I am trying to find out if Polycom (I am using IP601) can display the
speed dial list using last name first instead of first name first.
Currently, the speed dial list displays first name first.
Thanks.
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If you are an MFC/R2 user and want to help in the development of
chan_zap support for this signalling, please take a look at the
bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact
me. Currently just México support is built-in, if you want your
country variant supported, drop me
I will be out of the office starting 04/23/2008 and will not return until
04/29/2008.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Just want to know if anyone has used instant messaging using Polycom and
Asterisk.
From Google, I did not really see IM being mentioned at all. It appears
no one is interested to implement it in Asterisk. Or I guess people
would rather use Jabber or other IM messengers.
Hello,
Is it possible to somehow fork in the dialplan? Say a call comes in. Then I
want to wait 30 seconds and then write in a database, but at the same time
while I wait I want to go on with other commands too.
Thanks,
Best regards,
Tobias
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Hi,
I have softphone with a g723 codec, my question is how do i set it as Pass
thru in Asterisk?
cheers,
Aby Azid
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what kind of command do you want it to do in the background? The obvious
answer your question would probably be to use an agi script.
On Thu, Apr 24, 2008 at 11:51 AM, Tobias Ahlander [EMAIL PROTECTED]
wrote:
Hello,
Is it possible to somehow fork in the dialplan? Say a call comes in. Then
I
I use click2call. http://www.geocities.com/babarnazmi/index2.htm
It is an activex control though.
All of my testing has shown it be be pretty clean.
We have it on our contact us page of our website and we also give that url to
overseas (India, Germany, Japan) contacts and some
have used it.
I use click2call. http://www.geocities.com/babarnazmi/index2.htm (it is really
a click to talk, as I removed the dialing
capabilities and hardcoded the extension)
It is an activex control though.
All of my testing has shown it be be pretty clean.
We have it on our contact us page of our
Hello Moisés, thanks for your effort on this! I would love to use Digium cards
for MFC/R2 signalling in the future.
I added some info you might like in the bugtracker, you might take a look at it.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Moises Silva
There are much better solutions than doing a RAM drive. While it may
be stable (not in my experience, I advise using different servers for
different tasks (with redundancy obviously). A phone switch should be
just that, a recording server should also be just that (in demanding
You should also look at Darren's ASTPP, I am not sure if you missed
that earlier in the thread. It is basically ASTCC with major
improvements. It even has the ability to tie into OSCommerce which in
turn can connect to several credit card merchant accounts.
It is much more robust than ASTCC.
Do you have an example of it working on your website?
When I try the click2call websitenone of the demo's actually work?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED]
You can call an AGI script that will call another script. That last one would
wait 10 seconds and write in the database. The following example works for me:
/var/lib/asterisk/agi-bin/agi-test.agi:
#!/bin/bash
nohup /root/helloworld.sh 1/dev/null 2/dev/null
exit 0
/root/helloworld.sh:
Moises
Thats means, that we arent going to use unicall?
If that true i can test these weekend with a E1-Axtel.
Thanks
Ruben
Moises Silva escribió:
If you are an MFC/R2 user and want to help in the development of
chan_zap support for this signalling, please take a look at the
bugtracker at
allow=g723.1 or allow=g723 (I don't remember which).
aby azid wrote:
Hi,
I have softphone with a g723 codec, my question is how do i set it as Pass
thru in Asterisk?
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network
By your experience, please someone tell me which T.38 capable VoIP SIP
providers have you tested with success sending and receiving FAX with
Asterisk 1.4.
Thanks,
Ricardo Carvalho.
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I think I'm going to go about this a different way, if it works I'll post my
solution.
Essentially I'm going to limit the calls by grouping(didn't know you could use
categories until I did the research) and math. Limiting our corporate office
to 10 IAX calls, both incoming and outgoing
- Tobias Ahlander [EMAIL PROTECTED] escreveu:
Is it possible to somehow fork in the dialplan? Say a call comes in.
Then I want to wait 30 seconds and then write in a database, but at the
same time while I wait I want to go on with other commands too.
On Thu, 24 Apr 2008, Vin??cius
Atis Lezdins wrote:
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably
On Wednesday 23 April 2008 18:26, Brian J. Murrell wrote:
On Wed, 2008-04-23 at 08:52 -0500, Tilghman Lesher wrote:
Please understand that that's NOT the only security fix that has gone in
during that time. If this is the only thing that you fix, you're likely
to be vulnerable on several
On Thursday 24 April 2008 03:51, Tobias Ahlander wrote:
Is it possible to somehow fork in the dialplan? Say a call comes in. Then I
want to wait 30 seconds and then write in a database, but at the same time
while I wait I want to go on with other commands too.
