similar to: Simultaneous Inbound and Outbound calls on analog lines...

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,

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

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

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

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

Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Is this possible with Asterisk and what conf would I need for that? Thanks, -JE

Hello! Is it possible to assign any of the soft keys on the Polycom IP series handsets to a specific function in the feature menu? I'd like to assign one of the keys below the LCD to function as a Do Not Disturb button but I have not been able to find a helpful guide or proper documentation that is understandable for the task. Any help is greatly appreciated. Thank you! Tim Nelson

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Greetings! I'm hoping someone can help me with what should be the most basic of problems. Essentially, I want to have certain calls on an Asterisk 1.2.25 (Yes I know its old, upgrade, etc... its on my roadmap) install go out a couple of analog lines and all other calls go out a PRI. The analog lines are setup in Zaptel group 1 and the PRI channels are in Zaptel group 0. Here is my relevant

I have an Infortrend RAID box I'd like to see as one big 6TB partition, but I only can get 2.2TB partitions to work. I was trying to do this with an Adaptec controller but apparently they are only (any of them) 48 bits wide. Does anybody have a working system for SCSI/Centos over 2.2TB? Milt Mallory Topix.com 650-461-8316 Always consider the issues of progressive enhancement and

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

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:

This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ A brief read-over looks like it uses iaxclient and ruby to war dial a range of numbers and record audio samples to be analyzed to identify if the call was answered by a modem, fax machine, human, etc. The calls are placed through a PSTN termination

Greetings list- I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no problems and as expected. However, I would like inbound calls on that POTS line to go unanswered by Asterisk since I have other equipment on the line. I've setup zapata.conf for the channel without a context but the line is still answered. I've also setup a blank context with the same result.

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

I have an interesting observation which I thought I'd pass along to save other people from spending time trying to 'fix' it. One of my clients uses Charter's so called "business phone service". They provide 'analog' phone lines over IP. In general, they've worked OK. End users were saying that the phone are "cutting out" at times. What

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!

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:

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