On Mon, 13 Dec 2004 07:22:27 -0800 (PST), Thomas Miller
[EMAIL PROTECTED] wrote:
Can somebody suggest the easiest way to only allow outgoing long distance
calls to countries x, y, and z?
Set it up that way in your dialplan...instead of matching _011., match
_01144. for UK, _01130. for
On Mon, 13 Dec 2004, Christopher L. Wade wrote:
Mark Halverson wrote:
Sorry if this is the wrong list...
I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)
Anyone know of a service provider offering this?
Take a look at
Static configuration is where you can store regular *.conf files into the
database. These configurations are read at Asterisk startup/reload. Some
modules may also re-read this info upon their own reload (Ex. sip reload).
Right.
The table structure ast_config in Realtime Static Holds
On Mon, 13 Dec 2004 11:14:00 -0600, Adi Linden [EMAIL PROTECTED] wrote:
Hi,
How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco
claims both phones do SIP.
Both phones support SIP.
I can't speak for the 7912G, but I have several 7905G phones and these
work perfectly
We tried it using FreeBSD 5.2.1, and it worked ok for the most part. Used it
for SIP VOIP only, no hardware interfaces. We ran into a problem with the
purchased g729 codec from Digium. The registration program did not run. We
tried different versions of the LINUX emulation as well. So, we ended up
Im trying to get my asterisk box to register to a sip provider without much
success.
here is my console output in asterisk
Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration
for '[EMAIL PROTECTED]' timed out, trying again
-- Got SIP response 403 From: URI not
Hello all,
I have a problem with ASTCC. When I create all my
routes, I not able to get the destination pattern I desire. I see it come up,
but ASTCC seems to select the first available pattern, and not necessarily the
exact one I want. I found the MYSQL statement in astcc.agi:
SELECT *
Hi,
I have a problem with the queue cmd.
I am trying to redirect an incoming call to another phone when nobody in
the queue answer it within 18 seconds. Somehow the incoming call keeps on
retrying within the queue. The second part was never executed. Below is a
part of my extensions.conf
Hello there
I'm a learneron this stuff.
I have a Linux 8.0 (Redhat)
and try to install a Dialogic Card (D/41ESD)
ISA bus
i installed the release 5.1 from Intel and then the
SP1,
but cann't load drivers for the dialogic
card.
Cann anyone Help me ?
Thanks
On Tue, 14 Dec 2004, Kumaran Subramanian wrote:
I would like to know .
* Is it possible to build a VoIP system using ISDN up0 telephone lines?
I think you would normally use a NT device to get an S0 interface and
connect that to the Asterisk box. In Europe the U interfaces are ususally
Hi,
I have following setup:
BT100 Firewall/nat 1 (www.ipcop.org) Internet Firewall/nat2
(Vigor) Asterisk .
I'd like to use BT100 as local extension to Asterisk. I've done simple setup
and BT100 can call Asterisk and place outgoing calls. However I cannot set
him to qualify,
I am running it statefull because else I would have to open port 5060 on
my cisco AS5400's to the world, and that's too insecure
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Sikkema
Sent: Tuesday, December 14, 2004 8:07 AM
To: Asterisk Users
After running Asterisk in debug mode, the issue became
obvious.
My voicemail passwords have leading zeros, and since
the 'voicemail_users' table has password as an 'int'
these were stripped. Once I changed the password
field to 'varchar' then all worked fine.
Thanks for all of the help!
Jason
Check your FW-1 tracker and see if any sip packets are dropped during
call initiation.
I had this problem and it went away when I upgraded the BT's firmware to
the latest (16).
Beware, though, that people on the list claim that this firmware breaks
functionality of the message button and
Non Global SIP elements,
Non Global IAX elements,
Are there a way go get asterisk to wirte it's own hostname in the
sip/iax-friends table when a sip/iax-client connects ?
(When it update the ipaddress, client name eg.)
I have several *-servers using the same db, and the user can connect to
a
Hello,
I tried for almost 2 day to make the Function Keys
of Snom phone work. Now that they work, maybe
I can help someone.
The first two steps are simple and documented.
1) Go to Function Keys and make them "destination",
type in the extension to be
monitored.
2) add a "hint" statement
Hi,
I am having a few dial plan problems which I wondered if anyone would be
able to help with.
