1. They give direct support.
2. Their manuals for the Smartnode line are well thought out and very
heavily documented (4-500 pages I think).
3. The devices are highly configurable. You can do things with them like
schedules, and user matching on outbound calls to pick a "certain" POTS
lines for a certain user (if that matters anymore).
I have never had one lockup, they are very good at sensing disconnects
(faster than AudioCodes) and I've never had an audio problem with their
gear.
Since the config can be downloaded to a text file, you can also
"cut/paste" a config into a unit via serial/telnet. The interface
commands are robust and remind me of a Cisco/Juniper OS.
Did I mention they give direct support?
>>> "IT Services" <[EMAIL PROTECTED]> 10/31/08 7:22 PM >>>
v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);}
.shape {behavior:url(#default#VML);}
<![endif]--><o:smarttagtype name="PersonName"
namespaceuri="urn:schemas-microsoft-com:office:smarttags">
st1\:*{behavior:url(#default#ieooui) }
<![endif]-->Thank you for the help!<o:p></o:p>
<o:p> </o:p>
So with the Patton gateways, there were noproblems with the volume on
either end of the PSTN calls? What else did youlike about the
Pattons?<o:p></o:p>
<o:p> </o:p>
Thanks… tommy<o:p></o:p>
<o:p> </o:p>
Tony Graziano[mailto:[EMAIL PROTECTED]
Sent: Friday, October 31, 20083:38 PM
sipx-users@list.sipfoundry.org</st1:personname>
Subject: Re: [sipx-users] LowVolume on PSTN calls<o:p></o:p>
<o:p> </o:p>
No. That is the function of the phone itself. Once thegateway sends the
calls into your network it is digital. Gain is a function ofthe analog
gateway before it is converted and streamed. There is no way that Iknow
of to increase the volume of this until it hits your end user
device(phone), which is why people turn up the volume on their phone to
hear better.
I saw lots of problems with Audiocodes, volume, disconnect, lockups. I
turnedmine off and put in a Patton gateway and haven't had to touch it
since.
>>> "<st1:personname w:st="on">IT
Services</st1:personname>"<[EMAIL PROTECTED]> 10/31/08 6:24 PM
>>>
Hi there:
An update:
1. Volume on the receiver end of PSTN calls is fine.
2. Volume on the IP phones (the caller-side) of PSTN calls is very low.
3. Volume is fine for IP phone to IP phone calls.
4. Increasing the Gain did not help on the IP side.
Is there a setting on the gateway (audiocodes) that can increase only
the volume to the IP phone?
Thanks for your help!
tommy
-----Original Message-----
From: Picher, Michael [mailto:[EMAIL PROTECTED]
Sent: Friday, October 31, 2008 1:44 AM
To: <st1:personname w:st="on">IT Services</st1:personname>;
<st1:personname
w:st="on">sipx-users@list.sipfoundry.org</st1:personname>
Subject: RE: [sipx-users] Low Volume on PSTN calls
Look for a setting on the physical line in the gateway called gain.
Increase the gain in small steps. The more you amplify the analog line
the more you'll hear. There's good and bad on the analog line (static /
echo). Tweak it up slowly until it is acceptable. There will be a
happy medium in there somewhere.
Mike
> -----Original Message-----
> From: [EMAIL PROTECTED] [mailto:sipx-users-
> [EMAIL PROTECTED] On Behalf Of <st1:personname w:st="on">IT
Services</st1:personname>
> Sent: Thursday, October 30, 2008 6:47 PM
> To: <st1:personname
w:st="on">sipx-users@list.sipfoundry.org</st1:personname>
> Subject: [sipx-users] Low Volume on PSTN calls
>
> Hi all:
>
> I'm using 3.10.2 with audiocodes gateways. Internal calls are fine in
> terms of volume. But incoming and outgoing calls via PSTN have very
low
> volume. Is there anything I can do to increase the volume on these
> calls
> yet maintain the volume on internal calls?
>
> Thanks!
>
>
>
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