17 mar 2010 kl. 16.37 skrev Kevin Sandy:
We're having an odd issue with codec negotiation from one of our SIP
providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and G723.
In our response, we send back that we support G711 and G729.
19 mar 2010 kl. 03.41 skrev Philipp von Klitzing:
Hey hey!
My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.
Asterisk doesn't differentiate between a hard phone and a soft
On Sat, 20 Mar 2010, Daniel Bareiro wrote:
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Hash: SHA1
Hi all!
I'm testing with a Grandstream BT200 telephone and, according to I read,
it has a LED that blinks if for that extension messages were left.
In Voice Mail UserID, under the ACCOUNT tab, I put
Hy guys i am having so much hard time to setup asterisk on a virtual machine
that i got , i just want to know if i really need to use Dahdi and libpri on a
complete Digital PBX i just gonna use sip and iax.
I will never use any kind of analog line on this machine.
Wait for a feed back.
Daniel
On Sun, Mar 21, 2010 at 1:30 PM, Daniel Leite de Abreu
dlab...@gmail.com wrote:
Hy guys i am having so much hard time to setup asterisk on a virtual machine
that i got , i just want to know if i really need to use Dahdi and libpri on
a complete Digital PBX i just gonna use sip and iax.
I
even if i am using iax trunking?
On 21 Mar 2010, at 10:30 AM, Randy R wrote:
On Sun, Mar 21, 2010 at 1:30 PM, Daniel Leite de Abreu
dlab...@gmail.com wrote:
Hy guys i am having so much hard time to setup asterisk on a virtual machine
that i got , i just want to know if i really need to use
In that case, you need neither of them.
On 03/21/2010 08:30 AM, Daniel Leite de Abreu wrote:
Hy guys i am having so much hard time to setup asterisk on a virtual machine
that i got , i just want to know if i really need to use Dahdi and libpri on
a complete Digital PBX i just gonna use sip
Even if you're using IAX.
On 03/21/2010 08:48 AM, Daniel Leite de Abreu wrote:
even if i am using iax trunking?
On 21 Mar 2010, at 10:30 AM, Randy R wrote:
On Sun, Mar 21, 2010 at 1:30 PM, Daniel Leite de Abreu
dlab...@gmail.com wrote:
Hy guys i am having so much hard time to setup
My worries is about timing , because i know that ztdummy give me the timing
source , and i know that wend you install DAHDI also install ztdummy and i hear
that for iax trunking we need ztdummy fir timing it.
Must i use Dahdi or can i go free just on Asterisk?
Thanks
Daniel Abreu.
On 21
On Sun, Mar 21, 2010 at 12:48:27PM +, Daniel Leite de Abreu wrote:
even if i am using iax trunking?
IAX trunking does require Asterisk to have a source of timing. Up until
1.6.0 this requires DAHDI. As of 1.6.1 there is a difefrent source of
timing. Though res_timing_pthread is considered
On Sun, 21 Mar 2010, Daniel Leite de Abreu wrote:
Hy guys i am having so much hard time to setup asterisk on a virtual
machine that i got , i just want to know if i really need to use Dahdi
and libpri on a complete Digital PBX i just gonna use sip and iax. I
will never use any kind of
On Sat, 2010-03-20 at 21:08 +, Daniel Leite de Abreu wrote:
Sorry but i did not understand how did you built it?
Sorry that it was not clear. Here are the full steps for 2.2.0.2:
$ tar -xzf dahdi-linux-2.2.0.2.tar.gz
$ cd dahdi-linux-2.2.0.2/
$ DAHDIVERSION=2.2.0.2
On Sun, Mar 21, 2010 at 12:30:01PM +, Daniel Leite de Abreu wrote:
Hy guys i am having so much hard time to setup asterisk on a virtual machine
that i got , i just want to know if i really need to use Dahdi and libpri on
a complete Digital PBX i just gonna use sip and iax.
I will never
Hi Thanks very much for reply it and helping me out.
This is the out put
-bash-3.2# ls -l /lib/modules/2.6.18-164.6.1.el5xen/
total 1280
lrwxrwxrwx 1 root root 54 Nov 6 23:31 build -
../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64
drwxr-xr-x 2 root root 4096 Nov 3 17:31 extra
Virtual machine will not be able to access dahdi hardware anyways (don't
know about KVMs though) but for conferencing you do need it, disabling all
the other hardware info in its config and enabling ztdummy only. Libpri you
can ignore. Config I am refering to in 1.4 versions was
Hi David
I really try your tip but i have no joy.
Still stuck on the same error.
Do you have any others ideas?
Thanks
Daniel
On 17 Mar 2010, at 12:29 PM, David Backeberg wrote:
On Wed, Mar 17, 2010 at 10:16 AM, Daniel Leite de Abreu
dlab...@gmail.com wrote:
-bash-3.2# cd
Hello there,
I'm currently building a PHP-based software to help users make batch
calls. Basically, users provide a script and list of phone numbers.
The system calls those numbers and plays the script to whoever picks up
the phone.
