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Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn?t work and i was told that 1.6 can?t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best

...tamp: [branch trunk] 162350 > Blamelist: aaronballman,ahatanak,alexfh,anithab,asl,baldrick,bwilson,chandlerc,chapuni,criswell,ctopper,d0k,dblaikie,ddunbar,delesley,dgregor,djasper,dyatkovskiy,echristo,efriedma,evancheng,fjahanian,gps,gribozavr,grosbach,grosser,hanm,hans,hliao,jacksprat,jaykang10,jrose,jush,kcc,klimek,kremenek,matthewbg,mcrosier,mspencer,nadav,nicholas,panzer,pranavb,rafael,rdivacky,resistor,rjmccall,rsmith,rtrieu,spop,spyffe,stoklund,tbrethou,tnorthover,victorm,void,zaks > > BUILD FAILED: failed clang-test > > sincerely, > -The Buildbot > > >

...lokok,cdavis,chandlerc,chapuni,compnerd,ctopper,d0k,daenzer,dblaikie,ddunbar,delena,delesley,dgregor,dirty,djasper,dsanders,dvyukov,echristo,efriedma,enderby,enea,eugenis,faisalv,fangism,filcab,fjahanian,gribozavr,grosbach,hanm,hans,haoliu,hfinkel,jacksprat,jdennett,jholewinski,jimcownie,joerg,joey,jrose,jtsoftware,kevinqin,klimek,kremenek,kuba,labath,lattner,lhames,logan,lvoufo,majnemer,mgottesman,mkuper,mpopa,mren,mspencer,nadav,nicholas,nico,olesalscheider,papin_g,pcc,predmond,rafael,rdivacky,redstar,rengolin,resistor,revane,ributzka,rikka,rkotler,rlytton,rmitton,rnk,row,rsandifo,rsmith,rtrieu,r...

...at URI we accept for call completion PUBLISH requests. (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (Closes issue #18070. Reported by mav3rick. Patched by bbryant) * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the ca...

...at URI we accept for call completion PUBLISH requests. (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (Closes issue #18070. Reported by mav3rick. Patched by bbryant) * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the ca...

...Reported By Corey Farrell Posted On June 12, 2014 Last Updated On June 12, 2014 Advisory Contact Jonathan Rose < jrose AT digium DOT com > CVE Name CVE-2014-4046 Description Manager users can execute arbitrary shell commands with the MixMonitor manager action. Asterisk does not require system class auth...

...Reported By Jonathan Rose Posted On 20 November, 2014 Last Updated On November 20, 2014 Advisory Contact Jonathan Rose <jrose AT digium DOT com> CVE Name Pending Description The Asterisk module res_pjsip_acl provides the ability to configure ACLs that may be used to reject SIP requests from various...

...Reported By Corey Farrell Posted On June 12, 2014 Last Updated On June 12, 2014 Advisory Contact Jonathan Rose < jrose AT digium DOT com > CVE Name CVE-2014-4046 Description Manager users can execute arbitrary shell commands with the MixMonitor manager action. Asterisk does not require system class auth...

...Reported By Jonathan Rose Posted On 20 November, 2014 Last Updated On November 20, 2014 Advisory Contact Jonathan Rose <jrose AT digium DOT com> CVE Name Pending Description The Asterisk module res_pjsip_acl provides the ability to configure ACLs that may be used to reject SIP requests from various...

...Reported By Maciej Szmigiero Posted On March 04, 2015 Last Updated On April 8, 2015 Advisory Contact Jonathan Rose <jrose AT digium DOT com> CVE Name CVE-2015-3008 Description When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a commo...

...Reported By Maciej Szmigiero Posted On March 04, 2015 Last Updated On April 8, 2015 Advisory Contact Jonathan Rose <jrose AT digium DOT com> CVE Name CVE-2015-3008 Description When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a commo...

...s. (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/14...

...s. (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/14...

Hello, I'm running Asterisk 1.8.10 on Linux box, when I'm in a conference(meetme) with another person, and a third person join our conference when the third person leave the conference I get disconnected from the original conference with a second party. I hope this clear. This does not happen often, is random, anybody experience something similar? or any idea how to fix this problem?

Hello, is there anywhere an overview of SIP error codes and under which condition they are reported by Asterisk? There are general definitions for SIP error codes, but they are quite general and it's Asterisk that actually checks what's wrong and then reports an error. Now, currently I could check the source code to get more informations what could have caused the error, but that's