similar to: Register to h323 gk

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

Hello, I'm trying to register with a gatekeeper using chan_h323, I have a Login, Password and a user (telephone) number, This is my h323.conf [general] port=1720 bindaddr=x.x.x.x (my fixed IP) gatekeeper=x.x.x.x (gk ip) allowgkrouted=yes allow=all [xx] (my login) type=h323 e164=111 (my assigned phone number) secret=1111 (my password) context=incoming I'm getting this error when trying

Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing

Hello, Thank you for the job well-done. I installed the chan_h323 of the asterisk-1.2.7.1 and with lib pwlib-v1_10_0-src-tar.gz and openh323-v1_18_0-src-tar.gz and I used licensed g729 from digium. However, I am having a very funny behavour. 1. If I send a call on its ringing at the called side but the caller didn't get the ringing tone. 2. if the called picks up the phone, I am

Hello all, Please forgive me if this sounds a little (or a lot) ignorant as I am very new to asterisk. Right now I have two pc's connected back to back through an E100 card running asterisk. I have openh323 running as well and I am able to route calls through the E1 line. Next up I would like to be able to register asterisk with a gatekeeper. On another computer is running openGK. Using

[public_gk] ;exten => _070.,1,Set(CALLERID(number)=070333333${CALLERIDNUM}) exten => _070.,1,Dial(H323/${EXTEN}@59.120.139.119) exten => _070.,n,Hangup *CLI> h323 show peers Name Accountcode ip:port Formats 7000 ast_h323 203.160.252.147:1720 0x4 (ulaw) 88670333333 ast_h323 203.160.252.147:1720 0x4 (ulaw)

hello all, i have a conceptual question. i have a h323 gateway and it is connected to a h323 gatekeeper. my question is: can i connect my gateway to another gateway directly? i mean can these two gateways work with each other without working with gatekeeper? or when i have connection with a gatekeeper, all calls must be set through it? thanks in advance, SAM -------------- next part

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

Hi, Someone released a succefull interconnection in H323 with WTL equipement ? I'm trying to do that with an IPNx. But get dead air. With chan_oh323 it's fine, all works. With chan_h323 => dead air. The configuration is GW to GW. This is my configuration from h323.conf: [general] port=1720 bindaddr=my.ipaddr dtmfmode=rfc2833

Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper =

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

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'm trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with other PBX. The port use to connect is TCP 1720 but I can't configure this port on my * box. I'm using a H.323.conf file sample to activate the port but the * isn't listening there. Somebody have any idea or tip? This is mi H.323.conf [general] port = 1720 bindaddr =

Hi all, I try to connect two asterisk-server together. There is a server (obelix) which receives a call. This call should be transfered to another server. In my dialplan at obelix I have the following: exten => 920622201,1,Dial(SIP/outbound:geheim at asterix.local:${EXTEN}) exten => 920622201,n,Hangup exten => i,1,Congestion exten => t,1,Congestion If I call the number 920622201

Hi all, A week or so down the list, i read that not many people were using realtime on an Asterisk18, so i had this afternoon a go at it... [sorry for the inconveneant line-wraps] First i did: mysql> create database asterisk; mysql> grant all on asterisk.* to 'voipadmin'@'localhost' identified by next i used the info from the wiki: CREATE TABLE `sip_devices` ( `id`

Hi List, we are using openh323 gatekeeper for voip telefony. We also have a voip over ss7 TELES Switch for voip into POSTN Network. Know we want to use Asterisk for converting SIP to h323. Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do we need openh323 GK for astrerisk, too?. And how can i tell asterisk to sent all none SIP-ip calls to the gatekeeper over h323? thx