similar to: Asterisk 1.8.26.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 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 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 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 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 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 12.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.1.0 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: New

The Asterisk Development Team has announced the release of Asterisk 12.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.1.0 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: New

Hey All ~ 1, queue records on fairly unreliable. I would say about 40 - 60 percent of the queue calls are not being recorded and I'm not sure why. I don't seem to see any kind of pattern to the failure. I've added a sample of our queue config at the bottom. 2, cel_pgsql module seems to crash regularly. It seems every time I look at our asterisk server, the cel_pgsql module is

Hi, Anybody seen this before? (using a pre-compiled asterisk from the OBS on a sles11sp1) (I mean, i did the same with a 1.6 without any problem, but i need 1.8) after starting: kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk ended with

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code,

Hello asterisk-users, We've been having intermittent issues with chan_sip - it stops responding to cli requests, trying to reload chan_sip from cli doesn't seem to have any effect, initiated calls carry on for a short period, but no new SIP requests are processed ('sip show channels' hangs forever, server stops responding to SIP OPTIONS, or any other SIP messages). We have updated

Hello everyone, I would like to seek your advice regarding a build (or rather "configure") problem I am running into. For reference, tests are all relative to a build from a 1.8.26.0 tarball, on Debian Wheezy. I would like to understand if it is possible, and if any of you have tried, to build Asterisk from a subdirectory of the source tree, i.e., from a clean source tree # mkdir

Hi Richard, Thank you for the reply Correct, I did mean 13.15. I set no optimize and better backtrace through "make menuselect" and the output is now [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0) Got 26 backtrace records #0: [0x61923f] main/utils.c:2475

The Asterisk Development Team has announced the release of Asterisk 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.18.0 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 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.18.0 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

Hello, I've recently setup a small load test against an instance of Asterisks. I've tested on asterisk 13.5 and 14.6 with the same results. I am using PJSIP. My dial plan is, [test] exten => 1001,1,Answer exten => 1001,n,MusicOnHold(15) exten => 1001,n,Hangup I am using SIPP to test. I can share XML if desired but it simply waits on the line while music plays for 8

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.4.0 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: New

The Asterisk Development Team has announced the release of Asterisk 13.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.4.0 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: New

Hi! I have the following problem: winbindd getgrnam/getpwnam/lookupname calls do not return for a long time (> 30 seconds) which is obviously quite annoying to the users. This usually happens at the "first" login, i.e. subsequent connections are not delayed until a (probably cache) timeout. I've failed to determine who is responsible for the caching yet. I'll provide

Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint. I wonder to force asterisk to refresh the session in some cases when is needed . pjsip is able to