Firstly, I wanted to send 0800 calls through 1 sip provider and other
08xx calls through another. I have this:
exten = _0800.,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _0800.,2,Congestion
exten
If I have inbound SIP calls arriving from a provider's gateway to an
asterisk server on my LAN, which then routes the call back out via the
provider's gateway to a PSTN number, once the call is answered do all
the voice packets pass through my asterisk PBX, or is SIP intelligent
enough
Hi,
I'm currently looking for a softphone for windows, we have been using
X-Pro but it appears that X-pro doesn't support Message Waiting
notification.
Does anyone know of a well featured softphone that does support MWI ? I
can't seem to find one.
Any suggestions would be most appreciated,
I need to reopen this discussion because it's impossible to run spandsp
(and VoIP) under these circumstances.
With zaptel unloaded, I see the following vmstat 1 output:
no swapping, an occasional disk output, +/- 1003 interrupts/sec., less
than 10 context switches/sec., CPU idle 100%.
Hi,
I have a problem with ASTCC. When I create all my routes, I not able to get
the destination pattern I desire. I see it come up, but ASTCC seems to
select the first available pattern, and not necessarily the exact one I
want. I found the MYSQL statement in astcc.agi:
SELECT * FROM
Use ISDN lines!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ken
D'Ambrosio
Sent: Friday, December 10, 2004 11:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Should echo cancellation be a science or
anart?
Gents,
New to the list, and also new to Asterisk.
I was wondering - other than the official Asterisk Handbook(Draft 2) that
I have stumbled across, is there any other literature that you more
experienced asterisk users would recommend.
Also, are there any more resources with regards to
Hi
After modprobe hisax type=35 (Billion HFC PCI) on a Xorcom Rapid ISO I get:
HISAX
Dec 12 16:25:35 localhost kernel: HiSax: Linux Driver for passive ISDN cards
Dec 12 16:25:35 localhost kernel: HiSax: Version 3.5 (module)
Dec 12 16:25:35 localhost kernel: HiSax: Layer1 Revision 1.1.4.1
Dec 12
Hello,
Right now there is not a table build script at:
http://www.voip-info.org/wiki-Asterisk+RealTime+IAX
Therefore I have taken the SIP build script and added
a few fields that I use from my iax.conf (could be
more out there, please see the complete build script
below):
`dbsecret`
Does this phone have LEDs showing lines in-use?
Thanx!
Gary P.
Tracy R Reed wrote:
On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly:
I have been looking to see if this type of phone can be implimented in
*. I have found nothing conclusive. Is any out
On 13 Dec 2004 at 16:38, Rick Green wrote:
Asterisk'd ones are different from yours. Since the sources were
retrieved successfully, I don't suspect a problem with the different
cvs. The kernel-source versions are so similar, I suspect a typo on
your part? Mine is the current version from
On December 14, 2004 08:06 am, Doug Reid - Stormcorp wrote:
Use ISDN lines!
Blow it out your ear.
I have a PRI to the telco, and a PRI to my KSU. Echo still exists on some
calls. ISDN just means that *YOU* won't generate echo since there is no
hybrid. It's still very possible to get
Not being an asterisk expert, but having been around
the block once or twice when it comes to data and the
like, I have made some observations based on the examples
given on voip-info.org Sip configs.
it appears there is an adjustment to be made in
the sip_buddies example table:
name
Although set
Greetings All,
I've created a sample trouble ticketing management script in perl that
allows the management of trouble tickets, call routing, etc that
interfaces with the perldesk helpdesk application.
It is mainly for an example of what integration between asterisk and real
world apps is
What handsets are you using? Could be the firmware!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ian Chilton
Sent: Tuesday, December 14, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP registrations not staying registered
Hi,
I have
Would anyone care to offer opinions as to the FXO interface which sucks
the least :) I have an application in which it appears I must route
certain calls out an analog PSTN line. Presently, I am testing an
SPA-3000, but I can't seem to get the echo heard on the IP end of the
call down to a
Greetings to all Users at the List,
sorry for my not so perfekt english (german ist my favorite;))
I have Asterisk 1.0.3 (latest stables Release) and chan_capi 0.3.5 (latest
stable AFAIK), for the Connection to my Telco i Use an AVM B1 Aktive ISDN
Controller, my Linux Box runs with Suse Linux
thanks, Don.
