Hi
Am setting up * with R2/MfC support but am 90% done. I seem to be missing
something in my setup. Can you tell me what Linux distribution and the
packages you have used to complete your setup to a working level ?
/Sam
- Original Message -
From: "Miguel Cavazos" <[EMAIL PROTECTED]>
To:
Sorry
It works
Just had to reboot the phone
On Thu, 2005-01-13 at 08:40, Altus Snyman wrote:
> Good day all
> I got my snom 220 phone so that it displays on the buttons if someone is
> calling that extension
> I just added "exten => 403,hint,SIP/403" in my dialplan
> But
> These lights only comes
Paul Fielding wrote:
- Original Message - From: "Ronald Wiplinger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grandstream Bugetone 101 & mwi
I tried to use message waiting indicat
I have provisioned with iaxy.conf:
;
; IAXY Provisioning description
;
dhcp
codec: ulaw
server: 61.220.xx.xx
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623] ; IAXy
type=friend
host=dynamic
accountcode=aaabbb
disallow=all
allow=ulaw
secret=cccddd
callerid="IAXy at ELMIT" <623>
trunk=n
Good day all
I got my snom 220 phone so that it displays on the buttons if someone is
calling that extension
I just added "exten => 403,hint,SIP/403" in my dialplan
But
These lights only comes on if someone calls that extension,not if that
extension is busy are a call is made from that extension
Ca
hello there,
any one who used ASTCC in a real enviroment, or has successfully handled
above 1k simultanous calls. need some evalution of ASTCC. if any one has
such an experience please share it with the rest
thank you
Atif
___
Asterisk-Users mailing li
you need to set 'mailbox=extention' in the sip phone's context in
sip.conf
Paul
- Original Message -
From: "Ronald Wiplinger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grand
I saw that news but couldn't seem to find any price/availability info
yet.
I wonder how well they will work if everyone in the home or office has
one. I always liked the old-fashioned approach where switches were used
to explicitly select a channel. My guess would be that multiple wifi
sip pho
On Wed, Jan 12, 2005 at 11:06:18PM +0100, Remco Barende wrote:
> Hi List!
>
> Have a weird problem with ISDN in The Netherlands.
>
> The line that is coming in from the telco is 2 wires. The line is
> connected to an NT1 using the middle pair of a UTP connector. So far sop
> good.
The incommin
I tried to use message waiting indicator, by "Subscribe for MWI" in the
web menu of the phone.
However, it does not light up / flash, even if a voice mail is waiting.
Where is the switch to turn it to?
bye
Ronald
___
Asterisk-Users mailing list
Asterisk
Wired USB handset - quite plasticky looking/feeling really (no display)!
AND it is supplied with a software registration process which is a PITA
because each time you move the handset to a different computer you need
to unregister it from the old PC and re-register on the new PC.
I complained to th
I think some people are missing the point. You can't throw your cordless
phone in your pocket, go to your office, hotel or buddie's house, turn it on
and get a signal. You can with a WiFi phone, however
- Original Message -
From: "Kim Lux" <[EMAIL PROTECTED]>
To: "Asterisk Us
I don't know why people keep making the statement about range with wifi
versus a cordless phone. I can easily get a good wifi signal when I'm
over at my neighbors but can't get reception with our cordless (2.4GHz)
phone. (Both receivers are at home...) It seems to me that the wifi
range is at l
Assumptions:
-Working DHCP server
-Good LAN, everyone's happy and can see each other
-Working DNS server on LAN
-IP of Asterisk server: 192.168.1.46
sip.conf: (remove comments)
[550] 'extension number
callerid="Joe Blow" <550>
canreinvite=no
context=from-internal 'default AMP context, salt to t
Uniden and Vtech both just announced cordless phones with SIP
ATAs built into the base station.
You get better range and battery life compared to a WiFi
phone.
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Kim
Lux
To: Asterisk Users Mailing List -
On Wed, Jan 12, 2005 at 08:07:11AM -0600, Matt Schulte arranged a set of bits
into the following:
> Ok, we're trying to use Asterisk as a PBX in our office. Our original
> plan was to use a Cisco 7960 with a 7914 attached. Short story is, no
> one updated chan_sccp in a long time and the 7914 is q
asterisk v1.0.3, mpg123 v59r, shoutcast server.
