| 00:00.26 | tilt_ | I setup the webphone to use outbound proxy to asterisk over tls, but the peer shows as unreachable |
| 00:00.40 | tilt_ | it only allow peer to outbound call not inbound |
| 00:00.49 | beanie | navaismo - http://bin.cakephp.org/view/620293364 |
| 00:01.33 | navaismo | tilt_, that means is unregistered if you reload the web again do you see it as OK |
| 00:02.05 | tilt_ | yeah , i thought so too, but nope it stays unreachable, but is allowed to make outbound calls...weird |
| 00:02.48 | navaismo | beanie, seems like you don't have any peers, And you are using some distro based on FreePBX you need to go to Extensions, then add sip peer |
| 00:02.48 | tilt_ | though over udp (no outbound tls proxy) everythign works as expected |
| 00:03.44 | navaismo | tilt_, can make calls because the peer know the server IP and has the password but asterisk can't reach the peer to send calls |
| 00:04.02 | navaismo | You can enable the debug on the register process and we can see what happens |
| 00:04.17 | tilt_ | k ill check now thanks |
| 00:04.36 | beanie | navaismo is that within freepbx |
| 00:05.18 | navaismo | beanie, yes |
| 00:05.28 | navaismo | you dont edit files using freepbx |
| 00:05.32 | navaismo | never |
| 00:06.44 | beanie | navaismo i'll copy what ive got in extensions.conf if that's the file you're talking about? |
| 00:08.09 | navaismo | beanie, nope I was talking about sip.conf but remember dont edit the files when you are using freepbx, all configus are made by the GUI |
| 00:11.38 | beanie | navaismo - http://bin.cakephp.org/view/1476205047 |
| 00:12.53 | navaismo | beanie, forgot the files, and use the GUI |
| 00:13.14 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 00:13.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.35) |
| 00:13.42 | tilt_ | navaismo: do you configure your webphone with asterisk as an outbound proxy over tls? |
| 00:14.36 | tilt_ | I think asterisk is trying to reply to the sipml5 with "sips" and sipml5 is not responding to request |
| 00:14.42 | tilt_ | so then the peer becomes unreachable |
| 00:15.49 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.100) |
| 00:16.00 | beanie | ah ok |
| 00:16.11 | beanie | when you say the gui you mean log into free pbx? |
| 00:16.24 | beanie | (novaismo) |
| 00:16.35 | navaismo | beanie, Yes |
| 00:16.46 | navaismo | tilt_, no, I dont use TLS yet |
| 00:17.19 | tilt_ | ahh...bummer need to figure this out |
| 00:17.42 | navaismo | yes time to test here too |
| 00:17.52 | tilt_ | back to my first question is udp secure over the webrtc2sip gateway when using WSS? |
| 00:18.07 | tilt_ | I cant really find much documentation on it |
| 00:18.13 | beanie | novaismo - i've set up two extensions as generic sip device |
| 00:18.16 | beanie | already |
| 00:18.26 | navaismo | tilt_, acorrding to Mamadou(doubango's moderator) they have a Guide to setup |
| 00:18.59 | tilt_ | yeah I have read the guide, ill go look around the google group. thanks |
| 00:19.55 | navaismo | beanie, can you upload screenshots of the extensions page |
| 00:20.03 | beanie | yeah sure :-) |
| 00:20.10 | beanie | where can i upload them? |
| 00:20.53 | beanie | applications>extensions? |
| 00:23.36 | navaismo | no hoted site like imgur.com |
| 00:26.38 | navaismo | tilt_, on your xml file are you setyting the wss;*;10062 and set it in you javascript code with wss://ip:10062? |
| 00:28.11 | beanie | navaismo http://img43.imageshack.us/img43/471/f95z.png |
| 00:29.52 | beanie | navismo 2nd page http://img10.imageshack.us/img10/7838/74y.png |
| 00:30.42 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-fuhybtxzityqwvid) |
| 00:30.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:31.44 | beanie | navaismo - page three http://img6.imageshack.us/img6/7835/jq9.png |
| 00:31.49 | beanie | all the same extension |
| 00:32.22 | navaismo | beanie, you need to apply config changes |
| 00:32.52 | beanie | i have done previously, i just renamed the name to test 1 but didn't save it |
| 00:33.00 | beanie | so ignore that |
| 00:36.27 | tilt_ | navaismo: yeah, wss://hostname:10062, and it works, I just want to make sure the udp is being transported through the WSS, which I think it is. |
| 00:38.48 | navaismo | yes acoording to the doubango people that is the secure way |
| 00:38.56 | navaismo | but i never tried so i cant confirm |
| 00:39.41 | beanie | any ideas Navaismo? |
| 00:39.49 | tilt_ | ok great :) would be alot easier if I could figure our the asterisk hiccup with WSS |
| 00:40.17 | navaismo | well in the wiki of asterik maybe you will fins something about that |
| 00:40.51 | tilt_ | its pretty vague, couldnt find anything helpful. |
| 00:41.12 | *** join/#asterisk moos3 (~textual@cpe-72-224-215-87.maine.res.rr.com) |
| 00:42.32 | navaismo | beanie, this is the normal procedure -->http://www.youtube.com/watch?v=INBtlp6mopQ the part of entering in the cli is only to see if was created is not necessary to do it |
| 00:43.02 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 00:44.30 | beanie | navaismo - theres nothing there that i haven't done |
| 00:50.17 | navaismo | try in the asterisk cli: sip show peer 048600 load |
| 00:51.07 | beanie | peer 048600 not found |
| 00:52.09 | navaismo | apply again the changes in the GUI and see if your asterisk cli show a lot( lot... lot) messages then try again |
| 00:52.46 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 00:54.41 | beanie | navaismo http://bin.cakephp.org/view/856168087 |
| 00:55.13 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-fuhybtxzityqwvid) |
| 00:59.39 | *** join/#asterisk Draecos (~Draecos@203-59-205-139.dyn.iinet.net.au) |
| 01:00.10 | navaismo | sip show peers still empty? |
| 01:01.26 | navaismo | I dont know what is happening I do as the last vide and then I tes like this-->http://www.youtube.com/watch?v=xQqURU-l_gw and the peer was created and working correctly. |
| 01:01.52 | navaismo | Do you see warnings in the freepbx system status(dashboard)? |
| 01:03.26 | beanie | navaismo - http://bin.cakephp.org/view/1340661692 |
| 01:03.38 | beanie | navaismo - one heck of a lot of errors on the dashboard!!! |
| 01:04.46 | beanie | navaismo - dashboard screenshot -http://img542.imageshack.us/img542/129/w61.png |
| 01:06.44 | navaismo | Ok at least your peer now is loaded you can try to register with your phone |
| 01:07.13 | beanie | the previous screenshot before last contains the errors when i tried to login with softphone - it didn't work |
| 01:08.07 | beanie | ok i'm in! |
| 01:08.16 | beanie | what was the issue, |
| 01:08.24 | navaismo | username mismatch, have <048600>, digest has <IH048601> |
| 01:08.45 | beanie | but it was not even coming up as a sip peer before |
| 01:09.14 | beanie | all as i did was changed the display name |
| 01:09.17 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 01:09.30 | *** join/#asterisk tilt_ (~tilt@173-13-180-109-sfba.hfc.comcastbusiness.net) |
