2008/4/2, Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Apr 02, 2008 at 11:17:05PM +0200, Olivier wrote:
Hi,
Has anyone information about BRI hardware supported by 1.6 libpri ?
In another thread, I was told a basic BRI card with HFC chipset (Bewan
Gazel
128) was supported but I would
Hi,
I am using asterisk-1.4.15, and using AddQueueMember to add SIP
interface to the queue. Each sip interface is member of multiple
queues
The queue does not recognize that an agent is busy and keeps trying to
call the busy agent. I have identified two patches that can fix the
problem, one at
Sangoma cards are using a non-zaptel driver
(see http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation)
How does it compare to zaptel in analog, PRI and now BRI features ?
Regards
___
-- Bandwidth and Colocation Provided by
Dear all,
I am having a very strange problem with VoicemailMain. When using this
application to record unavail, greet, and busy, I an see the corresponding
file gets created in the ../default/SIP # directory. When pressing 1
to confirm the recorded message, the *.wav file gets deleted from the
On Tue, 01 Apr 2008 13:32:28 -0400, Jared Smith wrote:
On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
I call into the dialplan and try to play demo-congrats and I hear
nothing.
Firewall is disabled.
Everything is on the 192.168.1.X network for this simple configuration.
The
Tony Mountifield wrote:
snip /
Does anyone know what it would take to make the GUI compatible with IE
as well as FF?
That's a big question.
There are *so many* inconsistencies with IE. See
http://www.positioniseverything.net/ for a good accumulated list of many
of the bugs and hacks to try
You could always ask IE to emulate firefox
;)
Sorry, couldn't resist ...
Julian
Tony Mountifield wrote:
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of GUI!,
and We recommend using Firefox.
Does
If you cant power off the machine, look for a sip ata or channel bank.
USB/ TDMoE Channel banks:
xorcom.com
spidermux.com/
And for ata's or sip gateways, there are zillions of brands,
Zoa
Ronny Forberger wrote:
Thanks for that. What channel module do I have to use then ?
And can you
Thanks for that. What channel module do I have to use then ?
And can you recommend a card? Are there external ones, as I can't
really power off the machine. ;)
Thanks,
Ronny
--
Message from: Zoa [EMAIL PROTECTED]
Date: Mi 02 Apr 2008 21:54:31 CEST
Subject: Re: [asterisk-users] Analog modem
Hi,
I use asterisk 1.2.23
I have the following issue with transfer:
I call from from sipA to sipB
when sipB press transfer (not blanktransfer) sipA hear the music until sipB
put down the phone, in this time sipC is ringing but sipA don't hear
anything
can you tell me where to lookup the
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of GUI!,
and We recommend using Firefox.
Does this mean that it is known NOT to work under IE7, or just that it is
insufficiently tested to be guaranteed?
Just starting to play around with AsteriskNOW, and tried running conary:
Error reading config file http://rmirror.digium.com/conaryrc: No route to host
Anyone know if this is just a temporary error?
Being rather more familiar with RPM-based system than rpath/conary, I am
wondering whether I
Sounds good.
Thanks I will try.
-Ronny
--
Message from: zoa [EMAIL PROTECTED]
Date: Do 03 Apr 2008 11:22:18 CEST
Subject: Re: [asterisk-users] Analog modem as phone
If you cant power off the machine, look for a sip ata or channel bank.
USB/ TDMoE Channel banks:
xorcom.com
Hi all,
Is there anyway to have Asterisk to play greetings of different language
based on the local/time zone of the user?
I want to have the email body message to be sent using different languages
based on the user's country.
Also, users from different country should be hearing country-specific
asterisk-gui or not i always tell my client from a security stand point ie
is not a safe browser and recommend they migrate to firefox. most do so
willingly due to growing security concerns
On Thu, Apr 3, 2008 at 4:53 PM, Alan Lord [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
snip /
Does
Hi All;
Can I do transfer for the call from zap/1 to zap/2
(both are fxs)? All what I need is to add the t
argument for the Dail function?
And how can I transfer to be in that senario: zap/1
dial a code to transfer for zap/2, once zap/2
answered, then he can talk with zap/1 (where the third
Tony Mountifield wrote:
Just starting to play around with AsteriskNOW, and tried running conary:
Error reading config file http://rmirror.digium.com/conaryrc: No route to host
Anyone know if this is just a temporary error?
It is, the update servers are currently undergoing maintenance and
2008/4/3, Outback Dingo [EMAIL PROTECTED]:
asterisk-gui or not i always tell my client from a security stand point ie
is not a safe browser and recommend they migrate to firefox. most do so
willingly due to growing security concerns
Maybe but as OP's question remains unanswered : Tony, have
On Thu, Apr 03, 2008 at 10:53:24AM +0100, Alan Lord wrote:
Tony Mountifield wrote:
snip /
Does anyone know what it would take to make the GUI compatible with IE
as well as FF?
That's a big question.
There are *so many* inconsistencies with IE. See
What does it take to get ztdummy to work correctly?
I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel
1.4.9.2
Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs.
Problem is playback() does not work. So then I stop zaptel, asterisk
runs and playback()
Olivier wrote:
Maybe but as OP's question remains unanswered : Tony, have you tried IE
8 beta ?
The warning message should remain but I heard IE 8 standard compliance
improved a lot ...
The current AsteriskGUI (in AsteriskNOW) has a large number of problems
with IE (all versions) and is
It might be worth pointing out too that you can't use Firefox 2.x on OS X.
I had to solely use Ff on Win2k to do anything at all with the GUI. I
hadn't tried linux. YMMV.
peter
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Tony Mountifield wrote:
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of GUI!,
and We recommend using Firefox.
Does this mean that it is known NOT to work under IE7, or just that it is
On Apr 2, 2008, at 9:22 PM, Al lists wrote:
Bad memories from AudioCodec :)
Second this.
My favorite is Vega, but they have terrible support in US.
Have many Adit600 connected via Digium T1 - work great. Even FAX if
PSTN PRI connected to same card.
And no the Adit600 is not a switch,
In article [EMAIL PROTECTED],
Roderick A. Anderson [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
a message at the top Your browser is not supported by this version of
GUI!,
and We recommend using Firefox.
On Thu, Apr 03, 2008 at 06:06:19AM -0700, Roderick A. Anderson wrote:
IE was still on the desktop because they had to support a lot
of customers that used IE but for in-house stuff it slowly became a
Firefox place.
That's no excuse.
That's what IETab's for.
--
Tzafrir
Hello
I assume it's possible to do this with Asterisk: To train a new
worker who works remotely, I'd like to have him listen in on support
calls so that he gets to learn the kind of problems that come in, and
how they're solved.
When a call comes in and the support person thinks it's
Dean Collins wrote:
http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analytics.html
http://bp3.blogger.com/_jmYevHrBr6M/R_Pq-vgIjvI/Af0/PgE_8gFqrY8/s1600-h/World%2Bpopulation%2Bawake.png
Totally stumbled across this really interesting post
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
What does it take to get ztdummy to work correctly?
I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel
1.4.9.2
Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs.
Problem is
In article 47F4C604.1060301 at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users,
Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
/ What does it take to get ztdummy to work correctly?
//
// I have a new laptop HP
On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
Jerry, the first thing to check is cat /proc/interrupts and see if there
is an entry for rtc on IRQ 8. There should be, and the interrupt counts
on there should be going up at approximately 1024 per second.
To see it better:
You mean like FOP?
http://www.asternic.org
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Vincent
Sent: Thursday, 3
Look at the ChanSpy Application. It would be the easiest to implement
and it also allows the trainee to speak to the support person without
the customer knowing.
You can also use on-demand recording or simply record ALL calls
(legality and disclosure to calling parties are outside the scope of
I've used Adit600's almost exclusively for my installs. All have worked great
for me.
