00:00.19 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-kxlccfmnppqxmozm) |
00:06.03 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
00:17.20 | *** join/#asterisk Dovid (~Dovid@250.sub-70-192-71.myvzw.com) |
00:18.03 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-cwiafoopyejbfuar) |
00:20.33 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
00:38.41 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-fmqumjemzupkmjzd) |
00:52.04 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
00:52.14 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
00:59.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
01:03.26 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-hmgrrzigibxfbnxq) |
01:20.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.242) |
01:22.56 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
01:24.08 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
01:33.25 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
01:36.20 | *** join/#asterisk jrose_atDigium (~jrose_atD@24.214.220.38) |
01:45.25 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
01:54.21 | *** join/#asterisk SGjunior (~sgjunior@out-pq-247.wireless.telus.com) |
02:07.19 | *** join/#asterisk aruntomar (~Thunderbi@49.248.154.66) |
02:15.57 | *** part/#asterisk mjordan (~mjordan@75.76.55.191) |
02:16.15 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
02:17.12 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
02:21.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.105) |
02:35.56 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
03:04.04 | *** join/#asterisk SGjunior (~sgjunior@out-pq-247.wireless.telus.com) |
03:22.00 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.105) |
03:29.37 | *** join/#asterisk dreaded (~dreaded@108.234.73.68) |
03:44.52 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
03:45.03 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
04:10.45 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
04:22.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.47) |
04:22.52 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-jyziwxssmkqowblo) |
04:24.08 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-zyfsvkyteqqbeohv) |
04:26.20 | *** join/#asterisk dreamfighter (~dreamfigh@d6-54.rb2.clm.centurytel.net) |
04:32.26 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
04:32.34 | *** join/#asterisk SGjunior (~sgjunior@out-pq-247.wireless.telus.com) |
04:51.12 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
04:51.16 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
05:02.16 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
05:14.09 | *** join/#asterisk chuckf (~chuckf@fedora/chuck) |
05:23.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.47) |
05:34.28 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
05:34.47 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
05:54.02 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-gthalshzydbjzrvo) |
05:59.25 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
06:04.02 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
06:08.13 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
06:13.39 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
06:13.52 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
06:14.20 | bulkorok | hi |
06:19.45 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-llxjlwkcfprdgvdy) |
06:20.02 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-hmpmagoqhyswepvf) |
06:24.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.245) |
06:38.00 | v0lZy | hi |
06:49.43 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-tkqkmvjovcjtcqwi) |
06:55.50 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
06:57.21 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
06:58.03 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.241) |
07:26.01 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
07:30.30 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
07:34.06 | *** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl) |
07:44.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3cdf:be21:8cfe:bc97) |
07:44.59 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
08:13.37 | *** join/#asterisk jrose_atDigium (~jrose_atD@24.214.220.38) |
08:15.04 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
08:27.51 | *** join/#asterisk saintcajetan (~sodalitum@cpe-72-183-246-29.elp.res.rr.com) |
08:27.54 | *** join/#asterisk afournier (~admin@mx1.wisp-e.com) |
08:30.20 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
08:51.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
08:53.29 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
09:06.47 | *** join/#asterisk jkroon (~jkroon@dsl-165-145-94-58.telkomadsl.co.za) |
09:10.36 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
09:15.12 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
09:19.24 | *** join/#asterisk CeBe (~CeBe@port-92-206-120-64.dynamic.qsc.de) |
09:25.19 | *** join/#asterisk afournier (~admin@mx1.wisp-e.com) |
09:41.46 | *** join/#asterisk gerhard7 (~gerhard7@77.172.47.159) |
09:44.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
10:02.02 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
10:09.18 | *** join/#asterisk davlefouAMD (~david@197.15.217.90) |
10:25.40 | *** join/#asterisk afournier (~admin@mx1.wisp-e.com) |
10:42.45 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
10:45.07 | nam3l3zz | hi all, have a following, could anyone comment on how bad is it ? http://pastebin.com/Vvzn8spP |
10:54.04 | jkroon | non-issue probably. |
11:16.13 | *** join/#asterisk CeBe (~CeBe@dhcp-215-92.vpn.tu-berlin.de) |
11:16.30 | *** join/#asterisk eduzimrs (~eduzimrs@mail.aytycrm.com.br) |
11:21.51 | nam3l3zz | jkroon: thnx |
11:22.53 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
11:26.19 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.183) |
11:37.08 | *** join/#asterisk andrewya_ (~andrewyag@syd02s26-fw01.thecore.net.au) |
11:37.09 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
11:41.57 | *** join/#asterisk CeBe (~CeBe@port-92-206-120-64.dynamic.qsc.de) |
11:49.19 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
11:49.21 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
11:51.10 | *** join/#asterisk tzafrir (~tzafrir@213-193-105-70.static.cablecom.ch) |
11:55.37 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
11:58.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.65) |
12:03.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
12:07.00 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
12:12.51 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
12:24.19 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