Currently, the system does one call at a time via direct
On 03/21/2010 09:01 AM, Daniel Leite de Abreu wrote:
My worries is about timing , because i know that ztdummy give me the timing
source , and i know that wend you install DAHDI also install ztdummy and i
hear that for iax trunking we need ztdummy fir timing it.
Must i use Dahdi or can i go
hi:
only test
Best wishes!
Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7,
elastix, trixbox support.
website:www.cnasterisk.com, www.voip88.com
voip88
_
Test successful
On 2010-03-21 9:12 AM, card support asterisk asteriskc...@hotmail.com
wrote:
hi:
only test
Best wishes!
Asterisk Support group(sangoma, digium...), providing asterisk conf, pri,
ss7, elastix, trixbox support.
website:www.cnasterisk.com, www.voip88.com
voip88
Hi,
on one of our clients asterisk server we have the problem that you hear
nothing on external calls.
Here are the details abount the system:
Asterisk 1.6.0.22
DIGIUM Wildcard B410 quad-BRI card (rev 01)
dahdi-linux-complete-2.2.0+2.2.0
I have setup the following test extension:
Hi Olle!
The work I started during Christmas - Named ACL's - is a starting point
that other developers can use to develop all kind of schemes.
http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists
-asterisk-nacls/
Very interesting. Doesn't look like this has any chance
Daniel Leite de Abreu wrote:
Hi guys!
I am new here on the list , and this is my first question, and i want to say
tanks to you all for the help that i will get from you all.
Ok there is my problem.
I rent a virtual machine ( Server ) here in South Africa and this server is a
Centos
Hi,
On Sat, Mar 20, 2010 at 09:05:45PM +, Daniel Leite de Abreu wrote:
Hi Thanks very much for reply it and helping me out.
This is the out put
-bash-3.2# ls -l /lib/modules/2.6.18-164.6.1.el5xen/
total 1280
lrwxrwxrwx 1 root root 54 Nov 6 23:31 build -
On Sun, Mar 21, 2010 at 07:37:02AM -0400, Zeeshan Zakaria wrote:
Virtual machine will not be able to access dahdi hardware anyways (don't
know about KVMs though)
IIRC that was sorted out at around 2.2.0 .
Anyway, 2.2.1 works great here in kvm.
--
Tzafrir Cohen
icq#16849755
Hello All,
safe_asterisk just sent me an email saying Asterisk on bill exited on
signal 11. Might want to take a peek.. Looking at the
/var/log/asterisk/message doesn't show me anything...
This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and
it is routing calls from Nextone
Hello there,
I'm new to Asterisk and I'm trying to figure out a way to make the
Asterisk Manager Interface (AMI) accessible to multiple users at the
same time. Would anyone recommend an AMI proxy that could be accessed
from PHP code?
Thanks in advance,
Leo
--
Hi!
You can just add several users to manager.conf or you can use AstManProxy...
On 21 March 2010 20:27, Leo Burd l...@media.mit.edu wrote:
Hello there,
I'm new to Asterisk and I'm trying to figure out a way to make the
Asterisk Manager Interface (AMI) accessible to multiple users at the
Hello ,
I have a problem to install asterisk with ldap.
I am doing the following:
make clean
. / configure
make menuselect
LIBS =- lldap
export LIBS
make This is where my error
#make
CC = cc CXX = g + + LD = AR = RANLIB = CFLAGS = make-C
menuselect CONFIGURE_SILENT = - silent menuselect
On Sun, 21 Mar 2010, Zeeshan Zakaria wrote:
Virtual machine will not be able to access dahdi hardware anyways (don't
know about KVMs though)
It will depend on the virtualisation technology you use... I'm currently
using LXC and can easily give a container full hardware access if
required.
Hello,
maybe you could find a core dump file mostly in /tmp where you can use
gdb to find which thread has killed your asterisk.
have a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging
Backtracing a core dump file in /tmp
best regards
steve
Nitesh Divecha schrieb:
Has anyone done this with OpenSIPS? For example where it fronts an
Asterisk cluster with the load balancer module?
Thanks,
Gavin.
On 19/03/2010, Ryan Bullock rrb3...@gmail.com wrote:
Hey Philipp,
You can check out
http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
Good to know. I'll try that. I needed such solution for a client few months
ago.
On 2010-03-21 6:06 PM, Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
wrote:
On Sun, 21 Mar 2010, Zeeshan Zakaria wrote:
Virtual machine will not be able to access dahdi hard...
It will
...
Hi,
I've found the solution.
I remembered, that with IAX2 - DAHDI everything is fine.
Only SIP - DAHDI showed the problem.
It seems, that chan_sip does not open ealry audio,
if progressinband=yes in sip.conf.
progressinband=no is needed for early audio.
Strange!
Anyway, that's ok for me
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Hi, Gordon.
On Sun, 21 Mar 2010, Gordon Henderson wrote:
I'm testing with a Grandstream BT200 telephone and, according to I
read, it has a LED that blinks if for that extension messages were
left.
In Voice Mail UserID, under the ACCOUNT tab, I
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