similar to: Asterisk AND Cisco Phones in H323 cloud...problems with some models.

Hi, I've tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

Hi to all, I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper. I've tested h323 using ohphone and I can talk between them, then I've tested SIP with Twinkle softphones and function very well. Now I have to perform call from h323 to sip and viceversa. How can I do it ???? I receive h323 call from a Cisco Voice GW to my Asterisk and this call have to go to a SIP phone:

Hi to all, I would like to limit the numbers of inbound h323 connections for different extensions, for instance, I've the following rules in my dialplan: exten => 123,1,DIAL(H323/1100) exten => 234,1,DIAL(H323/2200) and I would like to limit to 5 the number of h323 connections for exten 123 e to 2 those for 234. The reason to limit the number of connectios is that these h323

Hi everybody, I have still problem with setting-up asterisk. I use asterisk with gnu gatekeeper and h323 phones. I read lots of much documents, but there's no any reference to setting-up "how to put on hold an incomming call". I mean: 1.) somebody call me from PSTN (via my ISDN BRI card in asterisk); 2.) asterisk routes this call to gatekeeper and finaly I pick up call by my

Hi, I'm trying to get h323 communication working between asterisk (0.7.1) and Innovaphone Gatekeeper + innovaphone phones. chan_323 installed OK with currently recommended pwlib_1.5.2 and openh323_1.12.2. Registration asterisk with the gatekeeper works OK, externsion for my test(sip) phone gets registered with gatekeeper. when establishing a call between a h323 phone and asterisk I run into

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

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

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

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,

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 =>

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

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

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

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: ***************************************

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

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