Cannot set up Asterisk for my SIP provider

Asterisk Asterisk SIP
22

I understand, thank you.
They are saying that they are not getting any 200 OK from me.
I also tried a different approach, I set up in the inbound route not an extension, but an announcement and hang up after.
The did not work neither.

23

Do a packet capture and verify that the packet is going out and such. If you see the packet leaving the machine, then it’s something upstream or a firewall.

24

Hello!
I made a tcpdump on port 5060 and this is what I get. My PBX tries to send the 200 OK 10 times, without success.
If needed, I can make a capture with a higher verbosity, but for a start I think this ok.

06:24:09.019163 IP sbc1.iristel.net.sip > freepbx.sangoma.local.sip: SIP: INVITE sip:our_phone_number@192.168.0.18 SIP/2.0
06:24:09.019943 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 100 Trying
06:24:09.138396 IP 192.168.0.11.sip > freepbx.sangoma.local.sip: SIP: REGISTER sip:192.168.0.18 SIP/2.0
06:24:09.139709 IP freepbx.sangoma.local.sip > 192.168.0.11.sip: SIP: SIP/2.0 401 Unauthorized
06:24:09.165348 IP 192.168.0.11.sip > freepbx.sangoma.local.sip: SIP: REGISTER sip:192.168.0.18 SIP/2.0
06:24:09.174904 IP freepbx.sangoma.local.sip > 192.168.0.11.sip: SIP: OPTIONS sip:200@192.168.0.11:5060 SIP/2.0
06:24:09.174941 IP freepbx.sangoma.local.sip > 192.168.0.11.sip: SIP: SIP/2.0 200 OK
06:24:09.206606 IP 192.168.0.11.sip > freepbx.sangoma.local.sip: SIP: SIP/2.0 200 OK
06:24:09.248997 IP freepbx.sangoma.local.sip > 192.168.0.6.sip: SIP: INVITE sip:777@192.168.0.6:5060;transport=udp;user=phone SIP/2.0
06:24:09.249092 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 180 Ringing
06:24:09.291655 IP 192.168.0.6.sip > freepbx.sangoma.local.sip: SIP: SIP/2.0 100 Trying
06:24:09.306003 IP 192.168.0.6.sip > freepbx.sangoma.local.sip: SIP: SIP/2.0 180 Ringing
06:24:09.306436 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 180 Ringing
06:24:16.319624 IP freepbx.sangoma.local.sip > 192.168.0.6.sip: SIP: OPTIONS sip:777@192.168.0.6:5060;transport=udp;user=phone SIP/2.0
06:24:16.349336 IP 192.168.0.6.sip > freepbx.sangoma.local.sip: SIP: SIP/2.0 486 Busy
06:24:19.187526 IP 192.168.0.6.sip > freepbx.sangoma.local.sip: SIP: SIP/2.0 200 OK
06:24:19.187793 IP freepbx.sangoma.local.sip > 192.168.0.6.sip: SIP: ACK sip:777@192.168.0.6:5060;transport=udp;user=phone SIP/2.0
06:24:19.188125 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:19.687732 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:20.688759 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:22.688451 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:26.687934 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:30.688459 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:34.687733 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:38.688449 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:42.688099 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:46.687914 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:50.234089 IP freepbx.sangoma.local.sip > 192.168.0.6.sip: SIP: BYE sip:777@192.168.0.6:5060;transport=udp;user=phone SIP/2.0
06:24:50.268959 IP 192.168.0.6.sip > freepbx.sangoma.local.sip: SIP: SIP/2.0 200 OK
06:24:50.688452 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 200 OK
06:24:55.502701 IP sbc1.iristel.net.sip > freepbx.sangoma.local.sip: SIP: CANCEL sip:our_phone_number@192.168.0.18 SIP/2.0
06:24:55.502888 IP freepbx.sangoma.local.sip > sbc1.iristel.net.sip: SIP: SIP/2.0 481 Call leg/transaction does not exist

25

The fault lies outside the Asterisk box.

26

Great news!
Although i checked my firewall and NAT rules to disable SIP translation, somehow i missed a global setting.

SIP%20transformations
SIP transformations.jpg742×542 64 KB

SIP Transformations was enabled and set to “Use global control”, that’s why my rules did not count.
After disabling this option, the calls are working normally. I can receive a call, pick up, have normal sound, and i can also initiate calls.
Thank you for your ideas and making it clear that the problem was not in the Asterisk.
I still have some issues with other features, but i think i will ask my questions in a different topic, because they are not related.
This topic can be marked resolved.