There isn't a fork, but there is
Atis Lezdins wrote:
Atis Lezdins wrote:
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller
In article [EMAIL PROTECTED],
Atis Lezdins [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6
More importantly, for it to pass-through you need something that
processes g723 on the other end. If Asterisk is terminating the call by
handing it off to the PSTN or to another phone that does not do g723
then Asterisk must transcode and that requires the license.
Eric Wieling wrote:
On Thu, 2008-04-24 at 09:13 -0500, Tilghman Lesher wrote:
Check the archives.
Indeed, you are correct. My apologies. I forgot that I temporarily
unsubbed from the -users list for a period of time where I was just
getting too much volume of e-mail and asterisk-users had to be one of
the ones
I have a macro that checks to see if a dundi route exists, if it does it
attempts to dial it. The remote end can set the chan as unavailable, or busy.
If it does the call immediately hangs up instead of returning to the macro for
more processing. Is there a way to force it to return?
Logic
Nevermind, helps when you reload the diaplan at BOTH ends :)
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann
Sent: Thursday, April 24, 2008 9:48 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Macro/Goto Help
I have a macro
Hello Moisés, thanks for your effort on this! I would love to use Digium
cards for MFC/R2 signalling in the future.
Currently you can use Digium cards with Unicall :-) , tho, having
MFC/R2 on chan_zap is more handy.
I added some info you might like in the bugtracker, you might take a look
Hello Ruben,
Yes, if you consider using R2 support in chan_zap Unicall is no longer
required. I will be not available online this weekend, please let me
know your feedback after your try it. We can also meet via MSN so I
can assist you in testing the next weeked (3-4 May).
Thanks for the help.
Gafachi is the only one we have had success with for T38 fax.
Jeff Johnson
NeturallySpeaking
Enterprise VoIP solutions at Small Business Prices
(866) 448-0038 ext 103
(813) 774-3570 direct
(813) 655-9049 fax
www.neturallyspeaking.com blocked::http://www.neturallyspeaking.com/
Every CPU core shows up as a separate CPU under Linux. For those that
have hyperthreaded processors, a single core processor will show up as
two processors - assuming you have hyperthreading enabled.
linuxian iandsd wrote:
top says asterisk 1.2.25 is using multiple cores:
Cpu0 :
LinkedIn
Asterisk,
I was playing around and found some option to cross-reference all gmail
contacts and linkedin people. It's a weird, enlightening list, so I figured I'd
check the boxes of people I actually might know (i.e., not random HR people,
website admins, tech
Dear Brian,
On Thu, Apr 24, 2008 at 08:23:29AM -0700, Brian Nehring wrote:
I was playing around and found some option to cross-reference all
gmail contacts and linkedin people. It's a weird, enlightening list,
so I figured I'd check the boxes of people I actually might know
(i.e., not
Hello
99% of all my users are calling from GSM phones, and my system
basically just plays some sound files back.
The PBX is connected to an ISDN-30 connection. Are there any modules
for playing MP3 files, so I can use them with commands like Play() and
Background()?
And will it have any effect
Hi All -
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm not able to get DTMF
working. I've tried using inband, rfc2833 and auto, and none of them
work. Maybe I'm missing something obvious? Here's my config:
Asterisk1
On Thu, 2008-04-24 at 17:50 +0200, harry wrote:
The PBX is connected to an ISDN-30 connection. Are there any modules
for playing MP3 files, so I can use them with commands like Play() and
Background()?
If I were you, I'd transcode the files to alaw and play back the alaw
version, so that
2008/4/24 Jared Smith [EMAIL PROTECTED]:
On Thu, 2008-04-24 at 17:50 +0200, harry wrote:
The PBX is connected to an ISDN-30 connection. Are there any modules
for playing MP3 files, so I can use them with commands like Play() and
Background()?
If I were you, I'd transcode the files to
Hi Harry -
99% of all my users are calling from GSM phones, and my system
basically just plays some sound files back.
The PBX is connected to an ISDN-30 connection. Are there any modules
for playing MP3 files, so I can use them with commands like Play() and
Background()?
See
On Thu, 2008-04-24 at 12:02 -0400, Noah Miller wrote:
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm not able to get DTMF
working.
If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it
appears that you
Hi All,
I'm trying to configure a ringgroup, which will ring the extension in the
group one by one. this is what i tried on my extension.conf
[macro-dial-ringgroup]
exten = s,1,Dial(SIP/${ARG1},15)
exten = s,n,NoOp( Dial Status: ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten
I use 1ezphone because its not activex and works all operating systems
and browser.Plus the codec is great and only uses 10k
- Original Message -
From: Steven
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Best Click-to-call client
Date: Thu, 24 Apr 2008
I upgraded one of our servers to 1.4.19.1 last evening, but ended up
having to drop back because of IAX calls failing at a near 50 % rate.
Here is the message that we would receive on the console (multiple
times), and then it would hangup the call.
Avoiding IAX destroy deadlock
Anyone else
Dinesh Nair пишет:
On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
The best option is to put a SIP Proxy in front of your Asterisk sever
and block REFER requests.
or just comment out the block in chan_sip.c which handles the refers.
Thanks to your answers, but i found
Hello Moisés, thanks for your effort on this! I would love to use
Digium cards for MFC/R2 signalling in the future.
Currently you can use Digium cards with Unicall :-) , tho, having
MFC/R2 on chan_zap is more handy.
Way more handy and will be much more reliable too. Steve Underwood did a
Hi,
I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the
specs of various cards I favor the Digium B410P and Sangoma A502D
because of hardware echo cancellation. Does anyone have any experience
with either card, good or bad? Which one would you choose and why?
Thanks for your
Hi...
Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the
asterisk version is 1.4.18.
If in the machine is all ok, i can stop start the asterisk service no prob,
my problem is when in another server (in my case, debian etch 4) using the
ssh the stop service is ok, but the
I have a box running a TE410P with echo cancelling and it works like a charm.
Set up once, forget about it.
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Patrick [EMAIL PROTECTED] escreveu:
Hi,
I need to setup an Asterisk box with 4x ISDN BRI links.
Way more handy and will be much more reliable too. Steve Underwood did a
great job implemeting it, but as far as I know the code isn't actively
maintained anymore. Of course your implementation of MFC/R2 will take a while
to become stable, but hey -- it's a start.
Agreed.
Russel pointed
Hi Jared -
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm not able to get DTMF
working.
If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it
appears that you are), you'll need to set
On Thu, Apr 24, 2008 at 05:01:53PM +0100, Bruno Pereira wrote:
Hi...
Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the
asterisk version is 1.4.18.
If in the machine is all ok, i can stop start the asterisk service no prob,
my problem is when in another server (in my
hi,
thanks for replying guys, I have a digium transcoder card installed and its
running on mixed mode. The softphone I have, is using g723.1 6.3k while the
transcoder card is using g723.1 5.3k...so it has different payload size..FYI
im using softphone from Adore. The guy from the Adore support
On Thu, 24 Apr 2008, Bruno Pereira wrote:
ssh etx9 'sudo /etc/init.d/asterisk start'
[EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk start'
start ini
Starting asterisk: [ OK ]
decrease the verbosity level to zero: OK
start fim
and just stays there,
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of the
Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4, and
1.6 soon). It works pretty well. In fact, it works more stable in 1.4 than the
original Steve driver in 1.2, and with better sound under
On Thu, Apr 24, 2008 at 11:23:21AM -0700, Steve Edwards wrote:
On Thu, 24 Apr 2008, Bruno Pereira wrote:
ssh etx9 'sudo /etc/init.d/asterisk start'
[EMAIL PROTECTED]:~$ ssh etx9 'sudo /etc/init.d/asterisk start'
start ini
Starting asterisk: [ OK ]
decrease the verbosity level to
Came upon a problem today that I thought I'd see if it's by design, if
I'm missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with
others, but for some reason when they would transfer an outside call
anywhere it would
I'm having a problem at a custom site where GotoIfTime doesn't seem to be
working for some reason. I had putty running and logging CLI output and below
is the call data:
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Ringing(Zap/3-1, ) in new stack
-- Executing Wait(Zap/3-1, 0) in
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of
the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4,
and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4
than the original Steve driver in 1.2, and with better sound
On 15:43, Thu 24 Apr 08, Lee Jenkins wrote:
I'm having a problem at a custom site where GotoIfTime doesn't seem to be
working for some reason. I had putty running and logging CLI output and
below
is the call data:
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing
Lee Jenkins wrote:
I'm having a problem at a custom site where GotoIfTime doesn't seem to be
working for some reason. I had putty running and logging CLI output and
below
is the call data:
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Ringing(Zap/3-1, ) in new stack
--
Lee Jenkins wrote:
-- Executing GotoIfTime(Zap/3-1,
08:30-17:00|mon-fri|*|*|?daytime_ivr|s|1)
Too many pipes. Mine is:
GotoIfTime(00:00-07:50|mon-fri|*|*?auto-paging,s,1)
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety,
For ABE support you really should contact Digium. BTW, there is no such
thing as a sip trunk. It's a sip peer or connection or account.
