| 00:00.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:22.32 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 00:22.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:26.38 | *** join/#asterisk Skten (~Sktenau@monitor.ic.ipera.net.au) |
| 00:28.15 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 00:50.09 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 00:53.36 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
| 01:06.20 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 01:14.38 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 01:34.27 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 01:35.19 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:7503:8ee3:6401:ad53) |
| 01:43.07 | *** join/#asterisk apb1963 (~apb@107-146-213-112.res.bhn.net) |
| 01:43.43 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 01:48.44 | apb1963 | Hello! How can I get * 1.8 to stop displaying manager connects in the CLI? |
| 01:49.41 | WIMPy | Decrease verbose level |
| 01:51.53 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:55.05 | *** part/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 02:10.54 | apb1963 | WIMPy: to what level please? |
| 02:11.18 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 02:18.15 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 02:18.46 | apb1963 | That's very interesting, I thought you were going to tell me to set "displayconnects = no" in manager.conf - but I guess you knew that doesn't work. I had to set verbosity to 1 to make it stop. I wonder what other info I will no longer get at verbose level 1. |
| 02:19.15 | apb1963 | WIMPy: |
| 02:19.17 | apb1963 | thank you |
| 02:26.45 | *** join/#asterisk raspberrypifan (~raspberry@181.113.111.232) |
| 02:30.34 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 02:32.46 | Penguin | Why wouldn't the display connects setting work? I'm sure it serves a purpose. |
| 02:37.33 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 02:54.56 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 02:54.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:18.11 | ChannelZ | It apparently doesn't work so good as a general option. |
| 03:18.28 | ChannelZ | Define it for your manager users and it does |
| 03:37.03 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 03:45.12 | apb1963 | Penguin: I don't know... bug? |
| 03:46.05 | apb1963 | ChannelZ: Yeah, I thought of trying that.... but just assumed that if it didn't work in general it wouldn't work for individual users... but I'll try it now that you say it works. Thank you |
| 03:53.16 | apb1963 | Yep... that works. Brilliant! Thank you! |
| 04:13.38 | ChannelZ | sure |
| 04:19.44 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 04:32.02 | *** join/#asterisk jetlag (~jetlag@pool-71-168-194-254.cmdnnj.east.verizon.net) |
| 04:40.51 | ChannelZ | WTF |
| 04:41.08 | ChannelZ | I'm VNC'd into a box at work and when I type < into a console, it types > |
| 04:59.57 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:13.20 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:16.11 | *** join/#asterisk hos7ein (~chatzilla@94.101.189.1) |
| 05:30.13 | apb1963 | What happens when you type > ? Maybe your keyboard got remapped somehow? Try restarting the connection? I don't use VNC so, no experience with it. |
| 05:36.09 | *** join/#asterisk cunningpike (~cunningpi@2602:304:6eaf:6280:9917:8482:e9d4:66d9) |
| 05:42.51 | *** join/#asterisk timahvo1 (~rogue@mail.cickenya.com) |
| 05:49.51 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:618c:ba1b:3804:4cce) |
| 06:15.10 | *** join/#asterisk joesuffceren (446aed33@gateway/web/freenode/ip.68.106.237.51) |
| 06:15.27 | joesuffceren | anyone had any luck getting Noojee click to work with recent versions of Firefox or Chrome? I can't get it to work with either browser and either astmanproxy or AJAM, but the plugins are not outputing anything to the console as they are supposed to |
| 06:15.39 | joesuffceren | so it's tough to debug with no logs. :-/ |
| 06:17.02 | *** join/#asterisk Takapa (vegard@junior.svanberg.no) |
| 06:26.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 06:28.24 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 06:57.01 | *** join/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag) |
