Than you
Hey I have 100 SIP phone with 2 E1 card and IVR feature but i am
not happy with my configuration so have u any configuration for advance level
Rgd
satish patel
Al lists [EMAIL PROTECTED] wrote: exten = _98XX,1,Dial(ZAP/(your
preferred E1)
exten =
On 20:24, Thu 19 Jul 07, Zoiper wrote:
Hello guys,
The so expected 2.0 release of Idefisk 2.0 softphone is a fact.
Idefisk and Zoiper became one - Zoiper 2.06.
Any indication when the linux and osx builds will be 2.0 ?
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG
I am also planning for IVR so u have any kind of script plz suggest me
David Ruggles [EMAIL PROTECTED] wrote: I have written a script that is
executed using ExternalIVR(). I am running
in to performance issues when I have four or more simultaneous calls running
this script. I'm running on a P4
Hi,
Here is the situation.. My Dad is working on contract in overseas.. He
has internet access in his hotel.. He wants to be able to talk to my Mum
but the calls are expensive..
I have an asterisk box setup for my business and it has a public IP
etc.. My Mum has access to a working phone
Noah Miller wrote:
Hi Arun -
Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents
this asterisk box is connected to another asterisk box using 5 IAX trunk to
load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my
cli start flooding with message: Maximum trunk
Noah Miller wrote:
You have to first uninstall your Asterisk1.2 like this--
First you have to stop your asterisk...using--
1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using.
In my experience, you don't need to do this step. In fact, you can
keep the old asterisk
On 7/21/07, satish patel [EMAIL PROTECTED] wrote:
Than you
Hey I have 100 SIP phone with 2 E1 card and IVR feature but
i am not happy with my configuration so have u any configuration for advance
level
Rgd
what kind of advanced level
Asterisk side or IP phone side
ram
Get portsip ( www.portsip.com ) its realtively easy to configure ( just
push in user/password and server name at startup ) .. there might be NAT
issue so make sure you have nat=yes in ur asterisk's sip.conf for the peer
definition . If it still doesnt work then you need to find a iax phone like
Hello list,
I have prepared a new tutorial for Astrecipes on how to compile the latest
Asterisk 1.4 with H323 support, Google Talk and Zaptel support, starting
from a stock TrixBox system.
You can find it here: http://www.astrecipes.net/index.php?n=286
I hope somebody will find it useful :-)
SIP has a lot of issues with NAT, I can only get it to work correctly
on my LAN with a softphone.
IDEFISK, now known as Zoiper, is IAX based and I have tested it
from all kinds of hotel rooms, even the free version supports
6 simultaneous calls :
i want asterisk extention.conf IVR plan so i want idea of IVR means how other
users use IVR in dialplan on asterisk
ram [EMAIL PROTECTED] wrote:
On 7/21/07, satish patel [EMAIL PROTECTED] wrote: Than you
Hey I have 100 SIP phone with 2 E1 card and IVR feature but i am
not
On 7/21/07, satish patel [EMAIL PROTECTED] wrote:
i want asterisk extention.conf IVR plan so i want idea of IVR means how
other users use IVR in dialplan on asterisk
Hi
Hint is Look at Agi Scripts
you can write small agi scripts to do your job
ram
Hi List;
I need help for the following senario:
Initiating a call from Asterisk to an extension and after it answers, IVR
prompts
will be played
Mohammad Mirzaee
+989121750530___
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satish patel wrote:
I am also planning for IVR so u have any kind of script plz suggest me
*/David Ruggles [EMAIL PROTECTED]/* wrote:
I have written a script that is executed using ExternalIVR(). I am
running
in to performance issues when I have four or more simultaneous
Does anyone out there know what version/release of solaris the g729
(v32) codec is built on? Is it built on
Solaris 10 GA,
Solaris 10 U1,
Solaris 10 U2,
Solaris 10 U3,
OpenSolaris (Nevada), which build?
I'm just trying to find out if my problem with the codec may be due to a
release difference,
Jay Wilton wrote:
gdb /usr/sbin/asterisk -c /tmp/core.4545
GNU gdb 6.3-debian
...CUT
This GDB was configured as i386-linux...Using host
libthread_db library /lib/libthread_db.so.1.
/tmp/core.4545 is not a core dump: File format not
recognized
Does the user running gdb have proper
I think n+101 worked in Asterisk 1.2.x but it doesn't work in Asterisk 1.4.x
use ${DIALSTATUS} if you want Asterisk to act depending the result of Dial()
I read that the variable has been disabled in SVN to be replaced by the
DEVSTATE function, I need to confirm that.
Well... an example:
exten
Hi Mohammad,
The best way is tu use .call files, check here :
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Cheers,
Yves.
