Ok, but i try for pjsip trunk, don’t have Sip quality option, only qualify frequency, i put to 0 but still not working.
I will retry in sip with sip qualify no …
I also try with sip and qualify=no, still same problem…
OK, with registration disabled and CHAN SIP entry with qualify=no, retry the call but grab both Asterisk and Freeswitch CLI logs and post them here so I can see what’s going on.
Ok i do and same problem .
Here asterisk log and freeswitch log
https://pastebin.freeswitch.org/view/7f474aef#iol8GqoidBii4LfTWWappptEClwZyVE7
Thanks
There’s no asterisk log there @johnx102, did you forget to paste the link?
Anyways, it’s an authentication issue you’re facing, something isn’t right with your peer details in FreePBX.
Remove all trunk setups in FreePBX linking to your ASTPP server and enter the following chan_sip trunk:
host=
username=
secret=
type=peer
context=from-ptsn
insecure=port,invite
nat=no
qualify=no
disallow=all
allow=alaw,ulaw
with no registration and all inbound details blank since they are covered by the above peer settings populating the host, username and secret lines with your astpp server IP, sip device username and sip device password accordingly. For the purposes of this test, ensure the password on the sip device contains nothing other than alphanumeric characters incase there’s an illegal character there somewhere that asterisk can’t handle.
Don’t add any other options to the peer details than those listed in the example above.
Once you’ve set that up, ensure your outbound route is set to use the correct trunk entry and re-test.
Thanks for answer, i put the asterisk log link, but i regive you .
I will try what you tell and update you .
Thanks
Assuming the username and secret match exactly on both ASTPP and FreePBX, then it appears your ASTPP solution is not picking up the SIP credentials hence the authentication is failing.
I’m guessing you’ve made changes to the sofia profile since your port has been changed. Have you made any changes other than the port?
No change at all.
Default sofia config on default astpp install.
If i put same user on voip phone , its working.
I dont know why with freepbx only ip auth work…