| 01:09.45 | navaismo | about the warning in the dashborad: the first 3 warinig said you need to chown those files to asterisk, I mean change the owner of those files whit chown asterisk.asterisk <filename in the warning> |
| 01:10.53 | navaismo | about the symlink warning move those files to <filenameinwarning>.conf to <filenameinwarning>.conf.old, for example: if warinig say about logger.conf move to logger.conf.old, do that to all files in the warning |
| 01:11.02 | Katty | ASTERISK. |
| 01:11.09 | beanie | oh crikey - this i'm not sure how to do novaiso |
| 01:11.18 | beanie | novaismo* |
| 01:12.06 | navaismo | beanie, I got to go before people sdtart to yell about this is #asterisk and not #freepbx but as always they cant help but they can YELL |
| 01:12.12 | navaismo | beanie, good luck |
| 01:12.29 | beanie | navaismo - i'm no further forward after 4 hours :( |
| 01:12.43 | beanie | it would help if at least i could get rid of these warnings :-) |
| 01:12.52 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 01:12.59 | navaismo | try to read or understan my last messages about that |
| 01:13.01 | navaismo | see you |
| 01:13.20 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 01:13.35 | *** join/#asterisk DBordello (~DBordello@unaffiliated/dbordello) |
| 01:14.12 | beanie | is there actually a room called #freepbx? |
| 01:14.14 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 01:15.28 | Katty | beanie: yesh. |
| 01:15.40 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 01:20.20 | beanie | could anyone guide me in how to move these files and sort the core warnings out - they are clearly causing the problems in asterisk :-) just spent 4 hours doing endless screenshots with nothing resolved :-( |
| 01:27.29 | Kobaz | what's the minimum asterisk version for sip video |
| 01:27.33 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 01:27.40 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 01:30.43 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 02:02.58 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 02:04.18 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 02:11.06 | *** join/#asterisk dave4925_ (~dave4925@24-116-139-191.cpe.cableone.net) |
| 02:27.08 | *** join/#asterisk tilt_ (~tilt@173-13-180-97-sfba.hfc.comcastbusiness.net) |
| 02:44.37 | chuckf | Kobaz: I think that 1.8 had some experimental support for video as I recall |
| 02:45.30 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 02:46.23 | tm1000 | beanie: go ask in #freepbx |
| 02:47.24 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 02:54.10 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 03:13.00 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 03:32.05 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pbluxzbcbbyriwit) |
| 03:44.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.193) |
| 03:55.43 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 03:56.57 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 03:57.04 | *** join/#asterisk mquin (~mquin@freenode/staff/mquin) |
| 03:57.14 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 04:03.13 | *** join/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 04:10.55 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 04:17.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.178) |
| 04:36.53 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:06.13 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 05:14.13 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:14.14 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:17.05 | *** join/#asterisk andrewyager (~andrewyag@CPE-144-132-193-27.nsw.bigpond.net.au) |
| 05:18.36 | *** join/#asterisk andrewya_ (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 05:41.37 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 05:42.19 | *** join/#asterisk suneye (~atcmmi@119.139.63.101) |
| 05:50.29 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:01.24 | *** join/#asterisk asvx (~svx@193.105.11.73) |
| 06:05.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.178) |
| 06:09.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 06:20.18 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-veayvhzuwtygzimv) |
| 06:22.05 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 06:24.56 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 06:30.53 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 06:33.50 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 06:40.10 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 06:40.33 | *** join/#asterisk Rumbles (~Rumbles@31.205.54.123) |
| 07:02.31 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-zqhordznmoulifpm) |
| 07:06.06 | *** join/#asterisk grEvenX (~even@110.84-49-69.nextgentel.com) |
| 07:10.48 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:10.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 07:18.00 | *** join/#asterisk LokiScarlet (~loki@crantrap.cranberrytrap.org) |
| 07:23.00 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 07:25.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:27.26 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.53.110) |
| 07:33.42 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
| 07:39.15 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 07:39.16 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
| 07:41.56 | *** join/#asterisk areski (~areski@80.174.255.25.dyn.user.ono.com) |
| 07:42.43 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 07:44.15 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:45e1:88e9:f9a6:e56a) |
| 07:52.55 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:53.20 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 07:54.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:56.02 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 07:59.02 | *** join/#asterisk btse (~btse@c83-253-253-142.bredband.comhem.se) |
| 08:04.36 | bombev | guys |
| 08:04.39 | bombev | <PROTECTED> |
| 08:04.51 | bombev | why I am getting this notice when try to register IAX extension |
| 08:05.47 | ChannelZ | bad auth? |
| 08:06.20 | kaldemar | it's not the auth, but ACL. |
| 08:06.56 | ChannelZ | bad ACLs are the worst. |
| 08:07.25 | kaldemar | so, check permit/deny/acl settings. |
| 08:09.51 | mirela666 | hey |
| 08:11.10 | mirela666 | is there a way to make a Queue like: caller stays in it 30 seconds and there are 3 or more Agents, first is ringing for 15sec and than second for 15sec than leave Queue... |
| 08:12.42 | mirela666 | Entering Queue(QueueName,n) with n option and timeout = 30 in queues.conf is making caller stay for 30 sec in qureue |
| 08:13.14 | mirela666 | I tried strategy=rrmemory |
| 08:13.36 | mirela666 | but can't limit ring to agent to 15 sec for example |
| 08:14.32 | bulkorok | mirela666: this is possible... |
| 08:14.50 | mirela666 | maybe timeout=15 and than Queue again, and increasing counter |
| 08:16.27 | mirela666 | I always have default ring of 27000 ms |
