Thanks, I don't play with web pages to much. It has a lot of great stuff
for a newbe like me.
Thanks, David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Tuesday, March 22, 2005 8:01 AM
To: Asterisk Users Mailing List - Non-Commercial
Has anyone configured the Zoom V3 to connect with Asterisk? I bought one
at Fry in San Diego for $99 bucks. I can't get it to register with
Asterisk..
Thanks, David
http://www.zoom.com/products/voip_products.html
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Hello All, I'm new to VoIP. I have a friend that has an Adtran 608 with
6 lines over a T-1. He likes my Asterisk box. Could I replace the Adtran
608 with an Asterisk box??? Any ideas on an interface card??
Thanks, David
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Hello All, Yep I'm a newbe.
I'm just started to play with asterisk.
What I have
Redhat Fedora Core 2 (New install)
3 X100P cards.
I installed
zaptel-1.0.3
libpri-1.0.3
asterisk-1.0.3
Where should I start??
--
Thanks, David
--
This message has been scanned for viruses and
dangerous content
Hello All,
I loaded [EMAIL PROTECTED] I have one X100P card. I try to dail out but get
rejected.
Any help...
Thanks, David
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This message has been scanned for viruses and
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KE6UPI thanks MailScanner for their support.
Please contact
on applications, just type show applications at your
; friendly Asterisk CLI prompt.
;
; 'show application command' will show details of how you
; use that particular application in this file, the dial plan.
;
[incoming]
include = default
exten = s,1,Dial,Zap/2
The reason, David, that nobody has responded
Hello All, I'm trying to dial out with no luck.
I'm using [EMAIL PROTECTED] defaults. I have one X100P card and SJPhone.
*CLI dial 96985628
No such extension '96985628' in context 'default'
Here is my exten
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten =
Well I guess I need to fix or create a channel now.
Asterisk Ready.
*CLI dial [EMAIL PROTECTED]
Jan 9 10:28:06 NOTICE[10750]: app_dial.c:743 dial_exec: Unable to create
channel of type 'Zap'
No luck when I dial [EMAIL PROTECTED]
David
David wrote:
Hello All, I'm trying to dial out
= s,1,Dial(SIP/300,10)
So what is s .
Thanks, David
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Hello,
Can anybody help me with this issue?
-- Called 999302
-- Got SIP response 488 Not
Acceptable Here back from 202.125.154.12
== No one is available to answer at this time
Why am I getting error 488. Im using Sipura SPA-2000
Thanks
David
Hello,
Can anybody help me with this issue?
-- Called 999302
-- Got SIP response 488 Not
Acceptable Here back from 202.125.154.12
== No one is available to answer at this time
Why am I getting error 488. Im using Sipura SPA-2000
Thanks
David
Christopher,
Any idea what causing Max retries
exceeded to happen?
Regards,
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher Dobbs
Sent: Saturday, January 19, 2002
8:31 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
Hi,Adi,
We provide the USB phone you wanted, it can access Asterisk natively. It
can support Skype,X-Lite,X-PRO,eyeBeam,StanaPhone,SJphone,Net2Phone,Firefly and
MSN too. To get more information about that, contact with me offline or goto
our website please.
Regards.
David
I think your remote peer should use the Eyebeam and enabled the video too.
Regards.
David
- Original Message -
From: Ing. Ignacio Ortega A. [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Thursday, January 27, 2005 10:07 AM
Subject: [Asterisk-Users] I need Help everyone I
Hi,Jason,
The TDM400P card
failed to get the Callee number or DID, so the * don't know how to route the
call. There are something difference between the analog line and the PRI
line.
Regards.
David
http://www.iaxtalk.com
- Original Message -
From
Hi,Edgar,
Config the agents.conf correctly and it will do what you want. For more
information, search it in the wiki please.
Regards.
David
http://www.iaxtalk.com
- Original Message -
From: Edgar de Leon [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Hi,
Where could I download the soxmix please? I
want to mix two .gsm files into one.
Regards.
David
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Hello,every one,
I have recorded the voice files
withmandarin (China).Where should I contrib the files ?
Regards.
David at iaxtalk.com
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Hello,
I have a version of asterisk running on my server for more than 1 year. I
wanna update it to the latest version without over-writing any of the
config files.
How can I do this?