| 06:59.50 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:01.34 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:68a3:c39d:8b7b:210d) |
| 07:13.25 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:14.49 | *** join/#asterisk LiohAu (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 07:14.51 | *** join/#asterisk W0rmDr1nk (~wormdrink@unaffiliated/wormdrink) |
| 07:21.27 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 07:40.16 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 07:50.19 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:03.50 | *** join/#asterisk shadej (~bzegeju@213.55.73.146) |
| 08:04.37 | shadej | I have a problem with connecting to asterisk using sip. when I run sudo netstat -nltp| grep 5060 on my asterisk CLI |
| 08:04.46 | shadej | I dont get anything. |
| 08:05.03 | shadej | how can I check if SIP is running on asterisk server |
| 08:11.39 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:12.48 | *** join/#asterisk madax (~madax@195.158.110.216) |
| 08:18.37 | shadej | when I execute the command console dial 200 |
| 08:18.49 | shadej | I am getting this error:[Sep 23 11:18:01] ERROR[22620][C-00000003]: app_stack.c:569 gosub_exec: Gosub address is invalid: '200,stdexten(SIP/200&IAX2/200)' |
| 08:21.20 | shadej | anyone? |
| 08:23.06 | zpotoloom | shadej: netstat -ntlpu |
| 08:24.09 | shadej | zpotoloom: okay,i got this. udp 0 0 10.0.20.207:5060 0.0.0.0:* 3948/sflphone |
| 08:24.22 | shadej | but that is for my sip client (SFLphone) |
| 08:24.43 | zpotoloom | you have to change th sflphone port or asterisk port, can't use same port for both |
| 08:27.11 | shadej | this is not my first time to use the sip client, it was working with same config. |
| 08:27.50 | shadej | the sflphone is trying to connect to that port it is not using it |
| 08:29.09 | zpotoloom | "SFLphone creates a default SIP UDP transport at startup on port 5060. If you want to change this transport's settings, you need to go in Edit > Preference > Ip Calls and then restart SFLphone." |
| 08:29.36 | zpotoloom | if you are running asterisk on the same machine then it fails to bind to UDP 5060 cause sflphone is already using that port |
| 08:32.06 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:32.23 | *** join/#asterisk areski (~areski@80.174.128.93.dyn.user.ono.com) |
| 08:33.56 | shadej | zpotoloom: solved! thank you very much you saved my life. |
| 08:34.23 | shadej | the problem was I was staring asterisk always that I can notice sflphone is using same port with asterisk. |
| 08:34.41 | shadej | *starting asterisk first |
| 08:40.37 | *** join/#asterisk NoobSaibot (~NoobSaibo@CPE-24-208-40-235.new.res.rr.com) |
| 09:03.37 | *** join/#asterisk hurdman (~ygcheny@c3p0.r0b0t.fr) |
| 09:04.31 | hurdman | hello all, i try too use repro ( using this tuto http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch ), to get a great early video media with sip |
| 09:05.37 | hurdman | if my call is directly send to repro, i get the early video media, if i call the proxy that route the call to asterisk and back to proxy, i doesn't get the early media. |
| 09:05.45 | hurdman | any idea of what i miss ? |
| 09:12.52 | *** join/#asterisk CeBe (~CeBe@p5488BD81.dip0.t-ipconnect.de) |
| 09:15.47 | skrusty | anyone here using pjsip? |
| 09:33.51 | hurdman | skrusty: what's your question ? |
| 09:34.25 | skrusty | not so much a problem, jsut trying to figure out if you can specify an interface to bind to an endpoint |
| 09:34.45 | skrusty | i.e. i want endpoint a to go out via eth0 and endpoint b via eth1 |
| 09:38.34 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 09:39.52 | hurdman | can't help you, never try, sorry :/ |
| 09:40.15 | skrusty | me either :) |
| 09:53.50 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 09:55.46 | *** join/#asterisk Skten (~Sktenau@ppp59-167-129-107.static.internode.on.net) |
| 10:09.29 | *** join/#asterisk aurs (~aurs@187.84-48-58.nextgentel.com) |
| 10:23.04 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 10:24.59 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 10:25.02 | wasanzy | hello |
