| 00:07.23 | nny | anyone here familiar with the Vega 50? |
| 00:07.37 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 00:08.31 | lorsungcu | anyone had luck with Snom phones and subscribing to contact lists? |
| 00:11.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:11.13 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:11.16 | nny | heh |
| 00:11.25 | nny | how am I suppose to express UTC -5 as a positive integer?! |
| 00:11.41 | nny | i assume i am just retarded |
| 00:13.23 | *** join/#asterisk Gitch_ (~Gitch_@49.176.1.44) |
| 00:14.59 | igcewieling | UTC +29? 8-) |
| 00:15.18 | ketas | hahaha |
| 00:16.47 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-158-043.pools.arcor-ip.net) |
| 00:20.12 | nny | lol |
| 00:20.27 | nny | i figured as much but doesn't that make it 24 hours into the future? |
| 00:20.30 | nny | er nm |
| 00:20.32 | nny | lol |
| 00:21.32 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 00:27.09 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 00:30.07 | *** join/#asterisk protoboardmx (~protoboar@201.120.24.10) |
| 00:33.56 | protoboardmx | Hello everyone |
| 00:35.14 | protoboardmx | I'm trying to use an Avaya 1608 IP phone with Asterisk 1.6.2.20 but so far I haven't had any luck |
| 00:38.24 | [TK]D-Fender | protoboardmx: Do you now have configs and debug to show us for your attempts? |
| 00:39.52 | [TK]D-Fender | ~pb |
| 00:39.52 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:39.53 | [TK]D-Fender | ^^^ |
| 00:41.17 | protoboardmx | ok, let me show you the h323.conf file |
| 00:41.35 | [TK]D-Fender | protoboardmx: And the CLI attempts with H323 debug enabled |
| 00:43.38 | protoboardmx | I'm having trouble figuring out how to see the output on the CLI |
| 00:43.47 | protoboardmx | the problem is I don't really know what to ask |
| 00:45.55 | [TK]D-Fender | asterisk -rvvvvvvvvvvv |
| 00:46.03 | [TK]D-Fender | you should then be at * CLI |
| 00:46.22 | [TK]D-Fender | the do : h323 <tab> and see what the syntax is to debug |
| 00:46.36 | protoboardmx | ok |
| 00:46.38 | protoboardmx | thanks |
| 00:48.05 | *** join/#asterisk igcewieling1 (~igcewieli@user-24-214-153-32.knology.net) |
| 00:59.31 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 01:00.43 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 01:02.31 | *** join/#asterisk zhando (~user@c-67-161-122-77.hsd1.wa.comcast.net) |
| 01:03.15 | zhando | still playing around with soundcards.. Anyone ever get them to work.. alsa/oss/console? |
| 01:03.57 | Phoebus | Night all |
| 01:04.01 | [TK]D-Fender | <PROTECTED> |
| 01:04.17 | [TK]D-Fender | zhando: I highly recommend you use a proper one instead |
| 01:04.17 | zhando | What is it about * that's so different from vlc and other media apps when it comes to the soundcard? |
| 01:04.42 | [TK]D-Fender | zhando: The fact that no attention has been paid to it's implementation. |
| 01:04.57 | [TK]D-Fender | zhando: And it's something that very few people use or need |
| 01:06.00 | zhando | [TK]D-Fender: ok.. another time I guess... |
| 01:06.07 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 01:06.56 | [TK]D-Fender | zhando: Just run a proper soft-phone on your server if you must and let it handle it |
| 01:07.04 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 01:07.14 | zhando | [TK]D-Fender: ok will do.. thanks.. |
| 01:08.03 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 01:09.32 | ketas | nny: do you have sip.conf for me? :P |
| 01:09.46 | ketas | nny: just wanted to see |
| 01:10.24 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 01:15.19 | gusto | hm |
| 01:15.21 | gusto | hi ketas |
| 01:15.24 | gusto | what's up? |
| 01:16.40 | gusto | zhando: what are you doing with soundcards? |
| 01:22.10 | rue_house | zhando, are you using the sound cards to get around your unfunctional mgcp? |
| 01:28.48 | ruben231 | :'(guys help |
| 01:33.35 | ruben231 | still not able to make my asterisk 1.8 to have color on its log..still purely white.. |
| 01:33.51 | WIMPy | xterm works for me. |
| 01:34.01 | [TK]D-Fender | .....log? |
| 01:34.10 | ruben231 | on me even i restart all whilte on teh asterisk log |
| 01:34.18 | [TK]D-Fender | LOG? |
| 01:34.19 | ruben231 | CLI> |
| 01:34.20 | WIMPy | But as I said, as far as I remember it has to fit when the daemon ist started as well. |
| 01:34.30 | [TK]D-Fender | Fix your termcap <- |
| 01:34.44 | ruben231 | asterisk -rvvvvvvvvvvvvvv ------> no color only white logs |
| 01:35.53 | ruben231 | WIMPy:ok bat what part could this be |
| 01:36.41 | WIMPy | What's yur TERM when starting the Asterisk daemon? |
| 01:40.47 | ruben231 | <PROTECTED> |
| 01:41.37 | WIMPy | echo 4 |
| 01:41.44 | WIMPy | oops |
| 01:41.49 | WIMPy | echo $TERM |
| 01:43.08 | WIMPy | or just export/setenv TERM to "linux". |
| 01:44.05 | ruben231 | root@asterisk:~# echo $TERM xterm-color |
| 01:44.58 | WIMPy | And now do ith wherte you start Asterisk |
| 01:45.22 | ruben231 | on the startup script |
| 01:45.24 | ruben231 | ..? |
| 01:45.36 | WIMPy | yes |
| 01:46.01 | WIMPy | And my xterm only set xterm. |
| 01:46.10 | WIMPy | No -color. |
| 01:49.40 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 01:49.44 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 01:54.53 | zhando | gusto: I'm back... Just want to be able to use * as a softphone.. Another tool in the box... mgcp?? |
| 01:55.17 | gusto | no idea |
| 01:55.29 | gusto | however, using asterisk as a softphone seems to be a very bad idea |
| 01:56.17 | [TK]D-Fender | zhando: Asterisk is not the tool for the job |
| 01:56.35 | zhando | gusto: nonsense.. the console should have the most functionality of all... |
| 01:57.45 | gusto | well, but i seem to be in the majority |
| 01:58.08 | zhando | I mean what's the typical solution these days for overhead pagers and *? |
| 01:58.08 | [TK]D-Fender | zhando: There is no basis for this belief of yours |
| 01:58.29 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:58.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:58.42 | [TK]D-Fender | zhando: Asterisk is not a softphone with PBX features. It is a PBX toolkit that if you're desparate you might be able to use the console for a test call. |
| 01:58.56 | [TK]D-Fender | zhando: It has no call handliong capabilities and is fugly |
| 01:59.07 | gusto | yes |
| 01:59.29 | zhando | ok guys I won't debate you on this.. |
| 02:00.13 | [TK]D-Fender | Most use FXS/FXO based solutions on an interface of some sort |
| 02:01.07 | [TK]D-Fender | zhandogusto: I'm back... Just want to be able to use * as a softphone.. Another tool in the box... mgcp?? <- and this is not as a "softphone". Calling out to it for paging is different from expecting call-handling and to use directly. |
| 02:04.12 | zhando | guys I lied a bit.. I was able to get chan_oss working but I had to open /dev/dsp to full permissions.. It was as [TK]D-Fender said "fugly".. |
| 02:04.35 | [TK]D-Fender | ... |
| 02:04.42 | igcewieling1 | zhando: how will you hangup your console "phone"? |
| 02:04.52 | zhando | console hangup |
| 02:04.57 | [TK]D-Fender | I'm not sure which lie is the truth now... |
| 02:04.59 | igcewieling1 | using a sound device may be useful at times. |
| 02:07.32 | zhando | Well at least I'm getting some negative feedback here (with one notable exception).. #asterisk-dev is dead silent.. |
| 02:08.07 | igcewieling1 | zhando: are you trying to modify Asterisk's source code? #asterisk-dev is not 2nd level support. |
| 02:08.39 | igcewieling1 | In any case, I bet they are afraid to respond to someone who is obviously crazy. 8-) |
| 02:09.14 | nny | oh lord |
| 02:09.26 | zhando | igcewieling1: I tried reading it... I even compared it against vlc.. What does snd_pcm_open do wrong in * that it doesn't do in vlc? |
| 02:09.29 | nny | tigase is downloading at a couple bits per second |
| 02:09.50 | nny | [TK]D-Fender: any hacks for having a phone sidecar monitor hints on two servers? |
| 02:10.14 | nny | [TK]D-Fender: tigase is turning into a pia pretty quick |
| 02:10.24 | zhando | igcewieling1: what's crazy about wanting to use existing code? |
| 02:11.42 | igcewieling1 | zhando: because there are so many other tools which are designed to be a softphone. You can use a hammer as a screwdriver, but that doesn't mean it is the right tool. |
| 02:12.03 | nny | 1.42K/s eta 1h 46m heh. Gues i'll just let it download |
| 02:12.08 | igcewieling1 | nny: what brand of phones are you using? |
| 02:12.18 | nny | igcewieling1: cisco 514g |
| 02:12.24 | nny | igcewieling1: so cisco SPA |
| 02:12.58 | igcewieling1 | I was hoping polycom, if you set two regisration servers on polycoms it will register to BOTH servers for each line appearance (as of 2.1.x firmware). |
| 02:13.08 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 02:13.15 | nny | igcewieling1: this cisco may do the same, I am going to test it |
| 02:13.40 | igcewieling1 | this is both REALLY cool and REALLY annoying depending what you are trying to do. 8-| |
| 02:13.56 | zhando | whois igcewieling1 |
| 02:14.04 | igcewieling1 | is aka ManxPower |
| 02:14.10 | nny | he is Vigo! You are like the buzzing oflies to him! |
| 02:14.12 | igcewieling1 | egads my 1 is sticking out |
| 02:14.30 | jpsharp | slimes nny. |
| 02:14.40 | nny | i need cab fare now |
| 02:14.50 | [TK]D-Fender | nny[TK]D-Fender: any hacks for having a phone sidecar monitor hints on two servers? <- reg to each. Read your phone's manual |
| 02:15.14 | nny | [TK]D-Fender: yeah i kind of knew that. Just wanted to say hi in the most annoying way possible |
| 02:15.26 | [TK]D-Fender | nny: Aim high. |
| 02:15.56 | zhando | igcewieling: how did sandy treat you? |
| 02:16.16 | nny | notes sandy is his girlfriend's name... awaits response |
| 02:16.55 | zhando | i have callcentric and were were out a few days.. voip.ms was ruthless about it.. |
| 02:17.17 | nny | nothing morbidly funnier than seeing "[GIRLFRIEND] Affects Thousands" in a newspaper |
