similar to: Software for my laptop to send Fax via H.323 ?

I have wanted to setup an Asterisk system for several months but have been unable to do so because I lacked the funds for a T1 card to connect my channel bank to the server. Recently, I acquired several Brooktrout T1/E1 interface (PRI-PCI48VC/PRI-PCI64V-C) cards for the PCI bus. According to the Brooktrout web site http://www.brooktrout.com/products/netaccess_pri_pci/specs.html Linux

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on

I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1 card, which, upon checking with brooktrout, is supported for linux 2.x, but before I do this, I want to check and see what the opinions of your, the list members, and Mark, of course, as far as asterisk being able to use this card. ANY information would be helpful, as this offer will expire to me very soon. Thank you. :)

Don't want the opening line of my post to be misconstrued. "Anyone familiar with the BrookTrout TR1034 Fax Board, supports T.38 and PSTN fax in an Asterisk environment?" That's a question, not a statement. Curious if anyone has played with one of these as an extension of an Asterisk server for T.38 faxing, did not mean to imply it could, should or would play well with

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout:

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the

> From: "Jason Aarons \(US\)" <jason.aarons@us.didata.com> > > Glad to hear you had a workaround. > > I would suggest re-queing your TAC case, perhaps you got a outsourced or > less experienced engineer at Cisco. Their support has varied depending on > which city/group you get. Some have more experience then others. > > While your 2600 from 2001

Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay "I Guess" When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like.

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

Hi, I am trying to make a call from SIP to H.323 using chan_h323. Asterisk CVS-HEAD-03/10/05-10:08:22. As given in chan_h323 readme I compiled pwlib and open h323 versions 1.8.1 and 1.15.1. Call seems to be get connected but no audio path. I see following; -- AGI Script Executing Application: (DIAL) Options: (H323/YYYY#XX112422428@XX.103.19.91/XX112422428|60|HS(63840)) -- Setting call

I'v got the following scenario: Two clients (ohphone) are calling (one at a time) the host with asterisk, which then connects to the SIP client. One of these hosts let's asterisk crash with a segmentation fault (i can provide the core file, if needed) in the second, the SIP client accepts the call. However .. if that client get's to the voicemail instead, because the SIP client is

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks

I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI>

Hi, I need an Asterisk with G729 support. Preference is with Asterisk 1.2(.18), but if not possible, then it can be 1.4. Question is, can I enable G729 for both protocols? do the H323 implementation allow it? I found the codec support for H323 in 1.2.18 very poor ... only got u/a-law to work ... not even GSM. Would the Digium G729 license be good both for SIP and H323? Cesc

hello everyone i have a simple question: i have an asterisk which is a h323 gateway and has a h323 connection to a cisco gatekeeper and a sip connection to a pbx. my question is: how can i send all calls to gatekeeper? i searched a lot and found that i should set gatekeeper=192.168.0.X (ip address of my gatekeeper) in h323.conf file. but what about extensions.conf file? should i define an

Hi all, I have two problems with h.323 in * The first one is, I can call my voip-phone, (exten => 59305004,1,Dial(H323/${EXTEN}@192.168.0.41)) BUT, I hear nothing in h.323 debug mode: *CLI> Allowed Codecs: Table: GSM-06.10{sw} <1> Set: 0: 0: GSM-06.10{sw} <1> -- Making call to 59305004@192.168.0.41. == New H.323 Connection created.

Hello, I'm trying to setup an asterisk extension to be attached to an H.323 gatekeeper so that an asterisk application (Astcc) answers H.323 calls from any terminal logged into the gatekeeper. I'm using asterisk's channels/h323 implementation, and I've configured the following in h323.conf: [general] port = 1720 bindaddr = AAA.BBB.CCC.DDD allow=all gatekeeper=XXX.YYY.ZZZ.AAA