I have been asked by the customer to
deilver a big PBX-system based on Asterisk. The requirements are approximately:
- up to 240 lines for making outside calls from the
building
- up to 1000 internal phone conversations (within the
building)
- scalable up to 300/1500 calls
Does
thanks Mark,
your right i have been getting confused between office and
station(given myself an uppercut and a slap to the back of the head),
thats sort of what im after, but does not really give any quick fix or
details for config files for making asterisk use this card.
i am a new
Ust go to eBay and find an X100P card. I usually pay $6.95 and usually buy
two at a time if they will ship them together so with shipping it will run
about $10-$12 each. If that will break the budget, I don't know if these is
a better solution.
-Original Message-
From: [EMAIL PROTECTED]
On Thu, 2005-11-24 at 21:24 +0200, Dan wrote:
Nov 24 20:55:13 NOTICE[25742]: chan_bluetooth.c:2227 try_connect:
Initialised bluetooth link to device W800
[AG] W800 AT+BRSF=23
Nov 24 20:55:13 ERROR[25742]: chan_bluetooth.c:2628 handle_rd_data:
Device W800: Expected '\n' got 13. state
Hi
thanks for the reply
i have visited the Site you have mentioned
in that site, they have mentioned how to config [EMAIL PROTECTED]
but i dont see any config installed with Source of Ast
does any one have step by step config
to configure IAX provider confg
and as well local extensions
ram
Anyone looking for the firmware? Please contact me off-list.
Best,
Stephen
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Your last few posts were about [EMAIL PROTECTED]. If you have it installed, either
from the ISO or the tar file, then all of the instructions I have already
pointed out are completely valid. If you need additional help, search through
the wiki at http://voip-info.org. If that
isn't enough,
2005/11/25, Bharath [EMAIL PROTECTED]:
have you added allow=speex allow = ilbc in the sip iax conf files ?
Yes. And the connection is stablished, the log says it is using speex
(i.e.), but I don't receive any sound. This weekend I will do more
tests.
--
Alejandro Vargas
thats only in the us!
they want over $50. AU for them here..
- without a credit card( not really a problem )
- just tried to log in on ebay server in the us, got the craps with the
login shit! every thing is taken user/pass.
and i tried some weird shit, even my name.?? there is another
Hi
thanks,
but later i posted that, [EMAIL PROTECTED] need new server to start with
as i have shortage of servers to go with that kind of setup
so i have downloaded the Ast Source not [EMAIL PROTECTED]
and compiled existing server setup, rather a new server
so iam asking what are config
Also try executing /var/lib/asterisk/agi-bin/a2billing.php from the
shell, most probably the path to php-cli is wrong or you don't have it
installed at all.
Jose M. Ramirez wrote:
Hi list, all. Please, I need help. Although already I installed
a2billing, simply I cannot initiate its
Vedran Dakic ha scritto:
I have been asked by the customer to deilver a big PBX-system based on
Asterisk. The requirements are approximately:
- up to 240 lines for making outside calls from the building
- up to 1000 internal phone conversations (within the building)
- scalable up to 300/1500
What sort of connection are you after for the Panasonic phone system? I
mean, what are you trying to achieve in connecting to it? Understanding
this might help with defining the best way to achieve a connection.
You should be able to locate a cheap card as per Kerrys post. Due to
variability of
You mean 240 / 1000 simultaneous calls or 240 outside lines and 1000
internal phones ?
I can only guess that I should have the ability to deliver a solution that
can do some 100/500 simultaneously. The only question is how powerful should
be a machine (or machines) that could do around 100/500
Stig Even Larsen wrote:
/ I'm having problems connecting my Sangoma cards to our PRI (E1)
// interface. It seems that the card get connected (green led), but
// Asterisk reports:
// Status: Provisioned, Down, Active.
//
// When I install another Sangoma card on the same system (pri_net),
Add a w (wait) to your dialstring. Chances are asterisk is dialing
before you are getting dialtone. You can put a butt set on the line and
listen while dialing out to verify this before adding the w. Sometimes
it may take ww to get it working correctly.
Thanks,
Steve
_
From:
It is all done through the web GUI, nothing is edited manually with the
exception of the ZAP files. Again, it is all in the manual that you
apparently are not reading.