| 02:22.28 | nny | nice! |
| 02:22.50 | nny | igcewieling: cisco allows it as well. Just set BLF field to @server 2 and add phone creds to sip.conf |
| 02:26.47 | ketas | hmm ok |
| 02:26.51 | ketas | it seems to work |
| 02:27.35 | ketas | but it sounds like crap |
| 02:31.54 | WIMPy | chan_alsa works very well for overhead paging. |
| 02:32.01 | *** part/#asterisk ruben231 (~OpenDial@112.198.90.187) |
| 02:33.08 | zhando | WIMPy: what do you use as your hardware thingy: hw:0,0 ?? |
| 02:33.10 | ketas | lot improvements now |
| 02:33.30 | zhando | WIMPy: default? |
| 02:33.33 | lorsungcu | i need to add headers to notifies when notifying for DND |
| 02:33.38 | lorsungcu | any ideas how? |
| 02:33.47 | WIMPy | plug:dmix |
| 02:33.58 | zhando | WIMPy: interesting! |
| 02:34.19 | zhando | WIMPy: I'll give that a try! |
| 02:34.47 | WIMPy | For input I use the dummy device and noaudiocapture=yes. |
| 02:34.51 | *** join/#asterisk bytemaster (~erw@host81-150-217-168.in-addr.btopenworld.com) |
| 02:35.52 | zhando | WIMPy: makes sense.. I'm not interested in audiocap at the moment myself. But I will be pretty soon.. Ever get it to work? |
| 02:36.23 | WIMPy | Yes, but I don't use it. |
| 02:36.54 | zhando | And your hardware spec? |
| 02:37.35 | WIMPy | Some sound card with 2 PCM channels. |
| 02:38.06 | WIMPy | ENS1370 |
| 02:38.30 | zhando | WIMPy: I meant output_device=??? |
| 02:39.28 | WIMPy | I'm using plug:dmix, but hw: works as well if the device isn;t in use otherwise. |
| 02:40.20 | zhando | WIMPy: I actually meant input_device=??? Let me guess hw: ... |
| 02:40.32 | WIMPy | yes |
| 02:41.35 | zhando | WIMPy: very good.. Thanks for the tips! |
| 02:56.43 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 02:58.02 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:58.02 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:00.22 | *** join/#asterisk vedic (~V@183.82.82.82) |
| 03:01.25 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 03:01.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:07.00 | *** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net) |
| 03:29.09 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 03:33.02 | ketas | hmm |
| 03:33.28 | ketas | i wonder what makes this crappy sound when i call out via sip into provider's network |
| 03:43.14 | *** join/#asterisk allover12 (47c3bc86@gateway/web/freenode/ip.71.195.188.134) |
| 03:43.45 | ketas | sounds strange |
| 03:44.24 | allover12 | hello everyone :) |
| 03:44.35 | ketas | intermittent wait-connecting tone |
| 03:45.35 | ketas | like something with terrible audio quality |
| 03:45.58 | ketas | it's almost ready |
| 03:46.08 | ketas | besides the weird sound when i call out |
| 03:46.21 | ketas | i don't like to hear it at all |
| 03:48.03 | ketas | but then again, some numbers i call to output normal sound |
| 03:48.42 | ketas | well i have no idea how calls are routed out there |
| 03:48.55 | ketas | and where does this shitty sound come from |
| 03:50.26 | allover12 | anyone here familiar with EAGI app? |
| 03:50.44 | allover12 | having a problem with it for going on 8 hours |
| 03:56.04 | allover12 | i dont want to spam the exact problem until someone is ready to help |
| 03:59.11 | lorsungcu | spam away |
| 04:00.21 | *** join/#asterisk allover12_ (47c3bc86@gateway/web/freenode/ip.71.195.188.134) |
| 04:00.44 | allover12_ | back :) |
| 04:01.43 | asr33 | ~ |
| 04:11.43 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.120) |
| 04:14.36 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 04:16.14 | allover12_ | Can anyone help me with a problem im having using eagi |
| 04:18.52 | *** part/#asterisk vedic (~V@183.82.82.82) |
| 04:30.26 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:32.02 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:34.22 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 04:37.14 | ketas | hm |
| 04:37.25 | lorsungcu | i need to modify what is sent as <state> in dialog-info xml for BLF |
| 04:37.26 | ketas | weird sounds with unknown origin :( |
| 04:37.37 | lorsungcu | is there an easy way to do that? |
| 04:47.33 | allover12_ | looking for help: http://forums.asterisk.org/viewtopic.php?uid=73135&f=1&t=85633&start=0 |
| 04:49.23 | *** join/#asterisk elico (~Thunderbi@bzq-79-176-187-20.red.bezeqint.net) |
| 05:22.27 | lorsungcu | allover12_: is that your actual script |
| 05:22.37 | lorsungcu | or is there a ton more you aren't showing |
| 05:34.50 | kuku | I have a call and I launch a bridge to a dummy context ( that does a playback of a sound ). However, the "other side" ( the number I called , disconnects. the sip phone stays connected. |
| 05:36.11 | lorsungcu | paste bin the dialplan |
| 05:42.17 | kuku | http://pastebin.ca/2311084 |
| 05:43.01 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 05:47.15 | ketas | wow, it works |
| 05:49.20 | kuku | <PROTECTED> |
| 05:51.00 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-fzczgtgimejjtfnq) |
| 06:01.19 | *** part/#asterisk zhando (~user@c-67-161-122-77.hsd1.wa.comcast.net) |
| 06:01.27 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 06:05.15 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-190-103.red.bezeqint.net) |
| 06:06.39 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 06:13.54 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 06:29.04 | ketas | magically it started working |
| 06:29.12 | ketas | or not magically, hmm weird indeed |
| 06:30.02 | ketas | hey, it all works... calls come in, calls go out, there is voicemail and call recording |
| 06:30.09 | ketas | and it even wasn't so difficult! |
| 06:30.10 | ketas | :) |
| 06:35.34 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 06:37.17 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.196.250) |
| 06:51.48 | *** join/#asterisk din3sh (~din3sh@41.136.248.114) |
| 06:51.56 | din3sh | gd mrning all |
| 06:52.34 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.196.250) |
| 06:55.06 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 07:02.58 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 07:10.32 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:10.57 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:14.23 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 07:16.01 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:17.07 | allover12_ | anyone know how i can add a non-user to a confbridge, and then run an application? |
| 07:17.28 | allover12_ | sorry if i worded that wrong |
| 07:18.27 | v0lZy | what do u mean by non-user? |
| 07:18.48 | allover12_ | well, one is dialed out... so it would be a person |
| 07:19.17 | allover12_ | the other would be added to the conference to listen in |
| 07:19.23 | allover12_ | maybe by a call file? |
| 07:20.11 | v0lZy | So, the non-user is actually a person, just not local to asterisk? |
| 07:21.01 | allover12_ | Sorry for being confusing man |
| 07:21.06 | allover12_ | http://www.voip-info.org/wiki/view/Asterisk+cmd+Ices |
| 07:21.40 | allover12_ | simular to what is on that page... with the dialplan and call files |
| 07:22.19 | allover12_ | let me pastebin my dialplan, maybe that will help :) |
| 07:23.10 | v0lZy | Sorry, I have no idea what this is |
| 07:23.47 | kaldemar | allover12_: how do you want to trigger all this? |
| 07:23.55 | v0lZy | at a quick glance, you want to stream conversations to a radiocast or something' |
| 07:24.08 | v0lZy | and then listen to them with VLC etc? |
| 07:24.19 | v0lZy | as if it was an internet radio? |
| 07:24.23 | allover12_ | exactly v01zy... |
| 07:24.27 | v0lZy | L |
| 07:24.31 | allover12_ | @kaldemar call file |
| 07:24.35 | kaldemar | why not listen on them with a soft phone? |
| 07:24.38 | allover12_ | or ami maybe? |
| 07:24.49 | v0lZy | Interesting |
| 07:25.20 | allover12_ | i want the conf bridge to be put out to internet radio |
| 07:25.26 | kaldemar | why call file? if you want to execute an app too, you might want to use a phone for all this. make an extension that originates using the Originate application and then use ChanSpy to listen on the confbridge or whatever. |
| 07:26.02 | kaldemar | so spying on it with a phone is not what you want. |
| 07:26.45 | v0lZy | one reason he might want to do this is like a radio talk show |
| 07:26.45 | allover12_ | i was using the EAGI application to get the audio |
| 07:26.57 | allover12_ | sorry if im confusing guys |
| 07:26.59 | v0lZy | people call in on a phone... |
| 07:27.27 | allover12_ | would i be able to use EAGI with chanspy? |
| 07:28.11 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 07:28.36 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 07:30.46 | allover12_ | Im confused on how to have AMI or a call file join in on confbridge.... once they are joined i can use EAGI to get the audio stream |
| 07:30.49 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.196.250) |
| 07:32.32 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 07:35.57 | allover12_ | kaldemar: maybe with async agi i could get what i want? |
| 07:36.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:36.54 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:48.24 | *** join/#asterisk din3sh (~din3sh@41.136.254.220) |
| 07:48.50 | din3sh | how to set jumper to E1 on a TE220 card |
| 07:59.54 | phix | attach one end to the card and the other to your car battery |
| 08:05.48 | kaldemar | din3sh: you can set the t1e1override module parameter to 0xff when loading the module. |
| 08:06.03 | kaldemar | din3sh: so touching the jumper is not required. |
| 08:08.10 | kaldemar | if you insist on using the jumper, get the manual for the card. |
| 08:08.38 | din3sh | i got it |
| 08:08.56 | din3sh | there's 4 sets of jumper pins |
| 08:09.08 | din3sh | but only 2 jumpers |
| 08:09.14 | din3sh | should i set all 4? |