-D
From: [EMAIL PROTECTED] on behalf of Steve Totaro
Sent: Thu 4/3/2008 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Its called Flash Operator Panel or FOP. It is install with freepbx, but
I think you can use it as a standalone app.
Jonn
Vincent wrote:
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Thank you.
Wow Drew, I had no idea someone from Oanda was a subscriber to the
Asterisk list (and therefore an asterisk user company?).
Just wanted to say you guys run a fantastic sight and I've been a long
time user for at least the last 2 years.
Now for the irony part of your email. I found it
Just Google Quintum Tenor AX. Well worth the money.
Thanks,
Steve Totaro
On Mon, Mar 31, 2008 at 10:03 PM, Al lists [EMAIL PROTECTED] wrote:
Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
Steve, what are my options for SIP to fxs?
thank you!
On 3/31/08, Doug Lytle
I am trying to send a DTMF digit automatically every 15 seconds to keep a
call connected to an alarm panel. I tried using the dial command L and
recording a dtmf tone for the beep, but obviously that didn't work. Does
anyone have a suggestion for merging the L option and the sendDTMF or the D
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Thank you.
___
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asterisk-users mailing list
To UNSUBSCRIBE or update
On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
/ Jerry, the first thing to check is cat /proc/interrupts and see if there
// is an entry for rtc on IRQ 8. There should be, and the interrupt counts
// on there should be going up at approximately 1024 per second.
/
To see it
To the Asterisk community,
Well this is as good a time to break cover as any, officially our launch
date isn't until next Wednesday the 9th of April, but based on this
discussion I feel that we have to comment.
We are going to announce CogoBlue, the 3rd generation of Asterisk GUI tools
(well I
Use call file to call out to the Alarm Panel and them put it in a
context that would do this:
[alarm-keepup]
exten = s,1,Answer
exten = s,2,SendDTMF(1)
exten = s,3,Wait(15)
exten = s,4,Goto(s,2)
You did not specify if you needed to do anything other than send the
digit to the alarm panel. If you
On Apr 3, 2008, at 10:32 AM, [EMAIL PROTECTED]
wrote:
uname -a shows x86_64 and Centos 5.1, 2.6.18-53.1.14.el5
You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If it's a lot of rtc errors, then
you'll likely need to upgrade your kernel
Anyone interested in this feature? I have a version 0.1 patch, which
is currently against 1.2.25-bristuffed, but which should port
trivially to almost any version. I am away until Tuesday 8th April,
but if there is enough interest, I will open a new-feature ticket
and upload the patch to the
On Thu, Apr 03, 2008 at 10:18:14AM -0400, Dean Collins wrote:
Wow Drew, I had no idea someone from Oanda was a subscriber to the
Asterisk list (and therefore an asterisk user company?).
Just wanted to say you guys run a fantastic sight and I've been a long
time user for at least the last 2
Yes Dean,
we do use Asterisk at OANDA. We've been running our office on it for 2
years now and the call centre's 1st Asterisk anniversary was April 1!
Glad you like our site, we've just launched FXGame Mobile
(http://fxlabs.oanda.com) for the gadget lover and FXGlobal Transfer for
ex-pat
Tzafrir Cohen wrote:
On Thu, Apr 03, 2008 at 06:06:19AM -0700, Roderick A. Anderson wrote:
IE was still on the desktop because they had to support a lot
of customers that used IE but for in-house stuff it slowly became a
Firefox place.
That's no excuse.
That's what IETab's for.
This
Hi John,
I think my history is well documented within the asterisk community that
moving Asterisk out of the geek zone and into the mainstream business
space is good for everyone.
It's good for customers, and it's good for programmers looking for
funding for the next generation of Asterisk tools
You might try this:
http://www.micpc.com/eventmonitor/
It is php, and you can easily disable what you don't want..
earl
On Thursday 03 April 2008 09:41:52 am Vincent wrote:
Hello
Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?
Update,
Still not sorted, I have checked some tools on the TrixBox and using
the wanrouter I was able to check the voltage on lines.
