| 00:02.26 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 00:02.55 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 00:04.38 | *** join/#asterisk cellkill (~kvirc@2001:470:c469:beef:c077:798d:9cf9:4e1a) |
| 00:07.37 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 00:08.38 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 00:23.12 | *** join/#asterisk sxpert (~sxpert@navsys.org) |
| 00:25.14 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
| 00:26.22 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 00:28.21 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 00:36.55 | *** join/#asterisk sxpert (~sxpert@navsys.org) |
| 00:39.01 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 00:44.56 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:46.18 | *** join/#asterisk hugogee (~hugogee@cpe-76-167-87-159.socal.res.rr.com) |
| 00:46.26 | hugogee | greets all :D |
| 00:48.16 | hugogee | What does one use to write asterisk scripts using python? |
| 00:50.22 | km2 | what controls the timezone of times in /var/log/asterisk/cdr-csv/*.csv? |
| 00:51.20 | cusco_ | someone earlier said something about some setting in cdr.conf |
| 00:51.20 | Freeaqingme | hugogee, you're probably looking for AGI |
| 00:51.59 | cusco_ | gmttime=yes or so |
| 00:53.36 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:54.59 | hugogee | Freeaqingme: Thanx just found a few libs. :D Off i go... cheers to all! |
| 00:55.35 | km2 | thanks, i'll look |
| 00:55.50 | Freeaqingme | hugogee, enjoy. Feel free to hang around though ;) |
| 00:58.28 | *** join/#asterisk Marvelous (~CE0@197.195.78.147) |
| 00:58.45 | Marvelous | HELLO |
| 00:59.37 | carrar | hi |
| 00:59.44 | carrar | Welcome to #Asterisk |
| 00:59.54 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 01:00.06 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:00.07 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 01:02.17 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 01:04.18 | Marvelous | i need to know which file content the configuration for Asterisk⢠Configuration Engine |
| 01:08.40 | p3nguin | I guess that would be asterisk.conf. |
| 01:10.59 | Marvelous | wrong file |
| 01:11.12 | Marvelous | <PROTECTED> |
| 01:11.50 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:12.02 | seraphie | Marvelous: you are talking about the Asterisk GUI. PLease see #asterisk-gui |
| 01:12.22 | carrar | <PROTECTED> |
| 01:12.54 | Marvelous | wait |
| 01:13.01 | carrar | no waiting |
| 01:13.03 | carrar | it's GO TIME |
| 01:15.50 | *** join/#asterisk guilhermebr (~Guilherme@189.63.44.75) |
| 01:15.54 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 01:16.17 | Marvelous | :d |
| 01:16.21 | Marvelous | okie |
| 01:16.24 | Marvelous | but the config is |
| 01:16.27 | Marvelous | http.conf |
| 01:16.40 | Marvelous | asterisk/http.conf |
| 01:24.17 | Marvelous | Starting asterisk: FD 32767 exceeds the maximum size of ast_fdset! |
| 01:25.01 | seraphie | Marvelous: Google first. |
| 01:25.27 | Marvelous | okie |
| 01:29.31 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 01:32.34 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 01:39.42 | *** join/#asterisk coppice (~chatzilla@m121-202-9-11.smartone-vodafone.com) |
| 01:48.54 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 02:00.40 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 02:02.32 | *** join/#asterisk Kumbang (~kumbang@180.245.175.76) |
| 02:11.47 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 02:14.02 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 02:15.42 | Marvelous | where i can get asterisk education |
| 02:19.06 | Freeaqingme | Marvelous, I think digium has some training programs |
| 02:19.20 | Freeaqingme | otherwise you could read ~thebook which is a great place to start |
| 02:19.23 | Marvelous | any online courses i could take !! |
| 02:19.24 | Freeaqingme | ~thebook |
| 02:19.24 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 02:34.12 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 02:39.03 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 03:01.57 | *** join/#asterisk sourcode (~code@ppp-115-87-231-86.revip4.asianet.co.th) |
| 03:17.25 | *** part/#asterisk jrrose (~Jon@207.98.203.74) |
| 03:21.13 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 03:22.38 | *** join/#asterisk rogersja1 (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 03:28.48 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 03:36.49 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 03:44.09 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 03:44.36 | *** join/#asterisk radic (~radic@dslb-178-002-230-138.pools.arcor-ip.net) |
| 03:47.22 | *** join/#asterisk pLr (~acer@unaffiliated/plr) |
| 04:00.50 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 04:08.25 | *** join/#asterisk m_tadeu (~quassel@89.180.67.188) |
| 04:12.17 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 04:14.35 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 04:15.23 | *** join/#asterisk trijezdci (~trijezdci@192.100.123.77) |
