similar to: Asterisk 11.12.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 11.12.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 12.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 12.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 1.8.30.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.30.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.30.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.30.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi, My asterisk segfaults a few times each day and the crash problem seems weird. When I run gdb on the core dump, it almost always segfaults on free() or malloc(). When I run the back trace, I see something weird. Here's one of the back traces. #0 0x4017f87f in _int_free () from /lib/libc.so.6 #1 0x4017e7f5 in free () from /lib/libc.so.6 #2 0x403be83e in __sip_destroy (p=0x8248c88,

Ok, Please, check your manager.conf and logger.conf for any clue about debugging options, into the Asterisk configuration directory. El 26 mar. 2017 14:52, "Telium Technical Support" <support at telium.ca> escribi?: > I tried that but it had no effect. Still see things like: > > > > [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining >

hi, any plans update patch for 11.12.0? |https://github.com/meetecho/asterisk-opus https://github.com/netaskd/asterisk-opus/ | patching file build_tools/menuselect-deps.in patching file channels/chan_sip.c Hunk #1 succeeded at 7659 (offset -98 lines). Hunk #2 succeeded at 11011 (offset -34 lines). Hunk #3 succeeded at 11050 (offset -34 lines). Hunk #4 succeeded at 11117 with fuzz 1 (offset

Hi, I'm running into a strange problem with commas is variables. I have the following contexts: [messages] exten => _+.,1,Noop(External SMS) same => n,Set(ACTUALTO=${CUT(CUT(MESSAGE(to),@,1),:,2)}) same => n,Macro(goip_sendsms,${ACTUALTO},"${MESSAGE(body)}") same => n,Hangup() [macro-goip_sendsms] ;Call Macro(goip_sendsms,number,message) exten => s,1,Noop(SMS

Hello, I looked at my log file this morning and I see a lot of stuff about ACLs that I don't understand and this notice: INTERNAL ERROR: Signal 11 in pid 23985 (3.0.9-1.3E.3) I suspect that it was caused when a user tried to change permissions on the contents of a directory or delete it (I'm trying to find that out now). The directory seemingly no longer exists, so the user may have

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Hi Ron, I don't remember right now, but you can try this command: cli> manager set debug off Cheers El 26 mar. 2017 3:58, "Telium Technical Support" <support at telium.ca> escribi?: I somehow cause AMI events to appear as output in the CLI, and I can?t figure out how to turn them off. Can someone offer a command which will suppress AMI events/commands from showing in

I have been shaking down a dialplan for SIP fax to efax. The basic senario is an ATA on the same subnet as the Asterisk 1.2 box (avoid RTP packet lose and thus fax crash), calling a 'fax extension' and envoking rxfax then email. I leverage off of context: from-internal-additional-custom, so as not to have it overwritten by FreePBX. In extension_custom.conf I have:

(please keep me on Cc when replying) I have a server running Xen that regularly spews the following. The box seems to survive fine regardless - just thought I''d let everyone know. Dec 8 12:19:26 server kernel: 0x47/0x7a Dec 8 12:19:26 server kernel: [alloc_skb_from_cache+70/243] alloc_skb_from_cache+0x46/0xf3 Dec 8 12:19:26 server kernel: [__dev_alloc_skb+70/92]

Yes: # getent group GROUP group:x:17573: # getent group group2 group2:x:11010: # getent group GROUP3 group3:x:21178: # wbinfo --group-info GROUP group:x:17573: # wbinfo -n GROUP S-1-5-21-948789634-15155995-928725530-7573 SID_DOM_GROUP (2)

Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a

My stat textbook tells me that using Shapiro-Wilk test for each variable one by one is not equal to a test for multivariate normality as a whole. Does R have a function of testing for multivariate normality? Thanks. Hongsheng (Hank) Liao, Ph.D. Lab Manager Center for Quantitative Fisheries Ecology 800 West 46th Street Old Dominion University Norfolk, Virginia 23508 Phone:757.683.4571

Attached is a histogram chart with a break on y-axis which my friend made for me using phot shop. Do anyone know how to make such a break using R? Thanks in advance. (See attached file: 2007 age distribution.doc) Hongsheng (Hank) Liao, Ph.D. Lab Manager Center for Quantitative Fisheries Ecology 800 West 46th Street Old Dominion University Norfolk, Virginia 23508 Phone:757.683.4571

Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten => TestTrakt,1,Dial(ZAP/1-2/517255333) exten => TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [TestTrakt at