similar to: Connecting 20+ asterisk servers together

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Does anyone have any experience ith configureing the sangoma A102 card for testing using a e1 cross cable i've configured and installed the cards properly even the lights on the card are green which proves that my cross cable is properly built too. my problem is with asterisk which gives me these errors PRI got event: HDLC Abort (6)on Primary D-channel of span 1 PRI got event: HDLC Bad FCS

In a pure voip envoirnment which uses a single codec say ulaw across all its phones can asterisk support 5000 voip sip phones on a dual / single xeon with 1 gb ram. If all the phones support reinvite (Send RTP stream directly to each other). Or would I need more than 1 system to support 5000 phones in the enviornment described above. also I am not talking about the phones using meetme or

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (http://www.vicramresearch.com)

I am planning to go in for a E1 line and whould like to know what questions i need to ask my service provider so i can connect that E1 to my asterisk box using the digium E1 card. what I mean is will my service provider give me info like LBO, framing , coding etc which i need to configure the span tag in the zaptel.conf and what about B and D channels am I allowed to setup whichever channel I

I configured agi-test.agi on extension 111 when i dial into asterisk extension 111 using a IAX softphone and hangup while the AGI is playing asterisk crashes. Does anyone have any idea why this happens. -- regards Vikram (http://www.vicramresearch.com)

After days of searching i've finally figured out that E1 lines in india use multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards which would work on R2. Also if anyone on this list is from INDIA and uses asterisk and E1 lines please let me know what type of cards (which vendor) do you use and what

I just came across libr2 anyone using it in its current state. Specifically someone from India or around India using it. Also does it work with the digium e1 cards or only the Dialogic cards. http://digium-cvs.netmonks.ca/viewcvs.cgi/libr2/ -- regards Vikram (http://www.vicramresearch.com)

Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing "#", and she heard a buzz noise in the phone and the

Hi All, Voiceroute http://www.voiceroute.net has released a new version of the DRUID Web-interface for Asterisk. There is a free trial edition available as well as a live online demo both can be accessed through the website. Tons of new features and extensive use of Ajax makes it a very easy and powerful application to use. Druid makes use of your asterisk configurations and does not damage old

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

After 20 cups of strong coffee and wasteing most of tonight and obviously doing lots of googling and emailing many people, i've concluded that dlink voip phones specifically DPH-80 dosent work with asterisk. If anyone has had anymore luck with these phones then me please let me know -- regards Vikram

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees, each with their own extension and DID, and at headquarters, we have about 70 people, again

> I have 2 PBX's, one in each office (say one in New York, one in Boston). I > have mobile users that can show up at either office and connect their soft > phones. > > > > Is there a very simple DUNDI config available which describes how to set > this up? > > Also, can I have the same outbound trunks setup in each office, so that > calls don't have to

Hello Folks; Again, excuse my cluelessness. I have an Asterisk server in the US - and I want to connect it to one in Europe. Here is my scenario: 1. call a phone number, my Asterisk box in the US answers 2. perhaps a 'please wait' voice message 3. it dials an extension on the other Asterisk box in Europe. I am not looking for someone to do this for me, I am just not really

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

All, I made some progress, but it seems the further I go with clustering the harder things get. Hmmm, I guess if it were easy, it would be documented...... Anyhow, I have 1 * server as the DUNDi peering master with a ttl=1. The only function of this server is to lookup where other sip peers are registered and forward that info on to the requesting * server. I have 4 * servers accepting

I am working on a project to unite several local school districts. We will have 14 different districts, every district would have their own asterisk box on location. We all have fiber lines running to a central location at our isd. This provides connectivity to all the districts. 1. would it be wiser to install a asterisk box at the central location of the ISD and have all asterisk boxes