The three result are when there is no call active
[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 1
--- Voltage Status (FXO,port 0) ---
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
/ Jerry, the first thing to check is cat /proc/interrupts and see if
there
// is an entry for rtc on IRQ 8. There should be, and the interrupt counts
//
/ uname -a shows x86_64 and Centos 5.1, 2.6.18-53.1.14.el5
/
You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If it's a lot of rtc errors, then
you'll likely need to upgrade your kernel to at least 2.6.23.11. That
worked for me.
You
WOW! Is this LONG overdue.
Why this wasn't done initially is beyond me
It has caused so many troubles and questions and posts from folks who
expected Asterisk to at least have a feature that has been in a dial up
modem for 10+ years.
Great job!
Many thanks
John Novack
Steve Davies wrote:
In article [EMAIL PROTECTED],
Norman Franke [EMAIL PROTECTED] wrote:
On Apr 3, 2008, at 10:32 AM, [EMAIL PROTECTED]
wrote:
uname -a shows x86_64 and Centos 5.1, 2.6.18-53.1.14.el5
You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
/ uname -a shows x86_64 and Centos 5.1, 2.6.18-53.1.14.el5
/
You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If it's a lot of rtc errors, then
you'll likely need to
I am sure they have until you for whatever business reason, you needed
to move 24 or 48 phones to a distant campus.
There is very little flexibility with T1 Adit solution. Yes it works
well, no it is not flexible, no you you will never even close to what
the Quintum Tenor AX offers as far as
On Wed, Apr 02, 2008 at 02:40:49PM -0600, Greg Woods wrote:
On Wed, 2008-04-02 at 21:18 +0200, Ronny Forberger wrote:
I want to use a analog V.92 modem to make outgoing (and possibly)
incoming phone call through a standard analog phone line.
When I asked this question, I was basically
Dean:
CogoBlue is currently only available on ISPBX's line of PBX appliances.
You can check out the hardware specs on
http://ispbx.com/product_matrix.shtml
We are growing our distribution channel and are actively looking
for qualified dealers to join us. Potential dealers wanting more
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
I have just installed ztdummy on a new system running 2.6.18-53.1.6.el5,
and it is incrementing fine. I didn't realise there was a newer kernel out;
I'll have to update and try again.
Just updated to 2.6.18-53.1.14.el5
On Thursday 03 April 2008 10:37:32 John Novack wrote:
WOW! Is this LONG overdue.
Why this wasn't done initially is beyond me
It has caused so many troubles and questions and posts from folks who
expected Asterisk to at least have a feature that has been in a dial up
modem for 10+ years.
Hi,
Is it possible for me to detect fax on a sip trunk?
my provider has a fax service that can send/receive
fax.
is it possible that i use a that trunk as a telefax?
meaning i will try to detect if it's a fax, if it is i
will forward it to an extension that can handle fax if
not will forward it
Hi Jerry,
Is that with ztdummy loaded or not? By default, Linux doesn't have
anything
that uses the RTC interrupt, so without ztdummy it will usually stay
at 1.
Once ztdummy and zaptel are loaded, then you should see it incrementing.
If not, that suggests a problem.
I have just
Hi Everyone,
My name is Matt Signorello and I'm responsible for wholesale dealers
sales here at ISPBX. (www.ispbx.com)
ISPBX is a New Jersey based systems developer marketing a series of
solid state Asterisk appliances since 2005. For the last year we've been
working on a new strategy for
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
OK If I modprobe ztdummy then the RTC does start incrementing... to like
9250. then stops...
What now?
Sounds like the module got auto-unloaded due to not being in use.
I have found the most reliable way to invoke
On Thursday 03 April 2008 10:56:30 John Signorello wrote:
CogoBlue is currently only available on ISPBX's line of PBX appliances.
You can check out the hardware specs on
http://ispbx.com/product_matrix.shtml
Now that you've announced your product, could you please move this
discussion to the
ounds like the module got auto-unloaded due to not being in use.
I have found the most reliable way to invoke zaptel/ztdummy is using the
proper init script:
1. In your zaptel source directory, do make config. That will create
/etc/rc.d/init.d/zaptel and the rcX.d links to it.