| 08:16.28 | bulkorok | so... you set timeout=15 in queues.conf for ringing queuemembers for 15 secoinds.... |
| 08:16.57 | mirela666 | bulkorok: ok... |
| 08:16.59 | bulkorok | and in dialplan you call Queue() with 30 seconds timeout |
| 08:17.28 | bombev | what about |
| 08:17.28 | bombev | ERROR[3633] chan_iax2.c: Call rejected, CallToken Support required. If unexpected, resolve by placing address 82.142.106.150 in the calltokenoptional list or setting user 280 requirecalltoken=no |
| 08:17.29 | bulkorok | so... the caller is max. 30 secs in queue and 2 members are called max. 15 secs... |
| 08:18.42 | bulkorok | bombev: I know that thios info for years ago comming up on the screen... setting the requiredcalltoken=no makes it disappear... but I never need to investigate what a calltoken is... ;-) |
| 08:20.36 | mirela666 | bulkorok: aha I thought that Application timeout will just overide the .conf timeout |
| 08:21.28 | bulkorok | mirela666: sample-config mentions that not that way... |
| 08:21.43 | mirela666 | bulkorok: ok thanks, i'll try it now |
| 08:22.07 | bombev | bulkorok I set up requiredcalltoken=no and now its okay |
| 08:23.11 | bulkorok | bombev: let me know what a calltoken is for, when you know it ;-) |
| 08:23.21 | bombev | oks |
| 08:23.34 | bombev | :) |
| 08:35.44 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 08:38.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:50.21 | *** join/#asterisk kresp0 (~kresp0@18.Red-83-42-125.dynamicIP.rima-tde.net) |
| 09:01.42 | mirela666 | bulkorok: not good, timeout is 15s and then queue is left |
| 09:03.49 | bulkorok | https://wiki.asterisk.org/wiki/display/AST/Application_Queue |
| 09:06.37 | bulkorok | and check the sample configs |
| 09:08.32 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 09:12.53 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-txsnnfhkwouaccac) |
| 09:17.53 | *** join/#asterisk osteenbergen (~osteenber@82.94.204.46) |
| 09:18.37 | kaldemar | bulkorok, bombev: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Security_id36001692.html |
| 09:18.47 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/IAX2+Security#IAX2Security-CallTokenValidation |
| 09:19.17 | kaldemar | there was a security advisory about that too, just don't remember when and what it was called. |
| 09:20.15 | bulkorok | ah ok |
| 09:20.30 | *** join/#asterisk vlad_starkov (~vlad_star@194.154.71.230) |
| 09:25.10 | MrQuist | Why is the documentation of asterisk so messed up? The wiki is wrong and voip-info is completely wrong / outdated |
| 09:25.24 | *** join/#asterisk vlad_starkov (~vlad_star@194.154.71.230) |
| 09:27.26 | bulkorok | ~book |
| 09:27.27 | 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 |
| 09:28.03 | bulkorok | mmh... this is old... 4 is out afair |
| 09:35.57 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 09:38.25 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:39.03 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 09:49.51 | ectospasm | bulkorok: 4 is still in OFPS |
| 09:51.37 | bulkorok | ah ok... |
| 09:56.49 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 09:57.12 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-uucpdrkuexuttxev) |
| 10:12.55 | kaldemar | MrQuist: what in the asterisk wiki is wrong? |
| 10:26.50 | *** join/#asterisk kresp0 (~kresp0@18.Red-83-42-125.dynamicIP.rima-tde.net) |
| 10:31.04 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:39.02 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 10:39.05 | wasanzy | hi |
| 10:39.13 | hebber | Wiki is wrong - its just hard to find it |
| 10:39.14 | wasanzy | how do you change sound formats? |
| 10:39.51 | hebber | sound files? |
| 10:42.04 | wasanzy | the sound is in gsm and I want to change it to wav or mp3 |
| 10:44.25 | MrQuist | kaldemar, https://wiki.asterisk.org/wiki/display/AST/Application_GotoIfTime |
| 10:44.33 | MrQuist | Syntax: GotoIfTime(timesweekdaysmdaysmonths[timezone]labeliftrue:labeliffalse) |
| 10:45.06 | MrQuist | GotoIfTime(10:00-12:00fri-monjun-marcall:message); |
| 10:45.09 | MrQuist | that won't work :) |
| 10:47.46 | wasanzy | any idea/ |
| 10:47.48 | wasanzy | ? |
| 10:50.46 | kaldemar | MrQuist: that's odd. "core show application GotoIfTime" prints GotoIfTime(times,weekdays,mdays,months[,timezone]?[labeliftrue][:labeliffalse]). |
| 10:51.07 | kaldemar | MrQuist: for syntax, asterisk itself is the best source. |
| 10:51.37 | kaldemar | wasanzy: download a sound package in another format. |
| 10:52.24 | wasanzy | ok but the existing one can not be converted? |
| 10:54.09 | kaldemar | sure you can convert them if you want, but why bother when you can just download a package here: http://downloads.asterisk.org/pub/telephony/sounds/ |
| 10:55.20 | MrQuist | kaldemar, i know, but i think a wiki is supposed to have correct information doesn't it? |
| 10:55.35 | MrQuist | kaldemar, i got the right information nevertheless, but i just think its bad |
| 10:56.30 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 10:57.39 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 10:57.42 | wasanzy | how do I convert them? |
| 11:01.04 | kaldemar | MrQuist: sure, it should have correct syntax. |
| 11:01.15 | kaldemar | wasanzy: why do you insist on converting them? |
| 11:01.16 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 11:01.37 | wasanzy | is a self recorded sound |
| 11:01.45 | MrQuist | wasanzy -> use soc |
| 11:01.47 | MrQuist | sox* |
| 11:02.09 | wasanzy | is it a linux command? |
| 11:02.29 | MrQuist | sox input.wav -r 8000 -b 16 output.wav |
| 11:02.31 | MrQuist | yes it is |
| 11:02.45 | MrQuist | it will convert a .wav to the proper asterisk-kind of file |
| 11:02.53 | MrQuist | cq. format |
| 11:03.07 | kaldemar | wasanzy: you could also convert with asterisk, "core show help file convert" |
| 11:03.42 | wasanzy | yea that is what I want thax, I will chck |
| 11:06.51 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 11:12.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:19.08 | *** join/#asterisk blee (~blee@50-89-200-235.res.bhn.net) |
| 11:32.08 | *** join/#asterisk vlad_starkov (~vlad_star@194.154.71.230) |
| 11:32.28 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:38.10 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 11:41.59 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 11:43.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:50.05 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 11:57.32 | *** join/#asterisk apurvtwr (b64722fb@gateway/web/freenode/ip.182.71.34.251) |
| 12:07.13 | apurvtwr | Hi everyone. I have got an asterisk1.8.21 server with a GrandStream GXP1400 SIP phone registered on it. When I make a call to this SIP phone from a PSTN line, the voice quality is not good ( the voice heard on the SIP phone breaks. I have checked if this is due to limiting upload/download speed, but the network speed doesn't look like an issue) |
| 12:07.55 | apurvtwr | I am confused on where to look next to debug this issue |