12:30.07 | *** join/#asterisk tzafrir (~tzafrir@213-193-105-70.static.cablecom.ch) |
12:46.02 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
12:47.26 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
12:55.06 | *** join/#asterisk mjordan (~mjordan@24.214.220.38) |
12:55.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
12:56.42 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
12:56.52 | *** join/#asterisk zigg (~matt@unaffiliated/zigg) |
13:00.04 | *** join/#asterisk tzafrir (~tzafrir@213-193-105-70.static.cablecom.ch) |
13:01.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
13:02.02 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
13:03.03 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
13:03.03 | *** mode/#asterisk [+o malcolmd] by ChanServ |
13:05.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
13:05.55 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
13:06.13 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
13:07.12 | *** join/#asterisk askin (~askin@pis-vi.tk) |
13:08.46 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
13:13.56 | *** join/#asterisk leedm777 (~leedm777@24.214.220.38) |
13:15.28 | *** join/#asterisk mjordan (~mjordan@24.214.220.38) |
13:15.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
13:17.03 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@186.7.193.173) |
13:17.33 | *** join/#asterisk miguel3239 (~miguel323@2001:470:1f06:12c4::2) |
13:18.07 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
13:18.07 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
13:18.19 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
13:20.41 | *** join/#asterisk infernixx (nix@unaffiliated/infernix) |
13:20.45 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
13:21.41 | *** join/#asterisk serafie (~erin@24.214.220.38) |
13:23.57 | *** join/#asterisk serafie1 (~erin@24.214.220.38) |
13:23.57 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
13:24.25 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
13:25.01 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3cdf:be21:8cfe:bc97) |
13:25.58 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
13:26.21 | askin | hi, i am using queue for a project, i have to dial some numbers after answer(to skip ivr menu). How can i do? |
13:27.02 | jmetro | core show application dial |
13:27.11 | leifmadsen | that ^^ |
13:27.18 | leifmadsen | there are flags for sending DTMF |
13:27.47 | askin | i use elastix, i didnt wrote any code. |
13:28.08 | leifmadsen | sorry to hear that |
13:28.21 | leifmadsen | this channel can't help you with GUI based projects |
13:28.29 | leifmadsen | you'll need to ask the #elastix people |
13:29.01 | *** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com) |
13:30.16 | jmetro | thats a shame, it would have been easy |
13:31.17 | askin | ok, i can write dial plan :) but i dont know how to start? is there any example? |
13:31.42 | mjordan | ~thebook |
13:31.42 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
13:31.52 | mjordan | askin: ^^^ |
13:32.01 | leifmadsen | get it while you can, because there won't be a 5th edition :) |
13:32.03 | jmetro | ^ Thats a good resource, but if you are using a GUI, they most likely mangle the dialplan in such a manner that you will be unable to modify it yourself. |
13:32.08 | mjordan | leifmadsen: ??? |
13:32.16 | leifmadsen | mjordan: too much work :( |
13:32.43 | mjordan | interesante |
13:32.43 | leifmadsen | plus, everyone is one 1.4 and 1.8 still... so that book will be valid for 5-6 years :) |
13:32.44 | askin | ok, thanks |
13:33.00 | mjordan | we should talk about that at some point :-D |
13:33.39 | leifmadsen | mjordan: talk about which point? :) |
13:33.43 | leifmadsen | mjordan: I'll be at AstriCon!@ |
13:34.16 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
13:40.35 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
13:42.02 | *** join/#asterisk Draecos (~Draecos@124-169-108-12.dyn.iinet.net.au) |
13:47.03 | *** join/#asterisk taylorbyte2013 (~cyberninj@139.218.237.26) |
13:48.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
13:48.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
13:48.55 | *** join/#asterisk dms (~dms@50-194-222-116-static.hfc.comcastbusiness.net) |
13:49.50 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
13:50.53 | jmetro | astricon seems fun |
13:51.02 | Qwell | jmetro: It is. You should go. |
13:51.08 | leifmadsen | it's super fun |
13:51.14 | leifmadsen | I'm excited. |
13:51.24 | leifmadsen | Qwell: you going this year? |
13:51.30 | Qwell | leifmadsen: heck yeah I am |
13:51.31 | leifmadsen | Qwell: you can be a booth babe! |
13:51.36 | leifmadsen | :D |
13:51.37 | jmetro | The engineers here are a cloistered group, i'd never be brought with |
13:51.37 | Qwell | OH SNAP |
13:51.40 | leifmadsen | perfect timing |
13:52.10 | *** join/#asterisk yano (yano@freenode/staff/yano) |
13:52.40 | Qwell | jmetro: Make them take you. Get some dirt on them. Or their wives. Or girlfriends. Or both! |
13:53.16 | jmetro | Could work.. |
13:53.22 | Qwell | oh, it works |
13:55.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
13:56.06 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
13:56.07 | raidghost | When asterisk just start stop doing its job, and not much info is to find in logs. It sounds like crappy hardware takingovah |
13:57.17 | raidghost | I reboot the computer. Asterisk works for some hours, and then suddenly without a notice it stop working propperly. I gonna try to set asterisk to debug mode to try figure out whats going on. Annoying when things happends i dont know why. |
14:00.45 | Qwell | "it stops working" doesn't really say anythign |
14:01.19 | jmetro | set your debug and verbose up and i bet youl lfind something |
14:01.22 | raidghost | Well. i dont get tone when i pick up the phone, And inbound and outbound calls does not give other than a "busy" signal |
14:01.38 | Qwell | What version of Asterisk? |
14:01.43 | raidghost | Latest one. |
14:01.45 | raidghost | 11.5.0 |
14:01.51 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
14:04.35 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
14:07.09 | Chainsaw | raidghost: And it's not just the bad bad internet routing calls to Palestine? |
14:07.28 | raidghost | Chainsaw: I dont do calls outside norway. So yes. |
14:11.02 | raidghost | But, i will reinstall asterisk on a fresh install of debian. And install the version of asterisk that debian have in apt. |
14:12.41 | jmetro | just get the new version, dont use the apt-get install |
14:13.10 | raidghost | jmetro: everything did work perfect out of the box. Until i upgraded from 11.3.something to 11.5.0 |
