similar to: asterisk, cisco 3640's and DIDs...

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

Hello togehter, i have a litte problem to convert a data.frame. My data.frame looks like this one A 1 R5000 2 R4800 3 R4700 4 3500 5 3800 I need now a command, which outputs all the numbers, without the character in front. The solution look like this one: A 1 5000 2 4800 3 4700 4 3500 5 3800 Thanks. Best regards. Mat -- View this message in context:

I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots

Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers.

Hi, I am using *Thinking Sphinx version 2.0.11 *in one of my project. To create an index I ran* **rake thinking_sphinx:index *and I got the following error Generating Configuration to /home/saidev/Ubuntu One/burgundy/ssearch/config/development.sphinx.conf rake aborted! At least one field is necessary for an index Tasks: TOP => thinking_sphinx:index (See full trace by running task with

Hi, I have been seeing data corruption on the ZFS filesystem. Here are some details. The machine is running s10 on X86 platform with a single 160Gb SATA disk. (root on s0 and zfs on s7) ...Sanjaya --------- /etc/release ---------- -bash-3.00# cat /etc/release Solaris 10 6/06 s10x_u2wos_09a X86 Copyright 2006 Sun Microsystems, Inc. All Rights

Duration: 3months + Location: SFO; CA The primary skills are: * Over all Red Hat Linux system admin - Patches using our own patch center - Spinning up approved version of the Linux - Migration of Red Hat 4.0 to 5.0 release in all our environment - Network trouble shooting & Design (mainly for our qa and dev environment) - Backup setup and Strategies - General scripting skills using bash,

Hi, I've been trying to set up Exin and Dovecot and think I've almost gotten there. However I'm not recieving mail to my inbox and I get messages like the following in /var/log/exim/main.log: 2005-09-28 04:14:47 1EKZtf-0003gQ-26 <= users-return-105525-paul= paulororke.net at openoffice.org H=s002.sfo.collab.net<http://s002.sfo.collab.net>( openoffice.org

Hi all Just after Puppet Camp we''re holding a Puppet developer training. If you want to extend, hack on or customize Puppet or write your own facts, types/providers or reports this is the place to learn how to do that. The details are: Rare Puppet Master Developer training (only held 1-2 times a year) October 11-13 (9 AM - 4 AM everyday) San Francisco, CA (Downtown area)

I would like to setup a system with two tennants with two seperate DIDs through one IAX provider account. Is it possible to route the calls into different contexts based on the DID dialed? I have searched and found nothing. I do not see anywhere in the console that says what DID was dialed so I am thinking two seperate accounts are needed to make this work. Can anyone confirm? Thanks

Hey there, I don't know who else has suffered broadvoices terrible service, but I am about to my end with them. The lack of a LBR codec, the outages, the changing of servers without notifying subscribers haspushed me to my end. Now most incoming calls are abbruptly cut off within a minute of the call starting. Anyone know of any other * friendly providers that have DID, besides Voicepulse,

I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the "smaller" cities. The question is how do I actually setup these DIDs? Thanks, Geoff

Does anyone know of a company that provides German DIDs (preferably Berlin) and termination of calls to Germany at reasonable rates? Thanks, Eric jacksch@tenebris.ca

I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up

Hello, one of the numbers where historically configured to act the following way: 123456: Ring All Desks 123456-1: Ring Desk 1 123456-2: Ring Desk 2 ... (I think you get the idea) Configuring asterisk to do the same isn't that hard, but I now have one problem, with users calling that number from PSTN. Those particular users go off-hook and start dialing the number. The ZAP Channel claims

Anybody know of good reliable Asterisk compatible DIDs in the 636 (Missouri, USA) area code? Voicepulse doesn't go there, and Broadvoice seems unreliable in my Asterisk installation -- so I'm reluctant to recommend it. Thanks, </edg>

Hi, I am looking for a provider that can provide DIDs in Atlanta, GA (national in future) and who can port an existing number. LiveVoip is a perfect fit but doesn't do LNP :-(. Currently I only need a handful of numbers ported, but will have more requirement in future. I'll need unlimited simultaneous incoming/outgoing calls (can pay for incoming as well). Any pointers are much

Anybody know a good IAX provider for Canadian DIDs? Mohit. -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't."

I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs

Anybody know where to find Thailand DIDs that can ring in to my * in the USA on SIP? Oh, and a good price, too! ;) Chris Coulthurst chris@shuksan.com