On 2006-09-20 23:57:09 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said:
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to
On 2006-09-20 10:23:01 -0700, Mr. Jones [EMAIL PROTECTED] said:
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to voicemail (which appear to need
to be g711)?
No, calls to voicemail
On Sat, Dec 17, 2005 at 07:44:29AM -0600, Rich Adamson wrote:
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
my extencion.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp
On Mon, Dec 19, 2005 at 06:36:16AM -0600, Rich Adamson wrote:
ok rick i will check this directives and write you again.. thanks
ok rick all of my conf...
asterisk 1.2.1
zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side
are xten with SIP.
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use
On Fri, Dec 16, 2005 at 05:08:07PM -0600, Rich Adamson wrote:
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all
I have been having this problem since day one of my * installation, and
have concluded that it's just the echo cancellation within asterisk. I
tried all the various options in the .conf files and the compile flags in
the source code, and various codecs, but it still sounds awful (although
the
I must be doing something wrong
i have installed gsm.rpm manually and tried to recompile, but i still
get the same error.
make[1]: Entering directory `/usr/local/asterisk-0.7.1/codecs'
make -C gsm lib/libgsm.a
make[2]: Entering directory `/usr/local/asterisk-0.7.1/codecs/gsm'
gcc -O6 -march=ppc
