search for: 20000ms

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

...c the disk image to the destination - shut down the guest - rsync the disk image again (to minimise the off-line period) - dumpxml of guest and copy to destination - define and start the guest on the destination With the methode of the live migration, the machine will be unresponsive for max 20s = 20000ms (or less if I chose to). The migration it self will take op to 5 hours but I guess I plan these nightly. With the other migration the guest will be really off-line for the duration of the (second) rsync. In my case with disk images from 50 to 150Gb, the rsync wil take up at least 30 minutes and ev...

...estination - shut down the guest - rsync the disk image again (to > minimise the off-line period) - dumpxml of guest and copy to > destination - define and start the guest on the destination > > With the methode of the live migration, the machine will be > unresponsive for max 20s = 20000ms (or less if I chose to). The > migration it self will take op to 5 hours but I guess I plan these > nightly. > > With the other migration the guest will be really off-line for the > duration of the (second) rsync. In my case with disk images from 50 to > 150Gb, the rsync wil take...

...hival purposes, I have setup the Intellitouch ITC-3002 (2006) SIP phones for use with asterisk (1.2.7.1). After a few "gotcha's", I was able to do transfer's, moh's, push a button to check voicemail, callerID, etc.. One big gotacha was the dial timeout (by default!) is set to 20000ms (20s) and its NOT configurable from the internal web gui on the phone. You have to download the config, edit it, then up the config again.. The line: dial_timeout=<your delay in ms> will set the dial delay Another gotcha is sip registration. Even though you set the authentication ID and pa...

...e a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here... PJ BILL-GW: bill*CLI> iax2 show peer ipbx-gw Status : OK (4 ms) Qualify : every 20000ms when OK, every 10000ms when UNREACHABLE (sample smoothing On) [Nov 4 11:06:58] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE [Nov 4 11:06:58] Timestamp: 00019ms SCall: 00002 DCall: 00000 [193.179.bbb.bbb:4569] [Nov 4 11:06:58] Rx-Frame Retry[ No] -- OSeqno:...

...9;n' option. I haven't gone through the entire code of app_queue to see exactly how membercount gets manipulated, but it seems from my log that these exitwithtimeouts events seem to occur right after an agent has let their phone ring without picking it up (see the "nobody picked up in 20000ms" message in my example above). Is it possible for qe.parent->membercount to be set to zero in a queue where all agents but one are on the phone and that one remaining agent lets their phone ring without answering it? -- j. -------------- next part -------------- An HTML attachment was sc...