Problem is that your mailboxes are in a different context than you are
checking.
Add @mb to your extension (mailbox) when checking.
-
>>[mb]
>>221 => 221,221,[EMAIL PROTECTED]
>>299 => 299,299,[EMAIL PROTECTED]
>>326 => 326,326,[EMAIL PROTECTED]
___
On Feb 28, 2006, at 9:51 PM, Andres wrote:
Ed Greenberg wrote:
I need to set up an office full of Cisco 7960 phones behind NAT with
the server out in Colo.
The first test phone registers fine, but the second one does not
register.
The first phone's registration looks like so:
/SIP/Registry
> Thanks for this example - it has really got me started!
>
> Short question - how can I put a variable into my perl script?
>
> I imagine it's something like
> exten => 780,1,AGI(agi_ret_val2.pl|${back})
>
> But how can I get my perl script to pick this value up?
>
> Again - thanks to everyone
On Tue, Feb 28, 2006 at 04:21:57PM +0100, Armin Schindler wrote:
> On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
> > On Tue, Feb 28, 2006 at 02:32:27PM +0100, Armin Schindler wrote:
> > > On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
> > > > Maybe chan_capi should have a timeout waiting for more
Hi All
I am working on voicemail , mailbox , after
reading documents,
I had setup of three users for mailbox
to make things simpler , I had kept the
user name and passord same for all the sip users, Now
I am able to record the message and I do get voicemail
in my email ,
Hi!
I'm looking for someone who has successfuly setup an asterisk in
austria with isdn in p2p mode and chan_capi.
There is is a special problem in austria with DID. If someone is
dialing the phone number of the asterisk pbx like 12345-0, zero is
passed as an DID, but in Austria u can dial 12345,
On 3/1/06, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
> Did you try the development environment:
>
> http://mirror.astlinux.org/astlinux-devel.tar.bz2
Kristian, this means I could create a bristuff'ed version of astlinux
by using this one? Cool!
cheers
On 3/1/06, mustardman29 <[EMAIL PROTECTED]> wrote:
> I am aware of Astlinux and the other embedded Asterisk solutions out there?
> Astlinux is nice but the problem is that when I hit a snag and need to
> incorporate a patch and what not I cannot do that with Astlinux because I
> cannot compile my o
Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a
VSP over a IAX2 trunk however we have recently purchased and connected a
TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make
outgoing calls via it fine, however incoming calls are droppe
Ed Greenberg wrote:
I need to set up an office full of Cisco 7960 phones behind NAT with
the server out in Colo.
The first test phone registers fine, but the second one does not
register.
The first phone's registration looks like so:
/SIP/Registry/3115552368
:64.169.xx.yyy:38836:3600:
I have about 10 cisco 7960/40 behind a Nat router no problem.
They work with all features.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Ed Greenberg
> Sent: Tuesday, February 28, 2006 5:30 PM
> To: Asterisk Users Mailing List - Non-Commerci
Perfection!
PaulH
On Tue, 2006-02-28 at 22:24 -0700, Darren Wiebe wrote:
> my ( $var1, $var2, $var3 ) =
> @ARGV;
>
> and so on and so forth.
>
> Good Luck
>
> Darren Wiebe
> [EMAIL PROTECTED]
>
> Paul Hales wrote:
>
> >Thanks for this example - it has really got me started!
> >
> >Short
my ( $var1, $var2, $var3 ) =
@ARGV;
and so on and so forth.
Good Luck
Darren Wiebe
[EMAIL PROTECTED]
Paul Hales wrote:
Thanks for this example - it has really got me started!
Short question - how can I put a variable into my perl script?
I imagine it's something like
exten => 780,1,AGI
The only way to do it 'automatically' would be to installing some dial-
out software.like vicidial.
PaulH
On Tue, 2006-02-28 at 13:47 +0100, nik600 wrote:
> hi
>
> i'm migrating a callcenter to asterisk, inbound calls, queue monitorig
> is ok, but how can i monitot outgoing calls?
>
>
On Feb 28, 2006, at 2:35 PM, [EMAIL PROTECTED] wrote:
On Tue, 28 Feb 2006, Cory Andrews wrote:
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other Zoom products
http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
I just found out,
Thanks for this example - it has really got me started!
