Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the VoIP-Provider and do SIP TLS-> SIP UDP and SRTP -> RTP translation (via *1.6.2 and the SRTP patch) Kristijan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091102/5bd4aa39/attachment.htm
Kristijan Vrban wrote:> Hello, short question: is there a possibility to use asterisk as an > outbound proxy? iam open for any suggestions, use asterisk trunk, dirty > patches, ugly workarounds, everything. > > What is want to build is: > > SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP > -> VoIP-Provider > > So Asterisk should just forward any incoming SIP messages (INVITE, > REGISTER) to the VoIP-Provider and do SIP TLS-> SIP UDP and SRTP -> RTP > translation (via *1.6.2 and the SRTP patch)It is highly unlikely that you'll be able to get Asterisk configured in a transparent enough fashion to appear as a proxy in this scenario. You'd be far better off to use an actual proxy, if that's the functionality you need. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpfleming at digium.com Check us out at www.digium.com & www.asterisk.org
"Proxy" is not the correct term to describe this scenario, but yes, it is possible in principle. Kristijan Vrban wrote:> Hello, short question: is there a possibility to use asterisk as an > outbound proxy? iam open for any suggestions, use asterisk trunk, dirty > patches, ugly workarounds, everything. > > What is want to build is: > > SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP > -> VoIP-Provider > > So Asterisk should just forward any incoming SIP messages (INVITE, > REGISTER) to the VoIP-Provider and do SIP TLS-> SIP UDP and SRTP -> RTP > translation (via *1.6.2 and the SRTP patch) > > Kristijan > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671