| 00:04.08 | *** join/#asterisk jetlag (~jetlag@pool-71-168-253-208.cmdnnj.east.verizon.net) |
| 00:04.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:11.21 | *** join/#asterisk CrazyTux[m] (~Brandon@108-64-56-113.lightspeed.irvnca.sbcglobal.net) |
| 00:11.59 | dijib | hey all does p3nguin ever come in here any more? |
| 00:15.57 | *** part/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 00:24.33 | *** part/#asterisk D3 (~d3@109.69.14.162) |
| 00:34.22 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 00:35.01 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 00:40.29 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 00:49.23 | *** join/#asterisk luckman212 (~irc@2001:470:8abb:0:8c4c:ca87:2fc8:2a46) |
| 00:49.25 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 01:15.52 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:15.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:22.15 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 02:18.49 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:18.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:47.56 | *** join/#asterisk jsjc (~Adium@49.Red-81-43-165.staticIP.rima-tde.net) |
| 02:52.09 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 03:03.51 | *** join/#asterisk CrazyTux[m] (~Brandon@cpe-66-27-167-245.socal.res.rr.com) |
| 03:18.00 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 03:21.55 | *** join/#asterisk zamba (marius@flage.org) |
| 03:32.34 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-gclbedbiescmbcmg) |
| 03:34.15 | *** join/#asterisk Digweed (digweed@2a00:f10:11a:764::c0ca:c01a) |
| 03:34.15 | *** join/#asterisk carrar (tim@osburn.com) |
| 03:34.15 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:34.15 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 03:34.15 | *** join/#asterisk enki (jbabiak@kahuna.routed.com) |
| 03:34.15 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 03:34.15 | *** join/#asterisk eronel (eronel@nutella.eronel.org) |
| 03:34.15 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 03:34.15 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:34.15 | *** mode/#asterisk [+oo file leifmadsen] by zelazny.freenode.net |
| 03:34.18 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pqstwvecqgoriazw) |
| 03:35.54 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 03:36.39 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 03:58.58 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 04:35.17 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 04:36.07 | *** join/#asterisk vn (~sys6x@nostalgeek.net) |
| 04:37.12 | vn | hi, I'm behind NAT and I use 2 VoIP providers, Line 1 works fine in/out, but Line 2 only works going out...any tips on how I should configure NAT? SIP Line 2 uses tcp 5061 and I think my provider doesnt like that |
| 04:37.22 | *** join/#asterisk MrNemus (~Nemus@c-76-23-30-50.hsd1.ut.comcast.net) |
| 04:40.11 | *** join/#asterisk jsjc (~Adium@49.Red-81-43-165.staticIP.rima-tde.net) |
| 04:41.26 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 04:48.24 | ChannelZ | are you talking about directly from a phone or something? |
| 04:50.29 | vn | nvm, I fixed it using EXT SIP port |
| 04:50.42 | vn | dual NAT port change...lame but it works |
| 04:50.49 | vn | but yep from a phone |
| 05:00.04 | *** part/#asterisk vn (~sys6x@nostalgeek.net) |
| 05:05.43 | *** join/#asterisk PCS (~root@50.8.206.176) |
| 05:06.30 | PCS | is there a history for this channel? |
| 05:07.15 | PCS | can someone recomend a good book? |
| 05:07.25 | PCS | for getting started with asterisk? |
| 05:12.57 | kaldemar | ~book |
| 05:12.57 | 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 |
| 05:13.32 | PCS | kaldemar: thanks! |
| 05:30.39 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 05:35.36 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:36.11 | *** join/#asterisk jsjc (~Adium@49.Red-81-43-165.staticIP.rima-tde.net) |
| 06:02.33 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 06:02.45 | v0lZy | morning |
| 06:05.30 | diab | Matin :) |
| 06:08.02 | ChannelZ | jumps off the table |
| 06:10.39 | kikohnl | Aloha kakahiaka |
| 06:12.41 | ChannelZ | Mecka-lecka hi mecka hiney-ho |
| 06:20.22 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:25.59 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:26.01 | schmidts | good morning |
| 06:28.01 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 06:31.13 | ChannelZ | Mecka-lecka hi mecka hiney-ho to you too! |
| 06:37.15 | bulkorok | hi |
| 06:38.06 | Roelt | And all the goyim say I'm pretty fly for a rabbi |
| 06:38.22 | *** join/#asterisk UQlev (~chatzilla@46.251.117.25) |
| 06:39.26 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 06:43.29 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 06:46.00 | v0lZy | ohayou gozaimasu! |
| 06:48.48 | ChannelZ | Gesundheit |
| 06:49.41 | schmidts | :D |
| 06:52.16 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 06:52.29 | v0lZy | :D |
| 06:59.52 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 07:11.12 | *** join/#asterisk kleszcz (tick@176.111.234.237) |
| 07:11.25 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b4ff:fbfa:fd7c:cbff) |
| 07:18.28 | v0lZy | anyone here ever tried freeswitch? |
| 07:19.55 | v0lZy | I thought asterisk had a lot of stuff to configure |
| 07:20.24 | v0lZy | im just reading the wiki.... its a kaleidoscope :D |
| 07:21.05 | *** join/#asterisk UQlev (~chatzilla@46.251.117.25) |
| 07:23.13 | diab | :))) |
| 07:23.40 | v0lZy | anyone tried? |
| 07:23.45 | v0lZy | any opinions? |
| 07:26.07 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e125.pool.mediaWays.net) |
| 07:41.15 | bulkorok | freeswitch is great |
| 07:42.06 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 07:42.30 | bulkorok | up 42 days, 1760357 sessions since startup |
| 07:43.21 | bulkorok | my personal best feature is failover to a second machine without dropping calls! |
| 07:44.21 | v0lZy | sounds nifty |
| 07:44.24 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 07:44.37 | v0lZy | thinking of trying it out |
| 07:44.49 | v0lZy | right now im trying to configure page to work though |
| 07:44.50 | bulkorok | let's me sleep very very good! |
| 07:45.09 | v0lZy | (with asterisk) |
| 07:45.22 | v0lZy | and its dropping :( |
| 07:46.07 | bulkorok | frustrating :-/ |
| 07:47.41 | bulkorok | http://wiki.freeswitch.org/wiki/Freeswitch_HA |
| 07:48.07 | bulkorok | (enough advertising... :) |
| 07:48.31 | v0lZy | whats the module to load? |
| 07:48.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 07:49.00 | v0lZy | app_page.so |
| 07:49.12 | v0lZy | i dont see it in /usr/lib/asterisk/modules :( |
| 07:49.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:49.51 | v0lZy | im using 10.5.1 though |
| 07:50.06 | bulkorok | mmh... 1.4 had it... I thnink |
| 07:50.31 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 07:50.50 | bulkorok | https://wiki.asterisk.org/wiki/display/~jcolp/Conversion+of+app_page+to+ConfBridge |
| 07:52.42 | bulkorok | this should be the one you need: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 07:53.27 | v0lZy | ah, so its no longer Page . |
| 07:53.31 | v0lZy | but confrence bridge |
| 07:53.33 | v0lZy | ok, got it. |
| 07:54.16 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:54.31 | kaldemar | Page hasn't gone anywhere, those are just different approaches. |
| 07:57.04 | v0lZy | I dont have it. |
| 07:57.08 | v0lZy | app_page.so .. . |
| 07:57.16 | v0lZy | so... have to use confbridge. |
| 07:57.36 | kaldemar | or you could just load/add it. |
| 07:59.07 | v0lZy | its not in the modules directory |
| 07:59.27 | *** join/#asterisk black187 (~black187@77.234.149.118) |
| 07:59.27 | v0lZy | i just trying a systems wide search for app_page.so |
| 07:59.28 | v0lZy | no hits. |
| 07:59.38 | kaldemar | don't bother. you don't have it compiled. |
| 08:00.30 | black187 | Hi, did anybody test WebRTC on Asterisk 11 beta -> how can I specify the port used for TCP Websocket connection. Only changes I see, is in the sip.conf - transport=ws or transport=wss? |
| 08:05.03 | v0lZy | kaldemar: From what I gather, if i use confbridge.conf... there i can define a type of conference bridge and a user profile.... then from my dialplan i start a conference at say 1234, and define that i want to use that bridge profile and users who are joining the conference to use the user profile i define. |
| 08:05.36 | v0lZy | then I use SipAddHeader(Call-info: sip:hostname\;auto-answer=0) |
| 08:05.39 | v0lZy | and use dial? |
| 08:06.01 | v0lZy | I mean.. how do i force join them. |
| 08:06.06 | v0lZy | to the conference bridge? |
| 08:12.47 | kaldemar | v0lZy: you'd have to originate calls to participants. |
| 08:12.57 | kaldemar | v0lZy: what version are you using again? |
| 08:13.04 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 08:13.16 | bulkorok | <v0lZy> im using 10.5.1 though |
| 08:14.47 | kaldemar | in 10.5.1 the Page app still depends on Meetme. |
| 08:15.01 | kaldemar | and Meetme depends on DAHDI. |
| 08:15.49 | kaldemar | either use those or use originates and a ConfBridge conference. |
| 08:15.50 | v0lZy | but i dont have it compiled. |
| 08:16.08 | v0lZy | originate calls to participants... |
| 08:16.35 | v0lZy | how do i do that? dial(SIP/1&SIP2&....,,o) ? |
| 08:17.37 | v0lZy | oh |
| 08:17.40 | v0lZy | theres an app Originate. |
| 08:21.43 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:23.02 | *** join/#asterisk lkcl (~lkcl@host86-131-180-106.range86-131.btcentralplus.com) |
| 08:29.00 | *** join/#asterisk Jasnejac (~EnorMouse@2001:b70:500:2:3947:cd6e:d194:e069) |
| 08:36.09 | v0lZy | Im reading the Originate desription |
| 08:36.49 | v0lZy | tech_data is the technology and extension im calling? |
| 08:36.54 | v0lZy | if i want to call extension 1234 |
| 08:37.03 | v0lZy | with sip tech |
| 08:37.15 | v0lZy | then i do Originat(SIP/1234 ? |
| 08:37.48 | v0lZy | or is this the extension I want to originate from? |
| 08:38.04 | v0lZy | say.. if i want to call from SIP/1234 I start like Originate(SIP/1234... |
| 08:41.50 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:43.57 | kaldemar | v0lZy: you need to learn what an extension is in asterisk. |
| 08:45.09 | v0lZy | the way i understand its just a holder for associating reactions to |
| 08:45.55 | kaldemar | if you want to "dial" and extension, that is done with Local/exten@context. |
| 08:47.22 | kaldemar | if you dial SIP/1234, it will dial a DEVICE by the name 1234 if you have such configured in sip.conf. else, 1234 is assumed to be an address. |
| 08:48.05 | kaldemar | extension 1234 would be an extension in some context in your dialplan. it may not have anything to do with SIP/1234. |
| 08:51.03 | v0lZy | ok |
| 08:51.12 | v0lZy | so Dial shoots it straight to the device over the network |
| 08:51.27 | v0lZy | if i specify it like SIP/1234 |
| 08:51.58 | v0lZy | dial(local/44@context) goes to context specified and provides 44 as the extension |
| 08:53.10 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 08:55.04 | v0lZy | im getting the behaviour I want now |
| 08:55.12 | v0lZy | with just using Dial(local etc |
| 08:55.15 | v0lZy | but |
| 08:55.19 | v0lZy | how do i mute the other end. |
| 08:58.06 | kaldemar | so the behavior you want is to make all phones ring but only pass audio to the first one that answers? |
| 08:58.20 | *** join/#asterisk jkroon (~jkroon@196.38.199.207) |
| 08:58.24 | kaldemar | that's what happens if you use a Dial. |
| 08:58.53 | v0lZy | no |
| 08:58.57 | v0lZy | all phones answer |
| 08:58.59 | v0lZy | automatically |
| 08:59.04 | v0lZy | and i pass audio to them |
| 08:59.06 | v0lZy | announcer speaker. |
| 08:59.13 | kaldemar | with Dial, the first one to answers gets the call. |
| 08:59.15 | v0lZy | but the thing is that some of them could have call forwarding activated. |
| 08:59.44 | v0lZy | right now |
| 08:59.46 | v0lZy | im doing |
| 09:00.26 | v0lZy | n,Dial(local/44@AppInternalCalls&loca/50@AppInternalCalls) |
| 09:01.24 | v0lZy | or rathern instead AppInternalCalls im using SIP16 there which basically only includes AppInternalCalls |
| 09:01.43 | v0lZy | problem is that AppInternalCalls does some call forwarding checking before it hooks things up |
| 09:02.20 | kaldemar | so you add an extra context to get some unnecessary processing and end up with the same result. |
| 09:02.46 | v0lZy | im just experimenting |
| 09:02.57 | v0lZy | ideally id want to dial them directly i think |
| 09:04.12 | v0lZy | but like u said, im only sending to the first one that picked up |
| 09:04.18 | v0lZy | which is not what i want |
| 09:04.21 | v0lZy | want to send to all of them |
| 09:05.46 | v0lZy | seems to me the way to go is to initiate a conference and then call all the phones and have them connected to it |
| 09:05.52 | v0lZy | with autoanswer in the sip header |
| 09:10.15 | kaldemar | too bad the SipAddHeader app does not affect invites that are the product of a following Originate app. |
| 09:10.40 | v0lZy | so in other words |
| 09:10.44 | v0lZy | its not gonna help |
| 09:11.14 | v0lZy | how can i go and to this then... |
| 09:11.22 | v0lZy | Have all phones activate speakers at the same time. |
| 09:11.42 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:11.57 | kaldemar | Page can do that. |
| 09:12.46 | kaldemar | or you can use an extension as the originate target. which would set the header for auto-answer. |
| 09:13.15 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:13.38 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:13.40 | v0lZy | Page is off the table so... second thing |
| 09:13.51 | v0lZy | use an extension as the originate target which would set the header for auto-answer... |
| 09:14.39 | v0lZy | im doing the SIPAddHeader right now |
| 09:14.54 | v0lZy | i just dont know what my next step should be |
| 09:15.09 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:15.55 | bulkorok | is t38 faxing working with g729 codec?! |
| 09:16.03 | v0lZy | logically, to call the phones? no |
| 09:16.42 | kaldemar | originate calls to an extension that sets the header and dials a phone. |
| 09:20.36 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:29.24 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:32.59 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:37.47 | v0lZy | this originate thing has me confused. |
| 09:37.50 | v0lZy | Im doing |
| 09:38.27 | v0lZy | Originate(SIP/44,AppAnnounce,AutoAnswer,1) |
| 09:38.29 | v0lZy | then i have |
| 09:39.17 | v0lZy | exten => AutoAnswer,1,SIPAddHeader(Call-info: sip:myhost.domain;\answer-after=0) |
| 09:39.26 | v0lZy | same => Dial(SIP/44) |
| 09:39.44 | v0lZy | when i call this extension with my phone, it rings 44 |
| 09:40.04 | v0lZy | 44 sees 'anonymous' as caller id... (ok, i havent set any variables so makes sense) |
| 09:40.07 | v0lZy | but its ringing |
| 09:40.10 | v0lZy | it doesnt auto answer |
| 09:40.12 | Gugge | bulkorok: t38 uses its own "codec" |
| 09:41.34 | kaldemar | v0lZy: you have a syntax error in your Originate. |
| 09:41.49 | v0lZy | ah |
| 09:41.53 | v0lZy | im forgetting ,exten, |
| 09:42.08 | v0lZy | so |
| 09:42.17 | v0lZy | Originate(SIP/44,exten,AppAnnounce,AutoAnswer,1) |
| 09:42.24 | v0lZy | same thing though |
| 09:42.31 | v0lZy | i dont think it ever hits AutoAnswer |
| 09:42.44 | kaldemar | v0lZy: and seems like you don't fully understand what Originate does. now that would dial SIP/44, and upon answer connect it to AutoAnswer@AppAnnounce, which would dial SIP/44 again. |
| 09:43.18 | v0lZy | ok, so the first one is the target... |
| 09:43.51 | v0lZy | i change 44 up there to say... 16 |
| 09:44.00 | kaldemar | where? |
| 09:44.05 | v0lZy | in the originate |
| 09:44.27 | v0lZy | then it should dial 16 and connect it to 44... right? |
| 09:44.42 | kaldemar | when SIP/16 answers, yes. |
| 09:45.05 | kaldemar | but that doesn't sound at all like what you need. |
| 09:45.25 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 09:45.30 | v0lZy | well what happens though is |
| 09:45.49 | v0lZy | that it calls sip/16, and when 16 picks up... i get a busy |
| 09:46.08 | v0lZy | and 44 isnt involved anywhere. |
| 09:46.19 | kaldemar | if you have a problem, show it. |
| 09:47.11 | v0lZy | What i want to do is |
| 09:47.17 | v0lZy | 1) set up conference |
| 09:47.22 | kaldemar | fail |
| 09:47.39 | v0lZy | ergh, let me rephrase |
| 09:47.56 | kaldemar | you don't need to "initiate" a conference. just configure it. then originate calls to the phones and connect the other end to your configured conference. |
| 09:47.57 | *** join/#asterisk mirela666 (~mirko@212.200.146.253) |
| 09:48.16 | v0lZy | When I dial *** from 16 i want it to enter a conference to which I then want to join another phone (44) |
| 09:48.30 | kaldemar | stop explaining and rephrasing, show what you really have. |
| 09:48.39 | v0lZy | I set up a conference with Confbridge(1234) |
| 09:51.08 | v0lZy | well its all messed up right now cause im trying to figure out whats going on |
| 09:51.10 | v0lZy | but this is it |
| 09:51.12 | v0lZy | http://bpaste.net/show/WTivMKjTdYkCjemwag4g/ |
| 09:52.40 | v0lZy | now i replace SIP/16 there with SIP/1234 |
| 09:53.39 | kaldemar | why? |
| 09:53.56 | v0lZy | well cause I want to bridge 44 there. |
| 09:54.11 | v0lZy | but as soon as i do ConfBridge... thats where the dialplan seems to stop |
| 09:54.16 | v0lZy | it doesnt do any originating |
| 09:54.20 | kaldemar | do you have a device by the name 1234, i.e. [1234] in sip.conf? |
| 09:54.28 | v0lZy | no |
| 09:54.30 | v0lZy | but |
| 09:54.38 | kaldemar | no buts. SIP/1234 is wrong. |
| 09:54.54 | mirela666 | v0lZy: yes cause the only way to leave ConfBridge is by hangup |
| 09:54.56 | v0lZy | when you do Confbridge(1234) it creates a conference bridge 1234... |
| 09:54.57 | kaldemar | in SIP/1234, 1234 is not an extension. nor is it a conference. |