Short question - how can I put a variable into my perl script?
I imagine it's something like
exten => 780,1,AGI(agi_ret_val2.pl|${back})
But how can I get my perl script to pick this value up?
Again - thanks to everyone who has helped me
On Feb 28, 2006, at 6:43 PM, James Harper wrote:
What about the fonebridge (http://www.red-fone.com/fonebridge.html)?
It uses POE, so you could hack something together to supply 48V @ 15W
if
you don't have access to a power point, and it appears to have a 2 port
switch built in so you could
On Feb 28, 2006, at 7:14 PM, Anton Krall wrote:
Anyway the phone can compensate? I don't think it works that way but
worth
asking..
If the phone has an input gain (for phone users voice) then adjusting
it down can help echo that is being generated at the far end. ie if
it's too loud coming
>You'd need to modify the code of app_dial.c for 1.0.9 to make this
>happen. There isn't a way to do it as it is now.
Dang. Thanks for that. Time to put my thinking cap on to do this in a more
unobtrusive manner.
___
--Bandwidth and Colocation provided
If memory serves me correct, Sangoma told me for their Remora A200 cards you
should budget about 100ma per ringing FXS port/phone. The Digium card
should be about the same. I believe that is at 12V via the molex connector
for both cards. The worst case is all ports ringing simultaneously. If
yo
http://www.google.ca/search?hl=en&q=dl380+g4+site%3Alists.digium.com&meta=
-Original Message-
From: Bartosz Supczinski [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 28, 2006 7:58 PM
To: Asterisk-Users; Asterisk-Dev
Subject: [Asterisk-Users] Problem with two cards Digium
Hello,
I`ve
Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped a
I am running Asterisk 1.0.9 and have been running
all my calls through a VSP over a IAX2 trunk however we have recently
purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through
Optus. I can make outgoing calls via it fine, however incoming calls
are dropped after a f
mustardman29 wrote:
I am aware of Astlinux and the other embedded Asterisk solutions out there?
Astlinux is nice but the problem is that when I hit a snag and need to
incorporate a patch and what not I cannot do that with Astlinux because I
cannot compile my own version.
How hard is it to crea
I looked into this 2 years ago when we were deploying our first Polycom
phones. Back then the answer was "no". Echo cancel was only supported
in speakerphone mode. However, I vaguely recall there was some
indication that the feature would be added.
The best solution would be to lower your t
Anyway the phone can compensate? I don't think it works that way but worth
asking..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Eric "ManxPower" Wieling
|Sent: Tuesday, February 28, 2006 7:37 PM
|To: Asterisk Users Mailing List - Non-Commercial
Guys, I think this might help everybody that has polycom phones.
Polycom never answered my email but after some testing I found the settings
to control default volumes after reboot:
For configuring the speakerphone volume after a phone restart (default
volume) you need to set in sip.cfg:
voice.g
hi , when i am compiling Intel G729 codec with Patch, i got this error. but i followed all the instraction and successfully patch those file.i saw some has the same error before in this forum and they solved it. but i cant find the patch file they gave.
can some one help me to fix this error?thanks
Ed Greenberg wrote:
My main problem local servers is that I need to (a) be able to forward
voicemails between users at multiple locations and (b) have all the
message waiting lights working.
Otherwise I'd be doing just that.
Connect the remote office to the main office via OpenVPN.
Doug
Hello,
I`ve got a problem which I can`t deal with. I own 2 cards - TDM2400P and
TE410P. I`ve put them into a HP Proliant DL380 G4 server, compiled the
drivers according to the manual. Unfortunetly there are both cards
channels are configured in zaptel.conf file the first module (in this
file)
I have a 4 PCI slot version. It worked well under Windows, but I could
never get my laptop to see any devices I stuck in it when running under
Linux. I have tried RHEL3 and RHEL4. They list their last supported
Linux version has Redhat 9, so doesn't appear they keep the Linux
version very up
Is the * server also behind a NAT?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ed Greenberg
> Sent: Tuesday, February 28, 2006 7:44 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] A
I've tried it both ways with no noticeable change.
--On Tuesday, February 28, 2006 7:25 PM -0700 Damon Estep
<[EMAIL PROTECTED]> wrote:
Your posted config had nat=1, not nat=yes. Are they interchangeable? I
thought I remembered nat=1 either doing something a little different or
not doing anyt
What about the fonebridge (http://www.red-fone.com/fonebridge.html)?
