On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote:
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
I'll eventually replace this crippled Linux with something better but I don't
time to play around with it as most divers and modules are still too new
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)- legacy
pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR option and
they are connected to the agents via the legacy pbx (which is in sync with
asterisk on
Hi,
Is it possible to get information about SIP destination channel (created
after Dial command) somehow?
For example I would like to know what codec was used. I can do this for
originating channel with:
${CHANNEL(audionativeformat)}
but not sure how to do the same for destination channel?
Sriram wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via
span2)- legacy pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which
is in
On Sun, Nov 16, 2008 at 4:28 AM, Sriram [EMAIL PROTECTED] wrote:
Hi
below are my configs:
pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)-
legacy pbx analog extensions.
my dial plan is like callers dial into asterisk(span1) , hear an IVR option
and they are connected
Looking at some old Polycom reference configs I find the following from
the phone.cfg
The users selection of the receive volume during a call can be
remembered between calls. This can be configured per termination
(handset, headset and hands-free/chassis). In some countries
regulations
On my eeePC I install windows, for the same reasons, sound video drivers...
Chris
Sent from my BlackBerry® smartphone with SprintSpeed
-Original Message-
From: Joseph [EMAIL PROTECTED]
Date: Sat, 15 Nov 2008 22:39:40
To: Asterisk Users Mailing List - Non-Commercial
hi Robert
followed your points - but problem persists...everything goes well for sometime
but after that - asterisk is unable to dial the pbx...
any more thoughts
thanks
Sriram___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Sun, Nov 16, 2008 at 9:41 PM, Sriram [EMAIL PROTECTED] wrote:
hi Robert
followed your points - but problem persists...everything goes well for
sometime but after that - asterisk is unable to dial the pbx...
any more thoughts
Post some outputs
or logs
ram
On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote:
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new
Hi
-Executing [EMAIL PROTECTED]:1] Dial(Zap/13-1,ZAP/g2/3901) in new stack
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Hangup(Zap/13-1,) in new stack
== Spawn extension (custom-app,1,2) exited non-zero on 'Zap/13-1'
-- Hungup 'Zap/13-1'
this is the
I checked the app store and haven't found anything promising, but I figured I'd
ask here.
Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will
communicate directly with a machine running Asterisk?
I know that there's at least one offering that seems like it's
On 11/16/08 10:09, Tzafrir Cohen wrote:
On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote:
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
I'll eventually replace this crippled Linux with something better but I
don't time to play around with it as most
On Sun, Nov 16, 2008 at 10:08:58AM -0700, Joseph wrote:
On 11/16/08 10:09, Tzafrir Cohen wrote:
On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote:
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
I'll eventually replace this crippled Linux with
Hello,
I'm running an Asterisk 1.4.14 on a linux machine.
Serving SIP Snom users.
I've noticed that each time Asterisk is restarted, for the first 5-10
minutes, the SIP users can dial but cannot be dialed until each phone
re-registers itself against the server.
So only after the Saved
Dear Sir,
I have an openSer+asterisk installed and running smoothly...I configured
OpenSer to send calls with a specific dial petterns to be routed to
Asterisk...What I need now is to create a dial plan that send all calls
coming from OpenSer to a specific provider...SO I have created a trunk
I did an svn check out of dahdi.
I am running centos 4.7
on the make all I get an error.
/usr/src/dahdi-complete/linux/drivers/dahdi/wcb4xxp/Makefile: No such
file or directory
make[4]: *** No rule to make target
`/usr/src/dahdi-complete/linux/drivers/dahdi/wcb4xxp/Makefile'. Stop.
In article [EMAIL PROTECTED],
Mikel Lindsaar [EMAIL PROTECTED] wrote:
On Sun, Nov 16, 2008 at 9:37 AM, Tony Mountifield
[EMAIL PROTECTED]wrote:
Actually, if Read() works, then WaitExten should have worked too. I expect
what was missing was the Answer(). So this ought to work as an
On 15/11/2008 3:58 a.m., equis software wrote:
I found this property in queue.conf
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
On Nov 16, 2008, at 9:07 AM, Lincoln King-Cliby wrote:
I checked the app store and haven’t found anything promising, but I
figured I’d ask here.
Does anyone know of a SIP or IAX client for a non-jailbroken iPhone
that will communicate directly with a machine running Asterisk?
Lincoln,
Hi,
I am new user of Asterisk. I am using Asterisk version 1.4.15, and I am having
the following problem:
I am using the Record application to record my SIP channel. With the timeout
option, if I don't record (don't speak anything) anything, after the timeout
period it comes out
On Sun, Nov 16, 2008 at 04:49:28PM -0500, Jerry Geis wrote:
I did an svn check out of dahdi.
I am running centos 4.7
on the make all I get an error.
/usr/src/dahdi-complete/linux/drivers/dahdi/wcb4xxp/Makefile: No such
file or directory
make[4]: *** No rule to make target
This process has been greatly improved in the latest versions of
Asterisk - might be time to upgrade.
PaulH
[EMAIL PROTECTED] wrote:
Hello,
I'm running an Asterisk 1.4.14 on a linux machine.
Serving SIP Snom users.
I've noticed that each time Asterisk is restarted, for the first 5-10
Jerry Geis wrote:
I did an svn check out of dahdi.
I am running centos 4.7
on the make all I get an error.
/usr/src/dahdi-complete/linux/drivers/dahdi/wcb4xxp/Makefile: No such
file or directory
make[4]: *** No rule to make target
Nice. Older version of the kernel. Kbuild won't do and we need o use
Makefile.
Could you please try copying over the Makefile fro wct4xxp?
cp -a drivers/dahdi/wct4xxp/Makefile drivers/dahdi/wcb4xxp/
Thanks that works.
Is there an easy way to make DAHDI only compile dahdi_dummy?
I
When I upgraded from 1.4 to 1.6 it seems my dialplan extensions.conf
is not loaded. in UPGRADE.txt I dont see any reason why.
^[[1;30m == ^[[0mParsing '/etc/asterisk/extensions.conf': ^[[1;30m ==
^[[0mFound
It shows its parsing with no errors.
dialplan show - does not show anything from
Rob Hillis wrote:
The solution for the problem of an IAX client is a SIP client.
That's not a particularly good solution if you have a NAT between your
client and Asterisk. IAX is still *much* easier to get working through
a firewall.
It's working fine here (Twinkle/Ubuntu over NAT/Netscreen).
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