Whenever I see updates in Asterisk-CVS, is that updates to
Head, or Stable?
cvs checkout zaptel libpri asterisk
vs
cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons
asterisk-sounds
Im just wondering, if some of those changes or bug fixes
interest me, which one of those
6:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject)
Deon Rodden wrote:
When do you think the last stable CVS will be available before lots of
stuff begins to change? I want to find the best possible Asterisk and
stick
Are there no permissions issues that will ever come up by running Asterisk
as a non-root user?
My Asterisk server is a dedicated/closed system, only I have access to ssh
into it. It's also behind an external firewall that only allows certain udp
ports through from the world. And ssh from my
Laugh. I use a bare-bones install of QMail on my main asterisk server. It of
course emulates sendmail and the like.
But on my remote Asterisk server, I use ssmtp, it came as a prerequisite to
Asterisk. When I emerged asterisk, ssmtp came with it. Works great.
Configured it to use my main Asterisk
for everybody, and
less latency, hence less jitter. I'm excited to see these developments,
as I believe it will make VoIP more reliable over these types of
networks. At the moment, there are simply too many variables to trust
it.
Brian
On Mon, 18 Oct 2004 17:36:14 -0400, Deon Rodden [EMAIL
I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.
But just 5 minutes ago I moved a user over to g729a. Changed the
SIP000.cnf file for the Cisco phone, but forgot to change the
My firewall script has something to the effect of:
# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045
When do you think the last stable CVS will be available before lots of
stuff begins to change? I want to find the best possible Asterisk and stick
with it, for some time, maybe until 2.0; If I get CVS right now, what if
tomorrow or the day after he comes out with a better CVS.
I wonder if Mark
Ummm... It used to be a while back there were 2 different CVS servers or
directories or something, 1 for Head and 1 for Stable. But some time ago,
only one version of CVS showed. I assumed they temporarily merged the 2,
every new release was just a new stable release.
I'm now on the download
If I had the choice of buying Sipura SPA-2000's or Linksys's PAP2-NA's, I
should go with LinkSys?
I thought linksys didn't support it working with any other provider other
than vonage? Changing it would void their warranty or tech support or
whatever.
You got the PAP2-NA's for ~$50? That's a
Have you tried voipsupply.com? Or even EBay? A lot of ppl sell them New or
Virtually New on EBay, with Buy It Now for instant purchasing. I find that
voipsupply.com is cheaper than the average EBay Buy it Now price. I also
find that voipsupply.com does business on EBay as b2tech or something like
Can you give me more info on general issues across the net ?
Yeah, VoicePulse seems to be having issues, it's usual though. I wish they
weren't the only place I knew to get flat rate incoming DID's Nationally in
the U.S from.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Try sip show channels or iax2 show channels instead of just show channels
Tells you what Codec is in use for the active channels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Henderson
Sent: Monday, October 18, 2004 1:58 PM
To: Asterisk Users
I wouldn't recommend a reload. More than once I've issued a reload to
Asterisk and it just sat there, never actually reloaded. Went into limbo,
wouldn't accept any future commands either. Had to kill -9 the process and
load it again. Plus certain changes you make, like certain changes to the
It's not the bandwidth. I have Sprint and am switching to Verizon with a
week. When I go online through my Sprint phone, I get 250+ms response times.
That can not be VOIP friendly. I have clocked downloads at up to 130 kbits
per second, so the speed is ok, but the ping response times are bad.
. At the moment, there are simply too many
variables to trust it.
Brian
On Mon, 18 Oct 2004 17:36:14 -0400, Deon Rodden [EMAIL PROTECTED]
wrote:
It's not the bandwidth. I have Sprint and am switching to Verizon with a
week. When I go online through my Sprint phone, I get 250+ms response
times
Either way. I've bought several devices from b2tech on ebay as well as
several devices direct from voipsupply.com so it wouldn't sway me much if
they were plugging their own company on this list, I already trust them.
