Re: [asterisk-dev] [Code Review] 3488: RAII_VAR: nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
On May 15, 2014, 4:50 a.m., Matt Jordan wrote: Hi Matt, Thank you for your review ! On May 15, 2014, 4:50 a.m., Matt Jordan wrote: /trunk/configure.ac, line 1092 https://reviewboard.asterisk.org/r/3488/diff/1/?file=57985#file57985line1092 red blob On May 15, 2014, 4:50 a.m., Matt Jordan wrote: /trunk/include/asterisk/utils.h, lines 989-993 https://reviewboard.asterisk.org/r/3488/diff/1/?file=57986#file57986line989 Coding style guidelines dictate not to use C++ style comments. That being said, since you've removed this, you should just delete the whole thing. Point taken. The patch i send in was meant as a proof of concept, not as a patch to be merged directly. I should have mentioned that in the description. Move changes will be necessary to make compiling using clang work correctly and nicely. I only wanted to show that the dependency on gcc nested fuctions should not preclude compiling with clang, where a block could be used instead. I remarked the original part to show how it could be replaced with a version that would work for both gcc and clang. - Diederik --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3488/#review11849 --- On May 15, 2014, 10:57 a.m., Diederik de Groot wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3488/ --- (Updated May 15, 2014, 10:57 a.m.) Review request for Asterisk Developers. Bugs: ASTERISK-20850 https://issues.asterisk.org/jira/browse/ASTERISK-20850 Repository: Asterisk Description --- RAII_VAR: nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. Making it possible again to use clang as a compiler, instead of depending on gcc completely. Compile instructions: ./bootstrap.sh CC=clang CXX=clang++ ./configure --enable-dev-mode Needed to set DISABLE_INLINE to get passed the double declaration error in api-inline.h, i guess someone needs to figure out how to get this passed clang, correctly make menuselect.makeopts menuselect/menuselect --enable DISABLE_INLINE Needed to suppresse some of the warnings to get clang to compile (for now), clang can be a little panicky, but for a good reason. -Wno-unknown-warning-option. When gcc doesn't know a compiler option it returns NON-ZERO errorlevel, clang returns ZERO errorlevel, which is annoying. Even the linux kernel developers group complained about this. Will be fixed/changed (hopefully soon). For now, when checking clang compiler options, you would need to grep and parse the error output -Wno-error needed to quite down clang being panicky (Standard asterisk -Werror is a good idea in general, but not when compiling against a 'new' compiler ) ASTCFLAGS=-Wno-unknown-warning-option -Wno-error make make install RAII_VAR seems to work, but i guess there is still a bit of work before clang support for the rest of asterisk can be announced. Diffs - /trunk/makeopts.in 413043 /trunk/main/Makefile 413043 /trunk/include/asterisk/utils.h 413043 /trunk/configure.ac 413043 /trunk/configure UNKNOWN /trunk/Makefile 413043 Diff: https://reviewboard.asterisk.org/r/3488/diff/ Testing --- Just a proof of concept, to show how asterisk could be made clang compatible again. Thanks, Diederik de Groot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3488: RAII_VAR: nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3488/ --- (Updated May 15, 2014, 10:57 a.m.) Review request for Asterisk Developers. Changes --- Fixes based on Matt Jordan's comments Bugs: ASTERISK-20850 https://issues.asterisk.org/jira/browse/ASTERISK-20850 Repository: Asterisk Description --- RAII_VAR: nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. Making it possible again to use clang as a compiler, instead of depending on gcc completely. Compile instructions: ./bootstrap.sh CC=clang CXX=clang++ ./configure --enable-dev-mode Needed to set DISABLE_INLINE to get passed the double declaration error in api-inline.h, i guess someone needs to figure out how to get this passed clang, correctly make menuselect.makeopts menuselect/menuselect --enable DISABLE_INLINE Needed to suppresse some of the warnings to get clang to compile (for now), clang can be a little panicky, but for a good reason. -Wno-unknown-warning-option. When gcc doesn't know a compiler option it returns NON-ZERO errorlevel, clang returns ZERO errorlevel, which is annoying. Even the linux kernel developers group complained about this. Will be fixed/changed (hopefully soon). For now, when checking clang compiler options, you would need to grep and parse the error output -Wno-error needed to quite down clang being panicky (Standard asterisk -Werror is a good idea in general, but not when compiling against a 'new' compiler ) ASTCFLAGS=-Wno-unknown-warning-option -Wno-error make make install RAII_VAR seems to work, but i guess there is still a bit of work before clang support for the rest of asterisk can be announced. Diffs (updated) - /trunk/makeopts.in 413043 /trunk/main/Makefile 413043 /trunk/include/asterisk/utils.h 413043 /trunk/configure.ac 413043 /trunk/configure UNKNOWN /trunk/Makefile 413043 Diff: https://reviewboard.asterisk.org/r/3488/diff/ Testing (updated) --- Just a proof of concept, to show how asterisk could be made clang compatible again. Thanks, Diederik de Groot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Asterisk CALLINGTON for SS7
Good morning, I have a problem with that variable: I use asterisk 11 + dahdi 2.7.0.1 + libss7 1.0.2 + libpri 1.4 in another box (with pri cards) I use CALLINGTON. But I recently found out that this variable is not populated for SS7. My question is: there's another variable I can use? I couldn't find anything in documentation/source. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3538: Partial fix to voicemail number maxmsg being overwritten.