| 09:55.18 | kaldemar | ConfBridge is a blocking application, you must execute Originate before it. |
| 09:55.33 | mirela666 | kaldemar is right |
| 09:55.35 | v0lZy | i can get the other user into conference 1234 if they call the AppAnnounce extension |
| 09:56.11 | kaldemar | AppAnnounce is a context, not an extension. |
| 09:56.59 | v0lZy | yes, but i call it with Gosub that matches *** |
| 09:57.03 | v0lZy | and then runs that. |
| 09:57.23 | v0lZy | i call *** from my phone which hits a gosub to AppAnnounce |
| 09:58.36 | mirela666 | you set up feature? |
| 09:58.44 | mirela666 | '***' |
| 09:58.51 | kaldemar | v0lZy: http://bpaste.net/show/DIqv8tScFeujJSh3dbky/ |
| 10:01.12 | kaldemar | v0lZy: do you now get what an Originate is? |
| 10:02.02 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 10:02.11 | v0lZy | no, i didnt setup a featuer |
| 10:02.17 | v0lZy | looking |
| 10:02.19 | v0lZy | sec |
| 10:02.54 | v0lZy | Originate(Local/connect44@AppAnnounce,app,Confbridge,1234) |
| 10:03.57 | v0lZy | it calls 44 and when it answers, it launches the confbridge app and tells it to use 1234 |
| 10:05.37 | mirela666 | kaldemar, I have one question for you :) |
| 10:06.40 | mirela666 | I use macros and exit code from macro to macro is not updating ${MACRO_EXTEN} and macro to context is updating ${EXTEN} |
| 10:08.44 | kaldemar | mirela666: use Gosub instead of Macro and use the STACK_PEEK function. |
| 10:09.27 | mirela666 | Gosub is isuing WARNING of call flow control |
| 10:09.40 | kaldemar | macros have been deprecated twice now. :P |
| 10:09.41 | v0lZy | uhm |
| 10:09.43 | v0lZy | can i use |
| 10:09.46 | v0lZy | ${EXTEN:10} |
| 10:09.54 | mirela666 | yeah i know :) |
| 10:09.55 | v0lZy | or does it only go up to 9 ? |
| 10:10.17 | kaldemar | v0lZy: now that's quick to test, isn't it? |
| 10:10.36 | v0lZy | yeah but not in my current context |
| 10:10.53 | mirela666 | v0LZy, you can use ${STRING:X} where X is as manu chars as you want to trim from begining |
| 10:10.53 | kaldemar | bullshit. just do it. |
| 10:10.54 | v0lZy | its returning 1 |
| 10:11.30 | v0lZy | hm |
| 10:11.32 | v0lZy | if i use |
| 10:11.44 | v0lZy | ${EXTEN} it says AnnounceAll |
| 10:11.56 | mirela666 | search for variables on voip-info.org |
| 10:12.04 | v0lZy | if i use ${EXTEN:10} it says 1 |
| 10:12.12 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 10:12.14 | v0lZy | should say l. |
| 10:13.10 | v0lZy | ah |
| 10:13.12 | v0lZy | stupid font |
| 10:13.15 | v0lZy | l and 1 look exactly the same |
| 10:14.34 | v0lZy | thing is though |
| 10:14.49 | v0lZy | originate calls the conference thing and i don tthink the plan continues. |
| 10:15.04 | mirela666 | hehe comicsans fonf ? :P |
| 10:15.48 | v0lZy | i dont get 44 into the conference |
| 10:16.00 | v0lZy | <PROTECTED> |
| 10:16.01 | kaldemar | v0lZy: CLI output with verbosity will show what happens. |
| 10:16.02 | v0lZy | <PROTECTED> |
| 10:16.03 | v0lZy | <PROTECTED> |
| 10:16.05 | v0lZy | <PROTECTED> |
| 10:16.06 | v0lZy | <PROTECTED> |
| 10:16.08 | v0lZy | SHINOBI*CLI> |
| 10:16.49 | kaldemar | and what made you think the dialplan would not continue after the Originate? |
| 10:17.13 | v0lZy | well... i mean, it does continue, but it doesnt dial 44 at any point as u can see. |
| 10:17.47 | v0lZy | it gets to originate with the whole call AutoAnswer44 and then confbridge 1234 |
| 10:18.05 | v0lZy | but then obviously, it doesnt call 44 |
| 10:18.13 | v0lZy | it does say confbridge-join.gsm there.. |
| 10:18.19 | v0lZy | on Bridge/0xb6d... |
| 10:18.28 | kaldemar | what do you think you should be doind at this point? |
| 10:18.29 | v0lZy | but no dialing of 44 what so ever |
| 10:19.01 | kaldemar | pastebin your dialplan... |
| 10:19.31 | v0lZy | http://bpaste.net/show/nh8dEsPGjiA1SsAQ3fGO/ |
| 10:20.49 | v0lZy | erm |
| 10:20.56 | v0lZy | Confbridge vs ConfBridge maybe |
| 10:21.03 | kaldemar | no. |
| 10:21.21 | kaldemar | dialplan show AppAnnounce |
| 10:22.25 | v0lZy | bpaste.net/show/dc58JrFs2S661URMzTij/ |
| 10:23.10 | v0lZy | http://bpaste.net/show/dc58JrFs2S661URMzTij/ |
| 10:23.12 | v0lZy | sorry. |
| 10:27.10 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 10:27.11 | kaldemar | the named extension screws you over there. |
| 10:27.18 | kaldemar | _[A][u][t][o][A][n][s][w][e][r]XX would work. |
| 10:27.30 | v0lZy | <PROTECTED> |
| 10:28.46 | v0lZy | so... thats failing too |
| 10:29.21 | v0lZy | sipaddheader thingy is on priority 1 |
| 10:31.46 | v0lZy | ah |
| 10:31.47 | v0lZy | doh |
| 10:31.49 | v0lZy | theres an n there |
| 10:32.09 | kaldemar | the "Spawn extension..." is not an error of any kind. |
| 10:32.22 | v0lZy | ok works now |
| 10:32.28 | v0lZy | it was an N thing |
| 10:32.53 | mirela666 | ...1,n,... |
| 10:33.20 | v0lZy | yeah |
| 10:33.22 | v0lZy | fixed that |
| 10:33.25 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:34.45 | v0lZy | now to sort out some other stuff |
| 10:34.52 | mirela666 | :) |
| 10:34.53 | v0lZy | when i drop the call |
| 10:34.56 | v0lZy | this call doesnt drop |
| 10:35.03 | v0lZy | the people i invited into the conference,... stay in the conference |
| 10:35.25 | v0lZy | when I hangup, they should hangup too |
| 10:35.53 | v0lZy | i tried adding n,Hangup() there |
| 10:36.05 | v0lZy | but i think i need to add a g or something to Dial |
| 10:36.45 | v0lZy | uhm.. nope |
| 10:38.31 | v0lZy | hopefully drop it in a sense that it wount indicate a busy tone... |
| 10:40.08 | kaldemar | v0lZy: what you should be looking at is ConfBridge. specifically parameters marked and end_marked. you need two user profiles. one for the one that initiates the conference and another for the rest. |
| 10:40.34 | kaldemar | v0lZy: startmuted is something you probably want. |
| 10:40.48 | v0lZy | it mutes everyone |
| 10:40.55 | kaldemar | no. |
| 10:41.08 | v0lZy | how do i pass more than 1 parameter to confbridge with Originate? |
| 10:41.16 | kaldemar | not everyone. only the ones that have it in their user profile. |
| 10:41.23 | v0lZy | right now im passing 1234, but i need to pass more |
| 10:41.28 | v0lZy | well its using the default profile |
| 10:41.34 | v0lZy | so im muting everyone |
| 10:42.29 | v0lZy | it says bridge_profile and user_profile are also passable |
| 10:42.44 | v0lZy | though how can i set one user as one user profile and the other as an admin or soemthing |
| 10:43.47 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 10:44.37 | kaldemar | app->exten |
| 10:46.34 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 10:47.57 | v0lZy | what do u mean? |
| 10:48.23 | v0lZy | i see theres an end-marked |
| 10:48.31 | v0lZy | to kick all users after the last marked user exits |
| 10:48.48 | kaldemar | change app to exten and do what you want in an extension. you can enclose multiple arguments to the app in quotes though. |
| 10:53.31 | v0lZy | mutliple arguments to the app in quotes |
| 10:53.33 | v0lZy | like |
| 10:53.46 | v0lZy | '1234','arg','arg2' etc? |
| 10:53.55 | v0lZy | or just |
| 10:54.02 | v0lZy | '1234 arg2 arg3 arg4' |
| 10:54.17 | *** join/#asterisk lkcl (~lkcl@host86-131-180-106.range86-131.btcentralplus.com) |
| 10:54.40 | mirela666 | ,app,Confbridge,1234,wAa |
| 10:54.44 | mirela666 | ... |
| 10:55.07 | mirela666 | like options for ConfBridge |
| 10:59.56 | kaldemar | mirela666: the wAa in that one would be ignored. |
| 10:59.59 | v0lZy | hm |
| 11:00.08 | kaldemar | "arg,arg2,arg3" |
| 11:00.16 | v0lZy | double quotes or single quotes? |
| 11:06.23 | v0lZy | i think i got it working |
| 11:07.46 | v0lZy | double quotes :D |
| 11:07.49 | v0lZy | thanks kaldemar |
| 11:07.52 | v0lZy | thanks mirela666 |
| 11:08.01 | v0lZy | i think i have all the bits to create an intercom now |
| 11:12.13 | mirela666 | no prob :) |
| 11:27.00 | v0lZy | btw, is there a way to sound a beep after a certain time? |
| 11:27.11 | v0lZy | after the beep is called that is |
| 11:27.16 | v0lZy | like putting it in the background |
| 11:30.37 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 11:32.42 | *** join/#asterisk sideffect_ (sideffect@gateway/shell/bshellz.net/x-osweqxhkadxrbbki) |
| 11:33.25 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:39.20 | *** join/#asterisk sideffect_ (sideffect@gateway/shell/bshellz.net/x-agpbholtaszfotwx) |
| 11:41.05 | *** join/#asterisk jsjc (~Adium@49.Red-81-43-165.staticIP.rima-tde.net) |
| 11:43.05 | *** join/#asterisk micols (~t@rlogin.dk) |
| 11:56.26 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:04.24 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 12:04.53 | carrar | Y*A*W*N* |
| 12:14.39 | dax_roc | Anyone connected TVSe to asterisk before ? |
| 12:18.12 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:19.27 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:28.16 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:38.51 | v0lZy | humh |
| 12:38.57 | v0lZy | mirela666: any idea why i get this |
