| 00:21.31 | tompaw | Day 4. I am still struggling with the ARI hangup of a Local channel ( https://issues.asterisk.org/jira/browse/ASTERISK-26716 ) |
| 00:35.35 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 00:38.36 | tompaw | I wonder if signing up for Digium support would help to resolve this. Not sure if they cover ARI in their plans. |
| 00:44.13 | Samot | Well you just submitted the debug details 25 minutes ago. |
| 00:44.21 | Samot | Might be wise to give them a chance to actually look at it. |
| 00:47.58 | [TK]D-Fender | A night. On a weekend. |
| 00:48.08 | [TK]D-Fender | Yeah, expecting instant results looks REALLY bad... |
| 00:50.44 | Samot | Specially when it was requested two days ago. |
| 00:51.19 | Samot | I get that high volume and live traffic make it hard to do the debug captures are a decent time but.. |
| 00:51.29 | Samot | s/are/at |
| 00:54.39 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
| 01:00.34 | Samot | Hrm. It seems the WWE owes me money. |
| 01:56.18 | tompaw | WWE? |
| 02:23.00 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 02:36.12 | Samot | World Wrestling Entertainment. |
| 03:18.03 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 03:20.35 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 03:46.36 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 04:13.01 | *** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta) |
| 05:48.45 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 06:40.44 | *** join/#asterisk jlnt (~jlnt@209-33-61-48.estlcmtc01.res.dyn.suddenlink.net) |
| 07:27.08 | *** join/#asterisk KaliLinuxGR (~KaliLinux@unaffiliated/kalilinuxgr) |
| 08:26.34 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:49.25 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 09:26.52 | *** join/#asterisk KaliLinuxGR (~KaliLinux@unaffiliated/kalilinuxgr) |
| 09:46.11 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
| 11:40.02 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 12:37.22 | *** join/#asterisk netman (~netman@185.94.249.77) |
| 12:44.29 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 13:20.14 | *** join/#asterisk tuxian (~tuxian@igilmour.plus.com) |
| 13:57.02 | tompaw | Samot: yeah I know but why would they owe you? |
| 14:11.22 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 14:19.54 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-djmiqdccninoadkw) |
| 14:20.56 | Samot | Because they over charged me for my subscription. |
| 14:21.39 | tompaw | I see. |
| 16:01.10 | tompaw | Is Local channel linked into main asterisk binary? |
| 16:42.57 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 17:09.19 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 17:25.14 | igcewieling1 | tompaw: "module show like local" in the Asterisk CLI. |
| 17:25.55 | tompaw | igcewieling1: found it, core_local is linked to "asterisk", but it only takes a second to rebuild, because Makefile is smart :-) |
| 17:26.19 | igcewieling1 | It is possible they rolled it into the main binary, but it is a separate module in Asterisk 11. "chan_local.so Local Proxy Channel (Note: used internal 0 " |
| 17:27.28 | igcewieling1 | core local? |
| 17:41.06 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:49.22 | tompaw | yep |
| 17:49.55 | tompaw | in 13 it's not showing as a module |
| 17:51.51 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 17:53.18 | Samot | chan_local is a module that can be loaded and unloaded. |
| 17:53.22 | Samot | In 13. |
| 17:53.42 | tompaw | well, local_hangup is defined in core_local.c, which is what I'm debugging |
| 18:20.39 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 18:22.55 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 19:06.16 | *** join/#asterisk zacky83 (~zacky83@AToulon-651-1-179-233.w83-113.abo.wanadoo.fr) |
| 19:28.33 | *** join/#asterisk BakaKuna (~BakaKuna@145.129.205.133) |
| 19:53.58 | *** join/#asterisk slack1 (~slack@103.254.153.99) |
| 21:20.19 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 21:21.10 | *** join/#asterisk Jack17 (~jackkk@212.34.23.123) |
| 21:21.52 | Jack17 | guys im using the originate having 1 small problem is when doing originate even when i hangup the call keeps ringing |
| 21:22.12 | Jack17 | how can i make sure the originate is terminated when the dialer hangs up |
| 21:25.51 | Jack17 | hello |
| 21:26.15 | [TK]D-Fender | What dialer? |
| 21:26.19 | [TK]D-Fender | hangs up how? |
| 21:26.26 | [TK]D-Fender | Show what you're actually doing |
| 21:26.31 | [TK]D-Fender | ~pb |
| 21:26.31 | infobot | i heard pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 21:26.32 | [TK]D-Fender | ^^^ |
