Thanks Richard for the pointer.
I tried all day but no luck.
I'll have another go at it in a few days.
Jerry
On 9/5/2022 3:39 AM, Richard Edmands wrote:
I know this works with an Asterisk configuration, not sure how
Freeswitch is any different yet. Haven't needed to poke Freeswitch.
https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srtp-bridge.cfg
Strip out the fun bits where Fred is re-writing the contact header.
Then in the route relay, add the following:
record_route();
handle_ruri_alias();
add_contact_alias();
In some configs, onreply_route manage_reply needs a:
add_contact_alias();
Lazy, yes. Works for me?, yes.
On Mon, 5 Sept 2022 at 14:16, Jerry Kendall
<[email protected]> wrote:
Hi There,
It's been a few years since I joind the group and I have watched and learnt and
played but, I am stuck...
I have searched hi & low but I can't seem to find what I am looking for.
Is is do-able to have SIP Phones proxy through Kamailio/RTPengine and have the
connection relayed to a freeswitch system where all the work is done.
I need to have Kamailio act as proxy ONLY - no need to do AUTH, and I need
Kamilio to have the RTPEngine have the media but only as proxy.
I have FreeSWITCH doing all the IVR/VM/CONF/etc....
So:
1) The SIP Phone sends everything to the Kamailio system, the Kamailio system
will relay everything to the FreeSWITCH system (requests and replies)
2) The FreeSWITCH system will send everything to the Kamailio system which will
relay everything back to the SIP Phone (requests and replies)
3) The SIP phones are behind NAT
4) The Kamailio and FreeSWITCH systems are on the public internet with public
IP addresses
I have found a few sample kamailio.cfg files but, I can get REG to make it
through but no SUBCRIBE wont get NOTIFYs back, etc...
I seem to get parts but not all the way.
If this is doable, and I assume it is, is there a config file someone can share
with me? or point me to?
Jerry
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