No sound recieved, call disconected, lack of RTP activity

General Help
1

Hello,
When i receive a call from outside on one of our extensions, I cannot hear anything but the other person can hear me. After 32 seconds the call is disconnected. In the log files i can read this:
NOTICE[2954] chan_sip.c: Disconnecting call ‘SIP/aaa-00000003’ for lack of RTP activity in 31 seconds .
However, if i receive the call when conected to the SIP trunk directly using YATE, this problem doesn’t occur.

2

Hello,
I am using FreePBX 2.11.0.23.
When i receive or make a call from outside on one of our extensions, I cannot hear anything but the other person can hear me. After 32 seconds the call is disconnected. In the log files i can read this:
NOTICE[2954] chan_sip.c: Disconnecting call ‘SIP/aaa-00000003’ for lack of RTP activity in 31 seconds .
However, if i receive the call when connected to the SIP trunk directly using YATE, this problem doesn’t occur.

3

Please read - http://www.freepbx.org/forum/general-help/read-before-posting

Then update your question.

You may also want to check out the wiki on NAT

4

You must not have read the “read before posting”

You did not include your Asterisk version or your NAT settings.

2nd time, read wiki on SIP NAT. Correctly set your NAT settings and your issue will be solved.

5

Hello,
I am using FreePBX 2.11.0.23, Asterisk (Ver. 11.8.0).
My NAT Settinngs are:

NAT:YES
IP Configuration: Static IP
External IP: our external IP (195.198.xx.xx)
Local Networks: 192.168.1.0/255.255.255.0

When i receive or make a call from outside on one of our extensions, I cannot hear anything but the other person can hear me. After 32 seconds the call is disconnected. In the log files i can read this:
NOTICE[2954] chan_sip.c: Disconnecting call ‘SIP/aaa-00000003’ for lack of RTP activity in 31 seconds .
However, if i receive the call when connected to the SIP trunk directly using YATE, this problem doesn’t occur.
I am so sorry for my previous incomplete posts. I tried to change my NAT settings, but I dont know what to change. I had spend 1 day reading about this issue, but no success so far.

6

There is a chunk specific to RTP. Which is where your problem lives.

7

Do you have any kind of SIP helper or ALG enabled on your router. Sounds like rep not getting through firewall.

Yate uses a STUN server to solve this issue. Asterisk is not as foegiving.