On Mon, 1 Mar 2010, Juan C. Villa wrote:
On 2/28/2010 10:21 AM, Gordon Henderson wrote:
On Sun, 28 Feb 2010, Juan C. Villa wrote:
Hey Guys,
I am considering leasing a new server in Germany to run my Asterisk
infrastructure and I was wondering how response time would affect the
performance
Am 02.03.2010 07:26, schrieb Zhang Shukun:
hi, all
i want to realize more secure communication between asterisk sip end users.
so i want to know Does Asterisk 1.6.2.1 Support SIP TLS encryption?
yes. But Asterisk does not support SRTP. Thus, only the SIP signaling is
encrypted, not the
Am 02.03.2010 08:50, schrieb Magnus Benngård:
Hi,
Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
problem to get outgoing calls to work but i have some problems with
incoming.
Did set srvlookup=yes in sip.conf. Sending all outgoing calls to
sip-corporate.tele2.se which
Thank you for your answer, Nic.
It seems that by putting rtcachefriend=yes, the qualify works as
expected and even changes made to my realtime MySQL-DB take affect
immediately without the need of a reload (I changed the username and
name).
However the old username and name are still valuable and
In article 3de056a31003010645x2c4481fbr5b05923d88614...@mail.gmail.com,
David Backeberg dbackeb...@gmail.com wrote:
On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote:
I am trying to get the usernum of a user when dialing in to a MeetMe
conference. Is there somehow a possibility to
Dear All,
How can we know the On board supports echo cancellation
I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02)*board
all working fine but sometimes i got echo when user are calling a PRI.
is there any way to know on board echo cancellation .
regards
Dhaval
--
I'd like to add to my thread that realtime SIP peers do not seem to be
surviving a sip reload.
step 1 : 2 realtime SIP peers are registered to Asterisk, they can make
a phone call to each other.
step 2 : I do a 'sip reload'
step 3 : the 2 realtime SIP peers are no longer able to phone to each
The problems we have with Asterisk Realtime:
1. After reload all registrations are void.
2. Without reload prune does not take effect.
Test it in your scenario also.
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL:
Asterisk 1.4.29
BLF-SUBSCRIBE go to internal IP (ngrep output):
U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 - 62.134.xxx.xxx:5060
SUBSCRIBE sip:1...@62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From:
Hi,
I need some help debugging a sip situation.
I started to have problems with sip trunks, using more than one trunk (and
sometimes using only one) the sip module seems to freeze.
My local extensions lost registration and also the trunks. The only way
that I can restart the sip is removing
jonas kellens wrote:
I'd like to add to my thread that realtime SIP peers do not seem to be
surviving a sip reload.
step 1 : 2 realtime SIP peers are registered to Asterisk, they can
make a phone call to each other.
step 2 : I do a 'sip reload'
step 3 : the 2 realtime SIP peers are no
Hi all,
We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to
any newer releases:
We use the following cli command to feed a wave/mp3 file into an existing
conference on an other serve:
/opt/asterisk/sbin/asterisk -r -x channel originate
Local/confgongad...@xy_features
Hi!
I started to have problems with sip trunks, using more than one trunk
(and sometimes using only one) the sip module seems to freeze... My local
extensions lost registration and also the trunks. The only way that I
can restart the sip is removing the trunks...
Have seen this also on
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02) board
all working fine but sometimes i got echo when user are calling a PRI.
is there
Hi!
Did a setup of 2 peers as Klaus suggested, it worked thx!
Has anyone thought about the possibility to add multiple ip/hosts to
host=?
I my case: host=130.244.190.42,130.244.190.46 or
host=sip-corporate1.tele2.se,sip-corporate2.tele2.se
Step 1 could be to send to the first ip/host and
There is a problem that bothered me for a long time:
Since one of the 1.6.0.x patch releases up until 1.6.2.5 a dialplan reload
works only once with a bigger dialplan.
If I issue dialplan reload again, it won't do anything. After doing so the
cli won't show responses
to any commands anymore.
So
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
In my experience, yes, that is normal behaviour. Generally any SIP phone
will try to reconnect with the server within 2 mins anyway.
