similar to: may convert SIP call in H.323 to words terminator??

I have P4 2.50GHz RAM 4GB, Asterisk how many calls handle using H.323 to SIP conversion on this server? Regards, --------------------------- Muhammad Asif Raza -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090210/6d5cb26b/attachment.htm

Hello, i have set up an asterisk sip to h.323 convertor, it is working OK. The only problem i have is this : For example when my identity is 12345@some.domain , and i call a sip number from a sip phone, the called party sees my identity (caller identity) as 12345@some.domain, which is the way it has to be. But when i call from the same phone with the same identity a h.323 endpoint (asterisk

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday).

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

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

Hi all, I'm a student (my thesis work consist in testing interopearbility SIP/H.323) and I begin to work with asterisk in this days. I have to testing to SIP/H.323, since today I have used Vocal system, but there are some problem for this features. In the asterisk mailing list, in the next message I've seen an e-mail """ [Asterisk-Users] Cisco

Hi, I am wondering whether I can use Asterisk as SIP/H.323 Signalling Gateway. The setup I envisage looks as follows: H.323 end-point ---------(ETH)--------- Asterisk ---------(ETH)--------- SIP Proxy/Registrar ---------(ETH)--------- SIP end-point (ETH: Ethernet) In principle, Asterisk would just be used to integrate H.323 end-points into a fully SIP-based core-network. Hence, there

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

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:

OK. So it would appear that my quest for FXO adapters unconvers more, and certainly more mature, H.323 based devices...not so many SIP devices. What would be the benefits of SIP over H.323 for a small office * server? All I need to do is bring 4 POTS lines into * with Caller ID, make outgoing local calls reliably without undo echo. FWIW, my * server is Fedora Core 1 on AMD XP2500 with 512 MB

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

we are trying to use asterisk for converting SIP to H.323 calls. asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper (gnugk version 2.0.8). the calls are going out through a cisco gateway. when I make a call from a SIP phone to a PSTN number reachable through the cisco gateway: asterisk diaplays Aug 5 23:24:26 WARNING[1255648560]: chan_oh323.c:2898

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

we got this installation : WinSip(demo version) -> ser(radius accounting) -> asterisk(from sip to h323 channel) -> gsm gateway(with 32 sims in it) we configured winsip to make 28 calls like from 28 different sip accounts, to 28 different cellular phones numbers after the first ten : -- Executing Dial("SIP/5060-081925b0", "OH323/33xxxxxx@gsm.gateway.ip") in

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

Hello, We want to make H323 calls between asterisk and avaya(lucent) pbx. We create node-name,H.323 signaling group,trunk, but we can not make H.323 calls to asterisk. Also no warnings exist in debug. Instead of giving the IP of Asterisk ,i give my computer's IP and run SJPhone ith H.323 GUI. In this time, connection is established. SJPhone accepts H323 calls but Asterisk does not. Do

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

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, is there a definite answer if asterisk can pass calls between SIP and h.323 protocols? Thanks, Yiannis.