> Thanks, but those extensions are listed in that list! > > I'm stumped > > Richard Adamson wrote: > > >Better read up on why a sip phone should register with asterisk. Do a 'sip > >show peers' and that will be the list of phones that can "receive" calls. > >------------------- > > > > I've double checked this. Everything is logging in fine, because I can > > make calls, check my voicemail, everything except recieve calls on the > > SIP devices. > > > > David Phelan wrote:
Okay, let's go through basic sip stuff then... In basic asterisk configurations, each sip phone is expected to register with asterisk. That register process essentially informs asterisk which IP address is associated with that phone (or extension). The register process will happen at the time the phone boots up AND about every 3600 seconds thereafter. (The exact time might be different then 3600 seconds and is either sip phone vendor dependent or configuable on some sip phones.) After the register process is complete, executing a 'sip show peers' should display something like: 3000/3000 211.222.191.73 D 255.255.255.255 5060 Unmonitored If there is an IP address shown, the registration was successful. If there is no IP, then the registration either expired or it failed. If an IP is displayed, it only indicates the registration process was successful at "some previous time", and that could have been ten minutes ago or 30 minutes ago. It does NOT indicate there is still contact with that phone. (Eg, unplug the sip phone ethernet cable and you'll still see that same display an hour later.) If you want to "ensure" the 'sip show peers' is always up to date and accurate, then include "qualify=2000" in that sip phone's definition within sip.conf. That will force asterisk to essentially test to see if that IP address is still reachable every 2000 milli- seconds (or every two seconds). Add that statement into your sip.conf definition and do your tests again. If that statement resolves your problem, then one of two things are occurring: 1. your loosing contact between the sip phone and asterisk, which is most likely related to nat issues (since I don't recall you stating that phones and asterisk are on the same wire). 2. you have another problem that none of us can even guess at since you've not provided any real technical information about your system. If the above does not help you resolve the issue, then you will have to post more technical info before anyone can actually help you. That should include: - your exact sip.conf entry for the sip phones in question, - your exact extensions.conf entries (including the 'context') associated with the problem, - output from 'sip show peers' - any command line debug statements shown during an attempted call to that extension. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
