Re: [OpenSIPS-Users] Call forking, branches, Record-routing

2022-04-01 Thread Karsten Wemheuer
Yes, I originally assumed that I could not use two TLS sockets on one interface. My last sentence in the last post showed that I understood it now. As you said, I now have port forwarding to another port and two separate sockets for public and private phones. socket = udp:10.0.2.3 socket = tls:10.

Re: [OpenSIPS-Users] OpenSIPS timers

2022-04-01 Thread Ovidiu Sas
Hello Bogdan, During my test, it was tm-utimer only. It was a typo on my side. I also see in the logs from time to time the other timers too, including tm-timer. What I noticed in my tests is that as soon as I increase the timer_partitions, the system is able to handle less cps until workers are

Re: [OpenSIPS-Users] Call forking, branches, Record-routing

2022-04-01 Thread Bogdan-Andrei Iancu
Oh, so you have traffic both from public and private network, right ? If so, you have 2 options: 1) use a single socket, without advertise and use the script advertise function depending on the source of the call - see set_advertised_address() [1] 2) use 2 sockets, one for public traffic, wi

Re: [OpenSIPS-Users] OpenSIPS timers

2022-04-01 Thread Bogdan-Andrei Iancu
Hi Ovidiu, Originally you mentioned tm-utimer, now tm-timerwhich one is ? As it is very important. When increasing the timer_partitions, what you mean by "instability" of the system? Yes, in the reactor, the UDP workers may handle timer jobs also beside the UDP traffic. While the timer

[OpenSIPS-Users] RTP Engine rpms for Centos 7

2022-04-01 Thread Alexey Kazantsev via Users
Hi list,   a small contribution to the community, not as a yum-repository yet, but still.   https://alexeyka.zantsev.com/rpm/   Often creating rpm packages of RTPEngine becomes a headache, so I decided to share.   Hope this will be useful!   --- BR, Alex

Re: [OpenSIPS-Users] Call forking, branches, Record-routing

2022-04-01 Thread Karsten Wemheuer
Hi Bogdan-Andrei, I tried with record_route and do some corrections. The part for calls from the PBX via the proxy to the phones now looks like this: route[TOPHONES] { record_route(); lookup("location"); t_on_branch("AST2PHONE"); if (!t_relay()) { sl_reply_error(); } } branch_rout