| 08:09.21 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 08:11.10 | kaldemar | din3sh: what does the manual say? what do you see printed on the board? |
| 08:14.26 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 08:15.38 | din3sh | open=t1 closed=e1 |
| 08:15.51 | din3sh | i have set the 2 jumpers to closed |
| 08:15.59 | din3sh | not sure about the remaining 2 |
| 08:29.29 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-190-103.red.bezeqint.net) |
| 08:35.57 | *** join/#asterisk niluje (~niluje@bdv75-4-82-227-67-242.fbx.proxad.net) |
| 08:39.23 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-158-043.pools.arcor-ip.net) |
| 08:43.36 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:58.31 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:02.26 | ketas | The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead |
| 09:02.30 | ketas | :( |
| 09:05.50 | ChannelZ | Why is that a problem? |
| 09:06.30 | ghost75 | when called person is busy, why i have extension s in cdr? |
| 09:06.55 | ketas | ChannelZ: i had idea of putting + there |
| 09:07.04 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 09:10.41 | kaldemar | ghost75: look what your dialplan does on busy. |
| 09:11.47 | kaldemar | ketas: then use _+. if it suits your needs. the idea is to not match special extensions. |
| 09:12.07 | *** join/#asterisk bruce_ (~bruce_@41.177.76.17) |
| 09:13.01 | ketas | i wish i could join _+. and _X |
| 09:13.03 | ketas | ? |
| 09:13.05 | ketas | _X. |
| 09:13.17 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 09:14.34 | ketas | _{+,X}. |
| 09:14.36 | ketas | :P |
| 09:16.09 | bruce_ | sup |
| 09:16.33 | bruce_ | anyone know anything about pattern matching in dial plans? |
| 09:16.36 | kaldemar | ketas: _[+0-9]. |
| 09:16.50 | ketas | kaldemar: ooh, this even works? |
| 09:17.01 | *** part/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 09:17.06 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 09:17.12 | *** join/#asterisk gajini (~gajini@61.12.12.132) |
| 09:17.28 | kaldemar | ketas: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.6 |
| 09:18.44 | ghost75 | kaldemar: nothing different |
| 09:19.30 | bruce_ | lol |
| 09:21.26 | ChannelZ | no, nothing. |
| 09:25.24 | bruce_ | shit |
| 09:25.43 | bruce_ | ChannelZ, have you used Elastix before? |
| 09:29.41 | *** join/#asterisk hegars (~hegars@061092248178.ctinets.com) |
| 09:31.00 | ChannelZ | no |
| 09:31.20 | ChannelZ | And I was joking. What is your actual question/problem? |
| 09:32.08 | din3sh | join #elastix |
| 09:32.18 | ChannelZ | you |
| 09:32.25 | ChannelZ | :P |
| 09:38.19 | din3sh | i have this strange problem |
| 09:38.35 | din3sh | in my realtime sip_users table in * 1.8.x |
| 09:38.57 | din3sh | i have added deny & permit collumns |
| 09:39.42 | din3sh | for some reason after i add 172.16.3.135/255.255.255.0 for an extension |
| 09:39.48 | din3sh | it is renamed to 172.16.3.135/255.255.255.0/255 |
| 09:40.13 | din3sh | or 172.16.3.2/255.255.255.0/255.2 |
| 09:41.28 | din3sh | any idea why ChannelZ? |
| 09:43.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:43.58 | hegars | din3sh, is it one host or a network? |
| 09:45.58 | din3sh | one host |
| 09:46.10 | din3sh | i have set ACL per host, not network |
| 09:46.28 | hegars | din3sh, then use 255.255.255.255 as the mask |
| 09:46.32 | hegars | not 255.255.255.0 |
| 09:47.00 | din3sh | why so? |
| 09:47.27 | din3sh | the network subnet is 255.255.255.0 |
| 09:48.23 | hegars | din3sh, its basic masking for networks, 255.255.255.255 matches only one host 255.255.255.0 matches the last 8 bits of the 32 bit network addresses |
| 09:48.28 | din3sh | why is the original saved value over-written anyway? |
| 09:49.53 | hegars | not sure what you mean there? |
| 09:49.53 | din3sh | hmmm |
| 09:50.16 | din3sh | ok but still doesnt explain why the value is over-written, does Asterisk over-write it? |
| 09:51.38 | hegars | your peers are in the database I'm guessing from what you said before? |
| 09:51.57 | hegars | *users |
| 09:52.26 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:52.58 | din3sh | yup |
| 09:53.13 | hegars | its your table order correct? |
| 09:53.39 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 09:55.20 | din3sh | order? |
| 09:55.54 | din3sh | permit/deny should have an order? these 2 fields werent present in the DB, i added them |
| 09:56.16 | hegars | deny should be before permit |
| 09:56.20 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9cff:302e:befd:b86c) |
| 09:58.33 | din3sh | it is before |
| 09:58.42 | din3sh | deny 0.0.0.0 |
| 09:58.47 | din3sh | then permit 172.... |
| 09:59.14 | *** join/#asterisk niluje (~niluje@bdv75-4-82-227-67-242.fbx.proxad.net) |
| 10:01.12 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 10:02.19 | din3sh | my questions made him quit |
| 10:05.18 | ChannelZ | You said 'after I add XXX it is renamed to YYY' -- are you doing this through a GUI? |
| 10:07.27 | kaldemar | and where are you looking at the values? where do you see 172.16.3.135/255.255.255.0 and where do you see 172.16.3.135/255.255.255.0/255? |
| 10:09.13 | *** join/#asterisk gajini (~gajini@61.12.12.132) |
| 10:10.23 | din3sh | not gui |
| 10:10.30 | din3sh | manually inserting in the db |
| 10:11.04 | din3sh | the ip remains same, somehow /255 or /255.2 is added at the end |
| 10:11.14 | ChannelZ | but then you do a SELECT on the same rows and it's returning something different, or are you talking about how it shows up within Asterisk? |
| 10:12.19 | kaldemar | din3sh: where are you looking at the values? |
| 10:13.18 | din3sh | am using navicat (mysql gui) |
| 10:13.27 | ChannelZ | wow haven't heard that name in awhile |
| 10:13.32 | din3sh | lol |
| 10:13.47 | din3sh | its working for me..so |
| 10:13.48 | ChannelZ | I used to own that (or whatever it was before it got bought by them) |
| 10:13.49 | din3sh | :p |
| 10:14.08 | ChannelZ | It seems like it's not really working for you |
| 10:14.33 | ChannelZ | It's either display corruption or it's not treating what you're entering as a string (varchar) and doing something bizarre.. |
| 10:15.27 | ChannelZ | Mascon! That's what it was. |
| 10:15.35 | din3sh | o.O |
| 10:15.48 | *** join/#asterisk hegars (~hegars@203186072235.static.ctinets.com) |
| 10:16.19 | kaldemar | so this really has nothing to do with asterisk in the first place? |
| 10:16.44 | ChannelZ | It was very weird. The guy who wrote Mascon decided to quit and then they gave people who owned it licenses to Navicat if memory serves.. but then the original guy turned around and wrote a new one called MyCon. |
| 10:17.13 | din3sh | how is the /255 being added ? |
| 10:17.14 | din3sh | :/ |
| 10:18.02 | ChannelZ | probably by that retarded SQL client |
| 10:18.19 | ChannelZ | You said you created the two new columns in your database because they didn't exist.. what type did you make them? |
| 10:19.03 | din3sh | varchar |
| 10:19.57 | ChannelZ | well get in a shell and connect to the database that way and see if it's the GUI thing screwing up what it's putting in the row or how it's displaying it. |
| 10:20.18 | ChannelZ | because it's either one or the other. |
| 10:21.25 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 10:28.15 | ChannelZ | bed time |
| 10:28.37 | din3sh | its 14:30 here |
| 10:28.42 | din3sh | definitely not bed time |
| 10:28.43 | din3sh | :p |
| 10:32.04 | ketas | echo 'Monitoring test' | mail user@misp-smsgw ; cp -a mobile.call /var/spool/asterisk/outgoing/ |
| 10:32.09 | ketas | lot of fun |
| 10:34.30 | kaldemar | ketas: using cp is likely to cause you issues. asterisk may read a partial file while it is being copied. use a temporary file and mv it to the spool dir. |
| 10:36.09 | ketas | sure, needs cp + mv |
| 10:36.28 | ketas | that's because i need that file again :P |
| 10:43.04 | *** join/#asterisk nacho2k (~Thunderbi@r190-64-14-98.ir-static.anteldata.net.uy) |
| 10:46.09 | wdoekes | ketas: ln without -s |
| 10:46.58 | ketas | oh |
| 10:47.00 | ketas | idea |
| 10:47.09 | ketas | but file wasn't on same fs |
| 10:48.16 | wdoekes | ah.. note that mv isn't atomic over fs boundaries either |
| 10:48.49 | ketas | indeed |
| 10:49.56 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 11:01.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:09.56 | din3sh | res_config_mysql.c: MySQL RealTime: Ping failed (2006) |
| 11:23.56 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 11:38.34 | *** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114) |
| 11:39.59 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 12:02.58 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 12:03.41 | bruce_ | cock suckers |
| 12:08.23 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:08.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:09.25 | din3sh | you drunk mate? |
| 12:09.26 | din3sh | :o |
| 12:13.32 | bruce_ | din3sh, you an african |
| 12:14.44 | bulkorok | my asterisk is segfaulting when ReceiveFax with res_fax_spandsp... is here somebody with a similar problem!? |
| 12:14.56 | bulkorok | I can't reproduce it! |
| 12:15.14 | din3sh | yes am an african, so? |
| 12:15.53 | bruce_ | are you a black african? |
| 12:16.08 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.222) |
| 12:17.21 | din3sh | indian african, why |
| 12:17.24 | din3sh | VERBOSE[28255] res_clialiases.c: == Aliased CLI command 'pri intense debug span' to 'pri set debug 2 span' |
| 12:17.33 | din3sh | is this wrong!? |
| 12:20.15 | kaldemar | why would it be wrong? |
| 12:22.36 | din3sh | how do you disable pri intense debug again? |
| 12:22.37 | din3sh | :o |
| 12:24.58 | kaldemar | pri set debug off span X |
| 12:25.45 | din3sh | thnks |
| 12:25.58 | kaldemar | the ye olde command was "pri no debug span X" i think. |
| 12:30.51 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 12:32.28 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:38.23 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.196.250) |