When first starting asterisk all is fine, moh/mpg
processes start, can see asterisk client connections on shoutcast
monitor as well and I've got mp3 streamed music on hold, cool!
After aprx. 32-105 seconds the asterisk client connections close on th
An unflattering zyxel review:
http://slacker.com/~nugget/asterisk3.php
I can't help but think my questions are out of place on this list... I'm
asking questions about SIP phones and everyone else is talking about
asterisk. Sorry.
On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote:
> My wife wa
hi
1 does asterisk do full proxy to all calls, or it can be configured
to do proxy signal only ? if yes, how to configure to proxy signal
only?
2 G.729 codec license: do i have to buy license if * doesn't do codec
conversion, just proxy the calls?
Jeffrey
My advice, not having used a WiSIP phone, is to use an FXS port and plug a
normal cordless phone in. It'll save you all potential problems, and if you
don't like the phone, you can just plug any other one on the market in.
In a home environment, I'm not quite sure I see the benefit of wifi. Esp.
How do I setup IAX peering between two Asterisk servers? I found a few
examples for the IAX client side that conects to a service provider. But
what does the service provider end look like. I would also like to use md5
authentication.
Adi
___
Asterisk-Us
On Wed, 12 Jan 2005, Kai-Uwe Jensen wrote:
> On Wed, 12 Jan 2005, Paul Rodan wrote:
>
> >> Yeah, it's a way for numbers to get sent faster, so you don't have to
> wait
> >> for the 3 second timeout before it gets transmitted to Asterisk. It's
> >> similar to the dial-plan in the Sipura devices.
My wife wants a cordless phone for around the house. We are going to be
using VOIP exclusively very shortly. Our current cordless phone is aged
and on the verge of replacement. The other phone we are going to use is
a SIP Budgetone.
Should I buy a SIP to POTS converter and a new cordless phone
On January 10, 2005 03:09 pm, Adi Linden wrote:
> How can I setup Asterisk to place calls if the same dial pattern can be
> routed through several PRI gateways. I have one way that I tried:
>
> So what happens is that if all channels on 172.17.99.5 are in use calls
> are routed to 172.17.99.6 and i
Hi,
I just tried this using a couple of iax peers and it works quite well. But
I did have to alter my dialplan. In my iax.conf I added 'qualify=5000' for
the nufone and voipjet peers.
[macro-gw-voipjet]
;
; This is the VoipJet IAX peer
;
; Use: Macro(gw-voipjet,${EXTEN}))
; Requires 'qualify=' st
It is a USB attached phone
It needs to be used with a soft phone
It does work as a handset for x-ten type soft phone or their own softphone
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kim Lux
Sent: Wednesday, January 12, 2005 9:09 PM
To: Asteris
I just realized that there is nothing wireless about this phone. I
think it is just a wired USB phone that looks like a wireless phone that
will be wireless is your laptop is wireless.
I think the picture of the guy holding the phone in his hand vaguely
shows a cord in his palm.
On Wed, 200
On Thu, 2005-01-13 at 11:51, [EMAIL PROTECTED] wrote:
> > >From searching the list archive I have come up with the following list
> >
> > 22 for SSH Should this be TCP, UDP or Both?
> TCP
> > 5060TCP Only
No, UDP
> > 1 -2 UDP Only
This last is for RTP and can be anythin
On Thu, 2005-01-13 at 12:38, Adam Goryachev wrote:
> On Thu, 2005-01-13 at 11:24 +1100, Howard Lowndes wrote:
> > On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
> > > On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> > > > I have a situation where I need to know which Zap channel an in
There is a mailing list dedicated to AMP that would be better suited to your
line of questions.
To answer your question, you can create your own files and name them
whatever you want and then use include statements. The most you will have
to re-input are the include statements.
- Original Me
http://www.pcphoneline.com/skype
"The VPT1000 is NOT a simple last generation USB phone audio device but
is rather a next generation integrated gateway and USB phoneset with
simultaneous dual mode Skype and SIP calling support. Skype is not
forecast to have "SkypeIn" available until June 2005 bu
> > > > > If nothing else, my efforts are documented for anyone else in the
> > > > > same boat.
> > > > > It seems that you can debug agi by typing agi debug at the *
command
> > > line.
