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?
Regards
Bilal
--- On Tue, 9/2/08, Erik Anderson [EMAIL PROTECTED] wrote:
From: Erik Anderson [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk Trunk and normal
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Tuesday
On Tue, Jul 15, 2008 at 3:22 PM, Olivier [EMAIL PROTECTED] wrote:
Hi,
How can I be notified anytime a given warning message appears in Asterisk
logs ?
Oliver -
This is a project I've had my eye on for a while:
http://www.splunk.com
I've never used it, nor have I set it up, but from reading
.
Ideas?
Thanks!
-Erik
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On Fri, Jun 20, 2008 at 12:47 PM, JR Richardson
[EMAIL PROTECTED] wrote:
So now the PBX is over 1.2 Gig for the installation. Typical PBX
installs are under 600 Meg. This makes me wonder about server
stability, reliability and performance as uptime creeps on and user
count increases over 50
On Wed, Jun 4, 2008 at 5:52 PM, Bob G [EMAIL PROTECTED] wrote:
None of them have features like hold, transfer, voice mail, dtmf, conference
as far as I know none of them has caller ID
Only 1ezphone.com has all that and the buttons are programmable for CRM
features.
Hrm:
- no apparent
On Thu, May 1, 2008 at 3:19 PM, Frank Tarczynski [EMAIL PROTECTED] wrote:
Is 384kB up too slow?
Probably not.
Is there any guidance for the minimum upload speed for an Asterisk box?
I'm guessing this is for just a few calls at a time, correct? I'd
guess that rather than these quality
On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when it was used. I played the wav file but I have never heard the
On Wed, Apr 2, 2008 at 9:37 AM, Ruben Zamora [EMAIL PROTECTED] wrote:
I just want to know if anyone have problems with server DELL 1600,
Like: Hangup Call.
Give us some more details of your setup and you'll probably have
better chances of getting an answer.
-Erik
On Wed, Apr 2, 2008 at 10:24 PM, Al Baker [EMAIL PROTECTED] wrote:
Clearly all of this not feasible in a IVR environment, so, in the
absence of all this, just how good , and how sophisticated of a voice
recognition can one achieve ?
Have you ever called Google 411?
1-800-GOOG-411
It'll
On Mon, Mar 24, 2008 at 1:56 PM, BerkHolz, Steven
[EMAIL PROTECTED] wrote:
I am not going to go into a sales pitch.
This is just an FYI to this opportunity.
Sorry, but one man's opportunity is another man's sales pitch.
To sign up to be a distributor , which is required to make money, is
On Wed, Mar 19, 2008 at 12:48 PM, Carlos Carvalhar
[EMAIL PROTECTED] wrote:
How do I install phpagi?
http://phpagi.sourceforge.net/
Since phpagi is really just a set of php libraries, all you need to do
to install is dump it somewhere and add that location to your php
include_path.
-Erik
On Wed, Mar 19, 2008 at 1:31 PM, Carlos Carvalhar
[EMAIL PROTECTED] wrote:
But when I download the gz file it doesn't uncompress as php files, the
phpagi-2.14.gz file returns a phpagi-2.14 file...and I tried with winrar and
7-zip that usually uncompress gzip files without problem.
How
On Wed, Mar 19, 2008 at 4:38 PM, Bill Andersen [EMAIL PROTECTED] wrote:
Although this is a users list, I think it is more of a list
for Asterisk resellers. I'd be interested in how many of you
are simply using Asterisk as your phone system and NOT selling
your services or an Asterisk
On Mon, Mar 17, 2008 at 12:09 PM, Brett Crapser [EMAIL PROTECTED] wrote:
Then I noticed how all the asterisk files/directorys had been 777'ed.
Ouch - I think I'll pass as well.