On May 14, 2014, 10:49 p.m., Matt Jordan wrote: /trunk/apps/app_voicemail.c, lines 4458-4463 https://reviewboard.asterisk.org/r/3538/diff/1/?file=58435#file58435line4458 This change avoids the problem you've discovered by completely discarding the maxmsg limit set for the voicemail user. That doesn't feel like the correct solution here. (As an aside, I think your editor is inserting spaces instead of tabs - you may want to review the Coding Guidelines on the wiki) I have been commenting on the issue tracker in regards to this. I have not been able to re-produce the issue and have been asking for more details on how to reproduce the described issue. - Michael --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3538/#review11891 --- On May 13, 2014, 3:30 p.m., Miguel Tavares wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3538/ --- (Updated May 13, 2014, 3:30 p.m.) Review request for Asterisk Developers. Bugs: ASTERISK-23713 https://issues.asterisk.org/jira/browse/ASTERISK-23713 Repository: Asterisk Description --- Although it doesn't solve the issue 23713 it makes it a lot less likely to occur. Another, maybe better, option might be to find the fist empty voicemail slot and use it to store the new message. This means changing the logic of how the messages are handled and as such as to be written and tested with more care. A brief description of the problem. Setting the maxmsg to 3 (so vmu-maxmsg = 3). When the user has 3 messages he/she deletes the first two. In the file system we would have the files msg0003.wav and msg0003.txt. If a new message is being delivered then the last_message_index will return 2 and the message will be left in the slot 3, meaning that the previous msg0003 will be overwritten. Diffs - /trunk/apps/app_voicemail.c 413892 Diff: https://reviewboard.asterisk.org/r/3538/diff/ Testing --- Thanks, Miguel Tavares -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3540: chan_local+app_dial: Propagagate call answered elsewhere over local channels.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3540/ --- (Updated May 15, 2014, 10:32 a.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 413949 Repository: Asterisk Description --- When dialing SIP/account_a + SIP/account_b, and account_b picks up, chan_sip sends out a Reason header with SIP;cause=200;text=Call completed elsewhere, signifying that the call was picked up. The SIP phone then does not show 1 missed call. However, then dialing Local/account_a + Local/account_b, this does not work. This review addresses that. When hanging up obsolete channels in chan_local, the answered_elsewhere flag is propagated to cancelled (parent) channel using the hangupcause. In app_dial, this hangupcause is checked and passed down to the other calls to be cancelled. Diffs - /branches/1.8/channels/chan_local.c 413892 /branches/1.8/apps/app_dial.c 413892 Diff: https://reviewboard.asterisk.org/r/3540/diff/ Testing --- Dialplan: [default] exten = 201,1,Dial(SIP/account_bSIP/account_c,5) exten = 202,1,Dial(Local/b@dialLocal/c@dial,5) ;; also tested with /n for no-local-channel-optimization, behaves the same as without [dial] exten = b,1,Dial(SIP/account_b) exten = c,1,Dial(SIP/account_c) sip.conf held 3 accounts: account_a, account_b and account_c. Before patch: 201 202 -- account_a calls these +---+---+ timeout | 1 missed call | 1 missed call | +---+---+ account_b | | 1 missed call | -- account_c sees these picks up +---+---+ After patch: 201 202 -- account_a calls these +---+---+ timeout | 1 missed call | 1 missed call | +---+---+ account_b | | | -- account_c sees these picks up +---+---+ Thanks, wdoekes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3535: bridge_native_rtp: Reconfigure bridge on removal of framehook; don't send re-INVITE to hungup channel