> > > > > Amazing! Here is the output. I am assuming that since astcc-tone
> > didnt
> > > > > play, the pro
On Wed, 12 Jan 2005, Paul Rodan wrote:
>> Yeah, it's a way for numbers to get sent faster, so you don't have to
wait
>> for the 3 second timeout before it gets transmitted to Asterisk. It's
>> similar to the dial-plan in the Sipura devices.
>>
>> I don't know where it's mentioned in their docum
Cool thanks. I got the system running today, and you were right, it was
pretty easy. There are several things I would like to change. What
files can be changed manually without AMP clobbering them?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.a
On Thu, 2005-01-13 at 11:24 +1100, Howard Lowndes wrote:
> On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
> > On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> > > I have a situation where I need to know which Zap channel an incoming
> > > call is on, so that the call can be answered a
Roger Hanson wrote:
I'm using CentOS - which is another Red Hat Enterprise clone, like WBEL
www.centos.org
I've had no problems of any kind with the OS
- Original Message - From: "Imran Sadiq" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 11, 2005 9:58 PM
Subject: [Asterisk-Users] What is
Anyone have a suggestion for a configuration example using Asterisk and SNOM
190 SIP phone?
I have read both sets of documentation and for the life of me, I can't get
the phone to register and work. I can use a IAX softphone and it works
perfectly. It is just the SIP thing I guess. Anyone have
Let me know how we can post message in your mailing list
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Cathey
Sent: Wednesday, January 12, 2005 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Unofficial
How to increase volume in line for music on hold?
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/
> just to make sure:
> when i have zaptel devices on my box and i also use meetme and iax2,
> do i need to have USB device enabled and it's modules loaded?
No
your zaptel device will provide the needed hardware timer
the USB timer hack is for when you don't have any digium card
___
Hi,
I've switched to fresh install and unfortunately changed two things at the
time:
- used fresh AMP install
- upgraded Grandstream bt100 to latest beta from Grandstream web site .18
I have one local IAX2 extension (IAX phone) and one bt 100 extension. I
cannot make calls between those two.
Wit
Well that didn't workI now get this error
Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569/5", "b") in new
stackJan 12 16:56:21 WARNING[4989]: app
On Wed, 12 Jan 2005, Andrei (MPI) wrote:
> Greg Boehnlein wrote:
>
> >Hello,
> > I have a mixture of Polycom SP IP 500 and 300 phones. I have been
> >reading through the administration manual to try and solve this problem,
> >but I do not seem to be able to find the answers to my question.
On Wed, 12 Jan 2005, Paul Rodan wrote:
> Yeah, it's a way for numbers to get sent faster, so you don't have to wait
> for the 3 second timeout before it gets transmitted to Asterisk. It's
> similar to the dial-plan in the Sipura devices.
>
> I don't know where it's mentioned in their documentati
> >From searching the list archive I have come up with the following list
>
> 22 for SSH Should this be TCP, UDP or Both?
TCP
> 5060TCP Only
> 1 -2 UDP Only
>
> Is this info correct or is there other ports or port type corrections
> above?
Yes, for SIP it is correct
I installed bristuff0.20-RC3 (which includes * 1.0.3 stable)
It works fine until I disconnect the phone jack for the ISDN line. Even
after plug it back in asterisk still reports that it could not create a
zap channel when I try to dial out and the line gives an engaged tone when
I try to dial.
On Tue, 2005-01-11 at 10:59 -0500, Vitalie Apostu wrote:
> Can you give me example of sip.conf and extention.conf which work with
> broadvoice? I want users who registered with Messenger through sip to be
> able to make a call thought broadvoice.
* for BV setup guide:
http://www.broadvoice.com/su
On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
> On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> > I have a situation where I need to know which Zap channel an incoming
> > call is on, so that the call can be answered appropriately when a SIP
> > phone displays the channel. These Za
I am having trouble building appradius from http://appradius.minitelecom.org/
I configure, make, make install cpprad-1.0, but when I configure, then
make appradius I get :-
obelix:/usr/src/appradius/appradius1.0 # make
make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib'
make[1]: Not
yah sorry dyslexic
this is what you want
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=71505&item=5742009370
&rd=1
and i'd guess this will close $2000-2500 the 1- TNT-SL-CT1 will do 8 t1
but you will need to add an extra 1-APX8-SL-96DSP to handle a full 8 t1
Pri
that should be about 1k
On Thu, 2005-01-13 at 10:40, John Dunham wrote:
> Just checking if anyone has experence with Integrated Networks IN1002 phone.
