similar to: Asterisk compatible E1 cards

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)

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 have 20+ asterisk servers and need to network them together so a phone on any of the servers can call a phone on any other server without any trouble. I can think of IAX trunks between every server. So every server will have an IAX trunk to every server and then prefix bases routing in the dialplan for each server (I can give a number to each server and use that as a prefix for that server).

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

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)

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)

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

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

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

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

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

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

Has anyone had any luck with voice recognisation using sphinx, if yes then could u please send some pointers. does the eagi app for sphinx really work cause i'v tierd it and sphix dosent seem to do anything -- regards Vikram (http://www.vicramresearch.com)

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

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

Does anyone have any input into integrating asterisk with a alcatel 4400 PBX. Acording to what i've found is that Alcatel uses R2 for E1 -- regards Vikram (http://www.vicramresearch.com)

> On Feb 22, 2016, at 6:27 AM, Vikram TV <vikram.tarikere at gmail.com> wrote: > > > > On Fri, Feb 19, 2016 at 10:46 PM, Vikram TV <vikram.tarikere at gmail.com <mailto:vikram.tarikere at gmail.com>> wrote: > Hi, > > Thanks for the reply. Few thoughts inlined. > > On Fri, Feb 19, 2016 at 8:00 AM, Adam Nemet <anemet at apple.com

Hi, Thanks for the reply. Few thoughts inlined. On Fri, Feb 19, 2016 at 8:00 AM, Adam Nemet <anemet at apple.com> wrote: > Hi Vikram, > > On Feb 18, 2016, at 9:21 AM, Vikram TV via llvm-dev < > llvm-dev at lists.llvm.org> wrote: > > Hi all, > > I have created a patch (up for review at: http://reviews.llvm.org/D17386) > that does Loop Fusion implementation.

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly