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hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

"Security only fixes: 2011-04-21" So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan

Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN & mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI & libpri but reading the source of the

...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. Re...

...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. Re...

...e -c flag when using $(INSTALL) (Closes issue #17695. Reported, patched by pabelanger) * Don't re-register CDR module on reload. (Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman) * Don't assume qlog is open. (Closes issue #17704. Reported, tested by vrban. Patched by pabelanger) * Expand the correct value within AST_OPTION_ONLY. (Closes issue #17703. Reported by stuarth. Patched by seanbright) * Allow for systems without locale support to be usable. (Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson)...

Hello, what about: "This feature means you can restart Asterisk after a failure (or asterisk restart itself with safe_asterisk), and keep existing calls up with only a few seconds of audio dropped." That would be a feature! there is a "other" pbx that has this feature... Anyone else would like to see that feature? just want to start some brainstorming, Kristijan

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

...k.org/svn/asterisk/tags/1.6.1.5/ChangeLog The following list of issues were resolved with the participation of the community, and these releases would not have been possible without your help! * Fix SIP transport type issues. (closes issue #13865. Reported by st. Tested by mmichelson, Kristijan, vrban, jmacz, dvossel. Patched by: mmichelson, vrban, Kristijan) * Fix an issue where the 'h' extension may occasionally not fire when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. Fixed by Tilghman. * Fix MWI NOTIFY if Asterisk listens on a non-standard port...

...k.org/svn/asterisk/tags/1.6.1.5/ChangeLog The following list of issues were resolved with the participation of the community, and these releases would not have been possible without your help! * Fix SIP transport type issues. (closes issue #13865. Reported by st. Tested by mmichelson, Kristijan, vrban, jmacz, dvossel. Patched by: mmichelson, vrban, Kristijan) * Fix an issue where the 'h' extension may occasionally not fire when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. Fixed by Tilghman. * Fix MWI NOTIFY if Asterisk listens on a non-standard port...

...riding options. (Closes issue #16855. Reported by davidw) * Modify queued frames from Local channels to not set the other side to up. (Closes issue #16816. Reported, tested by jamhed) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.18 Thank you for your continued support of Asterisk!

...s issue #16921. Reported, tested by whardier. Patched by seanbright) * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts (Reported, Patched, and Tested by alecdavis) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) * Fix ConfBridge crash when no timing module is loaded. (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6 Thank you for y...

...utter and dialrecall (Closes issue #17359. Reported, patched by alecdavis) * Make AgentComplete message more consistent (Closes issue #15638. Reported, patched by elbriga) * Missing fallback to audio fax feature when T.38 re-INVITE failed (Closes issue #16692. Reported, patched by vrban) * Don't hang up on a queue caller if the file we attempt to play does not exist (Closes issue #17061. Reported by RoadKill) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33 Thank you for...

...#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. * 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/telep...

...ic 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/telep...

...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.pd...

...by zvision) * Resolve issue where voice frames could be dropped when checking for T.38 during early media. (Closes issue ASTERISK-17705. Reported, patched by oej) * Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix. (Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson) Additionally security announcements AST-2011-010, and AST-2011-011 have been resolved in this release. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.42 Thank you for your continued su...

...riding options. (Closes issue #16855. Reported by davidw) * Modify queued frames from Local channels to not set the other side to up. (Closes issue #16816. Reported, tested by jamhed) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.18 Thank you for your continued support of Asterisk!

...s issue #16921. Reported, tested by whardier. Patched by seanbright) * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts (Reported, Patched, and Tested by alecdavis) * For T.38 reINVITEs treat a 606 the same as a 488. (Closes issue #16792. Reported, patched by vrban) * Fix ConfBridge crash when no timing module is loaded. (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6 Thank you for y...