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Hello Developer Team, My name is Rustam Khadipash, I am a forth year student in Computer Science at Pusan National University. This summer I would like to contribute to your project, however I do not have experience in contributing to open source societies so far. Therefore, I would like to start with not a difficult, in my opinion, proj...

Hello, samba I have configured samba in my net, and it works well. From computers in the same sub-net, using w95, in the "Network Area" they can see the samba server. However, from computers in other nets, I can connect with "Connect to net unit", but I can't see the server in th "Network Area". What's the problem? Sorry for my bad English angel

...me if the info below is still correct: Note: In order to obtain useful DIALSTATUS information when dialing a peer you will need to have qualify=yes in that peer's definition (e.g. in sip.conf or iax.conf). http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS THANKS!! -- Regards, Rustam Kovhaev

Hi there, I would like to setup up my Asterisk to do this: receptionist answers the call, caller says he wants to leave a voicemail message for Ashleigh, receptionist transfers the call to Ashleigh's voicemail I guess It has something to do with dynamic features, or probably blind transfer to special ext. might do it what would you recommend? cheers!

Hello, I have the very same problem, does anybody know a way? I am thinking of converting to BIND, modifying and then converting it back to Internal DNS implementation. >>>> Hello. How could one modify a SOA record in rc3? For example, NS part (not NS record) of SOA record points to an absent Windows server. This effectively breaks DNS updates, since there is no such server and if

...ERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK...

...ERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK...

...n I am being transfered to operator, and then suddenly I get ISDN DISCONNECT message. I had this type of problem some time ago, and I thought it was a problem on the other end. But now this is a second time it occurred and I want an expert to take a look at my ISDN logs. Thanks!!!! -- Regards, Rustam Kovhaev -------------- next part -------------- -- Making new call for cr 32801 -- Requested transfer capability: 0x00 - SPEECH > Protocol Discriminator: Q.931 (8) len=39 > Call Ref: len= 2 (reference 33/0x21) (Originator) > Message type: SETUP (5) > [04 03 80 90 a3] > Bearer Ca...

...- SIP deadlock when running automated queues tests (Reported by Steve Pitts) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24312 - SIGABRT when improperly configured realtime pjsip (Reported by Dafi Ni) * ASTERISK-24426 - CDR Batch mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to sof...

...- SIP deadlock when running automated queues tests (Reported by Steve Pitts) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24312 - SIGABRT when improperly configured realtime pjsip (Reported by Dafi Ni) * ASTERISK-24426 - CDR Batch mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to sof...

All, What would be the most efficient way to load the data at the following address into a dataframe? http://ratings.fide.com/top.phtml?list=men Thanks, David -- View this message in context: http://r.789695.n4.nabble.com/Loading-Chess-Data-tp4642006.html Sent from the R help mailing list archive at Nabble.com.

Dear all, When I compared the performance of some application on both a Xen domU and a standard linux machine (where domU runs on a similar physical mahine), I notice the application runs faster on the domU than on the physical machine. Instrumenting the application code shows the application spends more time on popen() calls on domU than on the physical machine. I wonder if xenlinux does some

...h mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24312 - SIGABRT when improperly configured realtime pjsip (Reported by Dafi Ni) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24413 - parking/parking_tests: Crash due to assertion in unit tests when MoH is started on channel in holding bridge (Reported by Matt Jordan) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-24321...

...erisk.org/jira/browse/ASTERISK-24312>] - SIGABRT when improperly configured realtime pjsip (Reported by Dafi Ni) - [ASTERISK-23846 <https://issues.asterisk.org/jira/browse/ASTERISK-23846>] - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) - [ASTERISK-24413 <https://issues.asterisk.org/jira/browse/ASTERISK-24413>] - parking/parking_tests: Crash due to assertion in unit tests when MoH is started on channel in holding bridge (Reported by Matt Jordan) - [ASTERISK-24393 <https://issues.asterisk.org/j...

Hi Guys, I am trying to perform live migration of a qemu+kvm guest using the libvirt java api binding. Unfortunately I keep getting this error Exception in thread "main" org.libvirt.LibvirtException: internal error unable to add domain task 1598 to cgroup: No space left on device the number after "domain task" keeps changing on every run, the problem is migration fails

thanks Brandon it helped me out! -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFFw@public.gmane.org To unsubscribe from this group, send email to