Posts Tagged chan_sip
Asterisk 1.8.8.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.8.0 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:
- Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required.
(Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/) - Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels.
(Closes issue ASTERISK-18090. Patched by Kinsey Moore) - Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
(Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose) - Fix crashes in ast_rtcp_write()
(Closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
Patched by: Russell Bryant - Fix for incorrect voicemail duration in external notifications.
This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration.
(Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443) - Prevent segfault if call arrives before Asterisk is fully booted.
(Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/) - Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
http://downloads.asterisk.org/pub/security/AST-2011-012.pdf
- Fix locking order in app_queue.c which caused deadlocks
(Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
(Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky) - Fix regression in configure script for libpri capability checks
(Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett) - Prevent BLF subscriptions from causing deadlocks.
(Closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/ - Fix deadlock if peer is destroyed while sending MWI notice.
(Closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky - Fix issue with setting defaultenabled on categories that are already enabled by default.
(Closes issue ASTERISK-18738)
Reported by: Paul Belanger - Don’t crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.
- Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
- Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 10.0.0-rc3 Now Available
Posted by admin in asterisk, Release Candidates on December 12, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the third release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 10.0.0-rc3 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 candidate:
- Add ASTSBINDIR to the list of configurable pathsThis patch also makes astdb2sqlite3 and astcanary use the configured directory instead of relying on $PATH.
- Don’t crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. - Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. - Fix a change in behavior in ‘database show’ from 1.8.In 1.8 and previous versions, one could use any fullword portion of the key name, including the full key, to obtain the record. Until this patch, this did not work for the full key.
- Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and
1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no.In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible. - Fixed SendMessage stripping extension from To: header in SIP MESSAGEWhen using the MessageSend application to send a SIP MESSAGE to a non-peer, chan_sip stripped off the extension and failed to add it back to the sip_pvt structure before transmitting. This patch adds the full URI passed in from the message core to the sip_pvt structure.
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.8.8.0-rc5 Now Available
Posted by admin in asterisk, Release Candidates on December 9, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the fifth release candidate of Asterisk 1.8.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.8.0-rc5 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 candidate:
- Don’t crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet. - Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. - Default to nat=yes; warn when nat in general and peer differ
AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header. In 1.4 and 1.6.2, this would mean if one setting was nat=yes or nat=route and the other was either nat=no or nat=never. In 1.8 and 10, this would mean when one was nat=force_rport and the other was nat=no.In order to address this problem, it was decided to switch the default behavior to nat=yes/force_rport as it is the most commonly used option and to strongly discourage setting nat per-peer/user when at all possible.
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.18-rc1 Now Available
Posted by admin in asterisk, Release Candidates, sip on February 28, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.6.2.18. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.18-rc1 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 candidate:
- Only offer codecs both sides support for directmedia.
(Closes issue #17403. Reported, patched by one47) - Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)
NOTE: Be sure to read the ChangeLog for more information about these changes. - Resolve deadlocks related to device states in chan_sip
(Closes issue #18310. Reported, patched by one47. Patched by jpeeler) - Fix channel redirect out of MeetMe() and other issues with channel softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) - Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) - Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip.
(Review: https://reviewboard.asterisk.org/r/1077/)
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.4.41-rc1 Now Available
Posted by admin in asterisk, Release Candidates on February 28, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.4.41. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.41-rc1 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 candidate:
- Only offer codecs both sides support for directmedia.
(Closes issue #17403. Reported, patched by one47) - Resolution of several DTMF based attended transfer issues.
(Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchuan, grecco. Patched by rmudgett)
NOTE: Be sure to read the ChangeLog for more information about these changes. - Fix channel redirect out of MeetMe() and other issues with channel softhangup
(Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. Patched by russellb) - Fix voicemail sequencing for file based storage.
(Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) - Guard against retransmitting BYEs indefinitely during attended transfers with chan_sip.
(Review: https://reviewboard.asterisk.org/r/1077/)
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.4.27, Asterisk 1.6.0.18 and Asterisk 1.6.1.10 released
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases resolve a large assortment of issues reported by the community.
For a summary of the changes in these releases, please see the release summaries:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.18-summary.html
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.10-summary.html
For a full list of changes in these releases, please see the ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.27
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.0.18
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.10
The following non-exhaustive list of issues were resolved with the participation of the community, and this release would not have been possible without your help!
- Seg fault in chan_local – local_pvt_destroy
(closes issue #15314. Reported by sroberts. Tested by davidw, lottc. Patch by davidw.) - T.38 reinvite started from Asterisk
(closes issue #15373. Reported by dcolombo. Tested by dcolombo, mbrancaleoni. Patch by mbrancaleoni.) - manager keeps creating /tmp/ast-ami-XXXXXX files (without deleting) when a single manager client remains logged in
(closes issue #15730. Reported by zmehmood. Tested by zmehmood. Patch by junky.) - BASE64_DECODE() adds garbage end end of decoded string
(closes issue #15271. Reported by chappell. Tested by kobaz. Patch by chappell.) - Fix ExternalIVR Documentation in 1.4
(closes issue #16220. Reported and patched by thedavidfactor.)
Thank you for your continued support of Asterisk!
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