I'm reading about Local channel and applications. One fancy idea came up:
if my generic dial plan uses Dial() with no timeout, can I assign it a
timeout for special purposes by
Dial(Local/[EMAIL PROTECTED],,20)
or even add other Dial() options.
Well, I can't. (Maybe a feature request?) So
can you show us you zaptel.conf zapta?
Younss AZ
KASTERISK.COM
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Hello,
I can't find asterisk-sounds in the svn.digium server, i ve been got asterisk,
zaptel,libpri,asterisk-addons (1.2 stable version)
Thank You
Younss AZ
KASTERISK.COM
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On Tue, Feb 13, 2007 at 09:29:26AM +, younss azzayani wrote:
Hello,
I can't find asterisk-sounds in the svn.digium server, i ve been got
asterisk,
zaptel,libpri,asterisk-addons (1.2 stable version)
Thank You
apt-cache search -n asterisk-sounds
It should be called
Hi, I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me. How can I search the
character in the end of line?
${REGEX([*]$ ${MACRO_EXTEN})
returns 0
${REGEX(*$ ${MACRO_EXTEN})
returns 0 with error
ango
i found someone that is called asterisk-sounds-main
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when i check this link
http://ftp.digium.com/pub/asterisk/old-releases/; you'll find a lot
of sounnds package realises, can i use one of them
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Hello
when i try to compile zaptel i get this error code line
any Help explain please :)
*
ipbx:/usr/src/zaptel-1.2#uname -r
2.6.8-3-686
ipbx:/usr/src/zaptel-1.2#make linux26
make: *** No rule to make target `linux26` . Stop.
All,
Using SIP with progressinband=yes I get Asterisk to generate the ringing
sound for callers. However, I was wondering if it is possible to
configure what is 'played back' to the calling party? i.e. instead of
just 'ring ring' could I potentially play back a song from an MP3, WAV
or GSM
Hi All,
One of my customer asked me if Asterisk can handle 7000 SIP users. They want
anyone that have access to wireless hotspot to make voice calls to the office
using software phone or SIP cordless phone.
Does anybody did such a setup? What are hardware requirements for server and
how
Copy and paste from my reply to a similar question a couple of weeks ago:
5000 sip registrations is quite a lot, but the more important thing is
the number of simultaneous calls.
If most of your calls is going to be SIP 2 SIP then I would suggest you
use openSER for the SIP registrations and
The 'm' option in the dial command, from memory.
PaulH
On Tue, 2007-02-13 at 23:25 +1300, Ray Jackson wrote:
All,
Using SIP with progressinband=yes I get Asterisk to generate the ringing
sound for callers. However, I was wondering if it is possible to
configure what is 'played back' to
Hi all,
I am looking for phones witch support POE, with a good relation between
quality and price to work with asterisk. I just see the Thompson st2030 and
the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave
you the best results in a productivity enviroment?
Thanks in
Snom 320.
PaulH
On Tue, 2007-02-13 at 12:24 +0100, voip crazy wrote:
Hi all,
I am looking for phones witch support POE, with a good relation
between quality and price to work with asterisk. I just see the
Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this
phones or another
Hi,
Following are the commonly used POE enabled phones
Sipura 942
Snom 320 and 360
GXP2000
Aastra 9133i and 480i
Ahsan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voip crazy
Sent: 13 February 2007 11:24
To:
On Tue, 2007-02-13 at 12:24 +0100, voip crazy wrote:
Hi all,
I am looking for phones witch support POE, with a good relation
between quality and price to work with asterisk. I just see the
Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this
phones or another ones gave you the
hi,
when i type
asterisk-1.4# ./configure
**
i got this error
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
*
# vi config.log
***
...;
cpp: installation
Whatever you take:
Stay away from cisco poe phones unless you're using cisco poe
switches.. and even then. Cisco doesn't always apply the POE standard,
older models are totally not conform the POE standard (they switched
the + and - poles at the socket).
Hi all,
I have installed some Asterisk machine, all with the same problem.
My typical configuration is:
- Asterisk 1.2.14 (or 1.4.0beta3)
- CentOS 4.4 server.
