On July 12, 2012 from approximately 11:00AM to 11:30AM (Central Daylight Time, GMT-5), the core routers that provide connectivity through to all Asterisk community services will be swapped out.
This will mean that these services will be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes.
The services affected include:
bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org
The Asterisk Development Team has announced the release of Asterisk 10.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.6.0 resolves several issues reported by the community like:
- format_mp3: Fix a possible crash in mp3_read(). (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)
- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis)
- Re-add LastMsgsSent value for SIP peers (Closes issue ASTERISK-17866. Reported by Steve Davies)
- Prevent sip_pvt refleak when an ast_channel outlasts its corresponding sip_pvt. (Closes issue ASTERISK-19425. Reported by David Cunningham)
- Send more accurate identification information in dialog-info SIP NOTIFYs. (Closes issue ASTERISK-16735.…
The Asterisk Development Team has announced the release of Asterisk 188.8.131.52. This release is available for immediate download at ttp://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 184.108.40.206 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- format_mp3: Fix a possible crash in mp3_read(). (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) * --- Fix local channel chains optimizing themselves out of a call. (Closes issue ASTERISK-16711. Reported by Alec Davis) * --- Update a peer's LastMsgsSent when the peer is notified of waiting…
The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): "P-Asserted-Identity", "Remote-Party-ID" or "From:". I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44.
I've installed 10.6.0-rc2 on two machines. On one of the machines (but not the other) /tmp gets filled with:
............... -rw-------. 1 asterisk asterisk 53661696 Jul 7 23:46 core.PBX-2012-07-07T23:46:10-0400 -rw-------. 1 asterisk asterisk 53891072 Jul 7 23:48 core.PBX-2012-07-07T23:48:55-0400 -rw-------. 1 asterisk asterisk 53469184 Jul 7 23:53 core.PBX-2012-07-07T23:53:00-0400 -rw-------. 1 asterisk asterisk 53739520 Jul 7 23:58 core.PBX-2012-07-07T23:58:25-0400 ...........
and finally fills up all the space.
grep -v ';' /etc/asterisk/logger.conf
[general] [logfiles] console => notice,warning,error messages => notice,warning,error
Any clue what to look for?