In the Zoiper softphone, it is set to 3600 seconds... I don't want my
customers have to do a lot of
Hi Erik, thanks for your help, I found a solution, but this problem only
happens when my server reboot.
I put permissions in:
chmod 777 /var/lib/asterisk/agi-bin/recirdingcheck ...and
chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi
I think the problem is FreePBX not Asterisk.
But
On Tue, Mar 02, 2010 at 08:28:52AM -0500, carem gyssell nieto wrote:
Hi Erik, thanks for your help, I found a solution, but this problem only
happens when my server reboot.
I put permissions in:
chmod 777 /var/lib/asterisk/agi-bin/recirdingcheck ...and
chmod 777
On Tue, Mar 02, 2010 at 01:56:37PM +0100, Andreas Brodmann wrote:
There is a problem that bothered me for a long time:
Since one of the 1.6.0.x patch releases up until 1.6.2.5 a dialplan reload
works only once with a bigger dialplan.
If I issue dialplan reload again, it won't do anything.
Hi Tzafrir,
yes, I will have to 'anonymize' the dialplan, is this list the right place
though?
-Andreas
2010/3/2 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Mar 02, 2010 at 01:56:37PM +0100, Andreas Brodmann wrote:
There is a problem that bothered me for a long time:
Since one of
jonas kellens wrote:
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
In my experience, yes, that is normal behaviour. Generally any SIP phone
will try to reconnect with the server within 2 mins anyway.
In the Zoiper softphone, it is set to 3600 seconds... I don't want my
Hi,
It's your S.O firewall disable? It's SeLinux disable? you can see that with
the 'setup' command if you are using some red hat distribution.
If you use 'sip show peers' command in your CLI you can see the sip
peers?if notthe problem is your manager connection between Asterisk
and
On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02) board
all working
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
If you are changing RealTime config in your DB you need to do a sip
prune realtime either directly from asterisk cli or using AMI. You
really do not need to do a SIP reload when changing the config of one
sip extension.
I notice that
Dear Asterisk users,
I have a simple question, but guess the answer is not that simple :-)
What I want to achieve is to hide time consuming processing by a
prompt (load of a customer history), stop the prompt and come
back to the dial plan when the information is available. I'm actually
using
Sip reload
Regards,
Mindaugas Kezys
Kolmisoft UAB
VoIP Billing Solutions
e-mail: mailto:i...@kolmisoft.com i...@kolmisoft.com
URL: http://www.kolmisoft.com http://www.kolmisoft.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Tue, 2 Mar 2010, Patrick wrote:
What I want to achieve is to hide time consuming processing by a
prompt (load of a customer history), stop the prompt and come back to
the dial plan when the information is available. I'm actually using AGI
script.
Many moons ago I wrote an AGI (written
- Brian brel.astersik100...@copperproductions.co.uk escreveu:
On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard
Gordon,
Thank you very much for the detailed insights! I really appreciate it. I'm
gonna test drive a server in Germany today. The main reason for choosing a
server in Germany is COST ($65 vs $200).
Thanks!
-
Juan C. Villa
Computer Engineering
Georgia Institute of Technology
On Tue, 2010-03-02 at 12:45 -0300, Vinícius Fontes wrote:
- Brian brel.astersik100...@copperproductions.co.uk escreveu:
On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu:
Dear All,
How can we know the On
Steve Edwards wrote:
On Tue, 2 Mar 2010, Patrick wrote:
What I want to achieve is to hide time consuming processing by a
prompt (load of a customer history), stop the prompt and come back to
the dial plan when the information is available. I'm actually using AGI
script.
This sort of
jonas kellens wrote:
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
If you are changing RealTime config in your DB you need to do a sip
prune realtime either directly from asterisk cli or using AMI. You
really do not need to do a SIP reload when changing the config of one
sip
On Tue, 2 Mar 2010, Brian wrote:
It would be nice to resolve this - but it's probably beyond my
understanding and ability.
Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi
directory?
Gordon
--
_
--
Hi all,
I need to check in realtime the calls that my asterisk is menaging:
1) SIP peers status and with who are talking.
