Thank you very much. This is a good tip. I will see openvpn.
Please have a look at the scientific miracles of CORAN: http://www.55a.net/
Thank you again.
On Thu, Feb 11, 2010 at 10:58 PM, Hans Witvliet h...@a-domani.nl wrote:
Forwarded Message
From: mosbah.abdelkader
On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote:
Hi,
I am breaking my fingers in configuring an asterisk (1.6) to
successfully transmit audio with the following setup:
asterisk, resides in local network, ip is 10.26.208.252
versatel business router (directly connected to a dsl,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
[general]
rtpstart=3
rtpend=30100
so 100 ports available. I know that up to 4 ports per
On Fri, 12 Feb 2010, Armin Schindler wrote:
I had a look at
netstat -nuap
and it shows that a lot of ports are still assigned, even if there is no
channel in use.
But sip show channels show a lot of (unused) entries with no
codec/Format and Last Message like INVITE, REGISTER, OPTIONS.
thanks brian,
yes, i am aware that sip is only responsible for signalling and therefor
my conclusion was, that it
has got something to do with nat / firewall / the router...
meanwhile i´ve got it solved... although the sip-provider tried to
convince me, that the misconfiguration
is on my
In this case, I suggest you modify the login script so that your agents
always start paused. It should be trivial to do.
l.
2010/2/8 Robert Grignon rgrig...@fleetone.com
Not a bad idea... We use queuemetrics and the login is done via Web GUI.
I could easily just send it to pause upon
On Thu, Feb 11, 2010 at 07:36:23AM -0500, Leif Madsen wrote:
Jason Parker wrote:
Brian wrote:
Each time the server is rebooted Asterisk duly
deletes the manually created /var/run/asterisk directory - quite why it
does this I just don't know - perhaps it is a bug?
Your assumption is
On Fri, 2010-02-12 at 14:15 +0200, Tzafrir Cohen wrote:
{snip}
So the questions to ask are, I believe:
* Should asterisk here be run as root? If so: why?
It suits me to run it that way on the device concerned.
* Where should the astvarrundir be?
If you leave the defaults as they are (dictated
On Fri, Feb 12, 2010 at 12:28:53PM +, Brian wrote:
On Fri, 2010-02-12 at 14:15 +0200, Tzafrir Cohen wrote:
{snip}
So the questions to ask are, I believe:
* Should asterisk here be run as root? If so: why?
It suits me to run it that way on the device concerned.
Could you please be
On Fri, Feb 12, 2010 at 11:15:17AM +1100, Lee, John (Sydney) wrote:
What is the output of 'cat /proc/dahdi/1' ?
I did not record it but it just shows every channel as 'red alarm'.
How many channels?
E1 or T1?
What do you have in /etc/zaptel.conf ?
loadzone=au
defaultzone=au
#
# For
My idea is to use a well know port like port 80 (that is not blocked).
Skype for example uses this port.
If you are in a situation where the ISP/government is blocking VoIP you
are probably going to have to encrypt it to get it through, and that may
not even work. I have a client who has
I have still not been able to resolve this problem. It only happens
when the call is routed through an agi script.
Whether the dialling itself is done within the AGI, or the AGI drops
into the dialplan to allow the call to originate there, the result is
the same.
Call is marked as answered and
Using FreePBX, is there a way to play a beep sound when you are connected to a
parked
call? Right now, it's dead silence and we can't tell if the call has
been connected.
--
_
-- Bandwidth and Colocation Provided by
As far as my experience, this problem occurs when the asterisk tries to take
a new channel and teco does not count with any available channels.
Contact your E1/T1 provider and work with them to search on the teco side.
2010/2/12 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Feb 12, 2010 at
hin lee wrote:
Using FreePBX, is there a way to play a beep sound when you are
connected to a parked call? Right now, it's dead silence and we can't
tell if the call has been connected.
I don't know about FreePBX, but under the features.conf, there is:
courtesytone = local/stutter;
Am 11.02.2010 21:09, schrieb Olle E. Johansson:
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
On Fri, 2010-02-12 at 15:18 +0200, Tzafrir Cohen wrote:
On Fri, Feb 12, 2010 at 12:28:53PM +, Brian wrote:
On Fri, 2010-02-12 at 14:15 +0200, Tzafrir Cohen wrote:
{snip}
So the questions to ask are, I believe:
* Should asterisk here be run as root? If so: why?
It suits me to
Hello,
Is it possible to use asterisk in T.38 pass through mode with reinvite?
My fax calls are getting disconnected if canreinvite=yes. It works
only if I make canreinvite=no. Normal calls work in both cases.
