On Sun, Sep 17, 2006 at 01:13:07PM +0800, unplug wrote:
How can I use a system cmd to get back the return value in dial plan?
Say, I want to run a script using system cmd to get the hostname.
System(hostname)
Reading previous posts helps.
On Sat, Sep 16, 2006 at 01:15:01PM +0200,
I have a number of different calls coming in to Asterisk from one SIP proxy.
All calls are currently allocated the same SIP Channel name but I want them
to be named differently. Note that Asterisk registers with the SIP Proxy,
not the other way around.
sip.conf
register=5551234:[EMAIL
On 9/16/06, Rich Adamson [EMAIL PROTECTED] wrote:
RR wrote:
All,
is there anyone who uses g726-32 ? If not, then does anyone know why
don't people use it?
I use g726 on iax links between systems and to teliax.com for LD calls.
Have no idea if its -32 or what though. What ships with asterisk
Thanks.
Could you tell me some detail how to compile and implement it with asterisk?
On 9/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Sep 17, 2006 at 01:13:07PM +0800, unplug wrote:
How can I use a system cmd to get back the return value in dial plan?
Say, I want to run a script
Hi,
I have to decide on hardware to buy real fast (being rocketed into the
situation).
We have 1 computer, we'll install hardware from digium in there to
connect with the ISDN phone lines (2)
It's a normal computer, I have no idea what type of card to take and
about the 3.3v vs 5v PCI.
I have a problem when making calls to and from my iax2 client(ifdefisk) I
only get one way audio and even when I make calls to a pstn line I have
tried the ff on iax.conf : allow=all and it does not seem to help.
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I am looking for Infos Tutorials for installing ISDN Karte PCI HST
Saphir III ML. Does sombody have infos links for me?
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On Sun, Sep 17, 2006 at 01:25:10PM +0200, Timothy Parez wrote:
We'll have about 10 internal phones.
One of the phones should be like a central station, where all other
calls can be monitored (if possible)
and from that phone the user should be able to press a button to take
over a call
I rebooted the server on which the Asterisk is hosted on. The * did not come back up and I get this message when I attempt to use CLI
[EMAIL PROTECTED] ~]# asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
___
[EMAIL PROTECTED] schrieb:
I rebooted the server on which the Asterisk is hosted on. The * did not
come back up and I get this message when I attempt to use CLI
[EMAIL PROTECTED] ~]# asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
Did you configure your
Hi,
obviously asterisk doesn't start with the installed(?) start script.
Try to start it manually and watch the cli for informations with
asterisk -vvvc
AFAIK a make config in the asterisk source should install the start script
for your system.
Hope it helps...
Guido
Master_PE schrieb:
I have changed in sip.conf the bindport (port=) to (bindport=) .
When i try to connect to my provider it doesn't work becourse it try's
to connect to port 5060. sip debug says
Retransmitting #5 (NAT) to 62.177.135.42:5060:
REGISTER sip:62.177.135.42 SIP/2.0
trion*CLI sip
On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
Hi,
obviously asterisk doesn't start with the installed(?) start script.
Try to start it manually and watch the cli for informations with
asterisk -vvvc
One warning: if your system is normally configured to run as
I still have problem with the latest implementation of rxfax for 1.2.
In fact it exit with non zero from the macro and it is not going to execute
the system script the have the fax2mail service.
Has someone the same experience?
Thanks to Steve U.
Rosario
- Original Message -
From:
try this:
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER
On 9/16/06, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hi Users,I'm new to Asterisk programming , I'm in working the Voip Technologies by using the OpenSER for my call routing process and Radius For
-Ursprüngliche Nachricht-
Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 17. September 2006 15:56
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Asterisk Server Down
On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
Hi,
Much higher, maybe double but that is when the agents start to complain
that their conversations start cutting in and out.
This is the main reason I am looking into building a re-invite solution
that moves the call into recording/cdr server's media path. Then we
just cap these servers at
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Gesendet: Sonntag, 17. September 2006 15:56
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Asterisk Server Down
On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote:
On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:
you're right, one should proof, under which user asterisk runs...
Besides security reasons, running asterisk as root, doesn't it allow a
higher prioritization of asterisk processes?
This is why we let asterisk setuid itself to user
I know it's not a digium product, but the 12 port A1200P card with a
single FXO module at pbxeq.com at first glance would seem to be the way to
get started for me with an in-system controller card. 4 ports seems too
small for expansion, the huge 24 port card a tad too big (and spendy).
So
I got the config working. Not sure if someone has
pre-recorded sounds for this app or not. Looked all over for them and I'm unable
to locate them.If anyone has sound file they would like to share that would help
me greatly.
Thanks
Sent: Friday, September 15,
2006 5:23 PMTo:
Hi Ron -
Is there a way to program one of the buttons on the 501 (Like the services
button) to do on the fly call recording? So in the middle of the phonecall
you can record the call without have to do a transfer type of setup. Ive
looked at the manual but cant seem how to do that, I only see
Hi, dont expect too much help providing only the Asterisk version you
are using. You need to tell us the call path, DTMF mode (inband,
outband, SIP INFO etc) used and call technologies involved (SIP, ZAP,
IAX2 etc).
