Hi Tim,
You could put he call back into the queue when the dial times out. Check for
the length of the CALLERID, if it's equal to the length of your internal
numbers then goto voicemail otherwise goto the queue.
thorben
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTEC
Michiel van Baak wrote:
Hi,
I read a lot about US providers that can terminate a PSTN
number for you and offer IAX or SIP connectivity.
Does anyone know such a company in The Netherlands ?
I read about Unet. Anyone with experience with them ?
Any information is welcome.
Michiel,
check out www.bu
For the Archives.
This document discusses analog recEive and transMit (E&M) Start Dial
Supervision signaling. Start Dial Supervision is the line protocol
that defines how the equipment seizes the E&M trunk and passes the
address signaling information (sends dual tone multifrequency (DTMF)
digit
TOUCH_MONITOR is the variable to set if I need to
specify my own options for ‘One Touch Record’ (filetype|filename|m).
I cannot get it to work.
Can you help?
Thanks in advance
___
Asterisk-Users mailing list
Asterisk-Users@list
No unfortunately a lot of the extensions do not have PC's near them or
in there offices, and the people involved are a little on the computer
illiterate side, although I am slowly training them.
They just want a phone that shows them extensions/lines and who is using
them That's why I am hopi
Hi,
I am sorry to be asking this but the wiki is down and has been for a
couple of days and I need to get this working before Monday to get my
live system setup.
Trying to get the Snom 190's and soon to arrive 3com 3102's to use the
function keys and for the life of me I can't work it out from
would an option where you could view it from a website or on your
computer be good?
coz there are a few ones out there like that, one in the wiki
James Bean wrote:
Sorry Newbie asking everyones option.
I am setting up a couple of small asterisk phone systems for my work, I
started using some
[EMAIL PROTECTED] wrote:
> Hi!
>
> Maybe I have just been looking on the wrong pages but there is a
> question that is very important for me. I already studied some
> Demo-Dialplans and made some basic experiences with Asterisk.
> But what I
> need to find out is how I can handle this.
>
> I am l
Hello, My colleague installed a Asterisk home as
company's SIP server and I would like to integrate the
Quintum gateway (SIP) but the calls don't get through.
Bellow is are the configurations on each side:
Quintum Primary Registrar =
202.69.190.244:5060 Prima
Steve Underwood wrote:
I had trouble with wink start (or delayed dial, or various other names
for the same thing) a long time ago. I think the switch was an AXE. The
spec said as soon as the seize came in I could start the wink. This
turned out not to be true. I needed to wait a while, or the PS
Hi Eric,
Eric Wieling wrote:
Does anyone know the default E&M Wink timings for Nortel DID ports?
The default settings on Asterisk are:
;prewink: Pre-wink time (default 50ms)
;preflash:Pre-flash time (default 50ms)
;wink:Wink time (default 150ms)
;flash: Flash t
On Fri, 18 Feb 2005, Robert Rozman wrote:
> I wonder which PRI interface card is most stable and supported for EuroISDN
> and Asterisk ? Are they stable enough ? Any tips ?
Digium TE410P and TE405P are well supported.
Peter
___
Asterisk-Users mailing
Well I fixed my setup by creating a seperate context for extension
defenitions for agents.
[agents]
exten => 1000,1,Dial(SIP/1000,20,rt)
exten => 1001,1,Dial(SIP/1001,20,rt)
exten => 1002,1,Dial(SIP/1002,20,rt)
and then had their regular menu accessable and inter office extensions
in defualt con
I fixed mine by updateing to the latest CVS-head. I spoke with David off
list and he no longer uses agent call back. I have noticed a slight
delay in transfers sometimes. But nothing major.
-Ryan
Hecken, Guido wrote:
I'm not shure, but I think something changed in CVS HEAD concerning the #
tran
Make sure you have the proper licenses to use the codecs:
g729
http://www.digium.com/index.php?menu=asterisk_g729
g723
http://www.dspg.com/technology/LicensePricing.html
On Fri, 18 Feb 2005 18:17:58 -0800, Nitesh Divecha
<[EMAIL PROTECTED]> wrote:
> Hello All,
>
> Any one has success with code
Hey,
Go to www.xten.com and download there X_lite dialer and create extensions!
You are ready to rock n roll
Nitesh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kiran Vahaja
Sent: Friday, February 18, 2005 5:22 PM
To: asterisk-users@lists.digium.com
Hello All,
Any one has success with codec g723 & g729?
I am having extremely hard time to setup this codec.
