similar to: Which H.323 for Stable?

Hi, I'm using OH323, mostly with success, to interface Asterisk to a provider's switch (World Telecom INX). I have noticed a particular effect, and I wonder whether anyone else has seen the same? The effect is audio flutter (almost like the flutter one gets on MF or HF radio sometimes) which only happens intermittently. Audio coming into Asterisk is unaffected, as proved by using the

Hi, I'm having trouble matching the symbols/color of the key to match those specifiec in the plot. Here is the code I used: xyplot(GCR+GCT ~ FRAC, data=RWF, type=c("g","p"),cex=1.2,pch=c(22,21), xlab=expression(italic("f")),ylab=expression(italic("S")), key=list(points=list(pch=c(22,21),cex=1.2), text=list(c("Targeted Deletion","Random

Hi, anybody knows what's missing or problem why i cant compile asterisk-oh323 in my machine? i got this compiled successfully ...Openh323 - v1.12.2 ...pwlib - v1.5.2 except ...asterisk-oh323 - v0.6.5 here's the output as i run make... mkoy@sambag:~/voip/asterisk-oh323-0.6.5$ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237

P.S. I _had_ read the other posts that suggest changing to unicast sntp mode. This didn't help. I eventually setup a new ntp server on my LAN, and used it as a sync source. Everything seems OK now. Obviously a problem with my other ntp server. Cheers. -------- Original Message -------- Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon

I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI -> Asterisk side to the Asterisk -> PBX side, or will a 4-port PRI card do the job? (I already have a spare one of

I have successfully been using OH323 v0.6.5 with Asterisk 1.0.x. I now need to move to CVS HEAD in order to use some features that are not in v1.0.x, and am trying to compile OH323 to use with it. On the InaccessNetworks site, it ways that OH323 v0.7.1 is for HEAD. However, when I compile it, it appears that it hasn't been updated since the channel structures were revamped. I get many errors,

I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- ==========================================

When I dial a PSTN (via * with Digium Quad-E1) number, I hear two sumultaneous ring tones. One is coming from *, the other I assume is from the ATA. Is there an intelligent way to get around this?

I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel

I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 ? ?Fax: +613 99401650 FWD: 512237 ? ? ? ? ? ? ? ? ?

I have googled until blie in the face, WiKi'd until physically exhausted and searched through every Asterisk repository that I can find, all to no avail... No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. I can receive faxes *perfectly* (sending from standard fax macine on POTS) into tiff file (via Digium analogue card),

August 5th, 2005: Craig Southeren announced today that OPAL (http://www.voxgratia.org) now provides support for the IAX2 protocol(Written by Derek Smithies and released under the MPL). This support allows you to use chan_woomera (http://www.pbxfreeware.org) driver developed by Anthony Minessale II to interconnect your asterisk systems and use the IAX2, SIP, and H.323 protocols. I would

Do broadvoice limit the number of concurrent calls that any given sip registrant can make? What about other similar providers?

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

The one I like: http://www.rhetorical.com/cgi-bin/demo.cgi is toast. I think they went broke or got aquired by someone. Also, is there a Festival voice that sounds as good as Rhetorical or the AT & T stuff? The default one is barely legible. Since Festival is a little brutal to configure, I'd like to get someone's recommendation then go through the pain of reconfiguring it only once.

Hello list, I put together a quick note about how to "see" oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca.

I know that this has been a hot topic in the past, and that there are a number of views about the overall suitability of RADIUS, which billing methodology is best, what is the meaning of life and such... What I would like to know is has anyone found an open-source billing platform that performs basic billing functionality (pre/post) from RADIUS and/or Asterisk CDR and is written

Hello,

I have already tried to contact to persons from Digium and I did not receive a response. I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A "T38 Gateway" of sorts.