I got past that hurdle last night. I've been using the asterisk-update.sh
script, in conjunction with reading the asterisk doc project book, and the
quickstart guide on onlamp.com.
While trying to simply find the zttool.c source and figure out how to
compile it separately, I
On December 14, 2004 09:00 am, Roy Sigurd Karlsbakk wrote:
a) will asterisk ever support silence suppression?
I believe this is an eventual goal, yes. Right now asterisk gets its SIP
timing from the data stream itself so it is currently not possible to support
it.
b) when getting these, Dec
On Wed, Dec 15, 2004 at 12:24:14AM -0200, Fabr?cio Zimmerer Murta wrote:
I'm thinking in sending a mail for asking WHY THE HELL they can't support
bare modems, even if they have voice support (I have an USR w/ voice, 56k
and ISA kind, and I simply can't use it for testing an * box).
Consider
On Mon, 13 Dec 2004 22:29:16 -0800 (PST), Gianni Veloce wrote:
Hi all,
I plan to install Asterisk at home and would like to
ask some question re firewalls(perhaps it sounds
stupid for experts, sorry
.)
I plan to connect Asterisk box to a ADSL line.
What Router/Firewall system to buy?
I think I
Eldon Balzer wrote:
snip
3. I am getting the following message a lot of the time on the * console:
ERROR[294928]: callerid.c:192 callerid_feed: fsk_serie made mylen 0 (-15)
WARNING[294928]: chan_zap.c:4657 ss_thread: CallerID feed failed: Success
WARNING[294928]: chan_zap.c:4699 ss_thread:
On Tue, 14 Dec 2004 10:48:01 -0500, Dorn Hetzel wrote:
Would anyone care to offer opinions as to the FXO interface which sucks
the least :) I have an application in which it appears I must route
certain calls out an analog PSTN line. Presently, I am testing an
SPA-3000, but I can't seem to
On Tue, Dec 14, 2004 at 06:44:04PM +, Jean-Michel Hiver wrote:
Dorn Hetzel wrote:
Would anyone care to offer opinions as to the FXO interface which sucks
the least :)
So far, for me, using VoIP - PSTN termination provider has been the
solution which sucked the least.
My FXO card
Ian,
When are the registrations failing? Rather what happened before
they started to not work?
A vague overview of SIP follows:
If you restarted Asterisk then it will take a while for the
phones to re-establish their registration as it is the phones
responsibility to
Jean-Michel Hiver wrote:
Hi List,
I've got * randomly hanging up on inbound or outbound calls on zap
channels. I use a Digitnetworks X100P clone card. Any idea of what might
be happening?
This problems is usually caused by callprogress=yes in
/etc/asterisk/zapata.conf or busydetect=yes. You
Jason Goecke wrote:
While most of IAX peers are working for
inbound/outbound, Voicepulse is not (requires the
'qualify' field set to yes). Not sure if this is a
problem on my side (although it works from the
iax.conf settings).
qualify= and mailbox= do not work with the realtime configuration
In my search for a better analog FXO interface, I'm looking into the
Voicetronix cards. Can someone let me know if they've been using the
OpenSwitch cards with Asterisk? Also, can anybody comment on the
differences in FXO performance between the OpenCall and OpenSwitch cards?
Thanks in advance,
Is there a cli timeout at all ? I normally leave a remote session connected
just to monitor and view what's going on. Yesterday, I went to my console
and found the message
pbx*CLI
Disconnected from Asterisk server
Now, I thought that may I had restarted the * server from somewhere else and
had
After searching the archives, I came acrross a few people mentioning
this, but I never saw anything about what became of it.
Has anyone tried to make a virtual modem that could be directly handled
by astrisk, I saw a while ago that someone was going to try and make one
using the same DSP
It seems that this is now fixed!
Looks like it was the NAT Keep Alive setting which needed to be set to
yes in my case.
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Me [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 10:45 PM
Something like this.. :-)
reg72 = wctdm_getreg(wc, card, 72);
/* Negative Voltage */
if (reg72 6) {
wctdm_setreg(wc, card, 72 , reg72 ^ 0x40);
wait_just_a_bit(HZ/10);
wctdm_setreg(wc, card, 72 , reg72 0x3F);
}
/* Positive Voltage */
Evening All,
I was wondering how I would go about enabling the usual # + ext transfer
on a call requiring user input followed by a #.