| 12:39.00 | v0lZy | -- Launching ConfBridge(1234,PageBridge,PagedUser,PageMenu) on Local/AutoAnswer44@AppAnnounce-8c36;1 |
| 12:39.01 | v0lZy | <PROTECTED> |
| 12:39.07 | v0lZy | the phone answers and immidiately drops |
| 12:39.55 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:40.46 | v0lZy | and the line it mentions |
| 12:40.48 | v0lZy | is just |
| 12:40.57 | v0lZy | same => n,Dial(SIP/44,,) |
| 12:41.40 | v0lZy | hm |
| 12:41.41 | *** join/#asterisk Smirker (~x@124-148-45-133.dyn.iinet.net.au) |
| 12:41.44 | v0lZy | now it works ok |
| 12:41.45 | v0lZy | intermittently apparently |
| 12:42.15 | v0lZy | sometimes works |
| 12:42.17 | v0lZy | sometimes doesnt |
| 12:42.19 | v0lZy | whats the catch? |
| 12:42.35 | mirela666 | v0lZy,what was the problem? |
| 12:44.46 | mirela666 | v0lZy, put 'core set verbose 3' and see or pase what was the ERROR or WARNNING message |
| 12:44.51 | v0lZy | cant exactly explain |
| 12:44.57 | v0lZy | it works halftime (the conference bridge thing) |
| 12:45.07 | v0lZy | im lookint at core set verbose 9 |
| 12:45.11 | mirela666 | :-/ |
| 12:45.33 | mirela666 | kk |
| 12:45.57 | mirela666 | was there any error before ==Spawn extension |
| 12:46.58 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 12:47.41 | v0lZy | -- Called SIP/44 |
| 12:47.42 | v0lZy | <PROTECTED> |
| 12:47.44 | v0lZy | <PROTECTED> |
| 12:47.45 | v0lZy | <PROTECTED> |
| 12:47.47 | v0lZy | <PROTECTED> |
| 12:47.48 | v0lZy | <PROTECTED> |
| 12:47.56 | jeffspeff | v0lZy, pastebinj |
| 12:48.01 | jeffspeff | *pastebin |
| 12:48.14 | v0lZy | http://bpaste.net/show/6G9DxpyCwZIiOxDzqifC/ |
| 12:48.17 | mirela666 | #pastebin |
| 12:48.32 | bulkorok | ~pastebin |
| 12:48.32 | infobot | A "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:48.34 | *** join/#asterisk chris_n (~Chris@nc-63-162-197-228.sta.embarqhsd.net) |
| 12:49.22 | jeffspeff | v0lZy, what are you trying to do? make a conference or page phones? |
| 12:49.41 | v0lZy | page phones with Confbridge |
| 12:49.48 | v0lZy | (I dont have app_page.so) |
| 12:49.55 | v0lZy | im paging one phone |
| 12:49.57 | v0lZy | and it workd |
| 12:50.02 | v0lZy | and for some unknown reason |
| 12:50.05 | v0lZy | its not working anymore |
| 12:50.12 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:50.32 | jeffspeff | let me guess, the phone just rings and doesn't auto answer? |
| 12:51.25 | jeffspeff | v0lZy, the very very easy way to do this is to just inall app_page |
| 12:51.30 | jeffspeff | *install |
| 12:51.39 | jeffspeff | my fingers aren't working well this morning |
| 12:51.57 | mirela666 | ;-) |
| 12:52.20 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:53.17 | jeffspeff | v0lZy, look in your /usr/lib/asterisk/modules/ and see if app_page.so is there |
| 12:53.26 | jeffspeff | if it is just set it load in asterisk |
| 12:53.47 | jeffspeff | you can also manually load it without interrupting the running service/process of asterisk |
| 12:54.34 | kaldemar | jeffspeff: it is not. he should compile DAHDI and the module to have it. |
| 12:54.46 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 12:54.51 | jeffspeff | kaldemar, true |
| 12:55.24 | v0lZy | jeffspeff: its not there thats why im using confbridge. |
| 12:55.35 | jeffspeff | v0lZy, why can't you install it? |
| 12:56.33 | v0lZy | cant find it |
| 12:56.36 | v0lZy | not compiled |
| 12:56.47 | v0lZy | never mind that though, i had this working exactly like i wanted it to |
| 12:56.50 | v0lZy | except |
| 12:56.54 | v0lZy | for some reason... it just started crashing |
| 12:57.08 | jeffspeff | Does anybody know why * would keep sip channels open? I have 2 active calls, and yet a sip show channels shows 72 channels... most of them have a last message of Rx: BYE but they don't go away |
| 12:57.31 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 12:58.03 | *** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:8c4c:ca87:2fc8:2a46) |
| 12:58.03 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 12:59.58 | *** join/#asterisk sideffec1 (sideffect@gateway/shell/bshellz.net/x-kclqypcdegmywqar) |
| 13:00.27 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:00.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:01.26 | v0lZy | this is extraordinarilly weird |
| 13:01.45 | v0lZy | i dont see typos |
| 13:01.55 | v0lZy | i dont remember doing anything special that would fuck it up this bad |
| 13:02.05 | *** part/#asterisk mirela666 (~mirko@212.200.146.253) |
| 13:02.46 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-rogyfkgyollpduat) |
| 13:02.46 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:04.52 | *** join/#asterisk mirela666 (~mirko@212.200.146.253) |
| 13:08.07 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 13:08.55 | *** join/#asterisk bchia (~Adium@nat/digium/x-kfdljzgvugfmahiv) |
| 13:10.29 | *** join/#asterisk nunne (~nunne@c-b9f4e355.021-109-73746f46.cust.bredbandsbolaget.se) |
| 13:13.41 | v0lZy | ah |
| 13:13.43 | v0lZy | channels arent dropped |
| 13:14.13 | v0lZy | wahts the command to kill chans from the cli again? |
| 13:14.26 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:14.50 | bulkorok | hangup [tab] channel |
| 13:15.04 | bulkorok | hangup request SIP/... |
| 13:15.13 | jeffspeff | Does anybody know why * would keep sip channels open? I have 2 active calls, and yet a sip show channels shows 72 channels... most of them have a last message of Rx: BYE but they don't go away |
| 13:16.41 | v0lZy | SHINOBI*CLI> hangup request SIP/44-0000000d |
| 13:16.43 | v0lZy | No such command 'hangup request SIP/44-0000000d' |
| 13:17.52 | jeffspeff | v0lZy, channel request hangup [TAB] |
| 13:21.13 | v0lZy | thanks |
| 13:21.18 | v0lZy | hm |
| 13:21.21 | v0lZy | still no idea whats going on |
| 13:21.25 | v0lZy | i have a feeling the cli is kind of lying |
| 13:21.26 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-urixuacqbfrebhjr) |
| 13:21.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:23.28 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-zwppsnybgelglyeg) |
| 13:26.16 | Katty | zombies |
| 13:26.51 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:27.16 | jeffspeff | Katty, are you saying my open channels are zombies? |
| 13:28.15 | Katty | YES |
| 13:28.20 | anonymouz666 | jeffspeff: there are some kind of SIP messages that opens channels also, but if this keeps increasing, you are in trouble. |
| 13:28.21 | Katty | remember, double tap! |
| 13:28.51 | dax_roc | like the pros |
| 13:29.04 | jeffspeff | anonymouz666, i don't think BYE messages should stay open like that |
| 13:29.32 | kaldemar | jeffspeff: are you sure they are the same dialogs that are hanging in there? |
| 13:29.42 | schmidts | jeffspeff what version do you have? |
| 13:30.00 | v0lZy | i think i somehow kill confbridge but i have no idea how |
| 13:30.06 | dax_roc | is there an option for legacy sip connections (protocol) ? |
| 13:30.14 | jeffspeff | schmidts, 10.3 |
| 13:30.28 | *** join/#asterisk serafie (~erin@nat/digium/x-khedwppuntejofcx) |
| 13:30.39 | anonymouz666 | jeffspeff: the RX Bye message should be destroyed within a few seconds after you hangup |
| 13:31.02 | dax_roc | Or does 2.0 negociate / backwards compatable ? |
| 13:31.35 | Katty | glomps serafie |
| 13:31.41 | schmidts | jeffspeff i have noticed the same problems but i have 10.7 |
| 13:31.46 | jeffspeff | kaldemar, my traffic isn't that high |
| 13:32.20 | jeffspeff | i have 2 active calls right now at this moment and at this moment i have 93 active sip dialogs |
| 13:32.43 | jeffspeff | some are the normal registers, acks, etc. but the vast majority are bye's |
| 13:33.06 | anonymouz666 | schmidts: are you under heavy load and sharing the device state? |
| 13:33.27 | v0lZy | apparently i dont know how to use ConfBridge or something with more than one parameter |
| 13:33.36 | anonymouz666 | v0lZy: time to learn then |
| 13:34.18 | jeffspeff | so, what would cause the channels to not be destroyed after the bye message? |
| 13:35.12 | anonymouz666 | jeffspeff: deadlocks, bad usage of scheduler, etc. |
| 13:35.27 | jeffspeff | here's what i'm looking at |
| 13:35.28 | jeffspeff | http://pastebin.com/MeFvva1R |
| 13:35.40 | v0lZy | well to me it looks fine |
| 13:35.48 | jeffspeff | what is scheduler? |
| 13:36.20 | schmidts | anonymouz6 what do you mean with "heavy load" and i dont use hints on this machine |
| 13:36.27 | v0lZy | im following this |
| 13:36.58 | anonymouz666 | schmidts: heavy load is a lot of current calls, i don't mean hints. I mean distributed device state. |
| 13:37.04 | v0lZy | andthis is what i have. http://bpaste.net/show/2gl1Bx9igLNrktV8eALa/ |
| 13:37.09 | schmidts | anonymouz666 36560 calls processed |
| 13:37.10 | schmidts | System uptime: 2 days, 7 hours, 3 minutes, 36 seconds |
| 13:37.10 | schmidts | ;) |
| 13:37.16 | v0lZy | uh |
| 13:37.17 | v0lZy | w8 |
| 13:37.19 | v0lZy | thats not right |
| 13:37.20 | anonymouz666 | not much then. |