Never bought Polycom from them though, although I plan to in the near
future.
version, should be out next week
Deon Rodden wrote:
I put FireFly on my mom's computer, but ran into a problem. She went
home and was able to place calls from it (using her headset and such).
But, she could not receive calls. I figured out the problem was with the
registration, firefly
I use FireFly w/ SIP all day long and it works great, except for the SIP
registration interval which I was just told will be fixed in next weeks
version.
Are you using GSM or g711u?
[remote-laptop]
context=remoteusers
type=friend
username=remote-laptop
secret=hiddenfromlist
qualify=yes
FireFly is awesome, it's not giving quality issues like X-Lite is. FireFly's
only problem was it wasn't registering with the server often enough, making
that NAT box forget the connection and not allow incoming streams.
Adam Hart said they would add it as an adjustable feature to the next
I put FireFly on my moms computer, but ran into a
problem. She went home and was able to place calls from it (using her headset
and such). But, she could not receive calls. I figured out the problem was with
the registration, firefly doesnt re-register often enough, so the
connection gets
better when
we were trying it out.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Thursday, October 14, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] FireFly SIP Registration Interval
I put FireFly on my mom's computer, but ran into a problem
If I relay through my Cisco 7940/7960, does it do QOS, even with a dumb
switch?
I know you can set quality/qos but only if you have a layer2/layer3
switch that supports the tagging. A simple little linksys 5 port switch
wouldn't know about QOS, it'd give everybody equal priority. If a
For almost 6 months now I've upgraded Asterisk every couple of weeks or
so and I've never had this problem. When I'm at the asterisk console
(asterisk -r) it shows me live status. Who called who, what it's playing
and when, etc. It logs to the screen. When I type reload, it says added
so and
I usually use safe_asterisk or /etc/init.d/asterisk start the
defaults have always worked for me.
Nick Barnes wrote:
Hi,
But recently I upgraded and now when I do reload all I see is Sep 14
12:55:25 NOTICE[393230]: indications.c:397
ast_unregister_indication_country: Removed default
We use a nice Polycom conference phone and plugged it into the Sipura
and it works crystal clear. Was cheaper than Polycom's conference phone
w/ built in VOIP capabilities.
Joe Dennick wrote:
If it were me; I'd opt for one of the Polycom Conference phones (they
are just regular analog phones),
=1ssPageName=WD1V
So for just over $200 (have to add shipping) you can have a nice
conference phone. A couple of our customers use this solution.
hank smith wrote:
what phone did you purchase and how much
- Original Message - From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk
Last night I updated to a custom 2.4.27 kernel, I also upgraded
asterisk. This morning I discovered Asterisk is no longer playing sounds
to users. ie when they go to the voicemail, asterisk says it's playing
vm-login but the user never hears it. It's not a firewall issue or
anything like this,
I believe this has something to do with the converter. With my
Sipura-2000 if I hit flash, it puts the person on hold and I get a new
dialtone to place a call. From there I can call another number, and if I
hit flash again, it 3 way calls them. If I hang up, it leaves the other
two people
Do you know where it got the 10.138.3.2 IP from? Is it configured
anywhere on the server? Do you have
externip defined in that config file?
Evert Meulie wrote:
Hi everyone!
situation:
Asterisk-server A: 192.168.11.6
Asterisk-server B: 192.168.2.44
server B contains a register = username:[EMAIL
What I did with Gentoo, which may work similarly with Debian, was first
I emerge'd the Asterisk 0.9.0 version that Gentoo offered. The benefits
of this was it downloaded and installed all the dependencies and then
installed Asterisk. I then used CVS to download the latest (with all the
bug
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine,
Here's my iax.conf and extensions.conf (I have not yet made the recent
changes they just emailed about a day ago, this is twice in a two month
period, jeesh) I have tested inbound and outbound dtmf. I use the g.711
codec and use inband.
iax.conf
We discussed this earlier and I believe the general consensus was that
it's personal choice. I've personally used Asterisk on Redhat 9.0,
Fedora Core 1 and Gentoo 2004.2
Each has required some minor securing and cleaning up, but Redhat/Fedora
tended to need more babying as far as securing
You can actually hear the hard drive noise when calling out or receiving
a call? A clicking sound, or like an electrical noise?