Thanks,
Steve
_
From: ram [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 4:35 AM
To: Asterisk
I have used Ebay hundreds of times and never got ripped off.
_
From: Phil Pritchard [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 4:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] configure intel modems.
thats
Is there a way to use a regular (analog) fax machine with
Asterisk? I suppose it coule be achieved by
using some ATA device, but is it possible without that?
Cheers,
Vedran.
___
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Hi
i have buy a used Cisco Phone 7910 for use with my asterisk.
The firmware version are 3.2(2.8), it's good for connect to asterisk ?
For update the fiormware, where i can get a new firmware ?
thanks bye
___
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No.
_
From: Vedran Dakic [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 8:07 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk fax
Is there a way to use a regular (analog) fax machine with Asterisk? I
suppose it
Hello all,
I'm trying to write a macro that'll handle blind SIP transfers nicely, since
at present, blind transferring to a busy SIP extension will give the
incoming caller busy tones.
Hopefully this will be of use to others on the list once it's working
correctly. Here's what I've got so far:
I use with success cisco ata 186, linksys
pap2 and Audiocodes MP108 with asterisk sending and receiving faxes with
regular analog fax machines, works very well. I have to set up all to use g711
because I can not make it work with t.38 with asterisk.
I was testing with asterisk 1.09, I
Noc Phibee ha scritto:
i have buy a used Cisco Phone 7910 for use with my asterisk.
The firmware version are 3.2(2.8), it's good for connect to asterisk ?
That is an old firmware. atest is 5.07
Try http://chan-sccp.berlios.de for the sccp channel driver
For update the fiormware, where i can
How about Sipura's 1001, 2002, 2100?
Anyone?
Cheers,
Vedran.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Federico Piergentili
Sent: Saturday, November 26, 2005
2:22 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE:
Thanks Mark,
phone system is a panasonic 1232 digital hi-bread thingy im fairly
privy to it and can reprogram etc. and with that, should be able to go
ether way,
- i thought that i could hang asterisk of one of the analog extensions
with a voice modem.
- the other way would be to hang it
Hello,
When asterisk receive a registration with a private
address is it possible to forward the sip request for
this agent to a sip proxy ?
Regards
Harry
___
Appel audio GRATUIT
Thanks Mark,
phone system is a panasonic 1232 digital hi-bread thingy im fairly
privy to it and can reprogram etc. and with that, should be able to go
ether way,
- i thought that i could hang asterisk of one of the analog extensions
with a voice modem.
- the other way would be to hang
Hi list:
Where i have to go to get callback.agi script i
searched every where but i didnt find it.
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
___
--Bandwidth and
Go to google and type callback.agi.
-Original Message-
From: jonny hashem [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 9:06 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] callback.agi script
Hi list:
Where i have to go to get callback.agi script
Denny Schierz wrote:
Most howtos discribes asterisk with capi or bristuff. How does the
extensions.conf and misdn.conf looks like, so that ISDN (basics) works,
for example phone internal from sip to isdn, isdn to sip or isdn to isdn.
Is there anybody, who has asterisk 1.2 with misdn running?
Kerry Garrison wrote:
pain to configure) have 4 ring types. I am guessing that I would need to
figure out how to tell this particular phone to use a different ring tone
unless there is a way to send a stutter type ring to the phones.
Hi Kerry, I'm also using grandstreams on a few places, have
Wayne Gemmell wrote:
Yes, make a 'default' to go directly to your fax-receive macro. (rxfax
witht the parameters)
At least you should hear a 'fax' answering.
Yes, I hear a fax answering, so at least I know its working.
Okay, so the ring detection goes wrong. Now at least you know what
Chris Bagnall wrote:
Hopefully this will be of use to others on the list once it's working
correctly. Here's what I've got so far:
Looks very nice, just have to find a way to get it integrated with AMP,
but looks promising!
It works fine with one exception - when the caller hits 1 whilst
Hi:
i used this callback.agi script and here what i ve
get :
Executing Wait(OSS/dsp, 1) in new stack
-- Executing AGI(OSS/dsp, callback.agi) in new
stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/callback.agi
Use of uninitialized value in string ne at
It means that there are sound files that need to be recorded.
