similar to: Interconnecting Cisco 1760 routers with Asterisk

Hi all, Our company has deployed a Mitel 3300 system (only about 2.5 years ago) and we are experimenting with setting up Asterisk in our head office (for business continuity, ie we have a bird flu epidemic and no-one can come in, therefore use SIP softphones at home to co-ordinate activity) and at a remote site in the Isle of Man (connected via 2Mbps SDSL) Ideally we'd like anyone on either

hallo everyone, fyi ... working SIP Trunk configuration between Asterisk and Mitel 3300 ICP attached. let's refine further, please test and share your feedback, regards, Joseph Okoegwale Abuja, Nigeria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070709/4f7f15e1/attachment.htm --------------

After I ran my program through Rubyscript2exe, the Random.exe file that it created is causing the following error: D:\Ruby projects>random.exe C:\DOCUME~1\JEDELS~1\eee\eee.Random.exe.2\app\Random.rb:59: undefined method ''ru n'' for Wxruby2::App (NoMethodError) from C:\DOCUME~1\JEDELS~1\eee\eee.Random.exe.2\bootstrap.rb:77:in `load'' from

Good day everyone, I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and from extensions on both sides are completing successfully (details on config coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel 3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN calls through it successfully? Here is an extract of the log on Asterisk whenever I

Hi, I have a Comapq T2400H model 242688-003 and do not have the special cable to connect it to my serial port. I have tried all sorts of cable wirings that I found on the net, including ones for the powerware 5119 which appears to be the same model. I have read the documentation that comes with NUT, tried to read the source code and looked at the genericups code too.? I have

Hi all, There is someone with some experience interconnecting a Panasonic digital PBX (KX-TD1232) with Asterisk? Thank you and best regards, Dan

Hi guys, I need to interconnect an asterisk server (1.4.17) with an alcatel pbx, because i want that an asterisk extension can call an alcatel extension and viceversa. I need to do this throug a NT port (of the digium card) for technical reasons (linked with alcatel hardware). Card installation was fine, but i'm not sure of the configuration i choose. In /etc/misdn-init.conf i put my

Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated.

Hello, I am a newbie in Asterisk IP PBX but I am very impressed by its functionalities. I have read that It can work over IP network and across the PSTN. I am not very sure how it works over the PSTN.. In case if people have not yet the Internet or SDL access, I would like just to know if it is possible to interconnect two IP LANs using the traditionnal analog Network and Asterisk PBX? In such I

Hi i am search a sample config (for asterisk and for cisco) for connect a cisco 1760 with a FXO card to my asterisk. Thanks for your help Jerome

https://bugzilla.mindrot.org/show_bug.cgi?id=1760 Summary: Timestamp offset using softflowd with nfdump Product: softflowd Version: -current Platform: ix86 OS/Version: Linux Status: NEW Severity: major Priority: P2 Component: softflowd AssignedTo: djm at mindrot.org ReportedBy: stephen at

hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card. I want to configure a bri card in a cisco 1760 on port 0/0, the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated, what ever I do, no shutdown command / shutdown command, etc , the green OK light never turn on,

I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file =================

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am hoping for a little help from the collective. Inbound I have 2 different numbers. I can call in on both

CentOS Errata and Enhancement Advisory 2014:1760 Upstream details at : https://rhn.redhat.com/errata/RHEA-2014-1760.html The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 34076a399db40f55bd0d2b86619e9799eab69d3d9e7662773914af9c2166b518 kmod-lpfc-10.2.8021.0-1.el7_0.x86_64.rpm Source:

I've had two people ask me (directly and indirectly) in the last 24 hours for Cisco 1760 config examples for FXO and/or FXS settings for handing off SIP calls to Asterisk. I know I vaguely recall helping someone set that up (via IRC) a few days ago - could you forward or post your example config for others? If you'd like, I can just put it in my example directory as well... JT

Don't know if anyone's got experiences on this they'd be able to share... I'm trying to obtain numbers for Kenya/Nigeria, but I'm struggling to find a company selling them. There's one for Nigeria listed on didx.com, but $22/month seems a little steep. Has anyone had any luck getting DIDs for countries in Africa, and are there any companies out there selling them? Thanks

I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to