similar to: GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

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

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No

Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

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

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so

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

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hi everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab

Hi list, I have SIP clients and H323 GK connected through h323 channel (Nufone). In h323 conf I gave prefix=09 so all call starting with this prefix are send to asterisk. The context is also given their as [fromh323] But now, in asterisk, I want to have the called number without this 2 leading digits so the exten variable will be according to my actual dialplan. Here's an exemple: In

Hi list, would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side,

Hi, the new Grandstream release for the ATAs allows the setting of the FXS impedence, the Onhook Voltage and the Polarity Reversal. Anyone know how these should be set in Germany? -- Best regards Peer Oliver Schmidt

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

Hi there. I am trying to connect Asterisk to a local danish ip-telephony provider. But is having some difficulties. First I thougt they were related to the provider. But then i started debugging on the Asterisk (aix2 debug) When I make a call using AIX to the provider everything seems to work just fine: *CLI> -- Accepting AUTHENTICATED call from 192.168.1.150, requested format = 1024,

Greetings- As most of you who monitor this list know, I've been messing about with Asterisk -HEAD, Cisco Callmanager, and the nufone H323 channel driver here for some time- with pretty decent success. I'm hoping to cash in a chip here- I've run into something that is probably a very simple answer, yet not found a decent reference to resolve it. Scenario- -HEAD as of last week

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

Hello all, We have been testing Asterisk RC2 with the native H323 channel driver. We followed the instructions with the needed OpenH323 and PWLib versions and everything compiled ok. Operation of the driver seems ok, except from 2 main points: 1) Audio is passed between the two ends of the call only after the call is answered. This was not the case with previous versions of Asterisk (0.9.2