| 04:16.27 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.63.166) |
| 04:28.13 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 04:28.21 | l1nuxman | how can I add a new voicemail message to the system without actually calling and waiting for the voicemessage system? I want to trigger "New Message" |
| 04:42.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 04:47.09 | gruvfunk_afk | hey l1nuxman |
| 04:47.28 | gruvfunk_afk | so, you are probably currently using the VoiceMailMain aplication, yes? |
| 04:48.12 | gruvfunk_afk | check out the VoiceMail application instead for your needs - create a new extension like exten => 9999,1,VoiceMail |
| 04:51.41 | *** join/#asterisk celthunder (~celthunde@rofltech.net) |
| 04:53.17 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:07.33 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:14.32 | gruvfunk | anyone up for a question? please settle this argument for me: can Asterisk stream audio from a PHP URL address like http://player.streamtheworld.com/liveplayer.php?CALLSIGN=ARNCITY |
| 05:18.19 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 05:20.50 | l1nuxman | gruvfunk, why VoiceMail? Whats that gonna do for me? |
| 05:21.33 | gruvfunk | what were you trying to do? |
| 05:21.56 | gruvfunk | leave voicemail without ringing the extension? |
| 05:22.07 | l1nuxman | without physically using a phone |
| 05:22.40 | l1nuxman | I was thinking maybe I could just 'touch a.wav' |
| 05:22.49 | l1nuxman | I wanted to test voicemail attachment email |
| 05:25.31 | l1nuxman | goodnight, I'll check your message tomorrow. |
| 05:40.18 | gruvfunk | can AGI PHP somehow help to stream a java/flash radio station? |
| 05:40.32 | gruvfunk | me scratches his head |
| 05:40.53 | gruvfunk | scratches his head |
| 05:41.43 | kaldemar | ices has been used to stream audio with asterisk. |
| 05:43.08 | Sertys | there's no need for agi php |
| 05:43.23 | gruvfunk | do tell more |
| 05:43.54 | gruvfunk | can I stream this: http://player.streamtheworld.com/liveplayer.php?CALLSIGN=ARNCITY |
| 05:44.27 | ChannelZ | if it's outputting a suitable format you could using MOH |
| 05:44.54 | ChannelZ | that particular URL appears to be flash though so I'm guessing not. |
| 05:44.55 | Sertys | well, yeah, it's basiclly about custom MOH |
| 05:45.21 | gruvfunk | I've done custom MOH before using mp3 streams, m3u |
| 05:45.24 | Sertys | i can think that vlc might grab the rtp data itself |
| 05:45.34 | gruvfunk | but a java flash? |
| 05:45.42 | kaldemar | that kind of music cannot be streamed. |
| 05:45.46 | kaldemar | anywhere. |
| 05:46.10 | Sertys | kaldemar: u're talking bullshit here |
| 05:46.11 | ChannelZ | that bad? Rap? |
| 05:46.42 | gruvfunk | the content is not in question, but the format |
| 05:46.57 | ChannelZ | No, I'd say the content is pretty questionable |
| 05:46.57 | Sertys | well, flash streaming is basically RTMP |
| 05:47.12 | Sertys | which is asao rtp |
| 05:47.13 | kaldemar | Sertys: i was referring to the genre not being aline with my musical taste. :) |
| 05:47.22 | Sertys | kaldemar: lol |
| 05:47.27 | Sertys | i just tuned in to the radio |
| 05:47.36 | Sertys | and i kinda feel u right now |
| 05:47.41 | ChannelZ | I guess they have morning zoo's in India too |
| 05:47.57 | Sertys | india has it all :) |
| 05:48.31 | ChannelZ | They're inescabable |
| 05:48.36 | Sertys | http://wiki.videolan.org/Flash_Video |
| 05:48.38 | ChannelZ | This is why I quit listening to the radio years ago |
| 05:48.50 | ChannelZ | but alas |
| 05:48.53 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 05:48.53 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 05:51.19 | Sertys | yeah |
| 05:51.48 | kaldemar | https://issues.asterisk.org/view.php?id=15484 |
| 05:51.52 | Sertys | gruvfunk: u're prolly wasting precious time right now trying to figure this shit out |
| 05:52.00 | gruvfunk | yepper |
| 05:52.27 | gruvfunk | some customers want the world |
| 05:52.43 | Sertys | it's good if they pay for it |
| 05:52.54 | Sertys | i can say it's doable though |
| 05:53.02 | gruvfunk | yeah? |
| 05:53.21 | Sertys | wireshark the traffic |
| 05:53.26 | Sertys | get the rtmp server |
| 05:53.35 | Sertys | and try to stream off it with a player |
| 05:53.45 | gruvfunk | yeah but that's not a practical way to build a solution with 100's of radio stations |
| 05:53.50 | Sertys | and if vlc works, u can use it to write to stdout |
| 05:54.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 05:54.24 | Sertys | gruvfunk: u gotta figure the basics in order to write a proper wrapper with it |
| 05:54.40 | gruvfunk | curious: then what? stdin to Asterisk.. |
| 05:54.59 | Sertys | then just use moh |
| 05:55.20 | Sertys | i might thing eagi might do the trick with pipe streaming |
| 05:55.32 | Sertys | but it would eat up res |
| 05:55.47 | Sertys | if there're 2+ concurrent users :) |
| 05:56.09 | gruvfunk | yeah... |
| 05:56.11 | Sertys | i don't have enough time right now to test it out |