-Original Message-
From: jonny hashem [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 10:12 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] problems with callback.agi script
Hi:
i used
http://www.google.com/search?hl=enq=sound+files+asterisk
-Original Message-
From: jonny hashem [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 10:12 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] problems with callback.agi script
Hi:
i used this
Hi :
i have the required soundfiles and i put it in
/var/lib/asterisk/sounds
and it stills request the sound file , and it give me
this message :
Use of uninitialized value in string ne at
/var/lib/asterisk/agi-bin/callback.agi line 56,
STDIN line 14.
do you have any idea what do i have to chnage
Hi :
i have the required soundfiles and i put it in
/var/lib/asterisk/sounds
and it stills request the sound file , and it give me
this message :
Use of uninitialized value in string ne at
/var/lib/asterisk/agi-bin/callback.agi line 56,
STDIN line 14.
do you have any idea what do i have to change
On Nov 26, 2005, at 8:15 AM, Steve Totaro wrote:
From: Vedran Dakic [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005 8:07 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk fax
Is there a way to use a regular (analog) fax machine with
you can use 'w' option with 'Dial' on 1.2.x
I don't think w do anything like 'wait', If I am wrong, correct me someone
please
According to app_dial.c
w- Allow the called party to enable recording of the call by sending\n
the DTMF sequence defined for one-touch recording in
Andrew Kohlsmith wrote:
On Saturday 05 November 2005 02:54, Tzafrir Cohen wrote:
Asterisk here falls under desktop: Much like interactive desktop apps
it need timely response rather than more time at the CPU.
I am not so sure; asterisk is a multithreaded app, yes, but I would think that
On Nov 26, 2005, at 4:35 AM, ram wrote:Hi thanks, but later i posted that, [EMAIL PROTECTED] need new server to start with as i have shortage of servers to go with that kind of setup so i have downloaded the Ast Source not [EMAIL PROTECTED] and compiled existing server setup, rather a new
Hello,
I am trying to enhance my cdr records.
What I am trying to are:
1. Add an option in Dial, Say R. to pass rate (price per minute) for the call
to do that I will have to modify app_dial.c
2. Dial would use option R to set cdr
3. I will also need to add one more function in cdr.c, say
On Fri, Nov 25, 2005 at 07:16:18PM +0100, asterisk183 wrote:
I have installed bristuff 0.3.0 for Asterisk 1.2 with kernel 2.4, but when I
doing :
insmod qozap.o
the shell show this messagge:
qozap.o: qozap.o: unresolved symbol free_irq_Rsmp_f20dabd8
qozap.o: qozap.o: unresolved
I am pretty sure it will wait if at the beginning of the dialstring like
this:
exten = _.,1,Dial(ZAP/g1,(${EXTEN}))
this amounts to a two second delay, each w = 500ms
Thanks,
Steve
-Original Message-
From: Innocent Evil [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 26, 2005
Hi folks,
This is what I am doing at this time :
exten = _,1,TrySystem(..command that sends a jabber message..)
exten = _,2,Set(calling=${EXTEN:0:4})
exten = _,3,ChanIsAvail(SIP/[EMAIL PROTECTED])
exten = _,4,Dial(SIP/[EMAIL PROTECTED],15,tr)
exten =
Stig Even Larsen wrote:
The thing is that I'm going to IP-enable our Alcatel PBX. This system is
currently working on our E1 PRI. I'm therefore 100% sure that the E1 is
working. Since the green led is showing. Is not that an indication that
the PRI is UP?
I believe that the led is yellow on
Innocent Evil wrote:
1. Add an option in Dial, Say R. to pass rate (price per minute) for the call
to do that I will have to modify app_dial.c
2. Dial would use option R to set cdr
3. I will also need to add one more function in cdr.c, say something like
ast_cdr_setrate(...)
None of this
I still continue to reboot my asterisk box everyday.
I posted a message on November 22, but it was on another thread and
no one answered me, so I try again here,
where a lot of people told be I was a bad administrator (Like a
Windows administrator and I don'0t want to resolve my problem)
Stig Even Larsen wrote:
The thing is that I'm going to IP-enable our Alcatel PBX. This
system is
currently working on our E1 PRI. I'm therefore 100% sure that the E1
is
working. Since the green led is showing. Is not that an indication
that
the PRI is UP?
