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...XXXXXXXXXXX' providing 5 concurrent calls<br> [Dec 28 15:18:08] NOTICE[2420] core.cpp: Skype For Asterisk Host-ID: XXXXXXXXXXXXXXXXX<br> [Dec 28 15:18:08] NOTICE[2420] core.cpp: Found a total of 5 Skype For Asterisk licenses<br> [Dec 28 15:18:08] DEBUG[2434] core.cpp: starting skyhost as: skypeforasterisk -z -f /var/spool/asterisk/skype/data -d /var/spool/asterisk/skype/skyhost-debug<br> [Dec 28 15:18:08] DEBUG[2434] core.cpp: skyhost environment is : HOME=/var/spool/asterisk/skype<br> [Dec 28 15:18:08] DEBUG[2433] core.cpp: starting skypewatcher as: skypewatcher 243...

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am curious to know if anyone has OPIE, S/Key, or any similar open source One-Time Password or Dual Authentication system working on Centos 4. Sincerely, Trevor Hammonds SkyHost / SkyNet Information Systems

I set up certain routes in my ASTCC application to be free of charge. When a user attempts to dial one of these numbers, the announcement plays the prompts "This call will cost", "nothing", and then terminates the script, dropping the call, leaving the card locked in the database as being in use. Any ideas? Sincerely, Trevor Hammonds -------------- next

I have an Avaya 4602 IP phone that was previously working with Asterisk. It was being used elsewhere for several months, and I recently set it up again to work with Asterisk. Everything works fine for several minutes -- I am able to receive and make calls as expected. However, after a few minutes, and every few minutes thereafter, I get the following message on the console: -- Got SIP

This message was just bounced back to me. I am not sure if it made it to the list originally or not, as I received no responses. Since this message was written, I have installed Zap hardware into this server. The Zap channels can be transferred to the Meetme conference. The IAX2 calls still cannot. Any suggestions will be greatly appreciated. Sincerely, Trevor Hammonds Trevor G.

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP