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inmediatly
2008 Jan 26
1
Loosing IAX/SIP user's registration with asterisk as no-root
...othing in the console, absolutely no
messages, only when another friend trys to dial an extension I can see this
on messages logs
[Jan 26 10:35:46] WARNING[23015] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
And in the asterisk console the call is inmidiatly redirected to user's
voicemail
When this happens, I issue a reload command and everything gets back to
normal.
This is wired, before I was using the same version installed as root, and
this behaviour wasn't present.
Any adias?
Thanks in advace!