| 00:09.40 | alexturner | pabelanger: Unknown? Though this exact config was working on another install - totally confuzzled |
| 00:13.18 | [TK]D-Fender | From: ""<sip:unknown@192.168.17.8>;tag=1837a804c6057ba6 |
| 00:13.42 | pabelanger | From: ""<sip:unknown@192.168.17.8>;tag=1837a804c6057ba6 |
| 00:13.46 | pabelanger | ya, that |
| 00:13.51 | [TK]D-Fender | Empty "from" and that is a multi-port gateway that is not configured for usedID auth. |
| 00:13.58 | [TK]D-Fender | userid. |
| 00:14.06 | [TK]D-Fender | Should be type=friend, not "peer" |
| 00:14.23 | [TK]D-Fender | Change your types, and fix the basic auth on that port |
| 00:24.26 | InHisName | I'm back now that kids are asleep. [TK]D-Fender: I captured this just as you were leaving earlier. http://pastebin.com/Nr0ckMt5 here is the sip debug of registration attempts. |
| 00:25.51 | [TK]D-Fender | InHisName: EvERY provider (I counted at least 3) are timing out. Serious networking issues. |
| 00:27.38 | InHisName | yes, I know that. Seems that asterisk cannot get ip addresses for its registrations. BUT in linux, pings of names works just fine. DNS working in linux but not with asterisk ? |
| 00:28.27 | InHisName | I am probably missing fixing a setting change somewhere with asterisk. |
| 00:29.13 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 00:30.13 | InHisName | What should be inside of dnsactual.conf file ? I changed it to my current outside IP. Maybe it wasn't quite that before and I wasn't paying close enough attention. |
| 00:34.45 | *** join/#asterisk bmg505 (~leon@196-209-163-108.dynamic.isadsl.co.za) |
| 00:36.29 | *** join/#asterisk Johnny- (~John@c-68-81-19-251.hsd1.pa.comcast.net) |
| 00:40.54 | *** join/#asterisk tzanger (tzanger@wallace.mixdown.ca) |
| 00:41.43 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 00:43.56 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:43.56 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:46.28 | [TK]D-Fender | InHisName: Do I know that the names it's resolving are WRONG? Because it is coming up with them... |
| 00:46.33 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:46.34 | [TK]D-Fender | This soulds like you are being blocked |
| 00:47.54 | InHisName | Is there something to do from linux command line to test to prove blocking or not ? |
| 00:53.02 | InHisName | Although, I don't think I can blame the ISP, I suspect it is in my asterisk setup somewhere that is causing my grief. |
| 01:01.31 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 01:01.31 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
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| 01:06.50 | [TK]D-Fender | Contact: <sip:s@96.25.37.57> |
| 01:06.54 | [TK]D-Fender | is that your server IP? |
| 01:07.11 | [TK]D-Fender | if so, it is sending to different WAN IP's for each host, so lookups seem to be doing their thing. |
| 01:07.34 | [TK]D-Fender | So either you're sending the wrong IP (which I figure ONE of them should have ignored), or you are firewalled somewhere |
| 01:07.51 | [TK]D-Fender | SIP ALG, if not raw port |
| 01:13.35 | [TK]D-Fender | heads out for the night |
| 01:16.07 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.64.195.dsl.dyn.forthnet.gr) |
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| 01:43.19 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 01:52.46 | alexturner | Hey guys, any idea why I'm getting auth errors from my FXO gateway? http://pastebin.com/UUrqXS0k |
| 02:04.01 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
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| 03:25.44 | ChannelZ | alexturner: No matching peer for 'unknown' from '192.168.17.200:5062' |
| 03:26.30 | ChannelZ | the peer you showed (asuming that's what the FXO is supposed to be) is set to 'dynamic' but apparently hasn't registered such that Asterisk knows that IP belongs to it. |
| 03:29.57 | alexturner | Hmm i see |
| 03:30.11 | alexturner | i've registered the host in sip.conf |
| 03:38.38 | *** join/#asterisk mokmeister (~mokmeiste@109.78.94.27) |
| 04:22.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:30.42 | ChannelZ | does it have a static IP? (should, it's LAN..) |
| 04:32.30 | alexturner | yeah it does, and i have set that accordingly |
| 04:32.56 | alexturner | well I'm running something simpler now |
| 04:32.58 | alexturner | still no luck |
| 04:36.59 | ChannelZ | show |
| 05:00.12 | *** part/#asterisk alexturner (~admin@onebig.lnk.telstra.net) |
| 05:14.44 | *** join/#asterisk kasanop (~kasanop@95-27-144-8.broadband.corbina.ru) |
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| 07:52.12 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 08:32.05 | onebitxajax | hi to all |
| 08:32.28 | onebitxajax | sip_budies is under sterisk or asteriskrealtime table? |
| 08:32.44 | onebitxajax | i want to setup an asterisk realtime user |
| 08:33.42 | onebitxajax | some tutoriall create it under asterisk other under asteriskrealtime |
| 08:50.03 | *** join/#asterisk oquidave (~oquidave@41.223.84.35) |
| 08:51.16 | oquidave | hello, am getting pbx.c:4235 pbx_extension_helper: No application 'DB_DELETE' for extension (subCountLoop, s, 2) ...is the DB_DELETE function supported in asterisk 1.8? |
| 08:51.45 | oquidave | DBdel and DBdeltree are working fine |
| 08:52.32 | oquidave | but when i use DBdel, asaterisk tells me it's decaprecated!! yet when i use DB_DELETE, it says not such application!! |
| 08:52.52 | kasanop | onebitxajax: you can name the database whatever you want as long as you write the same database name in the "sippeers" string in extconfig.conf |
| 08:53.24 | ChannelZ | oquidave: I'm thinking probably your syntax is wrong |
| 08:53.32 | kasanop | oquidave: DB_DELETE is not an application, it's a function |
| 08:54.00 | oquidave | ChannelZ: i've same=>n,DB_DELETE(test/count); |
| 08:55.29 | ChannelZ | yeah. As kasanop says, DB_DELETE is a function, not a dialplan application |
| 08:56.11 | oquidave | ChannelZ: that means i can't use it in the dialplan? |
| 08:56.15 | ChannelZ | I think you could get away with NoOp(${DB_DELETE(test/count)}) |
| 08:56.32 | oquidave | or i've to to use it inside an application |
| 08:56.38 | ChannelZ | right |
| 08:56.43 | oquidave | ChannelZ: okay |
| 08:56.55 | ChannelZ | A function is sort of like a variable that can do things. |
| 08:57.09 | ChannelZ | In the sense of where it can be used, anyway. |
| 08:57.23 | oquidave | ChannelZ: but DBdel and DBdeltree are applications, right?bse they work |
| 08:57.37 | ChannelZ | right |
| 08:57.53 | ChannelZ | see "core show applications" vs "core show functions" |
| 08:58.24 | oquidave | ChannelZ: that feels inconsistent to me! |
| 08:58.29 | ChannelZ | Not really |
| 08:59.00 | ChannelZ | For the example you provided, there's no reason to use DB_DELETE - just use the application DBdel |
