Ports being used are out of range set in rtp.conf

Any help to point me in the right direction would be greatly appreciated.

Running Asterisk 1.8 and FreePBX 2.8.1 The pbx runs on a server directly connected to the internet, behind firewall, accessed only from LAN.

When a place a call comes in from outside, if the call is routed directly to the sip device there is no problem. If the call passes through IVR or a greeting is played from Follow Me (7th setting down called Announcement), then there is only outgoing audio and the calls drops due to RTP time out. The log shows RTP errors. I have the RTP ports open on the firewall.

Thanks, Eric

I have the port range for rtp set as 10001-20000 in rtp.conf, yet from the log (see below), it appears it is using ports above that range. I’ve seen other times it was below that range. My firewall is blocking anything I haven’t permitted, and I have only permitted the 10001-20000 ports.

Is there anywhere else I can configure the rtp port range?

res_rtp_asterisk.c: RTCP SR transmission error to, rtcp halted Operation not permitted

Can anyone help?

System is working great. Only problem is if an audio clip is played before I answer the phone, then I get one way audio and a dropped call after 30 seconds…

As one example, if the inbound route goes to an extension, the call rings for 30 seconds, if no answer, then it goes to voicemail. If I answer, then everything works fine and there normal two-way conversation. If I add an audio clip to be played, before it begins ringing, then I get only outgoing audio (I don’t hear the other party but they hear me), and then the call drops after 30 seconds. There are RTP errors in the log.

[May 11 12:53:41] ERROR[6678] res_rtp_asterisk.c: RTCP SR transmission error to, rtcp halted Operation not permitted

Same happens if I use IVR.

For anyone else that gets similar issues to what I had, I wanted to let you know how I fixed it.

In freepbx, go to Tools and Asterisk SIP Settings. In Media and RTP Serttings, I changed Reinvite Behavior to “Update.”

The RTCP errors appear to have been unrelated to the initial issue of one way audio (happening only when greeting played before transfer to extension).

It may be a NAT issue.
I don’t know much about Asterix but I do know how SIP works and the problems it has through NAT. In this case a STUN server is needed but STUN also has issues through certain NATs. The following article explains in great detail why this issue is caused. It may be slightly different for you though as the article says there is one way audio from the beginning. If it is then this will be the cause.
If this works or is the cause please leave feedback here and on the site as it is a friend of mine and the site is pretty new.