| 05:56.38 | gruvfunk | the better solution is for them to figure out a media transcoder that then feeds Asterisk a friendlier format |
| 05:57.28 | Sertys | that's what i'm trying to make u do |
| 05:57.40 | gruvfunk | right |
| 05:57.57 | Sertys | every player is basically a media transcoder... |
| 05:58.04 | Sertys | and viceversa |
| 05:58.31 | gruvfunk | but the resources would be significant for multiple callers listening in |
| 05:58.46 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 05:58.47 | gruvfunk | assuming different stations |
| 05:59.20 | gruvfunk | i'll put some more thought on it tomorrow, thanks for the leads though, gears are turning |
| 05:59.38 | Sertys | np |
| 05:59.41 | Sertys | gtg |
| 06:05.27 | WiretapSeven | anyone here screwed around with a 7970 long enough to get BLF working? |
| 06:05.36 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:10.44 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:15.14 | *** part/#asterisk gruvfunk (~chatzilla@76.15.41.136) |
| 06:17.52 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 06:21.58 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 06:26.43 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:26.47 | schmidts | good morning |
| 06:40.15 | celthunder | morning schmidts |
| 06:42.01 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 06:46.26 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 06:51.09 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 06:55.49 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:56.19 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:03.31 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:04.07 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:04.15 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 07:04.53 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 07:09.07 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:13.46 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:19.04 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 07:24.30 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:32.59 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 07:33.25 | *** join/#asterisk mpe (~mpe@94.127.50.104) |
| 07:45.52 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 07:51.22 | *** join/#asterisk jizzzum6 (jizzzum6@2001:1938:232::100) |
| 07:57.27 | *** join/#asterisk teathsch (~chatzilla@ip68-5-51-145.oc.oc.cox.net) |
| 08:00.35 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:04.34 | *** join/#asterisk doneir (~cbrunker@203.206.237.186) |
| 08:05.07 | doneir | i've been goolging for an answer, but am still unsure - do the dahdi drivers (and cards i guess) support unframed e1 pri connections? |
| 08:06.06 | coppice | if they are unframed, they are not PRI |
| 08:06.24 | doneir | i have a pri setup, doesn't seem to be working, and am troubleshooting possibilities. Seems the only configuration of the line provisioning I can get shows: E1 mode, HDB3 line coding =, loopback disable, tsoffset disable, clock loop timing and.... Framer mode 4.unframe E1 |
| 08:07.31 | doneir | hmm... unframed mode would be BRI setup then or such? |
| 08:08.02 | coppice | unframed would be unframed |
| 08:08.59 | doneir | sorry, somewhat confused - i take it a 'framed' layout would then contain timing information and the like, which when you get a Blue Alarm it means you're receiving unframed 1s on the line (and no additional information, since the information can not be given outside of a frame?) |
| 08:16.07 | doneir | okay, so bottom line unframed == BRI, framed/fractional == PRI compatable? |
| 08:16.52 | doneir | i had thought some juniper and cisco cards could support telephony over unframed e1 connections, though probably mixed up with network (bri) based |
| 08:17.02 | coppice | unframed == HDLC/data, framed == PRI |
| 08:20.48 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 08:25.55 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 08:30.04 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:30.30 | Chainsaw | Okay, IPv6 to issues.asterisk.org is back up. Good to see. |
| 08:32.33 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:41.30 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59) |
| 08:43.47 | *** join/#asterisk sourcode (~code@ppp-115-87-231-86.revip4.asianet.co.th) |
| 08:44.55 | *** join/#asterisk Howlader (Howlader@soft.bdcom.com) |
| 08:45.10 | sezuan | Should chan_gtalk (* 1.8.3.x) work with the voice/talk plugin + googlemail? I think I've configured it properly, but it only works with the 'real' client. I tried it with googlemail + voice plugin on linux(firefox/googlechrome), mac(googlechrome), Windows 7(IE8). Signaling from * to the client works, when I accept the call * still rings, the call won't be established. When I call in the opposite direction, * shows no reaction. |
| 08:54.13 | *** join/#asterisk Pieplay (~pieter@ip-83-134-139-147.dsl.scarlet.be) |
| 08:54.44 | asterisk-learner | hi, is there a way to stop the execution of WaitForSilence() application while running on a channle without hanging up the channel and before timeout ? |