Thanks
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hi ..
i've just converted myself back to a newbie by trying to experiment with
some new stuff .. I have connected two grandstream Budgettone 100 phones
to my asterisk, and trying to experiment with them ..
I am trying to get into the asterisk sample basically ..
when I dial 1000 asterisk
hi,
can someone who has used Budgettone phones tell me how to do the
following:
an incoming call comes in and is answered by the receptionist.
she need to put the call on hold, speak to whoever the call is for,
and either (after that) pass on the call, otherwise speak again to
whoever was on the
So DIDs are sharing available channels.
In particular for ISDNs are DIDs sharing available channels?
--
David Kwok
CISSP,(ISC)2
61282315751 ext 1002
FWD#/IAXTEL# : 17001813482 ext 1002
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[EMAIL PROTECTED]
http
Service (DNIS) that is put on T1's for inbound 800 and 900
lines. This is an inband delivery of the last 4-digits of a dialed number
(800/900) that is passed into the PBX from the SPfor callcenter or other
routing. Does Asterisk support this?
- David Schlossman ([EMAIL PROTECTED])
of the Digium cards, DID signalling is
not supported.
Hope that helps a bit -- David Schlossman
its which is currently
permitted.
Hope this helps
David Schlossman
[EMAIL PROTECTED]
I
setup a ATA-186 with no problems at all by following the instructions from John
Todds excellent article at http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt
Hope
that helps
-
David Schlossman
[EMAIL PROTECTED]
ronald ramos wrote:
Hi,
For now i just turned off acpi. and it works now.
just dont know what's the connection of that though
:-)
i will try to do the things you guys suggested also
when i get the chance, thanks for you help!
regards,
nhadie
--- Tzafrir Cohen [EMAIL PROTECTED]
Joseph wrote:
On 04/05/08 05:16, bilal ghayyad wrote:
Hi All;
Till now I am not able to find a good IAX IP Phone or
Gateway that can be used with good quality.
Anyone can advise for good one?
Regards
Bilal
I've not seen IAX phone so your best option will be IAXy adapter from digum.
Roberto Milani wrote:
Roberto - I noticed in your original email you had the lines
something like
mailcmd=/opt/local/bin/msmtp -t ; --from blah
AND
serveremail=from=blah
In mailcmd everything after the ; will be ignored as a comment
In serveremail - well - it should throw
Yes, both Asterisk and Cisco are behind Nat.
My asterisk box is behind a dsl modem and router. All traffic is bridged
from the modem to the router. Here are the settings on the router;
http://dwabbott.com/pictures/port_forward.png
http://dwabbott.com/pictures/range_forward.png
The asterisk box
fateme fatah wrote:
Hi:
I want to install mpg123-0.59r on my asterisk server.I downloaded it
in /usr/src then untared it and I typed these command :
#cd /usr/src/mpg123-0.59r
#make linux
after run make linux ,I saw 2 errors in terminal:
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o
Have you configured and tested sendmail?
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Preetish Kakkar wrote:
But how would my calls be transferred to extension phones from
asterisk server. Would i need to connect those phones to Digium card
as well. What i mean is would digium card have a main extension where
i would connect main pstn line and other 3 port where i would
Hello, when with my client X-lite try to register in the server that
say me,
Registration error:501 Not implemented.
Google is your friend;
http://www.google.com/search?hl=enq=asterisk+register+x-litebtnG=Google+Searchaq=foq=
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/voicemail.conf
[default]
1000 = ,David Abbott,[EMAIL PROTECTED]
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and
retrieves alphanumeric data, plays/announces it to the user and deletes the row
from the database:
The SQL queries would look something like:
SELECT user, pwd FROM codes WHERE dialed = '111';
DELETE FROM codes WHERE user=$user AND pwd=$pwd;
Thanks,
David
.
Is there any way to bypass/ignore the fact that MySQL is installed separately
and enable the installation of the addons?
Thanks,
David
Got a little couch potato?
Check out fun summer activities for kids
reaches an extension that doesn't have an active
mailbox? Something like:
exten = _123105.,2,Playback(no-box,noanswer)
Thanks.
David.
Have a burning question?
Go to www.Answers.yahoo.com and get answers
Joseph wrote:
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
I'll eventually replace this crippled Linux with something better but I don't
time to play around with it as most divers and modules are still too new and
not fully available in all distros.
Joseph wrote:
It keeps complaining about /lib/tls/libc.so.6 'GLIBC_2.4' not found.
