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VoIP DotFuscate 11mo ago 100%

Help with Kamailio dispatcher

Hi, so my company decided to make a proxy for their internal and external phone call. so every call goes through the server (I'm using kamailio for it). then if the number called was internal, it send calls them. now for the external call, we had several SIP providers. in the past we use microsip to do external call with all of them. until the management are getting some problems like different dashboard for each providers, and too many calls to one provider while others are not being used that much. so here i am making a round robin for it. and would love any kind of help or pointers to make it.

my questions are :
- is there a way to ping the sip provider?, i saw a ping using OPTION but it doesn't seems to be reliable since most people would drop OPTION instead of replying them. i was thinking of console ping but not sure how to connect it to dispatcher database, how do i weight and prioritize them. along with when their server is on, but their asterisk are having problem.

- each providers had their own username and password, which they called an extensions. i think some might have a number prefix or suffix that needed to be added to phone number before i could use them. now how do i connect to it?, i tried UAC module but still unsure if it works

- when i called using JSSIP, i am getting sip uri like `123456789@sip.localhost.com` how do i replace the `sip@localhost.com` using uri that was used in the dispatcher that probably connected to UAC?.

my config the cfg is heavily adopting ChatGPT, i really had no idea what I'm doing.
the plan was to receive phone call only from TCP / websocket (we making our own softphone), then forward external phones to the real SIP provider. while rejecting UDP call. but it seems i need UDP port to send data to heplify / homer.

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