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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060811/2329be0b/attachment.htm
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.Here are some guesses from another relative newbie that has previously hit the problem :-) Have you set up a SIP trunk and incoming route for your stanaphone number? I think this is likely related to your freePBX config in that the SIP trunk is not in the right context. If you look in sip.conf you will see: ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying "The number you have dialed is not in service. Please check the ; number and try again." You might get more info about setting up things in Trixbox/freePBX on the amportal-users mailing list... The amportal-users mailing list is at Amportal-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/amportal-users Mike
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 10000-20000UDP (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
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. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 256 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060815/89b7538e/signature.pgp
Mojo with Horan & Company, LLC
2006-Aug-15 16:43 UTC
[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 <mojo@horanappraisals.com> Office Manager, Horan & Company, LLC (907) 747-6666 x112