I’ve got a new asterisk + freepbx install. The installation went fine but I ran into a problem. I created the sip trunk, the sip show registry shows that the trunk is registered.
But when I try to call out via a sip phone - xtenlite - it gives me the following error message:
[Feb 18 21:33:22] WARNING app_macro.c: Context ‘macro-dialout-trunk-predial-hook’ for macro ‘dialout-trunk-predial-hook’ lacks ‘s’ extension, priority 1
[Feb 18 21:33:22] WARNING chan_sip.c: No such host: trunk1
[Feb 18 21:33:22] WARNING app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[Feb 18 21:35:35] NOTICE chan_sip.c: – Re-registration for [email protected]
The asterisk version is 1.4.18 plus a patch from bugs.digium.com because the Cirpack sip gateway refuses to work without it. (It was the fault of asterisk negating the “Interval too brief” message).
asterisk-addons 1.4.5 also installed from source.
FreePBX version is 188.8.131.52 but the problem was the same with 2.4.0.
Server version Ubuntu 7.10, server install.
Any ideas? Is it a bug of asterisk or freepbx?