| 13:37.23 | v0lZy | thats not what mcedit is showing |
| 13:38.01 | v0lZy | this is the correct version. http://bpaste.net/show/u8sqTBMt2Qn74VasFNbq/ |
| 13:38.15 | anonymouz666 | schmidts: I asked that because I have seen sip stucks channels due a lot of contention in device state cache, but I am under heavy load and using res_corosync to distribute the cluster device state. |
| 13:38.19 | jeffspeff | schmidts, in 1 week 22 hrs, i've processed 3578 calls |
| 13:38.36 | schmidts | typically this server has around 100 concurrent calls up to 150 in busy hours |
| 13:38.43 | jeffspeff | nice |
| 13:39.30 | anonymouz666 | 547 active channels |
| 13:39.31 | anonymouz666 | 251 active calls |
| 13:39.33 | anonymouz666 | just one machine |
| 13:39.45 | jeffspeff | holy crap! |
| 13:40.11 | schmidts | anonymouz666 this is one of my gateway servers which runs on 10 but i have three hosted pbx servers with around 50 concurrent calls but 2200 sip subscriptsions runing on 1.8 without any problems |
| 13:40.25 | anonymouz666 | jeffspeff: I have seen issues in this machine that few people in the world saw |
| 13:40.39 | jeffspeff | i believe it |
| 13:40.41 | anonymouz666 | 1.8 is great |
| 13:41.10 | schmidts | i also have another machine which does just the same gatewaying with a little bit more traffic an no stuck channels, setup and dialplan is identic except asterisk verion |
| 13:41.12 | schmidts | 683512 calls processed |
| 13:41.12 | schmidts | System uptime: 3 weeks, 6 days, 7 hours, 29 minutes, 41 seconds |
| 13:42.22 | schmidts | jeffspeff my record while doing tests was 13500 concurrent calls :D |
| 13:42.41 | jeffspeff | how do you test that? |
| 13:42.54 | schmidts | sipp |
| 13:43.07 | jeffspeff | i've had nothing but problems trying to use that in the past |
| 13:43.58 | schmidts | anonymouz666 my problem is that when i try to enable thread debug and dont optimize that this machine will not have enough power left to work productive |
| 13:43.59 | _Corey_ | schmidts: How do you verify there are no audio bridging issues in your benchmarkinng? |
| 13:44.23 | schmidts | _Corey_ this testing was about sip signaling not audio bridging |
| 13:44.41 | *** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 13:44.42 | _Corey_ | schmidts: OK, so you were reinviting the media or whatever? |
| 13:44.56 | anonymouz666 | schmidts: I also make this mistake. My box does not run for 5 minutos in debug_threads mode. |
| 13:45.03 | anonymouz666 | *did |
| 13:45.04 | schmidts | _Corey_ no media at all ;) |
| 13:45.12 | _Corey_ | gotcha |
| 13:45.24 | anonymouz666 | _Corey_: I test with media |
| 13:45.26 | anonymouz666 | pcap play |
| 13:45.30 | _Corey_ | schmidts: yeah, I've heard of 50k+ numbers on that |
| 13:45.41 | anonymouz666 | only signalling for me does not count at all |
| 13:46.24 | mjordan | jeffspeff: we fixed a number of 'stuck' channel problems in recent versions. You may want to take at the version information for 10.6.0 (ASTERISK-19455) to see if it looks similar to the problem you're having |
| 13:46.39 | *** part/#asterisk mirela666 (~mirko@212.200.146.253) |
| 13:46.58 | jeffspeff | mjordan, thanks. i'll look at that now |
| 13:47.17 | anonymouz666 | I am running 1.8.16.0-rc1 in production :P |
| 13:47.42 | mjordan | jeffspeff: technically you'd want to run 10.6.1, but since that was a minor point release it won't have as much info :-) |
| 13:47.54 | anonymouz666 | there is an important fix regarding chan_spy and local channels |
| 13:51.13 | jeffspeff | chan_spy is my friend. :) |
| 13:51.22 | *** join/#asterisk Johnny- (~John@23-24-48-118-static.hfc.comcastbusiness.net) |
| 13:51.59 | anonymouz666 | jeffspeff: I can't say that. I was sleeping with my enemy (chan_spy) |
| 13:52.08 | jeffspeff | lol |
| 13:52.27 | jeffspeff | mjordan, why did you recommend the 10.6 release instead of 10.7 ? |
| 13:54.13 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:54.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:55.37 | v0lZy | sound_place_into_conference=none |
| 13:55.40 | v0lZy | thats apparently fucking up |
| 13:56.21 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 13:56.37 | v0lZy | well, gotta go. |
| 13:56.45 | v0lZy | will deal with this tomorrow |
| 13:56.47 | v0lZy | thanks for everything guys |
| 13:56.49 | v0lZy | bye |
| 13:58.57 | mjordan | jeffspeff: I didn't really recommend - that's just where that issue was fixed. 10.7 has that fix as well :-) |
| 13:59.21 | mjordan | jeffspeff: if that issue looks like it is related to the problems you're having, then upgrading to any release 10.6+ would have that fix |
| 13:59.52 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:00.30 | jeffspeff | mjordan, ok, |
| 14:00.49 | *** join/#asterisk vizzy (~volker@p54B1B709.dip.t-dialin.net) |
| 14:01.30 | vizzy | hello :) i got a small problem using gotoif in asterisk 1.2, i have the following line |
| 14:01.35 | vizzy | exten => 987,n,GotoIf($ [ "${CALLERID(num)}" = "413" ]?anwahl:affen) |
| 14:01.47 | vizzy | 981,1 is just a noop :) |
| 14:02.33 | vizzy | when i call from 413 it should dial sip/test, and in all other cases just play the monkeys |
| 14:03.14 | kaldemar | $ [ "${CALLERID(num)}" = "413" ] --> $["${CALLERID(num)}" = "413"] |
| 14:03.25 | vizzy | uhm |
| 14:03.36 | kaldemar | space your spacebar with those. |
| 14:03.51 | [TK]D-Fender | GotoIf($ [ " <- Stop adding extra spaces |
| 14:03.57 | vizzy | i got that example from a 'pro'.... |
| 14:04.08 | [TK]D-Fender | $[ != $ [ |
| 14:04.15 | vizzy | oh |
| 14:04.28 | vizzy | ill try that out immediately |
| 14:05.24 | vizzy | wooot |
| 14:05.39 | vizzy | thx folks, it works now, phew, took me 2h now |
| 14:06.25 | *** join/#asterisk mirela666 (~mirko@212.200.146.253) |
| 14:08.55 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:14.21 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:14.46 | *** part/#asterisk mirela666 (~mirko@212.200.146.253) |
| 14:15.11 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 14:21.06 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:22.07 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 14:24.17 | *** join/#asterisk Sophira (~sophie@neo.theblob.org) |
| 14:25.42 | *** join/#asterisk amessina (~amessina@h-66-166-108-205.chcgilgm.static.covad.net) |
| 14:29.18 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:30.56 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 14:36.54 | *** part/#asterisk StaRetji (~LittleAll@178.79.11.166) |
| 14:39.48 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-120-227.socal.res.rr.com) |
| 14:40.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:40.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:48.39 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 14:50.16 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 14:50.54 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 14:55.16 | *** join/#asterisk localhost (~IceChat9@cpe-76-188-161-222.neo.res.rr.com) |
| 14:55.38 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 14:57.06 | *** join/#asterisk esaym153 (~esaym153@cpe-70-123-194-182.satx.res.rr.com) |
| 14:57.22 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 14:57.34 | esaym153 | is this a good repo: http://packages.asterisk.org/deb/ ? Seems like everything on there is several releases behind... |
| 15:07.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:08.14 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 15:10.31 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 15:11.05 | nny | what's the best application for checking for the existence of a file and returning a variable that says it exists or not |
| 15:11.08 | nny | ? |
| 15:14.12 | leifmadsen | nny: STAT() |
| 15:14.32 | leifmadsen | esaym153: mjordan was working on getting them updated, but for now, they are indeed behind. |
| 15:23.21 | pabelanger | leifmadsen: I think I am just going to move them to launchpad or something |
| 15:23.25 | pabelanger | sound be easy to doo |
| 15:23.32 | pabelanger | and automate it |
| 15:24.41 | leifmadsen | pabelanger: awesome |
| 15:24.56 | leifmadsen | is tempted to build a mock server to build the RPMs |
| 15:25.10 | pabelanger | +1 |
| 15:25.13 | leifmadsen | since I'm bothered by the fact there are no RHEL/CentOS 6 RPMs |
| 15:25.37 | pabelanger | Is there a public RPM build server you can use, like launchpad? |
| 15:25.44 | leifmadsen | not that I'm aware of... |
| 15:25.51 | pabelanger | boo |
| 15:25.51 | leifmadsen | but that doesn't mean it doesn't exist :D |
| 15:25.57 | pabelanger | nods |
| 15:26.00 | leifmadsen | doesn't want to build the RPM version of launchpad :D |
| 15:31.24 | chuckf | leifmadsen: I'm not a developer, but perhaps https://build.opensuse.org/ can help you out? |
| 15:32.39 | leifmadsen | chuckf: thanks will check it out -- heh, I noticed a typo under Sponsors. Platin is probably wrong :) |
| 15:33.40 | leifmadsen | hmmm, open build service |
