[asterisk-users] Asterisk 11.5 not honoring RTP port change in RE-INVITE

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the

[asterisk-users] Interconnect Radio units using asterisk

Hi all, This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had this goal, but currently is not in the new Asterisk 1.8 or higher. Thinking in

[asterisk-users] Getting Asterisk 11.5 to use TURN

I've configured TURN in rtp.conf in Asterisk 11.5. The credentials are correct because I can get Chrome to get relay candidates and attach them to the SDP, but Asterisk doesn't want to play ball. There's little documentation -- at least from what I can tell -- on getting TURN working in

Re: [asterisk-users] Interconnect Radio units using asterisk

On Aug 26, 2013, at 8:11 PM, Manolo Quijano manolo.quij...@gmail.com wrote: Hi all, This my first mail in the community. My name is Manolo. I'm new in Asterisk. My objective is to control some radio using Asterisk via web. In google I could see that the application app_rpt had this