similar to: Asterisk 1.4.32 Now Available

The Asterisk Development Team has announced the final maintenance releases of Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to

The Asterisk Development Team has announced the final maintenance releases of Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to

The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.4.31. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.31 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved with the help

The Asterisk Development Team has announced the release of Asterisk 1.4.31. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.31 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved with the help

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.25. Asterisk 1.4.25 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves several crash issues, DTMF related issues, and CDR related issues. For a summary of the changes in this release, please see the release summary:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.25. Asterisk 1.4.25 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves several crash issues, DTMF related issues, and CDR related issues. For a summary of the changes in this release, please see the release summary:

The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release:

The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release:

For a long while, Asterisk administrators have had the desire to take PRI channels out of service and keep them that way, usually to create a window in which to perform maintenance or another such purpose. We are finally very close to having that functionality in Asterisk, but what we need is a few people who have PRI circuits to test out this functionality. Please, if you are an administrator

> On 18 May 2018, at 20.19, David Hubbard <dhubbard at dino.hostasaurus.com> wrote: > > Hello, given the 2015 revision date, I was curious if anyone can confirm https://wiki2.dovecot.org/Timeouts is still accurate where the 'before login' IMAP timeout remains hard coded? > > We're having an issue where blocks of IP's from China and similar locations are

It appears as though the 489 Bad Event response to the NAT keep alive event responds to the local address, instead of responding to the NATted address. This causes Linksys phones to go amber (no registration) after a short amount of time after placing calls. Turning the Linksys NAT keep alive off is a workound, but non-ideal in may situations. Apparently the asterisk devs don't even think

bugzilla-daemon@lists.xensource.com, le Thu 20 Mar 2008 23:23:35 -0700, a écrit : > http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1194 > > ------- Comment #3 from haicheng.li@intel.com 2008-03-20 23:23 ------- > This bug may be related to c/s #17266 which says that IDE should accept SETMUL > 0 as upstream qemu now does. It is following: > > ---

Hi I have configured IAX2 realtime in Asterisk 1.6.2.13. when I cannect a client to realtime extension, always the state of extension is "UNKNOW" like: * Name : marco Secret : <Set> Context : phones Parking lot : Mailbox : 2345 at default Dynamic : Yes Callnum limit: 0 Calltoken req: Auto Trunk : No Encryption : No

Hi, I have a 1.8.22 Asterisk (Box A) connected to a 1.4.32 Asterisk box (Box B) through SIP. The 1.4.32 box is then connected to the PSTN through PRIs. I've noticed there are occasions where I am seeing duplicated DTMF. I've verified from the SIP trace from the phone that there is only a single '3' being pressed. It appears as though the DTMF end (without a begin) that is detected

Asterisk 1.4.32 dahdi-2.3.0.1 Centos 5.5 Digium TE420 CAC channel bank (2) Cisco RVS4000 router Cox 50 Mbps/ 5 Mbps cable modem Flowroute provider codac G-711 90 % CPU idle callwaiting=no When there are 10-15 or more calls up the farend hears a callwaiting like beep every 3 to 6 sec. the duration of this "beep" is very short and would be no problem if it didn?t happen every few

asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel < 2.0 (from extensions.conf) exten=> 304,1,ChanSpy(Zap/4|q) exten=> 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304 at flash:1] ChanSpy("Zap/31-1",

If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn Asterisk's sip debug information on, you'll probably find errors like these in your logs: NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce received from

When I use an eagi script when play a message appear a lot of warning messages, but it play very well I?m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300