5.4 isn't yet available on the AudioCodes website, only 5.2. The
distributer I bought the box from went out of business.
-Ryan
On Jan 6, 2009, at 9:15 PM, Tony Graziano wrote:
Just looking that you are running 5.2 firmware on the gateway, while
it shouldn;t matter, have you looked at 5.4?
Just looking that you are running 5.2 firmware on the gateway, while it
shouldn;t matter, have you looked at 5.4?
>>> Ryan Quigley 01/06/09 8:18 PM >>>
Hi Steve,
Thanks for getting back to me. Attached are my AC call logs with debug
set to 5. working_log.txt is when Automatic Dialing is set to
Hi Steve,
Thanks for getting back to me. Attached are my AC call logs with debug
set to 5. working_log.txt is when Automatic Dialing is set to # 100,
broken_log.txt, when it's set to # 300 (or any user extension for that
matter). My config file is also attached.
Tony, I went with 3.11.9 be
Aastra sent me an automated response to my support ticket. It just said.
Of course, it does not tell me what part it does not like ;-) I have been
unable to find any published specs for the Broadworks format. I just see other
phone vendors also list Broadworks format for their BLF list featur
icy hot wrote:
> Hello,
>
> I'm testing inband DTMF with a Linksys PAP2 ATA against different SIP
> servers and noticed it doesn't work with sipXecs.
>
> Is there a way to enable this?
>
> Thanks!
> Joe
>
Hello, after some time I managed to get this working with my PAP2, these
are the settin
No, we primarily work with Netapp and in particular one of these clients uses
the low end S550 (formerly Storevault). Nice units with NFS/Cifs and iSCSI.
The other client is using an internal RAID5 array with I think just 10K SAS
drives. We have found that SATA drives just don't prefrom well
Hello,
We are currently setting up a new SIPXECS server, with currently an Asterisk
system online. We would like to setup SIP Trunking between Asterisk and
SIPXECS. If anyone can offer any assistance with ideas, it would be greatly
appreciated.
Thanks All,
Jason
_
Thank you all for the quick reply, I'll move my server to a dedicated
hardware, I'll try to wait till release 4.0 is out, hope it comes out soon.
Thanks again,
Jhony
Matt White wrote:
I should add we have two installation of Sipx on ESXi with "good" success.
Here is what we have found. Here
We are using Hyper-V in two different deployments and have no issues. We did
have to install the "server extensions" or the voicemail quality was bad. One
setup has about 40 users and gets several hundred calls a day. In that
environment we have 4 other HEAVILLY trafficked VM's on the server as
Hmm, when you say "hunt group" are you referring to an ACD queue on
sipX? "Hunt Group" is AudioCodes terminology (and others, but I haven't
seen it used in sipX unless that's new in v3.11). You would use a hunt
group on the AudioCodes side for outbound calls over your PSTN
connections, but you woul
On Mon, 2009-01-05 at 19:14 -0500, Matt White wrote:
> The admin manual states the BLF/List feature is complaint with the
> Broadworks format R11 and up.
Does it suggest that there is any documentation regarding the Broadworks
format?
Dale
___
sipx-us
Ryan,
You can however change the unbound number that the gateway is ringing.
All calls however then will hit that number. Usually you can set this
on a line by line basis.
So, manually configure the gateway and set the inbound number to
whatever extension you'd like. I would do this normally fo
Matt,
Are you folks using OpenFiler for your SAN stuff?
Thanks,
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt White
Sent: Monday, January 05, 2009 7:59 PM
To: sipx-users@list.sipfoundry.org
Subject: R
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