The only codec worked is g711a/u.
If I set g723 & g729 as first and second choice codec in my sip.conf, VM and
MeetMe stop working.
Sip.conf
[general]
port = 5060 ; Port to b
Are there any ACD softphones out there?
with features like :
hold
transfer
login/logout
line1, line2, line3 ...
ready
release (programable release codes)
wrap-up
mute
thanks
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.d
> Does anyone know if there is a way to get GotoIfTime to accept
> individual weekdays instead of a range?
>
> Example Dr. Office is closed on Thursday and Sunday.
Think the easiest option would be to use two statements:
exten => s,1,GotoIfTime(*|thu|*|*?closed,s,1)
exten => s,2,
Look at FireFly and Kphone.
Asterisk with IAX to FireFly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kiran
Vahaja
Sent: Friday, February 18, 2005 8:22 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VoIP Test Samples to test Asteri
I wonder which PRI interface card is most stable and supported for EuroISDN
and Asterisk ? Are they stable enough ? Any tips ?
Sangoma cards have _very_good_support.
You can use them for many different solutions, not only voice :)
I use them.
Maciej Kietlinski
__
I am offering a $100 bounty (or bribe) (payed via paypal or check,
your choice) to get this problem fixed before Monday 6am CST.
Rich Adamson wrote:
I'm confused. Digium cards do not support E&M trunks, which are older
interfaces that include either a two-wire or four-wire audio "plus"
two additi
Guys,
I am new to this group and asterisk. I downloaded the free asterisk
software and compiled successfully. I was able to get to CLI and type
'dial'. As usual because of sound card problem i could not hear
anything.
I do not have any hardware T1/E1 cards or equivalent to test out
asterisk. I wa
Surround the script with
~pp~
line 1 of script
line 2 of script
etc.
~/pp~
See example on this page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out+deliver+message
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
dean
collin
Andrew, thanks for the explanation ... see more
questions below. :)
--- Andrew Thompson <[EMAIL PROTECTED]> wrote:
> beonice wrote:
--- snipped some ---
> >
> > I guess the fundamental question is "why is a call
> > coming in from a DID any different?" And, of
> course,
> > "does a call
--On Friday, February 18, 2005 10:21 -0500 "Ferguson, Michael"
<[EMAIL PROTECTED]> wrote:
G'Day All;
So I purchased a Cisco 7960 and am now trying to get it configured for
*.
No can do without the variuos files/images through a FTPF server. I
configured the TFTP server on my RHES 3 box, now to g
--- Martijn van Oosterhout <[EMAIL PROTECTED]>
wrote:
--- snipped my quote of what 's' is --
> > I guess it
> > implies that calls coming from DIDs have digits
> > associated with them.
>
> Correct. On ISDN lines, E1, T1 and related digital
> protocols, details
> such as CallerID, Dialled Number
I would be very interested in the database table dumps, if you
wouldn't mind too much...
On Thu, 18 Nov 2004 23:45:46 -0600, Michael Shuler <[EMAIL PROTECTED]> wrote:
> SER is a stateless and/or stateless proxy. SER by itself it not very useful
> but SER teamed with Asterisk is how you make Aste
Robert, thank you very much for that informative
write-up. Of course, I now have more questions. The
first is really basic. I thought "extension" meant
something the caller dials _after_ reaching asterisk.
How come incoming DIDs have to be handled as if they
are extensions?
More questions follow:
Thanks for the reply.
It can't find the libzap.a library file, because it is not on my system
anywhere!
I did 'find / -name libzap.a' and it was not there.
I looked in the Makefile in my zaptel-1.0.4 source directory and found
no target of libzap.a
Any ideas as to how to get / build the requir
This script was developed by Mark Johnson.
All I did (Dean Collins) was type up the instructions and
make it easy to understand.
This Script will allow you to dial and extension number on
your [EMAIL PROTECTED] V0.6 or later pabx and have it read you the weather for your
city
How it d
Rich Adamson wrote:
My guess is the asterisk implementation for E&M signaling is probably
"one" end of that interface, watching the signaling bits and translating
those into something * can use to terminate the call. However, I'll
be the first to admit I'm not a programmer and probably wouldn't
r
- Original Message -
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To:
Sent: Friday, February 18, 2005 11:49 PM
Subject: Re: [Asterisk-Users] Which PRI card for EuroISDN ?
