Posts Tagged MOH
Asterisk 10.1.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 10.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 10.1.0 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:
- AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded.
(closes issue ASTERISK-19202) Reported by: Catalin Sanda - Allow playback of formats that don’t support seeking.ast_streamfile previously did unconditional seeking on files that broke playback of formats that don’t support that functionality. This patch avoids the seek that was causing the problem.
(closes issue ASTERISK-18994) Patched by: Timo Teras - Add pjmedia probation concepts to res_rtp_asterisk’s learning mode.In order to better handle RTP sources with strictrtp enabled (which is the default setting in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called ‘probation’. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source has been determined. - Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary. (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan
- Fix timing source dependency issues with MOH.Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patched by elguero - Fix RTP reference leak.If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali - Fix blind transfers from failing if an ‘h’ extension is present.This prevents the ‘h’ extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049)
- Restore call progress code for analog ports.Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller - Fix regression that ‘rtp/rtcp set debup ip’ only works when a port was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: Walter Doekes
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0
Thank you for your continued support of Asterisk!
Asterisk 1.8.9.0 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.9.0 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:
- AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded.
(closes issue ASTERISK-19202) Reported by: Catalin Sanda - Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop causes the loop to exit prematurely. This causes a variety of negative side effects, depending on when the loop exits. This patch handles the frame by essentially swallowing the frame in the local loop, as the current channel drivers expect the RTP bridge to handle the frame, and, in the case of the local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested by: Matt Jordan - Fix timing source dependency issues with MOH.Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero - Fix RTP reference leak.If a blind transfer were initiated using a REFER without a prior reINVITE to place the call on hold, AND if Asterisk were sending RTCP reports, then there was a reference leak for the RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali - Fix blind transfers from failing if an ‘h’ extension is present.This prevents the ‘h’ extension from being run on the transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049)
- Restore call progress code for analog ports.Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patched by Richard Miller
- Fix regression that ‘rtp/rtcp set debup ip’ only works when a port was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: Walter Doekes
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0
Thank you for your continued support of Asterisk!
Asterisk 1.8.3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3 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:
- Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman) - Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
(Closes issue #18464. Reported, patched by IgorG) - Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing.
(Closes issue #18350. Reported by gbour. Patched by Marquis) - When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman) - Resolve memory leak in iCalendar and Exchange calendaring modules.
(Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman) - Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) - Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb) - Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
(Reported internally. Patched by mnicholson) - Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - Resolve deadlock involving REFER.
(Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.17 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.2.17.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.17 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:
- Resolve duplicated data in the AstDB when using DIALGROUP()
(Closes issue #18091. Reported by bunny. Patched by tilghman) - Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman) - When using cdr_pgsql the billsec field was not populated correctly on unanswered calls.
(Closes issue #18406. Reported by joscas. Patched by tilghman) - Resolve issue where re-transmissions of SUBSCRIBE could break presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson) - Fix regression causing forwarding voicemails to not work with file storage.
(Closes issue #18358. Reported by cabal95. Patched by jpeeler) - This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems.
(Patched by tilghman) - Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes. - Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) - Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17
Thank you for your continued support of Asterisk!
Asterisk 1.8.3-rc2 Now Available
Posted by admin in asterisk, Release Candidates on January 26, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.3. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to those included in 1.8.3-rc1:
- Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) - Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb) - Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486.
(Reported internally. Patched by mnicholson)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc2
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.17-rc2 Now Available
Posted by admin in asterisk, Release Candidates on January 26, 2011
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the second release candidate of Asterisk 1.6.2.17. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.17-rc2 resolves the following issues in addition to those included in 1.6.2.17-rc1:
- Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo, shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes. - Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17-rc2
Thank you for your continued support of Asterisk!
Asterisk 1.4.39 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.39. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.39 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:
- Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos) - Fix bugs in saying numbers using the Swedish language syntax
(Closes issue #18355. Reported, patched by oej) - Fix not stopping MOH when transfered local channel queue member is answered.
The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm).
Patched by jpeeler. - Improve handling of REGISTER requests with multiple contact headers. Patched by jpeeler.
- app_followme: Don’t create a Local channel if the target extension does not exist.
(Closes issue #18126. Reported, patched by junky) - Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) - Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39
Thank you for your continued support of Asterisk!
Asterisk 1.4.39-rc1 Now Available
Posted by admin in asterisk, Release Candidates on December 15, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.4.39. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.39-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:
- Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos) - Fix bugs in saying numbers using the Swedish language syntax
(Closes issue #18355. Reported, patched by oej) - Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). Patched by jpeeler.
- Improve handling of REGISTER requests with multiple contact headers. Patched by jpeeler.
- app_followme: Don’t create a Local channel if the target extension does not exist.
(Closes issue #18126. Reported, patched by junky) - Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) - Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.14-rc1 Now Available
Posted by admin in asterisk, Release Candidates on September 21, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.6.2.14. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.14-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:
- Fix issue where session timers would be advertised as supported even when session-timers=refuse was set in sip.conf. Also fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel) - Fix issue with decoding ^-escaped characters in realtime (res_pgsql).
(Closes issue #17790. Reported by denzs. Patched by Qwell) - Parse all “Accept” headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel) - Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman) - Fix issue where MoH files were no longer rescanned on during a reload.
(Closes issue #16744. Reported by pj. Patched by Qwell) - Fix issue with dialplan pattern matching where the specificity for pattern ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.6.0.20 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.0.20.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.20 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
- clarify requirecalltoken option in iax.sample.conf (closes issue #16223), reported, patched by: bklang
- Prevent double closing of FDs by EIVR (closes issue #16305), reported by: diLLec, patched, tested by: thedavidfactor
- Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. (closes issues #16279, #16207), reported by: parisioa, dcabot, patched by: tilghman, tested by: parisioa, tilghman
- Send ack (response/message) after receiving manager action userevent (closes issue #16264), reported, patched by: dimas
- Make manager response to “Action: events” finish with empty line (closes issue #16275), reported, patched by: vnovy
This release also contains significant improvements to T.38 support. Anyone who has tried T.38 faxing in the past should try again as most problems should now be resolved.
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.20-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.20
Thank you for your continued support of Asterisk!
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