| 12:08.08 | apurvtwr | any help would be great. |
| 12:08.16 | *** join/#asterisk davlefouAMD (~david@41.225.212.107) |
| 12:09.43 | apurvtwr | I have also tried changing codec from G729 to ulaw, that didn't make much affect |
| 12:13.59 | ke-esc | Can anyone point me to a good "best practices" type document for Asterisk? I'm going to be building out a new installation and want to make sure I start doing it correctly from the get-go. Particularly I'm interested in any thoughts on how to best virtualize the entire setup- what aspects should be isolated on their own VM, etc. |
| 12:16.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:20.19 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 12:25.49 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:27.03 | *** join/#asterisk izbushka (~izbushka_@193.23.225.11) |
| 12:31.51 | *** join/#asterisk izbushka (~izbushka_@193.23.225.11) |
| 12:40.28 | carrar | *Y*A*W*N* |
| 12:48.14 | bombev | guys why rtptimeout doesn hang up the call |
| 12:48.22 | bombev | I mean stucked call |
| 12:50.55 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:51.58 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tasonpvmqatnbbuc) |
| 12:52.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:53.10 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.100) |
| 12:55.22 | WIMPy | MAyeb rtp doesn't go through Asterisk? |
| 12:55.31 | WIMPy | And it's always one second too late. |
| 13:05.18 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 13:08.26 | *** join/#asterisk italorossi (~italoross@ec2-54-232-199-58.sa-east-1.compute.amazonaws.com) |
| 13:08.50 | *** join/#asterisk italoros_ (~italoross@187.60.66.11) |
| 13:13.36 | *** join/#asterisk grEvenX (~even@110.84-49-69.nextgentel.com) |
| 13:13.41 | bombev | WIMPy well how to deal with stucked channels do you have any idea |
| 13:14.46 | *** join/#asterisk vandyk (~quassel@189.59.4.180) |
| 13:14.48 | Kobaz | bombev: deadlock? |
| 13:14.51 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.181) |
| 13:14.53 | Greenlight | What exactly do you mean by "stuck" ? |
| 13:15.24 | Greenlight | Does it show in "core show channels"? Can you hang it up from the CLI ? |
| 13:15.27 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:16.58 | bombev | sometimes yes |
| 13:19.44 | Kobaz | asterisk version? |
| 13:23.16 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 13:24.48 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 13:26.06 | Katty | snores |
| 13:26.41 | *** join/#asterisk LokiScarlet (~loki@crantrap.cranberrytrap.org) |
| 13:28.55 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 13:33.17 | bombev | 1.8.11 |
| 13:33.25 | bombev | Kobaz |
| 13:35.17 | Kobaz | upgrade? |
| 13:35.37 | Kobaz | 1.8.22.0 (2013/05/17) |
| 13:37.48 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:37.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:40.26 | *** part/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 13:40.59 | *** join/#asterisk serafie (~erin@nat/digium/x-zbtdtvswipalzfrf) |
| 13:45.26 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 13:48.22 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 13:50.12 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 13:56.45 | *** join/#asterisk Draecos (~Draecos@203-59-205-139.dyn.iinet.net.au) |
| 14:01.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.181) |
| 14:04.01 | *** join/#asterisk serafie (~erin@nat/digium/x-fxuzkbnzvzjpscdn) |
| 14:05.48 | *** join/#asterisk DruidZ (~darcy@dilbert.druid.net) |
| 14:09.05 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 14:09.30 | *** join/#asterisk davlefouAMD (~david@41.225.117.60) |
| 14:10.55 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 14:12.28 | bombev | Kobaz yes I did the upgrade up to 1.8.22 |
| 14:12.37 | bombev | but same thing happens again |
| 14:13.19 | [TK]D-Fender | bombev: Put an absolute limit on your calls or something then |
| 14:13.55 | Kobaz | bombev: compile with DEBUG_LOCKS |
| 14:14.08 | Kobaz | bombev: and when asterisk freezes, pastebin the output of 'core show locks' |
| 14:14.09 | bombev | [TK]D-Fender well how to put absolute limit |
| 14:14.12 | Greenlight | He's still not clarified what he means by stuck, or answered any questions |
| 14:14.27 | Kobaz | 2013-06-20 9:15 AM <Greenlight> Does it show in "core show channels"? Can you hang it up from the CLI ? |
| 14:14.30 | Kobaz | 2013-06-20 9:16 AM <bombev> sometimes yes |
| 14:14.30 | Greenlight | I don't think this is a deadlock |
| 14:14.38 | Kobaz | i think he answered that question well |
| 14:14.40 | Greenlight | To which of the three questions was that the answer |
| 14:15.30 | Greenlight | If it doesn't show in core show channels, then what makes him think it;s stuck |
| 14:15.58 | Greenlight | It sounds like it's not in the rtpstream and just not getting any indication that the call has cleared |
| 14:16.41 | Greenlight | But the vaugness of his answered has you asking him to enable DEBUG_LOCKS |
| 14:17.25 | [TK]D-Fender | bombev: "core show application dial" <--- |
| 14:18.17 | Greenlight | If he can't hangup the call from the CLI then it's something more unusual, and setting a limit on Dial won't help |
| 14:19.02 | Kobaz | he says sometimes |
| 14:19.09 | bombev | the call usaualy is getting stucked while is answered |
| 14:19.23 | Greenlight | "sometimes" what exactly? |
| 14:19.33 | Greenlight | What do you mean by stuck? |
| 14:20.19 | Kobaz | my assumption so far is that the channel sometimes shows in core show channels |
| 14:20.29 | Kobaz | but yeah, getting a further definition of stuck would be helpful |
| 14:20.33 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 14:21.28 | bombev | Greenlight |
| 14:21.34 | Greenlight | bomdev |
| 14:21.35 | bombev | when enter this command sip show channelstats |
| 14:22.19 | bombev | **.142.***.1** 1118162786@ 20:53:58 0000011647 0000000002 ( 0.02%) 0.0000 0000011649 0000000000 ( 0.00%) 0.0007 |
| 14:22.37 | bombev | pay attention on 20:53:58 this is the duration |
| 14:23.02 | Greenlight | Ok, and "core show channels" shows it as well ? |
| 14:23.53 | bombev | yes |
| 14:24.03 | Greenlight | Ok, now "channel request hangup <channel>" |
| 14:24.05 | Greenlight | Does that clear it? |
| 14:24.27 | bombev | sometimes yes, but sometimes not and has to restart asterisk |
| 14:25.00 | Greenlight | What is the channel doing? |
| 14:25.12 | Greenlight | Just inside a DIal ? |
| 14:25.18 | kaldemar | clear what? the channel from "core show channels" or the SIP channel from "sip show channelstats"? |
| 14:27.55 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-xskfwarpuoqhfgid) |
| 14:31.25 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 14:31.30 | *** join/#asterisk flapjacks (~flapjacks@wsip-184-183-148-253.ph.ph.cox.net) |
| 14:31.47 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:33.13 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 14:34.25 | bombev | but what is the reason sometimes to get stucking channels |