| 21:29.40 | Jack17 | https://gist.github.com/anonymous/1f7617567d18a11f9ae65f468a4ca2bb |
| 21:30.02 | Jack17 | [TK]D-Fender i did hangup but call is still ringing |
| 21:31.05 | tompaw | Samot: I don't see a file called chan_local anywhere in 13's source |
| 21:31.30 | Jack17 | https://gist.github.com/anonymous/7fa532a6171e8bbb48c0cf917e3b72bc |
| 21:31.54 | [TK]D-Fender | Originate has nothing to do with the channel at Originates it |
| 21:32.00 | [TK]D-Fender | You fail to understand the basics. |
| 21:32.10 | [TK]D-Fender | It launches a SEPARATE call |
| 21:32.17 | [TK]D-Fender | They are not tied |
| 21:32.25 | [TK]D-Fender | So what is your actual goal? |
| 21:32.39 | Samot | Jack17 = follow-me |
| 21:32.42 | Jack17 | if the person who is going to answer is human bridge |
| 21:32.47 | Jack17 | follow-me is my partner |
| 21:33.04 | [TK]D-Fender | Then this has nothing to do with originate |
| 21:33.08 | Jack17 | Samot you know him |
| 21:33.09 | Samot | tompaw: I might have been wrong. |
| 21:33.22 | Samot | We've tried working with him. |
| 21:33.23 | [TK]D-Fender | "core show application dial" <- |
| 21:33.29 | [TK]D-Fender | Run a macro and use AMD |
| 21:33.38 | [TK]D-Fender | Like we told follow-me a HUNDRED TIMES |
| 21:33.47 | Samot | ^ Which he was told from the start. |
| 21:34.00 | [TK]D-Fender | Wasted HOURS on an idiot who can't follow directions |
| 21:34.14 | Samot | Over a span of DAYS |
| 21:34.48 | *** join/#asterisk sl4ck (~slack@88-111-153-198.dynamic.dsl.as9105.com) |
| 21:35.16 | [TK]D-Fender | I also seriously doubt claims of this being about a "partner" |
| 21:36.00 | Jack17 | :D |
| 21:36.05 | Jack17 | i told him to use macro |
| 21:36.17 | [TK]D-Fender | Funny YOU aren't doing it |
| 21:36.29 | [TK]D-Fender | Yeah I'm not buying this story... |
| 21:36.36 | Jack17 | i do accounting but im about to learn the basic to do i t |
| 21:36.58 | [TK]D-Fender | We've given the instructions AGAIN. |
| 21:36.59 | Jack17 | [TK]D-Fender thanks for your precious time |
| 21:36.59 | [TK]D-Fender | Do it. |
| 21:37.35 | Samot | If you told him to use a macro, how are you *now* just learning the basics? |
| 21:37.36 | Jack17 | one more question , followme told me that with the macro the line will always open we need to bridge in one codition only and that if its a human |
| 21:37.45 | Samot | Because knowing to use a marco means you know something.. |
| 21:37.49 | [TK]D-Fender | Samot, SHH!!! quiet with your logic! |
| 21:38.05 | [TK]D-Fender | POLO! |
| 21:38.31 | [TK]D-Fender | <[TK]D-Fender> "core show application dial" <- |
| 21:38.31 | [TK]D-Fender | <[TK]D-Fender> Run a macro and use AMD |
| 21:38.33 | [TK]D-Fender | ^^^^^^^^^^^ |
| 21:38.39 | [TK]D-Fender | Last time. FOLLOW THE INSTRUCTIONS |
| 21:38.43 | Jack17 | thanks |
| 21:39.07 | [TK]D-Fender | Dial w/ macro. AMD. |
| 21:44.31 | *** join/#asterisk emel_punk|2 (~emel_punk@161.18.149.228) |
| 21:48.24 | Jack17 | [TK]D-Fender exten => _X.,n,Dial(SIP/provider/${EXTEN},M(amd) ) |
| 21:48.27 | tompaw | belive his issue is inside ast_unreal_hangup |
| 21:49.04 | emel_punk|2 | greetings, I am trying to compile asterisk 11. with console_video support and included the DHAVE_VIDEO_CONSOLE but when the make process is running I have an error with vgrabbers.c about PIX_FMT_RGB32. Can somebody help me? |
| 21:51.59 | Samot | emel_punk|2: You should use 13. 11 is EOL/SFO. |
| 21:53.44 | tompaw | * \note Batman. << what's that in ast_unreal_lock_all? <: |
| 21:58.02 | emel_punk|2 | oh no |
| 22:13.24 | emel_punk|2 | sorry but asterisk 13 didn't work, the same error. |
| 22:13.37 | emel_punk|2 | any other clue? |
| 22:20.41 | emel_punk|2 | got this vgrabbers.c:118: error: âPIX_FMT_RGB32â undeclared (first use in this function) |
| 22:31.05 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 22:51.34 | *** join/#asterisk emel_punk|2 (~emel_punk@190.67.252.41) |
| 22:52.30 | emel_punk|2 | apparently no other has report an error like this... or I am missing some silly dependency? |
| 23:05.00 | *** join/#asterisk scriptinghelp (~scripting@95.211.188.10) |
| 23:05.25 | scriptinghelp | hi all |
| 23:05.28 | emel_punk|2 | JOIN asterisk-dev |
| 23:06.05 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 23:06.47 | scriptinghelp | could someone please help me? i have a problem with my Asterisk Queue. Everything is working perfectly, but sometimes i see that Asterisk does not make calls to ALL of the members in the queue, only some of them. If i reboot Asterisk, it begins to make calls to all members of the queue again. |