| 08:56.14 | asterisk-learner | .join freepbx |
| 08:56.46 | sxpert | asterisk-learner: use a shorter timeout, and use the iterations option |
| 08:57.32 | sxpert | ah. no, can't do that... |
| 08:58.06 | sxpert | use shorter timeout in a loop |
| 08:59.07 | asterisk-learner | sxpert: I want to wait for 5 sec of silence, i am already using : 5,1,0 |
| 08:59.13 | asterisk-learner | wait 5 sec 1 time, for 0 timeout |
| 08:59.45 | sxpert | 0 timeout probably means "infinite" |
| 09:00.50 | sxpert | so you may want to use 5,1,5 |
| 09:00.58 | sxpert | or something |
| 09:01.15 | asterisk-learner | 5 ? |
| 09:01.27 | asterisk-learner | wait let me explain to you my scenario better |
| 09:01.47 | Pieplay | >0 not 0 |
| 09:02.16 | sxpert | besides, if you want 5 seconds of silence, it should be (5000,1,5) or something |
| 09:02.36 | sxpert | doc says "noiserequired" in milliseconds |
| 09:02.46 | asterisk-learner | sxpert: yeah, actually i am using 5000,1,0 |
| 09:03.28 | sxpert | but do explain your scenario ); |
| 09:03.33 | asterisk-learner | I am writing an application that gets called from the dialplan, when it executes (let's call it thread 1) it creates another thread (thread 2) |
| 09:04.09 | asterisk-learner | in thread 2 i am calling application "WaitForSilence" on the same channel, and in thread 1 i am looping over and testing channel variables |
| 09:04.40 | asterisk-learner | is there a way to notify thread 2 to stop executing WaitForSilence and end from thread 1 ? |
| 09:05.16 | asterisk-learner | because as far as i know pbx_exec() is blocking and wont return before the application returns.... |
| 09:05.56 | *** join/#asterisk jkroon (~jkroon@dsl-241-232-230.telkomadsl.co.za) |
| 09:06.04 | sxpert | how about looping over the channel variables only. it seems the waitforsilence is unneeded |
| 09:08.01 | asterisk-learner | sxpert: No, i need to run WaitForSilence in parrallel and depending on its returning values, i am setting other channel variables that gets read in thread 1 |
| 09:08.03 | *** join/#asterisk N101 (~pieter@ip-83-134-139-147.dsl.scarlet.be) |
| 09:08.19 | sxpert | asterisk-learner: ah... |
| 09:09.08 | sxpert | I actually fail to see the need of threads in that case ;) |
| 09:09.48 | N101 | Q: I did a clean install of astrisk via the repository, followed the tutorial on wiki.astrisk.org on creating Sip account, and now I can't get it to connect |
| 09:10.32 | N101 | Did I forget something? |
| 09:12.08 | asterisk-learner | sxpert: I am communicating with another server in thread1 and getting data from it, if i dont use thread2, thread1 will be blocked by WaitForSilence()... |
| 09:12.14 | kaldemar | N101: what kind of an account? |
| 09:12.23 | sxpert | asterisk-learner: ah yeah, ok, I get it now |
| 09:12.52 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:13.04 | asterisk-learner | sxpert: :-), so is there an easy way to do it without modifying WaitForSilence() code .... |
| 09:13.06 | sxpert | asterisk-learner: however, you really need the 5 secs of silence |
| 09:13.21 | sxpert | from the other side |
| 09:13.23 | N101 | A sip account by editting the sip.conf file (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) |
| 09:14.19 | asterisk-learner | sxpert: ??? |
| 09:14.29 | kaldemar | N101: what do you see in the CLI when the client tries to register or make a call? with verbosity enabled, e.g. "core set verbose 10" |
| 09:15.03 | sxpert | asterisk-learner: if you're using WaitForSilence, it's because you expect the line to be silent for 5 seconds |
| 09:15.40 | sxpert | well, you're actually waiting for signal to stop, then 5 seconds of silence |
| 09:17.20 | asterisk-learner | sxpert: In thread 2 I am probing to see if i have 5 sec of silence, but thread1 might stop this processing after 3 sec, or if i do have 5 sec, thread2 will end and notify thread1 that it found 5 sec to do some processing... |
| 09:20.02 | sxpert | hmm. seems like you can't stop WaitForSilence mid-check |
| 09:20.28 | sxpert | you'd have to create some loop or something |
| 09:20.40 | sxpert | like waitfor silence for 1 second |
| 09:20.51 | sxpert | then wait for noise with 1 second timeout in a loop |
| 09:20.56 | asterisk-learner | sxpert: or modify its internal for(;;) loop |
| 09:21.17 | sxpert | and count the seconds in a variable |
| 09:23.06 | asterisk-learner | sxpert: ok thx for the info, i will try it now, but i thought there might be an option/parameter that can stop WaitForSilence mid-check |
| 09:23.08 | N101 | kaldemar: not much when i get the account credentials wrong on purpose it goes nuts on local host, but not on lan. when I get it right on lan it does not connect what is normal because localhost is not a permited ip, |
| 09:23.35 | sxpert | asterisk-learner: doesn't seem to exist. guess you could add that feature though ;) |
| 09:24.11 | asterisk-learner | sxpert: but i feel my scenario is really weird and rare... :-P |