I believe that the led is
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Friday, November 25, 2005 9:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk + WiFi Phones
I eventually switched to using a Astra 480i CT desk phone
Has anyone ran into limits with chan_bluetooth and the maximum number of
calls that can be handled at the same time. I am sure there has to be
some limit within either bluetooth or the bandwidth of a dongle. I
would appreciate any info regarding which bluetooth dongle/card was used
on the server
Hello All,
I'm currently running CVS-HEAD-05/28/05-20:52:10 which is quite old and was
looking to upgrade to 1.2. I have three hopefully simple questions:
1) Do the patches from http://www.lusyn.com/asterisk/patches.html still
need to be applied for the X100P ?
2) I noticed on the site that 1.2
Hi All,
Does anybody know if there is Open Source Voice Recognition available for
Asterisk. I'm looking to add voice commands to my dial plan if possible.
Thanks in advance.
Phil.
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More specifically, you can make it work using an ATA or a TDM400P
card with an fxs port, but it is not likely to be reliable. If you
send a few faxes here and there, that shouldn't be a big deal. If you
are talking about an office where lots of faxing is done, the lack of
reliability will
I'm sure these questions have been answered at some point, but I'm too new
to this stuff to know the right words to plug into the search function to
find what I need.
I have never touched Asterisk before, but have wanted to for some time.
Now I finally think I'm going to bite the bullet, as I
Jason Marshall schrieb:
I'm sure these questions have been answered at some point, but I'm too
new to this stuff to know the right words to plug into the search
function to find what I need.
I have never touched Asterisk before, but have wanted to for some time.
Now I finally think I'm going
Hello all,
I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls as
expected. I have been trying to get atxfer working and am getting the error
message:
WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
whenever I try a transfer.
In features.conf:
On Sat, Nov 26, 2005 at 05:30:49PM +, [EMAIL PROTECTED] wrote:
Hello All,
I'm currently running CVS-HEAD-05/28/05-20:52:10 which is quite old and was
looking to upgrade to 1.2. I have three hopefully simple questions:
1) Do the patches from http://www.lusyn.com/asterisk/patches.html
On Sat, 26 Nov 2005 12:15:29 -0500, Steve Totaro wrote:
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Friday, November 25, 2005 9:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk + WiFi Phones
I
On Saturday November 26 2005 1:26 pm, John Millican wrote:
Hello all,
I have just upgraded to 1.2 from 1.0.9 and am receiving and placing calls
as expected. I have been trying to get atxfer working and am getting the
error message:
WARNING[19541]: res_features.c:844 builtin_atxfer: Did not
Hi,
Does anyone know if something like H standard extension exists in
Asterisk (I'm not talking about h standard extension). If yes what
does it do and what is the difference in comparison to h standard
extension.
One more thing; when I put h extension in my dial plan:
exten =
On Nov 26, 2005, at 12:22 PM, Manny A. Wise wrote:
On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote:
Great!!, this did the trick, now we have audio...
We are using a Sipura 2000 for testing
The Sipura now can call out and have audio...the only problem left
is that
the sipura can't receive
On Nov 26, 2005, at 12:47 PM, Andrew Nowrot wrote:
More specifically, you can make it work using an ATA or a TDM400P
card with an fxs port, but it is not likely to be reliable. If you
send a few faxes here and there, that shouldn't be a big deal. If you
are talking about an office where lots of
On Nov 26, 2005, at 12:48 PM, Jason Marshall wrote:
I'm sure these questions have been answered at some point, but I'm
too new to this stuff to know the right words to plug into the
search function to find what I need.
We'll let it go just this once... ;-)
I have never touched Asterisk
I have two fwd accounts, and I want them to behave differently. It
took me a while to figure out why it wouldn't work, but finally I
realized that the last definition in sip.conf is the one that steals
the show.
Simplified, I have this:
register = account1:[EMAIL PROTECTED]/88
register =
harry gaillac wrote:
Hello,
When asterisk receive a registration with a private
address is it possible to forward the sip request for
this agent to a sip proxy ?
