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 too, as the gateway for all the container instances.
We once used two of those containers to run asterisk on specific host
interfaces (one instance bridged to one nic, the other to the other).
The host asterisk would route calls out one container or the other, with
the effect you are looking for...
Cheers,
Jeff LaCoursiere
StratusTalk, Inc.
On 10/29/20 7:42 PM, David Cunningham wrote:
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 the
"extenip" setting, but it's really designed for NAT, and can only
appear in the [general] section.
Any suggestions would be greatly appreciated.
On Sat, 24 Oct 2020 at 09:43, David Cunningham
<[email protected] <mailto:[email protected]>> wrote:
OK, thank you George.
On Sat, 24 Oct 2020 at 03:16, George Joseph <[email protected]
<mailto:[email protected]>> wrote:
On Thu, Oct 22, 2020 at 4:13 PM David Cunningham
<[email protected] <mailto:[email protected]>>
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 only used
bindaddr in the [general] section before, but if it will
work in a device that could be the answer.
Sorry. I just assume chan_pjsip these days. Not sure how
you'd do it for chan_sip.
On Fri, 23 Oct 2020 at 00:13, George Joseph
<[email protected] <mailto:[email protected]>> wrote:
On Wed, Oct 21, 2020 at 9:16 PM David Cunningham
<[email protected]
<mailto:[email protected]>> 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 destination sees the SIP
packet as coming from 1.1.1.1 and the media
address in the SDP is 1.1.1.1, which is great.
However if we receive a call in to 2.2.2.2 then
the call dialled from Asterisk to an external
destination still comes from 1.1.1.1, whereas we
want it to come from 2.2.2.2. The source of any
dialled call (the IP packet and the SDP media
address) should be the same as the address the
related inbound call was received to.
For example:
INVITE received to 1.1.1.1:5060
<http://1.1.1.1:5060> -> Asterisk dials
[email protected]
<mailto:[email protected]> -> INVITE
sent from 1.1.1.1:5060 <http://1.1.1.1:5060> to
termination.com <http://termination.com>
INVITE received to 2.2.2.2:5060
<http://2.2.2.2:5060> -> Asterisk dials
[email protected] <mailto:[email protected]>
-> INVITE sent from 2.2.2.2:5060
<http://2.2.2.2:5060> to pstn.com <http://pstn.com>
Does anyone know how this can be achieved?
If termination.com <http://termination.com> is only on
1.1.1.1 and pstn.com <http://pstn.com> is only on
2.2.2.2, create 2 transports, one specifically bound
to 1.1.1.1, transport-1.1.1.1 for instance, and
another to 2.2.2.2 <http://2.2.2.2>:
transport-2.2.2.2. The names aren't important as long
as you can tell the difference. Then explicitly
configure endpoint termination.com
<http://termination.com>'s "transport" parameter to
"transport-1.1.1.1" and pstn.com <http://pstn.com>'s
"transport" parameter to "transport-2.2.2.2". In
your dialplan, you can see which endpoint the call
came in on, and route it out the same endpoint.
If both providers are available from both interfaces,
you can create 2 endpoint for each provider:
termination.com-1.1.1.1, pstn.com-1.1.1.1,
termination.com-2.2.2.2 and pstn.com-2.2.2.2; Then
configure each with the same transports as above.
Thanks in advance for your help,
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
--
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Check us out at www.sangoma.com
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_____________________________________________________________________
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Check out the new Asterisk community forum at:
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David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
Check out the new Asterisk community forum at:
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New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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To UNSUBSCRIBE or update options visit:
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George Joseph
Asterisk Software Developer
direct/fax +1 256 428 6012
Check us out at www.sangoma.com <http://www.sangoma.com/> and
www.asterisk.org <http://www.asterisk.org>
image.png
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
Check out the new Asterisk community forum at:
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To UNSUBSCRIBE or update options visit:
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--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
--
*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO
Phone: *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email: *[email protected]* <mailto:[email protected]>
Website: *https://www.stratustalk.com*
Address: *One Freedom Square
13th Floor
Reston, VA 20190*
<https://www.facebook.com/jeff.lacoursiere>
<https://linkedin.com/in/jeff-lacoursiere-884361>
<https://www.twitter.com/stratustalk>
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