I doubt this is being done through the motherboard, how close is the
card to the power supply and/or the power wires going into the hard
drives? Are they less (or
What does your host= line show in the iax.conf for fwd? I found that
iax.conf hates it when you use host=x.x.x.x so instead I had to use
host=dynamic and defaultip=x.x.x.x or something like that. It's very
finicky.
Storm D. J. Petersen wrote:
Hi,
I cannot seem to accept incoming calls from
them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of extensions
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I put the outbound rules so that the phones
can call
When I initially signed up with Packet8 and they sent their converter, I
used a X100P card in my Asterisk server so that it could send and
receive calls through Packet8, I suspect the same trick would work for
Vonage.
The benefit is you can then have several phones in the house, or one at
Have you considered relocating the hard drive (or Asterisk configs) and
the T100P card to a temporary machine? Even a lower class machine, just
to eliminate the SuperMicro as the possibility?
I'm interested in your research as we will be deploying some low end
$800 1U (very short) SuperMicro
Lol. Known issue, I spent an hour working on that problem. The phone's
current firmware is too hold and does not support longer filenames like
that. You have to increment the firmware versions, 2 or 3 firmware
upgrades and you'll be ready to use the latest and greatest.
Try upgrading to
,
When you say I've tested up to 6 inbound calls at the same time
with Broadvoice, is this with 6 $19.95 DID numbers that you have
assigned to *?
thanks
Doug
Deon Rodden wrote:
When I initially signed up with Packet8 and they sent their
converter, I used a X100P card in my Asterisk server so
Whenever I see the Maximum retries message it usually indicated a
communication problem, like one way traffic. Last time I got it, I
traced it to a bad firewall rule, dropped the firewall and it worked,
the time before that when I received it, it was due to a routing error,
the server could
I've been working with Asterisk for about 2 months now and am doing
well. However I decided to switch platforms from Fedora Core 1, that my
predacessor was using, to Gentoo, for obvious reasons. It just seems
faster and less bloated everything I need, nothing I don't.
Anyways, I've read what
I have several Cisco 7940's laying around, how do I piple the
speakerphone through external speakers? I understand the amplifier part,
but how do you get RCA/2.5mm outputs from the Cisco?
For now, we just configured a line 2 on all our phones with auto
answer and, using the trick found in the
Somewhat. You got the remote site right.
I have several Voice T1's at my main location, and it runs into a Cisco
router which converts it to SIp and sends it to Asterisk. I would like to
be able to push certain incoming phone numbers across IAX to another
Asterisk server at a remote site.
I setup extension 105 on my Asterisk server to
Dial(SIP/[EMAIL PROTECTED]) and then defined [sipserver_b] in the
sip.conf
So then I setup extension on sipserver_b's extensions.conf file to answer
with the auto attendant, and it simply plays a message asking what number I
want to dial. It then
We put a VWIC and a DSP in a Cisco 1720. The purpose will be for a customer
to use a T1 Crossover cable to connect the 1720 into their existing PBX
system. It'll be a Virtual T1 PRI type of thing. The Cisco 1720 will make
the conversion to SIP and send it to our Asterisk server. As far as his PBX
We have at least 3 customers with Cisco ATA186's plugged into a fax machine.
They can send and receive faxes perfectly. The config in Asterisk is no
different than any other ATA186. G711Ulaw is the codec we use.
Supposedly the Sipura SP-2000 we're now using can do faxes as well. Haven't
tested
er1,30,r)
exten =
1235551214,1,Dial(SIP/customer1,30,r)
Maybe I should put a "defaultip=x.x.x.x" in the sip.conf
section as well? Will this work?