It uses POE, so you could hack something together to supply 48V @ 15W if
you don't have access to a power point, and it appears to have a 2 port
switch built in so you could still chain to a VoIP phone (or more if you
daisy chain
Did you first dind the provider that will give you a mobile T1?
On 2/28/06, Damon Estep <[EMAIL PROTECTED]> wrote:
> It would be easier to find a laptop with a docking station that has a
> pci slot in it.
>
> Some laptop docks run off of the battery as well.
>
> I can't imagine a real-world situat
Your posted config had nat=1, not nat=yes. Are they interchangeable? I
thought I remembered nat=1 either doing something a little different or
not doing anything at all.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Greenberg
Sent: Tuesday, February
It would be easier to find a laptop with a docking station that has a
pci slot in it.
Some laptop docks run off of the battery as well.
I can't imagine a real-world situation where you would have a T1 and no
power outlet.
Watch out for performance issues on the PCI slot.
-Original Message--
I am using both TDM2406 and TDM2402 and using true 550W power supply. I couldn't find any indication from Digium about power comsuption.Daniel PizarroOn 2/28/06,
Melisa Teoh <[EMAIL PROTECTED]> wrote:
Kevin P. Fleming wrote:>Melisa Teoh wrote:For example, how much power would a server with 3 o
Hi,
I am using asteriskathome 2.5 and planing to
interconnect to a telco switch using MFCR2. I was able
to compile and load the channel on the asterisk. But
one thing I notice is that all of the unicall channels
disappear when I do a reload on the CLI of the
asterisk. And even worst is asterisk so
On Wednesday 01 March 2006 14:15, mustardman29 wrote:
> I am aware of Astlinux and the other embedded Asterisk solutions out there?
> Astlinux is nice but the problem is that when I hit a snag and need to
> incorporate a patch and what not I cannot do that with Astlinux because I
> cannot compile m
On 2/28/06, < Arnaud > <[EMAIL PROTECTED]> wrote:
> What are the options to hook a T1 card up to a laptop running * ? Are
> there USB or PCMCIA T1 cards ?
>
> Has anyone tried a USB to PCI adapter as such :
> http://www.mobl.com/expansion/products/cardbus_expansion/1slot/
> It looks nice but cost 1
Anton Krall wrote:
Guys.
I have about 20 Polycom 301, some 501 and some 600 and I really like the
phones, but I have a question and maybe somebody else has seen this. Seems
sometimes when people talk a bit loud, Polycom phones have a tiny bit of
echo, can this be controled with some kind of gain
Guys.
I have about 20 Polycom 301, some 501 and some 600 and I really like the
phones, but I have a question and maybe somebody else has seen this. Seems
sometimes when people talk a bit loud, Polycom phones have a tiny bit of
echo, can this be controled with some kind of gain or AGC or something
Title: RE: [Asterisk-Users] TDM400P digium card
I’d
suggest testing each set using a virtual “echo test” extension, this
will test your network between sets and asterisk box. If you have choppy
audio during the test, then I would look into the LAN for issues.
-vince
-Original
I am aware of Astlinux and the other embedded Asterisk solutions out there?
Astlinux is nice but the problem is that when I hit a snag and need to
incorporate a patch and what not I cannot do that with Astlinux because I
cannot compile my own version.
How hard is it to create my own version of L
Kevin P. Fleming wrote:
Melisa Teoh wrote:
For example, how much power would a server with 3 of the TDM2400P fully
loaded cards draw?
With what type of modules? The power consumption figures are on the
TDM2400P data sheet and on the Digium website, IIRC.
TDM2451E x 3 (60 extension
What are the options to hook a T1 card up to a laptop running * ? Are
there USB or PCMCIA T1 cards ?
Has anyone tried a USB to PCI adapter as such :
http://www.mobl.com/expansion/products/cardbus_expansion/1slot/
It looks nice but cost 1k
how about the linux drivers ?
the goal is to have a portab
My main problem local servers is that I need to (a) be able to forward
voicemails between users at multiple locations and (b) have all the message
waiting lights working.
Otherwise I'd be doing just that.
