search for: 1.4.26

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26. Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves a large assortment of issues reported by the community. 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 1.4.26. Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves a large assortment of issues reported by the community. For a summary of the changes in this release, please see the release summary:

Hi, I'm having a problem trying to get a Cisco 7965 phone registered on Asterisk 1.4.26. As we know, Cisco now, for security reasons, has made the phone ports non-symmetric, in that it sends out UDP requests on a high port and receives them on a different port. It seems that, even with 'nat' set to 'no', that Asterisk is not honoring the Contact header and keeps attempting to

Hello everybody, I was wondering what is postponing the 1.4.26 release? I thought it was scheculed for last week. Is there something we can do to help to release this version? There is no more issue reported on https://issues.asterisk.org/ for the time being. Best Regards, -- -- Marc LEURENT lftsy at leurent.eu -------------- next part -------------- An HTML attachment was scrubbed... URL:

Xapian 1.4.26 can now be downloaded from: https://xapian.org/download This release is mainly composed bug fixes and portability improvements. The wiki has the usual summary of the most notable changes: https://trac.xapian.org/wiki/ReleaseOverview/1.4.26 A big thanks to Sourav Saha, Baran Demir, A. Jiang, Qiu Yingbo, Arun Isaac and Ryan Schmidt for helping to make this release a reality. If

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release is primarily a fix for an issue (#14867, #14717) related to security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid registrations. Instead it sent

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc1 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release is primarily a fix for an issue (#14867, #14717) related to security fix AST-2009-001 where IAX was not sending REGREJ to terminate invalid registrations. Instead it sent

The Asterisk Development Team is pleased to announce the second release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc2 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes some issues reported by the community since the first release candidate. The most significant issues resolved include: * Treat an empty FORWARD_CONTEXT channel

The Asterisk Development Team is pleased to announce the second release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc2 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release fixes some issues reported by the community since the first release candidate. The most significant issues resolved include: * Treat an empty FORWARD_CONTEXT channel

Hello everybody, I try to install -Ubuntu 8.04 server- a B410P and a TDM2400P together with Asterisk 1.4.26-2, dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0. Problem I face is the following one: CLI> module load chan_dahdi.so == Registered application 'DAHDISendKeypadFacility' == Registered application 'ZapSendKeypadFacility' == Parsing

Hello, I'm trying to setup shared lines with Asterisk 1.4.26 and Snom phones. It seems that Asterisk works correctly (I get "State: SLA_TRUNK_STATE_RINGING" from the CLI) but the lamps on the phone are not blinking even if I setup one function key on my phones as shared line with number: <sip:line1 at 192.168.0.123> where line1 is my DAHDI/32 Any hints appreciated. :)

I'm new to asterisk, but have been at Linux since 1997 .. so kind of a newbie .. I have a good buddy who is supposed to be helping me with this darn install, but you all know how that works. I have a new mid-tower, AMD 64 x2, 4 gigs of memory and spoftware mirror. OpenSuSE 11.2 .. as I said Asterisk 1.4.26-rc6 and Asterisk-GUI 2.0 .. I have an 8 port TDM800P card from Digium. basic install

Hi, After upgrading a debian/lenny server to 1.4.26 I get this error: == Manager 'munin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'munin' logged on from 127.0.0.1 [Jul 26 17:45:12] ERROR[12354]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe repeated each time munin logs in. Should I be concerned?

--------------- Hi, I got hit with a funny one today; my configuration, which had been running fine for many weeks, suddenly stopped registering with Broadvoice's SIP. I switched among BV SIP hosts, verified my account was OK, even tried upgrading Asterisk. The weird thing was that I could associate OK with my WiFi SIP phone. So it looked like an Asterisk- specific SIP interoperability

Hi, I've created a tiny dialplan to test the return of a call on transfers, like this: (I had to use the DEVSTATE backport here) [phones] exten => _12XX,1,Dial(SIP/${EXTEN},6,tT) exten => _12XX,n,GotoIf($[ "x${BLINDTRANSFER}" = "x" ]?noBT) exten => _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)}); exten => _12XX,n,Goto(dRet) exten => _12XX,n(noBT),GotoIf($[

Hello list, For a customer I need to detect blue alarms on his T1 trunks. His server is in running asterisk and zaptel 1.4. I have a tool to generate all sorts of alarms, but on generating blue alarm, zaptel recognizes them as red alarms. This is not good for us as we need to get blue alarm as blue alarm. I checked using: cat /proc/zaptel/* zttool Are these versions of asterisk and zaptel

Hi list, I was going through my dialplan today and found these 2 oddities with meetme using DAHDI to join the conference. 1. Although music on hold is indicated, I don't get any sound until I press *. Then the conference menu plays and all is well except - 2. According to the instructions, pressing 8 should leave the conference. No dice. Problem persists with 1.4.30 Any

Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s.

Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live production. I cannot run valgrind and I do not have the right flags set in menuselect. I can however at the dead of the night run stress tests. I want to simulate x-amount of concurrent calls to both a dtmf dialplan, which is working, as well as MoH dialplan to see if this could be the cause of crashing. How

Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to