On Mon, 2003-10-20 at 05:52, Tilghman Lesher wrote:
On Sunday 19 October 2003 22:39, Drazen Vidakovic wrote:
Can I use modem on Linux box for making outgoing calls?
And receiving to?
http://asstricks.org/faq.html
Freudian slip?
I know how you feel, perhaps it should be moved to No 1?
Dear Members,
I am trying to make call from one Gnophone to another through the local
Asterisk Server.All the three systems have local IP Addresses
I created two users sheeba (extension 600) and test (extension 602) in
iax.conf file:
[sheeba]
type=friend
auth=plaintext
host=dynamic
secret=sheeba
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 ... which inspires trust and helps selling better
The only minus :
Missing a microphone to work handsfree (or i didn't
find
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 answered and then, using
Hi,
I try to use asterisk with SIP with Messenger
client.
Does anyones have already done this ?
Can we make alias like asterisk-oh323 channel
drivers ?
What sort of extension have I put in
extensions.conf ?
Regards
Rattana
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 pretty obvious what these two files do, but info about them is a
little scarce..
Is anyone using these scripts and could give me any details on
If you mean data calls (ppp dial-up for example) then there's no
problem, it works.
On Mon, 2003-10-20 at 07:59, Dave Cotton wrote:
On Mon, 2003-10-20 at 05:52, Tilghman Lesher wrote:
On Sunday 19 October 2003 22:39, Drazen Vidakovic wrote:
Can I use modem on Linux box for making outgoing
hi,
We have blind tranfers working well, and el-cheapo consultative
working too ...
Might be better ...
On Mon, 2003-10-20 at 13:57, Aaron Martin wrote:
Hi Marcel,
Good to hear that everything is working well for you.
Just one question, how do your users transfer calls to each other?
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 ... which inspires trust and helps selling better
The
Hello,
I am quite new to asterisk.
I managed to connect our 2 branch offices with asterisk.
In one side, our linux asterisk box is connected to the leased line going
to our other office and on the other side its connected to office PBX through a
channel bank.
This installation is running
Let's say I have 3 IP phones (A, B, C) and 3 MSNs
(1, 2, 3).
How can I define that the incoming MSN 1 is
redirected to A,2 to B and3 to C ?
And how can I define that the A phone uses the
outgoing MSN 1, etc ?
Actually, I'm using the CAPI channel driver, but
any help is welcome.
[EMAIL PROTECTED] wrote:
Hello,
I am quite new to asterisk.
I managed to connect our 2 branch offices with asterisk.
In one side, our linux asterisk box is connected to the leased line going
to our other office and on the other side its connected to office PBX through a
channel bank.
This
Jean-Christophe Heger wrote:
Let's say I have 3 IP phones (A, B, C) and 3 MSNs (1, 2, 3).
How can I define that the incoming MSN 1 is redirected to A, 2 to B
and 3 to C ?
And how can I define that the A phone uses the outgoing MSN 1, etc ?
Actually, I'm using the CAPI channel driver, but any
- 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 am quite new to asterisk.
I managed to connect our 2 branch offices with
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,
--
Make it idiot proof, and
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,
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 if
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
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
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, short of
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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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
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 MSNs
I was wondering if anyone knows if
Asterisk works in FreeBSD? I heard the problem was
that the digium cards werent supported in FreeBSD.
Thanks,
Alex
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
Never used in production - $3750/ea Email [EMAIL PROTECTED] if interested.
Cory Andrews
*
b2 Technologies
*
voice: 866-44-B2TECH X22
fax: 716.630.1548
email: [EMAIL PROTECTED]
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Alex Ayala wrote:
I was wondering if anyone knows if Asterisk works in FreeBSD? I heard
the problem was that the digium cards werent supported in FreeBSD.
Thanks,
Alex
AFAIK, Asterisk can be made to compile and run but as you mentioned I
think the problem is drivers..
I run into that # issue sometimes too
All I can dois 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
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 am quite new to
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 with
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 ...
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
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
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
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.
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
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 Monitor mode, ie:
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
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
---
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
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]
quote who=Kevin
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
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 any errors,
- 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 putting it
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
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
as
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
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
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:
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
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 pretty
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 is sat
--- 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 some
--- 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) Both 1
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
changes
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]
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
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 working
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
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
- 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.
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
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
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 transfers to
/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/spandsp-20031020/src/t30.h:96
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 the bandwidth
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 to see
Anyone can share me with Music Onhold Configuration sample?
Thanks in advance for your help,
Kang
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:: 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 going
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 web, that a
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
/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: [EMAIL
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
in
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
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.
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,Hangup
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
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
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
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 and outging affects the inUse counter
*/
case
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,
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
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-parking
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) Your
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 thought
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
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: Re:
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 =
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
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-extension-you
unsubscribe
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- 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 outgoinglimit work
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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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
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 with his skills
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
I would like to remove my mail address from
asterisk so pl let me know how to remove from the list.
Thanks
Venkateswaran
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
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
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