How do you install this library on EEE pc Xandros? (I know Xandros is Debian
based) but this is eee pc.
You should ask on another list but this should get you started;
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On the Python Tutor mailing list Kent Johnson uses a script to find the
top posters for the year. If this or something like it has been posted,
sorry for the noise;
2008
Steve Totaro 796
Tzafrir Cohen 749
Tilghman Lesher 496
Alex Balashov 354
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Totaro wrote:
| I would venture to guess that I would be in the top three (if not 1st)
| for the last five or more years. Would it be very hard to run the
| same script for years gone by? It would be interesting to see,
| especially when
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Bayardo Sanchez wrote:
| I need help need recording all call for my pbx but i am a novato in
| asterisk my confi for record is:
|
|
bilal ghayyad wrote:
Hi All;
Anyone knows an IAX IP Phone works fine and tested?
Does polycom support IAX IP Phone?
Regards
Bilal
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bilal ghayyad wrote:
Dear David;
At what price u get it?
Did u test it with IAX and SIP? Are u sure it is good? As really I did not
deal with chinese phone until now and I found it fine.
Regards
Bilal
--- On Mon, 1/19/09, David da...@linuxcrazy.com wrote:
From: David da
Try iaxLite or sipLite
- Original Message -
From: David fire
To: bilmar...@yahoo.com ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Monday, January 26, 2009 7:43 AM
Subject: Re: [asterisk-users] soft phone
there isnt any free soft phone wich support G729
Hi Sriram,
the customer should be billed a premium rate ex, Rs.9 per minute..
Will be billed by you or by telecomm company?
Regards
David
- Original Message -
From: Sriram
To: asterisk-users@lists.digium.com
Sent: Thursday, February 05, 2009 1:46 PM
Subject: [asterisk
perfectly without any problems.
What have I done wrong ? Is there a better way to implement a custom
transfer feature?
Thanks,
David
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Hello,
I am running asterisk 1.4. For argument's sake I have 4 telephones. 2
support video, 2 do not.
Calls between phones work fine and codecs are properly negociated. I
have videosupport=yes in sip.conf and when the two video phones
communicate I have video.
I call meet me with this
allow it between peers?
Thank,
David
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and password is.
David
Michiel van Baak wrote:
On 13:45, Thu 26 Mar 09, Lutgring, Sam wrote:
My preferred method is to use my own TFTP server. This makes changes to
accounts/phones very fast and easy. The whole process takes me about 5
minutes to deploy an entirely new phone.
1) I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John F. Ervin wrote:
| So, people have recommended building a system from scratch, start with a
| CentOS base and installing asterisk and all of the other utilities.
| I've only used Trixbox for my business system. I'm wondering what
| surprises I'd
,David Abbott,x...@.net
Thats all I have in there, asterisk will use my SMTP client without me
doing anything. I am using asterisk 1.4
- -david
- --
Powered by Gentoo GNU/LINUX
http://www.linuxcrazy.com
pgp.mit.edu
-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.11 (GNU/Linux)
Comment
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
jonas kellens wrote:
| David,
|
| what is your SMTP-client then ?
|
| Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it
| still /usr/sbin/sendmail ??
I don't have mailcmd in voicemail.conf, I was under the impression
Joseph wrote:
On 06/21/09 14:04, Joseph wrote:
When I call internal extension from PSTN line everything is working
correctly phones are ringing they way they should but internally when I try
to dial two
extensions on one sipura unit and my Digium IAXY unit rings only once and
call goes to
.
I use an Eutetcics IPP200 USB handset with linux usb audio drivers and kiax
for software.
http://www.eutecticsinc.com/news/news.html
It works ok but it depends on the audio drivers. I thought any USB handset
would work with linux sub audio drivers but that was just an assumption.
snip
David
on the callers profile? If yes, how?
- How (if at all) can I configure the voicemail to send the emails via an external SMTP server?
Thanks.
David
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Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some reasons I have an issue with the quality. The voice is a bit
scratchy. I have tried iLBC and SPEEX, but it didn't make any
Thanks Sean,
I can't really use ULAW, bcz I will have more than 20 calls at the same
time, and the entire path is a single codec (iLBC)
You have mentioned something about IAX timing. How can set this value?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
way to
configure * if there are some packet losts ?
