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Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd

...ocess? For any length phone number for that matter-- like dialing another extension. If I dial 7005, I'll have to wait a while.. but it's instant when I press the # key. Daniel ------------------------------ Message: 13 Date: Mon, 18 Jul 2005 18:17:37 +0200 From: Arnd Vehling <av@nethead.de> Subject: [Asterisk-Users] Comments on Areski Calling Card Solution plz To: Asterisk Users <asterisk-users@lists.digium.com> Message-ID: <42DBD621.4020403@nethead.de> Content-Type: text/plain; charset=us-ascii; format=flowed Hi, can anyone who has the Areski Calling Card solutio...

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hi There, we are developing a dialer application using the java lib to interface with the asterisk manager protocol. It works fine so far. The only problem we have is that if we use the "originate" command the user is required to pick up the fone _bevore_ asterisk will originate the call to the desired destination. What we would like to do is to place the call, check if the other end