14:13.21 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:8def:12f0:8906:9253) |
14:13.24 | raidghost | after that every little issue did show up over long time |
14:14.28 | [TK]D-Fender | raidghost: What are youcalling over? |
14:14.33 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
14:15.20 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
14:15.23 | raidghost | try to explain your question and i might understand better. |
14:16.47 | raidghost | IAX2 if that was what you wanted of info. Im standing here like a ? trying to understand what your looking for of info from me. |
14:17.20 | [TK]D-Fender | raidghost: You should be showing us debug for the calls then |
14:22.23 | raidghost | i thought sip set debug on should ben screaming out info |
14:26.05 | *** join/#asterisk newtonr (~newtonr@24.214.220.38) |
14:26.05 | *** mode/#asterisk [+o newtonr] by ChanServ |
14:30.45 | *** join/#asterisk saxa (~saxa@201-50-103-98.user.veloxzone.com.br) |
14:31.02 | saxa | hi, can somebody help me understand why I get this |
14:31.04 | saxa | http://pastebin.com/DYLrMgca |
14:31.40 | saxa | it seems a nat problem , but it was working. Afaik i have never changed anything |
14:35.24 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
14:36.01 | igcewieling | Yes, it is a problem. Retransmitting #1 (NAT) to 187.115.169.227:5060: |
14:36.10 | igcewieling | that means you have a networking or NAT issue. |
14:37.02 | saxa | ok |
14:37.34 | saxa | thats why it gets disconnected probably, but the thing is how to understand where the issue is ? |
14:38.01 | *** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com) |
14:38.05 | saxa | asterisk is behind a NAT and the casasip phone behind another nat at my home |
14:43.44 | igcewieling | set nat=yes externip= localnet= |
14:43.51 | igcewieling | set directmedia=off |
14:44.12 | igcewieling | make sure the localnet contains a network, not an ip |
14:44.21 | igcewieling | you know, all the standard NAT stuff. |
14:45.02 | igcewieling | also turn off SIP ALG and/or SIP SPI on your home router. |
14:45.04 | saxa | igcewieling: yep, thats all set up |
14:45.28 | saxa | i have to check my home router |
14:45.31 | igcewieling | saxa: put the [general] section of sip.conf on a pasebin |
14:46.13 | [TK]D-Fender | [10:22]raidghosti thought sip set debug on should ben screaming out info <- .... you just said your were using IAX2. What good do you think ***SIP DEBUG*** is going to do about that? |
14:47.51 | [TK]D-Fender | saxa: it is not setup right |
14:48.05 | [TK]D-Fender | saxa: * is giving your phone a LOCAL IP to connect to. |
14:48.36 | [TK]D-Fender | saxa: c=IN IP4 192.168.0.1 |
14:48.39 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
14:50.28 | saxa | http://pastebin.com/8aCP25q5 |
14:52.11 | saxa | [TK]D-Fender: but 192.168.0.1 is the * internal ip |
14:52.19 | [TK]D-Fender | saxa: braserv.chickenkiller.com <- ping it from CLI |
14:52.57 | Katty | good morning |
14:53.03 | *** join/#asterisk feeshon (~feeshon@ool-4a5a8ab9.dyn.optonline.net) |
14:53.09 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
14:53.19 | jmetro | braserv.chickenkiller.com doesnt seem to work. |
14:53.24 | jmetro | elo kati |
14:53.36 | wasanzy | after installing asterisk, where can I copy the sample configs like sip.conf etc from? |
14:54.06 | Katty | waves to jmetro |
14:54.36 | feeshon | Can anyone suggest a free VOIP call recording software? |
14:54.37 | [TK]D-Fender | wasanzy: "make sample" |
14:54.39 | newtonr | wasanzy, you can run "make samples" which will install them in /etc/asterisk |
14:54.54 | [TK]D-Fender | +s |
14:55.30 | newtonr | wasanzy, or just copy the ones you want from asterisksource/configs/ |
14:55.40 | wasanzy | oh ok thank you |
14:56.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
14:57.04 | saxa | jmetro: how you mean that ? |
14:57.23 | feeshon | The phone system isn't running asterisk so I would need something else..I know it isn't the prefect place to ask but any help suggestions would be great! |
14:58.16 | Katty | gives feeshon weird looks |
14:58.35 | Katty | SO. |
14:58.42 | Katty | i finally got around to making borscht |
14:58.49 | Katty | not too shabby. |
14:59.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
14:59.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
15:01.33 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
15:01.36 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
15:01.53 | pabelanger | Is the minregexpire and maxregexpire settings for chan_iax2 for the farside? Eg: control incoming registrations? |
15:03.07 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
15:06.10 | *** join/#asterisk skirge (~skirge@196-210-220-9.dynamic.isadsl.co.za) |
15:06.22 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
15:10.07 | *** join/#asterisk eirirs_ (~test0r@193.157.115.211) |
15:13.02 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
15:16.01 | saxa | [TK]D-Fender: http://pastebin.com/hhzMDn7q |
15:16.25 | [TK]D-Fender | saxa: .... |
15:16.34 | [TK]D-Fender | saxa: it resoles... AS YOUR LOCAL IP |
15:16.38 | [TK]D-Fender | resolves* |
15:16.43 | saxa | yes |
15:16.45 | saxa | oh ok |
15:16.54 | saxa | let me check my hosts file |
15:17.23 | saxa | ok i had an entry in there |
15:17.32 | saxa | that was probably the issue, let me check |
15:18.00 | [TK]D-Fender | s/probably/definitely |
15:18.54 | saxa | root@braserv:~# ping braserv.chickenkiller.com |
15:18.54 | saxa | PING braserv.chickenkiller.com (201.50.103.98) 56(84) bytes of data. |
15:18.54 | saxa | 64 bytes from 201-50-103-98.user.veloxzone.com.br (201.50.103.98): icmp_req=1 ttl=64 time=1.06 ms |
15:19.36 | saxa | great |
15:19.48 | *** join/#asterisk navaismo (~navaismo@189.241.68.101) |
15:19.49 | saxa | it works now |
15:19.59 | saxa | thanks [TK]D-Fender |
15:20.32 | saxa | i have the entry in my hosts file because on that same machine i have also a web server |
15:21.01 | saxa | and on my local net when i write braserv.chickenkiller.com it simply refuses to work if i do not have this |
15:21.18 | saxa | anyway it seems i forgot to comment it out once |
15:21.30 | saxa | sorry for bothering , but thanks to all |
15:21.56 | dpeloquin | love the domain |
15:26.16 | saxa | thats a free one on freedns.org iirc |
15:30.34 | wasanzy | anyone has idea about PIAF2? |
15:33.54 | igcewieling | wasanzy: Yes. |
15:36.46 | wasanzy | am trying to implement google voice which Wolfram provides, the question is do I need PIAF2 before that can be done? or I could go ahead and use my normal asterisk I installed from source? |
