Hi, i have a problem trying to register the codec G729, my licence is valid but
when i try to Register i got the following error:
"Registration unsuccessful... Error code: 110
ERROR!
Your Internet connection is probably behind a proxy and the
Registration program can't communicate with our server,
however it has created the file:
/var/lib/va-infoclient
which contains your machine signature and that you must send to
Voiceage to obtain a valid certificate for the g729 library."
My internet conection is fine and i dont have a proxy. I sent the file to
voiceage support but i dont have received any answer.
Thanks for your help.
Carlos Andres Medina
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You don't use that codec anymore. Its not the one thats being sold by
digium anymore... get the new one from ftp.asterisk.org/pub and the new register
program and then register the new one.
bkw
----- Original Message -----
From: Carlos Medina
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 23, 2004 2:39 PM
Subject: [Asterisk-Users] Codec G729 Registration problem
Hi, i have a problem trying to register the codec G729, my licence is valid
but when i try to Register i got the following error:
"Registration unsuccessful... Error code: 110
ERROR!
Your Internet connection is probably behind a proxy and the
Registration program can't communicate with our server,
however it has created the file:
/var/lib/va-infoclient
which contains your machine signature and that you must send to
Voiceage to obtain a valid certificate for the g729 library."
My internet conection is fine and i dont have a proxy. I sent the file to
voiceage support but i dont have received any answer.
Thanks for your help.
Carlos Andres Medina
------------------------------------------------------------------------------
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Yahoo! Mail - 50x more storage than other providers!
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I have a telephone number that points to my asterisk service and I want it to ring to my cisco 186 when I call it but for some reason when I dial it I get a busy signal and my asterisk server rejects it what am I doing wrong? Sip.conf [dstech4] type=friend username=dstech4 secret=atapassword qualify=200 nat=yes host=dynamic canreinvite=no context=HOME Extensions.conf [HOME] exten => 8103570767,1,Answer exten => 8103570767 ,1,Dial(SIP/dstech4,20,rt) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/312a3b7a/attachment.htm
Do you have the ATA "registering" with Asterisk? SIPRegOn set to 1. _____ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jeremy Kenney Sent: Wednesday, June 23, 2004 4:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk inbound assistance I have a telephone number that points to my asterisk service and I want it to ring to my cisco 186 when I call it but for some reason when I dial it I get a busy signal and my asterisk server rejects it what am I doing wrong? Sip.conf [dstech4] type=friend username=dstech4 secret=atapassword qualify=200 nat=yes host=dynamic canreinvite=no context=HOME Extensions.conf [HOME] exten => 8103570767,1,Answer exten => 8103570767 ,1,Dial(SIP/dstech4,20,rt) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/12738b92/attachment.htm
Ata186 are already registered with asterisk and I can call them ext to ext _____ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Kubat, Philip Sent: Wednesday, June 23, 2004 4:47 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk inbound assistance Do you have the ATA "registering" with Asterisk? SIPRegOn set to 1. _____ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jeremy Kenney Sent: Wednesday, June 23, 2004 4:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk inbound assistance I have a telephone number that points to my asterisk service and I want it to ring to my cisco 186 when I call it but for some reason when I dial it I get a busy signal and my asterisk server rejects it what am I doing wrong? Sip.conf [dstech4] type=friend username=dstech4 secret=atapassword qualify=200 nat=yes host=dynamic canreinvite=no context=HOME Extensions.conf [HOME] exten => 8103570767,1,Answer exten => 8103570767 ,1,Dial(SIP/dstech4,20,rt) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/5662c2f9/attachment.htm
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