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On Thu, Mar 13, 2008 at 4:04 PM, Mike Hammett [EMAIL PROTECTED] wrote:
I need to setup a small mail server on a local network. It only needs SMTP
ability as it's just so Asterisk can send out emails. The machine has
sendmail installed. My primary mail server seems to be rejecting the
On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote:
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
Hey Mike - give IPKall a try:
http://www.ipkall.com/
They'll give you a free Washington state DID along with free SIP to
your asterisk
On Mon, Mar 3, 2008 at 12:40 AM, NOC Ph [EMAIL PROTECTED] wrote:
[NOCPH] I have to open the SIP port and web. Another question, the SIP port
is 5060 UDP, how about the conference? Does it use the same port also?
That's a good start, but you'll also need to open the RTP ports as
well - these
On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED] wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK.
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On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Greetings,
How can I call cheap to UK cell phones. I am located in Toronto, Canada, but
need to call UK cell phones both from Toronto and London.
I'd guess you could get an account with one of these providers:
On Tue, Feb 26, 2008 at 2:10 PM, Matt [EMAIL PROTECTED] wrote:
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers don't work
with Linux, unless you load a new driver.They sell servers with a PCI-e
slot in them,
On Wed, Feb 20, 2008 at 8:49 PM, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
I currently have about 10 Asterisk servers scattered around the place each
hosting their own dynamic conference centre. Is there any way that when
people join these conference centres on each server that somehow
On Thu, Feb 14, 2008 at 8:38 PM, Al lists [EMAIL PROTECTED] wrote:
Always rely on free -m to see how much free memory you have not top.
You could install and use htop - it's a much more functional (and
informative) version of top. It shows the difference between
shared/buffer/cache memory.
On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
It also consumes more CPU.
True, a fraction more. If you have that little overhead on your
server, though, that this would cause a problem, you probably should
upgrade your hardware, IMHO.
-eriik
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On Feb 5, 2008 2:32 PM, Sanjoy Rath [EMAIL PROTECTED] wrote:
The Asterisk server is a linux server. There is no firewall between the
servers. It is in a DMZ.
My bet is that it's not a *true* DMZ. You're still dealing with NAT,
and that's what's causing the one-way audio.
This topic has been
On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote:
How about http://www.mgamble.ca/oss/iphone_asterisk/ ?
Hah! Cool, but quite ridiculous. :-)
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on the list: HP Proliant
DL360IBM x206IBM x346
Does anyone has a most recent list and I will be adding the digium cards for
T1 the 220 series with echo cancellation?
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On Jan 24, 2008 10:14 AM, Matt [EMAIL PROTECTED] wrote:
That worked... hrmm not that great... anyone know of any decent sounding
recording of Allison for Asterisk?
What's your definition of decent sounding? IMHO and that of many of
my co-workers, the default Allison recordings sound great...not
On Jan 20, 2008 7:14 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
I have added two extentsions. I am try to test connecting X-lite to the
server.
I have two extension one 1000 with password 1234 and one 2000 with password
2000.
Andrew - could you send us the relevent sections of your
On Jan 20, 2008 7:47 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
Here are my log information.
[Jan 20 12:34:00] NOTICE[2637] chan_sip.c: Registration from
'Andrewsip:[EMAIL PROTECTED]' failed for '192.168.3.116' - Device does
not match ACL
[Jan 20 12:35:33] NOTICE[2637] chan_sip.c:
On Jan 20, 2008 8:06 PM, Andrew Ladanowski [EMAIL PROTECTED] wrote:
Windows XP.
Andrew - you're going to need to get us your sip.conf before we can
really assist you any further.
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On Jan 16, 2008 6:39 PM, Steve Totaro [EMAIL PROTECTED] wrote:
Unbeatable price for a low end Asterisk server (or any server for that
matter)
http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=bednv4ks=bsd
I wonder if anyone has any experience with this box and Digium
On Jan 16, 2008 7:28 PM, Steve Totaro [EMAIL PROTECTED] wrote:
You can add the raid option for $199. I think I might pickup about ten of
them at this price. I can always resell them as general purpose servers or
even workstations if Asterisk/Zaptel/Linux does not like the boxen.
