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Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

I am setting up 2 nufone lines. I want to make them both availiable for dial-out. How do you syntax it in extensions.conf so that it figures out which one is avaliable and dials out on it. Also how do you setup the name part of callerid for the outgoing lines?

I have installed a Billion ISDN card in my Asterisk. Calls between sip and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten => s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten => 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814

I use http://www.voip-info.org/wiki-Asterisk+callingcard You only should compile the prepaid.c (look at readme file). Regards. Julio ----- Original Message ----- From: <clive18@webmail.co.za> To: <juliom@telelinkusa.net> Sent: Tuesday, April 20, 2004 12:47 AM Subject: fwd:Re: [Asterisk-Users] Asterisk prepaid debug > Hi > > Which pre-paid app are you using?. > Is it the one on the wiki? > > Any pointers will be apprciated?. > > Thanks an...

...pics: > > 1. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean) > 8. Re: Caller entered digits ignored during > wait.... (Stig Andersson) > 9. RE: Exception flag set - snom200 (jc) > >...

...pics: > > 1. Re: Asterisk Security Audit? (Steven Critchfield) > 2. DTMF Detection Problem (Ron McMillin) > 3. Re: Caller entered digits ignored during wait.... (Tilghman Lesher) > 4. Re: Sipcall.co.uk & [*] (Dave Cotton) > 5. Re: IAX2 trunk mode over satellite (clive18@webmail.co.za) > 6. Register vith SIP provider from behind NAT (Simon Brown) > 7. Can't talk on Cisco VIP 30 using Chan Skinny (Dean) > 8. Re: Caller entered digits ignored during > wait.... (Stig Andersson) > 9. RE: Exception flag set - snom200 (jc) > >...

Hello, I would like to buy a SIP IP phone, but I don't know wich one to choose... Can you tell me wich IP Phone is known to work with Asterisk please. I've seen the Cisco 7940, but I don't know if it works, and how expensive is it ? I'm french, so if you know some french resellers, tell me. Thanks a lot, ---------------------- Fabrice Tereszkiewicz Sawadka.org

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems