Great service. Hope it catches on.
In time there will probably have to be some adjustments:
- ratio 10/1 - this means everyone can make 10 calls for every call donated.
This doesn't add up in the long run.
- I get credited 10 calls for every call I make to my own dialplan. This
should be fixed.
Title: Message
Does anyone have a
workingiax1.conf for Gnophone (the softphone version) of www.voiceglo.com?
Thanks
__
Stig
Hess
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Still won't try to grab the ringtones on my Grandstream (Mac
ends with 0028B8)
Stig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: 3. juni
2004 00:14
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Where I can find
Voicepulse will use it. If I don't allow GSM Voicepulse will use
ILBC.
Does anyone know how to achieve this? Can "allow="
be put into each extension/provider definition?
Thanks,
__ºªº Stig Hess ºªº Iaxtel:
1-700-2854373USA: +1-212-400-7920Norge:
+47-51665681
Subject: Re: [Asterisk-Users] Iaxtel and
Voicepulse
You must use GSM with iaxtel and Voicepulse for now... I talked
to the guyfrom voicepulse and they said ilbc might be turned sometime in
the future.But not sure.bkwOn Tue, 14 Oct 2003, Stig Hess
wrote: I'm having trouble configuring
I found the answer: One can disallow a codec within
each context of iax.conf.
Stig
- Original Message -
From:
Stig Hess
To: [EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 11:43
PM
Subject: Re: [Asterisk-Users] Iaxtel and
Voicepulse
Voicepulse _has_
Greetings,
I was wondering if somebody is working on a
solution to the NAT/SIP-issues? It seems to me that the problem has been
identified, is that correct?
Just hoping that someone with more skills will
provide us with a solution sooner or later...
Regards,
Stig
I meant where Asterisk is behing a NAT... sorry for
the confusion.
Regards,
Stig H.
- Original Message -
From:
WipeOut
To: [EMAIL PROTECTED]
Sent: Sunday, September 28, 2003 8:21
PM
Subject: Re: [Asterisk-Users] NAT/SIP
solution?
Stig Hess wrote
