Thanks. Tell me, how intensive is it to use qualify? What type of packet/check is done with this? Is it reasonnable to use qualify for thousands of devices? Once the device is considered to be unreachable for any number of reasons, will another poll of the device be done to check if it became available again after the configured number of milliseconds? Or will it be considered unreachable until the next register attempt by the device? Regards, Mike
_____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Cennami Sent: Wednesday, August 01, 2007 17:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with the dial command qualify=yes in the sip.conf context for that device will change the device to unreachable and should send you directly to voicemail. There could still be a brief period where the device is timed out and the system hasn't qualified it yet, but outside of that, it will just continue trying to send to the device. On 8/1/07, Mike <[EMAIL PROTECTED]> wrote: Thanks Jared. It answers most of my question. Now, when the device doesn't register, the behavior is as expected. But eventually, any device that registers successfully might be unplugged, leaving Asterisk to wonder where the device has gone. So, what's the best approach to this? Should I put a timeout=x minutes for that SIP registration, and force the Polycom phone to reregister every y minutes (y being smaller than x)? How do I do this? Is this anyway to force Asterisk to consider the peer disconnected if Asterisk doesn't get a reply back within a second of trying a Dial command? Is this any other obvious option that escapes me? Mike -----Original Message----- From: [EMAIL PROTECTED] [mailto: <mailto:[EMAIL PROTECTED]> [EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Wednesday, August 01, 2007 14:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with the dial command On Wed, 2007-08-01 at 11:43 -0400, Mike wrote: > Aug 1 11:47:57 NOTICE[26107]: app_dial.c:1069 dial_exec_full: Unable > to create channel of type 'SIP' (cause 3 - No route to destination) This happens when Asterisk don't know where to find the peer (which is often the case if the device has failed to register to Asterisk, for example). > Sometimes, instead, the phone doesn't ring and I get a 15 second > silence on the calling end. After the full 15 seconds, Asterisk goes > to the next priority. This would happen, for example, if the phone registers with Asterisk but then gets unplugged from the network. Asterisk has an IP address for the peer and is trying to call it, but the peer isn't responding. -- Jared Smith Community Relations Manager Digium, Inc. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- <http://www.api-digital.com--> asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anthony Cennami
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