On Sat, 2007-07-21 at 18:08 +0430, mohammad mirzaee wrote:
Hi List;
I need help for the following senario:
Initiating a call from Asterisk
One I really like is the idefsk version that was a zip file, you could
extract the file configure the softphone, zip it up and email it out. Saved
the headache of walking someone through the process and even ran of thumb
drives.
On 7/21/07, WipeOut [EMAIL PROTECTED] wrote:
Hi,
Here is the
Great article here about the concept of open spectrum I posted a few
weeks ago. Could be very very interesting.
http://machinist.salon.com/blog/2007/07/20/google_fcc/
Maybe an Asterisk/OpenMoko tie-in?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
So I am looking for a softphone thats really simple to setup and as
foolproof as possible..
If SIP is likely to be problematic to setup then I have no problem
getting him to use IAX but will need suggestions of which IAX softphone
to use and also how to configure it in the iax.conf (haven't
On 7/21/07, Time Bandit [EMAIL PROTECTED] wrote:
You don't specify if he's on Windows, Linux or OSX. But if he is on
Windows, you can try my softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php
There is a version using INI file, so you can put all the settings
then zip it
I want my freedom to setup and configure PBX hardware and software how i
want, not how Digium or anybody else wants, so not interested in Asterisk
Appliances.
So anybody with experience with Supply Logics computers. Or any other
recommendations for asterisk pbx casings?
On 7/21/07, Zeeshan Zakaria wrote:
I want my freedom to setup and configure PBX hardware and software
how i want, not how Digium or anybody else wants, so not interested in
Asterisk Appliances.
So anybody with experience with Supply Logics computers. Or any
other recommendations for
Zeeshan Zakaria wrote:
I want my freedom to setup and configure PBX hardware and software how i
want, not how Digium or anybody else wants, so not interested in
Asterisk Appliances.
So anybody with experience with Supply Logics computers. Or any other
recommendations for asterisk pbx
Jay Wilton wrote:
gdb /usr/sbin/asterisk -c /tmp/core.4545
/tmp/core.4545 is not a core dump: File format not
recognized
Does the user running gdb have proper permissions to the
core file?
I was running gdb as root. Asterisk is running as the
asterisk user. I chmod 777'd the core
Is there any way to change COS bits for packets?
There is a tos option on sip.conf, does asterisk change COS bits considering
tos value?
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On Thu, 19 Jul 2007 12:14:53 -0500, Alvaro Parres wrote
Yes Moises, i was looking for it.
The main problem is only on the files for version 1.4... it give that
error when no CallerID is recive or a private caller id is recive.
The change i made is to add to Mexico variant on
Check this out: http://www.kapanga.net/IP/home.cfm
Very easy to create a self-installing pre-configured soft phone.
On 7/21/07, WipeOut [EMAIL PROTECTED] wrote:
Hi,
Here is the situation.. My Dad is working on contract in overseas.. He
has internet access in his hotel.. He wants to be able
Hello Everyone,
I have released AstLinux 0.4.7. This release includes Asterisk
1.2.22. More here:
http://www.astlinux.org/node/26
--
Kristian Kielhofner
___
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asterisk-users
Darrick, can you tell which mini-itx board you have and what processor it
has on it? I don't them with Pentium processors, instead they have some VIA
C3 and C7 processors, which are completely new to me and I have no idea how
will they perform with Asterisk.
On 7/21/07, Darrick Hartman (lists)
the best way attended transfer. See my feature.conf:
example:
[general]
; Call parking configuration
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in, need to
INCLUDE this in
On 7/21/07, Andrew Joakimsen wrote:
Check this out: http://www.kapanga.net/IP/home.cfm
Very easy to create a self-installing pre-configured soft phone.
I don't see IAX listed as one of the features, do you know
if it is supported ?
http://www.kapanga.net/IP/specs.cfm
--
Yes Moises, i was looking for it.
The main problem is only on the files for version 1.4... it give that
error when no CallerID is recive or a private caller id is recive.
The change i made is to add to Mexico variant on mfcr2.c this line
mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12;
It is possible that the hotel is only allowing certain
incoming/outgoing ports as well (i.e. just allowing DNS and HTTP
traffic). A VPN *might* help with that.
On 7/21/07, WipeOut [EMAIL PROTECTED] wrote:
Hi,
Here is the situation.. My Dad is working on contract in overseas.. He
has internet
On Sat, Jul 21, 2007 at 04:35:21PM -0700, Jay Wilton wrote:
Jay Wilton wrote:
gdb /usr/sbin/asterisk -c /tmp/core.4545
/tmp/core.4545 is not a core dump: File format not
recognized
Does the user running gdb have proper permissions to the
core file?
I was running gdb as root.
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