> On 23:11, Fri 18 Feb 05, Robert Rozman wrote:
> > Hi,
> >
> > I wonder which PRI interface card is most stable an
Yes, you are right, before using 8xxx numbers i was using 7 (i did
a copy/paste of my fedora box using 7xxx), then i got into this
trouble, i was using 2 xterms to compare fedora's asterisk and
Debian's. Eventually i decided to use 8xxx in debian's asterisk because
i was getting confused of
> >>>I'm confused. Digium cards do not support E&M trunks, which are older
> >>>interfaces that include either a two-wire or four-wire audio "plus"
> >>>two additional wires called the "E" and the "M" leads. So, Wink
> >>>timing is irrelevant (unless I'm misunderstanding your question).
> >>
> >>Uh
Specs: Fedora Core 3. Dual P3 600 (Dell PEdge 1300) SCSI
Disks
1x X100P (channel 1)
1x TDM20 (channels 2+3)
1x Knockoff X100P (channel 4)
I am looking to have all local and all toll free calls go
outbound through the Copper line, and all long-distance and international to go
out t
He may need zaptel for timing various other items such as meetme
conference, iax trunking, and probably a couple of other items
(which I do not recall). If those items aren't needed, then zaptel
isn't needed.
> You don't need the zaptel library if you aren't going to use
Hi,
I have been trying to learn how to enable
callers to my * to connect to a Mysql database, enter a unique number (
not a phone number, in this case a parcel number), and after verifying the
parcel number entered is valid and that it exists, give the caller the option to
change the parc
Hi,
I've installed a TDM400 card in an aging Dell Optiplex GXa (see my post a
few weeks ago). The machine powers the card okay, it shows up in an
'lspci', and asterisk runs fine with it. I've tried both 1.0.4 and 1.0.5.
The box in question is running a 2.4 kernel.
However... I'm having trouble
On 23:11, Fri 18 Feb 05, Robert Rozman wrote:
> Hi,
>
> I wonder which PRI interface card is most stable and supported for EuroISDN
> and Asterisk ? Are they stable enough ? Any tips ?
>
> Thanks in advance,
Hi,
We use AVM fritz! cards and they work wonderfull
--
Michiel van Baak
http://lunter
Thanks Jay,
For the Vonage information on how to make it work!
Just a quick question, what is the last number (99612) you specified in the
register string and beginning of sip parameter.
register => 16125551212:[EMAIL PROTECTED]:5061/99612
[sip99612]
Thanks,
Nitesh
-Original Message---
$5 DID? with who if you don't mind me asking?
On Fri, 18 Feb 2005 13:56:12 -0600, Jay Milk <[EMAIL PROTECTED]> wrote:
> Yes, it's doable, had this running for several months here. However,
> you'll need to get a softphone for $10/month from them, and they'll
> provide the sip-credentials on the
I’m trying to post a script on the wikki but it keeps
screwing up the text because it interprets the text as commands that cause
graphical errors.
Is there some trick to make the wiki think that the text is
just text?
Tia,
Dean
__
If I got you question, I think you can resolve your problem by separating your
incoming channels to particular contexts.
zapata.conf
context = line1
channel => 1
context = line2
channel => 2
extension.conf
[line1]
exten => s,1,Answer
exten => s,2,Whatever(YouWant1)
.
[line
On Fri, 18 Feb 2005 16:15:24 -0600
"Marco Castillo" <[EMAIL PROTECTED]> wrote:
You don't need the zaptel library if you aren't going to
use any digium
cards.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 1
Rich Adamson wrote:
On February 18, 2005 12:32 pm, Rich Adamson wrote:
I'm confused. Digium cards do not support E&M trunks, which are older
interfaces that include either a two-wire or four-wire audio "plus"
two additional wires called the "E" and the "M" leads. So, Wink
timing is irrelevant (unle
I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I
You don't need the zaptel library if you aren't going to use any digium
cards.
Marco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, February 17, 2005 8:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Zap
I don't think it would be logical (or efficient) have this run in a
dialplan macro at all; that would require creating a channel, copying
variables into it, etc.
I have been thinking about extending the Asterisk expression evaluator
to allow it to call out to res modules to do the evaluation (
Hi,
I wonder which PRI interface card is most stable and supported for EuroISDN
and Asterisk ? Are they stable enough ? Any tips ?
Thanks in advance,
regards,
Rob.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.co
Hi,
we have a client who seeks for help setting up and maintaining Asterisk
server (plain IAX trunk or SIP terminating calls on PRI card - nothing else)
in location Frankfurt, Germany. Should be operational in 30 days...
