Displaying 20 results from an estimated 3000 matches similar to: "Boosting Shared Internet Bandwidth for Asterisk"
2005 May 20
4
Boosting Internet Bandwidth for VOIP
There was errors when I tried to start the script
recommended by Andrew to boost bandwidth for voip
http://www.mixdown.ca/~andrew/dump/rc.tc.
This is the output I get :
./rc.tc start
RTNETLINK answers: File exists
RTNETLINK answers: File exists
RTNETLINK answers: File exists
RTNETLINK answers: File exists
RTNETLINK answers: File exists
RTNETLINK answers: File exists
RTNETLINK answers: File
2005 Jul 09
2
Modifying astcc
Hi:
Astcc is working fine, except for one thing. It
doesn't give the called party enough time to answer
the phone. If nobody picks up in two rings, astcc
reports back no answer and hangs-up. The only instant
NOANSWER "value" was mentioned in astcc.agi script is:
elsif ($res eq "NOANSWER") {
$res =
&mystreamfile("astcc-noanswer");
2005 Jun 30
5
wi-fi phone advice
Hi:
I want to connect a wi-fi phone to my Asterisk box
through a wi-fi AP so I can make voip calls.
please send me your recomendation about what wi-fi
phone I should be looking for. Anybody tried the
HOP1502 Wi-Fi IP phone. Its listed price $39.
Regards;
Chawki
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2007 May 07
4
iax to iax Reject Connection
Hi:
It's the first time I have this problem.
No matter how I configure my two IAX machines the
connection is rejected.
"chan_iax2.c:5550 socket_read: Call rejected by ****:
No authority found"
iax server A:
[saad_out]
type=peer
host=hostip
username=username
secret=secret
disallow=all
allow=gsm
iax server B:
[guest]
type=user
username=username
secret=secret
context=tele
2005 Jun 29
3
hidecallerid on analog line
Is there a way to hide the callerid on analog line on
outgoing calls. Any ideas whether it could be done
through configuration, a script or hardware.
Thanks;
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2005 Feb 08
4
In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Is the channel physically being hung up before the * tone is heard?
Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't
support Kewlstart-style disconnect notification.
The sequence I hear on the extension, when I plug in an analog phone, is the
click of the
2005 May 19
5
MusicOnHold probelms
This is my second attempt trying to get help and I am
hoping someone can. When the musiconhold extension is
matched, Asterisk attempts to execute musiconhold and
stops right away, this is what I gets:
Executing MusicOnHold("OSS/dsp", "") in new stack
-- Started music on hold, class 'default', on
OSS/dsp
-- Stopped music on hold on OSS/dsp
Is there a file that
2005 Aug 18
2
Searching For a Voip Provider
Hi:
Please advice me of a voip provider with reasonable
reseller program. I was using voipjet and it has a lot
of problems.
Did anyone experienced asteriskout.com service? They
have good prices.
Regards;
Chawki Hammoud
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2006 Mar 21
12
Fw: anybody has SIP realtime working ?
Hello,
I am just asking this because I am note sure if the problem
is on my side or not, I saw some comments on SIP realtime
today so I was wondering, has anybody has SIP realtime working
with a softfone ?
If yes, please confirm, that would give me a light.
My previous message to the list is below.
Thanks.
Frederic
----- Original Message -----
From: Frederic Jean
To:
2003 Dec 09
1
dialling peer problems
I'm trying to use Jeremy's suggestion of dialling using just the peer name
instead of user:pass@peer but I'm running into some really funky issues.
It does the same thing with both VoicePulse and another * server I have.
[voicepulse]
type=peer
host=gw5.voicepulse.com
trunk=yes
user=USERNAME
pass=PASSWORD
and in my dialplan:
Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r)
The log shows
2005 Jul 12
3
Help Configuring TDM04B
Hi:
I had an fxo card from Digitnetworks and it was
working fine on my Asterisk box. I then replaced it
with TDM04B. I changed the zaptel and zapata to
include the four channels. When I run ztcfg, I get
configuration errors:
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04:
2005 Mar 19
3
CallingCard Application
I appreciate any recomendation of a simple CallingCard
Application and resources of users manual.
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2004 Jan 16
2
Hardware for Asterisk
At 1/16/04 7:25 AM, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com>
wrote:
>That's pure bullshit -- I use software RAID *specifically* because I value
>my data. I don't want to buy two hardaware RAID controllers to have one
>sit on the shelf just in case the first dies... and if the second dies
>you're SOL because they've lasted long enough that
2005 Feb 04
5
IAX2 register Refresh
Hi all
I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file
thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec.
I need to get this down to 15 sec (nat /pat firewall issue)
any ideas?
thanks
Liaan
2005 Feb 23
3
Help With Adit 600 Configuration
Sorry to have had to post this, But I need urgent help with configuring one
adit 600 I picked up from e-bay.
Issues. I cannot access the console port, I am using HyperTerminal with
settings VT100, 9600, 8-N-1
I also do not have any user-manual so I am kind of stuck. Any help in
getting me started would be really appreciated. Any default settings like
Ethernet port address, that can help me
2005 May 27
3
G729 vs. gsm
I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm, G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?
I am interested in G729 because the internet in my
country is very expensive and I want to save every bit
possible. I want to use G729 because it takes less
bandwidth for
2005 Jul 16
3
Asterisk Interface with mobile phone
Hi:
I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can
2005 Feb 23
6
List tips for new subscribers
*spews coffee over keyboard*
- FUNNIEST - THREAD - EVER -
Also one of the most insightful.
Teddy, your gmail invite is on the way.
2003 Nov 25
1
Picking a channel (FXO port or SIP) for outb ound calls
> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Sent: Tuesday, 25 November, 2003 08:56
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for
outbound calls
>
>
> > Yep, we use it for international calling. Works great:
> > exten =>
2003 Oct 01
7
eBay Sip Phone Scam.
Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware.
http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch=
--
Costas Menico
Meezon Software Corp
201-224-8111
costas@meezon.com
--