| 12:45.33 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.196.250) |
| 12:54.18 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 13:05.03 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.196.250) |
| 13:10.43 | bruce_ | how would I setup a dial plan so that if a number like 11XX goes to 11XX |
| 13:10.49 | bruce_ | lol I can't explain this shit |
| 13:11.34 | *** join/#asterisk jkroon (~jkroon@dsl-244-22-121.telkomadsl.co.za) |
| 13:12.11 | bruce_ | it's some pattern matching dial plan or something |
| 13:12.12 | bruce_ | right? |
| 13:12.16 | Greenlight | Yes |
| 13:12.21 | Greenlight | Prefix the extension with _ |
| 13:12.26 | Greenlight | And it'll pattern match |
| 13:12.29 | Greenlight | So, say |
| 13:12.38 | bruce_ | I'm a massive asterisk/voip noob |
| 13:12.43 | bruce_ | can you explain that a little |
| 13:12.46 | Greenlight | exten => _11XX,1,Dial(SIP/${EXTEN}) |
| 13:13.07 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 13:13.15 | Greenlight | You should read the book |
| 13:13.55 | Greenlight | ~book |
| 13:13.55 | 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 |
| 13:16.01 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 13:16.22 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 13:16.53 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:b87c:ed1d:a781:b3f4) |
| 13:18.24 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 13:19.04 | jkroon | hi guys, asterisk 1.8.20.1 - netstat -nulp | grep 5060 indicates that asterisk isn't processing the receive queue on port 5060 |
| 13:19.15 | jkroon | this is on a clean startup ... any ideas/suggestions? |
| 13:19.59 | wdoekes | jkroon: core show locks |
| 13:20.39 | jkroon | wdoekes, no such command ... |
| 13:20.51 | *** join/#asterisk _zoom_ (~zoom@196.1.219.122) |
| 13:21.13 | [TK]D-Fender | jkroon, "sip reload" |
| 13:21.52 | _zoom_ | hi, have pri with over 100 caller id, I need to rotate setting these 100 numbers, any idea? |
| 13:22.03 | bruce_ | Greenlight, where would I put that in Elastix |
| 13:22.24 | jkroon | [TK]D-Fender, just did a restart three times and it didn't help, restarted again now and suddenly it works. |
| 13:22.29 | jkroon | [Feb 6 15:22:11] WARNING[25237]: chan_sip.c:3706 __sip_xmit: sip_xmit of 0x1719ac0 (len 414) to 1.2.3.4:5060 returned -1: Operation not permitted |
| 13:22.36 | jkroon | ooh, would that eventually cause problems? |
| 13:23.11 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:24.49 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:27.24 | [TK]D-Fender | Sounds like * is blocked off |
| 13:28.51 | jkroon | [TK]D-Fender, yea, but 1.2.3.4 is an illegal IP anyway afaik ... turns out one of my technicians thought it was a good idea to add 1.2.3.4 sip.iburst.co.za to /etc/hosts ... |
| 13:29.00 | jkroon | and then use iptables to drop outbound frames to 1.2.3.4 |
| 13:29.08 | jkroon | just pissed off about 300 clients for no good reason. |
| 13:30.56 | [TK]D-Fender | that would do it... |
| 13:31.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:31.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:34.17 | kuku | I have a call and I launch a bridge to a dummy context ( that does a playback of a sound ). However, the "other side" ( the number I called , disconnects. the sip phone stays connected. |
| 13:34.24 | kuku | http://pastebin.ca/2311084 |
| 13:34.40 | jkroon | [TK]D-Fender, ok, still, that error shouldn't cause the entire chan_sip to lock up should it? |
| 13:35.04 | jkroon | so I suspect there is still an issue in chan_sip itself that's causing NOBODY else to be able to access it either. |
| 13:38.10 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 13:40.30 | Ice_Strike | How many channels are possible with 1Mbit connection? |
| 13:40.40 | Ice_Strike | using ulaw codec |
| 13:41.04 | WIMPy | About 10. |
| 13:41.04 | Ice_Strike | 1Mbit upload, about 20Mbit download |
| 13:41.11 | WIMPy | Google for voip bandwidth calculators. |
| 13:41.28 | WIMPy | It also depends on the desired delay. |
| 13:41.33 | WIMPy | But not that muvh with ulaw. |
| 13:42.54 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 13:42.55 | [TK]D-Fender | 85kbps/direction |
| 13:44.29 | Ice_Strike | Hnmmmm |
| 13:44.30 | Ice_Strike | Thanks |
| 13:45.19 | WIMPy | I get about 105kbit on the LAN. |
| 13:46.04 | WIMPy | That's with 10ms frames. |
| 13:47.09 | [TK]D-Fender | WIMPy, The standard and default with * is 20ms mind you... |
| 13:47.28 | [TK]D-Fender | WIMPy, 10ms is better for latency, but worse on BW |
| 13:47.40 | WIMPy | That's why I mentioned it. |
| 13:47.57 | [TK]D-Fender | WIMPy, You're also the only person I've ever heard running that rate :p |
| 13:48.01 | [TK]D-Fender | EVER |
| 13:48.16 | WIMPy | Latency is a big issue, so I try ti minimize it as much as possible. |
| 13:49.22 | kuku | http://pastebin.ca/2311084 << can anyone tell me why the call hangs up ? |
| 13:50.09 | [TK]D-Fender | kuku, You didn't tell Bridge() what CHANNEL to grab |
| 13:50.14 | [TK]D-Fender | ^^^^ |
| 13:51.07 | kuku | I did |
| 13:51.19 | kuku | Launching Bridge(SIP/198-00000019,qB) |
| 13:52.11 | [TK]D-Fender | kuku, I don't see that in your code |
| 13:52.20 | kuku | fputs($socket, "Data: ".$channel.",qB\r\n" ); << this is where I pass it... unless im doing something wrong |
| 13:53.11 | [TK]D-Fender | kuku, Don't mind me... I missed it at the bottom |
| 13:53.12 | *** join/#asterisk rox (~rox@212.30.81.3) |
| 13:53.20 | rox | hello |
| 13:53.23 | [TK]D-Fender | kuku, You really should do things like this in order.... |
| 13:53.55 | rox | is there any way I can control the period of time after which the periodic announce message in a queue is first played? |
| 13:54.09 | kuku | [TK]D-Fender: what order should I do them in ? |
| 13:55.02 | [TK]D-Fender | kuku, And you should be looking with SIP DEBUG to see what happened to the call. |
| 13:55.08 | *** join/#asterisk serafie (~erin@nat/digium/x-rujltfkzhswujmrj) |
| 13:55.35 | [TK]D-Fender | rox, It's always after that amount of time has passed |
| 13:55.56 | rox | [TK]D-Fender: you mean after one period? |
| 13:56.35 | [TK]D-Fender | rox, Correct. Adjusted by your agent dial time. |
| 13:56.40 | rox | [TK]D-Fender: in my measurement it is not, if i set the period to 9 or above, it starts after 17 seconds, if i set it at 8 or below, it sarts after 7 seconds |
| 13:57.27 | [TK]D-Fender | <[TK]D-Fender> rox, Correct. Adjusted by your agent dial time. <- |
| 13:57.38 | [TK]D-Fender | If phones are ringing it won't start until after they stop |
| 13:58.09 | rox | agh, right, the tiem starts ticking only if the phones are not ringing? |
| 13:58.46 | [TK]D-Fender | No, it only starts when they're not. |
| 13:58.55 | rox | bah, i got a client, who wants the message played after exactly 10 seconds, so this means that I really can not control it in that manner |
| 13:59.12 | [TK]D-Fender | Ring=20, period = 90, then it'll only play on 100 |
| 13:59.51 | [TK]D-Fender | rox, You cannot control that. |
| 14:00.18 | jkroon | [TK]D-Fender, 10ms is actually a good idea - especially if you're using G.729 since the recovery characteristics of G.729 is that if you lose more than one consecutive 10ms frame it takes up to 7 or 8 more frames to properly recover. |
| 14:00.44 | kuku | [TK]D-Fender: I dont see anything in sip debug - just a bye |
| 14:00.45 | rox | [TK]D-Fender: ok, thank you very much, at least i got that figured out now |
| 14:00.50 | [TK]D-Fender | jkroon, And you have just shot yourself in the foot while doing it |
| 14:02.01 | jkroon | [TK]D-Fender, i run 20ms because all of my peerings does. it's good enough most of the time. |
| 14:02.15 | [TK]D-Fender | jkroon, You pick G.729 to SAVE BW. The packet overhead is 20kbps for 20ms frames. with 9.6kbps of payload. Imaging the WASTE by halving your data. |
| 14:02.26 | jkroon | and uses significantly less bandwidth, so well worth the risk :) |
| 14:02.51 | [TK]D-Fender | jkroon, You are wasing BW this way. |
| 14:03.24 | jkroon | [TK]D-Fender, it's a fine balance, when trunking (eg IAX/2) I reckon it's OK to use 10ms frames ... with rtp I would not dream of it. |
| 14:04.08 | [TK]D-Fender | jkroon, Ok, if you have enough channels at a time, then the cumulative impact could be worthwhile. |
| 14:04.32 | [TK]D-Fender | jkroon, That's the only real way to justify. |
| 14:05.07 | jkroon | yea, haven't actually done that math yet. only just found about the g729 characteristics like a month back. for now (and the foreseeable future) I'm sticking to 20ms :p |
| 14:05.27 | jkroon | if you have a link where you're losing frames you have much bigger issues anyway |
| 14:05.42 | [TK]D-Fender | jkroon, Do you have a PL issue? |
| 14:06.02 | jkroon | [TK]D-Fender, just had one an hour back ... |
| 14:06.16 | jkroon | vendor sorted out the link pretty quickly from reporting though |
| 14:07.04 | [TK]D-Fender | jkroon, Till then, good work on fixing problems you don't have |
| 14:07.28 | jkroon | hehe, thanks :p |
| 14:07.55 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:08.05 | jkroon | the chan_sip failing entirely one was a major PITA just now though. |
| 14:11.45 | phix | http://www.youtube.com/watch?v=EVcyNANK5cY |
| 14:14.16 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:14.16 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:19.29 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 14:23.25 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:24.46 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 14:26.27 | Kobaz | Set(CONNECTEDLINE(name)=Test); Dial(SIP/7777); |
| 14:26.35 | Kobaz | do i need anything else for connected line updates? |
| 14:27.03 | Kobaz | i have sendrpid=yes for my peer that's dialing (SIP/5610 is calling 7777) |
| 14:27.28 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:29.50 | Kobaz | Application.Exit(); |
