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 I found a few tutorials but found a lot of conflicting information and outdated sources, Once I have that I will have enough to work on to do what I want...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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