I got shi strange problem, if I type in cli: sip show peers this is what I get:
Name/username Host Dyn Nat ACL Port Status
trunk_1/0707736503 220.127.116.11 5060 Unmonitored
6001/6001 18.104.22.168 D N 54700 Unmonitored
6000/6000 22.214.171.124 D N 2048 Unmonitored
only the trunk is using port 5060
so when I try to call extension 6000 from ext 6001 no answer at all.
I read everything I could but I didn’t find the answer anywhere. I think I’m having NAT problems.
I have opened in our firewall this ports:
TCP 8088 (for the asterisk’s GUI)
UDP 13456:16482 (rtp ports)
In my asterisk GUI I have this fields:
Local Network Address:
NAT mode: yes
Allow RTP Reinvite
How do I have to fill them?
My dedicated server is remote and has his own IP.
Can you call voicemail from either phone? That is a simple test that will let you verify you can process calls and establish a voice channel to the phone while reducing complexity which helps a lot in debugging.
You also will need to turn off reinvite. The phones are not able to connect to each other since they are behind a NAT router and will need to use the Asterisk server to bridge the two phones.
Hi Phol and thank you for the answer,
I’ve solved everything now.
But I still got some problems, maybe you can help me:
1 - if I put more than one sip trunk from the same provider when calling a
number always rings the same extension. Maybe I should open more than just
5060 sip port?
2 - Voicemail is configured for extension 999 but it doesn’t works, this is
what it says: Executing [s-CONGESTION at macro-trunkdial:1]
NoOp(“SIP/102-0854e380”, “”) in new stack
== Auto fallthrough, channel ‘SIP/102-0854e380’ status is 'CONGESTION’
In my SNOM360 phone when I hit the mail key it doesn’t work but no message
3 - should I put IAX to yes in all the users?