15:37.31 | igcewieling | That wasn't my idea. My idea is that this is not the right channel for PIAF questions, and even if it was, PIAF sucks, I doubt version 2 is much better. |
15:37.40 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-qtwdoqallfarhvgm) |
15:38.06 | *** join/#asterisk serafie (~erin@nat/digium/x-wwlumuwthwyzunuj) |
15:38.14 | igcewieling | ~piaf |
15:38.38 | wasanzy | igcewieling: am not asking to use PIAF, because I don't want to use it so I was wondering if there is an alternative way |
15:38.57 | igcewieling | wasanzy: Do you understand what PBX In A Flash (PIAF) does? |
15:39.01 | *** join/#asterisk jylebleu (~jylebleu@78.208.220.3) |
15:39.08 | igcewieling | or core correctly, what it is |
15:39.36 | wasanzy | igcewieling: frankly, I don't understand |
15:39.55 | igcewieling | It is a gui distro like freepbx, which uses freepbx and adds lots of other stuff. |
15:40.00 | jmetro | you can make anything work in vanilla asterisk that works in a packaged asterisk |
15:40.14 | jmetro | packaged asterisk = elasticx, freepbx, pbiaf. |
15:40.32 | igcewieling | jmetro: don't forget trixbox! |
15:40.50 | jmetro | trixbox is the most fun of them |
15:41.31 | wasanzy | jmetro: I have asterisk installed from source (core) and I don't want to use any of those that is why am asking |
15:41.48 | igcewieling | wasanzy: no you do not need PIAF. Your question is answered. |
15:42.09 | igcewieling | you also do not need FreePBX, Trixbox, Elastix, or a camel. |
15:42.09 | wasanzy | right |
15:42.11 | wasanzy | thank you |
15:42.29 | igcewieling | though the camel would be the most useful of the 4 |
15:42.52 | wasanzy | so working with my sip.conf and extension.conf will be ok right? |
15:43.01 | jmetro | and possibly asterisk.conf |
15:43.24 | jmetro | i think they have you download modules to add to it |
15:43.31 | jmetro | but no pakages / guis / things that make it hard to use |
15:43.47 | wasanzy | jmetro: yea |
15:43.51 | igcewieling | wasanzy: unlikely |
15:44.09 | igcewieling | I believe google talk uses XMPP |
15:44.14 | boom^time | it does |
15:44.19 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
15:44.19 | *** mode/#asterisk [+o malcolmd] by ChanServ |
15:44.22 | igcewieling | then it is unlikely sip.conf will help. |
15:44.41 | igcewieling | I keep forgetting wasanzy doesn't read docs |
15:44.45 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-tcsrqdlxgrasnhwv) |
15:45.00 | wasanzy | igcewieling: do you have a link that can show me how to setup just a simple VOIP ? |
15:45.12 | igcewieling | ~book |
15:45.12 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
15:45.40 | igcewieling | I believe that talks about setting up google voice, If not a simple google search should turn up thousands of hits |
15:45.49 | wasanzy | igcewieling: I do read docs and when I don't understand the book is when I ask here |
15:45.50 | boom^time | I still have the first edition of that book somewhere that was giving out at the first astricon. |
15:45.53 | [TK]D-Fender | wasanzy: Google Voice is going to die shortly. |
15:46.10 | [TK]D-Fender | wasanzy: Google is moving it to Google Hangouts and dropping XMPP support |
15:46.26 | [TK]D-Fender | wasanzy: is is pretty much a waste of time |
15:46.36 | wasanzy | the main thing am working on is wolfram |
15:46.37 | igcewieling | [TK]D-Fender: google voice has been dying shortly for months. |
15:46.48 | igcewieling | wasanzy: what the hell is wolfram? |
15:47.05 | igcewieling | Is that the evil law firm from Buffy? |
15:47.24 | wasanzy | igcewieling: http://nerdvittles.com/?p=798 |
15:47.49 | igcewieling | ah. we can't really help with that |
15:48.47 | igcewieling | [TK]D-Fender: and honestly, based on his questions over the past few weeks I'm perfectly happy for him to waste says of time |
15:48.56 | igcewieling | s/say/days |
15:50.05 | [TK]D-Fender | igcewieling: Wolfram & Hart |
15:50.16 | igcewieling | [TK]D-Fender: ah, silly me. |
15:50.28 | [TK]D-Fender | igcewieling: And the was from Angel (the Buffy spin-off) |
15:50.55 | igcewieling | yup. Amy Acker is AWESOME |
15:50.57 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
15:50.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
15:51.17 | [TK]D-Fender | igcewieling: Agreed |
15:51.18 | saxa | ok, it does not work properly yet, i can hear the other side, but it seems the other side does not hear me. |
15:51.34 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-elgvtjpiahpviqdt) |
15:51.34 | *** mode/#asterisk [+o newtonr] by ChanServ |
15:51.37 | saxa | any ideas on where to check this ? |
15:51.45 | igcewieling | saxa: I must have missed the pastebin of your sip.conf [general] |
15:51.59 | saxa | probably i should set a lower resource consuming codec ? |
15:52.19 | [TK]D-Fender | igcewieling: She was in Dollhouse and still in Once Upon A Time (unless something has happened (NO SPOILERS!) |
15:52.43 | [TK]D-Fender | saxa: has nothing to do with codec. has everything to do with NETWORKING |
15:52.58 | saxa | oh ok |
15:53.02 | igcewieling | [TK]D-Fender: I watch neither, though if I'd known I might have watched them |
15:53.12 | saxa | so then i still have to check that thing |
15:53.27 | igcewieling | there is twice I asked. |
15:53.32 | *** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
15:54.23 | saxa | igcewieling: http://pastebin.com/8aCP25q5 <- here it is. |
15:55.13 | *** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax) |
15:55.26 | igcewieling | saxa: ther external ip address of your asterisk server is 201.50.103.98 ? |
15:56.09 | igcewieling | saxa: how many g729 licenses do you have? |
15:56.24 | saxa | yes, right now is that one 201.50.103.98 |
15:56.46 | saxa | igcewieling: do i have some ? |
15:57.07 | igcewieling | saxa: that is my question. how many g729 licenses did you purchase? |
15:57.11 | saxa | igcewieling: i never bought an g729 licence, unless it cames with the phone |
15:57.22 | *** join/#asterisk glambert (~glambert@37.157.50.80) |
15:57.26 | igcewieling | then you should not have g729 allowed in your codec list. |
15:57.32 | saxa | ok |
15:57.32 | igcewieling | not your current issue, but it will become one. |
15:57.41 | igcewieling | now do a new pastebin of a failed call with sip debug |
15:57.43 | saxa | let me disable it |
15:57.51 | saxa | ok a moment |
15:58.28 | glambert | Hi, we have an issue where the ptime the phone and server return in the invite is 20ms yet the RTP ptime from server is reaching 84ms and phone reaching max of 24ms, how do I work out why the server is no where near the negotiated ptime? |
15:58.33 | glambert | codec is g722 |
16:00.08 | igcewieling | glambert: what version of Asterisk |
16:00.26 | glambert | 1.4.44 |