| 10:25.45 | wasanzy | am getting the error: SIP/2.0 404 Not Found for incoming calls to my asterisk, what do I do? |
| 10:38.15 | *** join/#asterisk calum_ (~calum_@79-75-70-136.dynamic.dsl.as9105.com) |
| 10:41.42 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 10:42.17 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 10:47.59 | hurdman | wasanzy: full sip set debug on trace ? |
| 10:50.04 | *** join/#asterisk areski (~areski@130.98.135.37.dynamic.jazztel.es) |
| 10:50.17 | *** join/#asterisk Skten (~Sktenau@monitor.ic.ipera.net.au) |
| 10:52.25 | *** part/#asterisk hurdman (~ygcheny@c3p0.r0b0t.fr) |
| 11:01.42 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 11:05.11 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 11:06.09 | *** join/#asterisk hecatae (~hecatae@host-92-28-0-190.as13285.net) |
| 11:11.26 | wasanzy | hurdman: on the shell? |
| 11:11.38 | wasanzy | Yes I had sip debug on |
| 11:12.51 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 11:16.08 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:17.19 | wasanzy | http://pastebin.com/tdKKKDzq here is my config. when the user dial the extension from a cell phone, he get number does not exist. is there something wrong with my configs? |
| 11:36.04 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 11:52.46 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 11:54.52 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 11:58.14 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 12:00.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 12:12.24 | BakaKuna | wasanzy: How do you call the number? i.e. softclient connected to asterisk directly? |
| 12:16.06 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 12:19.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:28.54 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:30.36 | marceloamorim | morning, I am receiving a warning about file.c:830 ast_readaudio_callback: Failed to write frame. Am I doing something wrong? |
| 12:33.43 | wasanzy | BakaKuna: issue resolved thank you. |
| 12:35.32 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:68a3:c39d:8b7b:210d) |
| 12:36.35 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 12:41.35 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 12:42.45 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:05.33 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-jviellipuildczfq) |
| 13:05.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:21.10 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:33.55 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-vewhtolphshktpfc) |
| 13:33.56 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:48.33 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
| 14:11.01 | marceloamorim | guys, could I set label on special extension "h"? |
| 14:11.18 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 14:11.29 | [TK]D-Fender | yes |
| 14:11.38 | [TK]D-Fender | just like any other extension |
| 14:11.57 | [TK]D-Fender | You don't technically set it on an extension... you set it on a PRIORITY |
| 14:12.45 | marceloamorim | yeah, it works |
| 14:12.48 | marceloamorim | ty [TK]D-Fender |
| 14:21.51 | sip | woot |
| 14:27.40 | *** join/#asterisk raspberrypifan (~raspberry@181.113.111.232) |
| 14:28.33 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 14:29.10 | marceloamorim | guys I was testing here about the special extension "h" and if I have includes in my context and have "h" in one of them, any hangup that I have in my calls goto there, so I couldn`t have more than one special h extension |
| 14:29.22 | mjordan | hangup handlers |
| 14:29.36 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 14:30.00 | marceloamorim | in this situation should I program the dialplain without includes and more goto to another contexts? |
| 14:30.24 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:30.27 | marceloamorim | mjordan: I`m using the h extension because I`m trying to build my own cdr |
| 14:30.33 | mjordan | you can only have one 'h' extension per context. Includes are included within a context, so they fall into that boat. Alternatively, you can attach hangup handlers to a channel. You can have any number of those. They are executed when a channel hangs up in the same way that the 'h' extension is executed. |
