I'm trying to configure a spa3102 for outbound calls only. I know people
pull their hair out over these devices, but I wanted to give it a shot.
My only gateways I've worked with so far are sipxbridge and an
audiocodes configred from within sipx, so I haven't really done too much
manual FXO configuration.
I think I may be missing something on the sipx end, because I don't
think the call is ever making it to the spa3102. This is a new setup and
has no other gateways. I added the spa3102 as an unmanaged gateway. I
enabled all the dialing plans and added the gateway. I'm using a polycom
550, Sipx 4.0.4, bootrom 4.2.1, firmware 3.1.3C split. I would show a
siptrace, but the merged file doesn't really have anything in it. The
sipx server is at 10.81.1.5. The spa3102 is at 10.81.1.6. I tried
setting the gateway in sipx to UDP manually (that is what the spa3102
defaults to) and specifying port 5060, but that didn't seem to change
anything. There are only 2 logs created, so I attached those. Is there
something simple I'm missing? I read through this,
http://sipx-wiki.calivia.com/index.php/HowTo_configure_sipX_to_work_with_FXO/FXS_SIP_Gateways
but I don't see anything that sticks out at me. The only thig I thought
I might need to do is something in authrules.xml, but I'm still not sure
since the text around it refers to FXS and this is FXO. I sort of guess
there has to be some some sort of authorization for the spa3102 to know
the sipx call can be sent outbound, but I don't know where to do this.
Sorry if I'm missing something obvious here. I think the fact that I got
an audiocodes 8 port working inbound and outbound with no questions (and
clearly not much knowledge on the subject) is a testament to how well
sipx is able to configure it!
Thanks,
Matthew
"2010-02-18T16:27:06.266868Z":8:SIP:NOTICE:pbx.ma178.sipx.voip:SipRedirectServer-13:41419940:SipRegistrar:"ContactList::add():
[140-FALLBACK] SipRedirectorFallback added contact for
'sip:[email protected];user=phone':\n
'<sip:[email protected]:5060;user=phone;transport=udp?expires=60>;q=0.9'
(contact index 0)"
"2010-02-18T16:27:06.267309Z":9:SIP:NOTICE:pbx.ma178.sipx.voip:SipRedirectServer-13:41419940:SipRegistrar:"ContactList::set():
[999-AUTHROUTER] SipRedirectorAuthRouter modified contact index 0 for
'sip:[email protected];user=phone':\n was:
'<sip:[email protected]:5060;user=phone;transport=udp?expires=60>;q=0.9'\n
now is:
'<sip:[email protected]:5060;user=phone;transport=udp?expires=60&ROUTE=%3Csip%3A10.81.1.5%3A5060%3Blr%3E>;q=0.9'"
"2010-02-18T16:27:06.279507Z":10:SIP:WARNING:pbx.ma178.sipx.voip:SipRouter-11:41C5E940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"
"2010-02-18T16:27:06.363927Z":11:SIP:ERR:pbx.ma178.sipx.voip:SipUserAgent-2:41113940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"
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