For example, when I say ring a bank, they ask me for my account number,
I key it in and then press the # key. Of course this doesn't work if I
have the T option
On December 13, 2004 03:10 am, Soren Rathje wrote:
wait_just_a_bit(HZ/10);
I didn't want to wait inside the driver, likely a place where interrupts are
disabled...
-A.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
If I have inbound SIP calls arriving from a provider's gateway to an
asterisk server on my LAN, which then routes the call back out via the
provider's gateway to a PSTN number, once the call is answered do all
the voice packets pass through my asterisk PBX, or is SIP intelligent
enough to
Hi all,
this is my actuel setup
[SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900]
Linux CentOS 3.3 (2.4.21-20.EL.c0)
asterisk-1.0.1
asterisk-oh323-0.6.3b
openh323_1.12.2
pwlib_1.5.2
Calling from SIPphone to the extension 8900 works always.
Calling from 8900 to SIPphone works only
hi
is it possible to have asterisk connect to mysql with a
username/password in some config file and then, afterwards, just use a
global handle to the db? I don't see the point of connecting every time
I need to query it ...
roy
___
Asterisk-Users
Hello,
I am trying to setup an asterisk to store users datails in a mysql
database. Explained here
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers
All works well, I create a user in the database, but asterisk seems that
can't read the data, I create a mysql user with enough
This is not possible...
Hi
When I register a SIP or IAX client to asterisk and I dial to it from
another UA then there is no problem at all
But, when I register two or more clients to the SAME peer (with the same
user/pass) and I call to this peer.. Then only the UA which registered
the last
Hmmm that's bad...
This is the last issue I have which makes that I can't get rid of the
SER proxy in front of asterisk.. Want to get rid of it
Are there any plans to change this design?? (that multiple UA's can
register to one peer?)
Niels
-Original Message-
From: [EMAIL PROTECTED]
Roy Sigurd Karlsbakk wrote:
is it, or can it be possible to transfer stuff like HANGUPCAUSE or
RDNIS over IAX2? This is really a nessicity for multi-server setups to
become any good...
There is a patch floating around (on the mailing list and/or on the bug
tracker) that transports the HANGUPCAUSE
Same here. I've deleted and re-installed asterisk a few times and the
RealTime voicemail never works. The best I've gotten is the MySQL query to
execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
libpri asterisk asterisk-addons asterisk-sounds to download the latest
version
Yeah, I have been changing that line each time I re-install/re-configure
it but it never seems to work. My mysql.sock files is at
/var/lib/mysql/mysql.sock
The best I've gotten it to work so far is trying to get the voicemail
password from the Mysql table with the wrong context. For some
Is there a way to keep the incoming CallerID from the PSTN
and pass it onto the sip phone receiving the supervised call transfer?
The receptionist receives the PSTN callerID, performs a
supervised transfer, we get her local SIP callerID, not the original callers.
The main reason we
Hi,
I have problems with Park + Music on Hold. Sometimes is
working and the caller parked has moh and 95% of the time the moh doesnt
work L
Any suggestion is more than welcome.
Best regards,
Chris HARIGA
smime.p7s
Description: S/MIME cryptographic signature
Jim Van Meggelen wrote:
OK, look, you _might_ be able to free up enough IRQs on a PIC-based
motherboard -- if you disable the serial ports, mouse, parallel port and
USB. It's not recommended, but it's theoretically possible.
And if you have a MoBo that is APIC-compliant, you should be able to
Hi all,
I´d like to know if it is possible, in asterisk, to
modify the configuration setup from english language
to portuguese (Brazil) one.
If so, how can I modify it,ie which files do I need to
modify to get this setup working?
Thanks in advance
and best regards
Guild Jackson
At 09:59 AM 12/13/04, you wrote:
Get newest CVS. Its in there. Trust me. Oh..be sure your getting
asterisk-addons.
-Matthew
Got it thanks
Greg
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
On 13 Dec 2004 at 12:44, Rick Green wrote:
I am trying to do my first asterisk install on a SuSE 9.1 box, using
the asterisk-update script mentioned a few days ago on this list.
I did read the 'quickstart' document on onlamp.com, and made sure
the
following packages were installed
4) All of the preceding notwithstanding, I suspect that the real issue
has nothing (or little) to do with interrupt load, but, given that the
card uses CPU cycles rather than a DSP, the problem is more likley CPU
overload from data handling, which in turn, causes missed interrupts.