| 14:35.00 | Greenlight | ... |
| 14:35.05 | *** join/#asterisk skirge (~skirge@196.15.233.254) |
| 14:39.18 | Kobaz | Greenlight: still sounds like a deadlock |
| 14:40.09 | Kobaz | or it could be a bug in a channel state handling code chunk |
| 14:43.41 | igcewieling | OT http://www.itworld.com/it-management/361516/9-reasons-sys-admins-hate-you |
| 14:45.21 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 14:47.00 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 14:48.26 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:53.33 | Greenlight | It clears from "core show channels", and I can't see Dial deadlocking as we'd hear about it. |
| 14:53.41 | *** join/#asterisk Foxi352 (~Foxi352@v-172-77.access.restena.lu) |
| 14:53.58 | Kobaz | depends what you're doing |
| 14:54.14 | Kobaz | i've ran into enough deadlocks and bugs that i don't rule them out |
| 14:54.19 | Kobaz | especially when channels dont have up |
| 14:54.24 | Kobaz | that's usually a dead giveaway |
| 14:54.54 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:55.09 | Kobaz | especially when the word 'sometimes' comes into the picture |
| 14:55.27 | Kobaz | but that's just me... i tend to be pushing asterisk to the limit of complexity and i break it all the time |
| 14:55.37 | WIMPy | A very very bad word indeed. |
| 14:55.49 | Kobaz | and then i find the bug and submit the patch and then the world is good again |
| 14:56.01 | WIMPy | Oh, So far I thought that was normal. |
| 14:56.39 | *** join/#asterisk dvossel (~dvossel@user-24-214-93-113.knology.net) |
| 14:56.42 | *** part/#asterisk dvossel (~dvossel@user-24-214-93-113.knology.net) |
| 14:56.43 | Kobaz | mm this chedder cheese is really good |
| 14:56.59 | WIMPy | Hmm. When I try to fix something, I often run in to more trouble. Even if it's just bad documentation. |
| 14:57.51 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:57.51 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:58.34 | Kobaz | heh, aww |
| 14:58.42 | Kobaz | i havent done much documentation work |
| 14:58.49 | Kobaz | although i did help leif when chan_local docs |
| 14:58.52 | Kobaz | with |
| 14:59.02 | Kobaz | (still cant get used to this mac keyboard) |
| 15:06.36 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 15:18.28 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 15:21.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:21.18 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:21.32 | leifmadsen | Kobaz: mac pfffft |
| 15:23.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:25.16 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 15:26.08 | eirirs | leifmadsen: +1 |
| 15:26.35 | leifmadsen | of which, I think my macbook pro finally died |
| 15:27.09 | eirirs | leifmadsen: +1 |
| 15:27.46 | leifmadsen | it doesn't seem to spin up at all, doesn't get to the rEFIt screen or anything |
| 15:28.28 | eirirs | I love the fact that you need 20 hours and have a 200 step guide to change a harddisk on a macbook |
| 15:28.43 | eirirs | ok, maybe 2 hours |
| 15:31.03 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 15:31.29 | *** join/#asterisk Minotaur01 (~minotaur0@d72-38-1-47.commercial1.cgocable.net) |
| 15:32.47 | Minotaur01 | o/ morning everyone |
| 15:33.28 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 15:33.32 | leifmadsen | it's not really that hard on the version I have |
| 15:33.37 | leifmadsen | just a lot of little torx screws |
| 15:33.59 | Kobaz | leifmadsen: heh |
| 15:36.43 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:38.47 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 15:38.50 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 15:46.14 | *** join/#asterisk Minotaur01 (~minotaur0@d72-38-1-47.commercial1.cgocable.net) |
| 15:49.09 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:01.51 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 16:01.54 | *** join/#asterisk flapjacks (~flapjacks@wsip-184-183-148-253.ph.ph.cox.net) |
| 16:05.06 | *** join/#asterisk navaismo (~navaismo@189.241.95.6) |
| 16:09.58 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 16:11.33 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 16:12.32 | *** join/#asterisk kresp0 (~kresp0@18.Red-83-42-125.dynamicIP.rima-tde.net) |
| 16:12.54 | Katty | steals file's schnitzle |
| 16:13.07 | file | I could ask Doris to make one for you |
| 16:13.16 | Katty | but then she'd have to eat it. |
| 16:13.17 | file | although since I don't speak German I'd have to use altavista |
| 16:13.22 | Katty | i don't think it would survive the trip. |
| 16:13.22 | file | or rather, babelfish |
| 16:13.29 | Katty | oh, well i can help with that |
| 16:15.00 | lorsungcu | just checked |
| 16:15.05 | lorsungcu | altavista is a thing, still |
| 16:18.41 | *** join/#asterisk movl (~arares@unaffiliated/movl) |
| 16:21.03 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 16:23.49 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 16:24.45 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:28.35 | *** join/#asterisk Minotaur01 (~minotaur0@d72-38-1-47.commercial1.cgocable.net) |
| 16:32.10 | Minotaur01 | Has anyone experianced a problem with the inital voicemail setup prompt always playing everytime you access your voicemail? |
| 16:36.14 | _Corey_ | have you tried changing your password? |
| 16:37.09 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 16:37.16 | Minotaur01 | the voicemail password? |
| 16:38.03 | mmlj4 | that might be a file or directory permissions problem |
| 16:38.08 | _Corey_ | Yeah, it sounds like you have forcegreetings and/or forcename set to yes |
| 16:39.15 | Minotaur01 | mml14j: I've chowned the entire asterisk folder |
| 16:39.43 | Minotaur01 | Corey: Yah that is set to yes |
| 16:40.39 | Minotaur01 | Corey: forcename is set to yes |
| 16:40.52 | Minotaur01 | Will try changing the password |
| 16:42.02 | *** join/#asterisk tilt_ (~tilt@173-13-180-97-sfba.hfc.comcastbusiness.net) |
| 16:43.22 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 16:43.28 | igcewieling | if the password is the same as the mailbox and force* is enabled, then you will be prompted every time you log on. |
| 16:43.41 | igcewieling | I believe this is documented in voicemail.conf.sample |
| 16:44.24 | Minotaur01 | Thank you all that was the problem, I changed the password and its not asking anymore. |
| 16:45.48 | MLNoah | I'm trying to set up DUNDI - SIP for interconnect between multiple Asterisk 1.6.2 boxes, using the examples from Asterisk the Definitive Guide 4th ed. My mapping is: mapping-name => dialcontext,0,SIP,user:secret@10.6.2.220/${NUMBER},nopartial -- but when a system tries to place the call, I get a chan_sip.c warning - No such host: 10.6.2.220/212 |
| 16:46.26 | MLNoah | but even comparing the sample sip.conf dial strings in 1.6.2 and 11, i don't see any difference in the way the samples show. is the example wrong? is it something that upgrading to Ast 11 would fix? |
