I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to dia
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and such.
So please rate your i
I want to create a CGI that will allow me to make a call when a user
clicks on a URL in a webpage. I believe I need to create a file in
/var/spool/asterisk/outgoing that defines the number I want to call and
the phone I want to connect it to but I see no way to define the phone
number I want to dia
Notice the mailbox= entry in
http://www.fnords.org/~eric/asterisk/zapata.conf.html
Works the same in http://www.fnords.org/~eric/asterisk/sip.conf.html
On Mon, 2003-10-20 at 23:03, PBX wrote:
> I have a quick question...
>
> In the previous thread
> http://www.marko.net/asterisk/archives/0210/03
Voicemail2 already does the date and time.
On Mon, 2003-10-20 at 22:03, Kevin wrote:
> Has there been any discussion as to having asterisk voice mail play an
> optional message envelope with caller ID, date and time of message?
>
>
>
> ___
> Asteris
10 - A way to lock the phone settings (IP address, etc). It is too easy
to change the settings when in a public environment. The MENU button
should not be 1 press away from changing the settings, Use MENU + SOME
COMBINATION.
7 - Use the conference button to access Meetme. Like the Voice Mail
Use
7 - Ringer volume control
4 - plug in module of user programmable buttons for frequently called
numbers. Not everyone would need this so being able to add as an
optional module would keep the base phone cost effective.
9 - ability to switch back and forth between speakerphone and handset
7
On Monday 20 October 2003 21:38, John Brown (CV) wrote:
> Hi List,
>
> I had a wonderful meeting with GS's President last week
> and he is very interested in feedback on what top features,
> functions, bugs the community would like to see in upcoming
> firmware.
>
> Please keep in mind that adding
Go here
http://lists.digium.com/mailman/listinfo/asterisk-users
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, October 20, 2003 10:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unsubscrip
I would like to remove m
I'm pretty happy with mine, I've got 2 of them as basic extensions, but
I've found the following with daily use.
The phone needs more lower bandwidth codecs, starting with GSM or
ilbc scale 10
The blue backlight to stay on since the display doesn't tilt it makes it
easier to see. flashing it
Quoting Steven Critchfield <[EMAIL PROTECTED]>:
> On Mon, 2003-10-20 at 08:42, Herc wrote:
> > - Original Message -
> > From: "WipeOut" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, October 20, 2003 9:26 PM
> > Subject: Re: [Asterisk-Users] No detection of Line Busy
>
Dear Members,I am trying to make call from one Gnophone to another
through the localAsterisk Server.All the three systems have local IP
AddressesI created two users "sheeba" (extension 600) and "test" (extension
602) iniax.conf
file:[sheeba]type=friendauth=plaintexthost=dynamicsecret=sheebac
I have a quick question...
In the previous thread
http://www.marko.net/asterisk/archives/0210/0306.html it is mentioned
Mark added support for MWI to the chan_zap. Is this in the zapata.conf
and if so, if stutter is turned on then the MWI is turned on with it?
Geoff
_
I would like to remove my mail address from
asterisk so pl let me know how to remove from the list.
Thanks
Venkateswaran
Dear Members,I am trying to make call from one Gnophone to another
through the localAsterisk Server.All the three systems have local IP
AddressesI created two users "sheeba" (extension 600) and "test" (extension
602) iniax.conf
file:[sheeba]type=friendauth=plaintexthost=dynamicsecret=sheebac
> and the bits 1,2,4
>
> For the queue skillmask just keep multing the number by 2
>
>
> 1 = sales
> 2 = tech level 1
> 4 = tech level 2
> 8 = tech level 3
> 16 = advanced problems
> 32 = coperate
>
>
> to allow a queue member to be allowed to take the call just add up
> all the numbers that go wit
John,
I want the tftp configs done like cfgMACADDRESS.txt or compile
them into a binary form like the ATA's use. And stop trying to rip us for
the GAPS system. WHAT A RIP. It makes cisco so worth the extra cash!
Config refresh similar to the ATA.. refresh config every x seconds.
bkw
O
Has there been any discussion as to having asterisk voice mail play an
optional message envelope with caller ID, date and time of message?
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
- Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 21, 2003 10:29 AM
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
>
> Paul -
>A few questions and comments:
>
> 1) So, does this also make "incominglimit" and "outgoi
unsubscribe
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[EMAIL PROTECTED]
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On Mon, 2003-10-20 at 21:35, [EMAIL PROTECTED] wrote:
> What I want to do is have one phone number for multiple call bridges
> (meetme) so that first users are prompted for their call bridge ID then
> their password.