I
Hello
I would like to perform some Internet telephony Service Programming
using SIP CGI or CPL. Does anyone have an idea how integrate a SIP CGI
script for example with asterisk? I want the script to be invoked upon
the arrival of a INVIITE messages for example
thanks
moe smadi
You could try adding:
Dial(SIP/SIP/[EMAIL PROTECTED])
On Monday 13 December 2004 03:47 pm, Tom Ivar Helbekkmo wrote:
[EMAIL PROTECTED] writes:
This is the last issue I have which makes that I can't get rid of the
SER proxy in front of asterisk.. Want to get rid of it
Out of
Matt just for fyi incase you didn't know. I don't know what version of IOS
he is running, but on release 12.1 and 12.2 there were issues with one-way
audio on AS5300 and AS5800's if they were using certain combinations of
h.323 and SIP. You can probably search the cisco website for more info.
+++ Jay Austad [13/12/04 03:49 -0600]:
Anyone have an easy fix for making my music on hold to work properly?
It's very loud and has a lot of garbling in it. X is not running, and
the framebuffer is disabled.
I've tried just about every example I could find. I just uploaded
standard
Can anyone give me some recommendations for IP phones that work well with
Asterisk?
I'm hoping for something not much more then $100 bux or so.
Also does vonage service work directly through Asterisk or would I have to use
their hardware? Or are there any other suggestions for a VoIP
provider?
Hi,
I have been looking on that unit to be used as source of fxs ports.
Now I am not sure how I can get * box talking to it?
Thanks for advice.
robert___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Just so I understand the data structure
and what goes in
Static configuration is where you can store regular *.conf files into the
database. These configurations are read at Asterisk startup/reload. Some
modules may also re-read this info upon their own reload (Ex. sip reload).
The table
Hi,
I've recently got * working (thanks Clive and list!) at home. We have
2 PSTN lines
connected via X100P cards. I've got 3 x SIP phones (2 are Budgetone,
the other is
a Tecom SIP).
One of the lines is our standard home line, the other a business line. Presently
I've got * set so you dial 91tel
Cool things to do for home/small business use...
1. Bring up voicemail on your extensions.
2. Get a US phone number free from ipkall.com. Get some more free numbers
from various places. Maybe even pay for one or two. Call forward your POTS
lines on busy, so that people can be sent to voicemail
The company I work for is looking at vendors for a PBX, one of the
requirements is VoIP. I have been sitting there listening to people
pitch very proprietary implementations of VoIP where you are locked in
to their hardware, their interface...
I know a little bit about asterisk (set up a couple
Hi,
As you may have noticed, I've been playing around with a couple of ways
of displaying peer statistics in Windows.
There were previously 4 programs to do that.
I have now combined them into 1 uberprogram (392Kb uncompressed).
You can download the beta from:
Is there any way to transfer a call from host to host and keep the call's
variables intact? -- specifically, UNIQUE_ID and user created variables
like CARD_NUMBER, EXPIRATION_DATE, and CVV2?
Thanks in advance,
Steve
The small x means that the phone is not registered. Make sure you
have the correct secret. Look at asterisk cli debug output. sip debug.
Look for the response from asterisk. is it a 401 or 404?
You can also try setting host=dynamic.
and before all this check the basic networking stuff.
On Tue,
I've used them for a couple of months. My usage is very small, but I'm
really impressed. Especially compared to VoicePulse.
With Sixtel, when you call tech support, you get to talk to a person. That
person actually knows what they are doing. With VoicePulse, I could never
talk to a person and
Ref: Message: 10
Date: Mon, 13 Dec 2004 08:31:04 -0700
From: Damon Estep [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pitching Asterisk
http://www.millenigence.com/articles/asterisk-non-technical-review.pdf
Be careful about the last few paragraphs of this PDF, if possible, before
this paper is
NuFone.Net has 800 toll free IAX termination. They have good quality
of calls once everything is setup and running.