You might like to try aredfox.com and see if there is anything there
that might suit. I have HOP1002 phones and I am using the "1002" as a
clue here.
> We just got 100 o
On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> I have a situation where I need to know which Zap channel an incoming
> call is on, so that the call can be answered appropriately when a SIP
> phone displays the channel. These Zap calls are coming in over PSTN and
> don't have caller ID.
OK, I'm coming to think linphone is bullshitting me.
I now tried the following call paths
firefly -> * -> iaxcomm works
firefly -> * -> linphone works
sjphone -> * -> iaxcomm works, especially sip->iax works
sjphone -> * -> linphone works
The opposite paths work too except
linphone -> * ->
Scheda wrote:
If anyone knows of a linux applicable IAX softphone,
I'd be more than willing to give it a shot, but I haven't found one so
far.
Have you tried iaxcomm?
http://iaxclient.sourceforge.net/iaxcomm/
--
Cheers,
Matt Riddell
___
http://www.sineapp
Christian Savinovich wrote:
I can't believe you!!, how an incredibly rude person you are. In two
paragraphs you manage to imply that you belong to the group of gurus in this
list (my respects to you, oh major guru), that I don't know LookOut, and you
suggest I should drop learning linux and a
Sam Njenga wrote:
Hi Steve
Did that but still the same error :-(
PS. There is now unicall-0.0.2pre3. What are the changes in it ?
/Sam
pre3 has some bug fixes in the heart if the R2 protocol. The only file
different between pre2 and pre3 is the libmfcr2 tar file. There were a
couple of things
> I'm having an issue when I transfer a call to another SIP extension it sees
> that the sip phone is not there and goes to voicemail but in my case it
> transfers to the main voicemail instead of the users voicemail.
>
> Here is what my SIP extensions look like in the extension.conf file
>
> ext
Matt Riddell wrote:
Steve Underwood wrote:
Matthew Boehm wrote:
I know myself, SS7 will be a make or break for our continued use of
Asterisk.
Our make/break is FoIP support. If Asterisk had some form of T.38 for
reliable fax transmission..or even just T38 pass-thru..
One down, one to go.
Just checking if anyone has experence with Integrated Networks IN1002 phone.
We just got 100 of them in and no manual or passowrd to program the phone.
Also need some direction on the * sip.conf if anyone has experence with
these phones.
Thanks,
John Dunham
___
Steve Underwood wrote:
Matthew Boehm wrote:
I know myself, SS7 will be a make or break for our continued use of
Asterisk.
Our make/break is FoIP support. If Asterisk had some form of T.38 for
reliable fax transmission..or even just T38 pass-thru..
One down, one to go. The T.38 support wil
You can change the setting. I set mine for
every 1 min on a small system. The phones always work.
What is the register
interval in the grandstreams? The qualify=yes should keep the connection alive
as long as Asterisk is up, but if it goes down and then comes back up, the
phone
I'm having an issue when I transfer a call to another SIP extension it sees
that the sip phone is not there and goes to voicemail but in my case it
transfers to the main voicemail instead of the users voicemail.
Here is what my SIP extensions look like in the extension.conf file
exten => 3957,1,D
Did you ever have success copying your configs to the Xorcom box?
Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Monday, January 03, 2005 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-User
Hi,
I'm trying to determine why IAX2 calls are getting dropped after a 4-24 hours
of continuous connect time. My project requires that calls stay up for days at
a time. When I turn on IAX2 debugging, I see "max retries exceeded" for control
frames just before the connection is dropped.
My test
Nevermind... I presume you are refering to the
Lucent (Ascend) MAX TNT WAN Access Switch (correct??)
-- Mike
- Original Message -
From:
Michael
B. Murdock
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 5:23
PM
> > My question is simply, has anyone received a deposit from
> > these people once you return the equipment in good order?
> > I've been unable to contact them now for almost 2 whole months.
