I have no doubt it is a PEBKAC, I just don't where to look. I changed it
to 5061 (I now see that setting in the PSTN Line tab on the spa3102).
The logs look about the same to me. I don't see anything that even tells
me it is making it to the spa3102.
On 2/18/2010 10:54 AM, Eric Varsanyi wrote:
When I set mine up late last year the only issue I had making outbound calls
(that wasn't PEBKAC) was the thing didn't think there was a line attached and
returned something like 'resource not avaiable' to the invite. I had to change
the line voltage threshold down in the international settings box to fix this.
Ah, in the log I see you're using 5060, the FXO side by default is on 5061 (the
FXS is on 5060). LIkely that's your issue.
-Eric
On Feb 18, 2010, at 10:42 AM, [email protected] wrote:
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
<sipregistrar.log><sipXproxy.log>_______________________________________________
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"2010-02-18T16:59:02.920532Z":12:SIP:NOTICE:pbx.ma178.sipx.voip:SipRedirectServer-13:419A1940:SipRegistrar:"ContactList::add():
[140-FALLBACK] SipRedirectorFallback added contact for
'sip:[email protected];user=phone':\n
'<sip:[email protected]:5061;user=phone;transport=udp?expires=60>;q=0.9'
(contact index 0)"
"2010-02-18T16:59:02.920998Z":13:SIP:NOTICE:pbx.ma178.sipx.voip:SipRedirectServer-13:419A1940:SipRegistrar:"ContactList::set():
[999-AUTHROUTER] SipRedirectorAuthRouter modified contact index 0 for
'sip:[email protected];user=phone':\n was:
'<sip:[email protected]:5061;user=phone;transport=udp?expires=60>;q=0.9'\n
now is:
'<sip:[email protected]:5061;user=phone;transport=udp?expires=60&ROUTE=%3Csip%3A10.81.1.5%3A5060%3Blr%3E>;q=0.9'"
"2010-02-18T16:59:07.771052Z":14:SIP:NOTICE:pbx.ma178.sipx.voip:SipRedirectServer-13:419A1940:SipRegistrar:"ContactList::add():
[140-FALLBACK] SipRedirectorFallback added contact for
'sip:[email protected];user=phone':\n
'<sip:[email protected]:5061;user=phone;transport=udp?expires=60>;q=0.9'
(contact index 0)"
"2010-02-18T16:59:07.771391Z":15:SIP:NOTICE:pbx.ma178.sipx.voip:SipRedirectServer-13:419A1940:SipRegistrar:"ContactList::set():
[999-AUTHROUTER] SipRedirectorAuthRouter modified contact index 0 for
'sip:[email protected];user=phone':\n was:
'<sip:[email protected]:5061;user=phone;transport=udp?expires=60>;q=0.9'\n
now is:
'<sip:[email protected]:5061;user=phone;transport=udp?expires=60&ROUTE=%3Csip%3A10.81.1.5%3A5060%3Blr%3E>;q=0.9'"
"2010-02-18T16:59:02.932591Z":14:SIP:WARNING:pbx.ma178.sipx.voip:SipRouter-11:417C1940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"
"2010-02-18T16:59:03.018182Z":15:SIP:ERR:pbx.ma178.sipx.voip:SipUserAgent-2:416C0940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"
"2010-02-18T16:59:07.783250Z":16:SIP:WARNING:pbx.ma178.sipx.voip:SipRouter-11:417C1940:SipXProxy:"SipUserAgent::send
INVITE request matches existing transaction"
"2010-02-18T16:59:07.869833Z":17:SIP:ERR:pbx.ma178.sipx.voip:SipUserAgent-2:416C0940:SipXProxy:"SipUserAgent::handleMessage
SIP message timeout expired with no matching transaction"
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