2.
Matt,
I'm sure I won't be the only one to point out that your posts belongs in
asterisk-biz, not asterisk-users.
In article [EMAIL PROTECTED],
Matt Signorello [EMAIL PROTECTED] wrote:
[...]
however because some dealers having problems with Asterisk NOW
If that is the justification for
On Thu, Apr 03, 2008 at 04:45:27PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
OK If I modprobe ztdummy then the RTC does start incrementing... to like
9250. then stops...
What now?
Sounds like the module got auto-unloaded due
Lol and Asterisk NOW isn't a commercial product - The list nazi's are
out today.
Personally I'm ok with people announcing commercial products once and
once only on the user list and then taking it over to the biz list after
that as not everyone is a subscriber to the biz list.
Anyway - looks
Hi all,
Was curious to if for the 'make menuselect' command, there's a config file
hiding someplace that lets me quickly move a configuration to a new source tree
(much like .config in the kernel trees). I looked around after running
menuselect and compiling, but none of the files stood out
On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger [EMAIL PROTECTED]
wrote:
http://www.micpc.com/eventmonitor/
Thanks guys. I was also thinking of stand-alone apps like Jabber or
something. The call is simply to know if an extension is on- or
offline.
Hi all,
Noticed a curious issue in my testing setup for a faxing system I'm putting
together, but it looks like if I let the lines all sit idle for a few days (no
one uses this yet, so the whole thing really does sit idle until I do testing
on it or something else), something I believe on my
What about building an Adobe AIR application that can do this.
I'm kind of very curious about why developers haven't flocked to the AIR
platform for Asterisk apps yet.
I was looking to fund the development of an 'awareness application' for
asterisk based on AIR last year but this was dependant
I have a voicemail application that users can listen to messages and
leave messages. I am looking for a way to play a beep tone to a user
when a new message is received when they are on the phone.
Here is what I have come up with:
in extensions.conf:
[beepvoicemail]
exten = 1000,1,answer()
Well, it SHOULD have been obvious that IF I had degree of skill I would
have years ago.
Those kind of comments serve no purpose other than to anger and to boost
the already inflated ego of those who make the comments!
This is a USERS list after all!
John Novack
Tilghman Lesher wrote:
On
Hi,
I have been experimenting with NAT and Asterisk a bit now. Though I have
made progress along the way, I have come across the following problem. I'll
really appreciate if anyone can provide me any help or pointers. Thanks!
Successful Scenario:
---
All sorts of NAT calls are
Hi,
I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on
an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,
where an incoming line is plugged and also analog phone plugged to the
FXS port. Am faced with the problems below.
- For conversations between analog
On 4/3/08, Matt Signorello [EMAIL PROTECTED] wrote:
Hi Everyone,
My name is Matt Signorello and I'm responsible for wholesale dealers
sales here at ISPBX. (www.ispbx.com)
Matt,
As some others have already pointed out, this list is for
non-commercial discussion and you shouldn't have
No changes have been made to Asterisk.
CogoBlue is a PBX configuration tool.
It is only available at this time with ISPBX pbx appliances.
Kristian Kielhofner wrote:
On 4/3/08, Matt Signorello [EMAIL PROTECTED] wrote:
Hi Everyone,
My name is Matt Signorello and I'm responsible for
On Apr 3, 2008, at 12:45 PM, [EMAIL PROTECTED]
wrote:
You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If it's a lot of rtc errors, then
you'll likely need to upgrade your kernel to at least 2.6.23.11. That
worked for me.
I'd be surprised if
Vincent wrote:
On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger [EMAIL PROTECTED]
wrote:
http://www.micpc.com/eventmonitor/
Thanks guys. I was also thinking of stand-alone apps like Jabber or
something. The call is simply to know if an extension is on- or
offline.
