[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound
Jeff LaCoursiere
jeff at stratustalk.com
Thu Oct 29 20:49:23 CDT 2020
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
> <dcunningham at voisonics.com <mailto:dcunningham at voisonics.com>> wrote:
>
> OK, thank you George.
>
>
> On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com
> <mailto:gjoseph at digium.com>> wrote:
>
>
>
> On Thu, Oct 22, 2020 at 4:13 PM David Cunningham
> <dcunningham at voisonics.com <mailto:dcunningham at 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, 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
> <gjoseph at digium.com <mailto:gjoseph at digium.com>> wrote:
>
>
>
> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham
> <dcunningham at voisonics.com
> <mailto:dcunningham at voisonics.com>> 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
> destination at termination.com
> <mailto:destination at termination.com> -> 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
> destination at pstn.com <mailto:destination at pstn.com>
> -> 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
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> 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:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> 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:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> 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:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> 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: *jeff at stratustalk.com* <mailto:jeff at stratustalk.com>
Website: *https://www.stratustalk.com*
Address: *One Freedom Square
13th Floor
Reston, VA 20190*
<https://www.facebook.com/jeff.lacoursiere>
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