Thanks
--
_
--
Brian wrote:
If you leave the defaults as they are (dictated by (!) in asterisk.conf)
then the behaviour you get is Asterisk looking to /var/run/asterisk
which the OS deletes on reboot. The question is why is this the default
behaviour when it breaks systems that clear /var/run
Because it's
On Fri, 2010-02-12 at 10:20 -0600, Kevin P. Fleming wrote:
Brian wrote:
If you leave the defaults as they are (dictated by (!) in asterisk.conf)
then the behaviour you get is Asterisk looking to /var/run/asterisk
which the OS deletes on reboot. The question is why is this the default
On Fri, Feb 12, 2010 at 04:51:29PM +, Brian wrote:
On Fri, 2010-02-12 at 10:20 -0600, Kevin P. Fleming wrote:
Brian wrote:
If you leave the defaults as they are (dictated by (!) in asterisk.conf)
then the behaviour you get is Asterisk looking to /var/run/asterisk
which the OS
Can someone point me to a page about writing a text file to call an
external number and play a TTS with cepstral? I know it includes the
creation of a .call file but beyond that im a bit lost.
--
_
-- Bandwidth and Colocation
On Fri, 12 Feb 2010, mj wrote:
Can someone point me to a page about writing a text file to call an
external number
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
and play a TTS with cepstral? I know it includes the creation of a .call
file but beyond that im a
hello ,
First you check out
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Once you are done with auto dial out then look for Cepstral TTS.
http://www.google.com.pk/search?hl=ensafe=activeq=asterisk+with+cepstralbtnG=Searchmeta=aq=foq=
One more thing that there are other
On
],-
By
On Feb 12, 2010, at 12:45 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 12 Feb 2010, mj wrote:
Can someone point me to a page about writing a text file to call an
external number
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
and play a TTS
Thanks guys...
I already have Cepstral installed I guess I just need to figure out
where in the .call file and format to call cepstral and then the txt
for the message. Thanks again for all of your help!
On Fri, Feb 12, 2010 at 11:50 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
hello ,
As you have done Cepstral for in bound call you have developed contexts for
it, in the similar way in .call file you can specify the context and it will
start execution of commands in the context after dialing the number.
Hope this helps
On Fri, Feb 12, 2010 at 11:01 PM, mj sting3...@gmail.com
On Fri, 12 Feb 2010, mj wrote:
I already have Cepstral installed I guess I just need to figure out
where in the .call file and format to call cepstral and then the txt for
the message. Thanks again for all of your help!
Set the text as a channel variable in the call file. Specify a context,
On Fri, 2010-02-12 at 19:20 +0200, Tzafrir Cohen wrote:
Adding documentation is what you do when things fail to work. I suspect
you use an older init.d script.
You are just being silly now. Good documentation is essential to
everything.
With regards to your comments about the init.d script,
where in the .call file and format to call cepstral and then the txt
for the message.
Application and Data, respectively.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
I have created a call file as follows:
Channel: IAX2/trunk/cell number
Application: swift
Data: test call
it calls my cell then nothing...just hangs up.
On Fri, Feb 12, 2010 at 12:23 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 12 Feb 2010, mj wrote:
I already have Cepstral
On Fri, Feb 12, 2010 at 06:24:02PM +, Brian wrote:
On Fri, 2010-02-12 at 19:20 +0200, Tzafrir Cohen wrote:
Adding documentation is what you do when things fail to work. I suspect
you use an older init.d script.
You are just being silly now. Good documentation is essential to
On Fri, 12 Feb 2010, mj wrote:
I have created a call file as follows:
Channel: IAX2/trunk/cell number
Application: swift
Data: test call
it calls my cell then nothing...just hangs up.
Sounds like you don't have app_swift.so loaded.
What does your console look like?
--
Thanks in
Using the latest trixbox.
On Fri, Feb 12, 2010 at 1:44 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 12 Feb 2010, mj wrote:
I have created a call file as follows:
Channel: IAX2/trunk/cell number
Application: swift
Data: test call
it calls my cell then nothing...just hangs
i had configured asterisk with a minimum dial plan, made 10 extentions. below
is extensions and sip.conf
i want configure dial plan so that
Extention 2000-2005 can receive calls from outside and make calls outside and
can dial all ten extentions.
Extention 2006-2010 can only receive calls
i had configured asterisk with a minimum dial plan, made 10 extentions.
below is extensions and sip.conf
i want configure dial plan so
that
Extention 2000-2005 can receive calls from outside and make calls
outside and can dial all ten extentions.
Extention 2006-2010 can only
receive calls from
i had configured asterisk with a minimum dial plan, made 10 extentions. below
is extensions and sip.conf
i want configure dial plan so that
Extention 2000-2005 can receive calls from outside and make calls outside and
can dial all ten extentions.
Extention 2006-2010 can only receive calls
Hi all,
I am trying to implement call dropping funtionality in asterisk for
911. I mean if all lines are busy and someone wants to dial 911 at
least one line should be dropped. Here is my extensions.conf which i
copied from internet. Could somebody help me figure out what is wrong.
Thanks in
use two contexts, one for internal numbers, and one for outside, and include
the inside phones in the outside context.
