Notice that one of the prohibited items is:
# Phone Services - includes 800 or 900 phone services and audio text
services, prepaid phone cards, and prepaid phone services.
https://payments.amazon.com/sdui/sdui/about?acceptableuse
--
James Moore
ja...@restphone.com
http
`)
VALUES \
('2007-11-05
17:25:17','(removed)','(removed)','s','restphone_event_loop','SIP/icall-0075a2e0','','Read','Result|/var/lib/asterisk/sounds/restphone_cepstral/016d4fda5256dc9a944d7102fac4',25,15,'ANSWERED',3,'1\
','','');
What am I missing? I'm running 1.4.13.
- James Moore
On 11/5/07, Carlos Chavez [EMAIL PROTECTED] wrote:
Do you have userfield=1 in your cdr_mysql.conf file?
Thanks - that took care of it.
- James
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I'm slightly confused about how SIP security and authorization works.
I've looked at the Wiki (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer)
, but it's, um, flawed:
As of Asterisk 1.2, there is no reason to actually use 'user'
entries
any more at all; you can use
-1.2-r31555.
- James Moore
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I'm using meetme with three SIP calls, updated to the latest Asterisk CVS
(as of around 10am Nov 3). After a minute or two we start getting
substantial cutouts of the audio during the call. I'm using the ztdummy
timer, also the latest CVS release.
Suggestions for things to look at?
There are
What about doing something like say that the files localized for your
particular installation are in another language? Just do
something like language=enmycompanyname, put your modified sound files
in /var/lib/asterisk/sounds/enmycompnayname, et voila! No more
overwriting of your files, and you
in the standard
installation.
- James Moore
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