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Hi I am new to writing C code and am trying to write an R extension in C. I have hit a wall with F77_CALL(dgemm) in that it produces wrong results. The code below is a simplified example that multiplies the matrices Ab and Bm to give Cm. The results below show clearly that Cm is wrong. Am= 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 Bm= 1 1 1 1 1

...------------------- * ASTERISK-24032 - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined (Reported by Kilburn) * ASTERISK-24225 - Dial option z is broken (Reported by dimitripietro) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-24019 - When a Music On Hold stream starts it restarts at beginning of file. (Reported by Jason Richards) * ASTERISK-24211 - testsuite: Fix the dial_LS_options test (Reported by Matt Jordan) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod)...

...------------------- * ASTERISK-24032 - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined (Reported by Kilburn) * ASTERISK-24225 - Dial option z is broken (Reported by dimitripietro) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer (Reported by Badalian Vyacheslav) * ASTERISK-24019 - When a Music On...

...------------------- * ASTERISK-24032 - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined (Reported by Kilburn) * ASTERISK-24225 - Dial option z is broken (Reported by dimitripietro) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-24019 - When a Music On Hold stream starts it restarts at beginning of file. (Reported by Jason Richards) * ASTERISK-24211 - testsuite: Fix the dial_LS_options test (Reported by Matt Jordan) * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash Mohod)...

...------------------- * ASTERISK-24032 - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined (Reported by Kilburn) * ASTERISK-24225 - Dial option z is broken (Reported by dimitripietro) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer (Reported by Badalian Vyacheslav) * ASTERISK-24019 - When a Music On...

...s_rtp_asterisk: ICE role message even when ICE is not enabled (Reported by Joshua Colp) * ASTERISK-25427 - Callerid change does not always emit NewCallerid AMI event (Reported by Ivan Poddubny) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25410 - app_record: RECORDED_FILE variable not being populated (Reported by Kevin Harwell) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25396 - chan_sip: Extremely long callerid name caus...

...Badalian Vyacheslav) * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set (Reported by Jaco Kroon) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton) Improvements made in this release: ----------------------------------- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad de Wet) For a full list of changes in this releas...

...ns of files that do not end with newlines (Reported by Shaun Ruffell) * ASTERISK-24229 - ARI: playback of sounds implicitly answers channel, preventing early media playback (Reported by Matt Jordan) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname (Reported by Private Name) * ASTERISK-24147 - ARI: channel hangup c...

...ns of files that do not end with newlines (Reported by Shaun Ruffell) * ASTERISK-24229 - ARI: playback of sounds implicitly answers channel, preventing early media playback (Reported by Matt Jordan) * ASTERISK-24178 - [patch]fromdomainport used even if not set (Reported by Elazar Broad) * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks (Reported by Walter Doekes) * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname (Reported by Private Name) * ASTERISK-24147 - ARI: channel hangup c...

...posed to intersection of configured codecs and offered codecs (Reported by Taylor Hawkes) * ASTERISK-25825 - Crashes during shutdown when running CLI commands (Reported by Mark Michelson) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton) * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo) Improvements made in this release: ----------------------------------- * ASTERISK-25865 -...

...bridge appear to still be on hold (Reported by Mark Michelson) * ASTERISK-25384 - Regular Asterisk crashes when using Page application. "user_data is NULL" (Reported by Chet Stevens) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25410 - app_record: RECORDED_FILE variable not being populated (Reported by Kevin Harwell) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25396 - chan_sip: Extremely long callerid name caus...

...ERISK-24543 - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs (Reported by Taylor Hawkes) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton) * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo) * ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops audio (Rep...

...Configuration parser handles unsigned integers as signed integers (Reported by Gianluca Merlo) * ASTERISK-25825 - Crashes during shutdown when running CLI commands (Reported by Mark Michelson) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton) * ASTERISK-21301 - ERROR and failure to resolve socket address due to whitespace after port number in SIP Via header (Reported by Martin Vit) * ASTERISK-25857 - func_aes: incorrect use of strlen...

...bridge appear to still be on hold (Reported by Mark Michelson) * ASTERISK-25384 - Regular Asterisk crashes when using Page application. "user_data is NULL" (Reported by Chet Stevens) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25410 - app_record: RECORDED_FILE variable not being populated (Reported by Kevin Harwell) * ASTERISK-25396 - chan_sip: Extremely long callerid name causes invalid SIP (Reported by Walter Doekes) * ASTERISK-25399 - app_queue: AgentComplete event has wrong reason...

...terisk.org/jira/browse/ASTERISK-25612>] - Configuration parser handles unsigned integers as signed integers (Reported by Gianluca Merlo) - [ASTERISK-25407 <https://issues.asterisk.org/jira/browse/ASTERISK-25407>] - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) - [ASTERISK-25510 <https://issues.asterisk.org/jira/browse/ASTERISK-25510>] - [patch]Log to syslog failing (Reported by Michael Newton) - [ASTERISK-21301 <https://issues.asterisk.org/jira/browse/ASTERISK-21301>] - ERROR and failure to resolve socket address due to...