On 7/4/05, Ronald_Wiplinger <[EMAIL PROTECTED]> wrote: > Robert Goodyear wrote: > > >>>>>> I am confused about one of my installed server > >>>>>> > >>>>>> The dial plan seems to be ok, but sometimes NOTHING happens if I > >>>>>> try to dial an extension (from X-Lite), next time it is done. > >>>>>> > >>>>>> X-Lite does not have a tone, nothing and does also have no time > >>>>>> out. It seems it is not connected to the server. However, a sip > >>>>>> show users / sip show peers shows that the phone is connected. > >>>>> > >>>>> SIP clients generate their own dialtone, so if you've got no tone, > >>>>> that sounds suspicious of a problem with the client itself. I > >>>>> assume you've debugged the problem by registering a hard SIP > >>>>> client on that server? > >>>> > >>>> The CLI prompt does not show anything either. It is like the phone > >>>> is not talking to asterisk at all. > >>>> sip show users/peers does show the phone. > >>> > >>> ...shows the phone REGISTERED, yes? > >> > >> yes!!! > > > > > > ...yet no other information in the CLI or logs? C'mon, help us help > > you. The clue is in the question. > > > I cannot make up a CLI entry ;-) > There is nothing about it!!! > As I said it is like it is not connected!!!!! >
Do you have qualify=1000 or some value in the sip.conf? Are you getting a time when you do a sip show peers? It could be the phone is registering and then losing network, and if the registration time is an hour it would still show as registered even if it was uncontactable. (I think). IANAAE (I am not an asterisk expert.) e.g. 212/ 192.168.0.25 D 255.255.255.255 5062 OK (24 ms) 211/ 192.168.0.25 D 255.255.255.255 5060 OK (27 ms) 210/ 192.168.0.23 D 255.255.255.255 5060 OK (59 ms) 203/ (Unspecified) D 255.255.255.255 0 UNKNOWN Regards Mark _______________________________________________ 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
