using SIP

Hi All I am using RHEL , kernel 2.6.9-5, asterisk 1.2.4 , zaptel 1.2.3 installed , when I give modprobe for zaptel and ztdummy , I do not any error message my iax.conf contains the entry for trunking as [hoportal] type=friend host=192.168.20.32 secret=mysecret context=local trunk=yes my extensions.conf contains the entry for trunking as exten =>

Hi All I am working on voicemail , mailbox , after reading documents, I had setup of three users for mailbox to make things simpler , I had kept the user name and passord same for all the sip users, Now I am able to record the message and I do get voicemail in my email , But as defined in extensions.conf The Asterisk console messages, part of the sip.conf ,

Hi I had installed and configured 2 IAX server , users from 1'st server can dial to the second server and vice versa But when I make calls to users in other server , on my client , I get the caller if as asterisk@192.168.20.99 , the same I get when I try reverse , ie I get on my cleint caller id as astersik@192.168.20.32 Please guide me what

Hi I am looking for some advice or tips on how to make asterisk , to dial a number , when the asterisk server gets some mail to the asterisk user , Is it possible to do so Guidance requested Thanks Joseph John ___________________________________________________________ Switch an email account to Yahoo!

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

Hi I am experimenting Asterisk , so far I am able to talk from two sip clients under one server and in the same network, [ Thanks to the mailing list ] Now I want to have two or more Asterisk server and SIP clients from one server communicating to the other sip clients in another server when I had searched I found this link

On 12/14/14 21:17, Clayton Kirkwood wrote: > Thanks, Mark. Um, how's about from the commandline or how do I get, I guess we're still using X11, windows to load. > > Sorry, :<}}} > > Clayton > >> -----Original Message----- >> From: centos-bounces at centos.org [mailto:centos-bounces at centos.org] On >> Behalf Of Mark LaPierre >> Sent: Sunday,

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Thanks, Mark. Um, how's about from the commandline or how do I get, I guess we're still using X11, windows to load. Sorry, :<}}} Clayton >-----Original Message----- >From: centos-bounces at centos.org [mailto:centos-bounces at centos.org] On >Behalf Of Mark LaPierre >Sent: Sunday, December 14, 2014 5:50 PM >To: centos at centos.org; Mark LaPierre >Subject: Re:

Hi Vivek, Have you read the mailing list threads on this topic? I don’t believe we’re quite ready to make the switch yet. There was a discussion last October about what was left to be done. I’m sure it has been discussed since then too. Here’s a link to the start of the October discussion. http://lists.llvm.org/pipermail/llvm-dev/2017-October/118280.html If you’d like to get involved, one

a strange problem of multiple sip registrations and peer selection in sip.conf is calling for your suggestions!! let's examine this scenario: some numbers and passwords hidden with HHHs to protect the guilty :) I have 3 distinct sip subscriptions with cordiaip.net provider in US. For each of these i insert in sip.conf (with peer name differences and relefant number/password differences,

We are attempting to configure SIP trunking between asterisk 1.2.22 and a Mitel 3300 CX box. The Mitel machine will gateway to the PSTN for us. I found this earlier post about doing this from July: http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html Unfortunately the promised configs never came ;(. We're having the exact reverse problem: we can register with the Mitel

Hi I am using a 4FXO , TDM400P card I am able to call outside , after modifiying extensions.conf with exten => _9X.,1,Dial(ZAP/1/${EXTEN:1}) using this , I can only dial through one of the port , Actually I want to dial outside using round - robin search After reading the manuals , I have plans to modified the above line as exten =>

On 12/14/14 20:01, Clayton Kirkwood wrote: > Hello, > > It's been about 15 years since I've enjoyed working in the Unix space, and I > am trying to reintegrate myself. A few things have changed in the > intervening years. I've installed the first Centos6 iso without too much > difficulty, but I am rusty on commands and such. It appears the install > doesn't

Hello, It's been about 15 years since I've enjoyed working in the Unix space, and I am trying to reintegrate myself. A few things have changed in the intervening years. I've installed the first Centos6 iso without too much difficulty, but I am rusty on commands and such. It appears the install doesn't install mans so would somebody suggest a way to find and install them. It

I want a CentOS 62 (amd64) KDE desktop with KDM as the GUI login manager. My system environment is as follows: Host OS: openSUSE 11.4 .(amd64) with VirtualBox (64 bit) GuestOS: CentOS 6.2 (amd64) 10GB virtual disk I chose the "Customize Now" option in the installer and chose "KDE" group w/o the "Desktop" group. The system boots to a CLI console. I can login and

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

This is part of a rails project. The following method is part of the Ams class (a rails model). I''m a bit unsure of the rspec/bdd way of testing this method. def persist_as_domains @current_domains.each do |d| dom = Domain.new dom.domain = d dom.source_id = 1 dom.at = Time.now dom.save end end The following is what came out when I tried to write my test. Notice

Hello, I'm interested in knowing if anyone out there has successfully connected Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that we put in an Asterisk install, one of their sister companies who we don't control is putting in a Cisco UC 560. From my looking I think it can be done, but the vendor is telling them it can't. Thought I'd ask around here and see

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.