Some legacy devices and trunks just need to run on legacy CHAN_SIP resource. This resource is running and configured on Asterisk “module show”
If you check the configuration files, settings are configured too. The Web GUI show the configuration too:
But you can only create extensions and trunks on pure asterisk files, not the GUI. The GUI doesn’t have selectable resource “chan_sip” or “legacy” options.
Please search the forum. There will be no chan_sip anymore.
Use PJSIP. Everything should work as well.
Only other option: import sip devices via backup.
You will not be able to create new chan_sip devices.
Thanks for the information. We have some issues using PJSIP for trunks because if the remote inbound registrar send another register instead of using update method asterisk will reject that register because of an active contact (aor). But I can see on the GUI an option for max_contacts:
So this should be the solution to that issue.