| 14:35.47 | marceloamorim | let me try this,thx mjordan |
| 14:38.28 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-hflgahnceakkgbgj) |
| 14:38.28 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:42.46 | *** join/#asterisk CeBe (~CeBe@p5488BD81.dip0.t-ipconnect.de) |
| 14:46.18 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:46.33 | *** join/#asterisk catphish (~catphish@unaffiliated/catphish) |
| 14:47.32 | catphish | can someone briefly explain how 2-way audio is established when a sip client is behind a NAT? is it that asterisk force the audio in both directions to use the same UDP port? |
| 14:48.16 | file | many NATs will allow return traffic to occur if it is sent to the source IP address/port of traffic received from it |
| 14:48.47 | file | so provided Asterisk can receive the media from it, it can know the source IP address/port and switch where it is sending |
| 14:50.02 | catphish | but how does the client know to receive RTP on the same port it sends it? |
| 14:50.26 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-uqeteeyhhoqhvybt) |
| 14:50.50 | catphish | is that always the case, or does asterisk somehow force that to be the case? |
| 14:51.09 | file | everything I've come across uses the same port for sending as receiving |
| 14:51.20 | file | and Asterisk can't really force anything, it's all the client |
| 14:51.45 | catphish | ok, i guess that's how it works then, the clients all receive on the same port they send, so as long as the client sends first, asterisk can send back |
| 14:52.11 | catphish | just found this: http://www.voip-info.org/wiki/view/RTP+Symmetric |
| 14:52.44 | *** join/#asterisk cunningpike (~cunningpi@70-234-246-40.lightspeed.gdrpmi.sbcglobal.net) |
| 14:53.54 | *** join/#asterisk rmudgett (~rmudgett@nat/digium/x-wfrssypvztkzzuzd) |
| 14:59.55 | *** join/#asterisk retentiveboy (~retentive@2001:558:6011:78:5cc0:7fe:ddea:6cbf) |
| 15:06.20 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 15:07.38 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:09.36 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:18.04 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 15:35.53 | marceloamorim | nice mjordan, it works pretty well |
| 15:36.06 | *** join/#asterisk aross42 (~aross@CPE000e84807d02-CMbcc8100f7698.cpe.net.cable.rogers.com) |
| 15:36.30 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 15:51.28 | *** join/#asterisk tbsky (~tbsky@dns.annsky.com) |
| 15:59.06 | tbsky | hi, is that possible to change the codec when dial from SIP to DAHDI? I tried to use SIP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP codec change after dahdi answered the channel. so everything is broken. |
| 15:59.25 | tbsky | I got DAHDI/SIP legs final like this: |
| 15:59.40 | tbsky | <PROTECTED> |
| 15:59.40 | tbsky | <PROTECTED> |
| 15:59.40 | tbsky | <PROTECTED> |
| 15:59.40 | tbsky | <PROTECTED> |
| 15:59.40 | tbsky | <PROTECTED> |
| 16:00.29 | tbsky | although two legs finally use alaw both, but trascode in the middle.. is that fixeable? |
| 16:05.17 | tbsky | if that is not possible, then maybe I should give up using G722 as the preffered codec of ip phone. back to G711 seems much easier to make all legs with the same codec.. |
| 16:16.48 | *** join/#asterisk Stefan27 (~Stefan27@212.247.4.149) |
| 16:17.58 | Stefan27 | happy tuesday everyone |
| 16:18.48 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-chjfbxazqqryhxyz) |
| 16:21.55 | *** join/#asterisk supersoaker (~supersoak@users.757.org) |
| 16:22.26 | supersoaker | Hello asterisk team, I have an interesing bug in my dial plan, I was hoping for advice |
| 16:23.36 | supersoaker | I have a dial function that uses a GoSub. (U option) and the gosub my jump to a part of code where it goes to voicemail |
| 16:24.06 | supersoaker | works great, except if the phone is forwarded (a 301 is sent) |
| 16:24.17 | Qwell | Are you requiring people to leave a VM before even dialing a device? That's...special. |
| 16:24.26 | supersoaker | no? |
| 16:24.48 | supersoaker | there internal menu has an option to redirect to voicemail |