Ahh - nice.
On Jan 10, 2008 8:24 AM, Drew Gibson [EMAIL PROTECTED] wrote:
It has 5 ports! Although the ports are labeled as 1 Internet port and 4 LAN
ports, each can be assigned to a VLAN of your choosing and you can use them
as you please (at least you can under openWRT).
Yup - you can do the same with
On Jan 9, 2008 8:33 PM, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Does anyone know of a cheap (very cheap) dual port traffic shaping box
(i.e. sub $100) that can be configured for IAX/SIP?
Pick up a Linksys WRT54GL and install dd-wrt on it.
On Jan 9, 2008 9:40 PM, Matt Riddell [EMAIL PROTECTED] wrote:
Heh yeah that's what I was thinking of doing. What's the traffic
shaping like? Can I specify max bandwidth etc or use hfsc shaping?
DD-WRT will do both HTB and HFSC shaping, though I've only ever used HTB.
Here's the dd-wrt wiki
On Nov 28, 2007 9:44 AM, Dovid B [EMAIL PROTECTED] wrote:
So do I. I set SIP to high how ever the calls are still bad. I guess I need
to read up a bit more on the firmware and how to set it up correctly.
Are the calls poor quality in both directions or on just one of the
legs of the call?
On Nov 28, 2007 10:52 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
Do sangoma cards use the standard Zaptel drivers? Or do they have to be
compiled externally like Rhino cards?
Sangoma maintains a patchset that gets applied to the stock zaptel
drivers before compilation. They provide automated
-678-954-0671
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On Nov 26, 2007 7:51 AM, Dovid B [EMAIL PROTECTED] wrote:
Hi,
I am currently playing with DD-WRT and I like it. I am looking for something
that is more SIP Aware. Anyone know one those that are out there ?
Dovid - what exactly are you hoping this sip aware firmware will do
that dd-wrt doesn't?
On Nov 26, 2007 8:29 AM, David Boyd [EMAIL PROTECTED] wrote:
I struggle with the traffic shaping rules, would you be willing to
provide additional details as to what you have done in past?
Any additional information would be greatly appreciated.
Sure - I use the default HTB traffic
On Nov 26, 2007 3:07 PM, Bob Gibson [EMAIL PROTECTED] wrote:
VMukti.com
I have a few comments for you:
1. Your webserver has been throwing 500 errors all afternoon.
2. It appears that all you've been doing with your time all day is
spamming the list with VMukti.com.
3. Do you really think
On Nov 17, 2007 11:49 PM, Michael J. Liberatore
[EMAIL PROTECTED] wrote:
I figured that one side would be pri net and the other would be pri cpe,
well I chose pri cpe and the next question was asking for a switch type,
national isdn 2, att, nortel, etc - that sounds really wrong.
Pick
On Nov 15, 2007 12:55 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
In my experience, it's easier to combine them all into one syslog
server, and then utilize tools to filter them apart when necessary,
since there are more tools to do that than to *combine* them when that
is necessary, which
On Nov 14, 2007 4:15 PM, Richard Cahilig [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
cdr report. The message is when you open the cdr report is: - Call
Logs -
On Nov 13, 2007 11:21 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
Anyone knows what is wrong with this mailing list its a while all my new
posts appear as a reply (branch) for others post, is there any hints i
could prevent this issue??
I believe your posts are all showing up
On Nov 13, 2007 11:44 PM, Mohammad Shokuie [EMAIL PROTECTED] wrote:
HI Erik,
thanks for your post, Actually im sending new posts not replying but if you
see them correct, how come its wrongly viewed for me. Are you using a
speciall software to view mailing lists? Im just using firefox not
mailing list
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On 10/26/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Yes - a crossover *is* needed in
On 10/23/07, Joseph Begumisa [EMAIL PROTECTED] wrote:
Has anyone had any compatibility issues with a TE110P card installed on a
Dell Poweredge 1950? I noted the following error on the LCD display of the
Dell Poweredge 1950:
E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0.
