similar to: have two H323 connection: one with GK, one with other GW. is it possible?

Hello group, I am trying to register to a opengk h323 gatekeeper using chan_h323. The gatekeeper expects me to register a username like 31201234567@gatekeper.com with a password secret and an e164 of 31201234567. Thus I put the following in the config file: [general] gatekeeper=w.x.y.z. AllowGKRouted=yes [31201234567@gatekeer.com.com] type=h323 e164=31201234567 secret=geheim

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

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

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

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

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

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

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

[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)

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

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 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

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates

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, 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

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified