I've read a lot of the descriptions of handling NAT with Asterisk,
and the use of both the nat and canreinvite flags. I am very
familiar with Sip and NAT but have not seen an answer to the following
question.
My Asterisk server runs on a machine with two ethernets. One is
an external net,
All,
I have upgraded by home machine from Fedora Core 5
(FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and
Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with
845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686so we hve
this:
Hello... I have a big problem with asterisk. Every time i make a call
asterisk does not bridge the zap channels. The zap channel from which
i'm calling remains in state:ring and applicaton:dial and the zap
channel with the external line configured remains in state:dialling an
Hi!
For demonstration purposes I try to bridge an incoming video call from a
3G mobile handset to another 3G mobile handset using asterisk as switch.
On the incoming call leg I see all expected bearer capabilities
(Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call
leg the
Hi list,
I have been using the read command and I have
noticed that it behaves more like the playback command and not background. Is
there any way to set it up that I can enter a selection before Asterisk finishes
playing the file ?
Thanks.
Dovid
Hi List,
I have a F300. I have not yet been able to set it
up. Can anyone email me the exact configs on how they set it up to work with
Asterisk ?
Thanks.
Dovid
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What attributes are you talking about ? Depending
on what they are it may be real simple to set something up.
- Original Message -
From:
Curt Shaffer
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Monday, October 30, 2006 9:51
PM
Subject:
31 okt 2006 kl. 01.37 skrev je .:
Thanks for the diagram. Is it possible to get a more detailed
diagram. I'm looking for something a little more technical. In
other words, where does Asterisk stand when inviting a user, when
hanging up, when canceling an invitation etc.. Does it go
31 okt 2006 kl. 06.34 skrev je .:
Thank you for the link. Chapter 8 was most useful in explaining the
different types of connections (user/peer/friend) as well as the
register function such that users may know how to contact another
user. However I'm looking for something more
When user A is bridged with the client (right before it is done) set a
variable that the call was taken, this way when he presses 1 to accpet the
system checks to see what the value is, if its in taken mode he gets the
message and then gets hung up on.
- Original Message -
From:
Good day
Im look at
http://www.voip-info.org/wiki-Asterisk+GUI
And I see there are a few GUI for asterisk
What do you guys prefer?
What is the best and simplest? Id like something that give me
access to backend for a little bit of customization
Thanks for you help and time
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I think the (T) is for Trunk.
Regards
Fred
Hi Fred!
I believe that T is for trunk. Thank you.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
Brad Templeton wrote:
I've read a lot of the descriptions of handling NAT with Asterisk,
and the use of both the nat and canreinvite flags. I am very
familiar with Sip and NAT but have not seen an answer to the following
question.
My Asterisk server runs on a machine with two ethernets. One
Rajkumar S wrote:
-- Executing Queue(SIP/1002-74e9, Auth-Enq|t) in new stack
-- Started music on hold, class 'default', on channel 'SIP/1002-74e9'
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/1001||tS(30))
Hello,
I have a problem with a new provider and perhaps you can help me .
whenI send traffic to this new provider i have this error :
Dropping extra frame of G.729 since we already have a VAD frame at the end
and the quality of the voice is bad because some parts of words are dropped.
On Tue, 31 Oct 2006, Pedro Silva wrote:
Hello,
One problem is solved and another appears... :(
I cannot receive incoming calls on trixbox. I defined one incoming
route (any DID/any CID) and forwading these calls to a SIP extension.
With capi and sip debug in asterisk -r console i dont
Hi Pedro,
pls post your capi.conf! I'm not used with CAPI, but should have something like:
[interfaces]
incomingmsn=* ; Here you match MSNs arriving from telco, for debug
let it ' * '
controller=1
softdtmf=1
accountcode=
context=demo ;Set this to ext-did that should be the context
TRIXBOX
Can someone confirm if sip realtime is broken in
1.2.13 and if so when was the last release it wasn't?
heh
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i've been using it sucessfully
On 10/31/06, Don [EMAIL PROTECTED] wrote:
Can someone confirm if sip realtime is broken in 1.2.13 and if so when was
the last release it wasn't? heh
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From WIKI:
D(digits): After the called party answers, send digits as a DTMF stream,
then connect the call to the originating channel. (You can also use 'w'
to produce .5 second pauses.)
When I use the D option to send a call to my paging system and pick a
zone, the Tone is too early.
I have
Hi,
You have to modify the setup sent for the call outgoing...
Regards,
Tristan
Steffen Weinreich a écrit :
Hi!
For demonstration purposes I try to bridge an incoming video call from a
3G mobile handset to another 3G mobile handset using asterisk as switch.
On the incoming call leg I see
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I was wondering if anyone has implemented a feature that would allow a user
to
record a phone call and once the call has ended, the call is forwarded to
his voicemail?
Hi Tom!
I was looking for something like this, but I was unable to
not the sip.conf static realtime but sipusers etc?
- Original Message -
From: Marco Mouta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 31, 2006 7:26 AM
Subject: Re: [asterisk-users] sip realtime
Basically I would like a page that would
allow a user to log in and modify their extension only. So for example, I log
in for extension 102 once in there I can turn on or off my call waiting. Add a
number to call forward to. Change the email address my voice mail gets sent to.
Add any
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Did you ignore that is an english discussion list or did you just post
on the wrong list altogether?
Anyhoo, try relaxdtmf=yes in your zapata.conf.
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (MingW32)
Comment: Using GnuPG with Mozilla -
Hey,
This is probably a rather stilly question...
If I pick up my SIP phone that's registered to my asterisk server and dial a
number that asterisk recognises as destined for a SIP trunk (could be a
static route, or an ENUM lookup) or another SIP device registered on said
asterisk server
Hello list,
We have 2 asterisk servers (without firewall and NAT),
and We want to do :
From the first server, we have a .call file which dial out to the
second server. The second server automatically answers and Play a music
during X seconds, then it hangs up.
Is it possible?
Thanks
Anybody knows why ARI gives this error message when I enter extension number and password.
Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525
It doesn't show the
Which asterisk release are you running chan_skinny under?
- Original Message -
From: Will Roy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central
Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone
Before I got
You can make RTP pass through Asterisk, or not. Look in voip-info.org
about Native Bridge and sip.conf canreinvite option. And may be
this page will be usefull too:
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy
Regards
On 10/31/06, Mike Williams [EMAIL PROTECTED] wrote:
Tristan schrieb:
Hi,
You have to modify the setup sent for the call outgoing...
OK, is there a way to this from the dialplan of have i to modify source
for this?
cheeiro
Steve
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User that web server is running has to have read permissions to file
/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt
Easier option is to run apache as asterisk user
Zeeshan Zakaria wrote:
Anybody knows why ARI gives this error message when I enter extension
number and password.
On Tue, 2006-10-31 at 09:55 -0500, Zeeshan Zakaria wrote:
Anybody knows why ARI gives this error message when I enter extension
number and password.
Warning:
file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt):
failed to open stream: Permission denied
in
Hi All,
I
have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk
Management Portal (AMP) for web interface.
After installing properly when opening in the webpage it is
not parsing the index.php for the AMP. My Database is MySQL.and web server is
Apache 2.2.
Please let
it told you:Permission DeniedCheck the permissions on that directory.On 10/31/06, Zeeshan Zakaria
[EMAIL PROTECTED] wrote:Anybody knows why ARI gives this error message when I enter extension number and password.
Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed
We have a
centralized infrastructure where we deploy Asterisk servers in remote call
centers for authentication and transcoding. SIP g729a calls are then sent
over an MPLS VPN to a central Asterisk farm, from which calls
aresent/received via PRI.
To avoid placing two
servers in each call
I would like to know how to record all calls on a queue. Anu good sugestions?
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If you did make samples you should see an example in queues.conf. By default
it's commented out.
- Original Message -
From: Ed Nuñez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 31, 2006 5:31 PM
Hi Folks,
I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format, are there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.
Thanks
Rgds.
--
Omar E.P.T
Did you install PHP?On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote:
Hi All,
I
have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk
Management Portal (AMP) for web interface.
After installing properly when opening in the webpage it is
not parsing the index.php
Dear Jason,Please define better noisy? You talking echo issues? Is it on just your side or on the called party's side as well?This start happening immediately, or was the box working before and the problem just started?Also, a quick heads up, make sure before even beginning to troubleshoot an
If it is not parsing the index.php mean that you see the code in your
browser, install php
Alok Mohapatra wrote:
Hi All,
I have installed Asterisk 1.2.10 on Fedora 5. I have
installed Asterisk Management Portal (AMP) for web interface.
After installing properly when opening in
For the benefit of those outside of the USA or those
unable to make it to Astricon; I wanted to send out this email.
For those of you who attended Astricon in Dallas last week what was
the one thing that you saw that made the trip worthwhile?
(if we post enough information or
After installing properly when opening in the webpage it is not parsing the
index.php for the AMP. My Database is MySQL.and web server is Apache 2.2.
Please let me know is this configuration problem or this is the problem with
Apache (Apache 2.2) .
The problem is probably that you didn't
omar parihuana wrote:
Hi Folks,
I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format, are there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.
Thanks
Rgds.
Hi All,
I have one rather
annoying problem...my PBX can work great for weeks, when suddenly I start
receiving these messages when I try to make a zaptel call:
Oct 31 13:52:47
NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
Oct
Well it works.
If I have group=0 that includes all my channels, I can create group=1 which is
a subset and a simple reload makes this g1 available to dial on that subset.
Message: 12
Date: Mon, 30 Oct 2006 15:25:06 -0700
From: Alyed Tzompa [EMAIL PROTECTED]
Subject: re: [asterisk-users] Live
On Mon, Oct 30, 2006 at 06:54:40PM -0500, Vitalie Apostu wrote:
Greetings,
If somebody knows how to concatenate several .gsm files in one or create a
macro and use with background() please reply.
As simple as:
cat file1.gsm file2.gsm both.gsm
--
Tzafrir Cohen
Alok,
Two things:
1) You said you installed AMP. AMP has ceased development a while ago, but is survived by the FreePBX project. If you actually installed AMP and not FreePBX, I would suggest you get FreePBX running first. A lot of effort went into improving FreePBX from AMP.
2) You typically
Possibly a silly question, but do you have php installed
and configured in apache?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alok
MohapatraSent: 31 October 2006 15:45To:
asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk web
interface is not parsing
What about the AD in the VM. or running Open LDAP on the Asterisk server.
On 10/31/06, Adam Robins [EMAIL PROTECTED] wrote:
We have a centralized infrastructure where we deploy Asterisk servers in
remote call centers for authentication and transcoding. SIP g729a calls are
then sent over an
Hi Alok,
it seems like libapache2-mod-php is missing in your linux box.
Have you tried to make a simple index.php file to test?
Giorgio Incantalupo
Alok Mohapatra wrote:
Hi All,
I have installed Asterisk 1.2.10 on Fedora 5. I have
installed Asterisk Management Portal (AMP) for
Everytime a voicemail is recorded, a .txt file is generated. It was working fine before and permissions were automatically set. On my home server it is working perfectly fine. This is another server, with the same settings, and all of a sudden today it has started to give this error. Voicemails
Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need
to focus more in SIP and Asterisk compatibility and less in pricing
(yes, I know the Cisco are more expensive).
Are
Hi,
Has anyone here reprogrammed their Polycom features keys using
sip/ipmid.cfg?
If so I would be really grateful if someone could send me an example as
I have tried various entries for hours now and don't seem to be getting
anywhere.
Any help appreciated.
Hi, folks:
I need to be able to have a single DID ring multiple remote (IP and
PSTN) extensions, and then pass the call to whichever picks up first.
I'm sure this is old hat -- lots of providers offer it.
I see that Trixbox will do it, but it's not clear how it's doing it.
They use different
Asterisk does not work very well in a VM
due to the timeslicing. Dropped calls, jittery audio and echo can all creep in.
Good news is that an AD controller runs
just fine in VMware. Just make sure the box has enough RAM to keep it happy,
and use a physical second disk for the Windows
Hello All,
This is a great list post, I have blogged about it
here: http://www.asteriskvoipnews.com/asterisk_news/astricon_2006_followup.html
It would be great if people could post there
response on this post along with the list. I love reading answers to
questions like this. Thanks,
-Dal
Can anyone suggest any reasons why a zap (PRI) b channel
should not be a member of multiple zap trunk group definitions?
For example;
Group 1 = Channels 1 to 23
Group 2 = channels 1 to 12
Group 3 = channels 13 to 23
The purpose is to restrict the number of channels a
particular
Is there any advantage of getting a T1 card with a
channel bank over 2-3 FXO cards ?
Thanks.
Dovid
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On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
Have you tried setting the externalip and localnet parameters?
Localnet makes some sense, and is set (should be the default anyway, no?)
externalip, as I understand it, is for an Asterisk which is behind
a NAT. This asterisk is
Cisco Cisco or Linksys Cisco?
Cisco Cisco, I'd prefer the Snom. Linksys Cisco, it's a tossup.
I've worked with dozens of the Cisco 7960 phones, 25 of the Linksys, and 3
Snom.
My specific issues with the Cisco included poor echo cancellation, problems
with nat traversal, and no web interface.
Hello,
I'm working with supermicro servers, for the irq problems with Dell, any people have problems
Regards
On 10/30/06, Paul Hales [EMAIL PROTECTED] wrote:
How many analog lines are you looking at? Hundreds?PaulHOn Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote:
We have a number of clients
You can just seperate multiple phones with in the Dial command,
as the voip-info wiki page shows:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote:
Hi, folks:
I need to be able to have a single DID ring multiple remote (IP
When I look at TTY9 (using init.d and safe_asterisk to start the
asterisk process), I am getting some strange characters. When a
application is run the and the CLI shows the application executing the
languange almost looks russian...??
Anyone seen this before?
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct
DTMF tones 25% of the time. I have to call several times to enter an
extension. I have a router and a packet shaper and some other stuff.
Anyone have any other ideas why this might happen. I do not have any
Zap channels
firewall? i dont think so because sometimes the phone can register ok
and sudendly the appears unregistered
Leonardo Silva [EMAIL PROTECTED] ha escrito:
2006/10/31, Jon Farmer [EMAIL PROTECTED]:
Sergio R. D'Ippolito wrote:
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I
Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia
cars
Joao Pereira wrote:
Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt
between Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I
need to focus more
YES!
Many machines do NOT work well with multiple analog cards. Especially
the Digium ones.
Channel banks with FXO circuits are harder to come by on the used
market, though
Many all FXS channel banks can be had used, though.
If you want multiple FXO's and do not want to go the T1 route, look
Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO
cards ?
Thanks.
In my experience a T-1 port w/channel bank just works better. The more
cards you use, the more interrupts are generated.
My standard configuration for analog FXS ports is a T-1 card
Damon Estep wrote:
Can anyone suggest any reasons why a zap (PRI) b channel should not be a
member of multiple zap trunk group definitions?
For example;
Group 1 = Channels 1 to 23
Group 2 = channels 1 to 12
Group 3 = channels 13 to 23
The purpose is to restrict the number of
On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank
over 2-3 FXO cards ?
If you need enough ports to make a T-1 card cost-efficient, then you
might oughtta be looking at an Ethernet to FXO media gateway instead --
I think one of the differences is: We do pay attention to Asterisk and this
mailing list ;-)
CS
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira
Gesendet: Dienstag, 31. Oktober 2006 13:47
An: asterisk-users@lists.digium.com
Betreff:
On 26 Oct 2006, at 13:25, asterisk wrote:
Hi all,
Can tell me somebody what meen : channel.c: Avoided initial deadlock
Our customer makes calls with our softphone (with IAX2).
Sometimes the softphon freezes. The call is ACTIVE but the user
cant hang it up.
At this time in the log file
On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote:
Hi people,
I would like to read your suggestions as to where the issue might be.
ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS
port.
TDM04B= 4 FXO signal fxls
There is a 8FXO-to-SIP unit in this
On Mon, Oct 30, 2006 at 03:08:39PM -0500, Andre Courchesne - Consultant wrote:
Hi,
Is there a way to create trunk groups while asterisk is running.
For exemple let's say that zapata.conf defines g0 as channels 1-23
I would like (while asterisk is running) define g1 as 1-10 and g1 as
Brian Rogan wrote:
You can just seperate multiple phones with in the Dial command,
as the voip-info wiki page shows:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Thanks! It's not always clear where to look first for these things.
I'm repeatedly blown away by the ease of
On 26 Oct 2006, at 12:12, Nick Adams wrote:
I need to determine the number of active calls in a group from
outside of Asterisk. Currently I poll the manager API and parse the
channel status list but this is becoming too expensive on CPU.
What are my options? What is considered standard
On 26 Oct 2006, at 15:33, David Bandel wrote:
Folks,
Anyone know if Asterisk supports IPv6? If not, is support planned?
There was a talk at astricon on this. (I think the slides will be
available
from astricon.net).
The short answer is no, not yet, but folks are working on it.
Tim.
Is there any advantage of getting a T1 card with a
channel bank over 2-3 FXO cards ?
Thanks.