The problem is this:
When I start Asterisk with the default init script (/etc/init.d/asterisk
start) distributed with the source, and
I've just setup about 10 SPA942s. Great phones. Has the look and feel
of a Cisco phone. Documentation for configuring the phone remotely is
not easily accessible but (with the help of the all knowing Google)
found on the Internet. Have them connected to a POE switch also from
Linksys.
On
On Tue, Feb 13, 2007 at 01:35:50PM +0100, Andrea De Vita wrote:
Hi all,
I have installed some Asterisk machine, all with the same problem.
My typical configuration is:
- Asterisk 1.2.14 (or 1.4.0beta3)
- CentOS 4.4 server.
The problem is this:
When I start Asterisk with the default
On Tue, Feb 13, 2007 at 12:30:06PM +, younss azzayani wrote:
hi,
when i type
asterisk-1.4# ./configure
**
i got this error
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
*
#
I've got around 45 people who need to place calls from our inhouse app.
What is the considered best practice for placing these calls:
1) All clients connect to astmanproxy, and use AMI API Originate command
2) All clients connect directly to the astersik AMI and use the API
Originate command
Anybody,
I have download asterisk 1.4 via svn. whem I compiled it, I got the following
error:
/lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing
type-punned pointer
will break strict-aliasing rules
zttranscode.c:37:30: linux/page-flags.h: No such file or directory
If you asked this question on the biz list you would get a lot of people
that will tell you that they offer services where you can set the caller ID
to what ever you want. To name a few::
Nufone
Teliax
Voipjet
- Original Message -
From: Doug Crompton [EMAIL PROTECTED]
To: Asterisk
Sounds like you don't have the gcc-c++ package installed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, February 13, 2007 6:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] error when compiling
Hello all.
I'm setting up a new call center PBX using * v1.4, and figure it's better to
go with AddQueueMember over AgentCallbackLogin. The functionality of
AgentCallbackLogin still works, but without a firm idea of how long it will
be in the codebase, I'm wary of building a system on top of
i cant make apt-get install gcc-c++ it s result nothing
so i typed apt-cache search gcc-c++ no thing also
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Yes, thank you. I found one, callwithus which has excellant Asterisk
support, IAX/SIP and the support actually answered in minutes! So far good
connects (usig IAX) and good prices. Lets hope it stays that way.
I wonder why more companies can't be like that. This callerID thing is
stupid. If you
On Tue, Feb 13, 2007 at 02:37:05PM +, younss azzayani wrote:
i cant make apt-get install gcc-c++ it s result nothing
so i typed apt-cache search gcc-c++ no thing also
What do you have on /etc/apt/sources.list ?
--
Tzafrir Cohen
icq#16849755
I have installed both the Cisco 79xx and Linksys SPA942. I
personally have Cisco on my desk because there is quite a bit of
difference in the look and feel. Also the sound quality is better on
the Cisco phones.
The SPA942 is a nice phone for the price but is lighter and smaller
than the
when i type cat /etc/apt/sources.list | grep gcc-c++ i got nothing
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sorry,
#cat /etc/apt/sources.list
#deb file:///cdrom/ sarge main
deb ftp://ftp2.fr.debian.org/debian/ stable main
deb-src ftp://ftp2.fr.debian.org.debian/ stable main
deb http://security.debian.org/ stable/updates main
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On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
Lee Jenkins wrote:
Stefano Corsi wrote:
[snip]
The nice things about GUI's in my opinion is that routine chores
such as
setting up extensions, dialing extensions, hunt groups, etc. are less
likely to contain scripting bugs or typos. The
Hello All,
Hoping all of you might have an additional option for me to try at this
point. :)
My Goal:
To have a paging option that does the following When I press **_
it will send a ring-answer page to that person. The person on the other
end should be muted, so if they are in a
From: Rilawich Ango [EMAIL PROTECTED]
Date: Tue, 13 Feb 2007 17:43:05 +0800
Hi, I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me. How can I search the
character in the end of line?
${REGEX([*]$ ${MACRO_EXTEN})
returns 0
You must
From: Paul Hales [EMAIL PROTECTED]
Date: Tue, 13 Feb 2007 22:15:57 +1100
The 'm' option in the dial command, from memory.