>
>
> exten => 7001,1,Playback(/var/lib/asterisk/sounds/if-you-know-the-extensi
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and such.
So please rate you
What I want to do is have one phone number for multiple call bridges
(meetme) so that first users are prompted for their call bridge ID then
their password.
exten => 7001,1,Playback(/var/lib/asterisk/sounds/if-you-know-the-extension-you
want-to-dial-press-that-extension:gsm)
exten => 7001,2,s
Sorry:
exten => 2080,1,Answer
exten => 2080,2,Background,meetme1
exten => 2080,3,Authenticate(/opt/pass/pass_meetme1.txt)
exten => 2080,4,Meetme,1
Regards,
Gus
- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, October 20, 2003 10:12 PM
Subject
Joshua :
I don't know why you include 's,5,Hangup'...
I'm doing the same with:
exten => 2080,1,Answer
exten => 2080,2,Background,meetme1
exten => 2080,2,Authenticate(/opt/pass/pass_meetme1.txt)
exten => 2080,3,Meetme,1
meetme1 gsm file plays "Welcome to conference room number 1", and
/opt/pass/p
On Monday 20 October 2003 18:21, Paul Liew wrote:
> Hi All,
>
> This is the first time I'm submitting a patch, and I hope it fixes
> more than it breaks. I'm putting it here, since John Todd mentioned
> a while ago about the heavy load Mark and crew have at Digium (doing
> such good work), so I th
Sorry, to repost - but I left a "/*" comment - here it is again
Paul
[code block removed]
Paul -
A few questions and comments:
1) So, does this also make "incominglimit" and "outgoinglimit" work
as expected? The current method doesn't do quite what the average
user thinks it would do.
2) Y
Yes i'm using one of the workarounds.. but you can't do a native transfer
to the parking extension. # transfer yes.. but that is NOT a native sip
transfer.
On Mon, 20 Oct 2003, Juan J. Sierralta P. wrote:
> On Mon, 2003-10-20 at 16:44, Brian West wrote:
> > > - I couldn't get Asterisk call-parki
Hi Florian,
Florian Overkamp wrote:
Hi,
Citeren Steve Underwood <[EMAIL PROTECTED]>:
If it doesn't work for you, don't be too surprised. Feed back anything
you find, and lets try to make things better. I suspect, from experience
and things I have read on the web, that a lot of fax machines
Hello,
I have
a couple of 7960 and a quad T1 card on my asterisk box. I want to let
the phones to use g729 when they "talk" to each other, but to use g711
when I'm going to route the call out of my network using the T1 card.
Everything works just fine between the phones, b
Sorry, to repost - but I left a "/*" comment - here
it is again
Paul
--- chan_sip.c.save
2003-10-20 21:51:52.0 +1000+++ chan_sip.c 2003-10-21
09:26:41.0 +1000@@ -959,7 +959,9
@@
return 0;
} switch(event)
{+
/* Incoming a
Hi All,
This is the first time I'm submitting a patch, and I hope it fixes more than
it breaks. I'm putting it here, since John Todd mentioned a while ago about
the heavy load Mark and crew have at Digium (doing such good work), so I
thought all of us could test this first, and if ok submit for i
On Monday 20 October 2003 18:22, Joshua Heiks wrote:
> I also can not figure out what "Unknown RTP codec 19 received"
> means..
It means that your vendor still hasn't figured out that comfort noise
should be codec 13 and that codec 19 is reserved and should not
be used. And this isn't a recent ch
Once the dust settles here and there is more of a reliable build/install
recipe available, I'll have a closer look, but so far this sounds great !
I am not sure how the faxing standards interfaces are exposed right now to
asterisk, but I think rather than reinvent the wheel as far as the manag
How do I use the Authenticate application in my IVR menu, where do I put the
password?
here is my menu. I need to ask for a password before I let users log into my
conference room.
[conf1]
exten => s,1,Ringing
exten => s,2,Wait,2
exten => s,3,Answer
exten => s,4,Authenticate(1234)
exten => s,5,Han
My checked out source is not up to date with CVS so
my Rtp.c, Line 374 is not like yours but in general
what must have happened (given the text of the
message) is someone made a non-blocking
"read" system call when there was no data, saw that it
failed (-1 returned) and printed the log message.
Please see if this helps.