Please contact me offlist with offer ...
thanks in advance,
regards,
Rob.
Rich - thanks! Glad I am not the only one seeing this :)
Would be very interested in your results. No problems that I see yet
with these settings.
On Fri, 18 Feb 2005 10:21:08 -0600, Rich Adamson <[EMAIL PROTECTED]> wrote:
> > > That does not sound right at all. The difference between the two
> On February 18, 2005 12:32 pm, Rich Adamson wrote:
> > I'm confused. Digium cards do not support E&M trunks, which are older
> > interfaces that include either a two-wire or four-wire audio "plus"
> > two additional wires called the "E" and the "M" leads. So, Wink
> > timing is irrelevant (unless
True, but it also states that with no timeout value that it will dial
until the caller hangs up.
I have included my dial pattern - can anyone see anything that would
cause this, or something in my sip.conf or h323.conf files that would
override these settings?
Thanks,
Greg Oliver
[outbound]
ex
Hi,
Did you tried to set your DMA or SATA as described at message "Sangoma
A102 cards testing FIXED"?
Daniel
Kumak wrote:
Hello,
I have following problem with Sangoma A104 card:
CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Leng
On Fri, Feb 18, 2005 at 03:42:00PM -0500, Barry Porch wrote:
> Matt,
>
> Has this been incorporated into the regular libpri as opposed to the
> mattf-libpri? Are there any samples or documentation as to how the
> b-channel transfer and MWI would be configured and used?
This is against standard l
Good Day list,
Does anyone know if there is a way to get GotoIfTime to accept
individual weekdays instead of a range?
Example Dr. Office is closed on Thursday and Sunday.
Ron
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium
Bill Hamlin wrote:
The thing is I need to wait a few seconds. In fact, it's even worse, I need
to wait a few seconds, dial an extension, wait a few more seconds, and then
dial another!
It's perfect for something like
... Wait(5)
... SendDTMF("123")
... Wait(5)
... SendDTMF("456")
but the Dial comm
I have a Channelized Voice T-1 (FXO channels) configured as fxs_ks
signalling and am getting the following messages on the console:
Feb 18 15:22:33 WARNING[1862]: chan_zap.c:3428 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 2
Does anyone have any idea what is causing this and ho
On Thu, 17 Feb 2005 15:30:42 -0800, Robert Goodyear <[EMAIL PROTECTED]> wrote:
> That is pretty cool, except I'm not able to get my head around the
> usefulness of BT's 3~30 foot range limitation. Every time I think of a
> great use for BT, I then think about the consequences of walking down
> the
I am not sure it can be done. AS soon as the first number
answers, the channel is going to be cut through to the
original caller.
On Fri, 18 Feb 2005 16:26:09 -0500
"Bill Hamlin" <[EMAIL PROTECTED]> wrote:
The thing is I need to wait a few seconds. In fact,
it's even worse, I need
to wait a f
During the last week, we have had several support issues being reported
as bugs on the bug tracker. Since we are going into a final development
stage on version 1.1dev (CVS HEAD) in order to complete the 1.2 release
we are under pressure to fix bugs and handle a lot of reports in a short
time
Sipura 2100 is supposed to implement T.38
real-soon-now.
I've got a Multi-tech ATA with T.38 support on order
on the theory that Multitech has been making well regarded FAX modems for years
and might know how to actually do FAX reasonably well.
Jim
James H. Thompson[EMAIL PROTECTED]
Ah! You guys are right, the D option will do the trick I think.
Thanks,
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Thompson
Sent: Friday, February 18, 2005 4:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [
On Fri, 2005-02-18 at 12:32 -0800, Trevor Peirce wrote:
> Joseph wrote:
>
> >I'm testing new cordless Motorola phone and the handset is constantly
> >displaying message: MSG WAITING OFF
> >According to the manual this message suppose to go off but it doesn't
> >
> >Is it something that I can contr
The thing is I need to wait a few seconds. In fact, it's even worse, I need
to wait a few seconds, dial an extension, wait a few more seconds, and then
dial another!
It's perfect for something like
... Wait(5)
... SendDTMF("123")
... Wait(5)
... SendDTMF("456")
but the Dial command doesn't retu
Hi all,
I have call recording enabled via the Monitor command and it seems,
the call stops being recorded after the call is transferred. Is this
normal behavior? If so how can I continue recording of calls after
they have been trasnferred
thanks...
n Fri, 18 Feb 2005 16:06:54 -0500
"Bill Hamlin" <[EMAIL PROTECTED]> wrote:
I'm using Dial to place a call to a PBX. But then I
want to wait a few
seconds and dial an extension. Dial doesn't return
until the call is
disconnected though.
