|
qualify is not set in sip.conf at all. What
should the value be, or should it just be set to yes? The register interval is 60 minutes. The
Asterisk server is not going down, but the connection between the phone and the
server might go down for a few minutes, and when it comes back up the problem
occurs. From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Paul Rodan What is the register interval in the grandstreams?
The qualify=yes should keep the connection alive as long as Asterisk is up, but
if it goes down and then comes back up, the phone has to re-register with
Asterisk before asterisk can keep the connection alive. From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Norton Hi, I have several Grandstream phones connected to Asterisk,
some behind NAT and others not. If I reboot all the phones, everything is fine.
Should the connection go down, and then come back again, those behind a NAT are
still able to make calls, but are unable to receive calls. -- Executing
Dial("SIP/1239-ba74", "SIP/1242|60|t") in new stack Jan 12 23:45:19 NOTICE[21576]: app_dial.c:803 dial_exec:
Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) However, extension 1242 is still able to call 1239? Is this a configuration problem in Asterisk or in the
phones? Please help Regards David Norton -- This message has been scanned for viruses and dangerous content and is believed to be clean. |
_______________________________________________ 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
