Archive for category sip
Asterisk 1.8.0-Beta3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:
- Fix a regression where HTTP would always be enabled regardless of setting.
(Closes issue #17708. Reported, patched by pabelanger) - ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
(Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - Support “channels” in addition to “channel” in chan_dahdi.conf.
(https://reviewboard.asterisk.org/r/804) - Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
(Closes issue #17661. Reported by oej. Patched by mmichelson) - Fix inband DTMF detection on outgoing ISDN calls.
(Patched by russellb and rmudgett) - Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel) - Fixed IPv6-related SIP parsing bugs and updated documention.
(Reported by oej. Patched by sperreault) - Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
(Closes #17713. Reported, patched by gareth. Tested by tilghman)
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support in the SIP Channel
- Connected Party Identification Support
- Calendaring Integration
- A new call logging system, Channel Event Logging (CEL)
- Distributed Device State using Jabber/XMPP PubSub
- Call Completion Supplementary Services support
- Advice of Charge support
- Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3
Thank you for your continued support of Asterisk!
Asterisk 1.4.33-Rc1 And Asterisk 1.6.2.9-Rc1 Now Available
Posted by admin in Release Candidates, asterisk, sip on June 2, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced release candidates of Asterisk for versions 1.4.33 and 1.6.2.9. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These release candidates address issues that were reported by the community and resolved since the last round of bug fix releases.
The following is a sampling of issues resolved in these release candidates:
- Allow compilation on Mac OS X 10.4 (Tiger)
(Closes issue #17297. Reported by jcovert. Patched by tilghman) - Fix issue where IMAP backend would count messages twice in INBOX for MWI.
(Closes issue #17135. Reported, patched by edhorton. Patched by tilghman) - Fix segfault on logging.
(Closes issue #17331. Reported, tested by under. Patched by dvossel) - Fix crash when processing Cisco DTMF samples.
(Closes issue #17248. Reported by falves11. Patched by dvossel) - Fix crash in check_rtp_timeout.
(Closes issue #17271. Reported, patched by under. Tested by dvossel) - Fix transcode_via_sln option with SIP calls and improve PLC usage.
(Patched by mmichelson. Reviewboard https://reviewboard.asterisk.org/r/622/ )
For a full list of changes in the current release candidates, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33-rc1
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9-rc1
Issues found in any of these release candidates should be reported to the Asterisk issue tracker at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk 1.4.32 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.32 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 developers:
- Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by: geoff2010. Patched by dhubbard) - When StopMonitor is called, ensure that it will not be restarted by a channel event.
(Closes issue #16590. Reported, patched by: kkm) - Fix up hidecallerid feature in chan_dahdi.
(Closes issue #17143, #7321. Reported, patched by djenson99) - Resolve deadlocks in chan_local.
(Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) - Ensure channel state is not incorrectly set in the case of a very early answer.
(Closes issue #17067. Reported, patched by tzafrir) - Registration fix for SIP realtime.
(Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.32
Thank you for your continued support of Asterisk!
Asterisk Security Advisory – AST-2010-001: T.38 Remote Crash Vulnerability
Posted by admin in Asterisk Security Advisories, Security Advisories, asterisk, sip, t.38 on February 3, 2010
Asterisk The Open Source PBX & Telephony Platform
Asterisk Project Security Advisory - AST-2010-001
| Product | Asterisk |
| Summary | T.38 Remote Crash Vulnerability |
| Nature of Advisory | Denial of Service |
| Susceptibility | Remote unauthenticated sessions |
| Severity | Critical |
| Exploits Known | No |
| Reported On | 12/03/09 |
| Reported By | issues.asterisk.org users bklang and elsto |
| Posted On | 02/03/10 |
| Last Updated On | February 2, 2010 |
| Advisory Contact | David Vossel < dvossel AT digium DOT com > |
| CVE Name | CVE-2010-0441 |
Asterisk 1.6.0.22, Asterisk 1.6.1.14, Asterisk 1.6.2.2 Released
Posted by admin in Asterisk Security Advisories, Releases, Security Advisories, asterisk, sip, t.38 on February 3, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced security releases for Asterisk as the following versions:
These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001.
The issue is that an attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash will occur when the FaxMaxDatagram field is omitted from the SDP, as well.