Meanwhile I'll check out qualify=yes - I already have nat=yes.
Thanks,
--On Tuesday, F
> I've contact Polycom via their web form and they emailed me back
> around 2
> days later. We've purchased all of our phones via authorized
> resellers.
> Dunno if that makes a difference or not.
>
> My question was about the button programability and their response was
> that the manual was inc
Michael Collins wrote:
> I had never sat down to write an AGI script before - I hadn't needed one
> - but I thought, "How hard can it be?" Ugh. The Asterisk::AGI module
> is very handy, and I highly recommend it. I've only written one AGI
> script in my life (up to now) but I've written 10's of
Doug Lytle wrote:
Ed Greenberg wrote:
I need to set up an office full of Cisco 7960 phones behind NAT with
the server out in Colo.
The first test phone registers fine, but the second one does not
register.
The easiest way to do this is put an Asterisk server on the local side,
have the p
Try nat=yes and qualify=yes in sip.conf.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Greenberg
Sent: Tuesday, February 28, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] A room full of Cisco 7960s
Melisa Teoh wrote:
> For example, how much power would a server with 3 of the TDM2400P fully
> loaded cards draw?
With what type of modules? The power consumption figures are on the
TDM2400P data sheet and on the Digium website, IIRC.
The only significant power draw is the FXS modules, and then o
For example, how much power would a server with 3 of the TDM2400P fully
loaded cards draw?
We're trying to figure out what APC power we should have to achieve
approx 30 minutes back-up time.
Would appreciate any learnings or methods on calculating the power draw
and determining the appropria
On 2/28/06, Colin Anderson <[EMAIL PROTECTED]> wrote:
> Using 1.0.9:
>
> If I have:
>
> exten => s,1,Dial(SIP/&SIP/[EMAIL PROTECTED])
>
> How can I return the DIALSTATUS variable for the second SIP channel ONLY if
> the second SIP channel is busy, regardless of the dialstatus of the first
> SIP
Using 1.0.9:
If I have:
exten => s,1,Dial(SIP/&SIP/[EMAIL PROTECTED])
How can I return the DIALSTATUS variable for the second SIP channel ONLY if
the second SIP channel is busy, regardless of the dialstatus of the first
SIP channel? What I want is, if the second SIP channel is busy go to n+1
On Tue, Feb 28, 2006 at 09:44:35AM -0500, Chris Earle (CBL) wrote:
> Hi all,
>
> hard for me to explain this, but it keeps happening on a number of machines
>
> I attempt to upgrade zaptel, or do something to zaptel modules. and then
> I reboot the machine, and for whatever reason, it hangs o
I am currently running stock MeetMe configs at multiple locations on single P4 hardware (512M and 2G, depending on which location). I've seen no problems (no technical problems) up to and over 30 attendees, all coming in via
H.323 (chan_h323), using g.711u (no transcoding). Word of warning- abov
Ed Greenberg wrote:
I need to set up an office full of Cisco 7960 phones behind NAT with
the server out in Colo.
The first test phone registers fine, but the second one does not
register.
The easiest way to do this is put an Asterisk server on the local side,
have the phones register to it
On Tue, 28 Feb 2006, Cory Andrews wrote:
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other Zoom products
http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
I just found out, the 5801 does not support voip to FXO bridging. Only FXS
I need to read through the numbers in reverse order, so I can decide which
messages to play to people.
I was using a variable to mark how many messages they had read, and each
time read a number further back in the list.
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECT
I need to set up an office full of Cisco 7960 phones behind NAT with the
server out in Colo.
The first test phone registers fine, but the second one does not register.
The first phone's registration looks like so:
/SIP/Registry/3115552368
:64.169.xx.yyy:38836:3600:3115552368:sip:[EMAI
> That's getting pretty close - thanks for that.
>
> I just couldn't find any decent info on the web about working with
AGI.
>
Ditto. However, I pieced some stuff together by sifting through my
well-worn copy of TFOT and bouncing around between the wiki, the sample
AGI scripts and asterisk.gnu
hello,
I am trying to pass MWI from Asterisk to SER.my user agents register
with Ser.i am not able to figure out how to do this.
i added the changes for mailbox in sip.conf for ser peer entry.