Dovid
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jez . a écrit :
Dear all,
I've recently installed Asterisk and am trying to understand where
exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work
as a proxy. I am specifically interested in SIP. Could anyone perhaps
point me out to a diagram with SIP users and Asterisk to
Damon Estep wrote:
Can anyone suggest any reasons why a zap (PRI) b channel should not
be a
member of multiple zap trunk group definitions?
For example;
Group 1 = Channels 1 to 23
Group 2 = channels 1 to 12
Group 3 = channels 13 to 23
The purpose is to
Jason Walker wrote:
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct
DTMF tones 25% of the time. I have to call several times to enter an
extension. I have a router and a packet shaper and some other stuff.
Anyone have any other ideas why this might happen. I do not
On Tue, Oct 31, 2006 at 02:03:33PM -0500, Forrest Beck wrote:
When I look at TTY9 (using init.d and safe_asterisk to start the
asterisk process), I am getting some strange characters. When a
application is run the and the CLI shows the application executing the
languange almost looks
On 10/31/06, Joao Pereira [EMAIL PROTECTED] wrote:
Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need
to focus more in SIP and Asterisk compatibility and less in pricing
On Tue, 2006-10-31 at 13:52 -0500, Carlos Rojas wrote:
Hello,
I'm working with supermicro servers, for the irq problems with Dell,
any people have problems
I second the supermicro servers - particularly the opteron range based on
Serverworks HS1000 chipset.
Excellent stuff. Well
That and Cisco won't give you the time of day if you don't use their
stuff ;)
We have about 1600 of the Cisco's on campus, and unless you run them on
the call manager, you're not gonna have nearly as many features as any
other phone that's designed with SIP in mind. That said, if you need a
On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia
cars
Not really. Both are very good phones.
* My Clients prefer cisco because it looks more business-like. - The new
snom phones do look better though and the
Hi,
Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this:
Note: MySQL libraries are installed and the structure is as follows:
/usr/src/astsources/asterisk-1.2.13
/usr/src/astsources/asterisk-addons-1.2.5
in /usr/src/astsources/asterisk-addons-1.2.5 I do:
make clean
make
and
On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote:
On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
Have you tried setting the externalip and localnet parameters?
Localnet makes some sense, and is set (should be the default anyway, no?)
externalip, as I understand it, is for
Hi friends,
thank you for comments...
Marian
Tomislav Parčina napsal(a):
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I think the (T) is for Trunk.
Regards
Fred
Hi Fred!
I believe that T is for trunk. Thank you.
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000
I forgot to mention that the Carrier that owns the ATA box was not
willing to let me connect directly over IP, I was only allowed to use
the FXS port. He already ack that he has a problem with
disconnections.
On 10/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Oct 30, 2006 at
Looking at the number's now it seems that a T1 will be more.
Anyone here sell PRI's ?
- Original Message -
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, October 31, 2006 9:38 PM
Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with
Damon Estep wrote:
Damon Estep wrote:
Can anyone suggest any reasons why a zap (PRI) b channel should not
be a
member of multiple zap trunk group definitions?
For example;
Group 1 = Channels 1 to 23
Group 2 = channels 1 to 12
Group 3 = channels 13 to 23
The purpose is to restrict
Hi All,
I have a new client who has an existing Asterisk PABX and is looking
for us to install a TE110P for him, However he has a Dell SC420 and I
have never used one before.
I have had no problems with any other Dell servers which we use almost
exclusively.
Has anyone had any good/bad
Check out voip-info.org, there are quite a few GUIS some even generate nice graphs!On 10/31/06, omar parihuana
[EMAIL PROTECTED] wrote:Hi Folks,I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,arethere someopen source aplication
Where are these DTMF tones going? From where? Be specific, post the relevant config file sections I can't read minds and I'd be surprised if 0.1% of the people who read this can either
On 10/31/06, Jason Walker [EMAIL PROTECTED] wrote:
I have tried beta2, beta3 and now back to 1.2.12.1 and
I'm working with Supermicro as well.
-Original Message-
From: Carlos Rojas [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 31, 2006 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Server Recommendations
Hello,
I'm working with
I've done extensive testing, WDS is just as reliable as wired,however at first we had issues with some AP that would not respond and needed to be rebooted. But if its possible to wire the AP you should since WDS will eat alot of bandwidth and also decrease the range since most the AP will have to
I did this today with a Panasonic KX-TD1232 and a Digium TDM2401E
Card. I hope to put it on the wiki soon, if you need help just tell me
with what.
On 10/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If Someone did that, How I connect extensions.conf with this type of Hybrid
system to
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