Thanks,
Deon
550 Fairway DriveSuite 210Deerfield Beach, FL
33441Online: www.webunited.net
Deon
Is that Video phone really only $200? And it's SIP compatible with any
Asterisk server? Packet 8's was interesting but I never wanted packet 8
service, want to use my own server. Looks like the phone is only $200?
- Original Message -
From: Jeremy Jones [EMAIL PROTECTED]
To: [EMAIL
Is it only the guest speakers your interested in listening to? Or is it
specific vendos as they show off their products or enhancements to the
crowd? With so much noise and people talking at Astricon, how do people in
the conference expect to hear any one conversation, or one topic?
However, I
All you need is enough bandwidth to upstream one good signal, the users on
this list willing to donate bandwidth and equipment can then redistribute it
to the others. I don't think Dial up is a very good idea, but having access
to a shared T1 or even wireless internet access may be a possibility.
I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and
OpenNA Linux 1.0 and all give me an Error 1 after typing make but with
no real reason given. Just a few standard/non-critical warning messages, and
then suddenly Error 1
Anybody successfully compile Rate Engine? The least cost
It may sound bad, but I use Fedora Core 1. However, I installed using
reiserfs (my preferred filesystem) and I installed all the updates and had
to custom compile a new kernel (as the stock one that comes with Fedora is
too screwy, and the sources aren't done right and certain programs wouldn't
se the
BT100's for our cheap customer phones ($60, not bad), as they get beat up a lot,
they're far cheaper than putting 7960's in our customer waiting
area.
Regards,
Deon
550 Fairway DriveSuite 210Deerfield Beach, FL
33441Online: www.webunited.net
Deon R
Nevermind. Found http://www.voip-info.org/tiki-print.php?page=Budgetone
Sorry to disturb you.
550 Fairway DriveSuite 210Deerfield Beach, FL
33441Online: www.webunited.net
Deon Rodden Toll Free:
1-877-538-5969 x 208Phone: 954-418-8884 x 208Fax
I see the issue. Mine says Request Sent and just hangs there.
- Original Message -
From: Wolfgang S. Rupprecht [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 26, 2004 1:35 AM
Subject: Re: [Asterisk-Users] Broadvoice problems again
[EMAIL PROTECTED] (Rich Adamson) writes:
I've been using them for just over a month, and this is the first outtage
I've noticed with them. While I'm not happy about how long this outtage has
gone on, I'm willing to give them another chance.
I'm curious though, I'm paying $19.95 for unlimited local and long distance.
Why would you choose
Greetings,
C:\ping 147.135.8.129
Pinging 147.135.8.129 with 32 bytes of data:
Request timed out.
Request timed out.
Request timed out.
Request timed out.
Ping statistics for 147.135.8.129:
Packets: Sent = 4, Received = 0, Lost = 4 (100% loss),
Approximate round trip times in milli-seconds:
namesonly (ATT, Sprint, etc.)?
550 Fairway DriveSuite 210Deerfield Beach, FL
33441Online: www.webunited.net
Deon Rodden Toll Free:
1-877-538-5969 x 208Phone: 954-418-8884 x 208Fax:
954-418-8635E-mail: [EMAIL PROTECTED]
-
From: Adam Goryachev [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 8:20 PM
Subject: Re: [Asterisk-Users] Echo on a PRI
On Wed, 2004-07-21 at 02:03, Deon Rodden wrote:
I installed a server in Australia with a Wildcard X100P in it. From my
server in the U.S, I pushed a call
Awesome info, thanks. I will be attending, but I live in South Florida so I
doubt my company will fly me there, they'll probably make me drive or
something. Ohh well.
- Original Message -
From: Mike Reed [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 22, 2004 10:35 AM
I'm confused. In the end, overall, which is best for a T100P (or even a
TE405P) card? IDE or SCSI? Raid or No Raid?