2) IAX peers status and with who are talking
3) elapsed talking time
Some one could show me the way to realize that?
Any help are really appreciated
Thanks a lot in advance
On Tue, 2 Mar 2010, Juan C. Villa wrote:
Gordon,
Thank you very much for the detailed insights! I really appreciate it.
I'm gonna test drive a server in Germany today. The main reason for
choosing a server in Germany is COST ($65 vs $200).
I'm very surprised to hear that co-lo's in the US
On Tue, 2010-03-02 at 15:59 +, Brian wrote:
On Tue, 2010-03-02 at 12:45 -0300, Vinícius Fontes wrote:
- Brian brel.astersik100...@copperproductions.co.uk escreveu:
On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote:
- DHAVAL INDRODIYA dhaval.it01...@gmail.com
lore wrote:
Hi all,
I need to check in realtime the calls that my asterisk is menaging:
1) SIP peers status and with who are talking.
2) IAX peers status and with who are talking
3) elapsed talking time
Some one could show me the way to realize that?
Any help are really appreciated
Hi,
I have tried the new revision (769) with Asterisk SVN-trunk-r240667M (~ 1.6.2)
and it compiles without warnings or errors (see attached make output). It also
seems to work flawlessly. I can make and receive calls from/to the PSTN with
the Fritz card PCI via the BRI and route them to my
On Tue, 2010-03-02 at 16:27 +, Gordon Henderson wrote:
On Tue, 2 Mar 2010, Brian wrote:
It would be nice to resolve this - but it's probably beyond my
understanding and ability.
Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi
directory?
Gordon
Ah Gordon!
Dear Sirs,
I
installed AsteriskNOW 1.5 and the CDR is working with mysql as unique
ID, when you use call recording these files are stored in / var / spool
/ asterisk / monitor. but wanting to see through the ARI monitor leaves the
column blank.
What could be the problem.
Luis
On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote:
Dear All,
How can we know the On board supports echo cancellation
I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev
02) board
all working fine but sometimes i got echo when user are calling a PRI.
is
Hi,
thanks a lot for the reply,
yes I would like to put data in a web interface (maybe php made better
if already done :) ).
I'm reading something about dymanic realtime: could be ok for my needs?
Or is better spent my time on this docs :
http://www.voip-info.org/wiki/view/Asterisk+manager+API ?
On Tue, 2010-03-02 at 15:59 +0100, jonas kellens wrote:
On Tue, 2010-03-02 at 11:32 +, Ishfaq Malik wrote:
If you are changing RealTime config in your DB you need to do a sip
prune realtime either directly from asterisk cli or using AMI. You
really do not need to do a SIP reload when
Hi Philipe
Hi!
I started to have problems with sip trunks, using more than one trunk
(and sometimes using only one) the sip module seems to freeze... My local
extensions lost registration and also the trunks. The only way that I
can restart the sip is removing the trunks...
Have seen this
On Tue, 2010-03-02 at 16:49 +, Brian wrote:
On Tue, 2010-03-02 at 16:27 +, Gordon Henderson wrote:
On Tue, 2 Mar 2010, Brian wrote:
It would be nice to resolve this - but it's probably beyond my
understanding and ability.
Did you un-comment the 2 lines in Kbuild in the
On 3/2/2010 12:29 PM, Gordon Henderson wrote:
On Tue, 2 Mar 2010, Juan C. Villa wrote:
Gordon,
Thank you very much for the detailed insights! I really appreciate it.
I'm gonna test drive a server in Germany today. The main reason for
choosing a server in Germany is COST ($65 vs $200).
We are having an issue with Asterisk 1.6.1 and the MWI turning on when a
user doesn't have voicemail. We see random MWI lights come on and the phone
indicates a random number of messages (its been anywhere from 1-14) when a
server reload is done.
I just checked one user, they have no messages old
Does Asterisk know when it hits a voicemailbox ?
When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.
If Asterisk does not know it's a voicemailbox that has answered the
call, the voicemailbox will contain 60minutes of 'silence'. This is very
On Mar 2, 2010, at 2:37 PM, jonas kellens wrote:
Does Asterisk know when it hits a voicemailbox ?
