[asterisk-users] inbound SIP funnies

[Adrian Marsh]

Hi,

 

I've a few working asterisk servers, all seeing the same symptom, but
they are all based on the same configs.

 

A SIP inbound INVITE message is coming in to an extension  (not a peer)
eg  555 at ourserver.com

 

A tcpdump clearly shows the INVITE coming in, but asterisk seems to be
ignoring it (theres no reply outbound packet). All the source/dest IPs
and ports look good.

A "sip set debug trace ip <sourceip>"  is blank, showing nothing at all.

 

The sip.conf default context is incoming_pstn. The incoming_pstn context
is:

 

[incomming_pstn]

include => local-UK

include => local-US

include => test_numbers

 

and [test_numbers] includes:

 

exten => 555,1,Answer(0)                       ; Pick up phone instantly

exten => 555,n,Playback(vq51)         ; Let them know what's going on

exten => 555,n,Playback(vq20)

exten => 555,n,Goto(default,555,3)  ; repeat

 

 

So as far as I can tell, we should be accepting the connection and
playing the voicefile (yup - I know this would be open to the internet,
that's the intention).

 

Sip.conf also has:

 

allowexternalinvites=yes

allowexternaldomains=yes

 

so it should be working I think...

 

This is a 1.4.15 based asterisk

 

Thanks

 

Adrian

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