similar to: H323 ID's

Hello, Has anyone had any good or bad experiences with the new Sipura 2.0 firmware? The 1.0.33 works pretty well but there are a few more features I'd like to have. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com

Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards,

Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com Hello, I'm running a bone stock * box that only has SIP clients and about a dozen cisco T1 gateways. Some of the higher delay users complain that they occasionally hear echo on local and long distance calls, which eliminate the gateways as the source of the problem. Problem #2 is that it takes a long

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

I am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/${EXTEN}@192.168.1.2) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets being generated by * trying to communicate to 192.168.1.2. Can someone point out what I

Hi, I have about one year of experience with Asterisk, working with ZAP (digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite clear to me, the problem is that I have no experience with H323, but now, I need to use this also. The problem that I have is very trivial, so I think that this should be a very easy question for you guys whom know how it works. All I want to do,

Dear All. I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the chan_h323.so modules already included. When i try the SIP channel, asterisk works fine. In this case, i use the XLite softphone as client, i can hear the voice transfered through clearly. Asterisk is good!! :D The problem is when i try the H323 channel, the voice cannot be transferred through. I have

I try to get H323 to run, but have so far only partial success: There is a Gatekeeper GK, where asterisk connects to. The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper. From the Network on the GK, asterisk is reachable via the number 070333333. I have an extension on asterisk 6002, which is reachable. I try to call a number attached to the gatekeeper (070168177) with the

Driver chan_h323.so ---- If extension is exten => 0119823,1,dial(h323/0119823@10.10.10.1) then dial is OK: Executing Dial("SCCP/goran-00000002", "h323/0119823@10.10.10.1") in new stack ---- But if extension are something like: exten => _011xxxx,1,dial(h323/10.10.10.1/${exten:3}) exten => _011xxxx,1,dial(h323/${exten:3}@10.10.10.1) exten =>

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

Dears; Now using Asterisk H323 (which coming with Asterisk, I just compiled PWLIB and OPENH323), now I am placing a call from the IP Phone, the call comes to Asterisk, and it goes to the default context, but did not hear any voice of the played wave file. 1) Why Asterisk accepted the call without authentication? At least, it should be added to the h323.conf. 2) In case we found the method to

Hi list, After several tries to understand the subtil description in the h323.conf to be able to make the next scenario I was presented the following error messages by asterisk. Can somebody tell me please what I am doing wrong. Scenario: Gatekeeper (h323) --> Asterisk PBX -->(h323) Gateway Endpoints are connected to Gatekeeper. Call does come in like 999931235650087 with codec g711

Hi, I'm using D-Link DG-104SH (H323 4 port FXS gateway) with analig phone connected to it and X-Lite softphone as endpoints with * When I calling from X-Lite to analog phone it's ok When I dilaling X-Lite from analog phone, X-Lite si ringind but whei I picked up X-Lite connection drops IP of DG-104SH is 192.168.1.3, H323 ID is GW1 X-Lite number is 233 Here is * output: -- Executing

What is the difference between H323 and OH323 in Asterisk? I need Asterisk to have basic H.323 support so we can offer some simple H323 termination for some of our Cisco and Quintim hardware. Our upstream provider uses SIP, so I figured I'd use Asterisk as the go-between. I already setup Asterisk so it can push calls out through our providers via SIP. I just need a good/solid/very simple H323

Hi list, why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)? could anybody please give any idea to solve this issue? Please, let me know. Thanks in Advance. N.B. The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are: ***************************************

During debugging of H323 trunk side (using Jeremy Macnamara's H323 driver in ~/channels/h323) a couple of things come don't quite work as advertised... 1/ the following line in extensions.conf explicitly sets the outgoing caller ID (required in my case for downstream GK processing..) exten => _1NX.,1,SetCallerID,6400047602100 exten => _1NX.,2,Dial,H323/${EXTEN:1} what

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

Hi! i have to make pabx to direct calls to h323 terminals. i have an h323 gateway available and wish to use asterisk as the gatekeeper for call direction and queueing etc.I am a beginner at asterisk and to link openh323 with asterisk for my project i searched on net i found different compilation instructions from different sources. having no idea i followed two sources and issued commands as

Hi all: I don't know if exist any other mailing more apropiated for this question. If exist, please let me know. I need orientation for this situation: 1. 1.4.13-BRIstuffed with support for h323 with asterisk-h323 module 2. An analog Pbx with support por h323 make asterisk a call, that asnwer and put with MOH 3. At this point I want asterisk to make a native h323 transfer of the current

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the