PaulH
Also search for early media - I'm under the impression that you may not need
progressinband, as it is often undesirable.
Yuan Liu
On Tue, 2007-02-13 at 23:25
Try yum install gcc-c++
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: Tuesday, February 13, 2007 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] error when compiling asterisk-1.4
i
Hi.
I'm trying to get digits form the user via agi
something like this: this only should print result=asciicode
but none of the functions even wait until timeout ..
they just pass .. (after a nanosecond)
the las print is always timeout.
Any clue ..?
my $callerid = $AGI{'callerid'} ;
We are currently working to trunk from a Nortel 81C to an Asterisk
Server 1.4 running on Red Hat Linux. We have two PRI trunks which work
with the exception of the clock slips, which is causing the Nortel to
reset the PRIs once a hour. Thanks for any suggestions.
81C MSDL
but i can't work with yum am using debian not RH
2007/2/13, Savoy, Kevin - Williston, ND [EMAIL PROTECTED]:
Try yum install gcc-c++
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of younss
azzayani
Sent: Tuesday, February 13, 2007 8:37 AM
To: Asterisk
Does anyone have a solution to allow an agent to selectively end his
wrap-up time? We define a wrap-up time of 60 seconds to allow our
agents to finish their notes from a call. In some cases, the full 60
seconds is not needed and our agents would like to end their wrap-up time.
Thanks,
Jay
No one knows what the Notify answer on an owned channel is?
Anyone?
-Original Message-
From: Savoy, Kevin - Williston, ND
Sent: Monday, February 12, 2007 11:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] After upgrade to 1.4 transfers
this is the solution
apt-get install g++
it's work know
thnank you (all of you) :)
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On Tue, Feb 13, 2007 at 10:23:17AM -0500, Tom Rymes wrote:
On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
Lee Jenkins wrote:
Stefano Corsi wrote:
[snip]
The nice things about GUI's in my opinion is that routine chores
such as
setting up extensions, dialing extensions, hunt
Tom Rymes wrote:
On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
Lee Jenkins wrote:
Stefano Corsi wrote:
The nice things about GUI's in my opinion is that routine chores such as
setting up extensions, dialing extensions, hunt groups, etc. are less
likely to contain scripting bugs or
[EMAIL PROTECTED] wrote on Tuesday, February 13, 2007 10:17
AM
We are currently working to trunk from a Nortel 81C to an Asterisk
Server 1.4 running on Red Hat Linux. We have two PRI trunks which
work with the exception of the clock slips, which is causing the
Nortel to reset the PRIs once
Paul Hales wrote:
On Mon, 2007-02-12 at 22:14 -0800, George Pajari wrote:
On Tue, 2007-02-13 at 16:24 +1100, Dennis Kavadas wrote:
Hi all
I'm after a Digium card that will allow me to connect an Asterisk box to..
2 x sip providers
1 x company PBX
1 x POTS provider.
Can
voip crazy wrote:
Hi all,
I am looking for phones witch support POE, with a good relation between
quality and price to work with asterisk. I just see the Thompson st2030
and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones
gave you the best results in a productivity
Hi for all
I'm making some test and I can see an incorrect behaviour.
I have two asterisk with an IAX2 trunk. In asterisk 1 I have a queue and an
agent and, in Asterisk 2 I have three clients. When the clients make calls
to an asterisk 1, its calls entry in the queue. While they are waiting, an
On Tue, Feb 13, 2007 at 09:53:17AM -0700, Stephen Bosch wrote:
Tom Rymes wrote:
On Feb 12, 2007, at 7:13 PM, Stephen Bosch wrote:
Lee Jenkins wrote:
Stefano Corsi wrote:
The nice things about GUI's in my opinion is that routine chores such as
setting up extensions, dialing
I have experienced similar problems with AGI some time ago... sometimes,
the script just get to the 'WAIT FOR DIGIT' function and the streamed
audio file before it is a little 'long' and the timeout runs while you
are still listening the audio... Try testing with a very large timeout
number,
make a 'updatedb' , and look for 'page-flags.h' , i think that you might
be missing that file,
J. Espinal,
[EMAIL PROTECTED] wrote:
Anybody,
I have download asterisk 1.4 via svn. whem I compiled it, I got the following
error:
/lib/modules/2.4.33.3/build/include/asm/system.h:190:
Stephen Bosch wrote:
voip crazy wrote:
Hi all,
I am looking for phones witch support POE, with a good relation between
quality and price to work with asterisk. I just see the Thompson st2030
and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones
gave you the best results in a
Ciao Andrea,
Tzafrir is right...safe_asterisk is not very good. I had a discussed
with him time ago. It is better for you to develop it by yourself. I
made a version checking PIDs which is what safe_asterisk is lacking.