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3535/#review11894 --- Ship it! Minor optimization nit. /branches/12/main/channel.c https://reviewboard.asterisk.org/r/3535/#comment21779 Rather than calling the accessor funcition three times and potentially retrieving different values. Save it to a local variable and then test. flags = ast_channel_softhangup_internal_flag(chan); return flags == asyncgoto || flags == unbridge || flags == (both) - rmudgett On May 14, 2014, 9:14 p.m., Matt Jordan wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3535/ --- (Updated May 14, 2014, 9:14 p.m.) Review request for Asterisk Developers and Joshua Colp. Repository: Asterisk Description --- This patch fixes the currently failing pjsip/transfers/blind_transfer/caller_direct_media test (with a few small tweaks to the test as well). The test currently fails primarily for two reasons: (1) When Bob and Charlie (the transfer target and the transfer destination) enter a bridge together, the framehook remains on the transfer target channel until both channels are in the bridge. As it consumes voice frames, the initial bridge type is a simple bridge. The framehook is removed when both channels are in the bridge; however, this does not currently cause the bridging framework to re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a framehook is removed so the bridge can re-evaluate itself. (2) When a channel leaves a native RTP bridge, it may be leaving due to being hung up. Sending a re-INVITE to a channel that is about to be hung up is not nice - in fact, there's a good chance we'll send the BYE request before the channel has had a chance to send back a 200 OK. To be somewhat nicer, this patch makes it so that we only send the re-INVITE if there's a chance the channel will survive the native bridging experience. Diffs - /branches/12/res/res_pjsip_session.c 413948 /branches/12/main/framehook.c 413948 /branches/12/main/channel.c 413948 /branches/12/main/bridge_channel.c 413948 /branches/12/include/asterisk/channel.h 413948 /branches/12/bridges/bridge_native_rtp.c 413948 Diff: https://reviewboard.asterisk.org/r/3535/diff/ Testing --- Once some timing issues were removed from the test, it passes with this patch. Thanks, Matt Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Digium TE820 on BSD: bunch of missed interrupts
On Wed, May 14, 2014 at 1:40 PM, Łukasz Wójcik lukasz.woj...@zoho.comwrote: That's what I just did. It seems that irq handling routine takes 260-280ms during span setup. My measurements indicate that the most time is being spent in '__t4_set_timing_source_auto()' This is curious. There is no blocking in that function and it's just a small number of bit checks and two gpio writes. (..) Missed interrupt. Expected ident of 97 and got ident of 111 maany similar as above, differing with idents (..) It might be helpful to see the full output of the kernel log while the wct4xxp module is starting up. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk CALLINGTON for SS7
On Thu, May 15, 2014 at 9:45 AM, Alberto Rinaudo alberto.rina...@gmail.comwrote: Good morning, I have a problem with that variable: I use asterisk 11 + dahdi 2.7.0.1 + libss7 1.0.2 + libpri 1.4 in another box (with pri cards) I use CALLINGTON. But I recently found out that this variable is not populated for SS7. My question is: there's another variable I can use? I couldn't find anything in documentation/source. Thanks. That is the correct variable for the information. You could also use CALLERID(num-plan) or CALLERID(ton). Unfortunately that information is not extracted for SS7. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk CALLINGTON for SS7
If I'm right CALLINGTON is populated from a variable caller.id.number.plan that is equals to p-cid_ton I was comparing sig_pri.c and sig_ss7.c using asterisk 11.9 in sig_pri.c 279 caller.id.number.plan = p-cid_ton; 4210 pri-pvts[chanpos]-cid_ton = ast_connected.id.number.plan; 6263 pri-pvts[chanpos]-cid_ton = e-ring.callingplan; /* this is the callingplan (TON/NPI), e-ring.callingplan4 would be the TON */ this variable cid_ton is set equals to the number plan or the calling plan, depends on the flow. while in sig_ss7.c 192 caller.id.number.plan = p-cid_ton; 1071 p-cid_ton = 0; cid_ton is always 0, but in ss7 there are variables containing the number plan: sig_ss7.h 218 unsigned char gen_add_num_plan; so, what does this variable contains? it isn't a compatible value? *Alberto Rinaudo* *http://a.rinaudo.net http://a.rinaudo.net+44 (0) 741 496 5674 alberto.rina...@gmail.com alberto.rina...@gmail.comskype: a.rinaudo* On 15 May 2014 17:10, Richard Mudgett rmudg...@digium.com wrote: On Thu, May 15, 2014 at 9:45 AM, Alberto Rinaudo alberto.rina...@gmail.com wrote: Good morning, I have a problem with that variable: I use asterisk 11 + dahdi 2.7.0.1 + libss7 1.0.2 + libpri 1.4 in another box (with pri cards) I use CALLINGTON. But I recently found out that this variable is not populated for SS7. My question is: there's another variable I can use? I couldn't find anything in documentation/source. Thanks. That is the correct variable for the information. You could also use CALLERID(num-plan) or CALLERID(ton). Unfortunately that information is not extracted for SS7. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Segmentation fault error?