Regards,
Gus
noc2pbx*CLI> show application MeetMe
noc2pbx*CLI>
-= Info about application 'MeetMe' =-
[Synopsis]:
Simple MeetMe conference bridge
[Description]:
MeetMe(confno[|options]): Enters the user into a specified MeetMe
conference.
If the conference number
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
Would anyone have an idea on how I would be able to take the mic in on
the computer and put it as the "talking party" for a conference room.
I would then be able to set up a "listen only" profile for others to get
/etc/asterisk/musiconhold.conf
[classes]
default => mp3:/var/lib/asterisk/mohmp3
/etc/asterisk/extensions.conf
exten => 101,1,Answer
exten => 101,2,MusicOnHold(default)
That's about what is said in the manual (RTFM ;-) and it works great.
Jean-Christophe
- Original Message -
From: <[EM
Strange. I have a simple extension set up to do some Festival testing.
(Festival 1.4.3 /w Asterisk patch). My extension looks like:
exten => 1239,1,Answer()
exten => 1239,2,Festival(Welcome to the asterisk system!)
exten => 1239,3,Wait,1
exten => 1239,4,Hangup
Some times it work right (sounds c
On Mon, 2003-10-20 at 14:19, Florian Overkamp wrote:
> Hi,
>
> Citeren Steve Underwood <[EMAIL PROTECTED]>:
> > If it doesn't work for you, don't be too surprised. Feed back anything
> > you find, and lets try to make things better. I suspect, from experience
> > and things I have read on the we
::> I am trying to set up an IAX2 trunk between two
::> servers.
::[snip]
::> I am using bmtools to monitor the bandwidth usage, and
::> I am not seeing a difference.
::>
::Trunks don't seem to work with "host=dynamic", at least in my setup they
::don't. A good way to see if calls are actually goi
Anyone can share me with Music Onhold Configuration sample?
Thanks in advance for your help,
Kang
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On Mon, 2003-10-20 at 14:15, jerk face wrote:
> I am trying to set up an IAX2 trunk between two
> servers.
[snip]
> I am using bmtools to monitor the bandwidth usage, and
> I am not seeing a difference.
>
Trunks don't seem to work with "host=dynamic", at least in my setup they
don't. A good way t
Hi,
Citeren jerk face <[EMAIL PROTECTED]>:
> [ServerB]
> type=friend
> trunk=yes
> host=dynamic
> secret=myPassword
> context=myContext
>
> Server B has the following in extensions.conf:
> [outgoing]
> exten=>_40X,1,Dial,IAX2/ServerB:[EMAIL PROTECTED]/${EXTEN}
>
> I am using bmtools to monitor
ering directory `/usr/src/cvs/asterisk/apps'
gcc -O2 -g -Iinclude -I../include -I/usr/src/tiff-v3.5.7/libtiff -
I/usr/src/spandsp-20031020/src -c -o app_rxfax.o app_rxfax.c
In file included from app_rxfax.c:38:
/usr/src/spandsp-20031020/src/t30.h:96: parse error before `t4_state_t'
/usr/src
On Mon, 2003-10-20 at 16:44, Brian West wrote:
> > - I couldn't get Asterisk call-parking to work with this phone, transferring
> > to extension 700 doesn't work(and it works fine with my SNOM200) maybe just
> > a config change on my end, but I couldn't figure it out
>
> cant do native sip transfe
opened static nat for both * and client, one on each firewall
configured xten to connect to external (nat) address of asterisk
configured sip on asterisk on external (nat) address of client
I think this was all
- Original Message -
From: "Chris Albertson" <[EMAIL PROTECTED]>
To: <[EMAIL P
I am trying to set up an IAX2 trunk between two
servers.
Server A has the following in iax.conf:
[general]
...
[ServerB]
type=friend
trunk=yes
host=dynamic
secret=myPassword
context=myContext
Server B has the following in extensions.conf:
[outgoing]
exten=>_40X,1,Dial,IAX2/ServerB:[EMAIL PROTECTED
> - I couldn't get Asterisk call-parking to work with this phone, transferring
> to extension 700 doesn't work(and it works fine with my SNOM200) maybe just
> a config change on my end, but I couldn't figure it out
cant do native sip transfers to parking.
__
Eric Wieling wrote:
On Mon, 2003-10-20 at 11:31, Chris Albertson wrote:
Asterisk works perfectly fine in back of a NAT firewall, as long
as all of your SIP phones are also in back of that same fire
wall ;-)
Seriously, I'd fix this if I knew enough about SIP protocol.