Giorgio Incantalupo
Andrea De Vita wrote:
Hi all,
I have installed
You have the PRIs set up to recover clock from the Asterisk box, is that
what you want? If so, you certainly do *not* want span=1,1,0 or 2,2,0
since that will make Asterisk think the 81C should be clock master. Are
there any telco-timed PRIs somewhere? If so, set up the PRIs on the 81C
to be
Hello all,
I have been unsucessfully looking for conclusive information regarding
blocking collect calls in Brazil, using either MFC/R2 or ISDN E1 lines. So, I
would like to ask the list a couple of questions.
Regarding MFC/R2, there seems to be a patch for the chan_unicall module
On Feb 13, 2007, at 11:53 AM, Tzafrir Cohen wrote:
On Tue, Feb 13, 2007 at 10:23:17AM -0500, Tom Rymes wrote:
[snip]
Not to start a flame-war, but I completely disagree. Troubleshooting
a GUI is much easier, given that you don't have to scout for typos,
transposed numbers, etc throughout
On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote:
make a 'updatedb' , and look for 'page-flags.h' , i think that you might
be missing that file,
under the include/ directory in the linux kernel source directory.
J. Espinal,
[EMAIL PROTECTED] wrote:
Anybody,
I
I have made the change as stated
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
Where is your primary clock?
For the PRI, coming from the Asterisk PBX
Do you have an atomic clock supplying timing to your Nortel system?
No, timing picked up over the PRI from the provider
Does your Nortel get it's
These are common functions. Why Asterisk Manager doesn't provide
commands to hold and retrive an active channel?
If it must be implemented by AGI, could anyone give a direction or
steps?
Thanks in advance,
James
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I have asterisk 1.4.0 running.
I have a UIP200 that beeps when a second call is incoming.
I flip over with flash talk to that person then
hit flash to go back. The person is there for a short time
then the call is DROPPED.
This worked fine with this phone and 1.2.X
Is this a bug or anyone else
voip crazy wrote:
I am looking for phones witch support POE
Aastra 9133i and Aastra 480i
--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
www.netvoice.ca www.ip-centrex.ca
Hi .
I had the same problem but downloaded a test script and wait for digit
worked.
the only visible difference is that I wat nos using strict,
so I am rewriting the AGI with
use strict;
Hope this help.
On 9/14/06, Joel Lansden [EMAIL PROTECTED] wrote:
Hello all,
I have been trying to
Hi,
What's your favorite switchboard application software, for a 100-200 seats
company attendant ?
The setup is :
PSTN --- Asterisk - SIP Phones
|
- Attendant console
Has anyone used Icecom switchboard,
I have tried to compile asterisk with zaptel:
./configure --with-zaptel=/usr/src/zaptel-1.4.0
make
make install
however when I run asterisk it says that the zap command is missing.
What am I doing wrong? I have compiled and installed zaptel fine and it
is recognizing my card.
Thanks
On 13 Feb 2007, at 12:56, Julian Lyndon-Smith wrote:
I've got around 45 people who need to place calls from our inhouse
app. What is the considered best practice for placing these calls:
1) All clients connect to astmanproxy, and use AMI API Originate
command
2) All clients connect
Larry Shields wrote:
I recently read about the following new technologies from Digium. Has
anyone tried the new HPEC or knows when it will be available?
It's out now, and I've tried it - the difference between HPEC and MG2
from trunk is stunning - in situations with bad echo where MG2 can take
I am developing an ACD front end using Asterisk 1.2.14. I heard that
AgentCallBackLogin will be deprecated in future version of *.