On Fri, Feb 12, 2010 at 12:39 PM, cool dude cool_dudeof...@yahoo.co.inwrote:
i had configured asterisk with a minimum dial plan, made 10 extentions.
below is extensions and
Hi, I have a PRI problem where it appears that my system is not
responding to SETUP messages on a channel.
It seems to be retransmitting a significant number of RELEASE messages
to clear a call that is most likely
to be long gone.
This causes a huge issue because I get a bunch of hangup cause 102s
how to allow some extenstions to call outside and some extensions cant call
outside. i am attaching sipand extensions.conf
thx
Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!
http://downloads.yahoo.com/in/internetexplorer/
help
Description: Binary data
--
I got it working now. I was not including context ninioneone in default context.
On Fri, Feb 12, 2010 at 1:49 PM, mir shahnawaz shahnawaz...@gmail.com wrote:
Hi all,
I am trying to implement call dropping funtionality in asterisk for
911. I mean if all lines are busy and someone wants to
I know this is slightly off topic, but I was wondering if anyone can help
with a problem getting my PAP2's to connect to Asterisk. I use a
provisioning file, and I recently re-wrote the files for each PAP2. I had
a small typo and the PAPs logged it as a corrupt file. I corrected the
file, however,
Friday, February 12, 2010, 9:57:42 PM, cool wrote:
how to allow some extenstions to call outside and some extensions
cant call outside. i am attaching sipand extensions.conf
thx
Put the extensions into different contexts, and create outside call
extensions only in the allowed context.
We have inherited an installation with Ast 1.4 and Aastra phones. The
client complains that sometimes the call audio turns tinny and robotic...I
heard it and it sounds wierd.
Has anyone else experienced this? Cause? Solutions?
Thanks,
MD
--
internal numbers are in the internal context right?
[others]
include = internal
;extension rules for dialing out
also change your default context in sip.conf, that default context is hit by
anything incoming that doesn't match as one of your regular phones. set it
to notothers? ;)
Please keep
What codecs are you using? Are the calls internal(local network) only?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Michelle Dupuis supp...@ocg.ca wrote:
We have inherited an installation with Ast 1.4 and Aastra phones. The client
complains that sometimes
No, phone on LAN, through Asterisk box on LAN, through firewall, out to
fiber connection (lots of capacity there), to ITSP.
Codec is uLaw.
This only happens sometimes, so I'm wondering if it's an asterisk bug?
Aastra bug? Network latency? LAN capacity, etc. Never seen this before...
We experienced this a couple of months ago. It went away when
we upgraded the phones to the latest firmware.
Another symptom: temporarily putting the caller on hold cures
the problem sometimes.
Ron
Op 12-02-10 22:34, Tim Nelson schreef:
What codecs are you using? Are the calls internal(local
Since it is sporadic, my guess would be network latency / packet loss
/jitter to ITSP. You may have lots of capacity and they may claim to have
lots of capacity, but what about the links between you and them. Who knows
when/if there is loss and latency and jitter there. Setup wireshark to grab
On Fri, 12 Feb 2010, Peder wrote:
Since it is sporadic, my guess would be network latency / packet loss
/jitter to ITSP. You may have lots of capacity and they may claim to
have lots of capacity, but what about the links between you and them.
Who knows when/if there is loss and latency
There is a statistics area and you can select sip or voip calls to see
calls. It shows packet loss, jitter, latency, out of sequence packets, etc.
It can even play them back, so you can check where the loss is and play back
the call to see if the noise is in the same spot. Here is some info from
I'm still unable to do much with my new 1.6 installation. I just tried
reinstalling, and using the standard debian configuration files, with
just the necessary modifications, in case I had some legacy stuff in
there from earlier versions that was interfering. I'm testing in a xen
domU with
Hi,
I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.
Now I'm trying vanilla 1.6.2 with its official support for dialog-info
+xml notifications with no success. This is what i'm doing:
- Phone A has a key configured as type extension
Thank you Doug! I added courtesytone = beep and that worked!
From: Doug Lytle supp...@drdos.info
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Fri, February 12, 2010 6:40:17 AM
Subject: Re:
hi friend,
thx for the reply i am trying this way to achieve what i want i.e exten 2000 to
2005 can call outside and 2006 to 2010 cant call outside.
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
[2001]
type=friend
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
On Fri, 2010-02-12 at 17:23 -0400, Tim Johnson wrote:
I know this is slightly off topic, but I was wondering if anyone can help
with a problem getting my PAP2's to connect to Asterisk. I use a
provisioning file, and I recently re-wrote the files for each PAP2. I had
a small typo and the PAPs
We experienced this a couple of months ago. It went away when we
upgraded the phones to the latest firmware.
Another symptom: temporarily putting the caller on hold cures the
problem sometimes.
We have Snom 320 phones and had similar problems happening in the call
centre intermittently. We
hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can
make call outside and exten 2006 to 2010 can not make call outside. heres my
dial plan.
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic
Is there some reason why I keep getting this same message from cool
dude over and over and over? And under different subject lines?
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com http://www.rolandschorr.com/
67 matches
Mail list logo