The Dell
On 10/22/07, Vincent [EMAIL PROTECTED] wrote:
2008 might be a good year to update * - The future of telephony :-)
Version 2 of TFOT was just released a few weeks ago...
http://downloads.oreilly.com/books/9780596510480.pdf
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On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Why is that? I'm using the xml-style config and they're working just fine.
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http://andersonfam.org
On 10/19/07, Mike Clark [EMAIL PROTECTED] wrote:
Do they play well with Vista?
Hah - I have no idea. We installed Vista on one laptop here when Dell
started shipping it. That lasted about 3 days and 10 support tickets
from the user. Then we reverted back to XP. Haven't touched Vista
since.
On 10/19/07, Steve Totaro [EMAIL PROTECTED] wrote:
Any advice on softphones, handsets, or practical experience with this
sort of deployment? It would be very nice if there was a central way of
provisioning the phones.
I've deployed several setups internally using X-Lite and these headsets:
On 10/17/07, shadowym [EMAIL PROTECTED] wrote:
Ok so you use templates. I understand that. The problem is some people on
here seem to be claiming they type it all in from scratch in like 3 minutes.
Just call me out if you feel the need to. Please don't try and hide
behind the some people on
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
I don't do text editing so please indulge me. Why would someone want to do
that when a GUI makes life so much easier?
On a practical note, If someone was deploying 2 or 3 of these a week, most
of which have 5-10+ extensions doing all kinds of
On 10/16/07, shadowym [EMAIL PROTECTED] wrote:
So how long would it take you to vi a 20 extension office with custom
dialplan involving a medium level of complexity? Including time to debug
etc.
Well - there's a large amount of subjectivity in your question, but
perhaps I'll answer with not
On 10/12/07, D4rk F1ber [EMAIL PROTECTED] wrote:
Curious what others are using, and if anyone can make some
recommendations? Not sure if this has been covered already on the
list, and not sure if recommending companies are allowed, so maybe I
need get replies off list?
There are quite
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I don't think there is a formula like
cpu usage = loadavg / #cpus
A loadavg of 3 says that there are 3 processes waiting to
be executed.
Anyway, I'll admit that a loadavg of 3 /might/ be ok.
Here's a quote from this page:
On 10/12/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
I wouldn't be too happy about a system with a
loadavg of 3.
The system he mentioned had 8 cores, though. So a load average of 3
is less than 50% usage.
-erik
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On 10/11/07, Raúl Gómez C. [EMAIL PROTECTED] wrote:
At this point I was wondering if Asterisk gets real benefits on systems with
several cores (up to 8 in Dell PE2950) for a system that will handle up to
35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax
(Sangoma
On 10/8/07, Forrest Beck [EMAIL PROTECTED] wrote:
I was told that Asterisk was supported when we looked at the service.
Hey Forrest - thanks for the information. Might you be able to send
along the contact information for the TW rep who told you that
asterisk was supported? I've been in
On 10/9/07, Hans Witvliet [EMAIL PROTECTED] wrote:
Hi all,
Probably this is the wrong place to ask,
but is there an estimated time of arrival of the future?
i.e. TFOT--next generation dealing with * -1.4
I attended a workshop some time ago, and the book was part of the
package
The
.
Thanks!
-erik
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On 10/6/07, Matt Florell [EMAIL PROTECTED] wrote:
Do not use Dell. I have had issues with both Sangoma and Digium cards
on multiple brand-new Dell servers. This is the only vendor that has
consistently given me problems with telco-interface cards.
I'll have to refute this. Every single
On 9/30/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
I don't know about you, but I've had nothing but very good results with
VOIPSupply. I didnt do huge business with them, but I have purchased new and
refurb polycoms from them without so much as an ounce of pain.
