Hello, using Asterisk 1.4.28, I encountered a problem with SIP RTP port allocation.
I found some entries in mailinglist and bugtracker regarding this issue, but only old ones. My rtp.conf has [general] rtpstart=30000 rtpend=30100 so 100 ports available. I know that up to 4 ports per channel can be used and so up to 25 channels are possible. But even earlier I often get the error about "No RTP ports remaining". I had a look at netstat -nuap and it shows that a lot of ports are still assigned, even if there is no channel in use. But "sip show channels" show a lot of (unused) entries with no codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS. Why aren't RTP ports released when not in use? Or is there a possibility to configure this behaviour? Thanks, Armin -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