Noah Miller wrote:
Hi Jared -
For the first time, I'm setting up SIP trunking between two asterisk
boxes. The calls themselves work fine, but I'm
In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
The doc directory is the only official source of documentation for
Asterisk that I am aware of. Read it.
[EMAIL PROTECTED] wrote:
Dinesh Nair пишет:
On Wed, Apr 23, 2008 at 02:14:27PM -0400, Steve Totaro wrote:
There are much better solutions than doing a RAM drive. While it may
be stable (not in my experience, I advise using different servers for
different tasks (with redundancy obviously). A phone switch should be
just that, a
Hi All,
Quick question.
We have a customer with a T1 located in their data center, and then one TDM
card for local calls at their remote offices.
We would like to remove the local PBX and TDM card and have them register
directly to the main server.
For the remote office, that still uses one
Hello everyone.
I got a little problem in here: I want to set up a queue so that if anything of
these happens:
a) No agents logged in
b) All agents busy
Then the user gets diverted somewhere. I used this (for testing purposes only,
of course):
exten = 7080,1,Answer()
exten =
For ABE support you really should contact Digium. BTW, there is no such
thing as a sip trunk. It's a sip peer or connection or account.
shrug Semantics. IAX connections between two asterisk boxes are
often called IAX trunks. This is the same thing in SIP
flavor./shrug
Also, no offense
We have tested both and they work fine. The Sangoma is much easier to
install as it does not depend on any other driver, you just run
'setup-sangoma' and follow the instructions. You don't have to fiddle
with the linux kernel or zaptel or chan_misdn. It just works. Plus
its more modular.
24 apr 2008 kl. 23.01 skrev Noah Miller:
For ABE support you really should contact Digium. BTW, there is no
such
thing as a sip trunk. It's a sip peer or connection or account.
shrug Semantics. IAX connections between two asterisk boxes are
often called IAX trunks. This is the same
Most times it's easier to find something in google, than in your own
computer :)
2008/4/25, Eric Wieling [EMAIL PROTECTED]:
In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,
in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt
The doc directory is the
Forwarded Message
From: Noah Miller [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re:
Actually, Digium Support has been quite responsive in recent weeks, as
noted on this list 2 weeks ago:
http://lists.digium.com/pipermail/asterisk-users/2008-April/209457.html
We strive to be as responsive as we can, and have had some success on
this front recently. Please give us a
Tzafrir,
I'm sorry: I sent my previous message at your private address, it
was a mistake :-(sorry :-)
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be
using Zaptel (even your sangoma... the script might just be configuring it for
you)...
and btw, software echo cancel happens in the zaptel kernel driver... it has
nothing to do with the hardware (hence why its a
The Sangoma kernel drivers are different than Zaptel, while running
the install script you are asked if you would like to generate the
Zaptel configs but it is not required, you must also run wancfg to
configure the cards beyond the Zaptel configs. The Sangoma drivers
kind of run on top of the
any of you guys have used FOP for drag and drop transfer on 30 40 phones
environment?
how stable is that?
I'm playing with it but so far drag and dropping phone icon to another phone
disconnectes the call.
On Wed, Apr 16, 2008 at 2:02 PM, Lee Jenkins [EMAIL PROTECTED] wrote:
Al lists wrote:
No, it is not the same thing. An IAX2 Trunk is a version of IAX2 that
puts audio from multiple calls between the same two servers into a
single UDP packet. Fewer packets need to be sent so you use the
bandwidth much more efficiency because you don't have the packet header
overhead.
SIP does
Hi Olle -
Actually, there's a large difference between an IAX2 trunk and an IAX2
connection.
The IAX2 trunk multiplexes multiple media streams in one UDP packet,
therefore you can call it trunking. In order for this to work, you
need to enable a zaptel timer source in your system.
any of you guys have used FOP for drag and drop transfer on 30 40 phones
environment?
At one point, I used it for about 35 phones (25 users). I had to
really do some adjusting to the size of the buttons, but it worked
well. I thought it was very useful, as it showed MWI status, and was
great
how stable is that?
The version I used is probably a couple of versions old now, and it
was pretty reliable then. I imagine it would has probably at least
stayed as stabled if not improved a bit.
Mmmm. Me talk well english! At the risk of being redundant and
wasting list resources,
Hi All,
I'm tryng to test different scenarios for followme for different users:
[localext]
exten = 101,1,Set(FM = ALWAYS);
exten = 101,n,Macro(dial-ext|SIP/${EXTEN}|vm-1|moh-101|fm-101);
exten = 101,n,Hangup
exten = 102,1,Set(FM = NEVER);
exten =
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