| 09:24.23 | N101 | and eventually if I loop it trough my lan adress with the right credentials the client returns '' failed to register, method not allowed" |
| 09:24.31 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 09:25.57 | sxpert | asterisk-learner: agreed. so the better way for you is to hack up what you need out of existing stuff |
| 09:30.35 | Exten | is there a log i can see why asterisk falls everytime i start an AGI script ? |
| 09:30.41 | Exten | it just disconnects from server |
| 09:30.42 | Exten | " |
| 09:36.08 | kaldemar | N101: what does "not much" mean? nothing at all or do you get some message? |
| 09:37.17 | N101 | yes quite |
| 09:38.50 | *** join/#asterisk DennisG (~DennisG@541E88D0.cm-5-7c.dynamic.ziggo.nl) |
| 09:48.06 | *** part/#asterisk Pieplay (~pieter@ip-83-134-139-147.dsl.scarlet.be) |
| 09:49.01 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 09:54.47 | ferdna | how do you restart asterisk? |
| 09:54.51 | ferdna | asterisk -r |
| 09:54.58 | ferdna | CLI>restart now |
| 09:54.59 | ferdna | ? |
| 09:57.17 | kaldemar | core restart now in CLI or with init script. |
| 09:58.19 | ferdna | kaldemar, thanks |
| 10:13.12 | ferdna | register => <phonenumber>@sip.broadvoice.com:<password>:<phonenumber>@sip.broadvoice.com/<extension> |
| 10:13.19 | ferdna | is this correct? |
| 10:13.34 | ferdna | isnt phonenumber:password@domain.com |
| 10:13.36 | ferdna | ? |
| 10:14.11 | ferdna | register => username:password@my.service-provider.tld |
| 10:53.09 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:58.39 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 11:00.19 | ixyd_ | hi guys, is it possible to use Queue()`s gosub parameter with arguments, like Dial()`s U() parameter? |
| 11:02.55 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 11:07.08 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 11:14.59 | *** join/#asterisk devil_evoxxx (~d3v1l@157.27.182.73) |
| 11:15.02 | devil_evoxxx | hi guys |
| 11:17.05 | devil_evoxxx | i'have a asterisk machine ( asterisk 1.4.37 ) with 600 peers friends. When there are about 18/20 channels open the load of server is near 1.20 ( host machine is a debian lenny ). |
| 11:17.20 | devil_evoxxx | Asterisk run as single process or it have thread? |
| 11:17.52 | asterisk-learner | devil_evoxxx: asterisk is multi threaded |
| 11:18.48 | devil_evoxxx | there is anyway to reduce the load of the machine ? Or some tricks to have better performance? |
| 11:20.29 | devil_evoxxx | The machine is a dual Core Xeon HT 3.4 Ghz , 4 GB ram |
| 11:28.18 | *** join/#asterisk coppice (~chatzilla@m121-203-199-169.smartone-vodafone.com) |
| 11:32.10 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 11:35.08 | WiretapSeven | devil_evoxxx, more cores |
| 11:37.10 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 11:43.11 | *** join/#asterisk qCake (~qBaLL@41.160.153.78) |
| 11:43.45 | qCake | hi |
| 11:44.34 | qCake | anybody here? |
| 11:50.49 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 11:55.42 | tuxx- | guys, does the variable for a siptrunk `outgoinglimit` exist? I got some old configuration of siptrunk here, but i doubt this variable even does anything, cant find anything on google either, only for IAX peers it seems. |
| 11:56.58 | kaldemar | tuxx-: what version are you using? |
| 11:57.19 | tuxx- | the config is from an asterisk 1.4 |
| 11:57.37 | tuxx- | dont have the actual asterisk running though, the box is long gone. Just trying to figure out what these options are for. |
| 11:57.55 | tuxx- | but afaict they dont do anything :P |
| 11:58.50 | kaldemar | outgoinglimit does not exist in 1.4, 1.6.2 nor 1.8. |
| 12:03.09 | tuxx- | right, tnx :) |
| 12:05.56 | qCake | i've got a bit of a problem. I have a * server running v1.4.36 from which I register to another SIP provider for outbound calls. Another client connecting to my server using elastix (version not known) uses my server to make outbound calls. All is working well until the client running elastix phones any number that doesn't ring (ie IVR's, Phones going directly to voicemail, Auto Attendants, |
| 12:05.57 | qCake | Music on Hold etc.) there are no voice packets sent to the client. They have another SIP account registered using the same settings as what they are to connect to our server which is working so it is not being blocked on the firewall or natting issue...any ideas on what this could be? |
| 12:08.36 | *** join/#asterisk mawhiii (~trav@tcmsdev.com) |
| 12:09.39 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 12:13.03 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:14.55 | qCake | i've got a bit of a problem. I have a * server running v1.4.36 from which I register to another SIP provider for outbound calls. Another client connecting to my server using elastix (version not known) uses my server to make outbound calls. All is working well until the client running elastix phones any number that doesn't ring (ie IVR's, Phones going directly to voicemail, Auto Attendants, |