I'd recommend starting with a low dollar amount into your account
first, until you have everything working. I went through about 20
e-mails back and forth via their request
- Original Message -
From: Amer Nasir [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 12:48 AM
Subject: Re: [Asterisk-Users] Repost: Cisco 7960 and Asterisk...not
working
The small x means that the
On Mon, 13 Dec 2004 16:57:12 -0800 (PST), Steve Edwards wrote:
On Mon, 13 Dec 2004, Christopher L. Wade wrote:
Mark Halverson wrote:
Sorry if this is the wrong list...
I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)
Only tried it on X-lite, SIP, with ulaw and alaw.
On Dec 13, 2004, at 3:57 AM, Wilson Pickett wrote:
It's very loud and has a lot of garbling in it.
What/how many phones have you tried it on? What channels
(ZAP/SIP/IAX2) and what codecs?
___
Bob,
Have you looked at any of the products by Zyxel? With QOS, VPN wireless
support they have:
For ADSL: Prestige 652HW
Firewall/Router: Zywall 10W 30W
I'll be honest, I havn't used any of these yet. We were looking for similar
products to suuport our VOIP installs. We just ordered some
Does the built in skinny driver in Asterisk work yet with this phone?
All of the info I've seen tells me to use a different sccp driver for
the phone.
Anyone have any configs for using the built in driver?
~jay
___
Asterisk-Users mailing list
[EMAIL
I have used the 7912 and been pleased. I selected the 7912 over the 7905
because of the 100Mbit Ethernet switch. As I recall the 05 only supports a
10 Mb, as these were going in-line with existing PCs, I felt the 100 FX was
a better deal.
The Firmware upgrade is totally different than the
Go with the POLYCOM. I ve used these phones for a bit now and they are
the best VOIP phones I have used. (refering to the IP500 and IP600)
Just so long as your reseller is a certified Polycom VOIP reseller you
will be fine.
FTP the SIP loads to your phones adn zooom you're off.
either way
Good
Hi !
I would like to know how to choose SIP IAX login for customer account.
We will provide them DID, the good idea will be to said, login=phone number.
But in fact not, since a user can get call from different DID and may change
one day is phone number.
Also some user may not have a DID but a
ok why do you have the s extension commented out.
Try this, 1113 is a sip phone connected directly to asterisk. You can
fwd the incoming calls to a phone or let an auto attendant answer the
calls.
It really depends on how you want to handle incoming calls.
[incoming]
exten =
My company has started development on a Ethernet based channel bank.
Here are the (current) spec's
- 10/100 Ethernet Port
- Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired)
- Serial Console
- TDMoE
- IAX2
- EETP (A protocol that we have designed for IP Telephony)
We
Curious here,
What does that mean. Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher Dobbs
Sent: Monday, December 13, 2004 8:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Ethernet Channel
Ive been using it too and its working great. Still waiting for my DID
but as far as terminating to the US I am very impressed with sound
quality.
Lethol
On Mon, 13 Dec 2004 16:51:47 -0800 (PST), Steve Edwards
[EMAIL PROTECTED] wrote:
I've used them for a couple of months. My usage is very
Can you show me the simple example of this in asterisk words?
Thanks
On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
I would like to be able to dial in to my asterisk box.
Dial extension which would call two other people using the Sip channels.
We would like to be able to talk to
Never had a problem like that ionstalling asterisk on suse. maybe its
the cvs version try using 1.0.1, or 1.02
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[EMAIL PROTECTED] writes:
Because else people will complain that they can't register two
softphones anymore with same user/pass (because only one of the two
softphones can receive the incoming calls) :-)
That's not what I meant -- that bit was clear. I was wondering why it
is important to
Sorry, I misread that you wanted to connect 4 telephones and not for
telco lines. Obviously for 4 lines you will need a tdm04B opposite to
a tdm40b )FXO) which would be for 4 telephones (FXS).
On Mon, 13 Dec 2004 13:50:57 +0100, Wilson Pickett
[EMAIL PROTECTED] wrote:
Hopefully if you are
Ok, I believe the misunderstanding involves the use of CVS itself. When
I talk about CVS I am referring to using the CVS method of downloading
Asterisk vice FTP'ing a GZ file. I was not aware that you were referring to
a version named CVS. Are there any others besides CVS and STABLE.
When
I am trying to do my first asterisk install on a SuSE 9.1 box, using the
asterisk-update script mentioned a few days ago on this list.
I did read the 'quickstart' document on onlamp.com, and made sure the
following packages were installed via yast:
bison, cvs, gcc, kernel-source,
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