| 16:24.55 | *** join/#asterisk aross42 (~aross@CPE000e84807d02-CMbcc8100f7698.cpe.net.cable.rogers.com) |
| 16:25.14 | supersoaker | so it dials the devices and plays back options to the callee and they can redirect them |
| 16:28.44 | newtonr | supersoaker, how does it fail? can you pastebin an Asterisk log showing a working call and a failing one? |
| 17:02.52 | *** join/#asterisk mirela666 (~mirko.bra@95.180.126.160) |
| 17:06.35 | supersoaker | newtonr: Sure, do you need the dial plan as well? |
| 17:23.42 | supersoaker | newtonr: i gave you a link in pm, thanks, Ill get the non working log together |
| 17:26.14 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 17:36.34 | *** join/#asterisk Brixius (~Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 17:40.03 | marceloamorim | there is any thread about file.c: 830 ast_readaudio_callback when we use hangup_handlers? |
| 17:40.47 | newtonr | supersoaker, post the pastebins in here, i'm not sure when I'll get a chance to look at it |
| 17:41.17 | newtonr | supersoaker, yeah you should pastebin the dialplan as well. |
| 17:41.52 | *** join/#asterisk Brixius (~Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 17:45.28 | supersoaker | newtonr: the dial plan is kinda scary |
| 17:45.45 | supersoaker | :) |
| 17:45.59 | supersoaker | thanks newtonr ill repost them in chat |
| 17:46.49 | marceloamorim | I`m getting a lot of warning when I use extension "h" or hangup-handlers |
| 17:48.35 | *** join/#asterisk sdjernes (~sdjernes@75-135-54-221.dhcp.krny.ne.charter.com) |
| 17:49.34 | marceloamorim | Failed to write frame, every Playback I used now has a warning about to write frame |
| 17:50.18 | WIMPy | Well, you obviouselt can't play sounds to channels that no longer exist. |
| 17:51.03 | WIMPy | Neither SIP nor Asterisk know the difference between a disconnect and a release. |
| 17:52.07 | *** join/#asterisk riess82 (~riessma@188-22-233-163.adsl.highway.telekom.at) |
| 17:52.32 | marceloamorim | the audios working well, the problem is the warnings |
| 17:53.37 | WIMPy | Sounds like you're doing it to both channels. |
| 17:54.00 | sdjernes | marceloamorim: That warning is annoying but can be ignored. It usually means the other end hung up before you finished playing audio |
| 17:54.10 | WIMPy | Look at the channel names in your log. |
| 17:56.35 | sdjernes | marceloamorim: If you like me play something in your h extension. You could check the channel status first with a GoToIf |
| 17:57.13 | *** join/#asterisk areski (~areski@80.174.128.64.dyn.user.ono.com) |
| 17:57.48 | marceloamorim | let me check |
| 17:58.05 | marceloamorim | thx |
| 18:12.28 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 18:13.36 | *** join/#asterisk sdjernes (~sdjernes@75-135-54-221.dhcp.krny.ne.charter.com) |
| 18:15.12 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:15.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:30.27 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 18:43.07 | *** join/#asterisk ibercom (d562dfd6@gateway/web/freenode/ip.213.98.223.214) |
| 18:47.10 | *** join/#asterisk brezular (~brezular@bband-dyn132.178-41-33.t-com.sk) |
| 18:51.13 | *** join/#asterisk sdjernes (~sdjernes@75-135-54-221.dhcp.krny.ne.charter.com) |
| 18:54.37 | *** join/#asterisk bananapie (~pumpkinpi@207.134.195.126) |
| 18:55.17 | bananapie | Is there an option to see what exactly is consuming CPU in asterisk ? I am trying to bring down CPU and am wondering if there is a 'top' style command to tell me what asterisk is doing. |
| 18:55.39 | bananapie | I unloaded all the codec modules on the server and retested all the menus without the codecs loaded to make sure the server has the right file types already. |
| 18:57.04 | Qwell | bananapie: not really |
| 18:57.27 | bananapie | ok |
| 18:57.28 | bananapie | thanks |
| 18:57.36 | Qwell | You could profile it, but unless you're already really familiar with the Asterisk source, you aren't going to get far. |
| 18:58.02 | sdjernes | bananapie: Do you have a high call load |
| 18:58.36 | bananapie | No. I only have about 90 calls, but the CPU is at 40%. |
| 18:58.50 | bananapie | load average at 0.6 :| |
