Trying to get incoming calls to work on 10.04--even with a fairly good understanding of asterisk and the now missing freepbx--has been frustrating. Here is one clue if you are getting the message "this number is not in service, please check the number and dial again" when you call into your lmce phone:
Apparently the number detection is not yet working--or I haven't found a way to turn it on yet--and so asterisk thinks it is an anonymous call. In /etc/extensions.conf the second line under globals is set to no
ALLOW_SIP_ANON = no
this kicks any phone call into internal voice message that sounds exactly like a phone provider message informing the dialer that the number is not in service. If you change this to
ALLOW_SIP_ANON = yes
and the do a core restart gracefully inside of asterisk, external phone calls will now be allowed to go in to the system
Of course you are still not "home" yet as the system then tells you that pluto default voice mail does not exist and then just hangs not knowing what to do:
-- Launched AGI Script /usr/share/asterisk/agi-bin/lmce-callersforme.agi
-- AGI Script Executing Application: (NoOp) Options: (Finding if unknown is a caller for somebody)
-- <SIP/126.96.36.199-00000000>AGI Script lmce-callersforme.agi completed, returning 0
-- Executing [s@voice-menu-lmce-custom:4] Background("SIP/188.8.131.52-00000000", "pluto/pluto-default-voicemenu")
[Sep 16 09:08:59] WARNING: file.c:663 ast_openstream_full: File pluto/pluto-default-voicemenu does not exist in any format
[Sep 16 09:08:59] WARNING: file.c:958 ast_streamfile: Unable to open pluto/pluto-default-voicemenu (format 0x8 (alaw)): Permission denied
[Sep 16 09:08:59] WARNING: pbx.c:9781 pbx_builtin_background: ast_streamfile failed on SIP/184.108.40.206-00000000 for pluto/pluto-default-voicemenu
-- Executing [s@voice-menu-lmce-custom:5] Set("SIP/220.127.116.11-00000000", "TIMEOUT(digit)=10")
-- Digit timeout set to 10.000
-- Executing [s@voice-menu-lmce-custom:6] Set("SIP/18.104.22.168-00000000", "TIMEOUT(response)=20")
-- Response timeout set to 20.000
[Sep 16 09:09:18] WARNING: pbx.c:5203 __ast_pbx_run: Don't know what to do with 'SIP/22.214.171.124-00000000'
I'll post here if I find any way around this, but in the meantime, if people call you on your cell and let you know your phone has been taken out of service you will know why.
One more hint: when setting up your phone lines to your sip provider in the "prefix" field you should enter the number you want to dial out before the phone number, as in 91973-xxx-xxxx in the US. This seems like a great feature as it allows you to have several providers and dial automatically to that provider based on where the call is going and which prefix you use. I found this in the forum, but it was not intuitively obvious in the web admin.
Finally, do you think we could get a FAQ going just about how 10.04 asterisk is supposed to work and the gotchas? Searching the forum and/or wiki gives you all the flavors of asterisk used to date and is highly inefficient. Since freepbx is no longer implemented, searching the web isn't useful as it assumes you are using freepbx as part of the solution.
Just my 2 cents.