>
> Get in line. Refunds are difficult it seems -- best bet is to go
> through the credit card co. I c
I have a situation where I need to know which Zap channel an incoming
call is on, so that the call can be answered appropriately when a SIP
phone displays the channel. These Zap calls are coming in over PSTN and
don't have caller ID.
As far as I can make out my SIP phones (WuChuan HOP-1002) displ
Who makes the TNT-Max ??
-- Mike
- Original Message -
From:
TC
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 5:01
PM
Subject: Re: [Asterisk-Users] What's the
easiest way to get * to call PSTN?
ebayed TNT-M
Hi
ISDN wire:
From phone company you receive on two wire, this is called "U"
interface on this you can connect only one device, normaly the NT1 box.
On the NT1 there is a "S/T" bus that allows several devices (phones)
connected (in "TE" mode)!
Yes S/T is four wire !
HB
__
On Wed, 2005-01-12 at 14:39 -0800, Erik Espinoza wrote:
> Did you enable passthrough for the rtp ports on the asterisk box?
>
> I had the same problem until I enabled udp 1:2 on the firewall.
I did. That's why linphone -> * echo test works.
Maybe I made some progress however, by logging
On Wed, Jan 12, 2005 at 11:59:00AM -0500, Dave Morrow wrote:
> Thanks for the information Dennis, it is much appreciated. I think I am
> going to start from scratch (with AMP) also. It's just a bit of a pain
> is all. Do you have any expertise in regards to keeping current with *
> when new vers
ebayed TNT-Max with SIP for mere 8 pri you might
get that for 3K or better
I know i was offered a 4 pri tnt-max for
us1500
- Original Message -
From:
Michael
B. Murdock
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 1:
Did you enable passthrough for the rtp ports on the asterisk box?
I had the same problem until I enabled udp 1:2 on the firewall.
On Wed, 12 Jan 2005 22:06:18 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote:
> Hi folks
>
> an issue I don't understand. I'm running * stable 1.0.3 on public
>
We used the Nufone Implementation along with the OH323 implementation but, none
perform to a commercial level.
If there is a stable product out there, we would be keen on utilising it.
Quoting Paul Belanger <[EMAIL PROTECTED]>:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> James,
>
> Coul
Well, thank you to Tor for the SetGroup/Checkgroup config. It works
well! (Thanks to John for the contexts/dialplan version, too).
Unfortunately, the phone doesn't audibly ring when the second call is
coming in (just a visual prompt), and you have to press the line
appearance button, and then th
Hi,
I'm trying to have a queue with members that work like this:
Member = Local/[EMAIL PROTECTED]
Member = Local/[EMAIL PROTECTED],1 ; Penalty!
And a dialplan that looks like this:
[context]
Exten = 101,1,DBget(Channel=QM/101)
Exten = 101,2,Dial(${Channel})
Exten = 101,102,Busy
And similar for
Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
I will let anyone
Which MOH patch should I use? What is the best link to the best mpg123
replacement patch? Something with instructions please.
I found the original MOH patch, but then I remember somebody mentioning
another good patch that included the MOH patch within it, but I can't find
that post/link, anybody?
On Wed, 2005-01-12 at 15:50 -0500, Paul Rodan wrote:
> Hey,
>
>
>
> One of my remote Asterisk servers which has been up and running for a
> couple of months now suddenly stopped. I didn’t realize it because my
> monitoring system only pings the machine. But my remote office
> complained, I che
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
James,
Could you point me to the right source for some configuration settings?
~ I have searched http://voip-info.org, but could not see anything.
PB
James W. Coberly wrote:
| We have a stabilized product for H323 to SIP conversion using *. We
| have t
qualify is not set in sip.conf at all. What
should the value be, or should it just be set to yes?
The register interval is 60 minutes. The
Asterisk server is not going down, but the connection between the phone and the
server might go down for a few minutes, and when it comes back up th
hello all,
I have been lurking here for a while learning what I can, and reading
quite a bit, using the fine reference at asteriskdocs.org I have been
building my test server on an older compaq dl380, PIII-700, 1.2gig mem
[X not running].
My problems begin when trying to start * for the first tim
Hi List!
Have a weird problem with ISDN in The Netherlands.
The line that is coming in from the telco is 2 wires. The line is
connected to an NT1 using the middle pair of a UTP connector. So far sop
good.