| 08:59.48 | oquidave | ChannelZ: but DBdel is decaprecated! |
| 08:59.53 | ChannelZ | However, if you read the help for DB_DELETE, it returns the value of the key you are deleting, which might be useful in some other logic. |
| 09:00.29 | ChannelZ | I was just making a distinction between functions and applications |
| 09:01.29 | oquidave | ChannelZ: okay thanks...meanwhile NoOp() is some sort of print application right |
| 09:02.37 | ChannelZ | well it literally means 'do nothing' but it will output to the console so it's often used as a sort of debug printf |
| 09:03.15 | oquidave | ChannelZ: great thanks again |
| 09:03.24 | *** join/#asterisk jsjc (~Adium@96.pool85-51-81.dynamic.orange.es) |
| 09:03.36 | ChannelZ | Yup have fun |
| 09:05.21 | onebitxajax | kasanop: they don't appear in asterisk cli :/ |
| 09:05.33 | oquidave | so i should always use same=>n,NoOp(${DB_DELETE(test/count)}) instead of same=>n,DBdel(test/count); |
| 09:06.27 | onebitxajax | please can someone give a good guide to how can i setuo asterisk in realtime??? |
| 09:06.31 | onebitxajax | !help |
| 09:09.03 | ChannelZ | oquidave: you can use DBdel, just know that it will eventually go away and you'll have to update your dialplan if you upgrade Asterisk down the road. It's in 10, not sure about 11. |
| 09:10.04 | oquidave | ChannelZ: noted! |
| 09:15.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 09:28.45 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
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| 10:41.23 | kaldemar | onebitxajax: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html |
| 10:42.13 | onebitxajax | kaldemar: i read and tried 5 tutorials without success |
| 10:42.16 | onebitxajax | let's try this one |
| 10:42.43 | onebitxajax | kaldemar: dosen't work http://www.voip-info.org/wiki/view/Asterisk+RealTime |
| 10:43.00 | onebitxajax | kaldemar: either dosen't work http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example |
| 10:46.42 | kaldemar | onebitxajax: the latter is infested with typos in meaningful parts. no wonder if it does not work. |
| 10:47.21 | onebitxajax | kaldemar: i am with version Asterisk 1.4.19, |
| 10:47.22 | kaldemar | has never seen so many ways to misspell "extension" |
| 10:47.30 | onebitxajax | LOL |
| 10:47.32 | kaldemar | onebitxajax: that's something you should change. |
| 10:47.56 | onebitxajax | kaldemar: what do you mean? |
| 10:48.49 | kaldemar | onebitxajax: 1.4.19 is ancient even in the 1.4 branch, and the whole branch has no support at all. |
| 10:48.54 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 10:49.02 | onebitxajax | WOW |
| 10:49.31 | onebitxajax | i already say to our client "format and reinstall a good centos version" |
| 10:49.37 | onebitxajax | i don't want3 |
| 10:49.39 | onebitxajax | :/ |
| 10:50.24 | kaldemar | actually you shouldn't try with the link i gave you either, it is meant for 1.8. |
| 10:51.47 | kaldemar | this is an older edition of the book for 1.4 versions (in case you make the bad decision to stick with 1.4.19): http://downloads.oreilly.com/books/9780596510480.pdf |
| 10:56.59 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-7-211.broadband.tenet.odessa.ua) |
| 11:03.51 | onebitxajax | kaldemar: is that possibile tu upgrade to new version manually? |
| 11:05.10 | kaldemar | sure |
| 11:05.34 | kaldemar | there are even precompiled binary packages if you don't want to compile from source. |
| 11:06.01 | kaldemar | http://www.asterisk.org/downloads/yum |
| 11:09.52 | *** join/#asterisk funky1 (~jarvis@ip51cf100e.direct-adsl.nl) |
| 11:11.15 | funky1 | hi all, i have a 3g usb modem huawei, that is unlocked and has voice function enabled, i use my phone sim in there, in windows and ubuntu i can make and receive calls, but on my asterisk machine i have trouble getting the modem to work properly, is there any way that i could run it in my ubuntu machine and register the modem as an extension within asterisk? |
| 11:11.58 | funky1 | my asterisk machine is centos to be clear |
| 11:24.31 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 11:28.05 | *** join/#asterisk miltux (~miltux@62.1.222.223.dsl.dyn.forthnet.gr) |
| 11:55.06 | *** join/#asterisk dolcea-sgs3 (~androirc@232-197-235-201.fibertel.com.ar) |
| 11:57.39 | *** join/#asterisk dolcea-xoom (~androirc@232-197-235-201.fibertel.com.ar) |
| 12:07.27 | dolcea-xoom | Hi guys |
| 12:08.12 | dolcea-xoom | Is there a way to use a bluetooth device with asterisk? |
| 12:09.08 | WIMPy | funky1: That's not very helpfull for debugging. Do you need usb_modeswitch? Does that wotk on Centos? |
| 12:09.16 | WIMPy | dolcea-xoom: chan_mobile |
| 12:09.44 | WIMPy | dolcea-xoom: Or the other way? |
| 12:10.21 | funky1 | WIMPy: no, i believe you only need usb_modeswitch when your modem is not recognized properly, what more can i past for debugging info? |
| 12:10.33 | dolcea-xoom | Googling, thanks |
| 12:11.28 | WIMPy | funky1: No, modeswitch is needed to to configure the stick to the correct function. |
| 12:12.22 | funky1 | hm ok, i don't think i need it but let me read up on it to be sure |
| 12:12.58 | WIMPy | Have you tried to use it in other ways on the Centos box? |
| 12:14.02 | cusco_ | using chan_dongle ? |
| 12:14.26 | funky1 | modeswitch only switches when there is some storage or stuff like that on the stick,which i actually have but i have set that already before, so i don't need modeswitch (but have tried it anyways before) |
| 12:14.42 | funky1 | i can communicate through minicom with the modem |
| 12:14.52 | funky1 | tried chan_dongle trying to get that to work actually |
| 12:14.58 | funky1 | but does not work |
| 12:15.19 | WIMPy | Ok, if you can use minicom on that box, that's a good start. |
| 12:16.17 | funky1 | so what now then? |
| 12:17.07 | WIMPy | If the channel doesn't provide any debug, you have to ask a crystal ball :-( |
| 12:17.28 | funky1 | lol |
| 12:17.32 | funky1 | i guess i'll have to |
| 12:17.39 | WIMPy | Isn't chan_dongle a fork of chan_datacard? |
| 12:18.25 | WIMPy | That one did give me debug outpout. |
| 12:18.30 | funky1 | is there any way that you might now where i could attach the usb 3g stick to a windows machine use there the windows mobile partner application with which i can make and receive calls and register that as an extension in asterisk? |
| 12:18.59 | WIMPy | If the application supports it... |
| 12:19.14 | funky1 | i thought chan_datacard was old version, when using chan_dongle i see that it is trying to initializing the modem but fails and just gives error |
| 12:19.32 | WIMPy | What do you see? |
| 12:19.44 | funky1 | is there a windows application that supports this that you know of? or linux application? |