Is anyone willing to write up
Hello,
After receiving and finally being able to configure my new Polycom IP 600
phone Here are my initial experiences:
GOOD STUFF:
- The sound is wonderful from both handset and speaker (711U)
- The large multi-shade LCD screen is beautiful (for a phone)
- The phone has an integrated Ethernet sw
Chris Albertson wrote:
X-Lite _can_ dail out to FWD through a firewall but Asterisk can't.
SO this gives us the perfect chance to compare the content of
outbound packets where we have a working and non-working example.
X-lite when configured for FWD behind a NAT uses an outbound proxy
for that con
Can you please describe how you have your "el-cheapo" consultative transfers
working?
- Original Message -
From: "Marcel Prisi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 21, 2003 1:04 AM
Subject: Re: [Asterisk-Users] Success story
> hi,
>
> We have blind tranfers
Hi,
Citeren Steve Underwood <[EMAIL PROTECTED]>:
> If it doesn't work for you, don't be too surprised. Feed back anything
> you find, and lets try to make things better. I suspect, from experience
> and things I have read on the web, that a lot of fax machines do not
> follow the standards very
On Mon, 2003-10-20 at 16:36, Steve Underwood wrote:
> If it doesn't work for you, don't be too surprised. Feed back anything
> you find, and lets try to make things better.
At the moment I'm having the devils own job to get it compiled on MDK
9.0. or 9.2
--
Dave Cotton <[EMAIL PROTECTED]>
___
You should start by reading the specific SIP and RTP RFCs. SIP is less
of an issue than RTP (as someone else pointed out)
On Mon, 2003-10-20 at 12:47, Chris Albertson wrote:
> --- Eric Wieling <[EMAIL PROTECTED]> wrote:
> > Actually it requires CHANGING the SIP protocol. Asterisk already
> > cha
--- Tomica Crnek <[EMAIL PROTECTED]> wrote:
> to be more specific, I only managed to get xten softphone register to
> *
> behind the nat fw, but nothing else.
Where was the firewall?
1) Between xten X-Lite and the public Internet or,
2) Between Asterisk and the Publict Internet or
3) Bot
to be more specific, I only managed to get xten softphone register to *
behind the nat fw, but nothing else.
- Original Message -
From: "Tomica Crnek" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, October 20, 2003 7:44 PM
Subject: Re: [Asterisk-Users] SIP Nat Issue
> yes, re
--- Eric Wieling <[EMAIL PROTECTED]> wrote:
> Actually it requires CHANGING the SIP protocol. Asterisk already
> changes the SIP protocol when you use nat=yes and many clients also
> change the SIP protocol to work with NAT.
Is it really a change to the format of what is sent or is it that
only
yes, regarding sip, but I have stil problems with rtp
- Original Message -
From: "Mark Evans" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, October 20, 2003 4:49 PM
Subject: [Asterisk-Users] SIP Nat Issue
> Hi All
>
> Has anything been done to fix the issue where the * box
See: http://bugs.digium.com/bug_view_page.php?bug_id=343
What kind of details do you need?
Steve
On Mon, 20 Oct 2003, WipeOut wrote:
>Hi,
>
>I was just taking a look at the source code and noticed two files..
>
>retrieve_extensions_from_mysql.pl
>and
>retrieve_sip_conf_from_mysql.pl
>
>Its
"show application MusicOnHold"
Also "show applications" for more information on all the Asterisk
applications.
On Mon, 2003-10-20 at 11:13, Kevin wrote:
> Is there anyway for a sip station to play MoH out of the speaker?
> I know I can do it by calling the station and putting it on hold.
>
> For
Thanks, it works.
Now is their a way to use a sound card and use it's input to power music
on hold?
Or is that a feature that needs a little programming to get done?
Kevin,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Hajime Lanning
Sent: Monda
Good job.. now that the cat is out of the bag i'm sure you will get alot
of requests or ideas and maybe code!
bkw
On Mon, 20 Oct 2003, Steve Underwood wrote:
> Hi all,
>
> I would like to announce the availability of an initial test version of
> a totally software FAX facility, suitable for use
Title: Need to partner with someone in Hampstead London on a deal
The info
below was passed to me when looking for Digium products in the UK.
