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 <dy...@regtelco.com
<mailto:dy...@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information and outdated sources, Once I have that I will have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!
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
