similar to: Asterisk 1.8.30.0 Now Available

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 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 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:

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, 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

The Asterisk Development Team has announced the release of Asterisk 1.8.29.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.29.0 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 1.8.29.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.29.0 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 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)

Hi list! Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian 9). Well, I copied the configuration and changed what needed, so basically, it works, at least with my tests. But when Asterisk will be started, in the message log I get this error: [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel

Hi list! I'm using Asterisk 1.8.30.0 on a OpenWRT device and it works perfectly. I configured a voicemail and I receive an E-Mail with some information about the call. Again, wonderful! Now my wish: I'd like to have Asterisk to search the caller in a list file and send me the name corresponding to the number in the E-Mail of voicemail. Is it possible? I currently use ${VM_CALLERID} in

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

Max Grobecker <max.grobecker at ml.grobecker.info> schrieb: Hello Max, > I'm also a customer of the DTAG. > Yesterday, the messed a bit with their DNS entries... Maybe they tried again to repair a working system... :) > If you are NOT using their DNS resolvers you got a "wrong" IP address back > that was not working. Besides that, you should disable SRV lookups