I also want the caller to not hear any audio until the
D
Bill Hamlin wrote:
I'm using Dial to place a call to a PBX. But then I want to wait a few
seconds and dial an extension. Dial doesn't return until the call is
disconnected though.
Try this posting:
http://www.voip-info.org/wiki-Asterisk+cmd+dial?page=Asterisk%20cmd%20dial&comments_threshold=0&com
If there is a power failure, which cards other than x100p and
voicetronix openswitch provide "redundancy".
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update opt
I'm using Dial to place a call to a PBX. But then I want to wait a few
seconds and dial an extension. Dial doesn't return until the call is
disconnected though.
I also want the caller to not hear any audio until the DTMF has been sent.
This gets the caller to the right place and he doesnt have
Matt,
Has this been incorporated into the regular libpri as opposed to the
mattf-libpri? Are there any samples or documentation as to how the
b-channel transfer and MWI would be configured and used?
Thanks,
Barry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Be
Joseph wrote:
I'm testing new cordless Motorola phone and the handset is constantly
displaying message: MSG WAITING OFF
According to the manual this message suppose to go off but it doesn't
Is it something that I can control via * ?
It's probably made by the same Chinese company that made the ch
Eric Wieling wrote:
Rich Adamson wrote:
Does anyone know the default E&M Wink timings for Nortel DID ports?
The default settings on Asterisk are:
;prewink: Pre-wink time (default 50ms)
;preflash:Pre-flash time (default 50ms)
;wink:Wink time (default 150ms)
;flash:
Yes, it's doable, had this running for several months here. However,
you'll need to get a softphone for $10/month from them, and they'll
provide the sip-credentials on their website. It's a lousy solution if
you really just want one number, because then you'll have to pay
$15/month for their basi
Brancaleoni Matteo wrote:
Hi,
Il giorno ven, 18-02-2005 alle 12:44 -0600, Eric Wieling ha scritto:
The Digium Tx00P and TE*xxxP support E&M Wink
E&M is analogue, not digital...
digium cards support it over digital, like they supports fxs/fxo
to a channel bank . same from E&M
The interface describe
Thanks I will give it a try:)
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, February 18, 2005 1:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Send CallerID to PBX vi
Hi,
Il giorno ven, 18-02-2005 alle 12:44 -0600, Eric Wieling ha scritto:
> The Digium Tx00P and TE*xxxP support E&M Wink
E&M is analogue, not digital...
digium cards support it over digital, like they supports fxs/fxo
to a channel bank . same from E&M
The interface described here is analogue, afa
According to Motorola support the communication has to be setup at
1200pbs.
Does anybody have an idea how to do it?
The phone is connected to Sipura-3000
--
#Joseph
On Fri, 2005-02-18 at 11:42 -0700, Joseph wrote:
> I'm testing new cordless Motorola phone and the handset is constantly
> display
William Cruz wrote:
Hi everyone in the asterisk community. Am new to asterisk, while doing
the installation I notice that sip.conf examples were not clear for
beginners like me so I would like to share my current working
configuration with everyone.
Swifttel.net is a new VoIP service Provider out o
The Enable Call Features is set to NO!
>>> C F <[EMAIL PROTECTED]> 2/18/2005 12:10:00 PM >>>
looks like call forwarding is on localy on the phone.On Fri, 18 Feb 2005 08:55:53 -0700, Josh Wilson <[EMAIL PROTECTED]> wrote:> 1.0.5.16 - the latest version.> > >>> Michael 'Moose' Dinn <[EMAIL PROTE
go to:
http://www.grandstream.com/BETATEST/
Release 1.0.5.221/21/2005
 Changed polarity reversal logic per customer request and fixed the
polarity reversal issue
 Add support for syslog server (HT286 only)
 Change the choice between tftp upgrade and http upgrade to mut
Yes that's what I have in my current config... : context=incoming
signalling=fxs_ks echotraining=800 echocancel=yes
echocancelwhenbridged=yes rxgain=0 txgain=0 immediate=no busydetect=no
callprogress-no musiconhold=default usecallerid=yes callerid=asreceived
group=1 callgroup=1 pickupgroup=1 ch
You can try and contact the dealer that sold the Cisco phone to you,
and maybe they will give you the image.