| 15:36.37 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:37.00 | leifmadsen | chuckf: omg this is great :) |
| 15:37.04 | leifmadsen | chuckf: thanks! |
| 15:37.18 | chuckf | yw, glad to help |
| 15:39.05 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 15:39.28 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 15:43.57 | Katty | grooves |
| 15:44.16 | Katty | dances with chuckf |
| 15:46.21 | *** join/#asterisk jkroon (~jkroon@dsl-244-38-32.telkomadsl.co.za) |
| 15:46.36 | chuckf | enjoys the dancing with Katty |
| 15:51.49 | Katty | chuckf: is your head still bicced these days |
| 15:52.42 | n3hxs | says HI Katty, while it is still morning... |
| 15:52.56 | chuckf | Katty: if I knew what that meant, I'd give you an answer |
| 15:53.06 | Katty | razored. |
| 15:53.09 | Katty | hugs n3hxs |
| 15:53.11 | chuckf | ah, yeah |
| 15:53.27 | Katty | mister clean mister clean...do do do |
| 15:53.40 | chuckf | Katty: its been that way for ~12 years |
| 15:53.47 | Katty | i approve. |
| 15:54.13 | chuckf | yay! |
| 15:54.26 | Katty | it's a shame you two can't join us for friday night shenanigans. |
| 15:54.28 | leifmadsen | LUNCH |
| 15:54.40 | Katty | oh hey, it's almost lunch time. |
| 15:54.56 | chuckf | yeah, I see you post them and want to be there but its a long drive home... |
| 15:55.04 | Katty | yesh, a bit. |
| 15:55.26 | Qwell | it's only like 6 hours |
| 15:55.30 | Qwell | or something. |
| 15:55.40 | Katty | qwell you can join us for friday night shenanigans too |
| 15:55.57 | Katty | i have an air mattress |
| 15:56.10 | nny | leifmadsen: sorry missed the notification + multitasking, thanks |
| 15:57.02 | chuckf | its like 15 hours for me |
| 15:57.10 | Qwell | chuckf: road trip |
| 15:57.14 | Katty | that's forever away |
| 15:57.18 | Katty | which coast are you on |
| 15:57.22 | Katty | east? |
| 15:57.37 | chuckf | east, norht east of DC |
| 15:57.45 | Katty | brr! |
| 15:57.57 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 15:58.10 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:00.04 | chuckf | yeah. Though my sis in law did move to nashville recently so we may make a trip out there. If so, we'll see if shenanigans can be arranged for |
| 16:02.28 | Katty | ya i'll be going to gatlinburg here soonish |
| 16:02.35 | Katty | for touristy things. |
| 16:04.05 | Qwell | There's stuff to do in Gatlinburg? |
| 16:04.18 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 16:04.48 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 16:04.53 | chuckf | I always think gun battles when I hear hte name Gatllinburg |
| 16:06.19 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:09.35 | Katty | Qwell: of course! |
| 16:09.45 | Katty | Qwell: the ripley's aquarium, shopping in pidgeon forge... |
| 16:09.53 | Katty | Qwell: the microbrewery ^______^ |
| 16:10.02 | Katty | RIBS at that other place next door... forget the name of the place |
| 16:10.38 | Qwell | Why is there an aquarium in the middle of TN? |
| 16:10.42 | Katty | hiking in the smokey mountains! |
| 16:10.54 | Katty | because gatlinburg/pidgeon forge is a tourist trap like Branson, MO |
| 16:11.04 | Qwell | People intentionally go there? |
| 16:11.09 | Katty | i do. |
| 16:11.15 | Qwell | I meant Branson :p |
| 16:11.18 | Katty | yes, i do |
| 16:11.23 | Qwell | Why? O.o |
| 16:11.29 | Katty | i'm going to silver dollar city for a little christmas holiday |
| 16:11.30 | Qwell | I suppose it is like...right there |
| 16:11.40 | Katty | because it's adorable |
| 16:11.55 | Katty | and i love walking around and looking at the little shops |
| 16:11.58 | Katty | and getting ice cream ^_^ |
| 16:20.42 | Qwell | Katty: Did you go to the JD Distillery when you were around that area? |
| 16:25.54 | *** join/#asterisk lkcl (~lkcl@host86-131-173-121.range86-131.btcentralplus.com) |
| 16:26.13 | lkcl | hey folks - any chance of some help fine-tuning app_confbridge? |
| 16:26.29 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 16:26.38 | Katty | Qwell: not usually, no...but it's not like i'd be against going |
| 16:26.41 | drmessano | Don't know unless you ask something specific |
| 16:27.01 | Katty | Qwell: i did enjoy the smokey mountain brewery when i went :> |
| 16:30.47 | *** join/#asterisk CrazyTux[m] (~Brandon@108-64-56-113.lightspeed.irvnca.sbcglobal.net) |
| 16:31.21 | lkcl | e.g. i've set up "allowtransfer=no" because i believe that stops the call from being transferred away from the bridge and onto 2 users |
| 16:31.40 | lkcl | but i am seeing 11 users in the conference (!) |
| 16:31.51 | lkcl | all of them are the same IP address. |
| 16:32.04 | lkcl | dedi28*CLI> confbridge list |
| 16:32.11 | lkcl | fancybridge 11 0 unlocked |
| 16:32.17 | lkcl | oops :) |
| 16:32.38 | lkcl | so the more i call in (and end the call), the larger that number becomes! |
| 16:33.31 | Qwell | lkcl: sip show channels |
| 16:34.22 | lkcl | ta Qwell - it actually shows 12 channels (!) |
| 16:34.32 | lkcl | 10.8.0.161 lkcl ceee8a6e-e9e5-e (g722|mpeg4) No Tx: INFO lkcl |
| 16:34.33 | lkcl | 10.8.0.161 lkcl ea781902-eae5-e (g722|mpeg4) No Tx: INFO lkcl |
| 16:34.33 | lkcl | 10.8.0.161 lkcl 9e7ffcb7-22e6-e (g722|mpeg4) No Rx: INFO lkcl |
| 16:34.33 | lkcl | 10.8.0.161 lkcl 32c99fc2-2de6-e (g722|mpeg4) No Tx: INFO lkcl |
| 16:34.35 | lkcl | 12 active SIP dialogs |
| 16:34.56 | lkcl | with no expiry on them, hmmm |
| 16:36.33 | lkcl | those connections have been around for several hours. |
| 16:36.59 | drmessano | What version of Asterisk are you running? |
| 16:37.09 | lkcl | it probably doesn't help that ekiga segfaults at the end of a call.... |
| 16:37.31 | lkcl | drhonk: 10. checked it out of svn oooo 2-3 days ago |
| 16:37.38 | drmessano | ok |
| 16:37.53 | lkcl | drhonk, drmessano: sorry, xchat auto-completed there to someone else's name! |
| 16:38.37 | Qwell | lkcl: sip.conf, rtptimeout |
| 16:38.56 | Qwell | Your phone isn't closing the call, so Asterisk will keep it up. |
| 16:39.54 | lkcl | Qwell: ack. taa. |
| 16:40.07 | lkcl | hmm... that's weird. ekiga opens _two_ channels. |
| 16:40.14 | lkcl | let's see if that helps.... |
| 16:40.57 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:41.04 | lkcl | hmmm, would you recommend enabling rtpkeepalive as well? |
| 16:41.25 | Qwell | no |
| 16:41.29 | lkcl | it's UDP, after all... some of these firewalls tend to .... ok. |
| 16:44.08 | lkcl | ok, this is after ekiga segfaulted (only one person in the call, remember!) |
| 16:44.10 | lkcl | dedi28*CLI> sip show channels |
| 16:44.11 | lkcl | Peer User/ANR Call ID Format Hold Last Message Expiry Peer |
| 16:44.11 | lkcl | 10.8.0.161 (None) 3673b025-2fe6-e (nothing) No Rx: REGISTER <guest> |
| 16:44.11 | lkcl | 10.8.0.161 lkcl 3c13c326-2fe6-e (nothing) No Rx: SUBSCRIBE <guest> |
| 16:44.12 | lkcl | 10.8.0.161 lkcl 74e0142a-2fe6-e (g722|mpeg4) No Tx: INFO lkcl |
| 16:44.17 | lkcl | 3 active SIP dialogs |
| 16:44.19 | lkcl | !! |
| 16:44.25 | Qwell | Those are dialogs, not calls. |
| 16:44.36 | Qwell | They will go away. |
| 16:45.18 | lkcl | ok. |
| 16:45.26 | lkcl | they're gone. yay! |
| 16:45.33 | lkcl | [Aug 16 15:32:01] NOTICE[24110]: chan_sip.c:26871 check_rtp_timeout: Disconnecting call 'SIP/lkcl-00000000' for lack of RTP activity in 31 seconds |
| 16:45.35 | lkcl | ah ha |
| 16:45.36 | lkcl | ! |
| 16:45.40 | lkcl | problem solved. |
| 16:46.48 | lkcl | thank you drmessano, Qwell |
| 16:47.44 | lkcl | ok - would anyone like to help try and test this conference setup? anyone with a SIP videophone with an MPEG (MP4V-ES) codec? |
| 16:48.15 | lkcl | i've only been able to test with 2 people so far, but never with someone who has a working webcam, working speaker and working mic :) |
| 16:48.43 | lkcl | i have a couple of friends who don't have a webcam, or can't get their mic working ha ha |
| 16:50.40 | esaym153 | leifmadsen: thanks for the info |
| 16:50.42 | lkcl | i also don't have two computers in the house with 2 webcams and 2 working mics (only one) and it's too much network traffic for my broadband line anyway! |
| 16:51.43 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 16:51.44 | lkcl | 40k/sec which only is realistically available for "high-throughput" like HTTP. it all gets a bit spongey for voip. whoops |
| 16:56.01 | carrar | lifts his shades and glances over at Katty |
| 16:56.02 | carrar | PPAN GANGNAM STYLE !!!!!!!!!!!!!!!!!!!!!! |
| 16:56.07 | carrar | OPPAN GANGNAM STYLE !!!!!!!!!!!!!!!!!!!!!! |
| 17:01.48 | Katty | sorry, what? i was busy reading the AMA from the curiosity engineers on reddit |
| 17:02.01 | Katty | and dispensing hugs. |
| 17:07.39 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-wvdlqpxnxddpxrus) |
| 17:07.39 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 17:11.10 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:11.23 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:19.05 | *** join/#asterisk jakent (~jakent@c-71-63-6-140.hsd1.va.comcast.net) |
| 17:21.03 | beek | waves to Katty |
| 17:22.52 | Katty | hi beek :> |