Thanks
David
Senior Network Administrator
Call Center Development Services
(t) 514.731.5046 ext. 226
(f) 514-731.5834
(m) 514.814.0203
(e) [EMAIL PROTECTED]
(w) www.ccds.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto
-Commercial Discussion
Subject: Re: [Asterisk-Users] Voice Quality
David:
Bandwidth may be an issue; however, do you have any timing devices
installed? Digium's hardware (or any generic knockoffs) will provide this.
There are also some other ways, such as ztdummy or a usb controller (haven't
used
Thanks for your reply...
I was told to disable the jitter if using trunk=yes in iax.conf..
Have you guys had any experince with having jiiterbuffer=yes and trunk=yes?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent:
Title: Untitled Document
Hello
Guys,
Any
idea what this means:
WARNING[2138]: chan_zap.c:4409
my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 1 -
audio may have been lost
Thanks
Title: Untitled Document
Hi All,
I have configured Line1 (2011)and
Line2(2012)in SipuraSPA-2000 (latest Firmware)to use
G729. In sip.conf I have set disallow=all, allow=g729
IfLine1 is in use by an agent, then Line2 won't
work and viceversa (Inbound Calls Only).I have 40 license for G729.
Hi,I am trying to do the world's most simple install.I have a Wildcard TDM400P with 3 ports: 1 FXS on port1 and 2 FXOs on ports 3 and 4. (i'm not using port 3for now, put want it for expansion purposes)I simply (to start with) am looking to have the FXSphone ring when an FX0 port is dialed. I
is detected to arrive to FXO ports, will get to
incoming context and
will ring the receptionis.
I have no experience with FXS ports, but try what i
have just tell you
and post how is going so far.
best regards
On 6/30/05, David [EMAIL PROTECTED] wrote:
Hi,
I am trying to do
=friend, incoming calls doesn't
works. If the type is set to another value (for example peer) incoming
calls works fine, but outgoing calls doesn't works.
What can I do?
Thanks
David
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Thank you in advance.
David
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know you can pass info INTO AGI, but can you pass the info back OUT of AGI into the Asterisk extensions.conf dialplan?Many thanks. David
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On Friday 13 August 2004 22:06, Craig Guy wrote:
Hi,
Is an onramp 10 what is referred to as a 'channel bank'?
A channel bank is a device that would take the onramp 10 in one side a present
10 separate PSTN lines out the other.
--
Best Regards,
David Price
works just as
intended, but the crashes are making the system unusable.
I am pulling my hair out with this problem and my SO wants me to give up the
project. Any and all help will be greatly appreciated!
Thanks,
David
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It sounds like my lockups may be related since my TDM422b card has the FXS FXS FXO
FXO configuration and doesn't have an FXO in position 1 either.
My card is identified in software as Rev E/F and has the wire jumper on the back.
David
Richard Scobie said:
Maciej Kietlinski wrote
Nick,
I too battled a similar problem with my TDM400p. I solved it by putting the
following in the channel descriptions in zapata.conf:
stripmsd=0
Clearly this is not the default which I think should be obvious...
David
Nick Barnes said:
Hi all,
I've been batting my head against a brick
further sound stops. The
machine seems to be stalling, but I have noload on both oss and alsa modules (they
seem to be the culprit of all googled problems and I don't need a console).
David
Sep 11 20:00:12 VERBOSE[671762]: -- Goto (intern-post,18887452654,1)
Sep 11 20:00:12 VERBOSE[671762
I gather from the lack of response that no one has had a similar problem or knows
how to troubleshoot the problem. The Ooh, voice format changed to 4 is a mystery
to me since everything I find with that message has a coder format where I have a 4.
David
David said:
I have somewhat miraculously
calls.
David
Tim Robinson said:
Nick -
Put
nationalprefix=0
internationalprefix=00
in your zapata.conf file!
Magic!
Rgds
Tim
Nick Barnes wrote:
Hi all,
I've been batting my head against a brick wall for the best part of the day
and still haven't got any further (apart from getting
, and ironically, the file missing is missing).
Since I can't compile the cvs libiax, I am back to using the debian libiax0 and
libiax-dev. And since I can't get the CVS asterisk to run, I am back to RC2 and the
problems listed in my last email.
Please let me know what is going on here.
David
Sep
timeout never occurs, I never see MACRO_RESULT set, and the call is
connected even though it shouldn't be until the caller presses 1.
Any help (or explanation about why this doesn't work) will be greatly
appreciated.