Is this true? If it is, how can I use AddQueueMember to replace
AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have
multiple queues and
Will this work with SIP channels? I get zero echo out the PRI but I do
get it occasionally on a LD provider (SIP) we use. The stock * install
doesn't appear to be doing anything stopping echo on those channels.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi,
If I have a PRI with 23 channels on it.Can I setup Asterisk to start
outbound calls at 23 and hunt back to 1? I know I can individually do it
with gX/23/5551212 (or something along those lines). But is there a way to
make it hunt FROM 23 down to 1. By default it starts at 1 and hunts
g hunts low to high
G hunts high to low
John Novack
Matt wrote:
Hi,
If I have a PRI with 23 channels on it.Can I setup Asterisk to
start outbound calls at 23 and hunt back to 1? I know I can
individually do it with gX/23/5551212 (or something along those
lines). But is there a way to
Hi all, is it possible to to dumb down a FRITZ!Box Fon ata (
http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html
http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_ata/index.html##)
and have the two FXS ports AND the ISDN interface register with
Asterisk. In much the same way a
Oh interesting. I don't recall seeing that documented anywhere. Thanks!
On 2/13/07, John Novack [EMAIL PROTECTED] wrote:
g hunts low to high
G hunts high to low
John Novack
Matt wrote:
Hi,
If I have a PRI with 23 channels on it.Can I setup Asterisk to
start outbound calls at 23
At 10:09 11/02/2007 -0500, Gordon Henderson wrote:
Check the processor spec. carefully. [...] Also make sure you compile
asterisk for an i586
OK, I'll make sure it has enough cache and I'll recompile the code myself.
I'm thinking of getting an ML 8000
Where can I get a starting point for setting up sms via VoIP and via web.
I want to send SMS from VoIP or web to VoIP phones and GSM phones.
1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which phones can be used?
bye
Ronald
[EMAIL PROTECTED]:/usr/src/linux/include/linux$ pwd
/usr/src/linux/include/linux
[EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb
make: *** No rule to make target `updatedb'. Stop.
[EMAIL PROTECTED]:/usr/src/linux/include/linux$ ls -la page-flags.h
/bin/ls: page-flags.h: No such
Bill Gibbs wrote:
Will this work with SIP channels? I get zero echo out the PRI but I do
get it occasionally on a LD provider (SIP) we use. The stock * install
doesn't appear to be doing anything stopping echo on those channels.
Nope, it won't help - echo cancellation needs to be performed
On Wed, Feb 14, 2007 at 07:17:32AM +0800, Ronald Wiplinger wrote:
Where can I get a starting point for setting up sms via VoIP and via web.
I want to send SMS from VoIP or web to VoIP phones and GSM phones.
1. how to set-up?
2. which smsc should I use? (what is the price?)
3. which phones
We use a lot of mini-itx pc's, including the pCI slot. I don't think any of
the systems have shared an irq with the PCI slot
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vincent
Delporte
Sent: Tuesday, February 13, 2007 5:29 PM
To:
Ray Jackson wrote:
Using SIP with progressinband=yes I get Asterisk to generate the
ringing sound for callers. However, I was wondering if it is possible
to configure what is 'played back' to the calling party? i.e. instead
of just 'ring ring' could I potentially play back a song from an
Interesting,
Is this just a more advanced software echo canceller or software with
hardware hooks or software with hardware assisted processing?
How would it compare to a true hardware echo canceller like the one Sangoma
uses. Besides the extra CPU cycles required.
-Original Message-
Look in the list archives, I have posted a solution a while back. It
involves changing the source and recompling.
On 2/13/07, Rob Schall [EMAIL PROTECTED] wrote:
Hello All,
Hoping all of you might have an additional option for me to try at this
point. :)
My Goal:
To have a paging option that
http://lists.digium.com/pipermail/asterisk-users/2006-August/163798.html
On 2/12/07, Steve Davies [EMAIL PROTECTED] wrote:
On 2/12/07, Rob Schall [EMAIL PROTECTED] wrote:
Steve,
I posed a similar question to Shane, but maybe you'll know as well..
