similar to: E1's and span - what questions to ask my service provider

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 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).

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)

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

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

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

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 >

I'm working on an Asterisk system with all of it's PRI ports configured with the LBO setting at 5, like this: span=3,0,5,esf,b8zs As of yet, I am unwilling to change the LBO to 0 to where it probably should be because the system is working and I'm not sure exactly what the LBO does. I'm aware some changes were made to deal with low audio levels. Does anybody know what audio

I would look at using serveral machine splitting up the load using one 4 port card in each. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > janvb@caselaboratories.com > Sent: Tuesday, January 10, 2006 6:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion >

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

hi I want to compress and trunk 2E1 capacity over a satellite SCPC link using asterisk. I am nw to asterisk and I need suggestions on how to implement this. Best regards Arinze --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL:

MvPhone wrote: >Hi, > >You can put max of 2 4xE1 cards per dual CPU box. That means that you >end up with 4 dual CPU boxes > >4 boxes * 2 cards * 4 E1 = 32 E1's > >I sell affordable quadE1 cards that work well with asterisk. You can >check on pbxhardware.com > >As for the dual CPU boxes it depends what will you do with the traffic >/ calls ... Are you

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

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)

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

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