Re: [OpenSIPS-Users] [Blog] Traffic balancing – load, weights, round robin ??

2017-06-30 Thread Mundkowsky, Robert
Yeah, saw that. Looks real good. Robert Mundkowsky -Original Message- From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, June 28, 2017 3:43 PM To: Mundkowsky, Robert ; OpenSIPS devel mailling list ; OpenSIPS users mailling list Cc: busin...@lists.opensips.org; n...

Re: [OpenSIPS-Users] Pending OpenSIPS minor releases: Last minute bug fixes!

2017-06-30 Thread Bogdan-Andrei Iancu
As next week we will release the OpenSIPS Control Panel version related to OpenSIPS 2.3 , we will also change the versioning policy for OpenSIPS Control Panel Why? we what to create a tight and direct relation between the version of OpenSIPS Control Panel and the corresponding version of OpenS

Re: [OpenSIPS-Users] SIP URI User Parameters

2017-06-30 Thread Ben Newlin
Bogdan, Sorry for the delayed response, I am having some trouble reproducing this in a local test environment. Currently it is only occurring in our live environment. I do have some clarifications and answers to your questions: · The npdi parameter is not present in $ru in the failure

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

2017-06-30 Thread Bogdan-Andrei Iancu
Good, there is some progress :). On the incoming calls, if the WS get's the call, we can park the part with the auth (it seems your opensips script is accepting calls from unknown sources...we can address this security hole later. So, if a call drop after 30 secs it usually means there is no

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

2017-06-30 Thread Alex Megalokonomos
I think I set up uac_registrant correctly. I can dial out from a ws client and the ws extension rings from outside calls. However: a) on incoming calls, when ws client accepts, there is no sound and the line is dropped after 30 secs or so b) on outgoing calls, when the called extension accepts t

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

2017-06-30 Thread Bogdan-Andrei Iancu
I checked the script you mentioned and it does not help you - it has only UDP (no WS), it is really basic and it does not handle any REGISTER stuff, which is the trickiest - see https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/ or https://blog.opensips.org/2016/12/20/mid-regis

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

2017-06-30 Thread Alex Megalokonomos
Hello Bogdan, First of all, thanks for your time. Unfortunately my SIP/OpensSIPS skills are what I've managed to learn in the last couple of days. I am a programmer but I've never had to work on SIP stuff before. Frankly to me, both solutions sound equally difficult since I have no idea where to

Re: [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

2017-06-30 Thread Bogdan-Andrei Iancu
Hi Alex, To make a kind of WS<>UDP gateway you need a complete rework of the script presented in the tutorial, as it is a completely different SIP scenario. Not sure what are your SIP/OpenSIPS skills. But, there is a simpler alternative . Instead of a GW, you can make OpenSIPS as a sub-serve

Re: [OpenSIPS-Users] Forking Non-INVITE Requests

2017-06-30 Thread Bogdan-Andrei Iancu
Hi Chad, I would say the t_replicate() is what you are looking for : http://www.opensips.org/html/docs/modules/2.3.x/tm.html#treplicate Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2017, Houston, US http://opensips.org/tr