hello, I am trying to lanch asterisk in opnerisc 1200 processor in his simulator or1ksim, - Cross-compile install glibc [*] - Cross-compile install zlib[*] - Cross-compile install ncurses 5.9 [*] - Cross-compile install openssl [*] - Cross-compile install asterisk[*] - Configure linux + asterisk on it [*] - Running asterisk [X] when I tried to run asterisk as asterik -cvvv or with c option only I get a Segmentation faul error with c and g option Segmentation faul (core dump) error. I don't get it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk CALLINGTON for SS7
On Thu, May 15, 2014 at 11:56 AM, Alberto Rinaudo alberto.rina...@gmail.com wrote: If I'm right CALLINGTON is populated from a variable caller.id.number.plan that is equals to p-cid_ton I was comparing sig_pri.c and sig_ss7.c using asterisk 11.9 in sig_pri.c 279 caller.id.number.plan = p-cid_ton; 4210 pri-pvts[chanpos]-cid_ton = ast_connected.id.number.plan; 6263 pri-pvts[chanpos]-cid_ton = e-ring.callingplan; /* this is the callingplan (TON/NPI), e-ring.callingplan4 would be the TON */ this variable cid_ton is set equals to the number plan or the calling plan, depends on the flow. while in sig_ss7.c 192 caller.id.number.plan = p-cid_ton; 1071 p-cid_ton = 0; You want to use e-iam.calling_nai to assign to p-cid_ton. However, SS7 uses different code values so you will have to convert to the ISDN equivalent value that gets put there. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3541: res_http_websocket: Create a websocket client
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3541/ --- Review request for Asterisk Developers and Joshua Colp. Bugs: ASTERISK-23742 https://issues.asterisk.org/jira/browse/ASTERISK-23742 Repository: Asterisk Description --- Add client websocket capabilities to Asterisk. Diffs - trunk/tests/test_websocket_client.c PRE-CREATION trunk/res/res_http_websocket.exports.in 413541 trunk/res/res_http_websocket.c 413541 trunk/main/http.c 413541 trunk/include/asterisk/http_websocket.h 413541 trunk/include/asterisk/http.h 413541 Diff: https://reviewboard.asterisk.org/r/3541/diff/ Testing --- Created some unit tests. Thanks, Kevin Harwell -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3420: Testsuite: Call Files' Max Retries
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3420/ --- (Updated May 15, 2014, 1:11 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 5043 Bugs: ASTERISK-23218 https://issues.asterisk.org/jira/browse/ASTERISK-23218 Repository: testsuite Description --- These tests involved checking that call files max retries are functioning as planned through four tests: 1) The first test (call_file_retries_fail) required that the call file originates a local channel to a dialplan extension that will always fail, and checks to make sure that it ran through all of its max retries. 2) The second test (call_file_retries_success) involves a call file that originates a local channel that will fail once, but then is answered before it hits its max retries. 3) The third test (call_file_retries_alwaysdelete) consists of checking whether or not the call file was deleted from the [astspooldir]'s outgoing folder when the alwaysdelete option is set to 'no'. 4) The fourth and final test (call_file_retries_archive) consists of checking whether or not the call file was placed in [astspooldir]'s outgoing_done folder when archive is set to 'yes'. Diffs - ./asterisk/trunk/tests/pbx/tests.yaml 4990 ./asterisk/trunk/tests/pbx/call_file_retries_success/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_success/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/retries_fail.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_fail/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/retries_archive.