Regards
Harry
Hello,
I don't know anyway for this but what is your main target? Why you don't
try to register SIP proxy
Hi list,
I installed iaxmodem and Hylafax to see how it compares to rx/txfax; so
far I had 0 failure in my limited testing with a Philips HFC21 fax
machine that failed very often with txfax (same test platform, with
spandsp-0.0.2xxx).
So congratulation and thanks for this work !
(maybe some of
/ There is a big difference between the span being up/down and the
// D-channel being up/down. The span can be up, green and ready to go,
/but
/ the D-channel turned 'down'. Why that would be happening on your
/already
/ running span is beyond me... but Asterisk obviously thinks that the
Hi,
we're having quite some problems with new hardware we're testing - Parlay
Voxip ISDN-SIP gateway...
So we're curious if anyone is using this in connection to Asterisk and what
are experiences on this HW ?
Thanks in advance,
regards,
Rob.
We'll let it go just this once... ;-)
Thanks *8-)
I want all calls to come into the Asterisk box in the main office.
This is relatively easy, but how you do it depends on where the analog POTS
lines are terminated. At the central office or at the employees' remote
location? (I assume that
Hello,
i found in my system logs problem with handling IAX2 calls - its looks:
Connected to Asterisk CVS-Nv1-2-0-11/19/05-23:19:49 currently running on
ast-serv (pid = 13312)
Verbosity is at least 13
ast-serv*CLI show channels
Channel Location State Application(Data)
Hi
There was recently an update to the X100P patches to support 1.2.
If you look at the Lusyn website I think he's covered it (It was on this
list about 2 weeks back).
P
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Asterisk-Users
Tom Rymes ha scritto:
However, I do think it is fairly clear that using an ATA is a less
than ideal solution for any serious faxing, since the fax protocol
often doesn't play nicely with the tendency of VOIP to occasionally
lose packets. YMMV, though, so try it out
Yep I can confirm
[EMAIL PROTECTED] is believed to have said:
Jason,
I'm sure these questions have been answered at some point, but I'm too new
to this stuff to know the right words to plug into the search function to
find what I need.
well, yes of course.
I have never touched Asterisk before, but have
I am getting the same problems -- undefined symbols with
spandsp-0.0.2pre21 and the fax modules from the same directory --
wonder if there are anylater versions? I am not using /usr/local/lib
for any of these so its not in the ld.so.conf.
Any assistance would be appreciated.
on Wed, 23 Nov 2005
On Fri, November 25, 2005 18:44, Francesco Peeters said:
On Fri, November 25, 2005 9:29, Kristof Hardy said:
Francesco Peeters wrote:
I compiled 1.2 and bristuff 0.3.0 Pre1 yesterday late and that now
seems
to work! * is up and running *with* 2nd card in NT mode...
Nice to hear *1.2 and
Could someone assist me in connecting my bluetooth headset to my asterisk
box.
Doing the normal hcitool scan does not provide any connection that the
headset is available. It does pick up my other bluetooth devices that are
nearby.
I have tried puting the headset into pairing mode, but it
I'm not sure if anyone else is using this script. 411 must have changed their formatting a little.
I fixed the script by changing line 49 of web_lookup.php to
{ $result[ name ] = $parser-ExtractTo( /strong, TRUE );-- Is it something someone said, was it something someone said?
On Sat, November 26, 2005 23:19, Francesco Peeters said:
On Fri, November 25, 2005 18:44, Francesco Peeters said:
On Fri, November 25, 2005 9:29, Kristof Hardy said:
SNIP
Well, so far not good! * really seems to have problems with that second
card! :-(
When I have it run it's own clock,
Sphinx. All the info you need is on voip-info.org
Dean
From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED]
Sent: Sat 11/26/2005 12:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice recognition ...
Hi All,
Does anybody know if
Hi Chris,
I appreciate your getting back to me. This looks like the firmware for
optiPoint 400 Standard which I have already. I was looking for the
firmware for the latest optiPoint 410 and 420 series. Anyway I now have
them preinstalled in these sexy looking phones and they are being shiped
Ren
Can you email the exact configurations? By the way, do you have the SIP
application software for these phones and what version is it?
