I think you're in luck :) Sipura has a new firmware out for the 841 now,
version 3.1.1a. (Quite the jump from 0.9.5) It includes -6db, 0db, +6db
options for input gain on the handset, headset and speakerphone now. Release
notes and downloads are available on their site.
-James
-Origina
Wow... sort of a horror story there. Not sure I have a lot of input other
then we have two 841's that we've been using for about a month now without
out problems. (I've actually been rather impressed.) One thing that comes
to mind though, have you tried updating the firmware to the latest? Or i
Well, I'm using intercom with a couple SPA-841. Currently using * CVS from
a couple weeks ago and a macro in extensions.conf. It should give you some
idea:
[macro-intercom]
; ${ARG1} - Extension
exten => s,1,SetCDRUserField(intercom)
exten => s,2,SIPaddheader(Call-Info: \;answer-after=0)
e
Excellent, thanks for the info. That hiss is interesting; maybe all the
extra wire makes it tougher for the SPA to drive. In any event it sounds
like it will do what I want. Thanks.. :)
-James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent:
Hello all..
I have what I think is a rather simple question here. We're using * for
home and business lines with great success so far. (inbound, outbound,
several hard phones, voicemail, IVR, etc)
My next project is to get some of my traditional POTS phones at home on the
system. I understa
I've been running a pair of Sipura 841's with my business partner. So far
I've been very impressed for the money. Having multiple lines (with
multiple registrations) has also been really nice as we both have our "work"
line and "home" line off the phones. (And you can update them for 2 more
line
CTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] SJphone on PDA registering with Asterisk???
James Pooton wrote:
>I'm all so using SJphone on my x50v, works surprisingly well :).
>
>Is voip.elmit.com also in the 192.168.1.X NAT space that
That's surprising; I thought they were one of the larger outfits. I have
tried quite a few for outbound lately and the only one that has reliably
passed Caller ID (using our 406 area code) is simpletelecom.com. They are
also the only ones I've tried that respond to a support ticket in a
reasonabl
I'm all so using SJphone on my x50v, works surprisingly well :).
Is voip.elmit.com also in the 192.168.1.X NAT space that your PDA is in?
Do you have "host=dynamic" in your * sip.conf entry for 701 ? Actually might
help to toss your sip.conf entry out here for 701 without the secret.
Do you see
Well, considering I'm on topic, I shouldn't get flamed to badly for this. I
have a bunch of these working well in my home experiments:
http://www.laptops4me.com/product_info.php/products_id/1444
And yes that price is correct and they do arrive. :)
Not everyone can justify buying the "supported"
I think MOH works without a timing source, but can get end up sounding
poor without one. (stuttering, etc) At least that has been my experience.
Without a card, ztdummy becomes an option for timing if the box has the
correct USB controller to make use of:
http://www.voip-info.org/wiki-Asterisk+
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