16:02.39 | *** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com) |
16:03.53 | [TK]D-Fender | 1.4 is not supported, nor is 1.6.0, nor 1.6.1, nor 1.6.2 |
16:04.06 | igcewieling | I wish you the best of luck with that old version of Asterisk which is no longer supported and so few people use. |
16:04.15 | saxa | http://pastebin.com/b0tS7g1w |
16:04.34 | saxa | igcewieling: this is the new sip set debug peer casasip |
16:04.37 | boom^time | igcewieling, Hey! I have a lot of fond memories of that version of Asterisk. |
16:05.01 | igcewieling | boom^time: I have a lot of fond memories of my ex-wife, but sometimes it is time to move on. |
16:05.45 | glambert | we've used it for years and if it ain't broke and all that |
16:05.59 | [TK]D-Fender | glambert: It's broke now. |
16:05.59 | glambert | newer model of phone released and seem to have issues going through IVRs which is frustrating |
16:07.09 | [TK]D-Fender | 1.4 doesn't actually even transcode G.733... only passthrough |
16:07.13 | coppice | nothing seems to last like things used to :-) |
16:07.14 | [TK]D-Fender | G7.22* |
16:07.27 | [TK]D-Fender | gah |
16:07.38 | coppice | there's G.722 in 1.4 |
16:07.42 | glambert | thousands of calls going through it per day with no issues and MOS of >=3 on over 95% of calls doesn't indicate broken really, just an issue with this particular model going through IVRs, very strange |
16:07.55 | glambert | we've enabled g722, I believe we patched it |
16:08.01 | boom^time | igcewieling, I was speaking facetiously. I have a lot of fond memories of your ex-wife also. |
16:08.05 | glambert | but not me who would've done that |
16:08.28 | glambert | all of our calls use g722, internal, outbound or inbound |
16:08.36 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
16:08.54 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
16:09.22 | igcewieling | glambert: so no only are you using an old version of Asterisk, you are actually using an old version of Asterisk you applied custom patches to. |
16:09.30 | [TK]D-Fender | glambert: you are running a non-standard patched version of an Asterisk branch which has been off support for some time. Best of luck |
16:11.54 | glambert | There is a backport for 1.4 for g722 listed on voip-info.org which I can only assume we're using |
16:12.07 | boom^time | I don't understand how difficult could it really be to upgrade to 11.5? It's not like it would take _that_ long to handle the syntaxual changes |
16:13.10 | boom^time | within the configuration files |
16:13.42 | igcewieling | boom^time: doesn't matter. he won't upgrade to a supported version, most people here won't help with unsupported versions. end of story. |
16:14.44 | boom^time | But it's a terrible story. The premise is crap and the ending is depressing. |
16:15.43 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
16:23.22 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
16:32.51 | *** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com) |
16:34.43 | *** join/#asterisk psilikon (~joel@mail.vicimarketing.com) |
16:36.00 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
16:48.13 | dpeloquin | boom^time: it insists upon itself! |
16:54.29 | nam3l3zz | what could be the problem, x-lite /w username/pass/domain registers easy, but asterisk doesn't, exchanges /w OPTIONS packets, but no REGISTER ? |
16:54.37 | *** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com) |
16:55.20 | [TK]D-Fender | nam3l3zz: huh? |
16:55.40 | nam3l3zz | [TK]D-Fender: can i pm ? pastebins |
16:55.44 | [TK]D-Fender | nam3l3zz: you say registers easy then "no register".. Please be clear |
16:55.57 | [TK]D-Fender | just pastebin in public and mask ONLY passwords |
16:56.53 | nam3l3zz | [TK]D-Fender: x-lite a win app, sends a register request, but asterisk sends options request and register request doesn't follow, althou register field is defined in sip.conf |
16:57.37 | [TK]D-Fender | "register" in sip.conf is for ASTERISk to register to antoher service. this is NOT for setting up your x-lite\ |
16:57.48 | nam3l3zz | i know |
16:57.49 | nam3l3zz | :) |
16:57.51 | [TK]D-Fender | X-Lite registers to YOUR server, not the other way around |
16:58.23 | [TK]D-Fender | your are not being clear, and not showing configs an debug. |
16:59.04 | nam3l3zz | in the scenario, x-lite registers to a remote account , to which /w same account settings asterisk doesn't want to register |
16:59.17 | nam3l3zz | i will try to be clear |
16:59.34 | [TK]D-Fender | nam3l3zz: most services won't let you register MULTIPLE things to the same acount like that |
16:59.39 | [TK]D-Fender | nam3l3zz: SHOW US |
17:00.02 | nam3l3zz | i close x-lite and wait , then start asterisk |
17:00.07 | nam3l3zz | to avoid conflicts |
17:01.00 | nam3l3zz | but x-lite always registers, and wireshark i can see, that it actually starts from sending a register request packet, but asterisk starts from sending options request & never sends register after |
17:01.12 | [TK]D-Fender | Still not seeing configs.... |
17:01.18 | igcewieling | nor sip debug |
17:01.30 | [TK]D-Fender | and OPTIONS has NOTHING to do with registration to an outside service |
17:01.43 | [TK]D-Fender | that is due to "qualify" being set in a peer. |
17:01.57 | [TK]D-Fender | and your peer definitions have nothing to do with asterisk registering to anything |
17:03.44 | nam3l3zz | i get that, i'll try to be more specific, can asterisk be against me having quotes in the username ? |
17:03.44 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
17:03.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
17:03.51 | nam3l3zz | *dot |
17:04.08 | nam3l3zz | not quotes |
17:04.53 | [TK]D-Fender | awaits getting the configs |
17:05.04 | igcewieling | nam3l3zz: less talk, more do. |
17:05.37 | nam3l3zz | sent stuff with passwords to pastebin..... |
17:11.06 | nam3l3zz | http://pastebin.com/rPefaWAw |
17:11.07 | [TK]D-Fender | ... |
17:11.38 | [TK]D-Fender | nam3l3zz: your second register is DEAD |
17:11.52 | nam3l3zz | because of the quotes |
17:11.53 | nam3l3zz | ? |
17:11.59 | [TK]D-Fender | nam3l3zz: ALL registers have to appear after everything else in [general] and BEFORE the first entry |
17:12.10 | nam3l3zz | affirmative |
17:12.11 | nam3l3zz | thanx |
17:12.35 | *** part/#asterisk navaismo (~navaismo@189.241.68.101) |
17:12.43 | *** join/#asterisk navaismo (~navaismo@189.241.68.101) |
17:12.49 | nam3l3zz | found plenty shit examples /w register allover, thnx again |
17:13.10 | navaismo | Is there a way to ignore and no transmit dtmf when the call is bridged? |
17:13.11 | igcewieling | nam3l3zz: I bet the example in sip.conf.sample is correct. |
17:13.26 | igcewieling | doesn't ANYONE read the .conf.sample files?????? |
