Mike wrote: > 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] <mailto:[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:[EMAIL PROTECTED]> > [mailto: [EMAIL PROTECTED] > <mailto:[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 > ------------------------------------------------------------------------ > > _______________________________________________ > --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 Yes it continues to try to qualify after it is down. When it successfully qualifies again it triggers a peer reachable event. I personally have my servers qualify every peer, it does not add a noticeable amount of resource utilization.
Anthony _______________________________________________ --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