Regards
On 9/15/06, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
Can
Hi Kevin -
Has anyone used the Polycom expansion module with multiple lines?
My application is for 20 lines and read there was a limit of 7 at one point.
I heard rumors that the newest version of the polycom sip firmware
(2.01) would lift the limit of 7. It just came out, and I haven't had
RR wrote:
On 9/16/06, Rich Adamson [EMAIL PROTECTED] wrote:
RR wrote:
All,
is there anyone who uses g726-32 ? If not, then does anyone know why
don't people use it?
I use g726 on iax links between systems and to teliax.com for LD calls.
Have no idea if its -32 or what though. What ships
Nick Ellson wrote:
So has anyone used this card with Asterisk? I googled for reviews and
have not found anything, and I am tryingto find a way to search the
archives without looking at each month one at a time.
I was able to download the .gz files, extract them into SeaMonkey's mail
As for FOP, when clients come to meet you after seeing attractive interfaces
from other proprietary systems, its just embarrassing to show them such an
ugly interface like FOP.
FOP interface, altough it has some limitations (fixed button positions
and size for the flash client), it is pretty
When Asterisk (1.2.12.1) receives a SIP register message for a realtime
peer, the CLI reports Disconnected from Asterisk server. Asterisk has
disappeared:
asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
A look at the full log doesn't reveal much:
Sep
On 17 Sep 2006, at 12:25, Timothy Parez wrote:
Hi,
I have to decide on hardware to buy real fast (being rocketed into
the situation).
We have 1 computer, we'll install hardware from digium in there to
connect with the ISDN phone lines (2)
It's a normal computer, I have no idea what type
Thanks everyone it is working now.
-- Original message -- From: Tzafrir Cohen [EMAIL PROTECTED] On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:you're right, one should proof, under which user asterisk runs... Besides security reasons, running asterisk
I saw this termination company, www.BuyMin.comthe rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive.
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To
Yeah i was messing around last night and saw that! Now if I can only get the other caller to not hear the DTMF digits id be set! I didnt know you could remap the keys to DTMF digits, but since I can do that this will work perfect for the most part!
Thanks for the info!On 9/17/06, Noah Miller
Hi Again Ron -
Yeah i was messing around last night and saw that! Now if I can only get the
other caller to not hear the DTMF digits id be set! I didnt know you could
remap the keys to DTMF digits, but since I can do that this will work
perfect for the most part!
Let me qualify by saying that
Hi,
I am still working on trying to figure out why I cannot use the Dial
command from my AGI script. Can anyone tell me what I can do to get
more information about what's going on. I've tried asterisk -v with as
many v's as I can put on one line (like 40), and I was wondering if
there is
Brian Rogan wrote:
Hi,
many v's as I can put on one line (like 40), and I was wondering if
there is anything that I can do to debug this problem.
/etc/asterisk/logger.conf
Uncomment the full and restart Asterisk.
You'll find the log in:
/var/log/asterisk/full
Doug
-- Ben Franklin
Brian Rogan wrote:
Hi,
I am still working on trying to figure out why I cannot use the Dial
command from my AGI script. Can anyone tell me what I can do to get
more information about what's going on. I've tried asterisk -v with as
many v's as I can put on one line (like 40), and I was
Nick Ellson wrote:
I know it's not a digium product, but the 12 port A1200P card with a
single FXO module at pbxeq.com at first glance would seem to be the
way to get started for me with an in-system controller card. 4 ports
seems too small for expansion, the huge 24 port card a tad too big
Do some 7960s perform differently?
On 9/15/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the
RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone.
Well this would not be for comercial use.. I just want it for my own
cell phone to talk on my own asterisk system.
is that ok?
Tim Panton wrote:
On 16 Sep 2006, at 20:38, Net Nut wrote:
So with that said, can anyone recommend a way that I can get a sip
client on a cell phone that uses H.263
I'm using bristuff which patches lot of things, I did not try to use this patch ... may be I shouldjl2006/9/15, Gareth Owen [EMAIL PROTECTED]
:
I got a chance to patch my Asterisk server
this afternoon and was able to confirm that the directed call pickup feature is
working (at least
There has been several different hardware versions of the phone, but to
the best of my knowledge, the ringer has not changed. The cisco
documentation suggests there is a way to create your own ring tones, but
I've not tried that either.
The stock 7960 sip phone's built in ring tones are not
I tried one of these and pretty much got it working under visdn. If you
do decide to try one, make sure you get the HFC version. Earlier ones
used another chipset and definitely weren't supported using open sourced
drivers.
Please post back if you do get one and get it going though.
Thanks
Hi,
I have a problem installing func_odbc on asterisk 1.2.12.1 the message
that I received is:
./astxs func_odbc.c
make[1]: *** No rule to make target `apps_env'. Stop.
-I/usr/src/asterisk -I/usr/src/asterisk/include -c func_odbc.c -o func_odbc.o
make: *** [func_odbc.so] Error 255
then I've
As far as I know, it's 12.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Sun 9/17/2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] Polycom Expansion
Hi guys,
I have what is probably a very noob question. I've tried to search the
wiki, but my lack of knowledge is hindering me in finding the right
keywords:
I'd like to know what the packet size of an IAX2 packet is, if its using
the ilbc codec.
Now I'll tell you why, so you can tell me
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