| 12:20.04 | WIMPy | Asterisk :-) |
| 12:21.29 | funky1 | <PROTECTED> |
| 12:21.29 | funky1 | <PROTECTED> |
| 12:21.29 | funky1 | <PROTECTED> |
| 12:22.21 | funky1 | thgouht i got a question about dongle.conf i have the imei and imsi of my device and when i specify those do i still need to specify /dev/ttyUSB0 for audio and the other for data? |
| 12:23.03 | funky1 | in the conf it says i can omit those settings when i enter the imei and imsi but when i do that without the "/dev/" settings i see nothing about initializing the dongle |
| 12:24.36 | funky1 | anything else for windows or linux so that i can just configure it as an extension and link to my main asterisk box |
| 12:27.34 | WIMPy | That's not real debug. I used to see the complete communication. |
| 12:31.32 | funky1 | debug is set to 9 |
| 12:32.55 | *** join/#asterisk wonderworld (~ww@dsdf-4db5d237.pool.mediaWays.net) |
| 12:35.51 | *** join/#asterisk devyll (~devyll@89.47.55.72) |
| 12:36.44 | devyll | hello guys. Can you tell me how to destroy (force to expire) a sip registration? |
| 12:37.10 | kaldemar | sip unregister |
| 12:37.31 | devyll | it's not an otion.. I have 1.4.25 |
| 12:37.35 | devyll | *option |
| 12:38.32 | kaldemar | oh well. then you're out of luck. consider upgrading. |
| 12:56.03 | volga629 | Hello Everyone, channel.c:1498 __ast_queue_frame: Exceptionally long voice queue length queuing to what is mean ? |
| 12:57.07 | volga629 | this is dial ext http://fpaste.org/PfrS/ |
| 13:05.04 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
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| 13:54.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 14:06.53 | onebitxajax | Package asterisk requires libspandsp-0.0.3.so.0 |
| 14:06.55 | onebitxajax | Package urw-fonts requires chkfontpath |
| 14:06.55 | onebitxajax | Package urw-fonts requires mkfontdir |
| 14:06.55 | onebitxajax | Package urw-fonts requires mkfontscale |
| 14:06.56 | onebitxajax | Package system-config-keyboard requires firstboot |
| 14:06.58 | onebitxajax | Package system-config-keyboard requires gtk2 >= 2.6 |
| 14:07.01 | onebitxajax | Package libmfcr2 requires libspandsp-0.0.3.so.0 |
| 14:07.04 | onebitxajax | Package libsupertone requires libspandsp-0.0.3.so.0 |
| 14:07.11 | onebitxajax | can someone help me with myupgrade |
| 14:07.22 | onebitxajax | locate libspandsp-0.0.3.so.0 |
| 14:07.23 | onebitxajax | /usr/lib/libspandsp-0.0.3.so.0 |
| 14:07.23 | onebitxajax | /usr/lib/libspandsp-0.0.3.so.0.0.2 |
| 14:23.45 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 14:25.54 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::1) |
| 14:33.25 | onebitxajax | is there any solution to update manually asterisk? |
| 14:38.52 | cusco | how else would you update? |
| 14:51.44 | onebitxajax | cusco: http://www.asterisk.org/downloads/yum |
| 14:52.18 | onebitxajax | cusco: hw can i do it mnually, from source or from compilked rpm? |
| 14:52.57 | file | raises eyebrow |
| 14:53.34 | cusco | how did you install it last time? |
| 14:54.47 | onebitxajax | cusco: i installed an iso with asterink in it |
| 14:55.00 | cusco | ow, is that asterisknow? |
| 14:55.25 | onebitxajax | cusco: elastiks |
| 14:55.46 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:55.46 | cusco | you should refer to #elastix for that |
| 14:55.52 | onebitxajax | ah ok |
| 14:56.00 | cusco | elastix is based on asterisk, but not quite the same |
| 14:56.11 | cusco | upgrading any way we would recommend could break your system |
| 14:56.27 | onebitxajax | :/ |
| 14:56.31 | onebitxajax | i don't understand |
| 14:56.44 | onebitxajax | i tried everithingto make asterisk real time |
| 14:56.50 | onebitxajax | but i don't succed |
| 14:56.56 | onebitxajax | i don't know what to do T-T |
| 14:57.03 | cusco | ? |
| 14:57.12 | cusco | looked at extconfig ? |
| 14:57.31 | onebitxajax | cusco: |
| 14:57.33 | onebitxajax | yes3 |
| 14:57.45 | onebitxajax | cusco: do you have 10 min ute to help me with it? |
| 14:57.50 | cusco | so... what backend are you trying? |
| 14:58.24 | cusco | keep on asking and I'll help you as I see fit... lol |
| 14:58.28 | onebitxajax | what do you mean with backend? |
| 14:58.35 | onebitxajax | ah |
| 14:58.48 | cusco | backend for the configurations... mysql postgresql tds etc |
| 14:58.55 | onebitxajax | mysql |
| 14:59.02 | cusco | using 1.8 ? |
| 14:59.24 | onebitxajax | mysql Ver 14.12 Distrib 5.0.95, for redhat-linux-gnu (i686) using readline 5.1 |
| 14:59.32 | onebitxajax | Asterisk 1.4.19, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 14:59.58 | cusco | asterisk 1.4! ok I have no experience there.. do you have mysql addons installed? |
| 15:01.08 | onebitxajax | i have ollowed ths guide http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/installing_configuring_odbc.html |
| 15:01.22 | onebitxajax | however i think yes |
| 15:01.22 | cusco | odbc insteadl of mysql addon? |
| 15:02.06 | cusco | so.. is odbc working? |
| 15:02.18 | onebitxajax | i think no |
| 15:02.27 | onebitxajax | wait |
| 15:02.48 | onebitxajax | i mean when i enter in asterisk cli its say that he is connected to database |
| 15:03.05 | cusco | does it? |
| 15:03.12 | cusco | like 'odbc show status' or something ? |
| 15:03.51 | onebitxajax | no its say odbc command not found in the asterisk cli shell |
| 15:05.08 | cusco | I don't have a instalation with odbc on right now.. |
| 15:05.40 | cusco | do you have mysql addons? check with 'module show like mysql' |
| 15:05.47 | onebitxajax | cusco: to instal it wwith mysql addons i must only install with yum the package asterisk-addons-mysql.i386 |
| 15:06.27 | onebitxajax | cusco: |
| 15:06.29 | onebitxajax | Module Description Use Count |
| 15:06.33 | onebitxajax | res_config_mysql.so MySQL RealTime Configuration Driver 0 |
| 15:06.36 | onebitxajax | cdr_addon_mysql.so MySQL CDR Backend 0 |
| 15:06.39 | onebitxajax | app_cbmysql.so CBMysql conference scheduling 0 |
| 15:06.42 | onebitxajax | app_addon_sql_mysql.so Simple Mysql Interface 0 |
| 15:06.45 | onebitxajax | i got this with module show like mysql |
| 15:08.25 | cusco | seems that you alreaddy have it |
| 15:08.25 | cusco | :) |
| 15:08.25 | cusco | well I have a file in /etc/asterisk/ named: res_mysql.conf |
| 15:08.25 | cusco | do you have such a file? |
| 15:09.19 | cusco | inside it has details of mysql connector.. dbhost, dbname, dbuser, dbpass, dbport, dbsock |
| 15:09.51 | onebitxajax | yes there is |
| 15:10.05 | onebitxajax | cusco: i put it the root and mysql root password |
| 15:11.03 | cusco | lol, you should never do that |
| 15:11.17 | onebitxajax | cusco: i know |
| 15:11.19 | cusco | ok |
| 15:11.31 | cusco | then do you have the bd and table scructure created? |
| 15:11.43 | onebitxajax | yes |
| 15:11.59 | cusco | so... |