| 18:59.06 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:59.28 | bananapie | I think MixMonitor is killing performance, I passed a file name ending in wav, so suspect that asterisk is using CPU to merge the channels and transcode into a wave file. |
| 19:00.10 | *** join/#asterisk dan_j (~IceChat9@unaffiliated/danfromuk) |
| 19:01.01 | dan_j | Hi. Is it possible to find out the device_state of a virtual device from the CLI? |
| 19:01.28 | dan_j | Ive forgotten what the type of variable is called. Its the one that allows the control of hints using a variable. |
| 19:01.50 | sdjernes | Hmm 90 calls all using MixMonitor that can be a but piggy since I am pretty sure that it does transcode on write |
| 19:02.16 | dan_j | Ah, never mind. I found it. |
| 19:02.20 | Qwell | yes, MixMonitor is a huge performance hit |
| 19:02.26 | [TK]D-Fender | dan_j: it is not a variable. It is a function. |
| 19:02.30 | Qwell | also, the disk writes... |
| 19:02.57 | [TK]D-Fender | Monitor + ram disk/SSD |
| 19:03.01 | [TK]D-Fender | FTW |
| 19:03.11 | paulc | bananapie: I read a great article somewhere about using a RAM disk for the mixmonitor, then moving the file to somewhere else at the end of the call |
| 19:03.14 | Qwell | [TK]D-Fender: ssd is slower on writes than a normal disk, isn't it? |
| 19:03.15 | paulc | yeah - that ^^^ |
| 19:03.40 | file | Qwell, if you are using cheap SSDs... maybe... |
| 19:03.43 | [TK]D-Fender | [15:03]Qwell[TK]D-Fender: ssd is slower on writes than a normal disk, isn't it? <- LOL ... NO |
| 19:03.58 | bananapie | paulc, good idea. I'll do that and see if I can record to a ulaw file. If I can do that, it'll probably cause CPU usage to plumet |
| 19:04.03 | Qwell | shrugs and notes his information is a decade old |
| 19:04.15 | sdjernes | You may want to write the files call codec native and then convert it afterwards |
| 19:04.28 | Qwell | sdjernes: mixmonitor must transcode |
| 19:04.40 | Qwell | well, I mean, unless your call is natively SLIN, but nobody does that |
| 19:05.52 | sdjernes | I have done a few calls in slin they used up most of the bandwidth on the internet connection but it t was fun |
| 19:06.16 | bananapie | sdjernes, I think i'll do that. The script that sends the recordings to the file server could run a while [ load is more than 0.5 ] loop with a sleep 10 inside. |
| 19:06.25 | bananapie | before converting it. |
| 19:08.01 | bananapie | if that doesn't reduce CPU, I'll run asterisk with valgrind, valgrind seems to be able to do profiling. But I think that's what Qwell was suggesting |
| 19:08.32 | sdjernes | bananapie: as Qwell said it must transcode but if you got the disk space to keep it in slin for and convert to MP3 or wav later |
| 19:08.34 | Qwell | bananapie: Try just using Monitor, instead of MixMonitor, if you can. |
| 19:08.51 | bananapie | ok. |
| 19:08.53 | bananapie | thanks |
| 19:08.59 | Qwell | The slin>wav conversion will be pretty negligible anyways. |
| 19:09.15 | paulc | +1 for Monitor.. We did that so we'd have caller audio and agent audio in different files, then mixed them together externally, after the call, into a stereo wav --> MP3 file |
| 19:09.22 | Qwell | The big issue is the conversion to SLIN so that the files can be mixed. |
| 19:09.30 | bananapie | ok |
| 19:09.43 | bananapie | if i use monitor, it'll record as g711? |
| 19:09.50 | paulc | bananapie: you saw my PMs too? |
| 19:09.58 | Qwell | bananapie: It'll record as whatever you tell it. If native format is ulaw, use ulaw. |
| 19:10.15 | bananapie | ok |
| 19:10.16 | bananapie | thanks |
| 19:10.30 | Qwell | But honestly, if native format is ulaw, ulaw<>slin is pretty fast, and is not likely to be your problem with only 90 calls, unless you've got a seriously limited CPU. |
| 19:10.53 | Qwell | It'll help some, but I'd guess that the big hit is just the disk I/O |
| 19:11.07 | bananapie | but disk IO should not affect the CPU column in top. |
| 19:11.13 | bananapie | unless I am horribly mistaken |
| 19:12.01 | bananapie | Thanks for the help. I'll try everything you suggested. |
| 19:12.02 | Qwell | "CPU usage" in Linux isn't really as straight-forward as you'd think. |
| 19:12.06 | bananapie | ah. |
