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New 1.4 system: registered, but not responding to invite?

Posted by NaTel on
Forums
Thirdlane platform General Questions

I've turned on NAT everywhere I can think, but
even though I hear ringing on the calling
phone (different system) the called phone does
not ring.

Has anyone bumped into this lately?

Submitted by eeman on Mon, 12/21/2009 Permalink

congratulations, you've take the award for providing the least possible amount of information possible while asking for help. you'll need to take your 'verbose -5' setting and re-set it to around 'verbose +15'.

Submitted by NaTel on Mon, 12/21/2009 Permalink

Thanks, Erik.

That award includes the 4-week vacation to the Virgin Islands, doesn't it?
How many virgins do I get with that?

I did a packet capture and added some notes to point out
what is going on:

http://www.A7H.com/~stuph/TCPdump-2009-Dec-21-2304.txt

I just don't see where the extension responds to
the INVITE. What would prevent that?

By the way, I have a bunch of phones behind this
same router that work just fine on our old v1.2
system.

Submitted by eeman on Sat, 12/26/2009 Permalink

thats not what that setting does... that setting just specifies the default value when CREATING a sip extension. Existing entries in sip.conf will not be changed retroactively. You still must go into the specific extensions and change their settings. This is not technically a solution as much as it is a work-around. Your problem lies with being behind nat and having a router that likes to kill off your connections frequently. This problem could re-occur, less frequently, if the router is still too aggressive at pruning its port address translations. The only true fix for this is to proxy sip at the router where nat occurs so that the sip messages are not nat'd at all.

Submitted by eeman on Sat, 12/26/2009 Permalink

btw had you mentioned more information in the beginning you would have been helped sooner.

your comment implied you built a new asterisk 1.4 pbx and could not get calls to occurr between the new pbx and the existing thirdlane pbx. At no point did you mention a handset trying to communicate through a nat'd router.

next time try something like:

I just built a new thirdlane server running asterisk 1.4.26 and thirdlane 6.0.1.75 on a centos 5.3 distribution. I have a linksys 942 handset running firmware 5.3.5 behind a crappy linksys wireless router. I have nat=yes enabled and I can see the phone registered but after a few minutes beyond power-up I can no longer call this device

that would instantly tell everyone on the forum you were having problems with keepalives.