| 15:12.00 | onebitxajax | cusco: but there is a problem |
| 15:12.07 | cusco | yes? |
| 15:12.27 | onebitxajax | the structure of my tables are different from tthe structure of the tables that i find in the tutorials |
| 15:12.46 | onebitxajax | but i added 3 person in sipbuddies |
| 15:12.53 | onebitxajax | sip_buddies |
| 15:13.07 | cusco | ok.. so? |
| 15:13.12 | cusco | does asterisk read the mysql table? |
| 15:13.31 | cusco | realtime mysql status |
| 15:13.35 | onebitxajax | with sip show users there is no user |
| 15:13.55 | onebitxajax | no output for realtime mysql status |
| 15:14.35 | cusco | wait, prehaps it is still trying to connect? |
| 15:14.41 | cusco | also set verbose and debug to 15 |
| 15:15.25 | cusco | realtime load |
| 15:15.42 | cusco | oops nvm that |
| 15:16.01 | cusco | soo.. |
| 15:16.05 | cusco | when you start asterisk |
| 15:16.08 | cusco | looking at the full log |
| 15:16.19 | cusco | any relevant info regarding mysql connect attempt? |
| 15:16.25 | cusco | in extconfig what do you have? |
| 15:16.30 | *** join/#asterisk onebitxa1ax (~onebitxaj@unaffiliated/onebitxajax) |
| 15:16.37 | cusco | ow? |
| 15:19.28 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 15:19.57 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 15:26.25 | onebitxa1ax | cusco: i have a question, the new users i must put them in asteriskrealtime table and not in asterisk table, right? |
| 15:26.38 | onebitxa1ax | cusco: sorry disconnecte |
| 15:26.39 | onebitxa1ax | d |
| 15:27.04 | onebitxa1ax | 17:13 < cusco> realtime mysql status |
| 15:27.05 | onebitxa1ax | 17:13 < onebitxajax> with sip show users there is no user |
| 15:27.05 | onebitxa1ax | 17:13 < onebitxajax> no output for realtime mysql status |
| 15:27.33 | cusco | well |
| 15:27.44 | cusco | what do you have mapped in extconfig ? |
| 15:28.10 | cusco | like sippeers => mysql,general,tablename |
| 15:28.18 | cusco | and sipusers => mysql,general,tablename |
| 15:29.10 | onebitxa1ax | cusco: |
| 15:29.13 | onebitxa1ax | sippeers => mysql,general,sip_buddies |
| 15:29.13 | onebitxa1ax | extensions => mysql,general,extensions |
| 15:29.13 | onebitxa1ax | voicemail => mysql,general,voicemail_users |
| 15:29.15 | onebitxa1ax | queues => mysql,general,queue_table |
| 15:29.18 | onebitxa1ax | queue_members => mysql,general,queue_member_table |
| 15:29.20 | onebitxa1ax | meetme => mysql,general,meetme |
| 15:29.39 | cusco | ok... |
| 15:29.49 | cusco | so.. when you start asterisk |
| 15:29.50 | onebitxa1ax | sip_budies exist under asteriskrealtime table |
| 15:29.54 | cusco | do you see it trying to connect? |
| 15:29.59 | onebitxa1ax | yes |
| 15:30.00 | cusco | ? |
| 15:30.04 | cusco | sip_buddies is a tablename |
| 15:30.21 | onebitxa1ax | sorry database |
| 15:30.22 | cusco | exists under a database, specified in the general section of res_mysql |
| 15:30.24 | cusco | appointed db |
| 15:30.26 | onebitxa1ax | sip_budies exist under asteriskrealtime database* |
| 15:30.38 | onebitxa1ax | mmmmmm |
| 15:30.41 | cusco | and is asteriskrealtime in res_mysql ? |
| 15:30.58 | onebitxa1ax | yes |
| 15:31.09 | cusco | what does it say when you start asterisk? |
| 15:31.17 | cusco | have you looked at full log? |
| 15:31.43 | onebitxa1ax | no i don't look to log, where i can find them? |
| 15:32.05 | cusco | in /var/log/asterisk/full or where appointed in logger.conf |
| 15:32.54 | onebitxa1ax | [Oct 13 07:47:46] WARNING[6191] res_config_mysql.c: MySQL RealTime: Failed to query database. Check debug for more info. |
| 15:33.06 | onebitxa1ax | [Oct 13 07:47:46] DEBUG[6191] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '2456' |
| 15:33.09 | onebitxa1ax | [Oct 13 07:47:46] WARNING[6191] res_config_mysql.c: MySQL RealTime: Failed to query database. Check debug for more info. |
| 15:33.19 | cusco | there you go |
| 15:33.20 | onebitxa1ax | [Oct 13 07:47:46] DEBUG[6191] res_config_mysql.c: MySQL RealTime: Everything is fine. |
| 15:33.23 | onebitxa1ax | LOOOOOL |
| 15:33.43 | cusco | check debug for more info |
| 15:34.28 | onebitxa1ax | hoa can i do that? |
| 15:35.17 | onebitxa1ax | argh!!! [Oct 13 17:34:55] DEBUG[10870] res_config_mysql.c: MySQL RealTime: Cannot Connect (1045): Access denied for user 'root'@'localhost' (using password: YES) |
| 15:35.25 | cusco | :) |
| 15:35.34 | onebitxa1ax | mmmmmmmm |
| 15:35.38 | cusco | you found your problem |
| 15:35.42 | onebitxa1ax | wait |
| 15:36.01 | *** join/#asterisk vinhdizzo (~vinh@ip70-181-154-104.sd.sd.cox.net) |
| 15:37.18 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 15:38.14 | onebitxa1ax | cusco: |
| 15:38.14 | onebitxa1ax | [Oct 13 17:37:50] WARNING[10932] res_config_mysql.c: MySQL RealTime: Unable to select database: general. Still Connected (1049). |
| 15:38.18 | onebitxa1ax | [Oct 13 17:37:50] DEBUG[10932] res_config_mysql.c: MySQL RealTime: Database Select Failed (1049): Unknown database 'general' |
| 15:38.27 | onebitxa1ax | why he searcg general if i put asteriskrealtime?! |
| 15:38.29 | onebitxa1ax | :/ |
| 15:38.30 | cusco | -_- |
| 15:38.34 | cusco | wrong config |
| 15:38.46 | cusco | review your res_mysql.conf and extconfig.conf accordingly |
| 15:39.44 | onebitxa1ax | cusco: there is not general in all 2 files!!! |
| 15:40.04 | cusco | you should have a [general] section |
| 15:40.16 | onebitxa1ax | cusco: i must put asteriskrealtime eitther in the tag [general?] |
| 15:40.25 | onebitxa1ax | yes there is |
| 15:40.41 | onebitxa1ax | so i must put [asteriskrealtime] and not [general]??? |
| 15:40.42 | cusco | you're contraicting yourself |
| 15:40.45 | cusco | no |
| 15:41.06 | cusco | 16:09 < cusco> inside it has details of mysql connector.. dbhost, dbname, dbuser, dbpass, dbport, dbsock |
| 15:41.16 | onebitxa1ax | [general] |
| 15:41.16 | onebitxa1ax | dbhost = 127.0.0.1 |
| 15:41.16 | onebitxa1ax | dbname = asteriskrealtime |
| 15:41.16 | onebitxa1ax | dbuser = root |
| 15:41.16 | onebitxa1ax | dbpass = mypassword |
| 15:41.21 | onebitxa1ax | it's here |
| 15:41.28 | cusco | reload/restart |
| 15:41.34 | onebitxa1ax | where is wrong? |
| 15:41.36 | cusco | and show the output |
| 15:41.57 | onebitxa1ax | cusco: it's ok with asterisk -rx "restart now" |
| 15:42.01 | onebitxa1ax | ? |
| 15:42.10 | cusco | does that work? |
| 15:42.17 | onebitxa1ax | or i must restart all the machine? |
| 15:42.20 | cusco | no |
| 15:42.23 | cusco | only asterisk |
| 15:42.26 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:42.27 | onebitxa1ax | yes that restart asterisk |
| 15:42.36 | cusco | ok |
| 15:42.41 | onebitxa1ax | cusco: it say Disconnected from Asterisk server |
| 15:42.44 | cusco | ok |
| 15:43.23 | onebitxa1ax | mmmmmmmmmmm. |
| 15:43.27 | onebitxa1ax | there is a mega errore |
| 15:43.34 | onebitxa1ax | but it's never shown before |
