Hi
i have buy a Audiocode Median 2000 VoIP Gateway and connect it on :
1 E1 30 channels
1 Lan Port
Anyone use this equipements with asterisk ? because i am search a
config sample for AudioCode and for Asterisk (i am new in VoIP).
I want that all calls arrives on the AudioCode are sent
Hi all,
to be able to transfer calls I have set call-limit to 2 for all of my peers.
Now how can I determine if a peer is in busy state using the first line if I
don't want to route a second call to it?
Thanks in advance,
Oliver
--
What I found is when I use sip.conf instead of realtime and
On Fri, Sep 17, 2010 at 11:22 AM, Nyamul Hassan mnhas...@usa.net wrote:
While this is too many eggs in one basket, but can be useful if you have
too many E(T)1s say equivalent to a STM1 (OC3) or more. In that case, it
would be too many boxes at 8ports / box.
Somewhere in the mailing list,
I had to make a few minor edit in fail2ban to get it to work. I had to
change the logger messages format for asterisk to:
[general]
dateformat=%F %T
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Thank you for your info Moises. For those who want to have a high density
system, can you provide what modifications to the Dahdi (or anything else)
do you make?
Regards
HASSAN
On Sat, Sep 18, 2010 at 19:39, Moises Silva moises.si...@gmail.com wrote:
On Fri, Sep 17, 2010 at 11:22 AM, Nyamul
On Sat, Sep 18, 2010 at 6:46 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Anyone use this equipements with asterisk ? because i am search a
config sample for AudioCode and for Asterisk (i am new in VoIP).
Why would you want too? Asterisk can do everything, and more, then
the Audiocodes.
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On Sat, Sep 18, 2010 at 6:54 AM, unsero...@aol.com wrote:
What am I doing wrong?
Enable a SIP debug and find out.
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC:
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins d...@cognation.net wrote:
Any thoughts on why the lack of traffic?
Cheers,
Dean
Not enough applications to play immature bathroom sounds.
Just a guess.
-M
--
_
-- Bandwidth
Sorry i don't really understand your message ;=) my english are bad.
I am search a sample of configuration of the audiocode.
2010/9/18 Paul Belanger paul.belan...@polybeacon.com:
On Sat, Sep 18, 2010 at 6:46 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Anyone use this equipements with
On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collins d...@cognation.net wrote:
Any thoughts on why the lack of traffic?
Cheers,
Dean
Not enough applications to play immature bathroom sounds.
You could well be right, but consider for a
On Sat, 18 Sep 2010 00:54:38 +0100, dotnetdub wrote
Hi All,
Is it possible to specify more than 1 localnet? I know this is an odd
question. I have a customer that has multiple sites linked by VPN.
Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24
We want to allow some
Those modifications are done via regular Sangoma installation with a special
option to the Setup script.
http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#zaptel_adjustable_chunk_sz
http://www.sangoma.com/assets/docs/misc/2009_10_09_How_to_Reduce_Asterisk_System_Loads.pdf
Moises Silva
On 09/19/2010 12:06 AM, Darren Nickerson wrote:
On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote:
On Fri, Sep 17, 2010 at 11:52 PM, Dean Collinsd...@cognation.net wrote:
Any thoughts on why the lack of traffic?
Cheers,
Dean
Not enough applications to play immature bathroom sounds.
You
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