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 c
Thank you for the help!
So with the Patton gateways, there were no problems with the volume on
either end of the PSTN calls? What else did you like about the Pattons?
Thanks... tommy
From: Tony Graziano [mailto:[EMAIL PROTECTED]
Sent: Friday, October 3
No. That is the function of the phone itself. Once the gateway sends the calls
into your network it is digital. Gain is a function of the analog gateway
before it is converted and streamed. There is no way that I know of to increase
the volume of this until it hits your end user device (phone),
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)
On Fri, Oct 31, 2008 at 10:53:36AM -0400, Dale Worley wrote:
> On Fri, 2008-10-31 at 09:24 -0500, Richard Kolkovich wrote:
> > I'm having issues with the MWI on my phones. After digging into it a
> > bit, it looks like the NOTIFY is not getting from sipStatus to the
> > phone. I've attached a tra
On Fri, 2008-10-31 at 09:24 -0500, Richard Kolkovich wrote:
> I'm having issues with the MWI on my phones. After digging into it a
> bit, it looks like the NOTIFY is not getting from sipStatus to the
> phone. I've attached a trace which shows a phone registering and
> subscribing. You can see th
I'm having issues with the MWI on my phones. After digging into it a bit, it
looks like the NOTIFY is not getting from sipStatus to the phone. I've
attached a trace which shows a phone registering and subscribing. You can see
the NOTIFY coming back through sipXproxy, and sipXproxy wanting sip
Hi,
Please send the level 5 log corresponding to the sipx log you have
attached here. We should be able to get an idea whats happening to the
INVITE. From the looks of it, the INVITE transaction is timing out
somewhere. I am not sure if this is happening between sipx and OpenSBC
or with Ope
I'm afraid your logs don't mean anything, really. You need to increase the
loglevel of sipXproxy to INFO, at least, an then create a trace file.
See:
http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_Messages_to_display
when you get the trace data, take
No, problem. Thanks for your efforts in helping me.
What do the logs errors, that i have attached, mean. When this error is
happening?
On Thu, 2008-10-30 at 08:35 -0400, Picher, Michael wrote:
> Sorry, I'm not familiar with how to implement OSBC as I haven't used it
> personally. Somebody was su
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 th
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