[ser]
type=friend
mailbox=XYZ
also changes in chan_sip.c for asterisk but not seeing the notify
mes
> Hi,
>
>
>
> I'm trying to connect * to Nortel BCM 50, This PBX use H.323
> v3 to interface with other PBX. The port use to connect is TCP 1720 but
> I can't configure this port on my * box. I'm using a H.323.conf file
> sample to activate the port but the * isn't listening there. Some
IGNORE, mispost!
:-S
- Original Message -
From: "Chris Earle (CBL)" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, February 28, 2006 4:59 PM
Subject: [Asterisk-Users] Auto login via Remote User
> This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER
> requires web server
This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER
requires web server authentication right?
Correct me if I'm wrong, but this means that the contents of the User Table
would have to be in the passwd file defined through the .htaccess file
right?
.. because it passes whatev
same for me & ci$co router via chan_h323 (a have vad disabled on voip
dial-peer on router),
but I'm ignoring this notice messages from asterisk console and logs,
because calls are not disturbed ;-)
PJ
FaberK wrote:
Hi guys,
I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture
This is for the biz list. Please post there.
--- Sam Tam <[EMAIL PROTECTED]> wrote:
> Do anyone know who can provide some cheap PH
> routes/.'
>
> > ___
> --Bandwidth and Colocation provided by Easynews.com
> --
>
> Asterisk-Users mailing list
> To UN
That's getting pretty close - thanks for that.
I just couldn't find any decent info on the web about working with AGI.
regards,
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 0
Seems that in your iax settings you set the context to
mantra2. You need the same context in extensions.conf
(some one please correct me if I am wrong)
--- [EMAIL PROTECTED] wrote:
> Hi All,
>
> I was able to install Asterisk and make outgoing
> calls. Recently I purchased two
> DID's and I am f
Leo Ann Boon wrote:
Warren Burstein wrote:
I've observed a situation on my production system, and have managed
to recreate it on my test system (both running 1.2.4). I pick up a
phone connected to a TDM400B's FXS line. I dial a number (in my
tests, it was another local phone, but in product
I see these from time to time, I think it means that packets got lost,
or received out of sequence. It looks to me like asterisk manages to
deal with this, so unless your calls have also stopped working, I
wouldn't worry. (If we should be worrying, I expect someone will let us
know).
[EMAIL
Numbers from a text file - just need to read them back in one at a time.
PaulH
- Original Message -
From: "Michael Collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, March 01, 2006 5:14 AM
Subject: RE: [Asterisk-Users] Re: Asteris
We manually change the username on the phones, for example 'pbx' --
since you can do this before the phone contacts the boot server, it's
easy and fairly quick. The phones get the ftp server info from dhcpd,
but they use the username and password I've told them to.
Moj
Matthew T. O'Connor wr
On Tue, Feb 28, 2006 at 07:55:50PM +0100, Armin Schindler wrote:
> On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
> ...
> > - But I guess the workaround would yield to my current situation (I'm
> > running a patched version of 0.35 currently as mentioned at the start
> > of this tread): When a
>Kind of OT here, but just out of curiosity, how do you email them? Do
>you have an actual address, or do you just use the form on their web
>site? I've sent a bunch of requests via that form, and even though it
>says I should receive a response, I never have. I've tried going
>through a couple
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, February 27, 2006 7:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: Asterisk Question
>
>
> I w
Pasqualotto Enrico wrote:
Is possible that the Inbound routing routed only "from-pstn"? My FXO
(300) is in a from-internal!
Yes, is possible!
--
Pasqualotto Enrico
email: [EMAIL PROTECTED]
web: http://www.pasqualotto.org
-BEGIN GEEK CODE BLOCK-
Version: 3.12
GIT d? s: a-- C+++ UL
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, February 27, 2006 7:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: Asterisk Question
>
>
> I w
How can you check if transcoding is configured to work properly on a system?
Is there a way of knowing that transcoding is configured properly and is giving
some output to indicate so?
___
--Bandwidth and Colocation provided by Easynews.com --
Asteri
Greetings,
What is the recommended settings for using SPA-3000's FXO port for
dialing out to PSTN in regard of the DTMF?
The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports
registered to the Asterisk box with unique username/passwords.
