, but we use it on 1.2). It fixes the volume
issues. It does require sox to work though.
http://bugs.digium.com/view.php?id=6237
On Mon, 15 May 2006, Scott Bussinger wrote:
There's a been a long standing issue with voicemail volume levels for
files
saved in WAV49 format as compared
There's a been a long standing issue with voicemail volume levels for files
saved in WAV49 format as compared to WAV format. WAV49 is much smaller in
emails and that's great, but it's also less than half the volume level than
the exact same voicemail saved in WAV format. I've seen this
Help! I'm having an odd problem that I'm not seeing in any of the list
archives and thought I'd ask and see if anyone can help.
I've got Asterisk behind a NAT and an SPA-841 SIP phone behind a different
NAT. Everything works fine (incoming calls ring, outgoing calls work, audio
in both
I've got an SPA-841 SIP hardphone connecting to my asterisk server across
the internet through a couple of NAT routers. Everything works great (I can
initiate calls, receive calls, hear audio both ways, etc.) except for one
thing. When I hang up the phone, the connection in asterisk doesn't
Occasionally, one of our SPA-841's will spontaneously start up with
Welcome to Comedian Mail! on the speaker phone. No one is near the
phone or touching it. It is as if the Invisible Man walked up and pushed
the dial voicemail button. I have obviously been unable to reproduce
this problem,
Hi, we have put together a small application for Windows to allow you to
check IAX network statistics.
The application seems to work fine, but could you point to any information
that helps interpret the information that it displays? I've seen the little
bit that the new jitterbuffer
Could someone please look at this information and help me decipher what it
should actually mean to me? I've found a bit of information here and there
but I'd like to know what I'm supposed to be reading into this information:
pbx*CLI iax2 show channels
Channel Peer
With that said are there any forums that are well used
or that might even convert this email in a true
forum that is searchable and that doesn't require me
downloading every email.
Personally, I much prefer the mailing list approach to the online forums. In
my opinion mailing lists aren't
I've seen lots of comparisons between the various codecs with respect to
bandwidth requirements, but are there any comparisons with respect to
quality?
I'm currently using ulaw internally and gsm to connect to my ITSP, but
should I be using different codecs to get better sound? Could someone
After 20 posts, in 2005 the ideal setup for a new installtion
of a 50 user asterisk is:
Option1: IAX2 with softphone firefly
Option2: SIP with softphone
Option3: IAX2 with hardphones (which brand?)
Option4: SIP with hardphones.
As the other poster said, I doubt you'll find a consensus as
Forget older years but in 2005 do hard phones really add any
value over softphones.
The call center agents already have p4 2.4ghz with 512 MB ram
Win2K why not just get them a nice USB headset with a
softphone IAX client,
We just tried to go entirely with softphones in our office gave up
Did you consider vonage?
I played with Packet8 as a reality check before investing the time money
into the Asterisk solution. I certainly wouldn't use Packet8 or Vonage or
any other service instead of an Asterisk solution. It's easy to get
termination delivered by IAX now and is both cheaper
I'm testing a softphone-only setup (SJPhone with Plantronics
80 Headsets plugged into Soundcard) with around 40 users for
that are linked over LAN in an organization of around 300
people and never had any of the problems you described (the
test is going for over a month now).
I'm glad it
We're setting up some Sipura 841 phones and they're working pretty well, but
the microphone volume on the headset (not the handset) is too loud with our
Plantronics headsets. Is there some way to turn down the amplification on
the headset mic?
The microphones are picking up the sound of someone
I am looking for a good provider for IAX2/800 termination. I
I've been using TelIAX for a couple of months now for long distance and 800
service and they've had good quality and good support.
Be seeing you.
___
Asterisk-Users mailing list
Have you seen the user guide?
http://www.sipura.com/Documents/SPA841UserGuide.pdf
Yes, and it's actually not bad, though a bit wordy. There's almost not
information on actually programming the device though. I've been trying to
understand how all the configuration settings about
If you contact Sipura and prove to them you are a service
provider, they'll give you access to an area that contains
the manual and profile compiler for the 841.
I dropped them a note and they gave me all the details on how to prove I'm a
service provider, but I'm really just an end user and
Has anyone found an administration manual for the Sipura SPA-841 phones? I
found a quick start guide and a user manual at the website (good thing
because there was _nothing_ in the box), but I haven't found a manual to
explain the more complex features in the web setup pages.
Thanks!
I'd like some suggestions on the proper way to format a line in my
extensions.conf file.
In our office, most calls are sent to a group of extensions and whoever is
available answers the call. What I'd like to do is prefix the caller ID with
PRV: for those calls that are sent directly to a
Are there any settings in the VoicemailMain handler that would cause it to
play the voicemails in reverse order (i.e most recent voicemails first, then
second most recent, etc.)?
The reason is that we generally use email to get our voicemails, but
occasionally it would be nice to get them over
We're just getting our Asterisk server setup with TelIAX and it's working
fine. I did have to play with settings a bit. Basically I just used the
setting they recommended instead of the generic settings I started with.
Here are the significant settings we're using in IAX.CONF:
[general]
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