| 23:07.44 | WIMPy | Don't post duplicates to the wrong channel. |
| 23:07.55 | scriptinghelp | where should i be posting please? |
| 23:08.27 | [TK]D-Fender | Show us |
| 23:08.36 | WIMPy | here |
| 23:08.55 | scriptinghelp | ok sorry, where should i show you? Pastebin? |
| 23:09.34 | scriptinghelp | If I start asterisk, ALL members of the queue ring. After a while, if I de-register some members and re-register, they do not even try to ring anymore until I reboot Asterisk |
| 23:10.40 | [TK]D-Fender | SHOW US |
| 23:10.42 | [TK]D-Fender | PASTEBIN |
| 23:10.46 | [TK]D-Fender | just like befre |
| 23:10.48 | [TK]D-Fender | configs |
| 23:10.50 | [TK]D-Fender | status dumps |
| 23:10.53 | [TK]D-Fender | actual calls |
| 23:15.02 | scriptinghelp | ok, one moment please |
| 23:18.01 | emel_punk|2 | Greetings, I am trying to compile Asterisk 11 or 13 with console video support, I included the flag _ASTCFLAGS+=-DHAVE_VIDEO_CONSOLE on Makefile, but the compilation fails on module vgrabbers.c with vgrabbers.c:121: error: âPIX_FMT_RGB555â undeclared (first use in this function). Any clue of what should I do? |
| 23:18.20 | tompaw | core_unreal.c -> /* Owner side is hanging up. */ << that's where my problem must be. debugging this is a pain in the ass!! |
| 23:21.44 | scriptinghelp | http://cryptb.in/nXno#ed003cdc79e3b885b55d36e231f5e935 |
| 23:31.36 | [TK]D-Fender | I see a LOT of "absent" members |
| 23:32.05 | [TK]D-Fender | Also several "busy" ones |
| 23:32.21 | [TK]D-Fender | And you didn't include the STATUS DUMP I asked for |
| 23:32.34 | [TK]D-Fender | "queue show" |
| 23:36.27 | scriptinghelp | they are all absent except me as it is a test now |
| 23:36.38 | scriptinghelp | iâm posting the queue show now. |
| 23:36.52 | [TK]D-Fender | So how does this test show us the problem? |
| 23:37.38 | scriptinghelp | i have written the problem in the paste link very clearly |
| 23:38.02 | emel_punk|2 | show us a "queue show" with the real problem |
| 23:38.42 | [TK]D-Fender | Right away your member logic is a complete waste. |
| 23:39.35 | [TK]D-Fender | You have members using a local channel that one-to-one call a predictable SIP device. You could have givein it the SIP device for STATE detection to get rid of calling people who aren't there |
| 23:39.49 | [TK]D-Fender | And save the noise, system load, etc |
| 23:40.19 | scriptinghelp | http://pastebin.com/B7YA2mU0 |
| 23:41.00 | scriptinghelp | >> You could have givein it the SIP device for STATE detection to get rid of calling people who aren't there // I have no idea how to do that, and I am in a rush, and Asterisk consultant built this system for me.. since then I have been looking for another Asterisk consultant to pay, but they seem to be very busy or rarely available at least |
| 23:44.21 | *** join/#asterisk sl4ck (~slack@185.57.80.147) |
| 23:46.10 | scriptinghelp | All members ring.â¨Then i close my SIP phone, reopen it 10mins later, and it doesnât ring me anymore |
| 23:47.24 | scriptinghelp | if i restart asterisk, itâll start ringing all queue members again |
| 23:48.13 | [TK]D-Fender | member=Local/888111@from-queue/n,0,"888111",SIP/888111 |
| 23:48.42 | [TK]D-Fender | Fix them all up accordingly then restart * and show us the new configs & status dump, then the call |
| 23:49.03 | [TK]D-Fender | ; syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse] |
| 23:51.54 | scriptinghelp | should it just be |
| 23:51.55 | scriptinghelp | member => Local/888111@from-queue |
| 23:51.55 | scriptinghelp | ? |
| 23:52.36 | [TK]D-Fender | <[TK]D-Fender> member=Local/888111@from-queue/n,0,"888111",SIP/888111 |
| 23:52.36 | [TK]D-Fender | <[TK]D-Fender> Fix them all up accordingly then restart * and show us the new configs & status dump, then the call |
| 23:52.47 | [TK]D-Fender | I used YOUR code and handed you the sample |
| 23:53.06 | [TK]D-Fender | add the last parameter as above for each |
| 23:53.11 | scriptinghelp | oh, thank you very much |
| 23:54.49 | scriptinghelp | i am trying this now |
| 23:55.36 | scriptinghelp | oh my test phone rings now |
| 23:55.38 | scriptinghelp | without a restart :) |
| 23:55.48 | scriptinghelp | could it be that âsimpleâ ? [TK]D-Fender |
| 23:56.17 | [TK]D-Fender | I don't knwo which one was your test phone, but this is an immediately smart thing to do to avoid pointless members being called |
| 23:56.57 | scriptinghelp | thank you very very much you have saved me a lot of headache |
| 23:57.01 | scriptinghelp | i believe it is fixed |
| 23:57.14 | [TK]D-Fender | Let's see what it all looks like now |