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Adding gluster-users. On Wed, Jan 31, 2018 at 3:55 PM, Misak Khachatryan <kmisak at gmail.com> wrote: > Hi, > > here is the output from virt3 - problematic host: > > [root at virt3 ~]# gluster volume status > Status of volume: data > Gluster process TCP Port RDMA Port Online > Pid > ------------...

...functions and I compared the results with a more simple fit using a straight lm() command. The logarithmic fit works 100% correctly below. The problem is with the exponential fit: the nls command gives the wrong values and I have completely failed to see why this should happen. I can't see any misake in my self-start function, so why the mismatch?? Even worse, if I give nls a trivial fit by removing the "#&#&" on line 6, it fails to converge at all when the 'simpler' method with lm() easily gives the right answer (r=0.03 and A=5). I did the same exp and ln fits using...

Hello! Is it legal to use RedirectAction to redirect a call that is waiting in a queue? The idea is to have an external application manage a queue via manager API. The queue would merely collect calls and play moh. I've tryed this already but asterisk sends SIP/Forbidden to the channel in queue, after the channel has been redirected by RedirectAction, even though the response to

I anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the firmware, read below; I installed one today (turnaround time around 2 weeks to Australia, inc. economy freight in both directions... impressive!) and have noticed immediate, significant improvements. Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking,

Hello We have discovered a problem with DTMF on Asterisk. We have a setup with a T1 from PSTN going into an Asterisk box, and then out again on T1 and into a normal PBX (EADS) We use it to record all calls going to/from the PBX. The problem is that when we record the calls (with MONITOR command), DTMF tones gets obscured, and is not understood in the other end, if we dont Monitor, there are no

...14:56:17 DEBUG[4404] chan_zap.c: disabled echo cancellation on channel 6 Jan 9 14:56:17 VERBOSE[4404] logger.c: -- Hungup 'Zap/6-1' Jan 9 14:56:17 DEBUG[4404] app_dial.c: Exiting with DIALSTATUS=ANSWER. But neither the caller or the callee have made a hangup. -- Morten Isaksen http://www.misak.dk/blog/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060110/a231a81b/attachment.htm

...0512|SIP/080512 But this does not work in 1.6. It is expecting , and not | as the argument seperator. If I change the | to , then it does not work in 1.2. Is there any backwards compatible switch you can enable in 1.6, so it accepts | as a argument seperator in 1.6? -- Morten Isaksen http://www.misak.dk/blog/

...might want to try the 1.4.1 beta.??Info on how to > get the beta is available here: > > http://groups.google.com/group/Aastra-480i- > Users/browse_frm/thread/8f6f0f3419ef396d > > > I will try that and report back here. > > > -- > Morten Isaksen > http://www.misak.dk/blog/ > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users

...[3916]: Stopping retransmission on '07640a664ced6ffa7006f3916dac5cc2@194.255.106.3' of Request 103: Found Can someone explain what happend? How can we fix this problem. Will it help to upgrade to asterisk 1.0.9? Or upgrade the firmware on the TE410P card? -- Morten Isaksen http://www.misak.dk/blog/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050831/029990ef/attachment.htm

Hi, I have some interesting problem with htb , I set up root class and sub-classess: $TC qdisc add dev eth0 root handle 1: htb $TC class add dev eth0 parent 1: classid 1:1 htb rate 1990kbit ceil 2000kbit $TC class add dev eth0 parent 1:1 classid 1:10 htb rate 190kbit ceil 200kbit $TC class add dev eth0 parent 1:1 classid 1:11 htb rate 1400kbit ceil 1600kbit $TC class add dev eth0 parent 1:1