I have been pulling my hair out trying to get this to work.
Thanks,
David
To add to the mystery, if the cell phone answers and presses 1 as requested,
the
logs don't register priority 1,1 being executed. It is as if the macro has
prematurely aborted.
David
David said:
I just downloaded, compiled and installed Asterisk 1.2.9.1. I did this
specifically
to get
Thanks for the response! I used your template to write a similar one for us
and it
works great. I wonder if there is a bug in the macro timeout code.
David
whois wes said:
This may sound stupid, but I had a similar issue that I solved by
placing an Answer at the beginning of what would
Hello,In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system.Is there a way to manipulate this message, as
Hello,I'm trying to set the Asterisk to leave a video message to the mailbox, but there is some compatibility problem, although h263 is identified as the matching codec, as you can see in the debug messages below:Capabilities: us - 0x80100 (g729|h263), peer - audio=0x43f
terisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 30, 2006 2:15:03 AMSubject: Re: [asterisk-users] Pager Voicemail Message
Yes. It should be in that same file. Poke
around.
- Original Message -----
From:
David
Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 30, 2006 8:06:22 AMSubject: Re: [asterisk-users] Pager Voicemail Message On 10/29/06, David [EMAIL PROTECTED] wrote:I looked. There's nothing there.I even did a search under /etc/asterisk for files
Hi All;
Anyone can advise for a good IP Phone that has the
ability to support SIP firmware and IAX firmware?
Ofcourse, SIP there is a lot, but we need also the
ability to use IAX (as it is good for NAT).
Any advise.
Regards
Bilal
I am using an atcom at-530
and do not know where to go from here. I would really
appreciate it if someone could give me some pointers on where to go next, what
additionnal debugging steps I should perform. I would also really appreciate if
someone could propose a solution.
Please help!
David
Never give up, never surrender
signalling between the two calls.
Maybe something is different.
What I find really weird is that the DTMF is incorrectly sent from the
first asterisk only when the second asterisk bridges to DAHDI.
Any ideas?
David
On 11-04-23 11:48 AM, David wrote:
Hello,
I installed Asterisk 1.6.2.17.3
version.
Everything else is identical. So the problem appears to be caused in the
RTP and not in the SIP. So something about the RTP packets coming from
the DAHDI channel on asterisk-pri makes asterisk server send invalid DTMF.
David
On 11-04-24 11:42 AM, David wrote:
I did more testing.
Here
__ast_read: DTMF end
emulation of '#' queued on SIP/omnity-0023
I notice that the # key was repeated several times by the DTMF even
though the dialplan only calls # once. Why are these two different when
the DTMF sequence is exactly the same ?
Any ideas?
David
at a time because I want to validate the user's entry
at each key press.
Thanks,
David
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On 2011-04-27 13:34, vip killa wrote:
I just completed building a feature rich asterisk voicemail system
using perl, php, and mysql.
My only concern is that the system i built will not be able to handle
the call volume needed. Let me start by explaining my setup
and will give you lots of distortions on your
VoIP.
David
On 2011-04-28 11:25, Bruce B wrote:
Hi everyone,
How can I introduce some distortion, echo, chopping sound and all
other bad quality things that can happen to a SIP trunk? I have plenty
of bandwidth and crisp clear lines so the only
wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David
*Sent:* Thursday, April 28, 2011 10:32 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re
in debugging this issue ?
David
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I've created some images. I currently don't have a free Raspberry Pi so I
have not updated any images for a little while.
A how to on building your own.
www.klaverstyn.com.au/david/wiki/index.php?title=Asterisk_for_Raspberry_Pi
A how to on writing a pre-compiled image
http
Does anyone know how long the orders take?
I ordered some a couple of days ago and it said normally 24hours, and I
am guessing that the weekend cause's some delays but it did not say
anything abouy that.
Any one got any ideas on how long generally over the weekend it takes?
Thanks
David
-Original Message-
From: Julius Kidubuka [mailto:[EMAIL PROTECTED]
I need to be able to send an sms alert to one's mobile/cell phone. For
instance, when I receive a voicemail message in my inbox, I
also want to be able to get a message on my cell phone alerting me of this
e-mail.
one know what the exten line would be to be that generic or
point me to something that would explain it?
Thanks
David
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was happy with it. so thats my opion
and personal choice.
David
Any advice would be greatly appreciated
Gary
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