| 19:12.25 | Qwell | A lot of that "usage" is probably in the 'wa' column, or whatever, in top. |
| 19:13.13 | bananapie | Something I don't understand is it says 65% cpu for asterisk, but 98.0% idle at the top. Waiting for io is 0.0%. |
| 19:14.11 | bananapie | I think I'll go read about CPU usage and the fields in top and come back later |
| 19:14.11 | bananapie | thanks |
| 19:29.43 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:50.48 | mjordan | Now that everyone has left.... one of the benefits of MixMonitor is that it doesn't stay in the call path. Monitor does. If you have slow disk I/O, you're going to impact the real-time portions of your call. |
| 19:51.18 | mjordan | wanders off |
| 19:53.08 | *** join/#asterisk LiohAu (~LiohAu@bac69-8-88-165-4-152.fbx.proxad.net) |
| 19:54.51 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:22.36 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:23.36 | hecatae | bye mjordan |
| 20:25.18 | *** part/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 20:29.14 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 20:33.08 | *** join/#asterisk metasyntactic (~root@unaffiliated/kunwon1) |
| 20:40.29 | metasyntactic | I've got a problem with Asterisk 11.11.0 sending duplicate SIP INVITE to phones about a tenth of a second apart, why would it do that |
| 20:41.14 | metasyntactic | and they both have CSeq 102 |
| 20:42.41 | file | retransmission probably |
| 20:46.57 | metasyntactic | file: would asterisk attempt to retransmit invite after only 0.1 seconds? |
| 20:48.12 | file | potentially, I think chan_sip will use qualify round trip time when figuring it out |
| 21:02.49 | *** join/#asterisk Fanch (~fanch317@ks3313674.kimsufi.com) |
| 21:09.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:18.27 | marceloamorim | Guys, I used the example of https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause at this example I can get SIP with 200 OK, with asterisk cause code normal clearing, there is any way to define if the hangup was from the caller or the callee? |
| 21:21.07 | *** join/#asterisk Skten (~Sktenau@ppp59-167-129-107.static.internode.on.net) |
| 21:22.06 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 21:26.34 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 21:36.21 | *** join/#asterisk apb1963 (~apb@107-146-213-112.res.bhn.net) |
| 21:43.52 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 21:56.06 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 21:57.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:13.47 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jviellipuildczfq) |
| 22:14.27 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 22:23.18 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 22:30.05 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 22:33.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:45.29 | *** join/#asterisk Penguin (~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us) |
| 22:47.10 | *** join/#asterisk c0rnoTa (~c0rnoTa@kobratel.starlink.ru) |
| 22:47.34 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 22:56.40 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 23:04.34 | *** join/#asterisk tom_w (~telcomind@wsip-24-120-113-148.lv.lv.cox.net) |
| 23:05.00 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 23:07.32 | *** join/#asterisk Cyna (~cynagen@ip72-208-60-104.ph.ph.cox.net) |
| 23:08.33 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 23:10.47 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 23:10.49 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 23:13.44 | spicyramen_ | qq, when I do a calls per second calculation I f I look into CDRs, each CDR record contains calling number and called number each of this calls counts as 1? |
| 23:13.45 | spicyramen_ | or as 2 |
| 23:14.17 | spicyramen_ | 2 call legs |
| 23:27.08 | *** part/#asterisk woleium (~woleium@bc.io) |
| 23:27.54 | Penguin | If you had one device making a call to another device via asterisk, and if you then check core show channels, you'll see that it says 2 channels and 1 call. |
| 23:32.12 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 23:44.27 | *** join/#asterisk Skten (~Sktenau@monitor.ic.ipera.net.au) |