I am looking for someone to setup and configure Kamailio or Janus ( Preferably Janus) as a WebRTC to SIP Gateway in from of multiple Asterisk server. Janus / Kamailio should be able to load share to multiple Asterisk servers based on Caller Prefix.
...on ASTPP2 (freeswitch), there is an option for that in ASTPP. We suspect, that for interconnecting, freelancer needs to connect ASTPP2 to the DB of ASTPP1 and also create a sip profile of ASTPP2 on ASTPP1, to display the registered customers from ASTPP2 (but that is a hint only). The done job must be documented from the beginning, very detailed, how
We want to setup telegram to sip with this below link instructions - [đăng nhập để xem URL] Raspberry pi product link - [đăng nhập để xem URL] Is it possible for you to create this setup and make workable, if so let me know how much time you
...install to SIP trunks. - Main site has 2 CCM, 2 gateways with 2xPRI - 3 remote sites connected via MPLS have SRST routers and mixture of POTS lines Project requires: - Add Internet to each remote site (customer provisioning) - Add SIP trunks to each site to replace POTS - Provide failover scenarios (fault tolerance features enabled by SIP service provider)
We would like to integrate an third party software (SIP call) into our software, so that we can apply the SIP video call functions in our software. Our software is developed in Cordova and the third party is in Native so we need to convert their functions as cordova plugins. The third party software is Comelit intercome software, it can be found from