No communication before hold

Dear freepbx users,

I’m stuck at some point with my freepbx configuration of Asterisk on a Debian machine. First, I should probably list what I have and tell where I want to go.

  • Debian Linux 2.6.26
  • FreePBX
  • Asterisk 1.4.36
  • SIP Client X-Lite 4 on a Windows machine

I installed Asterisk and FreePBX without too many problems and could configure my SIP trunk (sipgate) well enough to make inbound and outbound calls work.

There is only one thing which I couldn’t resolve. Whenever I place a call (inbound and outbound calls have the same issue), the person at the other end (no matter if internal or external calls) cannot hear me, neither can I hear him - no communication possible.

When I place this call on hold, the person at the other end hears the MOH which has been set up… when I am put on hold, I can hear the MOH as well. When I pick this call up again from asterisk, communication is working fine… I can hear the person at the other end and he can hear me.

I tested what happens when I use an IVR to route calls to internal extensions based on the caller’s choice and after pressing 1 for extension xyz for example, the called extension rings and when the call is picked up, communication is working fine without having to put the person on hold first.

Anybody has an idea what went wrong during installation or configuration? Please excuse my english… i’m not a native speaker.

Thank you in advance and best regards!

How about UDP in the range of 10000-20000?

This is an old wiki but the router rules still apply. Take a look at this link:

Google is your friend.

Thank you first of all! I’m working with a cisco 2621 xm at home and have configured NAT for the following ports being forwarded to the asterisk server:

ip nat inside source static tcp 5038 interface FastEthernet0/0 5038
ip nat inside source static udp 5060 interface FastEthernet0/0 5060
ip nat inside source static udp 4569 interface FastEthernet0/0 4569

Would I need to add anything else to the NAT config of the router (adding NAT rules for each of the asterisk’s sip clients?)?

Thank you in advance and best regards

This sounds like pour NAT transversal through your router. Try “NAT=YES” in your extensions and trunks.