TelAppliant VoIP
Solutions (London)
Tan Aksoy
Voice: (44) 0845 004 4040
(local rate)
E-mail: [EMAIL PROTECTED]
WWW: www.telappliant.com
Actually it requires CHANGING the SIP protocol. Asterisk already
changes the SIP protocol when you use nat=yes and many clients also
change the SIP protocol to work with NAT.
On Mon, 2003-10-20 at 11:31, Chris Albertson wrote:
> Asterisk works perfectly fine in back of a NAT firewall, as long
> a
Kevin wrote:
Is there anyway for a sip station to play MoH out of the speaker?
I know I can do it by calling the station and putting it on hold.
You can make an extension that will play MoH:
exten => 5551212,1,MusicOnHold
Make your button dial that extention. Voila!
Jerimiah
Tularosa Communi
- Original Message -
From: "Kevin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, October 21, 2003 2:13 AM
Subject: [Asterisk-Users] MOH different question
> Is there anyway for a sip station to play MoH out of the speaker?
> I know I can do it by calling the station and put
Thanks,
The bug is submitted.
Cheers,
Steven
On Mon, 2003-10-20 at 17:52, John Todd wrote:
> At 6:19 PM +0200 10/20/03, Steven Poelmans wrote:
> >
> >Hello,
> >
> >I defined a global var in extensions.conf and tried to change it via the
> >SetGlobalVar application.
> >The application didnt return
> Is there anyway for a sip station to play MoH out of the speaker?
> I know I can do it by calling the station and putting it on hold.
>
> For example:
> On a samsung pbx with MoH, if you goto one of the workstaions and press
> a button
> The moh plays out of the speaker.
>
> Is there any way to
Title: Need to partner with someone in Hampstead London on a deal
I have made a contact with a company in London looking for various voip and ip telephony services. Is there someone local there who may help facilitate this opportunity?
Ray Burkholder
[EMAIL PROTECTED]
http://www.oneunifie
Asterisk works perfectly fine in back of a NAT firewall, as long
as all of your SIP phones are also in back of that same fire
wall ;-)
Seriously, I'd fix this if I knew enough about SIP protocol.
Is anyone willing to write up what is required at the "bit and byte
level"? One thing that could he
I'll own up to a patch - bug report 110. However, Mark peremptorily
dismissed my suggestion putting forward a solution I find illogical. I
guess more people need to ask for this feature!
I think my original patch was a bit over-engineered. The one below is
simpler.
Iain
--- res_parking.c.
Is there anyway for a sip station to play MoH out of the speaker?
I know I can do it by calling the station and putting it on hold.
For example:
On a samsung pbx with MoH, if you goto one of the workstaions and press
a button
The moh plays out of the speaker.
Is there any way to do this with aste
On Mon, Oct 20, 2003 at 08:49:42AM -0700, John Todd wrote:
> At 2:54 PM +0200 10/20/03, Louis-David Mitterrand wrote:
> >> Missing a microphone to work handsfree (or i didn't find it.) but
> >> strange enough their is a speaker ...
> >
> >Yeah, that's a real bummer. Cisco calls that "feature" Monit
There's a bug report on bugs.digium.com. Most people don't need to
transfer calls that go to outside numbers, only calls that come IN to
asterisk and calls between extensions and so most people don't run into
the problem.
On Mon, 2003-10-20 at 09:42, Louis-David Mitterrand wrote:
> On Mon, Oct 20
On Mon, 2003-10-20 at 08:36, Steve Underwood wrote:
> Hi all,
>
> I would like to announce the availability of an initial test version of
> a totally software FAX facility, suitable for use with Asterisk. This is
> a first public test release, so don't expect a solid polished product
> just yet
At 4:01 PM +0100 10/20/03, WipeOut wrote:
Mark Evans wrote:
Hi All
Has anything been done to fix the issue where the * box is sat behind a
nat firewall?
Regards
Mark
As far as I know it can't be done.. The server has to be on a public IP..
You could try using a SIP aware router like the inter
At 6:19 PM +0200 10/20/03, Steven Poelmans wrote:
Hello,
I defined a global var in extensions.conf and tried to change it via the
SetGlobalVar application.
The application didnt return any errors, however the value of the global
variable was still the same as the initial value.
Also the descriptio
On Mon, Oct 20, 2003 at 08:29:49AM -0700, Anthony Minessale wrote:
> I run into that # issue sometimes too
>
> All I can do is hit ## so the lady tells me there is no ext really
> fast and i may not miss any of the call the # still makes it to the
> real call too.
