similar to: IAX2 - Separate Signaling and Media?

Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Regards Bilal

Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every

Greetings All- I'm about to embark on some remote management testing and need a way to login to a remote system running CentOS 4.x/5.x via SSH, su to root (using a password), then execute a command. I currently login to the boxes using key based SSH like this: ssh -i ~/remote_key admin@$REMOTEIP Then, I SU to root. However, if I try to do this automatically like this: ssh -i ~/remote_key

Mark and Ed received word today that the long-awaited RFC for IAX2 has been approved by the IETF, and is now published: http://www.rfc-editor.org/authors/rfc5456.txt Thanks to Ed Guy, Mark Spencer, Brian Capouch, Frank Miller, and Kenny Shumard! Lots of revisions and discussions have paid off. JT --- John Todd email:jtodd at digium.com Digium, Inc. | Asterisk Open

Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 00000 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on:

starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us support at cdsportal.net -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm new to asterisk and this list. The ISO download appears to have 1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI. The only option for the Asterisk GUI is to use 1.4. Is it as simple as installing 1.6 only then using the yum repository to install the Asterisk GUI? If so, what packages are needed? Thanks!

Greetings list! Because of the sheer number of box (re)installations we do that have a CentOS base, I've converted our provisioning process to PXE. However, I've found that an alarming number of motherboards are coming with the Realtek RTL8168 ethernet controller onboard. When the system boots, it cannot find an ethernet controller and the installer process hangs. I made a new initrd

Greetings- I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Of course, logs on Box A were not set to show debug

Greetings list- I'd like the ability to hangup all calls for a particular extension from the system CLI. I understand this can probably be scripted using the AMI but I'm not familiar on how to do it. Help! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105

Can anyone recommend one to run under CentOS? Dnk Sent from my iPhone

Hi ? How can I do the dhcp to assign ip for eth2 network?only? ? eth1 and eth0 can igorn ? thank you __________________________________________________________________ Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I'm having a bit of an issue with one of my installations that I cannot figure out. For some reason, when two people are in a call (both local to the * box, same subnet, pure SIP), the call will randomly be placed on hold and provide MOH to the other party. We're using Polycom IP430 handsets almost exclusively for this installation. Can anyone think of a reason why a call would

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

Greetings- First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance. I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,

HY all, I am completely new to the asterisk so can any one help me with it as I have some questions queries 1. first n for most what are the tools/equipment that I need for eg a Computer, and a net connection is it all that I need for simple head start to get hands on the asterisk and cam please any one send me all the copies from vol 1 to vol 70 every issue Thank You and Best

Greetings all- I have a system running CentOS 5.5 x86_64. It's serving NFS for several 'frontend' boxes in a web application setup. All data is stored in specific dir but written by different users. When the webapp attempts to read this information, it may or may not have permissions to this data. What I'd like to do is set 'default permissions' on the data storage

It is widely known that Quickbooks has a horrible storage engine, "requires" Windows to host multiuser access, etc. However, I've successfully been storing QuickBooks files on Samba shares at a handful of locations with no real issues other than slight performance degradation. However, a somewhat recent thread here mentioned the proper use of oplock'ing in the Samba configuration

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a "PSTN" server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am hesitant to just turn on the jitter buffer in