| 16:58.30 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 16:58.56 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 16:59.32 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 16:59.45 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 17:04.14 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v049-058.mobile.uci.edu) |
| 17:07.50 | igcewieling | MLNoah: almost nobody uses DUNDI, you should check the mailing list archives |
| 17:08.15 | igcewieling | also did you read the UPGRADE-*.txt files? |
| 17:08.18 | MLNoah | ok, thanks |
| 17:09.06 | MLNoah | well, i was more confused, because TDG seems to basically be saying i should be able to dial SIP/user:secret@host/extension, but sip.conf.sample most certainly doesn't, for any version of Asterisk that I checked (1.6.2, 10, 11) |
| 17:12.26 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 17:12.53 | igcewieling | MLNoah: likely "TDG" is wrong. |
| 17:13.11 | MLNoah | but i read it on the internet! it must be true :P |
| 17:16.22 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 17:23.38 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:24.20 | *** join/#asterisk Weezey (~ohno@i.am.weezey.com) |
| 17:33.57 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:39.38 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 17:51.34 | *** join/#asterisk jalewis (~jlewis@solo.atlantic.net) |
| 18:01.07 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 18:03.05 | *** join/#asterisk edge (~edge@74-84-85-54.client.mchsi.com) |
| 18:03.44 | edge | Does anybody know how to end a conference room? Meetme session? We have two Dahdi lines stuck inside the meetme application |
| 18:04.00 | jalewis | anyone have advice re AMD Opteron vs Intel Xeon for Asterisk servers? |
| 18:07.32 | mmlj4 | if you can still get AMD, I'd use that |
| 18:12.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:19.32 | apb1963 | beep beep. I use RoadRunner for my cable service, they also provide me with VOIP phone service. Is there a way to tap into that with asterisk? |
| 18:20.11 | ChannelZ | I doubt they are using anything standard |
| 18:23.18 | [TK]D-Fender | Actually, chances are they are using H.3232 direct on the modem.... |
| 18:27.07 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 18:30.07 | thebmw | mine is done i am now seeding it |
| 18:30.26 | thebmw | sorry wrong channel |
| 18:30.30 | *** part/#asterisk thebmw (~thebmw@i.am.thebmw.net) |
| 18:30.33 | leifmadsen | wow... edge waited a whole.... 5 mins? |
| 18:30.44 | leifmadsen | I would have told him, "channel request hangup" |
| 18:33.20 | apb1963 | [TK]D-Fender: OK, but is there a way to tap into it? |
| 18:33.50 | [TK]D-Fender | apb1963: It's their life-mission to stop you from having any fun.... |
| 18:42.20 | igcewieling | You might as well try stopping ninjas. |
| 18:44.14 | apb1963 | I don't wanna have fun... I want to use the service I already pay for, to make outgoing calls without using my hardphone. |
| 18:45.49 | apb1963 | I not only have ninja insurance, I have ninja defense shields. And I know they work because I've never, ever had ninjas get through. |
| 18:47.11 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:49.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.183) |
| 18:49.36 | *** join/#asterisk przerull (~philip@philzerull.com) |
| 18:50.24 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 18:50.45 | przerull | hello, so I use ami to spool an outbound call to a sip peer. Is it possible to get the sip callid for that call? |
| 18:51.59 | jalewis | mmlj4: Dell still sells several Opteron-based servers |
| 18:52.02 | przerull | I'm particularly interested in the case when that call fails to connect due to telephony reasons (busy, congested, etc). I'm pulling ${REASON} but would like to get the sip callid |
| 18:52.15 | igcewieling | przerull: use a Local/ channel and handle it in the dialplan |
| 18:52.32 | igcewieling | (which is the answer to almost all spool related questions) |
| 18:53.46 | przerull | igcewieling: so would the ami command spool the call to the local channel which would then dial out using the dial command? |
| 18:54.18 | killown | could anyone help me, here is the debug http://paste.ubuntu.com/5784602/ my voip gets busy when someone try to call it after some minutes and it still keeps registered |
| 18:54.41 | przerull | igcewieling: thanks. that's a huge help |
| 18:54.45 | igcewieling | przerull: something like that. |
| 18:56.49 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 18:57.43 | killown | please? |
| 18:57.55 | *** join/#asterisk movl (~arares@unaffiliated/movl) |
| 18:58.05 | [TK]D-Fender | killown: Because they are almost certainly sending you UNAUTHED calls |
| 18:58.16 | navaismo | killown, No matching peer for '1122050075' from '201.86.87.21:5060' SIP/2.0 403 Forbidden |
| 18:58.18 | [TK]D-Fender | killown: And your peer is attempting to auth them. |
| 18:58.45 | killown | [TK]D-Fender, navaismo it works fine directly in a soft phone :( |
| 18:59.06 | killown | I am using these settings https://www.falevono.com.br/asterisk/ |
| 19:01.42 | [TK]D-Fender | killown: "insecure=port,invite" <--------------- |
| 19:01.56 | [TK]D-Fender | killown: Their guide is ancient crap |
| 19:02.02 | killown | [TK]D-Fender, ok |
| 19:02.14 | [TK]D-Fender | killown: And should be "type=peer" |
| 19:03.00 | killown | [TK]D-Fender, [ramal123] type=peer? |
| 19:03.15 | [TK]D-Fender | killown: whatever the name happens to be... |
| 19:03.28 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-025-221.pools.arcor-ip.net) |
| 19:04.30 | killown | [TK]D-Fender, worked, thanks a lot, like you said, their guide is ancient crap indeed |
| 19:16.25 | killown | [TK]D-Fender, ] same problem again http://paste.ubuntu.com/5784686/ |
| 19:16.44 | *** join/#asterisk chazzam (~cmoye@nat/digium/x-dguzohlcmrvnizuj) |
| 19:17.02 | [TK]D-Fender | show us your actual peer masking only the secret |
| 19:17.53 | killown | [TK]D-Fender, peer masking? |
| 19:18.14 | [TK]D-Fender | [15:17][TK]D-Fendershow us your actual peer, masking only the secret |
| 19:18.30 | killown | oh ok |
| 19:19.37 | killown | [TK]D-Fender, http://paste.ubuntu.com/5784697/ |
| 19:20.41 | [TK]D-Fender | killown: host=vono.net.br <- verify the IP that resolves to. Wouldn't seem to match the IP they are sending from... |
| 19:21.10 | [TK]D-Fender | set the IP direct if you have to |
| 19:21.34 | killown | 64 bytes from 201.86.87.35.sbc.static.gvt.net.br (201.86.87.35): icmp_seq=6 ttl=54 time=1572 ms << more than 1 second 0.o |
| 19:21.47 | killown | ok, I will set it directly |
| 19:24.44 | killown | [TK]D-Fender, I am using the ip now, for a while it's working |
| 19:38.40 | *** join/#asterisk sarsaeol (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 19:39.51 | pabelanger | Anybody still using an AA50 from digium? |