| 14:31.27 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 14:31.29 | Rico29 | hi all |
| 14:31.59 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:32.50 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 14:38.35 | *** join/#asterisk bchia (~Adium@nat/digium/x-durkxcrwssutjuba) |
| 14:39.40 | Greenlight | Any reason why Agent/ channels wouldn't always play the on hold music when a call clears? |
| 14:40.55 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 14:42.02 | nunne | Calendar support through EWS. Is it possible to specify which users calendar to view through the URL or a parameter? Because would like to use a global account that has read rights to all the calendars that has a password that never changes etc. |
| 14:43.53 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:45.03 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 14:48.54 | hwt | how can i set asterisk to forward something to TCP via an outbound proxy? |
| 14:49.41 | hwt | transport=tcp, host=...., outboundproxy=.... gives the correct R-URI, but still uses UDP all the way |
| 14:49.59 | hwt | the asterisk should speak UDP with the outbound proxy, which speaks TCP with the next hop |
| 14:51.34 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-190-103.red.bezeqint.net) |
| 14:58.10 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 14:59.29 | [TK]D-Fender | Greenlight, show us |
| 15:00.44 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 15:06.44 | Kobaz | ah |
| 15:06.55 | Kobaz | if you do CONNECTEDLINE before a dial then the dialer clobbers the connectedline |
| 15:06.59 | Kobaz | so you gotta do dial(I) |
| 15:08.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-awanwqkvuinbdguz) |
| 15:08.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:08.27 | *** join/#asterisk ggtest (cf869af1@gateway/web/freenode/ip.207.134.154.241) |
| 15:10.02 | hwt | is it possible at all? |
| 15:10.58 | Kobaz | hwt: yeah |
| 15:11.29 | hwt | Kobaz: oh, sorry, i was refering to the outbound proxy problem i described at all |
| 15:11.38 | hwt | Kobaz: eh, "at all" = "above" |
| 15:11.50 | ggtest | Hi ! |
| 15:11.53 | Kobaz | you can get a call on udp and send it tcp on another peer, that's fine |
| 15:12.40 | ggtest | When I try this: Action: Originate Channel: SIP/200 Application: Background Data: music_file, it works. |
| 15:13.41 | ggtest | But if I try this : Action: Originate Channel: MulticastRTP/basic/239.168.5.5:2000 Application: Background Data: music_file, I hear big noise in the speaker |
| 15:13.55 | ggtest | with asterisk 11.2.1 |
| 15:14.58 | ggtest | I don't know if multicastrtp channel suppor streaming like this |
| 15:17.22 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 15:19.30 | hwt | Kobaz: that's not what i mean. I have an edge proxy which talks TCP to the endpoint, and i talk UDP between Asterisk and the edge proxy |
| 15:20.59 | *** join/#asterisk bmill (~millski@199.167.196.110) |
| 15:26.52 | bmill | hey guys i've got a question.. one of our asterisk servers got hacked and i got blessed with having to analyze packet captures.. my question for you guys is what is i have a tcp stream for flirtmitmir on port 39925. Are any of you familiar with this service? |
| 15:27.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 15:28.08 | Kobaz | hwt: okay... so what's the issue? |
| 15:30.22 | bmill | wow that made no sense, i need more coffee :/ |
| 15:30.23 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 15:31.23 | igcewieling | Some days I want to go postal on a customer. |
| 15:31.38 | igcewieling | My cat could send in better trouble reports. |
| 15:31.40 | kuku | Running 1.8.20.1 - getting a lot of these: [2013-02-06 09:31:08] WARNING[5488]: chan_sip.c:4205 __sip_autodestruct: Autodestruct on dialog '5abdd5c51b4a4be3682f600b071fdc79@192.168.1.242:5060' with owner SIP/83-000008af in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 15:32.18 | Kobaz | mmm |
| 15:34.31 | *** join/#asterisk Defraz (~Defraz@67.60.210.130) |
| 15:36.08 | igcewieling | kuku: make sure ALL ip addresses configured on all interfaces on the Asterisk system are correctly listed in /etc/hosts I've seen this happen when DNS is down and the ips are not in /etc/hosts |
| 15:36.23 | igcewieling | i've seen it in other cases too, but that is the most common cause for us |
| 15:37.47 | kuku | ok - thanks |
| 15:40.58 | kuku | igcewieling: Tried that - didnt help |
| 15:42.41 | kuku | https://issues.asterisk.org/jira/browse/ASTERISK-19425 |
| 15:43.51 | igcewieling | kuku: are you running the latest 1.8.x? |
| 15:44.30 | [TK]D-Fender | <kuku> Running 1.8.20.1 - |
| 15:44.39 | hwt | Kobaz: it doesn't send the ;transport=tcp in the R-URI even if I have transport=tcp on asterisk |
| 15:44.55 | hwt | Kobaz: asterisk ---UDP---> outbound proxy ---TCP---> endpoint |
| 15:45.14 | igcewieling | kuku: according to that jira entry the specific issue addressed in the entry was fixed in 1.8.14.1 |
| 15:45.19 | hwt | Kobaz: and the asterisk is the only one which knows whether it should be TCP, UDP or TLS between the OP and the endpoint |
| 15:45.39 | igcewieling | assuming, of course, the patch ACTUALLY made it into Asterisk. They often don't. |
| 15:49.12 | *** part/#asterisk bmill (~millski@199.167.196.110) |
| 15:50.44 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:52.25 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 15:57.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:57.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:59.22 | Kobaz | hwt: and the problem is?..... |
| 15:59.44 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 16:07.36 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:07.36 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:23.41 | artyx | Is there any known issues on installing asterisk 2.11 on RHEL6 x64 base distributions... i did the addons like sqlite3 and posix |
| 16:24.37 | artyx | But i get error 127... i strace the asterisk binary and it seems to be complaining about the ast*.so.1 file, i set up a symlink on it, (and set up a modules.conf) and still no love... safe_asterisk or whatevers thats called still does not start up error 127 |
| 16:25.23 | *** join/#asterisk lorsungcu (~anonymous@75-144-37-241-Minnesota.hfc.comcastbusiness.net) |
| 16:25.36 | artyx | Ah nevermind, now its working ... and i must be full of it. lol |
| 16:32.23 | artyx | also In 11.2.1 ... is there supposed to be a call to undefined intenral ? (actual spelling) |
| 16:32.31 | artyx | or do you think it is supposed to read internal |
| 16:38.43 | Katty | hi lads. |
| 16:40.45 | Katty | who was i talking to yesterday about lexx? |
| 16:40.53 | artyx | The show? |
| 16:41.05 | artyx | I liked that show |
| 16:41.08 | Katty | apb1963_: YOU. |
| 16:41.12 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:41.12 | Katty | apb1963_: it was you i was talking to. |
| 16:41.33 | Katty | apb1963_: pretty sure when you said it was /odd/ that was the understatement of the month! |
| 16:41.54 | artyx | There is wierder shows out there... |
| 16:42.12 | Katty | idk i was expecting something similiar to firefly |
| 16:42.23 | Katty | not softcore porn...creepy softcore porn |
| 16:42.44 | Katty | sruffell was right :< should have started BSG! |
| 16:43.36 | artyx | HEy they dont make them like that anymore |
| 16:43.46 | artyx | Well they do, just not for broadcast tv ;) |
| 16:44.07 | Katty | i feel very disturbed. |
| 16:44.14 | artyx | Do you anime at all? |
| 16:44.28 | artyx | I seen some things man.... horrible things |
| 16:44.32 | artyx | Wouldn't recommend it |
| 16:44.57 | drmessano | I dont think thats considered "anime" |
| 16:45.48 | artyx | No but if you like the mental mind f* .. there are some great animes otu there that do a nice job of it |
| 16:45.49 | drmessano | Hentai ? |
| 16:45.55 | drmessano | lol |
| 16:46.11 | tzanger | hm, if I want to receive a SIP message (sip sms?) from a softphone to my asterisk 1.8 box, what is the secret sauce? Do I have a special extension in the receiving context of the dialplan? I'm having some trouble finding information |
| 16:46.25 | *** join/#asterisk timahvo1 (~rogue@aptilo.wananchi.com) |
| 16:47.42 | igcewieling | tzanger: in 1.8 SIP messages are only done in the context of a call. Later versions support messaging not as part of a call. Does that help at all? |
| 16:48.12 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 16:49.20 | tzanger | igcewieling: ah, I see. that's unfortunate (for me) |
| 16:51.39 | *** join/#asterisk timahvo1 (~rogue@62.8.87.225) |
| 16:58.17 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:04.00 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 17:19.18 | Greenlight | It's so annoying that the playing of music when Agent/ channels aren't connected is inconsistent |
| 17:24.55 | lorsungcu | wat do |
| 17:30.00 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 17:30.11 | SuperNull | Anyone running realtime for sippears on 11 ? |
| 17:31.14 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 17:31.15 | jmetro | http://tinyurl.com/b49lym7 sip pears. |
| 17:32.16 | navaismo | whats your isue SuperNull |
| 17:34.04 | Greenlight | http://lists.digium.com/pipermail/asterisk-users/2008-January/204812.html <-- This claims agent channels are obsolete; is this the case? |
| 17:40.29 | [TK]D-Fender | Greenlight, Yes |
| 17:42.18 | igcewieling | In asterisk 1.8 if I don't want to use res_timing_dahdi, what is the next best/reliable timing module? |
| 17:42.36 | igcewieling | timerfd and pthread are my options |
| 17:42.56 | navaismo | i was using timerdf |
| 17:45.05 | Greenlight | [TK]D-Fender: Hmm... damn, wish that was more documented. So, what's the alternative ? |
| 17:45.24 | [TK]D-Fender | Greenlight, AddQueueMember |
| 17:45.38 | igcewieling | hmm...on those install I don't see a timerfd timer. |
