Hello David,
you may start 2 Asterisk instances on the same machine,
one bind to 1.1.1.1 IP-address, the other to 2.2.2.2.
Just configure the appropriate settings in each instance asterisk.conf —
you’ll have to set correct directories like astspooldir, agi-bin and so on.
And of course create
Thanks for the suggestions. We'd prefer not to complicate the architecture
with additional proxies in front, so will try setting the Linux network
routes to see if that helps.
On Fri, 30 Oct 2020 at 16:24, John Runyon wrote:
> David, can you play around with the routing table and get the OS to
I didn't want to post this because its kind of ugly, but we *did*
actually do it a number of years ago to get around this issue with chan_sip.
Our original architecture was based on LXC, and we had large servers
running hundreds of containers, each running asterisk. The "host" ran
asterisk
I didn't want to post this because its kind of ugly, but we *did*
actually do it a number of years ago to get around this issue with chan_sip.
Our original architecture was based on LXC, and we had large servers
running hundreds of containers, each running asterisk. The "host" ran
asterisk
Run rtp proxy on the asterisk box (not sure if it would work since you
can't use the same ports).
On Thu, Oct 29, 2020 at 11:03 PM David Cunningham
wrote:
> Hi Dovid,
>
> We can change the SDP in Kamailio, but Asterisk will still send its RTP
> from its default address. The remote end is strict
David, can you play around with the routing table and get the OS to handle
it for you? So long as asterisk isn’t calling bind() (or is calling with
0.0.0.0) I would imagine adding a route for the peer, with your normal
gateway, and the correct device would work.
On Thu, Oct 29, 2020 at 10:04 PM
I didn't want to post this because its kind of ugly, but we *did*
actually do it a number of years ago to get around this issue with chan_sip.
Our original architecture was based on LXC, and we had large servers
running hundreds of containers, each running asterisk. The "host" ran
asterisk
Hi Dovid,
We can change the SDP in Kamailio, but Asterisk will still send its RTP
from its default address. The remote end is strict about accepting RTP from
the specified source and won't accept it. Have you any suggestions to solve
that problem?
Thank you.
On Fri, 30 Oct 2020 at 14:49, Dovid
Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
On Thu, Oct 29, 2020 at 20:44 David Cunningham
wrote:
> Hello,
>
> Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
>
Hello,
Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
address for its end of the communication for a specific device? Something
like:
[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44
This is for use on a server with multiple IP addresses. There is
OK, thank you George.
On Sat, 24 Oct 2020 at 03:16, George Joseph wrote:
>
>
> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi George,
>>
>> Thank you for the response. I'm a little unclear on what you mean by a
>> transport. We're using chan_sip,
On Thu, Oct 22, 2020 at 4:13 PM David Cunningham
wrote:
> Hi George,
>
> Thank you for the response. I'm a little unclear on what you mean by a
> transport. We're using chan_sip, not pjsip.
>
> Do you mean a device in sip.conf, using bindaddr to set the address to
> bind for that device? We've
Hi George,
Thank you for the response. I'm a little unclear on what you mean by a
transport. We're using chan_sip, not pjsip.
Do you mean a device in sip.conf, using bindaddr to set the address to bind
for that device? We've only used bindaddr in the [general] section before,
but if it will work
On Wed, Oct 21, 2020 at 9:16 PM David Cunningham
wrote:
> Hello,
>
> We have an Asterisk server with two public IP addresses, let's say 1.1.1.1
> and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
> dialled from Asterisk to an external destination. The external
Hello,
We have an Asterisk server with two public IP addresses, let's say 1.1.1.1
and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
dialled from Asterisk to an external destination. The external destination
sees the SIP packet as coming from 1.1.1.1 and the media address
15 matches
Mail list logo