similar to: Asterisk 1.6.1.0 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a

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:

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

The Asterisk.org development team is pleased to announced the release of Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release 1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons 1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of Asterisk-Addons is compatible with both Asterisk 1.6.1 and 1.6.2 branches. These releases are available for

Hi Jeff, I have a single-CPU system with P4 HTT-enabled processor (7.1-RELEASE-p3), kernel compiled with SCHED_ULE. Copyright (c) 1992-2009 The FreeBSD Project. Copyright (c) 1979, 1980, 1983, 1986, 1988, 1989, 1991, 1992, 1993, 1994 The Regents of the University of California. All rights reserved. FreeBSD is a registered trademark of The FreeBSD Foundation. FreeBSD 7.1-RELEASE-p3 #0:

my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says == Using SIP RTP CoS mark 5 -- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack -- Called trunk10 at 147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by

Hi All, I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP. On incoming calls from Avaya asterisk complains of 'unsupported crypto parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=-

Hello, is there anywhere an overview of SIP error codes and under which condition they are reported by Asterisk? There are general definitions for SIP error codes, but they are quite general and it's Asterisk that actually checks what's wrong and then reports an error. Now, currently I could check the source code to get more informations what could have caused the error, but that's

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a

http://bugzilla.mindrot.org/show_bug.cgi?id=1103 Summary: New option for ssh(1) to make it failing when remote port forwarding can't be established Product: Portable OpenSSH Version: 3.8.1p1 Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P2 Component: ssh

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

The Asterisk Development Team is pleased to announce the release of Asterisk-Addons 1.6.1.0. Asterisk-Addons 1.6.1.0 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release maintains compatibilty of the Asterisk-Addons package with the Asterisk 1.6.1 series of releases, since it is possible for API changes to occur between 1.6.x releases. Additionally,

The Asterisk Development Team is pleased to announce the release of Asterisk-Addons 1.6.1.0. Asterisk-Addons 1.6.1.0 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release maintains compatibilty of the Asterisk-Addons package with the Asterisk 1.6.1 series of releases, since it is possible for API changes to occur between 1.6.x releases. Additionally,

Hi, I have set up an asterisk with TLS and SRTP support. The SRTP is working with Phonerlite softphone. I have problem with the SRTP, when I make calls on Audiocodes gateway . I got the folloowing messages on asterisk: [Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80

Hi, I am trying to install Gatekeeper. I have installed pwlib and trying to install h323 plus. I have set the path as PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/open323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH I also configured /etc/ld.so.conf include /root/pwlib/lib include /root/h323plus/lib Then I tried running ldconfig. But it

When the support for h323plus was announced for Asterisk 1.4.25, I tried to build this support in Asterisk. For this, I checked out the h323plus CVS from SourceForge, which reported version 1.20.beta5, and also the ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a chan_h323 for Asterisk 1.4.25, which apparently loads correctly, but now I see that I get a segfault whenever I

The Asterisk Development Team is pleased to announce the fourth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc4 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release candidate improves the performance of the ast_event cache functionality, fixes issues with rwlock corruption that would cause deadlock like symptoms, fix queue weight behavior so

The Asterisk Development Team is pleased to announce the fourth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc4 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release candidate improves the performance of the ast_event cache functionality, fixes issues with rwlock corruption that would cause deadlock like symptoms, fix queue weight behavior so

The Asterisk Development Team is pleased to announce the fifth release candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc5 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release fixes a couple of issues with realtime music on hold that could cause Asterisk to crash, and an issue that caused hungup channels to stay up, leading to 100% CPU usage. Additionally,