17:13.30 | [TK]D-Fender | your provider entries should all be "nat=no", canreinvite was changed to "directmedia" Wway back in 1.6 |
17:13.31 | nam3l3zz | most likely, 2 much trust 2 google almighty |
17:13.59 | [TK]D-Fender | Google doesn't write Asterisk guides |
17:14.03 | [TK]D-Fender | don't say "google" |
17:14.16 | nam3l3zz | patent breach ? |
17:14.19 | nam3l3zz | :) |
17:15.56 | nam3l3zz | [TK]D-Fender: although I have nat=yes in general, i need to put nat=no in every peer ? |
17:16.10 | igcewieling | nam3l3zz: just the ones not behind nat |
17:16.43 | nam3l3zz | found myself being a noob in this one |
17:17.08 | nam3l3zz | peers are remote, /w global ips, so not behind nat... nice |
17:21.43 | *** join/#asterisk yano (yano@freenode/staff/yano) |
17:24.04 | nam3l3zz | igcewieling: u may know, i can see in debug of sip, sip server reply 401 for dialing a number via it, but asterisk keep retransmitting dial request |
17:25.06 | navaismo | anyone? some hints? |
17:25.21 | nam3l3zz | igcewieling: http://pastebin.com/0ZmyjVrJ |
17:26.51 | igcewieling | you still have a NAT issue |
17:27.09 | igcewieling | did you read https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ? |
17:28.05 | nam3l3zz | http://pastebin.com/uMVMY3W1 |
17:28.10 | nam3l3zz | it seems like a nat issue |
17:28.15 | nam3l3zz | will read, thanx |
17:30.23 | igcewieling | has been working on an ASCII over DTMF protocol. Why? Because I can. 8-) |
17:31.58 | *** join/#asterisk theron (~theron@access.vitalsite.com) |
17:34.14 | nam3l3zz | igcewieling: got different reply http://pastebin.com/snbz7M3X , could nat be sorted ? |
17:34.30 | coppice | I used to build systems that allowed ASCII and Chinese entry by DTMF |
17:34.42 | igcewieling | nam3l3zz: unlikely |
17:34.59 | nam3l3zz | got it , caller id doesn't match server's requirement |
17:35.26 | igcewieling | coppice: I'm converting the text to base-16 encoding, replace all "f" with * and all "e" with #, wrap it in a packet with crc and retransmits and send it |
17:35.53 | coppice | oh, automated entry. I used to let people enter messages by hand |
17:36.04 | nam3l3zz | no, actually it is a registered number :\ |
17:36.16 | igcewieling | coppice: *nod* this is mainly for machine-machine communication. |
17:36.33 | igcewieling | I think I get something like 5cps |
17:37.08 | igcewieling | nam3l3zz: forbidden is a config issue |
17:37.30 | *** join/#asterisk serafie1 (~erin@nat/digium/x-mlwvqnetnqlzucda) |
17:38.29 | saxa | igcewieling: had a chance to take a look at the second debug output ? |
17:40.04 | igcewieling | saxa: still have a NAT issue. since the source port is 5060 I can only assume SIP SPI or SIP ALG on the router |
17:40.25 | igcewieling | though it could easily be something like a broken softphone too. |
17:48.44 | coppice | igcewieling: I built a paging for the deaf system. deaf chinese were actually prepared to learn what are termed the chinese telegraph codes - numeric codes for each chinese character - and use those codes with a DTMF phone to send Chinese pages to their friends |
17:49.32 | igcewieling | coppice: nice! Users willing to learn! I didn't know they existed. |
17:50.06 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
17:50.17 | igcewieling | We need an Allison sound file "Watson come here, I want to see you." |
17:50.18 | *** join/#asterisk serafie (~erin@nat/digium/x-ghlnfdtmikftyoct) |
17:53.05 | *** join/#asterisk bananapie (~david@194.189.18.64.static.oricom.ca) |
17:53.50 | bananapie | My SIP provider gave me a backup IP, I have the primary IP in the "host" field. How do I enter the backup IP into the SIP peer without modifying the dialplan? Thanks. |
17:54.57 | igcewieling | bananapie: you can't. |
17:55.01 | coppice | igcewieling: deaf people are willing to do quite a lot |
17:55.14 | igcewieling | create another peer, use your dialplan to failover |
17:56.10 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
17:56.10 | *** mode/#asterisk [+o pabelanger] by ChanServ |
17:56.29 | bananapie | igcewieling : I read that solution already, I was hoping to not use it. Thanks! |
17:56.51 | bananapie | I think if I use a dialplan macro, I could do it in such a way that would be easy to use |
17:57.24 | igcewieling | better to use a gosub |
18:01.39 | saxa | igcewieling: ok, will check that thing, SIP SPI or SIP ALG on my router. But its not a softphone |
18:09.22 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
18:16.14 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
18:30.12 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
18:45.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.169) |
18:46.40 | igcewieling | "Thank you for your unsolicited contacting of me about a job. I'm not interested, but as a courtesy I've added your contact information to our Direct Marketing List" |
18:47.22 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
18:47.24 | jmetro | oh yes |
18:47.26 | jmetro | oh yes please. |
18:47.32 | jmetro | I have signed you up for Cat Facts. |
18:47.50 | igcewieling | jmetro: I'm being nice, only signeing them for OUR DMA list. |
18:55.47 | jmetro | http://tinyurl.com/84laak3 |
18:56.41 | SuperNull | jmetro someone on reddit was doing that to me for a week.. |
18:58.55 | jmetro | To cancel, reply "ÐØѱ" within 30 seconds |
19:01.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
19:10.33 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
19:20.27 | *** join/#asterisk SGjunior (~sgjunior@out-pq-194.wireless.telus.com) |
19:30.00 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
19:30.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
19:38.25 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
19:41.33 | *** join/#asterisk felipealmeida (~user@177.157.207.99) |
19:48.47 | *** join/#asterisk mokmeister (~mokmeiste@86-41-115-212-dynamic.b-ras2.lmk.limerick.eircom.net) |
19:52.24 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
19:59.27 | *** join/#asterisk [Outcast] (~anonymous@50-200-130-22-static.hfc.comcastbusiness.net) |
20:06.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
20:09.26 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
20:11.57 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
20:16.50 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
20:23.04 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
20:25.32 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
20:29.09 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
20:48.31 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
20:51.47 | *** join/#asterisk g_r_eek (~g_r_eek@78-23-63.adsl.cyta.gr) |
20:53.42 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