>
> If you knew in advance yo
At 3:42 PM +0200 10/20/03, Louis-David Mitterrand wrote:
Hi,
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with "Transfer?".
Is there a way to escape the pound key, short of disabl
At 2:54 PM +0200 10/20/03, Louis-David Mitterrand wrote:
On Mon, Oct 20, 2003 at 09:21:45AM +0200, Michael Devenijn wrote:
Justy to let you all know
that i tested 7905G phone with a SIP image (latest download) and it
works great ! for a reasonable price but with a good quality and a
brand ...
On Mon, 2003-10-20 at 09:42, Louis-David Mitterrand wrote:
> On Mon, Oct 20, 2003 at 02:55:18PM +0100, WipeOut wrote:
> > >This morning I found myself stumped when a remote interactive system
> > >asked me to enter some identification followed by the # key, and my
> > >local Asterisk interrupted wi
On Mon, 2003-10-20 at 08:42, Herc wrote:
> - Original Message -
> From: "WipeOut" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, October 20, 2003 9:26 PM
> Subject: Re: [Asterisk-Users] No detection of Line Busy
>
>
> > [EMAIL PROTECTED] wrote:
> >
> > >Hello,
> > >
> > >I
I run into that # issue sometimes too
All I can do is hit ## so the lady tells me there is no ext really fast and i may not miss any of the call the # still makes it to the real call too.
If you knew in advance you are calling that kind of system you could always
clone the ext you use to make
Alex Ayala wrote:
I was wondering if anyone knows if Asterisk works in FreeBSD? I heard
the problem was that the digium cards weren’t supported in FreeBSD.
Thanks,
Alex
AFAIK, Asterisk can be made to compile and run but as you mentioned I
think the problem is drivers..
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Cory Andrews
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Correct me if I am wrong, but if you use Agent Groups,
does this negate the strategies?
I assume these strategies are handled in app_queue,
and the groups are handled in the chan_agent.
Which of the strategies have been programmed? Last
I read, not all of them were in place.
Also, have the patches
I was wondering if anyone knows if
Asterisk works in FreeBSD? I heard the problem was
that the digium cards weren’t supported in FreeBSD.
Thanks,
Alex
Hi all,
I would like to announce the availability of an initial test version of
a totally software FAX facility, suitable for use with Asterisk. This is
a first public test release, so don't expect a solid polished product
just yet. People have shown interest in what I am doing, and here is the
Mark Evans wrote:
Hi All
Has anything been done to fix the issue where the * box is sat behind a
nat firewall?
Regards
Mark
As far as I know it can't be done.. The server has to be on a public IP..
You could try using a SIP aware router like the intertex range but I
don't know how much mil
Thanks, yes it helped a lot. I didn't understand the sense of contexts in
such cases.
Thanks and regards,
Jean-Christophe
- Original Message -
From: "WipeOut" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, October 20, 2003 3:33 PM
Subject: Re: [Asterisk-Users] Playing around
Louis-David Mitterrand wrote:
On Mon, Oct 20, 2003 at 02:55:18PM +0100, WipeOut wrote:
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with "Transfer?".
Is there a way to escape t
Hi All
Has anything been done to fix the issue where the * box is sat behind a
nat firewall?
Regards
Mark
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On Mon, Oct 20, 2003 at 02:55:18PM +0100, WipeOut wrote:
> >This morning I found myself stumped when a remote interactive system
> >asked me to enter some identification followed by the # key, and my
> >local Asterisk interrupted with "Transfer?".
> >
> >Is there a way to escape the pound key, shor
Florian Overkamp wrote:
Hi,
in following of a recent discussion I got to work on MGCP with the
Cisco ATA186 again, and got it to work very nicely. However, there is
a little thing with transfers I would like to get comments on:
Call comes in from PSTN and goes to an ATA186 (MGCP)
Call is answe
Hello,
I defined a global var in extensions.conf and tried to change it via the
SetGlobalVar application.
The application didnt return any errors, however the value of the global
variable was still the same as the initial value.
Also the description of the application is the same as with the Set
That's likely what's happening because it eventually
does make the call (after searching the remote
server). To get around the problem I just set up an
extension to send outgoing calls to my other server.
- Jerkface
--- Mark Spencer <[EMAIL PROTECTED]> wrote:
> Now it *is* notworthy that even i
Louis-David Mitterrand wrote:
Hi,
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with "Transfer?".
Is there a way to escape the pound key, short of disabling transfers?
Cheers,
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