RE: [Asterisk-Users] Cisco Gear

2004-01-09 Thread Arnold Ligtvoet
message posted on behalf Of Adthrawn [SNIP cisco stuff] I'll now feel ashamed, and sink into my seat :-) Best, Ad. Perhaps it would have been better to provide an email address or phonenumber where people can contact you directly. Now everybody who is interested has to reply to the list.

RE: [Asterisk-Users] FWD problems

2003-12-24 Thread Arnold Ligtvoet
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED] wrote: I've got it running through Asterisk - all working fine from a SIP standpoint. I can dial FWD numbers like 612/613/etc and everything works. However, if I dial *18005551212 or *408xxx (say, a USA

RE: [Asterisk-Users] Interconnecting Panasonic KX-TD1232 digital PBX and *

2003-12-19 Thread Arnold Ligtvoet
Dan wrote : Subject: [Asterisk-Users] Interconnecting Panasonic KX-TD1232 digital PBX and * Hi all, There is someone with some experience interconnecting a Panasonic digital PBX (KX-TD1232) with Asterisk? Ehh, what exactly do you want to do? I've got * 'interfaced' via the ISDN S0 bus of

RE: [Asterisk-Users] Asterisk and fwd

2003-12-15 Thread Arnold Ligtvoet
Shoval Tomer wrote : Hi, could anyone please provide a working sample of how to configure asterisk to connect to fwd? I've tried the one at www.loligo.com and it doesn't work. Not even when calling to 5. I presume you're looking at the asterisk console (ie. started asterisk with option

RE: [Asterisk-Users] Asterisk behind NAT How to do it.

2003-12-03 Thread Arnold Ligtvoet
;insecure=yes I'll wait your reply for the one-way sound 'issue' (probably me!) before posting to the bugtracker. Hopefully someone has some clue as to why my sip clients are not able to send sound. Thanks, Arnold Ligtvoet. ___ Asterisk-Users mailing

RE: [Asterisk-Users] Asterisk behind NAT How to do it.

2003-12-02 Thread Arnold Ligtvoet
of chan_sip.c is older than the one described, first use 'cvs update -C asterisk/channels/chan_sip.c'. Thanks, Arnold Ligtvoet. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] hold music =]

2003-11-21 Thread Arnold Ligtvoet
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Areski Sent: woensdag 19 november 2003 19:27 To: Asterisk-Users Mailing-list Subject: RE: [Asterisk-Users] hold music =] http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat FYI

[Asterisk-Users] Anybody using Sphinx

2003-11-18 Thread Arnold Ligtvoet
Hi, I'm trying to get sphinx to work with *. At the moment I believe that it won't work since there is no audio board in my server and it seems to me that sphinx expects one. Before I continue and try it with an audio board; - does sphinx really need an audio interface ? - what is the quality

RE: [Asterisk-Users] Anybody using Sphinx

2003-11-18 Thread Arnold Ligtvoet
Chris Albertson wrote: I've played around with it. I'm certainly NOT and expert but I'm pretty shur yu do NOT need a sound card. Sphinx will be happy to read an audio recording from a file. Oke this might explain it. I was trying to run sphinx-server, but got an error there leading me to

RE: [Asterisk-Users] Updated Asterisk-NL

2003-11-17 Thread Arnold Ligtvoet
://www.tric.nl/nl/Asterisk Cees, how do I use the patch included in the tar.gz? I have downloaded the * source from cvs, so the * source in in /usr/src/asterisk/ ? Secondly, can you tell me how to convert the *.wav to *.gsm ? TIA, Arnold Ligtvoet