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...G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike 1-877-237-1777 x1000 sales at contacttel.com <http://www.contacttel.com/> http://www.contacttel.com <http://www.contacttel.com/> logo_small_ctel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/c0...

Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090406/9514bc47/attachment.htm

Hi all, We have already bought 30 Digium G.729 licenses to install in several machines, but Digium only has provided on key to use in registration. We suspect that the registration has assigned the whole 30 licenses to the same server. Do anyone know how to distribute the licenses among several servers? Thanks in advance. -- Arturo D?az Contact me on FWD: 870436 Skype: arturo.diaz.almagro

Hi all I have been hacked but no idea how!!! I noticed somebody in Eastern Europe came from an American IP and tried to call loads of international numbers. Thankfully I had no credit with my VOIP out provider so the calls went nowhere. But if I had credit it would all have been used up. I noticed hundreds of calls being made from clid and src being either UNKNOWN or as ASTERISK. Here are a

...Coursiere) 5. Re: Dialplan Priorities and Sort Order... (Alex Samad) 6. Re: Step-by-Step Asterisk and MeetMe Help (Jimmy Ezell) 7. Re: Open source SIP client (marek cervenka) 8. Re: Step-by-Step Asterisk and MeetMe Help (Jonathan Thurman) 9. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business) 10. Re: Channels configuration with DAHDI (Daniel Bareiro) 11. 1.4.24.1 -> 1.6.0.9: segfault (sean darcy) 12. Voicemail playback NEWEST first vs. OLDEST first (Karl Fife) 13. Re: Step-by-Step Asterisk and MeetMe Help (Jeff LaCoursiere) 14. Re: Step-by-Step Asterisk and MeetMe...

I am about to connect to a new provider who requires 20ms payload sizes in g729a. Is this configurable on asterisk? Is 20ms the default? Cheers, j

Who knows who decided to put insecure as the name for that option ? Not only does it confuse noobs, it really has nothing to do with security, as iirc its to accept calls from a device regg'ed or authed even if on diff ports, and for the invite.. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

Here's a little story on all the cheap guys trying to get the best rate on any route out there ( lcr and others). Anyone have 0.000001 to Mexico billed 1/1 ????? " When customers call us to ask if we sell Cuba termination for 50c/min, I sometimes joke and tell them "sure, I'll give you Cuba for 1c/min, but it will have 0% ASR". Or some other times, I

I'm having hard times with paging intercom Heres my dialplan exten => 777,1,Goto(intercom,777,1) [intercom] exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0) exten => 777,2,Page(Local/308 at page& Local/309 at page& Local/310 at page) [page] ; Paging context exten => _X.,1,Macro(page,SIP/${EXTEN}) [macro-page] ;