| 17:23.05 | beek | How goes it Katty? |
| 17:29.04 | *** join/#asterisk localhost (~IceChat9@cpe-76-188-161-222.neo.res.rr.com) |
| 17:30.47 | Katty | not too shabby, how about you beek |
| 17:31.40 | beek | Doing okay. Busy busy busy. |
| 17:38.09 | carrar | HOW BUSY ARE YOU |
| 17:40.43 | Katty | steals a lemon from [TK]D-Fender's life. |
| 17:41.09 | [TK]D-Fender | MYYYY PRRRREEEEEECCCCIIIOUUUSSS!!!!!!! |
| 17:41.21 | [TK]D-Fender | hisses and claws them back |
| 17:42.45 | Katty | :< |
| 17:42.48 | Katty | gives lemon back |
| 17:45.38 | beek | carrar: busy-cubed |
| 17:47.37 | carrar | thats BUSY |
| 17:47.39 | Katty | i hate busy cubed days. |
| 17:47.44 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:47.47 | *** part/#asterisk esaym153 (~esaym153@cpe-70-123-194-182.satx.res.rr.com) |
| 17:54.48 | beek | Already had two blame-storming sessions today... looking forward to another at 3:30. |
| 17:55.02 | Katty | i will call you in dead. |
| 17:55.05 | Katty | and we can go shopping. |
| 17:55.20 | beek | I'm already dead so they wouldn't believe you. |
| 17:55.31 | Katty | well i'll call you in mostly rotten. |
| 17:55.38 | beek | eeeewww |
| 17:55.43 | Katty | ^_^ |
| 17:56.58 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 17:59.44 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:00.03 | *** join/#asterisk lkcl (~lkcl@host86-131-179-7.range86-131.btcentralplus.com) |
| 18:03.06 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:07.09 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 18:09.14 | *** join/#asterisk jakent (~jakent@c-71-63-6-140.hsd1.va.comcast.net) |
| 18:12.02 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:14.17 | *** join/#asterisk navaismo (~navaismo@189.191.94.251) |
| 18:16.24 | *** join/#asterisk LemensTS (~Administr@h20.246.23.98.dynamic.ip.windstream.net) |
| 18:17.33 | LemensTS | im trying to get realtime mysql working with asterisk on centos. it seems it is not pulling my settings from res_mysql.conf http://pastebin.com/A1Egtn3x |
| 18:18.15 | *** join/#asterisk jakent_ (~jakent@109.sub-70-192-192.myvzw.com) |
| 18:18.31 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:20.55 | LemensTS | I can do 'realtime mysql status' at the cli, and this is what it gives me: http://pastebin.com/28V0pnFD |
| 18:27.12 | leifmadsen | LemensTS: did you module reload res_mysql.so ? |
| 18:27.18 | leifmadsen | err res_config_mysql.so |
| 18:28.58 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:29.54 | *** join/#asterisk jakent_ (~jakent@c-71-63-6-140.hsd1.va.comcast.net) |
| 18:32.36 | *** join/#asterisk Panic (~Panic@70-89-113-113.for.infostatic.com) |
| 18:43.34 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 18:47.44 | LemensTS | leifmadsen: yea i did that and restarted asterisk, and restarted mysqld |
| 18:47.56 | leifmadsen | shrugs |
| 18:48.18 | leifmadsen | I only use ODBC |
| 18:52.14 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:59.47 | *** join/#asterisk jakent (~jakent@c-71-63-6-140.hsd1.va.comcast.net) |
| 19:00.11 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:03.00 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 19:03.37 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:08.50 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:21.18 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:23.07 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-193.mobile.uci.edu) |
| 19:29.38 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:30.06 | *** part/#asterisk LemensTS (~Administr@h20.246.23.98.dynamic.ip.windstream.net) |
| 19:30.44 | *** join/#asterisk stupidnic (~foo@cpe-70-94-232-191.sw.res.rr.com) |
| 19:31.59 | stupidnic | I want to troubleshoot an AGI script that doesn't seem to run from inside Asterisk (I can run it from the command line as the asterisk user, so I know it is something specific to AGI). How can I troubleshoot it? I have looked at "agi exec" but I can't wrap my head around it. |
| 19:36.01 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:47.10 | *** join/#asterisk bitglue (~pfrost@207.8.235.242) |
| 19:48.23 | bitglue | Is there a way I can detect in the dialplan that a SIP handset has responded with a 302 "moved" response? I'm trying to allow users to enable forwarding on their handsets, without having problems where cell phone voicemails pick up queue calls. |
| 19:49.11 | bitglue | I'm aware i can configure the queue to require agents to confirm they are human with a DTMF response, which is good, except I don't want to require that except when the call has been forwarded to an external device. |
| 19:54.35 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 20:02.28 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 20:03.05 | navaismo | stupidnic, active agi debug from asterisk cli and see if show errors: agi set debug on |
| 20:03.24 | stupidnic | navaismo: Yeah I have that turned on but I am not seeing anything |
| 20:03.35 | stupidnic | Just the standard call and the tx variables |
| 20:04.10 | navaismo | then you can set some verbose from your script to send asterisk and see where the script stop |
| 20:04.11 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 20:04.15 | stupidnic | so I know it is getting called, but I can't replicate why it won't work from the AGI call. As I said it works when I su asterisk and call it direcly from the CLI |
| 20:04.53 | stupidnic | It's literally two lines, althogh one call is fetchmail, and I suspect that is what is messing with things |
| 20:14.43 | leifmadsen | try running it not from asterisk |
| 20:14.55 | leifmadsen | debug the script first then it should work fine from the AGI |
| 20:15.23 | leifmadsen | add some additional debugging for what is passed so you can check what you're expecting to be in the script to exist |
| 20:16.13 | stupidnic | leifmadsen: I have done that. It works perfectly as the user asterisk (I just su asterisk and then run the AGI and it works fine) |
| 20:16.42 | stupidnic | however for whatever reason when it is run via AGI it doesn't work |
| 20:17.02 | stupidnic | What is really weird is that it worked fine in 1.4 |
| 20:17.19 | stupidnic | but then the drive for the server died, and so I figured okay... time for an upgrade anyways |
| 20:17.35 | stupidnic | moved to Debian 6 and upgraded to 1.8 and it no longer works |
| 20:17.56 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 20:17.57 | stupidnic | the fetchmail script isn't doing anything more than flushing a mailbox |
| 20:18.10 | stupidnic | nothing fancy like festival etc |
| 20:21.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:23.50 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 20:24.47 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 20:38.10 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 20:42.05 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:44.54 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 20:45.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:46.02 | Hive | Hello, I'm trying to get my musicclass setting of queues.conf to work, however it is apparently overridden by Set(CHANNEL(musicclass)=<themusic>). Is there a way to set music on hold without setting CHANNEL(musicclass)? |
| 20:47.49 | [TK]D-Fender | Hive: show us your configs and the call |
| 20:53.46 | Hive | [TK]D-Fender: http://pastebin.com/ZuARE7SF |
| 20:54.14 | Hive | i want it to play musicclass 18, which is configured in queues.conf, but it plays default, which is from the set musicclass |
| 20:54.49 | Hive | i could change the musicclass before the caller is queued, then change it back after they leave the queue... but that seems like cheating |
| 21:01.06 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 21:01.10 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:10.37 | [TK]D-Fender | Show me that the music class was actually loaded |
| 21:20.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:27.08 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:31.14 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 21:35.35 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-48-76.bois.qwest.net) |
| 21:35.40 | *** join/#asterisk craig3smith (~craig3smi@ip67-90-212-69.z212-90-67.customer.algx.net) |
| 21:36.06 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-48-76.bois.qwest.net) |
| 21:37.21 | craig3smith | Can someone point me in a direction with running 11 beta? |
| 21:37.24 | craig3smith | asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_websocket_add_protocol |
| 21:37.35 | Qwell | file: PING |
| 21:37.42 | file | what |
| 21:37.45 | file | oh |
| 21:37.46 | Qwell | that ^ |
| 21:38.10 | file | craig3smith, is res_http_websocket.so loaded? if not load it first - I believe there is an issue open to make that optional |
| 21:38.36 | WIMPy | Optional sounds good. |
| 21:38.58 | craig3smith | Great thanks, where would have been a good place to look that up? |
| 21:39.35 | Qwell | ~asterisk issues |