| 15:43.54 | onebitxa1ax | [Oct 13 17:43:29] DEBUG[10975] res_config_mysql.c: MySQL RealTime: Database Select Failed (1049): Unknown database 'general' |
| 15:44.01 | onebitxa1ax | there is either this |
| 15:44.03 | cusco | :| |
| 15:44.17 | onebitxa1ax | cusco: what i make wrong |
| 15:44.19 | onebitxa1ax | ??????? |
| 15:44.47 | cusco | dunno |
| 16:08.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:22.02 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 16:37.43 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:59.57 | *** join/#asterisk beefcafe (~beefcafe@unaffiliated/beefcafe) |
| 17:01.02 | beefcafe | hi, I'm trying to enable g722, but it seems asterisk isn't sending g722 in INVITE. why is that? |
| 17:04.28 | *** join/#asterisk funky1 (~jarvis@ip51cf100e.direct-adsl.nl) |
| 17:05.27 | jpsharp | Do you have it properly enabled in your sip.conf? |
| 17:13.16 | [TK]D-Fender | Obviously not... |
| 17:16.41 | *** join/#asterisk Alex_Bkash (b4ea5412@gateway/web/freenode/ip.180.234.84.18) |
| 17:23.57 | beefcafe | jpsharp: allow=g722? |
| 17:24.34 | jpsharp | yes. |
| 17:24.48 | jpsharp | But what else do you have in the sip.conf for allow/disallow? |
| 17:25.27 | beefcafe | yes, I have that, right below disallow=all |
| 17:27.45 | [TK]D-Fender | beefcafe: PASTEBIN is your friend.... |
| 17:29.45 | beefcafe | http://pastebin.com/Gk7Rbw3k |
| 17:30.45 | jpsharp | does g722 show up as a valid translation if you do "core show translations"? |
| 17:34.12 | beefcafe | it does. |
| 17:36.39 | [TK]D-Fender | beefcafe: that is one lieelt bit of the communication and I do not see the CALL. You should be showing the * CLI w/ SIP DEBUG for it. |
| 17:36.59 | [TK]D-Fender | little* |
| 17:38.08 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 17:38.30 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 17:40.39 | beefcafe | [TK]D-Fender: problem solved by adding allow=g722 in [general] |
| 17:41.03 | [TK]D-Fender | beefcafe: that meens that calls aren't even matching your peers which is pretty much a BIGGER problem |
| 17:42.49 | beefcafe | I guess so. I'm only a few hours into asterisk. I need many more hours to know how it really works. |
| 17:43.13 | [TK]D-Fender | beefcafe: Well the first thing you should be doing is learning to lok at your calls. |
| 17:43.34 | [TK]D-Fender | beefcafe: "sip set debug on". |
| 17:43.49 | [TK]D-Fender | Well ... maybe second or third. |
| 17:43.56 | [TK]D-Fender | But should not be put off |
| 17:44.16 | beefcafe | I'm not, yet. :P |
| 17:44.22 | [TK]D-Fender | Otherwise you become the guy who just starts shove random parameters all over the place in hopes that something magically starts working... |
| 17:44.51 | jpsharp | throws pickles at the wall to see what sticks. |
| 17:44.56 | [TK]D-Fender | And then opens up his whole system to the world at large and gets defrauded |
| 17:45.27 | beefcafe | I'm trying to connect asterisk to cisco CME. I had the sip messages turned on on cisco but didn't know how to do it in asterisk |
| 17:46.08 | beefcafe | defrauded how? if I don't have any outside trunk or pstn? |
| 17:46.43 | [TK]D-Fender | Well that would change what they could have access to... |
| 17:46.59 | [TK]D-Fender | [13:43][TK]D-Fenderbeefcafe: "sip set debug on". |
| 17:48.26 | beefcafe | [TK]D-Fender: thanks. it pratically shows the same messages, but in reverse direction:P |
| 17:49.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:49.35 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:49.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:49.46 | [TK]D-Fender | beefcafe: ... this is the part where you show us so we can tell you what you DON'T see in there... |
| 17:54.13 | Alex_Bkash | [TK]D-Fender http://pastebin.com/5KJU4xE3 |
| 17:54.40 | Alex_Bkash | [TK]D-Fender plz check this |
| 17:55.00 | Alex_Bkash | this helps me round robin call distributation |
| 17:55.20 | Alex_Bkash | but how can i add failover with this? |
| 17:56.00 | [TK]D-Fender | Alex_Bkash: look at a call that clearly didn't get answered. Choose to dial something else |
| 17:57.55 | Alex_Bkash | why? |
| 17:58.04 | Alex_Bkash | whr is thr problem? |
| 17:58.47 | Alex_Bkash | if u can correct it that will be helpful |
| 17:59.06 | [TK]D-Fender | Alex_Bkash: what is there to correct? |
| 17:59.21 | [TK]D-Fender | Alex_Bkash: You haven't shown something that has failed. |
| 18:00.00 | Alex_Bkash | if gateway1 is congested the call drops |
| 18:00.33 | [TK]D-Fender | Alex_Bkash: I don't SEE a call being dropped... |
| 18:00.35 | Alex_Bkash | in that situation how can i route the call to next gateway? |
| 18:03.12 | Alex_Bkash | so will it route to next gateway if gateway1 is congested? |
| 18:03.15 | [TK]D-Fender | Alex_Bkash: I don't SEE a call being dropped... |
| 18:05.21 | Alex_Bkash | i really dont understand this part very well. so will it do the failover ? |
| 18:06.10 | [TK]D-Fender | Alex_Bkash: I don't SEE a call being dropped... |
| 18:06.32 | beefcafe | [TK]D-Fender: thanks for the help, it was a matching problem. it's using g722 without the global allow=g722 now |
| 18:06.46 | [TK]D-Fender | beefcafe: Glad you found it... |
| 18:11.26 | beefcafe | [TK]D-Fender: thanks. how do I add chan_motif? I installed 11.0 from svn. chan_motif.so can't be found in /usr/lib/asterisk/modules/ |
| 18:11.44 | [TK]D-Fender | beeNever touched it. Don't do G-talk, skype, etc.... |
| 18:11.59 | beefcafe | why? |
| 18:12.43 | [TK]D-Fender | No need. |
| 18:12.57 | beefcafe | I want to use GV for free outbound:P |
| 18:13.08 | [TK]D-Fender | I consult with people using real providers with butter supported services. |
| 18:13.26 | [TK]D-Fender | Yes, and you can be happy to have gotten something when GV gets pulled out from under you. |
| 18:13.41 | [TK]D-Fender | beefcafe: Those I know don't want to have to worry about their services disappearing |
| 18:14.15 | beefcafe | I'm not depending on GV so it's not a concern for me |
| 18:23.43 | [TK]D-Fender | something to try if you have the time I guess... |
| 18:29.38 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 18:45.28 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-hzuybvubevflzpqh) |
| 18:48.08 | *** join/#asterisk onebitxajax (~onebitxaj@unaffiliated/onebitxajax) |
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| 19:05.48 | *** join/#asterisk danfromuk (~IceChat77@2.27.39.234) |
| 19:06.32 | danfromuk | Hi, is there any way to change the recording format in mixmonitor? I have a client complaining about recording quality. |
| 19:07.48 | [TK]D-Fender | danfromuk: And what are you recording from, and what are you saving it ass? |
| 19:13.11 | blitzrage | the quality is just whatever the codec is that you record in |
| 19:13.18 | danfromuk | SIP to SIP call. |