20:57.31 | ChannelZ | BORED |
21:02.37 | saxa | igcewieling: this is a third debug i'm sending , i doublechecked that no SIP SPI or other thing is enabled on the phone side |
21:02.41 | saxa | http://pastebin.com/ZcmesZji |
21:03.15 | saxa | igcewieling: on the * side I'm 99% sure there is no one filter or anything related to SIP enabled. |
21:03.52 | saxa | igcewieling: i will double check it again, but i'm nearly 100% sure i have disabled all those things |
21:04.15 | saxa | another thing is that on the phone side a even disabled the firewall. |
21:10.53 | *** join/#asterisk iTrojan (~Adium@41.233.28.50) |
21:13.21 | iTrojan | please help |
21:13.27 | iTrojan | i keep getting this error ; |
21:13.28 | iTrojan | <PROTECTED> |
21:13.34 | iTrojan | http://pastebin.com/YzFu9Bd1 |
21:13.42 | iTrojan | this si my configurations |
21:14.02 | [TK]D-Fender | and just like it says... no match |
21:14.03 | iTrojan | it is driving me nutes |
21:14.06 | iTrojan | nuts |
21:14.10 | [TK]D-Fender | "nutes" indeed |
21:14.26 | [TK]D-Fender | because it isn't spelled EXTERN either.... |
21:14.45 | iTrojan | what is it then!! |
21:14.52 | [TK]D-Fender | and [new] should not "include" .... ITSELF |
21:14.59 | [TK]D-Fender | EXTEN <----------- |
21:15.06 | iTrojan | oh lol |
21:15.08 | [TK]D-Fender | ho do you not know this? |
21:15.12 | [TK]D-Fender | how* |
21:15.18 | [TK]D-Fender | ~book |
21:15.18 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
21:15.19 | [TK]D-Fender | ^^^^^^^^ |
21:15.25 | iTrojan | hmm maybe because I just learned about this 10 minutes ago? |
21:15.45 | [TK]D-Fender | I'd be interested to see who else wrote it that way |
21:15.47 | iTrojan | I already have a book |
21:15.51 | iTrojan | but I want to do this hello world |
21:15.56 | iTrojan | to get a feeling of pprogreessing |
21:16.06 | iTrojan | I googled alot |
21:16.10 | iTrojan | nothing works |
21:16.24 | [TK]D-Fender | "nothing works"? |
21:16.33 | [TK]D-Fender | what kind of "nothing"? |
21:16.59 | iTrojan | can you leave your saracsm ? |
21:17.07 | iTrojan | sarcasm !? |
21:17.07 | igcewieling | edges away and gets out the garlic |
21:17.16 | [TK]D-Fender | I'm just not sure what you mean by "nothing works" |
21:17.31 | [TK]D-Fender | so far I saw adialplan error I already gave you an answer for |
21:17.38 | [TK]D-Fender | Have you already applied the changes and retested? |
21:17.49 | iTrojan | http://pastebin.com/YzFu9Bd1 |
21:17.51 | iTrojan | oh not this |
21:18.01 | iTrojan | allowexternaldomains=yes |
21:18.05 | iTrojan | like setting this option |
21:18.17 | [TK]D-Fender | that's the same pastebin |
21:18.27 | iTrojan | yeah, I said not this.. |
21:18.34 | iTrojan | I meant to paste the option |
21:18.36 | [TK]D-Fender | So go fix it and see if it works |
21:18.42 | iTrojan | what to fix? |
21:18.46 | [TK]D-Fender | your dialplan |
21:18.52 | [TK]D-Fender | I already told you what your typo was |
21:18.55 | iTrojan | to change what? |
21:18.58 | [TK]D-Fender | Didn't you go and fix it? |
21:18.59 | iTrojan | ah that is it? |
21:19.01 | iTrojan | exten ? |
21:19.03 | [TK]D-Fender | EXTERN is not right |
21:19.06 | [TK]D-Fender | EXTEN |
21:19.08 | [TK]D-Fender | is |
21:19.10 | [TK]D-Fender | no "R" |
21:19.14 | iTrojan | ok ok let me do it |
21:19.17 | [TK]D-Fender | all those lines were wrong |
21:19.40 | iTrojan | aleast extern has a meaning |
21:19.47 | [TK]D-Fender | EXTENsion <- |
21:19.54 | iTrojan | EXTERNal |
21:19.57 | [TK]D-Fender | has a meaning that isn't related |
21:20.01 | [TK]D-Fender | that is an EXTENsion |
21:20.10 | [TK]D-Fender | as in EXTENSIONS.conf |
21:20.29 | *** join/#asterisk asteriskmonkey (~darlek@server.xcel.on.ca) |
21:20.30 | iTrojan | hm that makes sense |
21:20.41 | asteriskmonkey | does callerid routing not work in realtime? |
21:20.51 | iTrojan | wow that works |
21:20.57 | asteriskmonkey | i have exxten => did/callierd .... dosnt seem to work |
21:21.12 | iTrojan | although you're not so 'friendly' , but thank you [TK]D-Fender |
21:21.55 | [TK]D-Fender | iTrojan: do remove that "include" as well |
21:22.03 | *** join/#asterisk SGjunior (~sgjunior@96.127.222.20) |
21:22.32 | *** join/#asterisk zozeer (~zozeer@192.222.104.1) |
21:23.35 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
21:24.28 | zozeer | I have a random question about call forwarding. if 20 calls 21 and 21 is forwarded to 22 and 22 is forwarded to 23 and 23 does not answer will it drop me in 23's mailbox, or 20's mailbox? |
21:25.13 | [TK]D-Fender | zozeer: it goes wherever your dialplan tells it to go. |
21:25.22 | [TK]D-Fender | zozeer: what did you tell it it do for that? |
21:25.33 | zozeer | so I can nest as far deep as I want? |
21:25.54 | [TK]D-Fender | these are phones bouncing calls.... |
21:26.10 | [TK]D-Fender | it bounces as far as your devices keep pointing elsewhere |
21:26.21 | zozeer | http://www.voip-info.org/wiki/view/Asterisk+call+forwarding example 5 |
21:26.25 | [TK]D-Fender | or until where you bounce them to doesn't try to dial another |
21:26.37 | zozeer | I was having a debate with our old nortel guy. |
21:27.03 | [TK]D-Fender | * call processing has nothing to do with the "Nortel Experience" |
21:27.17 | [TK]D-Fender | You can an infinite loop any time you want |
21:27.25 | [TK]D-Fender | Dialplan = programming = whatever you tell it to do |
21:27.41 | [TK]D-Fender | So YOU are the limit of what you want to do |
21:27.47 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
21:28.01 | [TK]D-Fender | So if you copied that code verbatim.. trace it. Where do you see it going? |
21:28.22 | zozeer | [TK]D-Fender, would that form a loop? as in it just keeps bouncing down the line? (I am a router guy so some of this is a little counter to what I know) |
21:28.33 | [TK]D-Fender | trace the code... |
21:30.32 | [TK]D-Fender | That sample is also pretty broken |
21:30.51 | *** part/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
21:31.27 | [TK]D-Fender | And this is a sample for "dialplan controlled callforwarding" as a concept |
21:31.33 | [TK]D-Fender | As opposed to "device-based |
21:33.31 | asteriskmonkey | does pattern matching not work in realtime? |
21:34.38 | jmetro | everything works in realtime |
21:34.43 | navaismo | asteriskmonkey i have used realtime static and that works |
21:35.03 | navaismo | you should show us your cli output |
21:35.05 | asteriskmonkey | i have in my realtie db did/callerid and it dosnt seem to get hit |
21:35.30 | asteriskmonkey | callerid based routing with the / method dosnt seem to get hit in my realtime :/ |
21:35.33 | asteriskmonkey | is that a bug? |
21:35.40 | [TK]D-Fender | asSHOW US |
21:35.54 | [TK]D-Fender | and be specific on versions |