However, the outgoing ports on the NT1, should they be wired with 2 wires
or 4 (2 pairs)?
We have a stabilized product for H323 to SIP conversion using *. We
have tested it up to 500 lines per box without many troubles using G729.
Depending upon the hardware selected, it should scale into the DS3+
range.
James-
On Wed, 2005-01-12 at 13:34 -0800, William Boehlke wrote:
> Yes, it is.
On Wed, 12 Jan 2005, Asterisk wrote:
> Sometimes things are so obvious that you miss them. "just view a single LF
> as the field separator and a double LF as the record separator" is, of
> course, the point that makes me look soo stupid.
Note that the order of the elements is only defined by
What is the register interval in the
grandstreams? The qualify=yes should keep the connection alive as long as
Asterisk is up, but if it goes down and then comes back up, the phone has to
re-register with Asterisk before asterisk can keep the connection alive.
From:
[EMAI
On Wed, 12 Jan 2005, Alex G Robertson wrote:
> If I got the matter, unstructured framing is used for Data (2M full) and
> structured for "64k circuits".
You can run data over a channelized link, or even over pri. There are lots
of flexibility in how an E1/T1 can be configured. For voice there a
i'll join the list. my DID is 716, but i continuously have issues with
BV..the good thing is i've kinda learned a lot about how they are
setup. The bad thing is i'm still plagued with horrible jitter/warble,
downtime, and dial-out capability..
On Wed, 12 Jan 2005 09:44:59 -0500, Mike Cathey
<[EM
That's probably a timeout problem in the nat
box.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
NortonSent: Wednesday, January 12, 2005 6:44 PMTo:
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] Cant receive calls after network g
On Wed, 12 Jan 2005, Matthew Boehm wrote:
> ahh..american arrogance. I assumed you were in the US.
> We pay $2000 a month for DS3/SS7 to national carrier. We will soon be
> dropping the SS7 and turning that voice DS3 into a bandwidth DS3. We will
> still use the carrier but all calls will terminat
Hi,
I have several Grandstream phones
connected to Asterisk, some behind NAT and others not. If I reboot all the
phones, everything is fine. Should the connection go down, and then come back
again, those behind a NAT are still able to make calls, but are unable to
receive calls.
Greetings all-
For whatever reason of personal insanity, I've decided to start an
Asterisk bookclub. Basically, we'll pick three books every month (a
users book, a developers book, and another general interest book) and
then read and discuss on IRC in the #asterisk-bookclub channel.
The users
> My question is simply, has anyone received a deposit from
> these people once you return the equipment in good order?
> I've been unable to contact them now for almost 2 whole months.
Get in line. Refunds are difficult it seems -- best bet is to go
through the credit card co. I cancelled a l
Yes, it is. Ugly but possible.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Belanger
Sent: Wednesday, January 12, 2005 1:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?
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We are also looking for a high density
SIP<->TDM gateway (signalling & media) as an alternative to putting
the ISDN PRI cards in the * box. Ideally it should support up to 8 ISDN Pri's
with NFAS on the TDM side and 100baseT/1000baseT on the IP side.
Has anyone had experience with this type
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Hello,
I have been successful in getting Digest authentication to work with my
Mitel 5055 IP Phones, however I'm wondering if Asterisk still supports
Simple authentication? I know it has been depreciated in the RFC, but I
have some phones with don't su
Hi folks
an issue I don't understand. I'm running * stable 1.0.3 on public
internet, with following iax.conf / sip.conf entries:
iax.conf
[100]
type=friend
username=Foo
context=default
auth=md5,plaintext,rsa
secret=secret
host=dynamic
callerid="Foo" <100>
qualify=no
sip.conf
[10]
ty
You can use RealTime to store the mgcp.conf file. This does not get you
"realtime" abilities as you still need to reload mgcp when u make a change.
Matthew
- Original Message -
From: "Michael Baird" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: W
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Hello all,
I was looking for some information about using Asterisk to convert an
incoming H.323 call to and outgoing SIP call. Is this possible?
PB
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Hi All
I am setting up * for use as a voicemail. I have discovered that if I dial the
phone system and send "#+Extension+messagenumber" (dtmf) that the "msg" light
will
come on on the phones, if 00 messages the light will go off. they are an old
tie
onyx vs system. So how can I get asterisk t
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