| 19:41.48 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:50.13 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:52.51 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 19:58.48 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.166) |
| 20:02.53 | *** join/#asterisk capnkooc (~capnkooc@ns3.tribuddb.com) |
| 20:08.26 | *** join/#asterisk dlynes (~dlynes@bas9-hamilton14-3096722323.dsl.bell.ca) |
| 20:22.09 | *** part/#asterisk capnkooc (~capnkooc@ns3.tribuddb.com) |
| 20:32.50 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:34.14 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:46.22 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 20:52.23 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tasonpvmqatnbbuc) |
| 21:14.09 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 21:14.57 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 21:21.19 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 21:27.36 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 21:29.23 | *** join/#asterisk ani216 (~ani216@142.197.233.126) |
| 21:29.27 | ani216 | Hello everyone |
| 21:30.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.149) |
| 21:30.44 | ani216 | I have a question, last night at about 3:30am my phones started ringing with a call from extension 1000. I dont have an ext 1000 and when i picked up no one was there, and i was left with 39 empty voicemails...anyone have any idea as to what this was? |
| 21:32.34 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 21:40.50 | drmessano | Yep, SIP attack |
| 21:42.02 | drmessano | You can disallow guest or block the offenders IP address |
| 21:42.40 | WIMPy | Or configure youre extensions correctly. |
| 21:42.57 | drmessano | that too |
| 21:44.45 | drmessano | Some people have a fascination with having a "catch all" for <insert protocol or service>. If something is coming to my domain and a valid user isn't specified, return to sender |
| 21:54.11 | SuperNull | im getting this over and over: WARNING[26982]: chan_sip.c:4210 __sip_autodestruct: ..... |
| 21:54.16 | SuperNull | should i be pooping my self |
| 22:00.06 | *** part/#asterisk sarsaeol (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 22:08.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:12.09 | *** join/#asterisk blixt0 (none@177.43.29.231.dynamic.adsl.gvt.net.br) |
| 22:17.40 | *** join/#asterisk Synx|hm_ (~Synx@unaffiliated/synx-hm/x-1623004) |
| 22:18.12 | *** join/#asterisk beanie (beanie@94.14.66.167) |
| 22:18.54 | beanie | hullo, just wondering what time to live settings need to be in dyndns if your setting up extension access of site |
| 22:19.15 | beanie | im intending on setting up my router to connect with dyn dns |
| 22:20.13 | beanie | off* |
| 22:20.28 | navaismo | #asterisk try with dydns support or Google is your friend |
| 22:22.07 | beanie | hello navaismo again hope your well :-) |
| 22:22.28 | beanie | i think because it needs to be compatible with asterisk they may point me back here |
| 22:22.42 | [TK]D-Fender | no |
| 22:22.47 | [TK]D-Fender | It has nothing to do with Asterisk |
| 22:22.49 | [TK]D-Fender | .... |
| 22:22.52 | [TK]D-Fender | or FreePBX. |
| 22:22.59 | [TK]D-Fender | that cvlient does its own job |
| 22:23.24 | [TK]D-Fender | you set the REFRESH rate in Asterisk SIP settings as to how often * should VERIFY it. |
| 22:24.08 | beanie | ah i see :-) are you familiar with time to time to live - i don't want asterisk to go offline and be unreliable :-) |
| 22:24.21 | beanie | for long periods |
| 22:25.12 | [TK]D-Fender | So set it for something reasonable. |
| 22:26.55 | beanie | what's reasonable this is the thing!! |
| 22:27.12 | [TK]D-Fender | This ain't Raw-Cat Sigh Hence |
| 22:27.22 | [TK]D-Fender | Is HOURLY good enough for you? |
| 22:27.35 | [TK]D-Fender | Or are you going to check every five seconds? |
| 22:27.58 | beanie | ah i'm just a lost soul my friend - i just want to make sure i don't cock up here |
| 22:28.26 | beanie | it's no good having a decent pbx if it goes down all the time |
| 22:28.42 | [TK]D-Fender | How about getting a fixed IP then? |
| 22:28.52 | beanie | can't afford it |
| 22:28.59 | beanie | i do charitable work, don't get paid for it |
| 22:29.21 | beanie | have a small network of homeworking volunteers |
| 22:33.03 | navaismo | if I have a dollar for every "charitable work" |
| 22:33.25 | beanie | navaismo - that's a bit harsh |
| 22:33.41 | beanie | imagine, i'm totally genuine, trying to do a decent job in homeless outreach |
| 22:34.00 | beanie | need to train volunteers, keep in touch with them, liaise with local authorities and i don't get a penny piece for doing it :-) |
| 22:34.33 | beanie | if I had money, don't you think i'd just pay someone to do it all for me? |
| 22:34.36 | [TK]D-Fender | heads out for a while.... |
| 22:34.36 | navaismo | Good karma points |
| 22:35.14 | beanie | sure, but here I am now, working on this, no money for a static ip and so i need to work with dyn dns and just not sure how to set this time to live properly |
| 22:35.53 | navaismo | tranlating [TK]D-Fender--->I'ts up to you |
| 22:36.11 | beanie | sure, does it not matter in the slightest for the purposes of asterisk? |
| 22:36.19 | beanie | or could it seriously increase downtime if i get it wrong? |
| 22:36.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.178) |
| 22:38.26 | navaismo | my advise for you is to use a dyndns client on your pbx |
| 22:39.31 | apb1963 | inadyn works for me |
| 22:40.38 | navaismo | ^ |
| 22:40.52 | apb1963 | and for that matter... ani216: Get thyself a whitelist! |
| 22:41.04 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 22:41.05 | saint_ | hi all |
| 22:41.15 | apb1963 | starts to pray |
| 22:41.36 | saint_ | If I want to use the 2nd line button on a D40 to be a "speed dial" to another extension, how do I do that ..? |
| 22:44.03 | beanie | whats the benefits of using a dyndns client on my pbx as opposed to using dyndns connected to the router |
| 22:44.39 | ShaunR | I've got a bunch of polycom 550's, I was just going to upgrade the firmware on one of them to the latest which appears to be 4.1.0 and started reading somthing about having to purchase licensing for lync? |
| 22:44.58 | ShaunR | whats this all about? Can i not upgrade the phone without having to purchase some type of new license? |
| 22:51.32 | ShaunR | ah, it looks like lync is for some microsoft stuff |
| 22:51.57 | beanie | ok, got a dyndns which file do i need to edit - is it sip_nat.conf |
| 22:52.55 | navaismo | in the GUI advanced sip settings |
| 22:54.58 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 22:59.58 | beanie | so i have hostname, primary dns, secondary dns tertiary dns, dns search path? |
| 23:00.51 | beanie | this is in dns configuration within the gui |
| 23:02.30 | navaismo | asterisk sip settings<----correct menu |
| 23:05.13 | beanie | that won't run |
| 23:05.18 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
| 23:05.18 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
| 23:05.58 | navaismo | uh? thats a menu in the GUI |
