The call is established but there is no dialing tone.

this is the log, WHATS MISSING?


-- Executing [s@macro-dialout-trunk:13] Set("SIP/103-08d7cc20", "OUTNUM=0682002085") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/103-08d7cc20", "custom=SIP/Abissnet") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/103-08d7cc20", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/103-08d7cc20", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/103-08d7cc20", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/103-08d7cc20", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/103-08d7cc20", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/103-08d7cc20", "SIP/Abissnet/0682002085|300|") in new stack
-- Called Abissnet/0682002085
-- SIP/Abissnet-08dc06c8 is making progress passing it to SIP/103-08d7cc20

Welcome to Voipuser forums,


If you were trying to make an outgoing call from Voipuser then the problem may be that outgoing calls are shut off at present as the servers are upgraded. It is expected to be running again very soon.

Meanwhile, when you post it would be useful to give a little more information about your post. Is this an extract from an Asterisk or similar log? What is Abissnet?

This is the exact log from the asterisk server. Abissnet is the name of my Sip server.

The server is not being upgraded.

In that case the first thing to check is that you do not have a port problem with your router. Make sure that all of the required ports are open and available to the Asterisk server.

The configuration from the firewall side is like this:

from inside to outside permit any
from outside to inside port range forwarding to asterisk server
"port range 5000-32767"

Is there any other port for SIP+RTP?