For more information about the details of this vulnerability, please read the security advisory AST-2010-001, which was released at the same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.22
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.14
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.2
Security advisory AST-2010-001 is available at:
http://asterisk.net.ru/en/2010/02/03/asterisk-security-advisory-ast-2010-001-t-38-remote-crash-vulnerability/
Thank you for your continued support of Asterisk!
Asterisk 1.6.0.21 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.0.21.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.21 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
- Fix to Monitor which previously assumed the file to write to did not contain pathing.
(Closes issue #16377, #16376. Reported by bcnit. Patched by dant. - If EXEC only gets a single argument, don’t crash when the second is used.
(Closes issue #16504. Reported by bklang. Patched by tilghman.) - Avoid a crash with large numbers of MeetMe conferences.
(Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) - Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10).
(Patched by seanbright.) - Allow “REMAINDER” to function properly in expressions.
(Closes issue #16427. Reported, Patched by wdoekes.) - Shut down the SIP session timers more gracefully, in order to prevent a possible crash.
(Reported, Tested by corruptor. Patched by tilghman.) - Fix channel name comparison for Bridge() application.
(Closes issue #16528. Reported, Patched by telecos82.)
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.21-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.21
Thank you for your continued support of Asterisk!
Read the rest of this entry »
Asterisk 1.4.29 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.29.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
- Fix to Monitor which previously assumed the file to write to did not contain pathing.
(Closes issue #16377, #16376. Reported by bcnit. Patched by dant. - Propertly set T.38 attributes and don’t return before T.38 ports are configured when T.38 is found but no audio stream is found.
(Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) - Avoid crashes with large numbers of MeetMe conferences.
(Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) - Change in ‘sip show channels’ display format allowing more digits for CID.
(Closes issue #16459. Reported, Patched by Rzadzins. - Revise documentation on disposition values to the actual values used.
(Closes issue #16289. Reported by wdoekes.)
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.29
Thank you for your continued support of Asterisk!
Read the rest of this entry »
Asterisk Security Advisory – AST-2009-005: Remote Crash Vulnerability in SIP channel driver
Posted by admin in Asterisk Security Advisories, Security Advisories, asterisk, sip on August 11, 2009
Asterisk The Open Source PBX & Telephony Platform
Additionally note that while this can crash Asterisk, execution of arbitrary code is not possible with this vector.
Upgrade Asterisk to one of the releases listed below.
Product | Asterisk |
Summary | Remote Crash Vulnerability in SIP channel driver |
Nature of Advisory | Denial of Service |
Susceptibility | Remote Unauthenticated Sessions |
Severity | Critical in 1.6.1; minor in lesser versions |
Exploits Known | No |
Reported On | July 28, 2009 |
Reported By | Nick Baggott < nbaggott AT mudynamics DOT com > |
Posted On | August 10, 2009 |
Last Updated On | August 10, 2009 |
Advisory Contact | Tilghman Lesher < tlesher AT digium DOT com > |
CVE Name | CVE-2009-2726 |
Fax For Asterisk
T.38 fax for Asterisk
Digium‘s Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry’s leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium‘s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.
Fax For Asterisk provides two components: res_fax and res_fax_digium. Res_fax is an Asterisk resource module that adds fax termination and origination functionality in Asterisk. It provides Asterisk dialplan functions and dialplan applications to enable the user to build highly-customizable fax solutions. Res_fax_digium provides core fax processing functionality in the form of several supported fax modems — V.21, V.27ter, V.29, and V.17 — to achieve speeds up to 14400bps.
Asterisk says Hello to Fax
If you ask Google about faxing for Asterisk, with the search keywords asterisk and fax, and you ask Google to omit similar entries, you’ll end up with 52 pages of results.
If you ask Google how many times fax has been mentioned on an Asterisk mailing list, by setting the site parameter to lists.digium.com, then Google tells you there are 1120 utterances.
Yesterday, if you asked Digium for help in faxing documents through Asterisk, we’d have apologized and said that we didn’t offer a fax solution for Asterisk.
That was yesterday.
Today, Digium is pleased to announce Fax For Asterisk.
Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry’s leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium’s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.
Wait, I’ve forgotten something…okay, not really. There’s also Free Fax For Asterisk. Free Fax For Asterisk provides a single-channel only, per Asterisk, version of Fax For Asterisk, for free. Want to use Free Fax For Asterisk now? Visit the Digium webstore and get a license, free of charge.
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