| 12:14.56 | qCake | Music on Hold etc.) there are no voice packets sent to the client. They have another SIP account registered using the same settings as what they are to connect to our server which is working so it is not being blocked on the firewall or natting issue...any ideas on what this could be? |
| 12:22.10 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:25.02 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 12:29.11 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 12:30.27 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 12:30.44 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:31.08 | qCake | i've got a bit of a problem. I have a * server running v1.4.36 from which I register to another SIP provider for outbound calls. Another client connecting to my server using elastix (version not known) uses my server to make outbound calls. All is working well until the client running elastix phones any number that doesn't ring (ie IVR's, Phones going directly to voicemail, Auto Attendants, |
| 12:31.09 | qCake | Music on Hold etc.) there are no voice packets sent to the client. They have another SIP account registered using the same settings as what they are to connect to our server which is working so it is not being blocked on the firewall or natting issue...any ideas on what this could be? |
| 12:34.43 | *** join/#asterisk rymkus (~pcccp@mx.okhtaform.ru) |
| 12:34.54 | rymkus | hi everybody! |
| 12:35.37 | rymkus | I'm use next dialplan construction |
| 12:35.37 | rymkus | <PROTECTED> |
| 12:35.37 | rymkus | <PROTECTED> |
| 12:37.12 | rymkus | Can anyone tell me how can I stop dialplan execution on the first priority if pickup was successfull? |
| 12:37.57 | *** join/#asterisk DaneoShiga (~danilo.sh@187.39.191.252) |
| 12:40.29 | ixyd_ | @rymkus i would expect asterisk to auto stop on successful pickup? |
| 12:42.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:42.10 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:48.02 | rymkus | @ixyd_ yes, you understood my poor english correctly =) |
| 12:48.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:48.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:49.28 | ixyd_ | @rymkus which version of asterisk do you use? iam pretty sure i have some similiar extensions on a 1.4 based setup which works as expected...but i'll take a look...moment pls :) |
| 12:49.37 | wolfe | ixyd_: this is IRC, not twitter :) |
| 12:49.41 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-kozyoiqtaxrtzchr) |
| 12:49.41 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 12:49.57 | Sertys | wolfe: lol |
| 12:50.05 | Sertys | g point @wolfe :) |
| 12:51.03 | ixyd_ | hehe ah sure ;) |
| 12:51.13 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 12:51.15 | rymkus | ixyd_ 1.8.3.3 with pickup deadlock diff applied |
| 12:51.16 | wolfe | silly Twatter users |
| 12:51.36 | *** join/#asterisk m_tadeu (~quassel@89-180-67-188.net.novis.pt) |
| 12:51.38 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:52.08 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 12:52.32 | qCake | anybody: i've got a bit of a problem. I have a * server running v1.4.36 from which I register to another SIP provider for outbound calls. Another client connecting to my server using elastix (version not known) uses my server to make outbound calls. All is working well until the client running elastix phones any number that doesn't ring (ie IVR's, Phones going directly to voicemail, Auto |
| 12:52.32 | qCake | Attendants, Music on Hold etc.) there are no voice packets sent to the client. They have another SIP account registered using the same settings as what they are to connect to our server which is working so it is not being blocked on the firewall or natting issue...any ideas on what this could be? |
| 12:52.47 | leifmadsen | on IRC using an @ in the front signifies a channel op :) |
| 12:53.22 | rymkus | hadn't used IRC for couple years =) |
| 12:53.34 | ixyd_ | dito :) |
| 12:54.52 | rymkus | back to my problem - does pickup application return something that can be used in expression for gotoif func? |
| 12:55.49 | rymkus | I tried to use the i priority, but no luck |
| 12:57.43 | ixyd_ | @rymkus my setup is different, using ael and asterisk 1.4 ... :-/ |
| 13:00.28 | *** join/#asterisk ketema (~ketema@ketema.net) |
| 13:00.58 | *** part/#asterisk ketema (~ketema@ketema.net) |
| 13:01.48 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 13:02.57 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 13:03.08 | rymkus | anyone? |
| 13:04.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:07.14 | *** join/#asterisk serafie (~erin@nat/digium/x-vrxlwcioldmkkrus) |
| 13:15.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:18.16 | malcolmd | rymkus: not that i see, no. |
| 13:18.32 | malcolmd | certainly, it's not documented...https://wiki.asterisk.org/wiki/display/AST/Application_Pickup |