| 23:06.12 | navaismo | you should start to see the freepbx wiki |
| 23:08.39 | beanie | so i type setup |
| 23:08.43 | beanie | it opens the gui right |
| 23:09.39 | navaismo | seriously why you are obsessed with that setup command |
| 23:09.58 | beanie | well whats the difference - you have told me a few different ones and it comes up with same gui |
| 23:10.02 | navaismo | You are using FreePBX use the GUI, WEB GUI |
| 23:10.06 | igcewieling | navaismo: happens when you are in the wrong channel. Something about the mind not being to cope with the space-time rupture. |
| 23:10.24 | navaismo | igcewieling, i need to learn to use your favorite irc command |
| 23:10.29 | beanie | how enlightening |
| 23:10.35 | igcewieling | navaismo: indeed |
| 23:10.46 | beanie | so what am I doing here? what am I typing? where am i going :-) |
| 23:11.02 | navaismo | go to freepbx wiki and read read alot |
| 23:11.36 | beanie | i did, i typed sip settings into search and it said there was no direct link, i looked through the other relevant articles and couldn't find anything |
| 23:12.19 | beanie | so where do I need to be in this command prompt |
| 23:13.09 | beanie | as far as i know, so far, the gui is that setup |
| 23:13.12 | navaismo | 1) When you talk about asterisk forget command line, since you are using a GUI 2) ask this question in the # freepbx channel 3) learn to differ linux stuff from asterisk stuff |
| 23:13.33 | navaismo | since morning you are mixing linux with asterisk stuff |
| 23:13.37 | navaismo | asterisk is not linux |
| 23:13.46 | navaismo | is only a program |
| 23:13.55 | beanie | no but centos, dedicated to asterisk seems to be totally intertwined |
| 23:14.28 | beanie | so when you talk about the gui, do you mean the free pbx control panel accessible via the ip address? |
| 23:14.52 | igcewieling | must. resist. |
| 23:14.58 | navaismo | We now you are a newbie but that doesn't mean that you want to fly before walk, so my advise for you is, Learn Linux, read book and use google. Then learn asterisk, read the online guide(bible) and then try to setup your pbx |
| 23:16.09 | beanie | I don't have an issue with doing that but ultimately 1) the information i'm reading does not make a lot of sense to me 2) you have twice today sent me to links that don't actually answer the question |
| 23:16.16 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 23:16.16 | navaismo | for example i dont know how to fix a car, so you dont see me offering services to fix cars, first i need to know how to drive it |
| 23:16.45 | beanie | so, when you talk about the gui, are you talking about the free pbx control panel? |
| 23:17.18 | navaismo | http://wiki.freepbx.org/display/F2/Asterisk+SIP+Settings |
| 23:17.47 | navaismo | must resist to use the F in the sentence READ THE MANUAL |
| 23:18.30 | saint_ | does someone know where I can find a list of high charged numbers or prefix to block in the USA ? Like porn, etc etc.. |
| 23:19.06 | navaismo | beanie, download and read the pdf and as you can see the link WORK |
| 23:23.01 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 23:26.29 | beanie | does local networks need to be completed when i have set a dynamic ip? |
| 23:26.46 | beanie | i hit auto configure and it came back with 192.168.0.0 |
| 23:28.45 | beanie | and i must resist using the F in the sentence I have read the manual, I don't understand it very well and so I came to ask for some support :-) |
| 23:29.24 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:29.58 | flapjacks | your asterisk box should have a static IP that is outside of the range of your local DHCP server |
| 23:30.25 | beanie | ah yes - the one i log in to freepbx with :-) |
| 23:30.40 | beanie | (i.e. the gui) |
| 23:30.46 | beanie | ? |
| 23:32.48 | navaismo | igcewieling, im like a stalker do you have twitter? |
| 23:33.16 | igcewieling | navaismo: I'm not on any social networking sites |
| 23:33.22 | navaismo | ¬¬ |
| 23:35.44 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.138) |
| 23:36.24 | b0ot | Is there a way not to pass dtmf tones on specific trunks? |
| 23:41.30 | beanie | not getting anywhere with this local networks setting? |
| 23:41.39 | beanie | and, before you ask, i have looked on google |
| 23:46.20 | navaismo | http://literature.schmoozecom.com/asterisk_sip_settings-module/userguides/asterisk_sip_settings-module-userguide.pdf |
| 23:46.38 | navaismo | read the local networks section |
| 23:47.00 | beanie | ok :-) |
| 23:48.56 | beanie | is what i am trying to do effectively a vpn? |
| 23:49.00 | beanie | i did read this before |
| 23:49.05 | beanie | and then came back here |
| 23:49.25 | navaismo | what? uhhh no in anyway you are building a vpn in that web page |
| 23:49.52 | navaismo | you are only configuring the nat settings for asterisk |
| 23:50.07 | beanie | so i need to add a local network |
| 23:50.13 | beanie | but what is the ip? |
| 23:50.20 | beanie | the same one as i log in to freepbx with? |
| 23:50.40 | navaismo | facepalms |
| 23:51.00 | beanie | well it would help if your last comment made sense :-p Beanie puts his head in his palms to |
| 23:51.06 | navaismo | your Local-network!!! |
| 23:51.22 | navaismo | i.e 192.168.0.0/24 |
| 23:51.34 | navaismo | the pdf say it very clear |
| 23:51.37 | beanie | you could just answer "yes" or "no" - is this the same address as i use to log in to the free pbx from my web browder |
| 23:51.44 | beanie | browser |
| 23:52.07 | navaismo | nothing to do here....shhhhhhhh |
| 23:52.25 | beanie | do you actually know? |
| 23:52.31 | b0ot | So I have this device that sends * and # when talking. When I call from that device to a softphone such as SJPhone it works fine, however when I call to my Yealink T-26P it plays the * and #s which is very annoying (although audio from call still works) |
| 23:52.40 | b0ot | Any ideas on how I could fix this problem |
| 23:52.48 | b0ot | Phone setting, trunk setting? |
| 23:53.04 | navaismo | my face right now ----> http://24.media.tumblr.com/71e41bcad13d02c1b12b1bb6267e3659/tumblr_mhxze13HR81s5nruoo1_500.jpg |
| 23:53.09 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 23:53.14 | TJNII | Why on earth is is sending # and * on its own? |
| 23:53.29 | b0ot | It especially annoying since the phone sends a constant # until the person on that end talks |
| 23:53.38 | b0ot | It is meant to integrate into systems that are half duplex |
| 23:53.51 | b0ot | which use * and # to signal when to talk and when to listen |
| 23:53.56 | TJNII | Aah |
| 23:54.01 | navaismo | no beanie i don't know any shit about asterisk or freepbx im just a troll what pretend to help people but actually I only mess around the info |
| 23:55.23 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip68-104-246-231.ph.ph.cox.net) |
| 23:57.42 | b0ot | changed dtmf type to INBAND |
| 23:57.45 | b0ot | fixed the issue |
| 23:59.19 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
