similar to: sphinx voice recognisation

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 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

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 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)

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).

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. >> >> So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do?

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

forgot to mention Gluster version 3.10.6 On 18 October 2017 at 13:26, Alastair Neil <ajneil.tech at gmail.com> wrote: > a short while ago I experimented with tiering on one of my volumes. I > decided it was not working out so I removed the tier. I now have spam in > the glusterd.log evert 7 seconds: > > [2017-10-18 17:17:29.578327] W [socket.c:3207:socket_connect] 0-tierd:

a short while ago I experimented with tiering on one of my volumes. I decided it was not working out so I removed the tier. I now have spam in the glusterd.log evert 7 seconds: [2017-10-18 17:17:29.578327] W [socket.c:3207:socket_connect] 0-tierd: Ignore failed connection attempt on /var/run/gluster/2e3df1c501d0a19e5076304179d1e43e.socket, (No such file or directory) [2017-10-18

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)

Hi, All, We create a GlusterFS cluster with tiers. The hot tier is distributed-replicated SSDs. The cold tier is a n*(6+2) disperse volume. When copy millions of files to the cluster, we find these logs: W [socket.c:3292:socket_connect] 0-tierd: Ignore failed connection attempt on /var/run/gluster/39668fb028de4b1bb6f4880e7450c064.socket, (No such file or directory) W

Hi, We create a GlusterFS cluster with tiers. The hot tier is distributed-replicated SSDs. The cold tier is a n*(6+2) disperse volume. When copy millions of files to the cluster, we find these logs: W [socket.c:3292:socket_connect] 0-tierd: Ignore failed connection attempt on /var/run/gluster/39668fb028de4b1bb6f4880e7450c064.socket, (No such file or directory) W [socket.c:3292:socket_connect]

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

Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with conferences (using both meetme and app_conference) and the audio sent out to an

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like also to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.