I was anticpating putting a single Quad-Port TE405P inside a Dell PowerEdge,
Dual 1.3ghz Processors, SCSI Hard Drive (No Raid). Was going to run 4 Full
T1 PRI's into it, either all
Never been a fan of D-Link. However it's interesting they'd make something
like the DPH-80S. I have family in Australia and might be able to get one,
but I'm curious on why it's so popular, how does it to compare against the
Grandstream Budgetone 101?
- Original Message -
From: Kanuri,
.
What are you looking to do with this system? what kind of traffic will be
going through these 4 T1s?
MATT---
-Original Message-
From: Deon Rodden [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 22, 2004 12:41 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RAID/SCSI/IDE/SATA
I installed a server in Australia with a Wildcard X100P in it. From my
server in the U.S, I pushed a call via IAX to the server in Australia which
then pushed it out that card. Severe echo, only I could hear it though. The
remote side heard nothing. Definately been reading up on this echoing
Turn off dhcp first. Option 25 in network configuration.
- Original Message -
From: xfastjackx [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 12:34 PM
Subject: *SPAM FOUND* [Asterisk-Users] how to configure my cisco
7960?!
hi everybody,
just tried to
You want Asterisk to take a call in via SIP and pass it
to the PSTN? Via what hardware?
I may be able to help. E-Mail me via drodden at webunited
dot net
550 Fairway DriveSuite 210Deerfield Beach, FL
33441Online: www.webunited.net
Deon Rodden
What problems are you having? Did you modprobe zaptel,
then wcfxo? Is it configured correctly in /etc/zaptel.conf and
/etc/asterisk/zapata.conf ?
You should not have to scrap your configuration or even
recompile Asterisk. Just load the right modules, run ztcfg and Asterisk should
take it.
Sorry, I've been on voip-info.org but I still can't get a clear definition
of what IAX trunking is. It says you need the timing from a zaptel device
(or ztdummy or zaprtc) to make it work, but nothing specific about what it
is or what it does. Maybe I'm looking in the wrong place.
Right now, I
I have a Cisco 7960 phone, I just updated to P0S3-07-1-00 image/firmware,
due to a ton of fixes from P0S3-06-3-00 which we were running. But now when
I call my phone using X-Lite, the second I answer, it reboots. I tried
upgrading to the latest X-Lite but nothing. So I then tried FireFly, and the
Is this done automatically when using IAX2?
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 04, 2004 9:37 AM
Subject: Re: [Asterisk-Users] What is IAX Trunking?
On Sun, 2004-07-04 at 08:24, Deon Rodden wrote:
Sorry, I've been
It wasn't a corrupted load, tried this on 5 different phones. For whatever
reason, it's because I had canreinvite=yes on, and nat=no
The phones are on a 10.0.10.0/24 network and my workstation is on the
10.0.0.0/24 network. There is a firewall device linking the 2 subnets.
Either canreinvite=yes
9:32 AM
Subject: Re: [Asterisk-Users] What is IAX Trunking?
On 04/07/2004, at 11:24 PM, Deon Rodden wrote:
Sorry, I've been on voip-info.org but I still can't get a clear
definition
of what IAX trunking is. It says you need the timing from a zaptel
device
(or ztdummy or zaprtc) to make
While your frustration is understandable, FWD is a free service.
It's plausible they had a busy night or whatever, and they couldn't handle
that kind of traffic. Although I've never had a problem personally with the
conference rooms, I rarely use them.
If you really needed the conferencing
Weird problem.
We have 3 PRI's and 1 5 year old Channelized (Channel bank?) T1 (24 lines,
not pri, no caller id support). Incoming calls run into a Cisco, from there
it gets sent to the Main Asterisk server.
Now, when I have it go to an extension, and have |m at the end to play
music during the
What's your iax.conf config files look like on both end? And your dial
statements in the extensions.conf file? Also, what version of Asterisk are
you running locally, remotely?
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Thursday, July
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