Ditto - I've never had a
On 10/1/07, Robert DeVries [EMAIL PROTECTED] wrote:
Anyone have a list of the files that would need to be moved? (Obviously the
*.conf files in the Asterisk directory, I can think of some others, but if
someone ever did a list that would be a great help.)
You'll probably want to move the
On 9/28/07, William Stillwell (Ki4swy) [EMAIL PROTECTED] wrote:
What is the recommend Digium Card for a PRI in NA ?
William - this has been discussed ad nauseam on the list recently.
Some will suggest that you forget Digium and use instead a Sangoma
card. I personally have only used Sangoma
On 9/27/07, Doug [EMAIL PROTECTED] wrote:
http://www.atacomm.com/
Heh - yah I pulled up their website earlier today with the hopes of
purchasing a Polycom SIP conference phone. Oh well...
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On 9/24/07, Steve Davies [EMAIL PROTECTED] wrote:
The phones can send a parameter to the provisioning server to indicate
that they want an Update if they do this, and you send no network or
other major config parameters, the phone does not reboot.
Look at the Linksys provisioning PDF for
On 9/18/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hi all,
Does Asterisk contain a full fledged ISDN packet sniffer. By giving the
command
pri intense debug span 1 , does it debug every packet received
(control and voice/data packets) ?
To get the equivalent of a packet sniffer, you'll
(one of the
above options or otherwise) is best to keep your sip.conf sane?
Thanks!
-Erik
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On 9/18/07, C F [EMAIL PROTECTED] wrote:
Use the extension, and use grep to determine which account uses which
phone. For example I provision my spa9xx phones from a subdirectory on
apache called spa which on slackware is at: /var/www/htdocs/spa/
doing:
grep 123 /var/www/htdocs/spa/* will
On 9/7/07, Mike Hammett [EMAIL PROTECTED] wrote:
If it has nothing to do with Asterisk, then why does every other device work
as its supposed to?
You never answered as to whether or not you're able to get out past
your gateway with any other network applications on your asterisk
server. Fire
On 9/6/07, Mike Hammett [EMAIL PROTECTED] wrote:
I have multiple upstreams in my office. The primary upstream is having some
issues with latency\jitter. I want to move the VoIP traffic to another
interface.
I have the router set to send all traffic destined for local networks out
the
.
Is there currently any script out there that would facilitate this
sort of testing?
Here's my current config:
linux-2.6.21
asterisk-1.4.10
zaptel-1.4.4
wanpipe-3.1.3
libpri-1.4.1
Thanks!
-Erik
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On 8/28/07, Brian West [EMAIL PROTECTED] wrote:
What exactly are your needs? I can provide you some sipp scripts
that might help you.
Brian - thanks for the reply. If you read my email, I believe I make
it fairly clear what my needs are. I have a 4-port Sangoma PRI card
installed. Crossover
On 8/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Another more creative tool would be to place an ad in the Penny Saver or
whatever your local equivalent is for a free 42 inch LCD TV, you haul
and list your number. I bet that would generate alot of calls. You could
put them through and IVR,
On 8/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Also, you seemed to miss Brian's main point, keeping calls up is not
going to tax your box or prove anything really, you want to create as
many short calls as possible. Run BOINC in the background for a CPU
burn-in test.
Well - with this
Off the cuff, I can't recall if asterisk can listen for (in this case
I assume) SIP on multiple ports. It would be quite easy to do this
redirection with iptables, though.
On 8/15/07, Walter Willis [EMAIL PROTECTED] wrote:
hot to asterisk multiport...???
example 5060, 5061, 5080
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On 8/13/07, John Meksavan [EMAIL PROTECTED] wrote:
Asterisk Users,
I have never done a dial plan for this scenario before. Is it possible to
have Sip Phones make outbound calls through the PSTN? What would the call
routing/dial plan would look like?
