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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)

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)

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 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).

...t; anymore. Lots of other ways to address boat-loads of speakers >> and long cable runs with current technology. >> >> Rich > > > > ------------------------------ > > Message: 4 > Date: Fri, 27 Aug 2004 19:22:29 +0200 > From: Vikram Rangnekar <vicky@freebsdcluster.net> > Subject: [Asterisk-Users] libr2 > To: asterisk-users@lists.digium.com > Message-ID: <20040827172229.GA40041@freebsdcluster.org> > Content-Type: text/plain; charset=us-ascii > > I just came across libr2 anyone using it in its current state. Specifically > some...

...nts with sip urls, ser will have to be set up. Also the audio i.e. rtp stream, traverses asterisk i.e. it acts as a middle man holding onto the call, and if you want the audio to go peer to peer (which it ideally should with sip), ser is also needed. Aisling. ---- Original Message ---- From: vicky@freebsdcluster.net To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP Date: Thu, 13 Jan 2005 17:50:39 +0100 >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...

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 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)

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 am currently installing an Audiocodes MP-108 with Asterisk in Singapore. I am able to make outgoing calls but incomming calls just get cut by the MP-108 I can see the corrosponding frontpanel led light p but the call gets cut so I'm very confused about what could be wrong. Could this problem have something to do

Hi guys, Does anyone know of a way where I can bring a third person in on my conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM and am speaking to someone now I want to quickly bring another SIP or IAX extension into this call so the three of us can speak to each other. I know I could do this by transfering the first person into a meetme then calling the second

Hi, We have recently setup a Live Demo of DRUID our Asterisk management interface product. Also I'd like to thank all of you that took the time to download the trial edition and give us your feedback. WE've tried to incorporate as much of that feedback into our new updated release. Feel free to download the trial, checkout the live demo or buy a copy :)

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

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

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)

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