I was able to get app_page to work. So when
Hi:
Say I want to build an IVR application which sends an SMS message to a
mobile telephone when the caller responds to a prompt in certain way.
I think I can manage the part about generating the message and building
something to actually send it. The part I'm foggy about is: how would I
by your .ca address I assume your in Canada..
both Telus and Rogers have a email-to-SMS gateway...
Stephen Bosch wrote:
Hi:
Say I want to build an IVR application which sends an SMS message to a
mobile telephone when the caller responds to a prompt in certain way.
I think I can manage the
Yeah, it's hard to know what it would be filed under. However, if you
use zap trunks then you'll want to know about this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+Channels
BTW, see Dialing a Group for specifics on 'g' vs. 'G' as well as other
cool stuff.
-MC
Singer Wang wrote:
by your .ca address I assume your in Canada..
both Telus and Rogers have a email-to-SMS gateway...
Well, those are notoriously unreliable. I've had messages take hours to
arrive when sent by the email-to-SMS gateway. I was kinda hoping for
something more direct. Rogers
shadowym wrote:
Interesting,
Is this just a more advanced software echo canceller or software with
hardware hooks or software with hardware assisted processing?
A more advanced software canceller (there's no magical thing that makes
hardware echo cancellers better, it's still software, but
James Zhang wrote:
These are common functions. Why Asterisk Manager
doesn't provide commands to hold and retrive an active channel?
If it must be implemented by AGI, could anyone give a direction or steps?
Sure the Manager API provides all thing to do that.
Maybe you are just using the wrong
On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote:
Singer Wang wrote:
by your .ca address I assume your in Canada..
both Telus and Rogers have a email-to-SMS gateway...
Well, those are notoriously unreliable. I've had messages take hours to
arrive when sent by the email-to-SMS
Quoting Patrick [EMAIL PROTECTED]:
On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote:
Singer Wang wrote:
by your .ca address I assume your in Canada..
both Telus and Rogers have a email-to-SMS gateway...
Well, those are notoriously unreliable. I've had messages take hours to
arrive
Thanks good info on that page.At times I think the wiki has grown out of
control. There is almost too much info there... that even with a search
engine you can miss some. Oh well.. Thanks for the pointer.
On 2/13/07, Michael Collins [EMAIL PROTECTED] wrote:
Yeah, it's hard to know what
Does anyone have any experience with any SIP or IAX providers that
support E911? I'd love to convert entirely to Asterisk at my house,
but the lack of emergency dialing has been a major hold-up for me.
Thanks in advance for any suggestions!
--
Kyle Sexton
Nic Bellamy wrote:
Ray Jackson wrote:
Using SIP with progressinband=yes I get Asterisk to generate the
ringing sound for callers. However, I was wondering if it is possible
to configure what is 'played back' to the calling party? i.e. instead
of just 'ring ring' could I potentially play
Quoting Ray Jackson [EMAIL PROTECTED]:
hey, why not just play a wav or mp3 of the other person talking instead of
actually placing a call at all ? Wouldn't that be even cooler ? :)
reminds me of the old dial 811 to hear a duck quack - someone with too much
time on their hands.
Nic
Hi Kyle,
Vitelity.net does it for me... There are a few others too. I tried a
half dozen, but none seem to have the elusive Customer Service, E911,
and good Voice quality. I use multiple providers. Les.net is great for
everything but E911. Origination-- Les.net . Termination-- Les.net,
At times I think the wiki has grown out of control.
I hear you. I'd pay money to anyone willing to create and maintain a
master index!
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, February 13, 2007 7:09 PM
To: Asterisk Users Mailing
Drop me an email
I know some GSM Gateway that has a direct serial port for SMS
Sam
-Original Message-
From: Jon Pounder [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 14, 2007 10:50 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMS from Asterisk
Hello All
This month we would like to offer our GSM Gateway range for less to clear up
some spaces.
CT-GSM-1000 Basic GSM Gateway (RJ11) Single Sim £69
CT-G01GSM Gateway with SMS Feature (RJ11) Single Sim £99
CT-G04GSM Gateway (RJ11) Quad Sims
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