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_archive/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/test-config.yaml PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/retries_alwaysdelete.py PRE-CREATION ./asterisk/trunk/tests/pbx/call_file_retries_alwaysdelete/configs/ast1/extensions.conf PRE-CREATION ./asterisk/trunk/lib/python/asterisk/pluggable_modules.py 4990 Diff: https://reviewboard.asterisk.org/r/3420/diff/ Testing --- Thanks, Scott Emidy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- (Updated May 15, 2014, 9:13 p.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- Review request for Asterisk Developers. Bugs: ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3542: Documentation: Wiki page for Maintenance and Upgrades, including sub pages.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/#review11896 --- In updating or upgrading asterisk, you use the terms first, then later explain what they mean. It would be better to define the terms first, then use them. - Scott Griepentrog On May 15, 2014, 4:13 p.m., rnewton wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3542/ --- (Updated May 15, 2014, 4:13 p.m.) Review request for Asterisk Developers. Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-23396 None Description --- We needed a wiki page that at least gives an overview of tasks associated with maintaining an Asterisk system. Such as backups, updating or upgrading and other miscellaneous maintenance related tasks. I wrote a few pages to get this started. Please give it a quick read and a ship it, if it looks good. I'm sure I said something dumb or inaccurate somewhere. :) https://wiki.asterisk.org/wiki/display/AST/Maintenance+and+Upgrades Plus two sub-pages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Backups https://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk I'd like to know about: * Things to expand on * Things I said too much about or went out of scope on * Things missing If you already have wiki edit access feel free to edit typos/logic errors straight-away, otherwise just report them on here. Diffs - Diff: https://reviewboard.asterisk.org/r/3542/diff/ Testing --- Thanks, rnewton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3543: app_meetme: Don't interrupt MOH on waitmarked users.
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3543/ --- Review request for Asterisk Developers. Bugs: AST-1349 https://issues.asterisk.org/jira/browse/AST-1349 Repository: Asterisk Description --- Occasionally, when the last marked user leaves the conference, waitmarked users don't get MOH if MOH is supposed to be played while a waitmarked user is waiting for another marked user. * Made not interrupt MOH when the user is a waitmarked user. The waitmarked user doesn't need to hear any leave announcements from the conference as the user would have already heard different leave announcements if they were enabled. Apparently DAHDI occasionally sends unending non-silent streams to these users or a normal user still in the conference has continuous high background noise. These non-silent streams cause MOH to be suspended while the never ending announcement is played. Issue caused by ASTERISK-13680. Diffs - /branches/1.8/apps/app_meetme.c 413994 Diff: https://reviewboard.asterisk.org/r/3543/diff/ Testing --- Added debugging statements to indicate when the MOH would have been suspended on the waitmarked user. Got confirmation that the waitmerked user would not be suspended when the marked user leaves the conference. Thanks, rmudgett -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Segmentation fault error?
On Thu, May 15, 2014 at 06:08:27PM +0100, Mohammed Essaid Mezerreg wrote: hello, I am trying to lanch asterisk in opnerisc 1200 processor in his simulator or1ksim, - Cross-compile install glibc [*] - Cross-compile install zlib[*] - Cross-compile install ncurses 5.9 [*] - Cross-compile install openssl [*] - Cross-compile install asterisk[*] - Configure linux + asterisk on it [*] - Running asterisk [X] when I tried to run asterisk as asterik -cvvv or with c option only I get a Segmentation faul error with c and g option Segmentation faul (core dump) error. I don't get it? You have a core dump. Where did it crash? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
