similar to: Asterisk 11.8.0 Now Available

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 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 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 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

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

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

-----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,

Dear R experts, I'm trying to install rgdal in my R 2.11.1 (Ubuntu 10.10). I have as an ouput the following messages. It seems that there are problems with libgdal1.7.0 or sqlite3... Could someboby help me? Thanks a lot! Ale > install.packages() Warning in install.packages() : argument 'lib' is missing: using '/home/ale/R/i686-pc-linux-gnu-library/2. 11' trying URL

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

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

Hi. In my setup exim delivers mails to dovecot using LMTP. In one LMTP session exim can deliver up to 200 recipients (batch_max set to that value). Now the problem is that sometimes 256MB is not enoug for dovecot lmtp to handle incoming emails. My questions: - how big memory limit should be for lmtp? I was thinking that lmtp (more or less) simply reads from one descriptor and writes to file,

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

Hi, after upgrade from 11.8.1 to 11.9.0 on our test server, and from 1.8.26.1 to 1.8.27 on production one, some CLI commands like "sip reload" or "iax2 reload" does nothing. We opened bug 23683 but it was immediately closed by Matt Jordan, telling that he can't reproduce it. But we can. Example: - switching back to 11.8.1 respectively 1.8.26.1 does the job working

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

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