Displaying 20 results from an estimated 3000 matches similar to: "Best QoS for Linux"
2009 Sep 23
3
SIP/WiFi handsets?
Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.
Thanks!
-Ken
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2009 Oct 05
6
Receptionist GUI?
Hey, all. Just wondering if there's a GUI out there -- preferably OSS,
but I'll take what-have-you -- that
a) can run on an Ubuntu/Debian box, and
b) allows a receptionist to see what calls are in-process, and forward
calls from their phone to somewhere else.
Thanks!
-Ken
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2009 Oct 29
3
Unable to set TOS to 184?
I don't understand this message:
[2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184
2009 Sep 02
4
More Echo
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
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<body bgcolor="#ffffff" text="#000000">
<span class="postbody">Greetings,<br>
I am running Asterisk 1.4.25 with Dahdi Complete
2.2.0, on a Digium TE121B PCI express card with a </span>VPMADT032<span
2011 Feb 03
1
PLEASE REMOVE ME FROM YOUR LIST !
--
*Larry "Yako" Junior*
*:On Air Analyst/personality
:Online Radio Broadcaster "YakoRadio"
**:V.O. Artist*
*:Image/Com Producer*
*:Audio Engineer
:Media Talent**
**facebook.com/yakoradionetworks <http://www.facebook.com/yakoradionetworks>
twitter.com/yakoradio
myspace.com/yakoenturadio
yakoradio.com
youtube.com/yakoenturadio
soundcloud.com/yakoradio
2007 Nov 08
3
Asterisk as a SIP to XMPP Jingle voice gateway
Hello,
I'm looking for a SIP to XMPP Jingle voice gateway.
I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client.
Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa?
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
2007 Dec 10
2
asterisk linkedin group
asterisk linkedin group
I have created an asterisk linkedin group for anyone interested.
http://www.linkedin.com/e/gis/45252/66270A773F53
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Board member of
Connectech Greater Detroit
www.connectech.org
________________________________
Please visit us on the web at www.hirotecamerica.com
HIROTEC AMERICA Ph.
2004 Sep 21
2
SIP termination in Brazil
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination solution.
But who as it up and running ?
Best regards,
Han
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2007 Dec 11
1
Appending two voice files
Does anyone know how I can append to different user recorded voice files within a dial plan? For example Asterisk ask caller a question and records the answer, then ask another question record the answer to the end of the first answer - so when it's played back, all the answers are in one playback.
TIA
Bart
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2008 Jan 12
2
Asterisk RFC2833 to SIP INFO DTMF conversion erros.
Hi,
I am using asterisk 1.4.17 which is connected to a SIP trunk supporting
rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have
set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for
SIP clients I have set dtmfmode=info. So when I make a call to a cell number
using the sip trunk and then press digits I can see the 2833 dtmf events
coming to asterisk
2008 Apr 07
0
[LLVMdev] Newbie
On Mon, Apr 7, 2008 at 5:56 AM, Vania Joloboff <vania.joloboff at inria.fr> wrote:
> We do dynamic binary translation. We are in a similar situation to qemu
> except we are SystemC / TLM compliant for hardware and bus models. Our
> current technology is somewhat like qemu, we translate the binary into
> "semantic ops", which are pre-compiled at build time, like qemu.
2006 Jan 28
2
Trunk is not released
Hi!
I have this little problem here and i really don't know how to solve it.
This is the scenario:
I've setup a IVR, using my mobile phone I call my asterisk server and after pressing "1" the call is directed to my softphone at extension 100. The phone at extention 100 will ring until a certain time, and my mobile phone will cut off due to no one picking up my call. However,
2005 Aug 05
1
Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start:
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15'
Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1
== Unregistered
2018 Feb 23
4
Stale mailbox lock file
Hi all,
I would like to understand the meaning of "Stale mailbox lock file detected"
In index-storage.c (https://mis.dopa.go.th/atmail/server_source/dovecot/src/lib-storage/index/index-storage.c) there is this notice "Stale mailbox lock file detected" ...
I have a java application which connects via javamail to a dovecot server and I have some anomalies when that message
2008 Jan 31
2
Analog Adapters ?
I have a friend with a small business running a small SIP based phone
system. He was looking into providing some SIP phones for a couple of
remote teleworkers, but as he started to look around and ask me questions he
ran across analog adapters which made him curious.
He proceeded to ask me if there was an analog adapter that provided the
following functionality in which my reply was simply,
2006 Mar 31
1
Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't
get it to work.
-David
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J.
M.
Sent: Friday, March 31, 2006 1:44 AM
To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: BRI
2008 Apr 07
2
[LLVMdev] Newbie
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Thanks to all those who responded to our email.<br>
<br>
Tilmann Scheller wrote:
<blockquote
2007 Oct 19
2
Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday
As usual, we'll be jawing about any and all asterisk-related subjects
with the usual gang and any new people are always welcome, regardless
of your level of expertise. You can even come and ask questions, it's
guaranteed to be a more pleasant experience than it will be on IRC ;)
http://VoipUsersConference.org/topics.php
IRC; Freenode.net #voip-users-conference
2005 Sep 24
1
unable to use misdn group dial
I have set up a * box with two hfc ISDN pci cards
using mISDN both in TE mode with PmP mode.
(using $MODPROBE hfcpci protocol=0x2,0x2
layermask=0xf,0xf)
I have no problem dialing out by explicitly naming the
mISDN port, ex: Dial(mISND/1/${EXTEN},60)
or Dial(mISDN/2/${EXTEN},60)
But it does NOT work when specifying the mISDN group:
exten =>
_(outpattern),1,Dial(mISDN/g:TEmode/${EXTEN},60)
2010 Apr 20
1
Portech MV-374 does not register
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<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
has anyone experience with the Portech MV-374 GSM-gateway ?<br>
<br>
I'm