Yes - certainly possible. There's
On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote:
I've been going back and forth with my telco for several days, trying
different configurations to get a new PRI to come up. The bchannels
are all up and the T1 is not in alarm status. The dchannel refuses to
come up however. We've tried ni2
On 8/9/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote:
Is the OpenVPN the ideal solution to set a tunnel between two asterisk
servers or there is a better solution.
There is no global ideal solution. The solution that is ideal for
*you* depends on many factors:
- What will the tunnel be used
On 8/9/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote:
Hello,
Have i to install OpenVPN in each Asterisk server or it is enough to install
it in one side only?.
Both.
You best take any further questions to the OpenVPN mailing lists.
You'll get much better information and help there.
On 8/7/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
In your wanpipe1.conf see if you have
TDMV_DCHAN = 0
Nope. I have it set to 24.
-erik
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On 8/7/07, Jeremy Mann [EMAIL PROTECTED] wrote:
In Zapata.conf, if my PRI is NI-2 configured, do I still use
switchtype=national ?
Yup:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf#ISDNPRISwitchConfiguration
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On 8/7/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I want to have conference call(meetme) service with asterisk and 30
users.Now do I use 1E1 or 30 analog lines with due attention to high price
of E1 line?And which interface card do I use?
I'm not sure what analog prices are in your area,
firmware
and I'm running wanpipe-2.3.4-12 for the sangoma drivers.
Any ideas?
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On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
Call Sangoma and give them root if you can. They will fix it quickly or
at least give you ammunition that it is the telco's issue.
Good idea - I just emailed them. Hopefully they'll respond quickly. My
normal contact there (Jignesh) is either
On 8/6/07, Darryl Dunkin [EMAIL PROTECTED] wrote:
wanpipemon is the way to do it as far as I know.
For starters, what do your zaptel/zapata configs look like?
lpdlnx04*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: Nortel DMS100
Type: Network
I know
On 8/6/07, Steve Totaro [EMAIL PROTECTED] wrote:
I have done a conference call with the telco guy, myself, and a Sangoma
tech at the same time. I was just quite and let them battle it out. It
turned out to be a telco issue but the Global Crossing tech wanted to
blame me and my equipment.
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
also in asterisk do:
pri intense debug span 1
Then you should see UA's and SABME's, If you don't, your not talking to
them.
I see plenty of SABMEs, but nothing else:
[ 02 01 7f ]
Unnumbered frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
You should never be the signaling source, you are always a slave to the
provider, go with pri_cpe and see if things go better.
That's what I've experienced in the past, but they were adamant about
me being the network end. I tried switching
On 8/6/07, Anthony Francis [EMAIL PROTECTED] wrote:
Yeah you are sending the SABME's because you think you are the master,
they are not replaying with a UA because they think they are the master,
you should def be pri_cpe.
Tried it...no go.
There is one other potential cause here, you may
On 8/2/07, Don Kelly [EMAIL PROTECTED] wrote:
Hi, Erik,
Never heard of call-by-call trunking.
Are you in Minnesota? What carrier are you using?
Yes I am...this is for one of our branch offices, though, outside of Boston, MA.
-Erik
___
--Bandwidth
anything.
Thoughts?
This is a Sangoma A102 card, by the way. In this case, though, I
don't think that's of any relevance.
-Erik
--
Erik Anderson
http://andersonfam.org
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk
On 8/1/07, C F [EMAIL PROTECTED] wrote:
what channel are they putting the Dchannel on?
Post your zapata.conf and zaptel.conf
The D channel is on 24.
zaptel.conf:
loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
zapata.conf
lpdlnx04 asterisk # cat zapata.conf
;autogenerated
On 8/1/07, John covici [EMAIL PROTECTED] wrote:
I had some troubles -- try setting the timing parameter to 0 (second
one in your span) and see if that helps.
If I'm reading the docs correctly, this param should only be set to 0
if you *never* want to use the T1 connected to this port for
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