| 19:13.20 | blitzrage | and whatever the source format is |
| 19:13.22 | blitzrage | ok... |
| 19:13.28 | blitzrage | what is the codec? and what format are you recording into? |
| 19:13.41 | danfromuk | recording into a wav file |
| 19:13.52 | danfromuk | One moment while I check the codec |
| 19:14.58 | danfromuk | alaw |
| 19:15.47 | blitzrage | well that's about all you can do unless you can change the source to g722 or something |
| 19:16.20 | blitzrage | 8 bit, 8kHz mono can only sound so good :) |
| 19:16.20 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-hgvxkuxdofnlvprx) |
| 19:16.28 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 19:16.30 | blitzrage | I'd suggest changing the expectation of the quaity |
| 19:16.33 | blitzrage | quality of the recording |
| 19:16.40 | blitzrage | it's a layer 8 issue |
| 19:16.46 | danfromuk | Whats the next level up from alaw? |
| 19:16.52 | blitzrage | g722 |
| 19:16.57 | blitzrage | HD audio basically |
| 19:17.05 | blitzrage | phone needs to support it though |
| 19:17.20 | danfromuk | What level of bandwidth does g722 use? |
| 19:17.35 | blitzrage | google will tell you for sure, but it's comparable to ulaw/alaw |
| 19:18.04 | danfromuk | ok, no probs. thanks I'll check with the provider as both sides of the call are on their system. |
| 19:18.10 | blitzrage | k |
| 19:18.15 | blitzrage | runs off to build a crib |
| 19:22.24 | [TK]D-Fender | danfromuk: What exactly is on each end of the call? |
| 19:22.59 | [TK]D-Fender | danfromuk: If it's ALAW end-to-end and is on the PSTN then that's all you're going to get. |
| 19:24.50 | WIMPy | G.722 uses the same or less than G.711 |
| 19:26.57 | [TK]D-Fender | And changes nothing where your PSTN is only running G.711 |
| 19:27.05 | [TK]D-Fender | LCD |
| 19:30.31 | danfromuk | [TK]D-Fender: its a mobile to mobile call |
| 19:32.08 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 19:35.08 | [TK]D-Fender | danfromuk: then you have each call coming in over a shitty wireless code to the cell co, then over the PSTN to your ITSP where it is shuffled to you. Then mixed for recording and recompanded into ANOTHER format |
| 19:35.45 | [TK]D-Fender | danfromuk: It's going to suck. Nothing to "fix" |
| 19:36.14 | WIMPy | Shitty? It will probably be one of the better codecs. |
| 19:36.18 | danfromuk | [TK]D-Fender: I have checked the actual call quality and it sounds fine. Just the recordings sound muffled. |
| 19:38.16 | *** join/#asterisk boynas (~boynas@wsip-72-214-214-130.ph.ph.cox.net) |
| 19:38.40 | [TK]D-Fender | danfromuk: the MIXING is the part that will suck more from it |
| 19:38.44 | beefcafe | Unable to add Google ICE candidates as ICE support not available or no candidates available |
| 19:39.03 | beefcafe | what does this mean? |
| 19:41.07 | danfromuk | [TK]D-Fender: whats the alternative to mixing? is there one? |
| 19:41.21 | [TK]D-Fender | Don't |
| 19:41.43 | [TK]D-Fender | force left/right on the streams |
| 19:42.04 | [TK]D-Fender | And it'll have to be stereo |
| 19:45.16 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
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| 20:11.10 | boynas | I am having problems with dahdi not detecting a third quad PRI card. I have Red alarms, anybody feels like helping a guy in trouble? :) |
| 20:12.08 | WIMPy | Does the card have a working interrupt? |
| 20:12.13 | [TK]D-Fender | boynas: If you have alarms it's because it DOES see your card.... |
| 20:13.32 | jpsharp | red alarm = seeing your card, not seeing your T-span. |
| 20:14.27 | WIMPy | Do the lines work in one of the other cards? |
| 20:24.59 | *** part/#asterisk onebitxajax (~onebitxaj@unaffiliated/onebitxajax) |
| 20:45.23 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 20:47.57 | beefcafe | is there a way to cut the "+" in "+180000000000"? |
| 20:54.25 | [TK]D-Fender | ${VAR:1} <- returns the value of the var less 1 char |
| 20:54.35 | [TK]D-Fender | beefcafe: Keep readin your dialplan basics. |
| 20:54.40 | [TK]D-Fender | ~book |
| 20:54.40 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:55.04 | beefcafe | [TK]D-Fender: found a way. thanks:) |
| 20:56.03 | *** join/#asterisk mathi (~Matthew@ip-213-49-238-253.dsl.scarlet.be) |
| 20:56.05 | mathi | hi |
| 20:57.05 | mathi | if I understood well, if I answer the call (with Answer() application), and then Dial(), I pay the cost of the transfer. But if I don't Answer() and Dial(), then I don't have to pay anything ? |
| 20:58.25 | [TK]D-Fender | mathi: Each leg costs you whatever it costs you |
| 20:58.49 | mathi | can you for once speak normal?) |
| 20:58.49 | [TK]D-Fender | Answering just means that the DIAL time is guaranteed to be billed |
| 20:59.09 | boynas | Yes guys, the lines worked with other cards. |
| 20:59.26 | mathi | [TK]D-Fender, so what I said was correct ? |
| 20:59.35 | boynas | It was an interrupt problem. Moved one of the cards to a different slot and worked. Thanks to all. |
| 20:59.40 | [TK]D-Fender | No.... |
| 20:59.55 | [TK]D-Fender | What you said was vague and says nothing about when the call does get answered |
| 21:00.12 | mathi | [TK]D-Fender, you speak to me like I was a native english. Can you say things more simple, please? |
| 21:00.27 | [TK]D-Fender | ... |
| 21:01.03 | [TK]D-Fender | ANSWERING before a dialout means that you will be billed for the dialing duration of the 2nd leg REGARDLESS of if they answer or not. |
| 21:02.03 | boynas | TKD: Just so you know, I was having red alarms with no service.. On the lights of the card there was a green light on the first Span the rest were off. this are TE420s (Thank you though) |
| 21:02.19 | mathi | and if I don't Answer() and Dial() ? |
| 21:02.31 | mathi | I mean, if I dial without aneswering |
| 21:02.43 | [TK]D-Fender | Then you probably won't be charged for the dialing time before answer on the remote end |
| 21:03.14 | mathi | [TK]D-Fender, but then I can Dial a number at the other end of the world, and the caller would have to pay? |
| 21:03.17 | WIMPy | mathi: The first call will get billed as soon ans it is answered. Either by you using the Answer() application or by the 2nd call being answered if you use Dial(). |
| 21:03.38 | [TK]D-Fender | mathiif I understood well, if I answer the call (with Answer() application), and then Dial(), I pay the cost of the transfer. But if I don't Answer() and Dial(), then I don't have to pay anything ? <- this statement however is DANGEROUSLY vague. If you call out and the other side answers you ARE paying for BOTH ENDS. |
| 21:03.45 | WIMPy | mathi: If YOU dial, YOU pay. |
| 21:03.59 | [TK]D-Fender | If you answer incoming = YOU PAY |
| 21:04.11 | WIMPy | Everyone pays what he dials. |
| 21:04.12 | [TK]D-Fender | When you dialout and bridge.. you pay BOTH. |