When calling to a cell-phone or GSM, after some rings and no pickup you
arrive at a voicemailbox.
If Asterisk does not know it's a voicemailbox that has answered the call, the
voicemailbox
Please need help!!.
thx
From: lcr_2...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: ARI problem with monitor
Date: Tue, 2 Mar 2010 16:54:51 +
Dear Sirs,
I
installed AsteriskNOW 1.5 and the CDR is working with mysql as unique
ID, when you use call recording these files
On Tuesday, March 2, 2010, Andreas Brodmann andreas.brodm...@gmail.com wrote:
Hi Tzafrir,
yes, I will have to 'anonymize' the dialplan, is this list the right place
though?
-Andreas
How big is your dialplan? How many lines / file size, etc. Are you
using ael or lua or just the original
I've just got Uverse installed. I had dsl, but ATT insisted I couldn't
keep my old dsl, but had to switch to Uverse internet - vdsl.
My setup:
linux box as router : 10.10.11.252
asterisk box: 10.10.11.180
10.10.11.252 is multihomed and connected to the Uverse Residential
Gateway.
On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
I've just got Uverse installed. I had dsl, but ATT insisted I couldn't
keep my old dsl, but had to switch to Uverse internet - vdsl.
My setup:
linux box as router : 10.10.11.252
asterisk box: 10.10.11.180
10.10.11.252 is
Hi Warren,
the dialplan currently holds 1792 lines. It's a plain old .conf file.
-Andreas
2010/3/2 Warren Selby wcse...@selbytech.com
On Tuesday, March 2, 2010, Andreas Brodmann andreas.brodm...@gmail.com
wrote:
Hi Tzafrir,
yes, I will have to 'anonymize' the dialplan, is this list
Hello all!
I having some trouble with a OpenVox A1200P card equiped with 5 FXO
and 7 FXS ports, all ok with FXO ports, but the FXS ones are having
some strange problem:
With a telephone connected to any FXS port, when i dial some
extension number on this phone, i receive a busy
Fred Posner wrote:
On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
I've just got Uverse installed. I had dsl, but ATT insisted I couldn't
keep my old dsl, but had to switch to Uverse internet - vdsl.
My setup:
linux box as router : 10.10.11.252
asterisk box: 10.10.11.180
You need to set your firewall public ip to dhcp in order for Uverse
dmz to work.
Thanks,
--Warren Selby
On Mar 2, 2010, at 8:53 PM, sean darcy seandar...@gmail.com wrote:
Fred Posner wrote:
On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
I've just got Uverse installed. I had dsl, but ATT
Hi all,
What i'm going to do is that enable caller sing while playing a
background music. My approach is using Monitor and Meetme app.
Caller make a call to asterisk, asterisk join caller in to a voice
conference and create a dummy caller which will play music. Monitor app
record both music and
Hi all,
The question has already asked here,
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg98176.html
but it's been two years since then, so is there any better solution with
latest release version?
Quyps
--
Hi,
Carlos
I checked dmesg on my server and i found following message
what is meaning for this ? i cant understand
VPM400: Not Present
VPM450: echo cancellation for 128 channels
VPM450: hardware DTMF disabled.
VPM450: Present and operational servicing 4 span(s)
regards
Dhaval
On Tue, Mar 2,
Hi All,
I have followed below steps to enable echo cancellation.
# cd /usr/src
# wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
# tar xjf linux-2.6.28.tar.bz2
# tar zxvf dahdi-linux-2.1.0.4.tar.gz
# ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi
# mkdir
On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote:
Hi All,
I have followed below steps to enable echo cancellation.
# cd /usr/src
# wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
# tar xjf linux-2.6.28.tar.bz2
# tar zxvf dahdi-linux-2.1.0.4.tar.gz
# ln -s
On Wed, 2010-03-03 at 11:31 +0530, Chandrakant Solanki wrote:
Hi All,
I have followed below steps to enable echo cancellation.
# cd /usr/src
# wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2
# tar xjf linux-2.6.28.tar.bz2
# tar zxvf dahdi-linux-2.1.0.4.tar.gz
# ln -s
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