| 13:25.41 | qCake | anybody: i've got a bit of a problem. I have a * server running v1.4.36 from which I register to another SIP provider for outbound calls. Another client connecting to my server using elastix (version not known) uses my server to make outbound calls. All is working well until the client running elastix phones any number that doesn't ring (ie IVR's, Phones going directly to voicemail, Auto |
| 13:25.41 | qCake | Attendants, Music on Hold etc.) there are no voice packets sent to the client. They have another SIP account registered using the same settings as what they are to connect to our server which is working so it is not being blocked on the firewall or natting issue...any ideas on what this could be? (If I register a IP phone from another site it works fine and if I register a softphone on my |
| 13:25.42 | qCake | notebook it also works fine) |
| 13:26.06 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 13:26.24 | Dovid | does a dual processor machine help asterisk or it only uses one CPU ? |
| 13:27.05 | sunfone | Dovid: IMO by the time you need a second processor you should probably be thinking about another box |
| 13:27.42 | kaldemar | rymkus: if you're using a version with COLP, you could try to check the connected line information with func CONNECTEDLINE before and after the pickup. |
| 13:27.56 | *** join/#asterisk IamTrying (~IamTrying@d51A42D64.access.telenet.be) |
| 13:28.54 | Dovid | sunfone: My question is if it will help (like with transcoding etc.) also there are custom scripts on the machine. quesiton is should i get 1 cpu thats better or 2 less quality |
| 13:28.56 | IamTrying | Does Asterisk supports: 1) Audio: G722, Video: H.264, Presentation: H.239 2) All over SIP ? |
| 13:29.58 | Dovid | IamTrying: 1) Yes 2) Don't recall but supports one of em, dont think supports H.239 |
| 13:30.08 | Dovid | one of the vid codecs |
| 13:31.25 | IamTrying | Dovid, so no H.239 supported for SIP? |
| 13:31.38 | sunfone | Dovid: I don't believe so - asterisk is multi-threaded, but single process. For transcoding you might consider the Digium card. When I architect such things I try to keep external code running on external machines (fastagi), and let the asterisk server do what it does best |
| 13:32.31 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 13:33.15 | leifmadsen | no H.239 |
| 13:33.36 | leifmadsen | H.264 and H.263 yes |
| 13:34.06 | sunfone | Anyone using cacti to monitor call volume? Point me to some templates? |
| 13:34.18 | leifmadsen | sunfone: no, just use opennms for that |
| 13:34.28 | coppice | H.239 is the multi-camera protocol, isn't it? |
| 13:34.35 | sunfone | hmm haven't heard of it... will look at it thanks |
| 13:34.40 | IamTrying | leifmadsen, what else can be a solution to do get h.239 working? any idea or thoughts/tips.. |
| 13:35.20 | leifmadsen | IamTrying: you could write a patch to provide the functionality? |
| 13:35.23 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:36.28 | leifmadsen | sunfone: I wrote about it in the monitoring chapter of Asterisk the definitive guide as well -- works quite well |
| 13:36.30 | Dovid | sunfone: using proprietary software that has agi's on the box. |
| 13:37.30 | *** join/#asterisk rogersja (~RogersJa@S0106001346faf79f.gv.shawcable.net) |
| 13:39.50 | IamTrying | leifmadsen, if bypass media is used, is it really necessary to have h.239 supports? |
| 13:40.01 | sunfone | Dovid: bummer :) But perhaps your AGI's, since they are spawned as separate processes, will take advantage of your extra CPUs |
| 13:40.12 | leifmadsen | IamTrying: I know nothing about h.239 |
| 13:40.19 | sunfone | leifmadsen: I have yet to checkout your book - keep meaning to do so :) |
| 13:40.37 | IamTrying | leifmadsen, its all about SDP file, bypass media should do already those in peer 2 peer |
| 13:40.37 | sunfone | Looking at the OpenNMS demo now... pretty impressive! |
| 13:41.10 | *** join/#asterisk bmcspadd (~bmcspadd@198.117.194.51) |
| 13:44.34 | sunfone | I just wish it wasn't java based (sigh). Java seems to make everything all that more complicated and takes so much underlying resources |
| 13:46.22 | leifmadsen | sunfone: ya I just started playing around with it some more -- I might start using it to monitor some customer boxes as well via SNMPv3 then enable the notification stuff |
| 13:46.40 | leifmadsen | sunfone: ya, I run it inside a VM to try and limit the amount of resources it can use |
| 13:47.09 | sunfone | To date all of our monitoring is hand-tooled mrtg :) |
| 13:47.43 | sunfone | Just installed cacti the other day and have been very impressed, mainly because of the wealth of pre-built templates for stuff... our underlying infrastructure is all wireless mikrotik |
| 13:47.54 | sunfone | And it hardly uses any resources |
| 13:48.14 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 13:48.15 | rymkus | kaldemar: isn't COLP QSIG and PRI related feature? I use only SIP and IAX lines. |
| 13:48.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:49.26 | Chainsaw | I have turned off "URL dialling" on Polycom 670 phones using FEATURES. They respect that for normal calls. However, on Transfer, they like to default to URL dialling regardless. Has anyone here managed to persuade those handsets otherwise? |
