[OpenSIPS-Users] OpenSIPS and Asterisk on same system

[Bogdan-Andrei Iancu]

Check
https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/
https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 5/24/23 1:00 AM, Dylan Cruz wrote:
> Still looking for possibly a template/example code on this.
>
> I am setting a bounty of $150 for anyone willing to help.
>
> You can reach out to me via E-Mail or phone at 407-999-0000
>
> Thanks!
>
> On Mon, Mar 13, 2023 at 8:26 PM Dylan Cruz <dylan at regtelco.com 
> <mailto:dylan at regtelco.com>> wrote:
>
>     I'd love a sample OpenSIPS Config that would let me accomplish
>     using it as a transparent proxy to Asterisk running on the same
>     system. I found a few tutorials but found a lot of conflicting
>     information and outdated sources, Once I have that I will have
>     enough to work on to do what I want... Basically I would like
>     OpenSIPS to sit between the outside world and Asterisk, Incoming &
>     Outgoing would both transparently be proxied through it. OpenSIPS
>     would be running on port 5060 & Asterisk would be running on port
>     5090, So for example to register to a SIP Trunk from a VoIP
>     provider my Asterisk sip.conf would look like this: (I know
>     chan_sip is deprecated...)
>     *[general]*
>     *nat=no*
>     *bindport=5090*
>     *outboundproxy=127.0.0.1:5060 <http://127.0.0.1:5060>; Route
>     everything through OpenSIPS*
>     *tos_sip=cs3*
>     *tos_audio=ef*
>     *trustrpid=yes*
>     *canreinvite=yes*
>     *directrtpsetup=yes*
>     *allowguest=no*
>     *allowoverlap=yes*
>     *srvlookup=yes*
>     *disallow=all*
>     *allow=ulaw*
>     *[inbound-pstn]*
>     *type=peer*
>     *host=191.122.31.32*
>     *insecure=invite,port*
>     *qualify=yes*
>     *context=from-inbound*
>     *[outbound-pstn]*
>     *type=peer*
>     *host=191.122.31.33*
>     *insecure=invite,port*
>     *qualify=yes*
>     I would then be able to talk to both of those trunks from Asterisk
>     and have inbound & outbound calls working all the way through to
>     the VoIP provider.
>     My purpose for wanting to do this is I want to play around with
>     the SIP-I module in OpenSIPS to interwork ISUP IAM fields by
>     breaking them out into SIP Headers that I can then manipulate
>     easily in Asterisk.
>
>     Full disclosure: I am a complete OpenSIPS noob! This would be my
>     first OpenSIPS project, I am very impressed with its capabilities
>     and by having a little sample config it would allow me to
>     experiment and start my journey of getting my feet wet with it!
>
>     Thanks in advance!
>
>
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