[Asterisk-Users] false busy using sipura spa-3000 with asterisk on solaris

[Logan O'Sullivan Bruns]

I'm new to asterisk and already a fan. Please forgive me if my
questions are covered by some FAQ and thanks in advance for any
pointers anyone can give me.

The basic problem that I'm having is that sometimes outgoing calls
result in a busy signal when the outgoing line is free. I'm thinking
that the channel is timing out or something but haven't figured out
how to debug or gather additional detail yet. (I've only been using
asterisk for a couple of days. So, again, please pardon me if these
are stupid questions.)

My home configuration is as follows. I have asterisk 1.0RC1 running on
a sparc (solaris 10). I have an Sipura SPA 3000 on the same
network. It has an one SIP accessible FXS port (5060), one SIP
accessible FXO port (5061) and ethernet port. I have my land line is
attached to the FXO port. All my home phones are attached to the FXS
port. I want to set it up to route long distance through some voip
provider like vonage but right now I set up to route calls from the
land line or internet to my home phones or asterisk hosted voice
mail. Incoming calls work perfectly. Voice mail works perfectly (with
the minor exception that volume on gsm attachments is lower than I'd
like.) Outgoing calls works sometimes and other times it just goes
straight to a busy signal.

Despite the fact that outgoing works some of the time I'm wondering if
I have the configuration wrong. Here is the outbound fragment from
extensions.conf.

[outbound]
; Press * to reach voice mail
exten => *,1,VoiceMailMain(${PHONES1VM})
exten => *,2,Hangup

; local calls
exten => _NXXXXXX,1,Playback(transfer)		; "Please hold while..." 
exten => _NXXXXXX,2,Dial(SIP/${EXTEN:0}@phone.gedanken.org:5061,20)
exten => _NXXXXXX,3,Congestion

; long distance
exten => _1NXXNXXXXXX,1,Playback(transfer)		; "Please hold while..." 
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN:0}@phone.gedanken.org:5061},20)
exten => _1NXXNXXXXXX,3,Congestion

I'm not sure if this relevant but I have wondered if the sip incoming
configuration for the FXO port might be interfering or conflicting in
some way. Probably not but here is the sip.conf fragment anyway.

[5785]
type=friend
context=inbound
host=dynamic
secret=secret
defaultip=10.0.1.190
port=5061
nat=no

I haven't seen any errors of messages of note in the /var/log/asterisk
logs. I'm not positive that asterisk is generating the busy signal. I
think so but I guess it could be possible that the spa3000 is. I know
the remote end isn't busy because I'm testing by calling my cell
phone.

I know Solaris isn't a well tested platform and I did have to make
some minor code changes to get to compile on my sun box. However, the
fact that almost everything works so perfectly makes me think that it
is a configuration problem not a porting problem. 

Again, any advice, things to try and such would be greatly
appreciated.

Thanks,
  logan