On Fri, 18 Feb 2005 10:21:21 -0500, Ferguson, Michael
<[EMAIL PROTECTED]> wrote:
> G'Day All;
>
> So I purchased a Cisco 7960 and am now trying to get it configured for
> *.
> No can do w
Just the voltage
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ed
Greenberg
Sent: Friday, February 18, 2005 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Difference between a TE410P and TE405P?
Can anybody
Ed Greenberg wrote:
Can anybody tell me the difference between a TE410P and a TE405P? Is it
JUST the 5v vs 3.3v pcis slot spec, or is there some thing else?
That's all.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.c
looks like call forwarding is on localy on the phone.
On Fri, 18 Feb 2005 08:55:53 -0700, Josh Wilson <[EMAIL PROTECTED]> wrote:
> 1.0.5.16 - the latest version.
>
> >>> Michael 'Moose' Dinn <[EMAIL PROTECTED]> 2/18/2005 8:14:41 AM
> >>>
>
> What firmware are you running on your 101?
>
>
>
I have updated to 1.0.5.22 and I still get the same problem.
Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-a873 is busy>>> [EMAIL PROTECTED] 2/18/2005 10:41:14 AM >>>
Can you send me that update?>>> Michael 'Moose' Dinn <[EMAIL PROTECTED]> 2/18/2
On February 18, 2005 12:32 pm, Rich Adamson wrote:
> I'm confused. Digium cards do not support E&M trunks, which are older
> interfaces that include either a two-wire or four-wire audio "plus"
> two additional wires called the "E" and the "M" leads. So, Wink
> timing is irrelevant (unless I'm misun
On Fri, Feb 18, 2005 at 11:06:16AM -0700, Chris Modesitt wrote:
> I am terminating a PRI, NI2 signaling into a PBX (My company's PBX not the
> PSTN) from an Asterisk server. Caller ID number appears to be transmitted
> caller id name is not being transmitted is their a compile time flag on
> libpri
I have a bunch of Grandstreams ATA286 for sale. They are brand new in
the box loaded with the latest code and tested. Price is $50.00. This
will work with the Asterisk PBX. They have one 10/100 Ethernet port and
one FXS port.
Send email to [EMAIL PROTECTED] from more details.
_
Hi everyone in the asterisk community. Am new to asterisk, while doing
the installation I notice that sip.conf examples were not clear for
beginners like me so I would like to share my current working
configuration with everyone.
Swifttel.net is a new VoIP service Provider out of Georgia. Their web
Freddi Hansen wrote:
what about letting this logic be followed by a dialplan macro that gets
called to make the decision. May sound weird but we have a 'network-id'
attached to each sip-user. That network-id contains the public fixed-ip
of the users network plus a unique name. If the network id
On Fri, 18 Feb 2005 18:11:26 -
"Brett, Gary" <[EMAIL PROTECTED]> wrote:
Hello all
I am relatively new to asterisk and am sure this will be
a simple question
to answer. I have a TDM400p card and I am in the process
of creating my dial
plan, however I am a bit stuck on one thing. I have 2
ana
Rich Adamson wrote:
Does anyone know the default E&M Wink timings for Nortel DID ports?
The default settings on Asterisk are:
;prewink: Pre-wink time (default 50ms)
;preflash:Pre-flash time (default 50ms)
;wink:Wink time (default 150ms)
;flash: Flash time (defa
I've watched the dialogue about how asterisk has to manipulate the
packets from an IAX2 connection to a SIP client. That said, I'm
wondering if a previous problem I've been trying to diagnose could be
related to that process. In short, here's how I describe it:
Outbound:
SIP 7960 > Asterisk > I
I'm testing new cordless Motorola phone and the handset is constantly
displaying message: MSG WAITING OFF
According to the manual this message suppose to go off but it doesn't
Is it something that I can control via * ?
--
#Joseph
___
Asterisk-Users mai
Can anybody tell me the difference between a TE410P and a TE405P? Is it
JUST the 5v vs 3.3v pcis slot spec, or is there some thing else?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-
Hi Ismael,
Thanks for the reply, i did the install again this time with "make
samples" and this is what i get now:
What am i doing wrong?
Thanks
Edward
[EMAIL PROTECTED] asterisk-1.0.5]# ./asterisk -vvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.con
They emailed me about the phone and want to arrange for us to test and
certify it on the network.
As soon as we get one I will post the results. We try to test anyone's
equipment that asks and will
not even consider using something that we have not tested. Our last email
was yesterday. I think
t
1 - 100 of 238 matches
Mail list logo