Not web
Cute :)
I was thinking about getting something more complex developed but yes
FOP is a great product though getting a little old.time for the next
version?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original
On Thu, 3 Apr 2008 09:39:55 + (UTC), [EMAIL PROTECTED] (Tony
Mountifield) wrote:
nothing was shown in the main pane. So there is definitely something
wrong with IE compatibility.
s/ compatibility//
There. I fixed your post :)
--
Godwin Stewart - Horwich IT services
Kristian,
I completely pulled a my-bad and posted to the users list instead of
biz. I even had re-posted to the biz list and apologized for the
mix-up. (This time I originally replied to your questions on the biz,
not the users.. its been a long day..)
To answer you question, we do not modify
I've noticed that FWD updated but IAX is not registering with FWD server; nor
the log-in page exist.
Is FWD still supporting IAX?
How to check their IAX server status?
--
#Joseph
___
-- Bandwidth and Colocation Provided by
On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
What about building an Adobe AIR application that can do this.
Any application that connects to the manager interface can do that. It
can be AIR, or FIRE or GROUND.
The FOP exists and does that.
--
Tzafrir Cohen
On Thu, Apr 03, 2008 at 07:22:55PM +0200, Vincent wrote:
On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger [EMAIL PROTECTED]
wrote:
http://www.micpc.com/eventmonitor/
Thanks guys. I was also thinking of stand-alone apps like Jabber or
something. The call is simply to know if an
FOP is quite clunky!
Also the flash is almost un-usable with a large number of extensions
Would love to see something in PHP/Ajax which could be lightweight and
fast.
We are working on something along those lines which we should be able to
release in a few months.
On Thu, 2008-04-03 at 14:42
On Thursday 03 April 2008 02:59:07 pm faraz wrote:
FOP is quite clunky!
one reason i wrote the event montor... which is in PHP (and Ajax or rather
Ajap) and does not poll the asterisk manager (which in my opinion overloads
asterisk)
Also the flash is almost un-usable with a large number of
Quote Re-reading your answer it appears you provided advice about a topic
where you hadn't actually utilized the product.
No - I asked a question to find out peoples experience.
I did not Offer advice to anybody, I ASKED for advice vis-a-vis where They had
actually
used the product.
And yes -
Drew Gibson wrote:
SNIP
I suspect that this is due to the call
billing structure in Europe. They make the North American telcos look
positively philanthropic.
Yes indeed!
Flat rate calling plans? What are those?
Flat rate Mobile Internet? non-existant!
We pay per minute/SMS/MB on every
I wanted to try AsteriskNOW plus a few others to see which I can wrap my
head around the quickest.
The issue so far is I can't figure out how to use my Vitelity account
with it. I went so far as to put their Asterisk configuration in the
sip.conf file but still no joy.
Any pointer as to
Do anyone has an idea about an open source SIP API written in C# that can
communicate with Asterisk, to call out?
Regards,
Sanjay.
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Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word transfer, I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could
There is a .NET 1.1 library out there... I've played with it a little bit, but
not enough that I could comment on how feature rich or stable it is...
http://www.voip-info.org/wiki/view/Asterisk+.NET
It'll more than likely not be compatible with AMI 1.1 however, which I believe
is included in
This work with Asterisk Manager Interface. I want to implement basic phone
functionality in C#.
Regards,
Sanjay.
- Original Message -
From: Matt Watson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 4,
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven
Matt Watson escribió:
There is a .NET 1.1 library out there... I've played with it a little bit, but
not enough that I could comment on how feature rich or stable it is...
http://www.voip-info.org/wiki/view/Asterisk+.NET
It'll more than likely not be compatible with AMI 1.1 however, which I
On Thu, Apr 3, 2008 at 11:35 PM, [EMAIL PROTECTED] wrote:
Do anyone has an idea about an open source SIP API written in C# that can
communicate with Asterisk, to call out?
There are a few C# SIP stacks around that will let you do that.
Creating a call from such a stack to Asterisk will be
Can you Please refer me to any, the one that I found are all either in Java/C.
Or if they are in C# they are not opensource.
Regards,
Sanjay.
- Original Message -
From: Grey Man [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
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