search for: russellb

I'm no longer on the DEV mailing list, but: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk svn: URL 'http://svn.digium.com/svn/asterisk/branches/1.4' doesn't exist http://svn.digium.com/svn/asterisk/branches/ -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL:

...bout 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...

...bout 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...

...formation about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Resolve an issue with the Asterisk manager interface leaking memory when disabled. (Reported internally by kmorgan. Patched by russellb) * Support greetingsfolder as documented in voicemail.conf.sample. (Closes issue #17870. Reported by edhorton. Patched by seanbright) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb....

...formation about these changes. * Resolve deadlocks related to device states in chan_sip (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) * Resolve an issue with the Asterisk manager interface leaking memory when disabled. (Reported internally by kmorgan. Patched by russellb) * Support greetingsfolder as documented in voicemail.conf.sample. (Closes issue #17870. Reported by edhorton. Patched by seanbright) * Fix channel redirect out of MeetMe() and other issues with channel softhangup (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb....

...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...

...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...

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help maintain this code, please speak up. As it stands right now, we'll likely wait a week or two remove committing the patch. [1] - https://reviewboard.asterisk.org...

...lock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (Closes issue #18031. Reported by rain. Patched by bbryant) * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * 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. Reporte...

...lock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (Closes issue #18031. Reported by rain. Patched by bbryant) * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * 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. Reporte...

...equested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. (Patched by rmudgett) * Split _all_ arguments before parsing them. This fixes multicast RTP paging using linksys mode. (Patched by russellb) * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list the special CEL variables that are available for use in the mapping. There are also several other CEL fixes in this release. (Patched by russellb) Asterisk 1.8...

...equested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. (Patched by rmudgett) * Split _all_ arguments before parsing them. This fixes multicast RTP paging using linksys mode. (Patched by russellb) * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list the special CEL variables that are available for use in the mapping. There are also several other CEL fixes in this release. (Patched by russellb) Asterisk 1.8...

...y 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: * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * 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. Reporte...

...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.) * Re...

Asterisk Users Conference Fri 12:30 PM EDT ** see http://x2z.eu for instructions on how to join or listen RussellB from Digium and others will be around to tell us about the recent devcon and hopefully answer any questions about the near future of asterisk branches, what's new, etc. Hope you can make it! ** exten => AUC,1,Dial(SIP/123@66.212.134.192,60,D(22622#${YOUR_PIN}#))

...y 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: * Fix cache of device state changes for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * 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. Reporte...

...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.) * Re...

...7 09:59 AM, Deka, Rajib IN MAA SL wrote: > > Is the following trunk has development version of out-of-call messaging capability, also what is the version of asterisk, > http://svn.asterisk.org/svn/asterisk/trunk/ > I don't believe the branches has been merged into trunk, you can use russellb's branch [1]. [1] http://svn.digium.com/svn/asterisk/team/russell/messaging/ -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org ------------------------------ Important notice: Th...

...t(). 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 s...

...t(). 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 s...