21:35.57 | [TK]D-Fender | you know better. |
21:36.03 | asteriskmonkey | lol |
21:36.20 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:98bc:4a61:946:6c89) |
21:37.09 | asteriskmonkey | version 10.3.0, dialplan (switch statement then a catchall, the realtime i have an entry based on callerid which isnt getting hit, the catch all in teh flat file is taking it) |
21:37.56 | asteriskmonkey | exmple in my realtime would be 1234/555 Playback ttweasels.. |
21:38.15 | asteriskmonkey | the call iv nooped i can see correct criteria.. so i know its just not catching in realtime |
21:38.24 | [TK]D-Fender | real DB dump, real call with core debug |
21:38.33 | [TK]D-Fender | none of this pseudo nonsense... |
21:39.09 | [TK]D-Fender | because that first answers comes back as "your DB values are wrong and has more errors than syntax" :p |
21:39.18 | asteriskmonkey | thanks, ill do further beating and dump, though id just do a quick poke to see if the exten/callerid was buster like it was in 2006 :) |
21:40.13 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:98bc:4a61:946:6c89) |
21:40.21 | navaismo | Im going to ask for the last time..today: Is there a way to ignore & dont transmit dtmf to the called party on a bridged call? |
21:41.15 | jmetro | hm |
21:41.22 | jmetro | you mean dont let the called party hear dtmf |
21:41.23 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:98bc:4a61:946:6c89) |
21:41.31 | jmetro | but also asterisk doesnt hear dtmf? |
21:45.53 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ltymotqbdmouplgy) |
21:45.53 | *** mode/#asterisk [+o newtonr] by ChanServ |
21:46.36 | *** join/#asterisk navaismo (~navaismo@189.241.68.101) |
21:46.58 | navaismo | jmetro yes |
21:47.21 | navaismo | dont let the called party hear dtmf and also asterisk ignore them |
21:47.43 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
21:48.43 | jmetro | I would expect it to be in core show app dial |
21:48.45 | jmetro | but i dont see anything |
21:48.59 | navaismo | nope there is nothing about that |
21:49.01 | imcdona | I'd like to remind everyone that freenode supports connecting via SSL. If you have trouble connecting via SSL on the default port, you can try these: on ports 6697 (SSL only), 7000 (SSL only), 7070 (SSL only) |
21:50.00 | jmetro | navaismo: sorry nav, dont see anything in my google-fu |
21:50.18 | navaismo | no problem, neither do i |
21:50.51 | *** join/#asterisk iTrojan (~Adium@41.233.28.50) |
21:54.15 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
21:58.37 | saxa | ok, i understand that in some way my dtmf tones do not get transmitted |
21:58.46 | saxa | or at least they do not reach asterisk |
21:58.59 | saxa | dtmfmode=auto |
21:59.26 | saxa | but when i try to access my mailbox it doesnt recognise the password |
22:01.25 | saxa | any ideas what should i look at ? |
22:05.19 | newtonr | saxa, You should figure out what style of DTMF your endpoint is using when talking to Asterisk. You should also look at https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information. With that information you'll be headed down the right path |
22:06.41 | saxa | [Aug 19 19:55:40] WARNING[21114]: app_voicemail.c:9834 vm_authenticate: Couldn't read username |
22:06.59 | saxa | newtonr: thanks |
22:14.21 | saxa | http://pastebin.com/4z7v5FVH |
22:14.46 | *** join/#asterisk mardok45 (~mardok@74-140-49-109.dhcp.insightbb.com) |
22:15.00 | saxa | This is a sip debug on the casasip peer who seems doesnt get recognised the dtmf tones |
22:16.06 | [TK]D-Fender | don't assume "auto" will work |
22:16.17 | [TK]D-Fender | Set the proper mode yourself |
22:18.58 | mardok45 | I'm using Asterisk for my Google Voice phone. Is there a way to also is it for SMS? I'm using FreePBX. |
22:19.08 | mardok45 | to also use it for SMS?* |
22:20.38 | jmetro | you should check #freepbx |
22:20.46 | jmetro | as we cannot accurately support freepbx config files here |
22:20.55 | tm1000 | jmetro: +1 |
22:21.02 | tm1000 | a plesant response! :-) |
22:21.42 | jmetro | Well who says anyone has to be nasty anyway :3 |
22:22.29 | mardok45 | Linus Torvalds? |
22:23.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
22:23.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |
22:23.33 | tm1000 | mardok45: ha! |
22:24.36 | [TK]D-Fender | No, with linus it's simply bundled with the package |
22:24.56 | [TK]D-Fender | And no, FreePBX does not support SMS |
22:25.04 | saxa | [TK]D-Fender: thx |
22:27.54 | newtonr | saxa, get a pcap of SIP and RTP traffic, if your endpoint is doing rfc2833 then read up on that and look at your traffic to see if its hitting Asterisk. |
22:28.31 | newtonr | saxa, also, if it is hitting Asterisk, turning on DTMF debug (see logger.conf) may give you additional info. |
22:31.07 | saxa | http://pastebin.com/L13rqUbf this one is done with dtmfmode=rcf2833 |
22:31.43 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
22:31.57 | saxa | newtonr: ok many thanks, will try also with the dtmf debug |
22:32.41 | saxa | goes to eat a pizza, l8r i'm back here, thx for now. |
22:37.25 | *** join/#asterisk SGjunior (~sgjunior@out-pq-194.wireless.telus.com) |
22:50.03 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
22:51.09 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
23:03.13 | igcewieling | Nothing wrong with gently pointing people to #FreePBX, but if they don't listen we get out the garlic and wooden stakes and take care of the problem. |
23:05.55 | *** join/#asterisk mchou_ (~quassel@unaffiliated/mchou) |
23:09.25 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
23:10.51 | *** join/#asterisk PreludeZzzz (~preludeZz@66.49.157.140) |
23:10.54 | PreludeZzzz | hello everyone |
23:11.12 | PreludeZzzz | a quick question if someone can guide me in the right direction |
23:11.29 | PreludeZzzz | i want to send an audio tone to a particular live asterisk channel ( one side i guess ) .. how do i do that ? |
23:11.50 | PreludeZzzz | say someone is on a call and its been more then say 1/2 hour.. maybe i wnat to send a warning beep that they have exceed a particular time window on a call |
23:11.56 | PreludeZzzz | how does one send a tone to a live channel that way ? |
23:15.58 | igcewieling | for that SPECIFIC thing use the options to Dial() |
23:16.03 | ChannelZ | Dial() can do time limits with warnings |
23:19.27 | *** join/#asterisk iTrojan (~Adium@50.7.50.53) |
23:23.27 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
23:35.20 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.94) |
23:45.04 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
23:55.35 | *** join/#asterisk SGjunior (~sgjunior@out-pq-194.wireless.telus.com) |