Re: [asterisk-users] Abstraction for a newbie
Dominic,I am not familiar with Trixbox, however a similar service from Sipgate in the UK is configured in Asterisk quite easily. Check out this link. http://www.sipgate.co.uk/faq/index.php?aktion=artikelrubrik=715id=540lang=dehighlight=asteriskFrom what I understand, these type of 'trunks' are really just SIP accounts. Having Asterisk or Trixbox register the account gives you control of how to use the line, as opposed to configuring a SIP account on a telephone. Hope this helps,ColinOn 8/14/06, Dominic Son [EMAIL PROTECTED] wrote: Thank you Mark. I've went from The number you are dialing is not in service, please check the number and dial again to a fast busy tone...I think I'm getting closer..-- Anything else, let me know. -Dominic Sonwww.DominicSon.com On 8/12/06, Mark Phillips [EMAIL PROTECTED] wrote:Sounds to me like you don't have a proper connection with Stanaphone. The only time you'll get these problems is when they cannot contact youto forward the call to your system.Double check you firewall settings. They need to be able to reach yoursystem on port 5060UDP (assuming SIP) as well as ports 1-2UDP (Asterisk default media ports).They'll contact yo when a call comes in. You'll accept the call and atthe same time tell them which port to send the incoming audio to.They'll also tell you where to send your outgoing audio. Hope that helps.MarkOn Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abstraction for a newbie
Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? I had a similar problem. Turned out the extension was not correctly configured. If you check the console with asterisk -r and run with debug/verbosity up around 5 you will see that the call is hitting your asterisk box, but asterisk doesn't know what to do with it. Make sure the extension is correctly configured in extensions.conf, reload and try again. Hope that helps. Regards, Austin. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abstraction for a newbie
Generally, the turn off voicemail function you used tells the provider *what to do when they can't get ahold of your asterisk* -- this doesn't typically mean 'send all calls to my voicemail until I turn this feature off' All calls should attempt to contact your asterisk server (or at least check if it's registered recently) before sending the caller to voicemail. As Mike pointed out in another post, this issue is almost certainly either with your sip.conf or your firewall config. Moj Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. !DSPAM:500,44dd080a25292693510148! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,44dd080a25292693510148! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abstraction for a newbie
Thank you Mark. I've went from The number you are dialing is not in service, please check the number and dial again to a fast busy tone...I think I'm getting closer..-- Anything else, let me know. -Dominic Sonwww.DominicSon.comOn 8/12/06, Mark Phillips [EMAIL PROTECTED] wrote:Sounds to me like you don't have a proper connection with Stanaphone. The only time you'll get these problems is when they cannot contact youto forward the call to your system.Double check you firewall settings. They need to be able to reach yoursystem on port 5060UDP (assuming SIP) as well as ports 1-2UDP (Asterisk default media ports).They'll contact yo when a call comes in. You'll accept the call and atthe same time tell them which port to send the incoming audio to.They'll also tell you where to send your outgoing audio. Hope that helps.MarkOn Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Abstraction for a newbie
Sounds to me like you don't have a proper connection with Stanaphone. The only time you'll get these problems is when they cannot contact you to forward the call to your system. Double check you firewall settings. They need to be able to reach your system on port 5060UDP (assuming SIP) as well as ports 1-2UDP (Asterisk default media ports). They'll contact yo when a call comes in. You'll accept the call and at the same time tell them which port to send the incoming audio to. They'll also tell you where to send your outgoing audio. Hope that helps. Mark On Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