| 17:46.03 | Greenlight | I want to keep the users connected inbetween calls, like agent channels did |
| 17:46.09 | igcewieling | Greenlight: you should read all the UPGRADE*.txt files in the Asterisk source code so you know when the docs you are reading are out of date. |
| 17:46.47 | Greenlight | I must have missed the section regarding agent channels usage |
| 17:49.24 | [TK]D-Fender | Greenlight, no longer exists |
| 17:49.37 | Greenlight | ? |
| 17:51.19 | Greenlight | At present I'm using Queues, Local channels and ConfBridges to connect agents and customers. However I'm hitting some odd performance problems, and I was advised that agent channels were a better method to facilitate this. |
| 17:51.22 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 17:52.39 | [TK]D-Fender | Greenlight, Get new advice.... |
| 17:52.48 | Greenlight | ^^ |
| 17:52.49 | [TK]D-Fender | Greenlight, chan_agent, AgentLogin = dead |
| 17:53.01 | Greenlight | Okay |
| 17:53.08 | Katty | ATTENTION. |
| 17:53.10 | Katty | Lumpia is delicious! |
| 17:53.12 | Katty | that is all. |
| 17:53.51 | Greenlight | I'm actually thinking that the performance issues could be related to the Queues if anything, so maybe I can still avoid using them, without using agent channels. |
| 17:54.14 | Katty | oh. performance issues. |
| 17:54.16 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:54.17 | Katty | i'm sorry to hear about those. |
| 17:54.32 | Greenlight | heh |
| 17:54.36 | Katty | you know they make something call stallion potion, if you can find a good alchemist. |
| 17:55.08 | Greenlight | I tried some of that already, and smoke started coming out... |
| 17:55.39 | Katty | sounds like you should visit the college of winterhold |
| 17:55.42 | jmetro | for calls lasting longer than four hours... |
| 17:55.44 | Katty | you may be doin it wrong |
| 17:55.57 | Greenlight | I'll have to try that :) |
| 17:56.31 | Greenlight | Oh well, in the meantime, back to the drawing board, and to pull out my "change" to agent channels... have that to look forward to tomorrow. |
| 17:56.34 | Katty | calls lasting longer than four hours?! gwai gwai long duh dong! |
| 17:56.57 | Katty | kwai chur hun-rien duh di fahng! |
| 17:57.21 | jmetro | i think the translator broke. |
| 17:57.22 | Katty | Greenlight: yes. |
| 17:57.32 | Katty | Greenlight: for tonight...there is Vodka. |
| 17:57.42 | Katty | Greenlight: optionally iced tea if you're not feeling Vodka. |
| 17:57.47 | Greenlight | Great advice! Laters all.... :) |
| 17:57.52 | Katty | tata. |
| 17:57.54 | apb1963_ | Oh Katty.... it was a funny show. |
| 17:58.05 | Katty | jmetro: yes. that would be cursing from firefly |
| 17:58.16 | Katty | jmetro: i googled it tho. it's not like i'm actually smart. |
| 17:58.33 | Katty | apb1963_: if by /funny/ you mean quite /disturbing/ |
| 17:58.39 | Katty | apb1963_: wait. are we talking about firefly or lexx? |
| 17:58.44 | apb1963_ | lexx |
| 17:58.52 | Katty | apb1963_: then yes. most disturbing so far. |
| 17:58.56 | Katty | apb1963_: on episode 3. |
| 17:58.59 | apb1963_ | It didn't disturb me at all |
| 17:59.03 | Katty | possibly 4 |
| 17:59.14 | Katty | i just got done with the ones about the Worms. |
| 17:59.19 | Katty | who liked the bright green Pattern |
| 17:59.23 | apb1963_ | I don't remember specifics.. it was quite some time ago |
| 17:59.39 | Katty | ah...well it reminded me of Soylent Green...which may have been the point |
| 17:59.44 | apb1963_ | lol |
| 17:59.45 | jmetro | wait, Lexx = Farscape? |
| 17:59.53 | [TK]D-Fender | ... |
| 17:59.54 | [TK]D-Fender | no |
| 17:59.54 | Katty | Lexx =/ Farscape |
| 17:59.57 | apb1963_ | not by a longshot jmetro |
| 18:00.05 | Katty | i remember farscape being cool. |
| 18:00.09 | Katty | with pretty boys. and big guns. |
| 18:00.10 | apb1963_ | it was |
| 18:00.13 | jmetro | ^ |
| 18:00.21 | Katty | there are no big guns in Lexx. |
| 18:00.25 | apb1963_ | umm... I don't know about pretty boys |
| 18:00.27 | Katty | just pretty girls and suggestive soft porn content |
| 18:00.29 | [TK]D-Fender | Katty, Just one |
| 18:00.33 | jmetro | Chiana. |
| 18:00.46 | apb1963_ | Lexx IS a big gun |
| 18:00.52 | apb1963_ | It destroys planets |
| 18:01.04 | apb1963_ | they don't get much bigger than that |
| 18:01.06 | Katty | Lexx isn't pretty. |
| 18:01.28 | apb1963_ | no but the woman is.. I forget her name |
| 18:01.38 | Katty | zeb? |
| 18:01.41 | apb1963_ | maybe |
| 18:01.43 | [TK]D-Fender | Zev / Xev |
| 18:01.51 | Katty | ah yes. Zev |
| 18:01.52 | apb1963_ | That sounds familiar |
| 18:02.31 | *** join/#asterisk shido6 (~shido6@c-98-234-178-147.hsd1.ca.comcast.net) |
| 18:02.35 | artyx | Farscape was jim hensons work no? Great stuff |
| 18:02.44 | artyx | I thik it was also aussie |
| 18:03.22 | apb1963_ | It was poorly named. I avoided watching it for a very long time because I thought it was some kind of geography show or something. |
| 18:03.42 | apb1963_ | I think that's why it didn't do so well and got cancelled |
| 18:04.10 | artyx | I feel that if it went as long as stargate did it would have lost its appeal |
| 18:04.10 | apb1963_ | There's an episode in stargate where they make reference to it. |
| 18:04.30 | Katty | never did get into star gate. |
| 18:04.32 | artyx | I mean.. i dunno about you, but you can only do the same universe for so many episodes before your writers have constipation |
| 18:04.41 | jmetro | Farscape was excellence. |
| 18:04.51 | artyx | Hell yes, crackers dont matter ! |
| 18:04.51 | Katty | not enough eye candy in stargate i supposed. |
| 18:05.02 | apb1963_ | I prefer unsalted |
| 18:05.27 | artyx | I have the farscape bluray collectors edition, it was given to us by the distributor for the purposes of my wife reviewing it |
| 18:05.31 | Katty | crackers don't have nearly enough fiber for that |
| 18:05.39 | artyx | I also have the PK wars on oldschool dvd... that was a good followup |
| 18:05.49 | artyx | Well normal crackers dont katty. but space krackers |
| 18:06.04 | Katty | ah. i don't get the reference. but that's ok! don't ruin it. |
| 18:06.08 | Katty | i will get it eventually, surely. |
| 18:06.16 | artyx | I think its somewhere in the end of season 2,early season 3 |
| 18:06.28 | artyx | forget exactly. its a title of an episode |
| 18:06.58 | [TK]D-Fender | Katty, Personal onte on it : S4 EP 10 = most important piece of enlightenment. Look forward to it... |
| 18:07.01 | [TK]D-Fender | note* |
| 18:07.06 | chuckf | Farscape was great, Lexx the movies were more pornish than the series, and the series had its moments |
| 18:07.25 | artyx | What about that 80's show max headroom |
| 18:07.36 | artyx | and if your on the subject of sci-fi . how about V the series |
| 18:07.41 | artyx | loved V |
| 18:07.55 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-tgzbdzhcwhmolaqo) |
| 18:07.58 | chuckf | I'm surprised that Max Headroom hasn't been rerun in a while |
| 18:08.13 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 18:08.24 | psykon | chuckf: I was just think that |
| 18:08.25 | artyx | I think there were licensing issues regarding the holder |
| 18:09.00 | psykon | "thinking" that is |
| 18:09.09 | jmetro | enders game - the movie. |
| 18:09.19 | artyx | Im holding out for that.. i have high hopes but low expectations |
| 18:09.26 | jmetro | harrison ford |
| 18:09.33 | artyx | As mazer? |
| 18:09.34 | chuckf | The thread of the show would be neat to see. Much like today but the real screens we carry around are much smaller |
| 18:09.36 | artyx | graft? |
| 18:09.51 | artyx | graff even, oops |
| 18:10.01 | jmetro | Graff yeah |
| 18:10.07 | jmetro | I wish he was mazer. |
| 18:11.46 | chuckf | Blake's 7 is one that I haven't seen in ages too |
| 18:13.05 | coppice | Blake's 7 led to charity events to send the main characters to acting school :-) |
| 18:14.17 | chuckf | heh, I saw it as a kid, thought it was good then |
| 18:14.54 | *** join/#asterisk AviMarcus (~avi@bzq-79-183-162-87.red.bezeqint.net) |
| 18:15.07 | AviMarcus | Has anyone seen an spa-2102 request the provisioning file every 40-70 seconds for an hour? (And then lose the connection) |
| 18:15.26 | coppice | chuckf: well, it didn't become a classic through the superiority of its acting or special effects :-) |
| 18:15.48 | artyx | does libpri install the zaptel dependancies? |
| 18:16.17 | artyx | i installed dahdi-linux-complete and wanpipe still isn't working right. |
| 18:18.09 | artyx | only thing i can think of is i set it up for dahdi and not zaptel (which should be the same damn thing no?) |
| 18:18.38 | WIMPy | ... in different times, yes. |
| 18:20.09 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:20.10 | artyx | It used to be called zaptel, and now its dahdi. wanpipe presents me with an option to specify dahdi or zaptel and the dir. and after specifying the install location (/usr/local) where dahdi resides |
| 18:20.41 | artyx | or is it /usr/local/include. if orget... It builds and int heory just works. at least thats how its always worekd out for me before. .. but THIS box is giving me grief |
| 18:20.48 | igcewieling | artyx: no. It wants the location of the DAHDI srouce code, not the installed files. |
| 18:20.54 | artyx | oops |
| 18:21.17 | artyx | your right, its linking to usr/local/src |
| 18:21.18 | artyx | my bad |
| 18:21.24 | artyx | but still.. its failing :P |
| 18:22.09 | artyx | got this B600 in mint conditionf or like a hundred bucks |
| 18:22.33 | artyx | 4 port fxo 1 fxs with hardware ec and $100. score =) |
| 18:25.20 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 18:27.33 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 18:29.02 | *** join/#asterisk Tymanthius (~Tymanthiu@wsip-98-175-23-210.br.br.cox.net) |
