It is not a matter of preference, it is actually a rule [1]. Top-posting
is also an annoying practice [2] and NOT the general accepted way to reply.
And that's been the case for at least TWO DECADES. I find it amazing that
this is still being argued now.
--
On Jan 15, 2011, at 2:01 AM, Carlos Chavez wrote:
The problem is that Asterisk simply stops responding. No calls in or out
and you cannot even get to the CLI. The process seems to be running but there
is simple no activity. All I see in the log files is:
[Jan 14 16:30:46]
I'm going out on a limb here, as I'm still pretty new to Astrisk and running
my own VOIP server, however I believe there is a bug or flaw with the Music
on Hold feature.
I have it all configured and it should work, and it did briefly several
weeks ago, however now, it doesn't work at all and only
Paul Belanger wrote:
It is not a matter of preference, it is actually a rule [1]. Top-posting
is also an annoying practice [2] and NOT the general accepted way to
reply.
[1] http://www.asterisk.org/community/rules
[2] http://linux.sgms-centre.com/misc/netiquette.php#toppost
Thanks for
On Jan 15, 2011, at 9:29 AM, Don Kelly wrote:
That said, of course I want to follow this list's etiquette. I've posted a
couple times asking how I can interleave responses in Outlook or what other
approach can I take to make it practical to stop top-posting. Any
suggestions?
Don:
Tom Rymes wrote:
On Jan 15, 2011, at 9:29 AM, Don Kelly wrote:
That said, of course I want to follow this list's etiquette. I've
posted a couple times asking how I can interleave responses in
Outlook or what other approach can I take to make it practical to
stop top-posting. Any
I have it all configured and it should work, and it did briefly several
weeks ago, however now, it doesn't work at all and only plays the default
hold music.
If it is playing the default music, then the MOH function is working. What
do you get from moh show files in Asterisk?
--
Hi. I am using asterisk-1.8 and I am having problems getting
conferencing to work properly. I did modprobe on dahdi and did load =
chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing,
but meetme says
[Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available
On Sat, Jan 15, 2011 at 9:36 AM, James Miller paramedi...@gmail.com wrote:
Forgive me, but how do I do moh show files?
Basically what is occurring is:
If you enter a queue and are waiting to be answered, you will hear the
streaming MOH
If you call another extension on the system,
On Fri, Jan 14, 2011 at 6:31 PM, Tim Nelson tnel...@fudnet.net wrote:
You've been officially added to my kill file [1]. The lists are here to get
suggestions and assistance with various issues [2]. They are *NOT* your one
stop shop for everyone doing your homework [3][4][5][6][7][8][9]. You
Hello,
Our Asterisk runs with multiple remote sites (12 over an MPLS network),
everything works fine except for the last site we have juste installed.
When VOIP flows comes/goes from/to this site, there are sound quality
issues, persistent, 100% reproducible, on every call. This is not a
Hello,
Can you record audio at different locations on its route? Our experience
would suggest (of course) using intrusive or non-intrusive perceptual voice
quality evaluation at different parts of the network to localize the one
where it drops down.
Best regards,
Sevana Oy
Le 15/01/11 20:50, Sevana Oy a écrit :
Hello,
Can you record audio at different locations on its route? Our
experience would suggest (of course) using intrusive or non-intrusive
perceptual voice quality evaluation at different parts of the network
to localize the one where it drops down.
Yes
I am sure there are RTP packets losses somewhere, except RTP debug in
the asterisk CLI, how can i determine where the problem come from ?
If it is possible to make a network trace in a Wireshark compatible
format, Wireshark can parse all the SIP and RTP messaging and give you
lots of
Hello all,
In app_fax WATCHDOG_TOTAL_TIMEOUT is set to 30 minutes to kill a fax channel
regardless of whether or not it completes. In my case I have a fax that really
would take longer than 30 minutes to complete. Is there any way to disable the
WATCHDOG_TOTAL_TIMEOUT so that it runs to
From the command you suggested to enter:
Class: default
File: /var/lib/asterisk/moh//reno_project-system
File: /var/lib/asterisk/moh//macroform-robot_dity
File: /var/lib/asterisk/moh//manolo_camp-morning_coffee
File: /var/lib/asterisk/moh//macroform-cold_day
Hello List,
I've been trying to compile Asterisk with H.323 support and, after
correctly installing PTLib and H323plus (OpenH323), the Asterisk
configure script still doesn't detect the dependencies as installed.
I know they are correctly installed because after going into
On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote:
Hi. I am using asterisk-1.8 and I am having problems getting
conferencing to work properly. I did modprobe on dahdi and did load =
chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing,
but meetme says
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Jan 15, 2011 at 11:50:32AM -0500, cov...@ccs.covici.com wrote:
Hi. I am using asterisk-1.8 and I am having problems getting
conferencing to work properly. I did modprobe on dahdi and did load =
chan_dahdi.so in
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