| 21:04.29 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 21:06.48 | mathi | ok so whenever I use Dial(), I will pay the call from my server to the number I dial ? |
| 21:07.01 | [TK]D-Fender | of course |
| 21:07.09 | WIMPy | Sure |
| 21:07.40 | mathi | ok, I got that, but then the call from the user's telephone to my server, I will never have to pay that right? that is the user paying |
| 21:07.54 | WIMPy | Yes. |
| 21:08.11 | WIMPy | Unless you have a 800 number, off course. |
| 21:08.12 | [TK]D-Fender | mathi: If YOU pay for inbound then you pay for that call |
| 21:08.47 | [TK]D-Fender | WIMPy: You are assuming inbound is free, and perhaps even an assumption that its the same service as it used by outbound.... |
| 21:08.59 | mathi | why wouldn't inbound be free ?? |
| 21:09.19 | mathi | I mean, if you call me on my phone, I don't have to pay (at least if it's local call) |
| 21:09.22 | [TK]D-Fender | mathi: If your service CHARGES for inbound, then you pay for that. If your service charges for outbound, you pay for that. That's 2 legs to account for. |
| 21:09.40 | mathi | I didn't know that it exists that you can pay inbound |
| 21:09.46 | WIMPy | mathi: US-American idiosyncracy |
| 21:09.59 | [TK]D-Fender | mathi: You are assuming what kind of product people use. |
| 21:10.14 | mathi | it's only analog phones, PSTN |
| 21:10.26 | mathi | the inbound calls are only from PSTN in my case |
| 21:10.29 | [TK]D-Fender | WIMPy: Really, what about Australia? How about China? South Africa? |
| 21:10.50 | [TK]D-Fender | mathi: from PSTN HOW? |
| 21:11.05 | WIMPy | Do they have such misfeatures as well? |
| 21:11.09 | mathi | [TK]D-Fender, I have a DIgium card getting inbound calls from PSTN |
| 21:12.03 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-92-39.pppoe.mtu-net.ru) |
| 21:12.18 | [TK]D-Fender | WIMPy: As the region where your viewpoint becomes increasingly smaller, everyone else is the norm and YOU are the "misfeature" |
| 21:12.52 | WIMPy | Was that a "no"? |
| 21:12.59 | [TK]D-Fender | mathi: If the call is coming in on that line you should already know what it costs you. When you dial out, you should also already know what it costs you. |
| 21:13.10 | mathi | [TK]D-Fender, "If you call out and the other side answers you ARE paying for BOTH ENDS." <- here you say that I would have to pay inbound call AND outbound call ? (as you say both end), but most generally I wouldn't have to pay inbound |
| 21:13.46 | WIMPy | If *YOU* call... |
| 21:14.05 | [TK]D-Fender | mathi: Your description was WEAK at the start and took several questions just to find out HOW the calls are arriving. |
| 21:14.35 | [TK]D-Fender | mathi: Your analog lines have their own rules. The ITSP's of the world also have their own. |
| 21:14.39 | mathi | i'm a programmer, I may not use the terms specific to telephony right, that's why the misunderstandings |
| 21:15.18 | [TK]D-Fender | mathi: If you think that the precise means you use for calling aren't important to answering that question... you have a REAL problem. |
| 21:15.45 | [TK]D-Fender | mathi: This is basic logic. If you're copper in -> copper out.. then you should already know what it costs and when. |
| 21:17.21 | [TK]D-Fender | [17:12]WIMPyWas that a "no"? <- it means that your question becomes devalidated once you become the tiny fringe element. And then every time you advise others based on that same limited world-view. |
| 21:17.59 | mathi | I would like to know when I need to answer (with Answer() application) the call. Because in my script I Answer() always right from the beginning, even if I Dial() right after. In this case I wonder if I need to use Answer() application |
| 21:18.15 | [TK]D-Fender | mathi: Depends if you have a REASON to answer or not. |
| 21:18.28 | [TK]D-Fender | mathi: Do you? |
| 21:18.42 | [TK]D-Fender | MatIs this call IN from your card and out another line on the same? |
| 21:19.23 | mathi | [TK]D-Fender, whether I use Answer() before Dial, or Dial imediately without Answer, there is no difference? As I would always have to pay the communication costs of server -> dialed number. So I can always Answer() in the very beginning ? |
| 21:19.37 | [TK]D-Fender | .... |
| 21:19.39 | [TK]D-Fender | you CAN. |
| 21:19.42 | [TK]D-Fender | You asked if you SHOULD |
| 21:19.47 | [TK]D-Fender | Please PAY ATTENTION |
| 21:20.12 | [TK]D-Fender | "In this case I wonder if I need to use Answer() application" <- Need != can |
| 21:20.58 | [TK]D-Fender | There are consequnces for each |
| 21:21.19 | mathi | what are the consequences ? I don't understand the effect of Answer() |
| 21:21.34 | [TK]D-Fender | Asterisk isn't the CONSEQUNCE. |
| 21:21.36 | [TK]D-Fender | its your LINE |
| 21:21.46 | [TK]D-Fender | Suppose *I* get billed for calling you..... |
| 21:22.08 | [TK]D-Fender | And you try bridging me out. The outbound call rings 60 seconds and I decide to give up. |
| 21:22.28 | [TK]D-Fender | Well if you ANSWER immediately, I get billed for the 60 seconds of the outbound ATTEMPT that didn't even get answered |
| 21:22.35 | [TK]D-Fender | I PAY = consequnce. |
| 21:23.20 | [TK]D-Fender | Do you have VOICEMAIL on that line? If you DON'T answer the call and just dial out... after X amount of time the telco will STEAL the call back for VM and *poof*, I'm gone |
| 21:24.08 | [TK]D-Fender | Asterisk isn't the "catch"... your CALLS are. Think about YOUR LINE. |
| 21:24.19 | [TK]D-Fender | This shouldn't have to be a question.... |
| 21:24.21 | mathi | I didn't understand the last part, steal the call and "VM?" |
| 21:24.59 | [TK]D-Fender | .... |
| 21:25.05 | [TK]D-Fender | VOICEMAIL. |
| 21:25.15 | mathi | ah I don't have a voicemail |
| 21:25.46 | [TK]D-Fender | Do you understand what will happen if you DON'T answer an incoming call for 2 minutes of attempted rining? |
| 21:25.52 | [TK]D-Fender | YOU should know if that's BAD or not. |
| 21:26.14 | [TK]D-Fender | And that if you answer, maybe the person who calls you DOES have to pay if you DO answer... but then waste time dialing out without a confirmed bridge |
| 21:26.17 | [TK]D-Fender | This is common sense |
| 21:26.22 | mathi | [TK]D-Fender, I got it... I think ti's bad because my line will be busy. So I better and Answer and put a limit ? |
| 21:26.38 | [TK]D-Fender | No, it's BUSY... RINGING |
| 21:26.42 | [TK]D-Fender | so busy either way |
| 21:27.00 | [TK]D-Fender | Still failing at common sense here.... |
| 21:28.34 | mathi | Do you understand what will happen if you DON'T answer an incoming call for 2 minutes of attempted rining? |
| 21:28.37 | mathi | => I don't know |
| 21:28.40 | mathi | can you tell me? |
| 21:29.00 | [TK]D-Fender | mathi: Do you think I know where you live or how your specific telephone company works? |