The inbound PSTN DTMF works excellently
bb9", "Using CallerID "ht488" <300>")
in new stack
-- Executing SetVar("SIP/300-3bb9", "FROMCONTEXT=exten-vm") in new
stack
-- Executing Macro("SIP/300-3bb9", "record-enable|204|IN") in new stack
-- Executing GotoIf(
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
...
> - But I guess the workaround would yield to my current situation (I'm
> running a patched version of 0.35 currently as mentioned at the start
> of this tread): When a caller uses overlap sending (e.g from a POTS
> line) instead of block dia
Here is a link to some additional resources which may be helpful in
configuring the 5801 and other Zoom products
http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/
Cory Andrews
Purchasing Manager
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buff
Hi,
I’m
trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with
other PBX. The port use to connect is TCP 1720 but I can’t configure this
port on my * box. I’m using a H.323.conf file sample to activate the port
but the * isn’t listening there. Somebody
On Wed, Mar 01, 2006 at 12:45:45AM +0800, Sam Tam wrote:
> I think I have seen a post about that before. But can't find it
> again
> Can some people light me up with the detail
GSM extenders I don't think are legal in the UK, except if
installed/operated by a GSM network operator (
Hi guys,I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk.Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller.
Immediately, I though into an
Paul,
Just curious - what kind of stuff are you reading from the file?
-MC
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, February 27, 2006 7:53 PM
> To: Asterisk Users Mailing List - Non-Commerci
Hi Jordan, We are planning on building the same thing. We are still waiting for the hardware. We are using a Dell PE 2850 3GHz with 2G of RAM and a TE210P. I asked Digium support if this can suupport 50 users in one conference and the tech support guy said yes. Here's also a response f
On Feb 28, 2006, at 6:50 AM, Ash Thakrar wrote:
Hi Mark,
Thanks for your reply.
For the phase you have indicated the time it took was immediate, no
delays
there.
I have seen on the list several discussions of how additional delay on
ringing can be due to Asterisk trying to get caller ID
I think I have seen a
post about that before. But can’t find it again
Can some people light me
up with the detail
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We have just installed Asterisk in our new office and we have some
teething problems, but so far nothing we did not
expect/could not handle. However, our CEO was very attached to a
function in our old Nortel PBX that I am not sure
how to approach. If someone could point me in the right directi
On Tue, Feb 28, 2006 at 04:21:57PM +0100, Armin Schindler wrote:
> > See above: Seems the dialplan match isn't detected??
> It will be detected when chan_capi gets the signal to check it
> (SENDING-COMPLETE/SETUP INFO_IND, or normal CONNECT_IND in MSN mode with
> immediate=yes).
Hmm. As already m
I'm chasing down a pop/click type of disturbance on a PBX system.
Strangely, the disturbance is only heard by the outside caller, the
internal recipient hears the caller crystal clear. This seems to have
crept up when upgrading the zaptel driver to the 1.2 series while
running 1.0.10. I went ahead
Dear all,
Did anyone successfully test T38 fax pass thru to Cisco
as53xx? We’ve tried 1.2.4 with latest patch and latest svn trunk
and T38 patch but still not work. Reinvites from Cisco are correctly passed
back to the originating gateway, but fax never able to connect.
Cisco I
Can be as low as 15€cents from us on fix and 20€cents for mobiles
We don't have dids yet for Philipine
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Johnathan
Corgan
Envoyé : mardi 28 février 2006 18:07
À : Asterisk Users Mailing List - Non-Commercial
i think we can help. we do have there some contacts and if the total volume would be significant, we can give a nice quotation.
let me know off list what you exactly need.
BTW, $0.23/minute is much much high compared to our solution.
On 2/28/06, Johnathan Corgan <[EMAIL PROTECTED]> wrote:
S
On 08:05, Tue 28 Feb 06, Colin Anderson wrote:
> I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD,
> but no go
Do this, pick the kernel you want to load, and add: single
So in my laptops case it sais: Linux single
That will boot your pc into singleuser mode and it won't
en
Sam Tam wrote:
> Do anyone know who can provide some cheap PH routes/.’
I've been looking myself. Cheapest DIDs in Metro Manila I've seen are
$27.50/month; cheapest termination to same (non-mobile) from US I've
seen is $0.23/minute.
Expensive chismis :-)
-Johnathan
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