Previously I was looking for the
app/firmware for the latest optiPoint 410 and 420 series. Anyway I now
get them preloaded in these sexy looking phones
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
Kristof Hardy schrieb:
This was with a quadBRI card from junghanns, but I've now switched that
testmachine to bristuff 0.3.0pre1.
i know this howto, but it doesn't answer the main questions. :-/ It
seems, that nobody has it running.
cu denny
I've just heard about DECT which is used for about 50 million phones in
Europe and is just starting to appear in the US.
DECT stands for Digitally Enhanced Cordless Telephone
and supposedly has much greater range than other cordless telephony.
Additionally, you can purchase repeaters that will
Before using asterisk I can see that all the channels are set correctly
on the digium wildcards. But when running Asterisk doing a 'zap show
channels' shows them as unconfigured. There is three cards total and
all are seen outside of asterisk with a total of 12 channels, but when
in asterisk it
I've just heard about DECT which is used for about 50 million
phones in Europe and is just starting to appear in the US.
I didn't realise they'd not been around in the US for long. I've had DECT
phones for at least 5 years now...
In my house, a Uniden 5.8 and Panasonic 2.4 cordless system
Here are some more thoughts on this one:
1. Make use of the ${BLINDTRANSFER} varialbe to detect that it realy
is a blindxfer.
2. If it is a blindxfer, use the ${BLINDTRANSFER} varialbe for
fallback, if that fails then go to the queue.
On 11/26/05, Kristof Hardy [EMAIL PROTECTED] wrote:
Chris
Hi
thanks for your suggetions
iam working on it
will do the same and post you any more help required
ram
On 11/26/05, Tom Rymes [EMAIL PROTECTED] wrote:
On Nov 26, 2005, at 4:35 AM, ram wrote:
Hi
thanks,
but later i posted that, [EMAIL PROTECTED] need new server to start with
as i have
On 11/25/05, Julio Tejera [EMAIL PROTECTED] wrote:
Hello:
Looking for help ...
I need to setup an * box in order to swap my office from
an old pbx..., the only thing that I can't figure out on how
to do with * is to have something like this:
- User A and customer are in a bridged call
-
Use the setgroup checkgroup apps.
On 11/25/05, Rafael Canchola [EMAIL PROTECTED] wrote:
Hi.
I have a problem or require in my Asterisk, I need limit the out calls from
my outgoing context. I have configure a outgoing peer in sip.conf
[outgoing-xxx], but I need that in this peer out
Just fire up vi and start typing.
On 11/25/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi list:
what are the steps to do to asterisk to be ready fro
callback system?
__
Yahoo! Music Unlimited
Access over 1 million songs. Try it free.
Hello
I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk
1.2 libraries, must be oh323-0.7.3, now I have compiled this version
but when reload asterisk i have this error:
[chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe
Any idea???-- rrgv
The problem is your's. pre21 works with Asterisk 1.2, although I haven't
yet updated the Makefile patch to apply cleanly. Try looking for other
versions of spandsp left behind on your system.
Steve
John Covici wrote:
I am getting the same problems -- undefined symbols with
thats right
On 11/27/05, C F [EMAIL PROTECTED] wrote:
Just fire up vi and start typing.
On 11/25/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi list:
what are the steps to do to asterisk to be ready fro
callback system?
__
Yahoo! Music Unlimited
I think I discovered a bug.
I have a dual Xeon machine running * 1.2.0
I have a queue defined to play the default music on hold class, which
simply plays an mp3 file.
When a call comes into the queue (note that there are no agents
logged in, but I have joinempty=yes and leavewhenempty=no
Does someone have a working example of extensions.conf using the new
page command?
Thanks
___
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HI,
I have been playing around with the A2billing agi for a few days( I
really think its a great application) and trying to understand its
working. As far as I have understood, the only debugging mechanism is
going through the logs it generates.
It would be really cool if I can run the agi in
1. Add an option in Dial, Say R. to pass rate (price per minute) for the
call
to do that I will have to modify app_dial.c
2. Dial would use option R to set cdr
3. I will also need to add one more function in cdr.c, say something
like ast_cdr_setrate(...)
None of this is necessary.
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