| 21:29.07 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:29.15 | [TK]D-Fender | mathi: This is YOUR telco. How do YOU not know? |
| 21:29.31 | [TK]D-Fender | mathi: Have you just moved to some foreing place and started working on a server there? |
| 21:29.36 | [TK]D-Fender | foreign* |
| 21:29.54 | WIMPy | 120s used to be the standard ringing expiry time. |
| 21:30.03 | mathi | what can be the problem that it is ringing for 2 mins ? |
| 21:30.53 | [TK]D-Fender | Some telcos may give up and say FUCKIT... even though they'd LIKE to try for longer |
| 21:31.04 | nightrid3r | if you keep me ringing of half that time i'll take my bussiness somewere else |
| 21:31.35 | [TK]D-Fender | mathi: You should really go and lean how your own phone lines work.... |
| 21:31.56 | [TK]D-Fender | maybe your CALLER has a timeout limitation imposed on them <- |
| 21:32.47 | ChannelZ | .. |
| 21:32.57 | ChannelZ | oops |
| 21:33.21 | mathi | well, what is your point? Ok the ringing time expires, the call is ended, and how is that a problem? |
| 21:34.32 | jpsharp | Whoops. "An error occurred during a connection to www.asterisk.org. Peer's certificate has been revoked." |
| 21:34.55 | mathi | I prefer to Answer() and Dial() with a expiry time in seconds, and after 15 secs say *FUCKIT*, rather than having my line busy for e.g. 2 mins (depends on tel company as you said) |
| 21:36.13 | [TK]D-Fender | mathi: Hope you thought long and hard on that..... 15 seconds to call the outside world sounds BAD given so many VM's may only trigger in 20s..... |
| 21:36.24 | [TK]D-Fender | 2 min may be long, but 15s sure is SHORT |
| 21:36.29 | WIMPy | Ah, that was the question. Well, the answer is: Go digital. Any way. |
| 21:36.29 | [TK]D-Fender | Time to start THINKING here.... |
| 21:36.54 | [TK]D-Fender | WIMPy: that isn't even an analog VS digital question |
| 21:37.04 | [TK]D-Fender | Dialing/billing time is just time. |
| 21:37.09 | WIMPy | Sure it is. |
| 21:37.12 | [TK]D-Fender | medium is unimportant |
| 21:37.20 | [TK]D-Fender | facepalms... |
| 21:37.33 | WIMPy | On analog you can't end a call without accepting it. |
| 21:37.48 | mathi | 15 secs short?? Nobody is going to wait more than 15 secs anyway, it seems long enough to me!! |
| 21:38.27 | [TK]D-Fender | WIMPy: Excepet maybe a circuit cut. "maybe" |
| 21:38.43 | [TK]D-Fender | WIMPy: I made a module for someone to do that. |
| 21:38.47 | [TK]D-Fender | a LONG time ago |
| 21:39.21 | mathi | and I don't use Voicemails |
| 21:39.28 | [TK]D-Fender | And this was one odd scenario. In his case the worst would mean an answer+immediate hangup |
| 21:39.28 | WIMPy | Maybe. Probably not. |
| 21:39.42 | [TK]D-Fender | WIMPy: True, YMMV on that. |
| 21:39.42 | WIMPy | yes |
| 21:40.29 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 21:46.46 | *** join/#asterisk bandroidx (~bandroidx@205.185.117.117) |
| 21:50.02 | *** join/#asterisk factormystic (~factormys@c-24-128-2-238.hsd1.ma.comcast.net) |
| 21:53.45 | [TK]D-Fender | WIMPy: /me heads out for a while |
| 21:53.50 | [TK]D-Fender | heads out for a while |
| 21:53.52 | [TK]D-Fender | dang leftovers... |
| 21:54.09 | factormystic | is the motif channel driver part of asterisk 11? I don't see motif.conf in /etc/asterisk and I can't tell if that's ok or if it means motif is missing |
| 21:58.02 | file | it is, but the configuration file is only installed if you do make samples |
| 21:58.09 | file | cause that overwrites ALL config files |
| 21:58.22 | file | you can copy the sample one in manually and modify it, same goes for res_xmpp |
| 22:01.32 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-92-39.pppoe.mtu-net.ru) |
| 22:02.14 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.88.210) |
| 22:03.12 | factormystic | thanks |
| 22:26.54 | factormystic | followup question, if I run "module show like chan_" I don't see anything called chan_motif... but I do see chan_gtalk and chan_jingle which I thought motif was supposed to replace |
| 22:27.06 | factormystic | that's not really a question I guess but a statement of confusion :v |
| 22:30.09 | factormystic | for context I'm going through https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google with Asterisk 1.8.10.1~dfsg-1ubuntu1 installed via apt-get in ubuntu 12.04 |
| 22:35.54 | ChannelZ | Check your 'make menuconfig' and make sure it's enabled and built (or look in your asterisk lib directory and see if it's there) |
| 22:38.27 | file | Asterisk 1.8 does not have chan_motif |
| 22:38.52 | ChannelZ | I thought he said earlier he was on 11 |
| 22:38.58 | factormystic | I thought I was |
| 22:39.10 | ChannelZ | if you are using a package, you are not. |
| 22:39.29 | ChannelZ | 11 is still beta |
| 22:40.29 | factormystic | ok, so in order to get asterisk 11 I need to build from source looks like, so I'll need to remove the package install first I guess |
| 22:40.29 | ChannelZ | 1.8 or 10 would still be using chan_gtalk |
| 22:42.28 | ChannelZ | (which is fine by the way) |
| 22:43.26 | factormystic | oh, the red header at the top of the wiki page implied that 11 was the way to go |
| 22:43.54 | factormystic | "The old page documents behavior that is not functional" |
| 22:44.01 | ChannelZ | It will be in the future |
| 22:44.24 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:44.38 | ChannelZ | Albeit soon future.. Asterisk 11 looks like it's in release candidate. But all I'm saying is that chan_gtalk and jabber work fine, I'm using them. |
| 22:44.45 | factormystic | ok |
| 22:46.28 | ChannelZ | that said if you're just farting around there's no reason to NOT play with Asterisk 11 either. |
| 22:46.36 | factormystic | I'm going to try for pstn -> google voice -> google talk -> asterisk -> sip soft phone |
| 22:47.08 | *** join/#asterisk jsjc (~Adium@96.pool85-51-81.dynamic.orange.es) |
| 22:47.13 | factormystic | I've go the first three stages there working, its asterisk onwards which is new to me |
| 22:49.23 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 23:23.55 | factormystic | ah geez sip.conf is big |
| 23:25.43 | ChannelZ | The sample configs are like manuals; they contain EVERY possible option and many examples. It's not as bad as it looks. |
| 23:26.18 | ChannelZ | An average peer can be setup in less than 10 lines |
| 23:35.26 | robl^ | factormystic: a real world sip.conf on one of my boxes is about 40 lines total. that includes 2 softphones, linking to another pbx and to a SIP itsp and a handful of comments |
| 23:37.15 | robl^ | I missed the announcement. I am still on 11.0.0 Beta 2 |
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| 23:57.25 | *** join/#asterisk dolcea-xoom (~androirc@232-197-235-201.fibertel.com.ar) |
| 23:57.58 | dolcea-xoom | Hi, is there a way to restric a sip-user to a macaddress? |