Is Kamalio running on the same system as Asterisk?
On 21/01/2014 2:41 PM, David Cunningham wrote:
Hi Larry,
Thanks for the reply. We have all of those settings left out of our
sip.conf, so this should allow everything, right?
On 21 January 2014 17:38, Larry Moore mailto:lmo...@omninet.net.au
Cool
That looks like it is arriving at Asterisk - are you sure asterisk is not
getting it? If you turn on sip debug in asterisk can you see the SIP packets?
It maybe asterisk is ignoring them or replying to them but its going out an
interface you hadn’t thought of, I have had that a few times.
Hi Larry,
Thanks for the reply. We have all of those settings left out of our
sip.conf, so this should allow everything, right?
On 21 January 2014 17:38, Larry Moore wrote:
> Have you checked your localnet=, deny=, permit=, contactdeny= &
> contactpermit= settings?
>
> My 2c worth.
>
>
> On 2
Have you checked your localnet=, deny=, permit=, contactdeny= &
contactpermit= settings?
My 2c worth.
On 20/01/2014 10:51 AM, David Cunningham wrote:
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem i
Hi Duncan,
Thank you for your reply. Here's the netstat:
[root]# netstat -udpln | grep asterisk
udp0 0 0.0.0.0:50000.0.0.0:*
6672/asterisk
udp0 0 0.0.0.0:45200.0.0.0:*
6672/asterisk
udp0 0 0.0.0.0:50600.0.0.0:*
On 21/01/2014, at 6:40 pm, David Cunningham wrote:
> Hi Paul,
>
> Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
> and arriving at the Asterisk server. This is why it's a mystery that Asterisk
> doesn't see the call coming in. We tried removing the firewall (so ip
Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried removing the firewall (so
iptables -L shows no rules at all) but that didn't help unfortunatel
Hi Eric,
Thanks for the suggestion. It was on bindaddr of 0.0.0.0, but we tried
removing that too, and Asterisk still doesn't see anything.
On 21 January 2014 09:18, Eric Wieling wrote:
> Make sure you do NOT have any *bindaddr options set in your sip.conf. If
> you do, you are telling Aster
Ok, so now I just feel kind of stupid. After I got home I decided to play with
this a little more.
After far too long I realized that part of the issue was Asterisk parsing the ;
as a beginning of a comment (hindsight=duh).
A little bit more experimenting and (though I could swear I tried this b
On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham
wrote:
> Hi Paul,
>
> The ngrep on the Asterisk server does show it being received. Have you any
> idea what would prevent it getting from the network stack to Asterisk on
> that machine?
>
Well, you need to use tcpdump on each hop across your netw
Make sure you do NOT have any *bindaddr options set in your sip.conf. If you
do, you are telling Asterisk to not allow the OS to pick the source IP and
hence the routing.
The *bindaddr options are seldom useful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:
Hi Duncan,
The Asterisk machine also has a VPN IP address, so it has a route for 172.x
addresses to go to tun0 VPN interface.
On 21 January 2014 08:30, Duncan Turnbull wrote:
> On 21/01/2014, at 10:24 am, David Cunningham
> wrote:
>
> Hi Paul,
>
> The ngrep on the Asterisk server does show i
Hi All,
In the midst of trying to pilot a deployment of Microsoft Lync (mainly for
non-voice collaboration, specifically IM) and integrate it with our Asterisk
(11.6.0 if it matters) deployment and a "everything in one place" tool when
people are out of the office.
I have everything on the voi
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
I'm looking for the best / recommended solution for automatic discovery
of phone numbers for a multiple Asterisk system. This would be for an
administration, with many branches (~30), but a common infrastructure
(DNS, LDAP). Most branches wou
On 21/01/2014, at 10:24 am, David Cunningham wrote:
> Hi Paul,
>
> The ngrep on the Asterisk server does show it being received. Have you any
> idea what would prevent it getting from the network stack to Asterisk on that
> machine?
>
>
Have you got a static route on asterisk or your defaul
Hi Paul,
The ngrep on the Asterisk server does show it being received. Have you any
idea what would prevent it getting from the network stack to Asterisk on
that machine?
On 21 January 2014 05:30, Paul Belanger wrote:
> On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham
> wrote:
> > Hi,
> >
>
What is dahdi.auto_assign_spans and why should you care?
In later versions the kernel module dahdi[Q] includes a new parameter:
auto_assign_spans. It defaults to 1, and if you set it to 0, DAHDI can
start behaving in strange and completely expected ways. Chances are the
default will be set to 1 in
On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham
wrote:
> Hi,
>
> We have a Kamailio and Asterisk cluster, both machines being on a real 103.x
> IP address and also on a 172.x OpenVPN address.
>
> The problem is that when Kamailo receives a call from the VPN and forwards
> it to the Asterisk serv
On 01/19/2014 08:40 AM, Steve Murphy wrote:
Here's another idea! How about changing your port from 5060 to something
different, maybe 7067 or some other number that is not popularly being
used?
You'll provision your phones to use this port, and the scanners will not
find you. Seems a much simp
>
>
> I see MANY of these in my log files:
>
>
> [Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from '"202"
> ' failed for '37.8.12.147:26832' - Wrong password
> [Jan 15 03:06:19] NOTICE[14129] chan_sip.c: Registration from '"5001"
> ' failed for '37.8.12.147:21268' - Wrong password
> [Ja
Hi every body
our Calls are begging dropped for no reason and it starts with the sound
quality dropping and then the caller unable to hear our call center agents.
Then the call drops or the caller hangs up unable to hear.
I could see following lines inside full log
--
Dears,
There is a PSTN line shared between 2 asterisk servers, (openvox 4FXO
lines)
The outgoing call of the one server may be conflict with the established
call of the other one,
is any way to force the Asterisk or Dahdi to dial after hearing the Dial
tone ?
--
Pezhman Lali
--
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