similar to: devicestate / inuse issue with 1.4.21.1

Hi, I've uploaded a new patch at https://issues.asterisk.org/view.php?id=16732which adds two new AMI commands, called "DeviceStateSet" and "DeviceStateGet". These commands let you update Custom device states, and read all devicestates from AMI. It would be very nice if someone could help me test this feature, and report back to the issue tracker. To test, log into AMI

I am currently using a copy of Asterisk checked out as the code of 10 days ago from Asterisk and the: sip show inuse reports that I have 3 incoming connections to one of the Grandstream phones, even though that isn't the case. I believe I have tracked the problem down to the following error message, which also (conveniently) showed up 3 times: -- Got SIP response 481 ""

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

Hello, On (03/30/16 16:12), Minchan Kim wrote: [..] > +static int get_zspage_inuse(struct page *first_page) > +{ > + struct zs_meta *m; > + > + VM_BUG_ON_PAGE(!is_first_page(first_page), first_page); > + > + m = (struct zs_meta *)&first_page->mapping; .. > +static void set_zspage_inuse(struct page *first_page, int val) > +{ > + struct zs_meta *m; > + > +

Hello, On (03/30/16 16:12), Minchan Kim wrote: [..] > +static int get_zspage_inuse(struct page *first_page) > +{ > + struct zs_meta *m; > + > + VM_BUG_ON_PAGE(!is_first_page(first_page), first_page); > + > + m = (struct zs_meta *)&first_page->mapping; .. > +static void set_zspage_inuse(struct page *first_page, int val) > +{ > + struct zs_meta *m; > + > +

The Asterisk.org development team has released Asterisk version 1.4.21.1. This release includes a critical bug fix for 1.4.21. All users that experienced lockups when upgrading to 1.4.21 should have their issues resolved with this update. Asterisk 1.4.21.1 is available for download from the downloads site: * http://downloads.digium.com/pub/telephony/asterisk Thank you for your continued

The Asterisk.org development team has released Asterisk version 1.4.21.1. This release includes a critical bug fix for 1.4.21. All users that experienced lockups when upgrading to 1.4.21 should have their issues resolved with this update. Asterisk 1.4.21.1 is available for download from the downloads site: * http://downloads.digium.com/pub/telephony/asterisk Thank you for your continued

Hi, I just installed 1.4.21.1, and from Mantis is looks like there isn't big issues with it specifically (at least nothing like I've seen). I had all my phones go "UNREACHABLE", but after that the CLI was unresponsive. No command would do anything or give out any info, including "restart now". Anything I should be looking at? Or is this a one-time bad luck? Mike

Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Hello, I just upgraded my asterisk to Asterisk 1.4.21.1 I am getting this Notice can any one tell me what i need to see in order to fix this problem. [Jul 18 18:27:08] NOTICE[9779]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' [Jul 18 18:27:09] NOTICE[9780]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' -- With Best Regards,

Hi All, I upgraded a PBX from 1.2. to 1.4.21.1 and I'm noticing that the hints for SIP channels are not updating the phones 100% of the time. The hints seem to work for some time, then the notification on the phone will hang in either and on or off state. During this condition, on the PBX, >core show hints, indicates the correct presence state for the SIP channel. Also if multiple

Currently, we store class:fullness into page->mapping. The number of class we can support is 255 and fullness is 4 so (8 + 2 = 10bit) is enough to represent them. Meanwhile, the bits we need to store in-use objects in zspage is that 11bit is enough. For example, If we assume that 64K PAGE_SIZE, class_size 32 which is worst case, class->pages_per_zspage become 1 so the number of objects in

I have recently upgraded to 1.2.7.1 from 1.2.4. I can no longer use "sip show inuse". Below is the output... I know there are current calls: redhat*CLI> sip show inuse * User name In use Limit * Peer name In use Limit Does anyone have an idea why this isn't working? Thanks, bp

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

I have patched astcc.agi with the HUP patch, but it still hangs from time to time. Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running Linux on 2006-05-07 00:31:09 UTC bye Ronald

Migration without --p2p works just fine, ie. the below works: $ virsh migrate --verbose --copy-storage-all \ --live cvm1 qemu+ssh://kashyapc@devstack3/system Migration: [100 %] Result: - On the source host, the guest is shut off - On the destination host, the guest is live migratied successfully Migration with "--p2p" fails, a simple test below:

Hello all! I have the following problem with the %in% command: 1) I have a data frame that consists of functions (rows) and genes (columns). The whole has been loaded with the "read.delim" command because of gene-duplications between the different rows. 2) Now, there is another data frame that contains all the genes (only the genes and without duplicates) from all the functions of

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the