| 21:39.41 | Qwell | infobot: stupid bot |
| 21:39.41 | infobot | Stupid human. |
| 21:39.49 | Qwell | craig3smith: https://issues.asterisk.org/jira/ |
| 21:39.52 | Qwell | infobot: I can kick you. |
| 21:39.53 | infobot | You can kick you.? |
| 21:39.57 | Qwell | crap. |
| 21:40.10 | WIMPy | LOL |
| 21:40.11 | file | issues are contained there, otherwise most individuals use auto loading of modules which loads all installed and 'stuff just works' |
| 21:40.12 | craig3smith | Thanks Qwell |
| 21:40.41 | craig3smith | autoload = yes |
| 21:40.43 | WIMPy | If you build all modules, that is. |
| 21:40.54 | craig3smith | ok gotcha |
| 21:40.56 | file | WIMPy, truth! |
| 21:41.15 | file | if there is no issue for that create one and I'll nudge the powers that be to fast track it to me |
| 21:41.23 | Qwell | file: That should be pretty easy to force |
| 21:41.26 | file | at no extra cost to you! |
| 21:41.35 | Qwell | <use type="module">res_http_websocket</use> |
| 21:41.37 | Qwell | just FYI! |
| 21:41.57 | Qwell | (or something) |
| 21:42.27 | file | Qwell, that doesn't make it optional within the code |
| 21:42.29 | WIMPy | await the questions about why Asterisk no longer has sip support. |
| 21:42.57 | craig3smith | not from me I got that one figured out |
| 21:43.43 | WIMPy | No, but from those who deselect websockets and then end up without sip. |
| 21:44.30 | craig3smith | <PROTECTED> |
| 21:45.01 | file | yeah that was discovered today, it shouldn't be - slipped by |
| 21:45.14 | craig3smith | I'm good thanks all! |
| 21:45.22 | file | peace! |
| 22:08.02 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 22:08.13 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 22:15.46 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 22:17.17 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-urixuacqbfrebhjr) |
| 22:20.02 | *** join/#asterisk bmg505 (~leon@196-209-171-201.dynamic.isadsl.co.za) |
| 22:22.21 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 22:36.11 | *** join/#asterisk jakent (~jakent@237.sub-70-192-195.myvzw.com) |
| 22:38.06 | *** join/#asterisk skor_ (~skor@unaffiliated/skor) |
| 22:56.47 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 23:00.35 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 23:06.38 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 23:07.48 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 23:09.47 | paulc | POP QUIZ! And sanity check... If I take 451470 calls a month, and each lasts 30 seconds, using g711.. around how much data transfer will get used, in gigs? I've got my number, I want to see if anyone else agrees/comes close.. |
| 23:12.31 | *** join/#asterisk wonderworld (~ww@dsdf-4db5e125.pool.mediaWays.net) |
| 23:18.40 | carrar | 0.148 GBytes |
| 23:19.10 | paulc | carrar: This is where the teacher says "now show me your workings.." - cos I had way higher |
| 23:19.56 | WIMPy | per direction. |
| 23:20.14 | paulc | or was that per call? |
| 23:20.46 | carrar | ((((451470*30)*88000)/1000000000000)/8 |
| 23:21.03 | carrar | 30 seconds |
| 23:21.17 | carrar | 88000 bps |
| 23:21.21 | carrar | for g711u |
| 23:21.32 | carrar | <PROTECTED> |
| 23:21.36 | carrar | <PROTECTED> |
| 23:21.39 | paulc | Yeah, I went with 80000 bps = 10000 bytes per second |
| 23:22.23 | carrar | err |
| 23:22.24 | paulc | so.. 451470*30*10000 then /1024 for KB, /1024 for Mb and /1024 for GB? |
| 23:22.24 | carrar | (((451470*30)*88000)/1000000000000)/8 |
| 23:22.34 | carrar | one too many left paren |
| 23:23.07 | carrar | bc |
| 23:23.09 | carrar | scale=4 |
| 23:23.13 | carrar | (((451470*30)*88000)/1000000000000)/8 |
| 23:23.19 | carrar | result == Gbytes |
| 23:23.23 | carrar | GBytes |
| 23:23.28 | WIMPy | GiB |
| 23:24.17 | carrar | You are missing one part in your MATH |
| 23:24.19 | carrar | and thats |
| 23:24.20 | carrar | OPPAN GANGNAM STYLE !!!!!!!!! |
| 23:24.33 | carrar | WOP WOP |
| 23:24.57 | paulc | Ok, I may regret asking this (he said with a grin), but what is "oppan gangnam style" (and will it make my head explode?!) |
| 23:25.12 | carrar | I will leave that to you to google |
| 23:25.36 | paulc | And... umm.. aren't there 3 zeros too many in there? |
| 23:25.53 | paulc | 148GB would be closer to what I had, per dierction |
| 23:26.15 | carrar | maybe |
| 23:26.16 | WIMPy | That's what I see. |
| 23:26.25 | carrar | Hey I didn't make math up |
| 23:26.31 | carrar | I just missuse it |
| 23:26.49 | paulc | I like that line :) |
| 23:27.08 | carrar | it's only 0 |
| 23:27.12 | carrar | just divide more |
| 23:27.14 | paulc | And yes - oppan gangnam style.. I was thinking it was some math geek joke thing like "reverse polish notation".. google's great, I'm all good |
| 23:27.27 | carrar | or multiple more |
| 23:27.32 | carrar | multiply |
| 23:27.39 | carrar | (((451470*30)*88000)/1000000000)/8 |
| 23:27.42 | carrar | try that on |
| 23:28.17 | carrar | (((451,470*30)*88,000)/1,000,000,000)/8 |
| 23:28.23 | carrar | bc really needs to accept commas |
| 23:28.27 | paulc | 149GB give or take.. multiple by 2 for both legs.. yeah.. with your 88kbps versus my 80kbps.. tallies, more or less.. |
| 23:28.43 | paulc | I had 252GB all in.. |
| 23:29.13 | carrar | who counts bothways? |
| 23:29.27 | carrar | in 95th % billing it's whatever is the greateast of in and out |
| 23:30.15 | paulc | Good point.. and I think the place my co-worker was looking at gives unlimited inbound bandwidth to the server, and just bill on data transfer outbound.. |
| 23:30.18 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 23:30.19 | carrar | and then cop off the top 5% |
| 23:30.22 | carrar | chop |
| 23:30.41 | carrar | ((((451470*30)*88000)/1000000000)/8)*2 |
| 23:30.41 | carrar | 297.9702 |
| 23:31.08 | paulc | I find that whole thing funny.. bandwidth versus data volume.. it should kinda be one or the other really, yet cable companies and telcos like to do both.. "Sure, you can pay more for higher throughput, but we'll still limit how much you can do of that" |
| 23:31.23 | carrar | You get billed for upload AND download added together? |
| 23:31.27 | paulc | likes carrar's bc command line skillz |
| 23:31.46 | carrar | Once I get the zeros right it works nice :) |
| 23:32.02 | paulc | some companies do.. others say "this is the limit, it applies to both directions, equally" (ie same up and down data transfer limits) |
| 23:32.31 | paulc | nice when the upstream is same as the downstream speed, not some bullshit crippled thing (*coughs* fiber to the home so why am I limited to 1Mbps up like I was on ADSL?!) |
| 23:32.56 | carrar | fire up RTG on your hom router and start doing your own accounting |
| 23:32.59 | carrar | home |
| 23:33.13 | carrar | it can do totals or 95th |
| 23:33.35 | carrar | http://rtg.sourceforge.net |
| 23:34.32 | paulc | Yeah, I remember doing some work with RTG years ago (like 12 years?) when I was back in the UK.. some stuff for an ISP.. |
| 23:35.00 | paulc | on my list of things to do is replace the telco supplied router with something a bit "better" and do some pretty graphing.. I just haven't gotten round to it yet.. |
| 23:35.22 | paulc | I had Tomato on my WRT54G at my old place, that did cool stats for usage.. |
| 23:35.24 | carrar | It's all about pretty graphs |
| 23:35.25 | *** join/#asterisk xpl (~xpl@204-232-210-62.static.cloud-ips.com) |
| 23:35.31 | paulc | alas.. ActionTec.. nuff asid. |
| 23:35.34 | paulc | or said, even |
| 23:35.38 | carrar | and espresso |
| 23:38.04 | carrar | ActionTec DSL routers? |
| 23:38.23 | carrar | modem/routers |
| 23:38.33 | paulc | Yeah.. fiber --> ONT --> ActionTec |
| 23:39.21 | carrar | get a cisco and VDSL wic |
| 23:40.54 | carrar | Cisco VDSL2 wic supports up to 200 mbps |
| 23:40.58 | paulc | I was thinking MikroTik RB750GL and some cleverness.. still up in the air whether Telus does the same VLAN stuff that Bell does for their Mediaroom IPTV deployment. Need to sniff some stuff |
| 23:41.22 | carrar | The 1-port VDSL2 HWIC is supported on the Cisco 1941, 2901, 2911, 2921, 2951, 3925, and 3945 ISR G2 starting with Cisco IOS Software 15.0.1M1. |
| 23:41.33 | carrar | expensive setup :) |
| 23:42.09 | carrar | ebay it's not too bad |
| 23:43.25 | carrar | $500 for the WIC, $855 for the router (2901) |
| 23:43.51 | *** join/#asterisk tamiel (~tamiel@c-69-181-254-90.hsd1.ca.comcast.net) |
| 23:47.36 | paulc | I'd like to learn some more Cisco stuff.. but I don't have $1400 burning a hole in my pocket right now ;) |
| 23:55.46 | carrar | can't you sell some of your children or something? |
| 23:55.49 | carrar | :) |
| 23:56.14 | WIMPy | You could even sell potential children :-) |
| 23:56.49 | carrar | sell futures |
| 23:57.00 | carrar | done all the time on the stock market! |
| 23:58.35 | paulc | The problem with child selling is that he's not actually my child.. |
| 23:58.52 | paulc | and on that note - it's time to do the commute! Thanks for the validation this afternoon :-) |