BroadVoice-Busy Signal for Incoming Calls


Incoming calls get a busy signal. Call log shows a succesful connection to the Asterisk Server, caller id, etc, but shows the call “disposition” as “no anwser”. I can call back and forth between the Sip phones and I can also dial out with SIP phones. Below is a Log from the debug.

Sep 24 21:09:06 VERBOSE[1274]: – Executing Wait(“SIP/”, “1”) in new stack
Sep 24 21:09:07 VERBOSE[1274]: – Executing Goto(“SIP/”, “from pstn|s|1”) in new stack
Sep 24 21:09:07 VERBOSE[1274]: – Goto (from pstn,s,1)
Sep 24 21:09:07 WARNING[1274]: Channel ‘SIP/’ sent into invalid extension ‘s’ in context ‘from pstn’, but no invalid handler

Thanks for any help


I have set up two asterisk@home bxes and had the same problem with broad voice. Both times, it was due to minor misconfiguratona in the incoming call routing, hence calls could not get through to the box and generated a busy signal.

Try checking your inbound routes again, and also the sip.conf files for the following

[quote];context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
context = from-pstn[/quote]
which solved the problem for me on the box I just completed

Good luck


I concur that this is a configuration problem, had the same thing
happen to me not just with Broadvoice but with other providers.

Here is how I try to eliminate this sort of thing now:

Make sure that in sip.conf, in the definition for the incoming BV trunk
(or in the BV “friend” config, if you only have one combined config)
you specify a clearly identifyable context, i.e. “from-broadvoice”.

If you have other SIP providers, do the same thing for them (i.e. I have
a context “from-sipgate”, and one “from-babble”, etc.)

Then, in your extensions.conf, start the context with
[from-broadvoice], and then match, accept and route calls coming
from BV.

If you have multiple incoming SIP lines which eventually all need to
be treated the same, i.e. go to the same context, it is still useful to have
a separate context for each line/provider, but all it needs to contain is a
"Goto sip-incoming,s,1". Then all the actual work takes place in
[sip-incoming], but if you DO need to do anything specific to a
particular line or provider, you have a place to do it., and it also makes
it a tad easier to trace in the debug output from Asterisk.