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i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, i need an asterisk board that can support up to 30 FXS ports. i found the following on digium: TDM400P. It only has 4 port, does it mean that i have to buy 8 of those to have my 30 FXS line? Or is there any channel bank solution that digium or other providers can provide us with in order to do that. Experienced asterisk user, how do you manage to make PBXs for 30 to 100 regular phones?

Hello all, I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering if by any luck, i could make some magic happen with asterisk ... If asterisk does not support it, is there any PSTN to H323 or PSTN to SIP gateway that support this dialogic card and that can be connected to an Asterisk Box? Digium, I PROMISE that I will buy my cardf rom you once my tests are conclusive

Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

...; Regards, > > Patrick Anderson > Digium Hardware Support Specialist > Digium, Inc. > 150 West Park Loop > Suite 100 > Huntsville, AL 35806 > support at digium.com > Toll Free 1-877-LINUX-ME (1-877-546-8963) > Local 1-256-428-6000 > > > > -- Kido NOAGBODJI Directeur NTIC C.A.F.E. Informatique & T?l?com' Cit? Maman N'Danida Route de Kpalim? B.P. 12596 Lom? Togo P. +228 225 55 55 F. +228 225 66 66 GSM. +228 946 55 33 GSM. +228 935 55 33

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to