| 18:29.44 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:32.18 | Tymanthius | Hello! I was wondering if anyone could help me figure out why a script won't run from my dial plan? It spits out this error: [2013-02-06 12:28:03] WARNING[12410]: app_system.c:138 system_exec_helper: Unable to execute 'receivedfax.sh, <arg1>, <arg2>, <arg3>' The script runs fine from the command line. |
| 18:34.59 | igcewieling | Tymanthius: su -l asterisk -c 'fullpathtoscript' |
| 18:35.13 | igcewieling | also use the full path to the script in your dialplan |
| 18:36.46 | kaldemar | Tymanthius: show the line in your dialplan |
| 18:36.55 | Tymanthius | Tried useing the full path to script in the dial plan, and it gave the same error. EVen had it in a /test dir that was owned by asterisk:asterisk, and the dir & file were chmod 777 |
| 18:40.01 | Tymanthius | igcewieling: That command doesn't spit out errors, but doesn't seem to run the script either. |
| 18:41.04 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 18:41.10 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 18:41.47 | igcewieling | Tymanthius: aha! remove the spaces after the commas |
| 18:41.55 | Tymanthius | kaldemar: I'm using the ael file, I believe this is the relevant section: h => { |
| 18:41.55 | Tymanthius | <PROTECTED> |
| 18:41.55 | Tymanthius | <PROTECTED> |
| 18:41.55 | Tymanthius | <PROTECTED> |
| 18:41.55 | Tymanthius | <PROTECTED> |
| 18:43.48 | Tymanthius | igc: Will do, brb |
| 18:45.49 | kaldemar | Tymanthius: core show application System |
| 18:47.17 | kaldemar | arguments for the command are not separated by commas |
| 18:47.54 | igcewieling | generally arguements to applications and functions should not have spaces in them |
| 18:49.21 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 18:51.07 | Tymanthius | grrrr . . . faxzero seems to be down so I can't send myself a fax. |
| 18:51.22 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 18:59.40 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 19:03.11 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ckllwcvpndsudhye) |
| 19:03.52 | Tymanthius | Ok, got a fax sent in, and even with the spaces removed, it gave the same error. |
| 19:04.47 | *** join/#asterisk lorsungcu (~anonymous@12.40.176.42) |
| 19:07.04 | Tymanthius | ANd just to be clear, I get the faxes (they are sitting in my /var/pool/asterisk/faxes dir). The script just refuses to run. |
| 19:11.21 | Tymanthius | Kaldemar: Jsut read the core show line - everything comes up "Not Available" I'm on an ubuntu 12.04.1 headless server, installed from repo's. |
| 19:12.07 | *** join/#asterisk kleszcz (tick@linuxmafia.pl) |
| 19:15.39 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 19:16.05 | *** join/#asterisk lorsungcu_ (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 19:20.13 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 19:21.11 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 19:21.15 | kaldemar | Tymanthius: spaces? you were supposed to remove commas. system takes a plain shell command. |
| 19:23.03 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 19:23.09 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 19:23.17 | Tymanthius | Ok, I'm trying no comma's, just spaces. Although it was like that before. I (mis)read a config on the wiki. I think I read for agi commands, and that's why I had the commas. |
| 19:24.30 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 19:29.17 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 19:29.29 | Tymanthius | Ok, I've got it. It's a permissions issue. I moved back to the one in /test, after checking my permissions there, and THAT is working. Thanks guys. |
| 19:30.10 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:32.27 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 19:36.34 | leifmadsen | sees some AEL in the backlog, pukes a little in his mouth |
| 19:36.56 | igcewieling | leifmadsen: AEL is AWESOME! |
| 19:37.05 | leifmadsen | I disagree, but ok :) |
| 19:37.09 | igcewieling | 8-) |
| 19:37.21 | leifmadsen | maybe I'm just an old and crusty Asterisk admin now |
| 19:37.33 | igcewieling | leifmadsen: We use AEL mainly with Dial, most everything else is in AGIs |
| 19:38.53 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 19:39.31 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 19:39.53 | Tymanthius | lol - I can read ael more readily than I can conf exten lines. But I'm still getting MASSIVE help from the web, and friendly peoples. |
| 19:40.30 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 19:41.12 | gavimobile | res_calendar_icalendar.so says that it supports .ics intergration, does this mean I just add the url to the ics location? |
| 19:41.45 | gavimobile | can't find anythiny sample calendar.conf samples online doing this.. not sure what it means |
| 19:44.43 | lorsungcu | leifmadsen: know of any way to change the dialog-info bits of notify packets? |
| 19:46.14 | leifmadsen | gavimobile: did you read the calendar stuff written in the book? |
| 19:46.25 | leifmadsen | lorsungcu: use something like opensips or kamailio |
| 19:46.46 | gavimobile | leifmadsen: yap! |
| 19:47.06 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 19:47.40 | lorsungcu | yeah that's sort of what I was leaning towards |
| 19:47.41 | lorsungcu | thanks |
| 19:47.46 | gavimobile | leifmadsen: this specific module is NOT discussed in the book |
| 19:47.53 | gavimobile | caldav is mentioned |
| 19:47.54 | leifmadsen | lorsungcu: basically if you need to manipulate SIP packets directly, move that out of Asterisk |
| 19:48.07 | leifmadsen | gavimobile: sorry, guess never used whatever part you're talking about |
| 19:48.08 | gavimobile | but icalendar is NOT |
| 19:48.17 | leifmadsen | correct, it is NOT |
| 19:48.23 | leifmadsen | although I don't understand the need for the emphasis |
| 19:48.42 | gavimobile | no emphasis :-) |
| 19:48.58 | gavimobile | just trying to get my calendar back up and running. im on like day 8 |
| 19:49.22 | lorsungcu | sounds like its working fine, if you know what day you're on |
| 19:49.28 | gavimobile | http://forums.asterisk.org/viewtopic.php?f=1&t=85623&p=183332#p183332 |
| 19:49.46 | gavimobile | this is my post on the forums.. someone else claims the same this is a server issue |
| 19:51.03 | leifmadsen | <gavimobile> but icalendar is NOT |
| 19:53.00 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 19:53.11 | gavimobile | leifmadsen: im sorry? |
| 19:55.21 | gavimobile | I wonder what else I can use other than google and exchange |
| 19:55.36 | gavimobile | can't find any information to connect to apple |
| 19:55.57 | *** join/#asterisk ghost75 (~trechber@dslb-178-002-147-114.pools.arcor-ip.net) |
| 20:11.35 | ghost75 | is it possible to use thunderbird with tapi and asterisk? |
| 20:12.12 | [TK]D-Fender | ghost75, I'd probably google "asterisk TAPI" and see what shows up... |
| 20:12.19 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:12.36 | jmetro | Id google "Google TISP" |
| 20:12.36 | ghost75 | a lot ... |
| 20:15.24 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.165.130) |
| 20:17.04 | ghost75 | jmetro: thats at least 5 years old |
| 20:19.32 | jmetro | still a very fat pipe for a 5 year old connection |
| 20:21.08 | ghost75 | better get wireless cable |
| 20:23.03 | drmessano | I have a 1000ft box of wireless ethernet cable |
| 20:23.06 | drmessano | $75 |
| 20:23.10 | drmessano | Paypal in PM |
| 20:24.00 | ghost75 | i 'll send you nigerian agent |
| 20:28.08 | Katty | 75 bucks? |
| 20:28.13 | Katty | for ethernet cable? |
| 20:28.19 | Katty | i wouldn't touch that with a 10ft pole for a grand. |
| 20:28.22 | ghost75 | wireless! |
| 20:28.41 | Katty | i didn't see the wireless bit. |
| 20:28.44 | Katty | still sounds like work ;) meh |
| 20:29.05 | Katty | drmessano: maybe your wife will volunteer. |
| 20:29.50 | *** join/#asterisk din3sh (~din3sh@196.20.241.110) |
| 20:29.55 | din3sh | hello all |
| 20:30.33 | chris_n | offers drmessano a cable stretcher to go with that wireless cable |
| 20:30.38 | chris_n | $15 |
| 20:30.53 | din3sh | wct4xxp 0000:0b:08.0: TE210P: RECEIVE slip NEGATIVE on span 2 |
| 20:31.05 | din3sh | what does this mean? |
| 20:31.15 | *** join/#asterisk kresp0 (~screspo@178.Red-80-33-64.staticIP.rima-tde.net) |
| 20:40.24 | kresp0 | Hi all, I'm having trouble making dahdi work with a X100P FXO card |
| 20:40.31 | kresp0 | this is what I've done so far: |
| 20:40.35 | kresp0 | http://pastebin.com/3zAtkDrL |
| 20:42.07 | *** join/#asterisk tc (~travis@rrcs-67-78-243-170.se.biz.rr.com) |
| 20:43.36 | tc | Hi. Does anyone know whether Asterisk correctly implements DNS SRV now (e.g. priorities, weights, and ports of the various records are respected)? If so, do you know what version that happened in? |
| 20:43.52 | tc | I found conflicting information online. |
| 20:45.26 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-190-103.red.bezeqint.net) |
| 20:45.31 | ghost75 | i remember something "1st record only" |
| 20:45.44 | kresp0 | dahdi_hardware locate the card, but dahdi_genconf dont. what should I try next? |
| 20:47.02 | tc | ghost75: Yes, I found that mentioned someone too. I found it so hard to believe that hadn't been fixed by now I thought I'd ask. |
| 20:47.22 | tc | ghost75: I checked in the bug tracker and couldn't find the right one. |
| 20:48.01 | tc | s/someone/somewhere/ |
| 20:49.16 | kresp0 | ok, solved. I've rebooted and now it works :) |
| 20:49.17 | ghost75 | no clue i dont use fqdn in dialplan |
| 20:49.18 | file | tc, it will sort and pick one - but only one and stick with it |
| 20:50.29 | navaismo | kresp0, the modules file in /etc/dahdi/ contains the wcfxo module uncommented? |
| 20:50.35 | navaismo | ha got it |
| 20:50.42 | tc | file: Thanks, I see. I have a customer running 1.6.0.26 and it's not using the port number in the SRV record. Should I tell them to upgrade to a particular version. |
| 20:51.01 | *** join/#asterisk horatio (~horatio@unaffiliated/horatio) |
