Hi
I'm having the same issue as someone else who wrote into this list but
feel I have more information to add and that this is possibly a new bug.
We have multiple servers, most running 1.8.7.0 and one running 1.8.18.0
which is our next upgrade candidate (all servers running CentOS 5).
We have
Le 06/02/2013 23:15, kepin sinatra a écrit :
Hi, I tried it the implementation of TLS in asterisk 1.8.4.3 on ubuntu
10.04. I follow the tutorial:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial.
and I use blink as a softphone in ny client in windows. for regular
My apologies if this topic was already discussed in the past.
Here is my scenario:
Network A - 192.168.1.0
1 Asterisk
1 Digium phone
Router does NAT from the public IP to asterisk, and forward ports
5060tcp/udp and 10k-20k udp
Network B - 192.168.1.0
1 Digium phone, registering to the public
For the phone on the public network. you might need to set canreinvite=no.
My guess is that if you listen really closely you would have about a
quarter second of audio before it cuts out. Whenever I have had this
happen it is because the packets didn't know how to reroute from the IP
address
On Thursday 07 February 2013, Frank wrote:
My apologies if this topic was already discussed in the past.
Here is my scenario:
Network A - 192.168.1.0
1 Asterisk
1 Digium phone
Router does NAT from the public IP to asterisk, and forward ports
5060tcp/udp and 10k-20k udp
Network B -
On Tue, 2013-01-29 at 08:32 -0600, Matthew Jordan wrote:
On 01/29/2013 02:52 AM, Ishfaq Malik wrote:
On Wed, 2013-01-16 at 08:06 -0600, Matthew Jordan wrote:
On 01/16/2013 05:31 AM, Ishfaq Malik wrote:
On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote:
Hi Everyone
This issue has
Hello Kepin,
I don's know if there's a difference, I changed order with the same result.
Did you find a different script with CentOS?
Elder
On Wed, Feb 6, 2013 at 6:16 PM, kepin sinatra insanlaks...@gmail.comwrote:
hi daniel, are you sure the command in debian and ubuntu same?
On Wed, Feb
AJS,
That is a solution that I am envisaging.
But I would really love to try to work out with my issue first. It will
allow me to deploy more phones in separates buildlings in the future. If
I do the IAX solution, it means that for every building, I need a box..
Which I would like to prevent.
On Thu, Feb 7, 2013 at 10:26 AM, Frank fr...@efirehouse.com wrote:
AJS,
That is a solution that I am envisaging.
But I would really love to try to work out with my issue first. It will
allow me to deploy more phones in separates buildlings in the future. If I
do the IAX solution, it means
The easiest thing to is renumber one of the networks so they are not using the
same address block.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Thursday, February 07, 2013 12:27 PM
To:
I thought about that.
I will give it a shot tonight and will post back my results in here.
Thanks
On 2/7/13 12:39 PM, Eric Wieling wrote:
The easiest thing to is renumber one of the networks so they are not using the
same address block.
-Original Message-
From:
I don't see how that would really solve anything - instead of the server
sending the 192.168.x.x packets onto the local network, it will send them up
toward the internet and get black-holed. What probably makes more sense would
be to switch the subnet on one of the networks, AND put up a vpn
Or if it's just a couple phones, you might be able to setup a vpn connection
directly on the phone itself - have it vpn into 'HQ' and get an address on that
network. I'm not sure which phones you're using though or what phones support
that setup.
Justin Killen
-Original Message-
Digium phones, which (as far as I can tell with my experience) do not
support VPN yet.
On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen
jkil...@allamericanasphalt.com wrote:
Or if it's just a couple phones, you might be able to setup a vpn
connection directly on the phone itself - have it vpn
I'm using Digium Phones.
I still do not understand why it's not possible to do it the way the
networks are right now.
If the options I mentioned in my sip.conf are enough, then both phones
should use Asterisk as a proxy, and Asterisk should handle all the media.
I will run tcpdump traces
Did you set canreinvite=no in sip.conf on the phone in network B? A phone
that can connect but loses audio is almost a sure sign that it is
reinviting and your rtp packets are not making it to the phone. By turning
canreinvite off, it will keep asterisk in the middle of your sessions and
I'm not sure, but it looks like a command in centos and ubuntu are same ...
i'am also trying to configure TLS on ubuntu but always error on the
softphone blink: transport error.
On Fri, Feb 8, 2013 at 12:23 AM, Daniel - Asterisk earohua...@gmail.comwrote:
Hello Kepin,
I don's know if there's
i think canreinvite is not part of Asterisk 1.8 anymore.
Asterisk 1.8 added directmediapermit and directmediadeny to limit which
peers can send direct media to each other.
On 2/7/13 1:15 PM, Kevin Larsen wrote:
Did you set canreinvite=no in sip.conf on the phone in network B? A
phone that
And actually I did not have directmediadeny=0.0.0.0
But I had directmedia=no.
So I will add the directmediadeny line, and will check it out again
tonight.
On 2/7/13 1:22 PM, Frank wrote:
i think canreinvite is not part of Asterisk 1.8 anymore.
Asterisk 1.8 added directmediapermit and
Hi,
exten = _0614.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}_*${CALLER}*
.wav|av(0}V(0))
This should append the caller number in your recorded file name.
Ensure that you save the callerid in the variable before you're changing it
to MY_CALLERID
exten =
when i start sip reload, doesn't appear about SSL certificate ok, i
install asterisk with :
./configure --enable-xmldoc
make menuselect
make make install
make samples
make config
ok, maybe i try using tshark later...
yes, i'm sure blink is configured for TLS. and i've installed the
certificate
I did follow instructions in debian without problems, this issue arise when
trying with Centos 5.8 and 5.9.
On Debian 6.0.6 i wrote:
./ast_tls_cert -C 10.200.x.y -O Company -d /etc/asterisk/keys/
and I got ca.cert which is working on my Blink phones.
If you have any news please let me know,
On 8/02/2013, at 6:49 AM, Frank fr...@efirehouse.com wrote:
I thought about that.
I will give it a shot tonight and will post back my results in here.
Thanks
On 2/7/13 12:39 PM, Eric Wieling wrote:
The easiest thing to is renumber one of the networks so they are not using
the same
Hello,
I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme)
with another person, and a third person join our conference when the third
person leave the conference I get disconnected from the original conference
with a second party. I hope this clear.
This does not happen
motty cruz wrote:
Hello,
I'm running Asterisk 1.8.10 on Linux box, when I'm in a
conference(meetme) with another person, and a third person join our
conference when the third person leave the conference I get
disconnected from the original conference with a second party. I
hope this clear.
Got it to work tonight.
So once again this is my network:
Network A: 192.168.1.x
Network B: 192.168.1.x
In between, the internet.
Asterisk is in Network A.
1 Digium phone is in network A.
Router from network A does NAT and forward (for now):
- 5060 TCP/UDP to internal IP of asterisk
- 10k-20k
Hi, everybody,
Where can I get the manual or user guide of latest asterisk version,
1.11.x?
I want to know the syntax and usage of all the supported functions or
something like that in the latest version.
Thanks in advance
Ding Peng
--
On Fri, Feb 8, 2013 at 11:05 AM, Ding Peng roc.dingp...@gmail.com wrote:
Hi, everybody,
Where can I get the manual or user guide of latest asterisk version,
1.11.x?
I want to know the syntax and usage of all the supported functions or
something like that in the latest version.
Thanks in
-Original Message-
From: Carlos Alvarez car...@televolve.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk
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