| 20:51.16 | file | I don't know, nothing immediately springs to mind on any issue about that - best bet is to search JIRA |
| 20:51.32 | horatio | Is there anyone around who knows anything about the configuration of NEC PBXs? Specifically on an sv8100 how I pick a particular line to be in DISA mode rather than DDI mode? |
| 20:51.36 | tc | tc: I also had them try turning on dnsmgr=yes as that helped some, but not all people in ASTERISK-17722. |
| 20:51.40 | tc | file: https://issues.asterisk.org/jira/browse/ASTERISK-17722 |
| 20:52.11 | file | then it's entirely possible that is indeed an issue |
| 20:52.13 | kresp0 | ty navaismo ;) |
| 20:52.56 | *** join/#asterisk JohnnyAsterisk (~cianmaher@154.50.194.130) |
| 20:53.38 | tc | file: Thanks. It just seemed so unlikely for a compatibility issue like that to be outstanding so long that I thought I'd ask. Thought perhaps it could have been fixed but the bug report abandoned. |
| 20:54.00 | file | bug reports are never abandoned, sometimes just forgotten if someone fixes it randomly |
| 20:54.12 | JohnnyAsterisk | ok so this is frustrating…. compiling dahdi 2.4.1 complete everything goes fine, it compile i make the config and then start asterisk…. but there is no dahdi, when i do a search for the dahdi module its no where to be found on the server |
| 20:54.17 | tc | file: Yes indeed. |
| 20:54.25 | JohnnyAsterisk | tried with 2.6.1 and 2.4.1 any ideas? |
| 20:54.31 | file | not likely something would ever touch randomly |
| 20:55.13 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:55.49 | tc | file: Probably true. Thanks again. |
| 20:56.24 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 21:01.27 | JohnnyAsterisk | dahdi compile find but doesn't drop our chan_dahdi.so |
| 21:02.27 | navaismo | that is generated by asterisk IIRC |
| 21:04.33 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:22.43 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:24.07 | *** join/#asterisk ideaman55 (~ideaman55@173-10-29-217-BusName-utah.ut.hfc.comcastbusiness.net) |
| 21:28.22 | *** join/#asterisk Dovid (~Dovid@246.sub-70-215-64.myvzw.com) |
| 21:30.02 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.165.130) |
| 21:42.56 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 21:42.57 | igcewieling | JohnnyAsterisk: when you build asterisk if it finds dahdi installed in the system then it will generate chan_dahdi.so |
| 21:47.08 | *** join/#asterisk BrokenArrow (~Lp@unaffiliated/brokenarrow) |
| 22:09.14 | *** join/#asterisk rdm (~rdm@unaffiliated/qubix) |
| 22:25.13 | *** join/#asterisk wwalker (~wwalker@208.92.232.27) |
| 22:25.57 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:44c9:c90b:a530:9b49) |
| 22:26.20 | wwalker | would someone ping the bot and have it tell me about provider lists? I can't find that URL |
| 22:27.47 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:27.47 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:27.51 | jmetro | ~providers |
| 22:27.51 | infobot | extra, extra, read all about it, providers is http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
| 22:27.52 | jmetro | ? |
| 22:29.29 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 22:29.33 | cj | hey folks |
| 22:29.40 | wwalker | <PROTECTED> |
| 22:29.45 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:29.56 | jmetro | yep, bots eat ~'s |
| 22:30.42 | cj | when I press a key my analog phone's keypad, the ATA it attaches to sends it immediately to my asterisk server. what should I look for to have the delay wait for a bit before assuming I've finished entering the full number? |
| 22:30.56 | cj | this is a grandstream 24-port ata, in case anyone knows that platform |
| 22:32.34 | cj | hmm... maybe it is out of time sync and I need to tell it the right network settings so it can a) resolve the ntp hostname and b) find a route to said ntp host ;-) |
| 22:34.51 | wwalker | What I really need is a toll free number _in_Canada_ that forwards to my cell phone. I'm not worried about cost as I will only use it to verify that our sip trunk providers and candidates to become sip trunk providers. We've had problems calling Canadian toll free numbers repeatedly with multiple providers. |
| 22:35.58 | wwalker | I didn't find any country info on the voipreview pages, so I'm hoping someone has used a service that provides Canadian toll free? |
| 22:36.52 | leifmadsen | voip.ms and unlimitel.ca |
| 22:36.58 | leifmadsen | both of those companies should |
| 22:37.01 | wwalker | leifmadsen: thank you! |
| 22:37.03 | leifmadsen | also potentially les.net |
| 22:39.18 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:40.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:40.16 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 22:42.33 | dwayne | wwalker, I don't believe les.net has a 9 AM Eastern support option if that matters to you |
| 22:43.59 | *** join/#asterisk jsjc (~Adium@164.Red-2-136-102.dynamicIP.rima-tde.net) |
| 22:44.30 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 22:46.05 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 22:46.44 | sezuan | I need some debugging help. I've upgraded my from an older 1.8.x to the 1.8.19.1 and 1.8.20.1. It seems that asterisk doesn't respond to network packets anymore. I can see with netstat that data is waiting. |
| 22:47.38 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-awanwqkvuinbdguz) |
| 22:49.37 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 23:01.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:04.31 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 23:07.47 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 23:09.42 | igcewieling | sezuan: does ifconfig show anything inteesting? |
| 23:12.05 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:20.17 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 23:27.15 | *** part/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 23:31.50 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 23:36.51 | cj | okay, so it wasn't clock skew |
| 23:37.03 | cj | maybe I need to put the DSLAM into interleaved mode |
| 23:39.12 | sezuan | igcewieling: no, all error counters show zero. iptables looks ok, no insane fail2ban rule. |
| 23:39.33 | sezuan | the asterisk <-> jabber connection is fine at the same time. |
| 23:45.26 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 23:50.14 | Mon|A|rch | so, I'm trying to detect answering machines in my IVR, that way the IVR won't go ahead and leave uselsss messages on people's answering machines |
| 23:50.16 | Mon|A|rch | http://pastebin.com/zgFLvzPU |
| 23:50.33 | Mon|A|rch | it detects answering machines very well |
| 23:50.53 | Mon|A|rch | but it apparently can't tell the difference between an answering machine and a real person |
| 23:51.09 | Mon|A|rch | am i going to need to overwrite the parameters from amd.conf manually in the application call? |
| 23:51.18 | Mon|A|rch | i can paste amd.conf |
| 23:51.21 | WIMPy | So what part is it, it does very well then? |
| 23:51.37 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 23:51.59 | Mon|A|rch | when an answering machine answers, it recognizes it (apparently) and heads to the hangup extension |
| 23:52.16 | Mon|A|rch | when a human answers, it thinks an answering machine is on the line |
| 23:52.32 | Mon|A|rch | i guess very well isn't accurate |
| 23:52.33 | Mon|A|rch | but w/e |
| 23:52.35 | darkdrgn2k | hey guys question |
| 23:52.36 | darkdrgn2k | http://pastebin.ca/2311299 |
| 23:52.41 | darkdrgn2k | why am i getting 403 on that one? |
| 23:52.48 | WIMPy | Sounds like no detection to me. What am I missing? |
| 23:52.56 | darkdrgn2k | is it complainig about codec? |
| 23:53.17 | Mon|A|rch | WIMPy, you're probably right, i just don't know the amd application well |
| 23:53.29 | [TK]D-Fender | darkdrgn2k: No. |
| 23:53.44 | darkdrgn2k | so whats the 403 mean? does it actualy matter? |
| 23:53.50 | WIMPy | darkdrgn2k@ What do you make of the word "Forbidden" after that 403? |
| 23:53.56 | igcewieling | Mon|A|rch: tuning AMD is a black art. I used to work for a company which considered the AMD settings they came up with as a company secret. |
| 23:54.16 | Mon|A|rch | igcewieling, oh god |
| 23:54.20 | igcewieling | They were all complete nutjobs, but the product was very popular. |
| 23:54.24 | darkdrgn2k | WIMPy: not sure... |
| 23:55.09 | [TK]D-Fender | darkdrgn2k: it's documented. |
| 23:55.13 | Mon|A|rch | igcewieling, there any semi-effective examples online? |
| 23:55.20 | igcewieling | They may or may no have patched AMD, I don't recall. |
| 23:55.33 | darkdrgn2k | forbidden usualy is user login/password isses no? |
| 23:55.39 | Mon|A|rch | I'm in version 1.8, if that makes a difference igcewieling |
| 23:55.43 | igcewieling | Mon|A|rch: no idea. I was never crazy enough to try. |
| 23:55.53 | Mon|A|rch | lol |
| 23:56.03 | Mon|A|rch | i see |
| 23:56.11 | Mon|A|rch | well, it's a robot going out and calling people |
| 23:56.16 | Mon|A|rch | so I guess I'll have to find something |
| 23:56.20 | Mon|A|rch | thanks for cluing me in |
| 23:56.35 | darkdrgn2k | <PROTECTED> |
| 23:56.36 | darkdrgn2k | http://pastebin.ca/2311300 |
| 23:56.39 | igcewieling | In my experience most companys which need to detect answering machines are either collection agencies or telemarketers and I try not to consort with the devil. |
| 23:57.10 | Mon|A|rch | we're medical, we're using a robot to call people asking if they need a refill on their medication/med supplies |
| 23:57.19 | [TK]D-Fender | darkdrgn2k: Thre is no "acceptance" in there |
| 23:57.19 | Mon|A|rch | but, you're right |
| 23:57.49 | darkdrgn2k | so its an auth issue? |
| 23:58.24 | Mon|A|rch | i think he means that the server isn't giving you the option to log in |
| 23:58.27 | Mon|A|rch | it's forbidden to you |
| 23:59.02 | [TK]D-Fender | darkdrgn2k: Did you look up what SIP 403 is defined as? |