| 13:52.04 | *** join/#asterisk nite613 (~chris@CPE001839c16d35-CM00237453c586.cpe.net.cable.rogers.com) |
| 13:53.49 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-tkckienqtysdcrsx) |
| 13:54.09 | nite613 | Hi guys. * 1.4 here. Looking for a way to manually and gracefully hang up a call that never got a BYE message (I suppose it was dropped or never generated by provider, not sure). There is an executing AGI script doing a recording and I'd like it to end gracefully as if it were a normal hangup |
| 13:54.22 | nite613 | I've considered crafting and injecting the SIP packet, but checking for an easier way ;) |
| 13:55.05 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 13:55.10 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:56.02 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:56.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:57.42 | leifmadsen | nite613: soft hangup? |
| 13:58.31 | *** join/#asterisk e7e5 (~rudenko@77.239.239.27) |
| 14:00.35 | nite613 | leifmadsen: what do you mean? |
| 14:02.08 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 14:02.16 | leifmadsen | nite613: there is an asterisk CLI command called 'soft hangup' in 1.4 that you could use to request the channel be hun gup |
| 14:03.00 | nite613 | Woohoo! :) Thank you, that seems to be exactly what I was looking for! |
| 14:03.23 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 14:04.54 | _Corey_ | nite613: you might also look into 'rtptimeout' |
| 14:07.34 | nite613 | Corey, don't talk to me about rtptimeout ;) That's a whole 'nother issue I'm fighting |
| 14:07.40 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 14:08.08 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 14:08.16 | _Corey_ | lol |
| 14:08.20 | nite613 | I'm supporting a dictation system so most of the time audio is just one way, and I'm having a hell of a time getting RTP packets to be generated during silence while in the Dictate() application. It's not in the source, but adding it hasn't been as easy as it looked |
| 14:08.30 | leifmadsen | nite613: you can probably access that command (or a similar hangup command) from the Asterisk manager interface as well |
| 14:08.56 | nite613 | Thanks again, leifmadsen, the CLI command will work fine for me |
| 14:09.22 | leifmadsen | sounds good |
| 14:11.49 | *** join/#asterisk kdekorte (~kdekorte@63-163-137-113.iprev.kci.net) |
| 14:15.22 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:16.47 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 14:19.59 | *** part/#asterisk IamTrying (~IamTrying@d51A42D64.access.telenet.be) |
| 14:23.59 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:29.02 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 14:29.37 | *** join/#asterisk mclaro (~mclaro@190.183.222.194) |
| 14:31.14 | *** join/#asterisk arnotixe (~arnotixe@cl-205.udi-01.br.sixxs.net) |
| 14:33.48 | Aut0ExeC | hi guys anyone here use voip.ms ? |
| 14:35.14 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 14:35.20 | psilikon | Aut0ExeC, yeah I use them |
| 14:35.50 | Exten | yip |
| 14:36.26 | *** join/#asterisk arnotixe (~arnotixe@cl-205.udi-01.br.sixxs.net) |
| 14:39.34 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:40.53 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:52.07 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:54.16 | m4xx | when compiling on freebsd i'm getting http://paste2.org/p/1398829 |
| 14:54.25 | *** part/#asterisk nite613 (~chris@CPE001839c16d35-CM00237453c586.cpe.net.cable.rogers.com) |
| 14:58.11 | Aut0ExeC | psilikon: nice... thats like prepaid right? |
| 14:58.20 | Aut0ExeC | psilikon: with roll over? |
| 14:58.38 | m4xx | i've been using didforsale.com, seems ok so far |
| 14:59.44 | psilikon | Aut0ExeC, yes |
| 15:02.02 | m4xx | ooops perhaps i should install dahdi ;[ |
| 15:02.17 | *** join/#asterisk gruvfunk (~chatzilla@user-160uac8.cable.mindspring.com) |
| 15:02.49 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:09.20 | *** join/#asterisk pierre` (~pierre@87.98.140.216) |
| 15:12.04 | *** join/#asterisk sled-dog (~luser@65.124.95.55) |
| 15:15.54 | Aut0ExeC | psilikon: thanks bro |
| 15:16.15 | Aut0ExeC | psilikon: only thing is I see there is I think a 25 dolloar minimum |
| 15:16.35 | Aut0ExeC | psilikon: but considering that you can roll over year after year... I guess I will never loose it |
| 15:16.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:16.59 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 15:20.49 | *** join/#asterisk andyoutside (6161c2a5@gateway/web/freenode/ip.97.97.194.165) |
| 15:21.43 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 15:21.49 | andyoutside | Has anyone noticed in 1.8.3.3 that when you park someone you do not hear wherethey are parked to |
| 15:22.46 | gruvfunk | hey Aut0ExeC did you finish out that setup? |
| 15:22.57 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:25.18 | *** join/#asterisk andyoutside_ (6161c2a5@gateway/web/freenode/ip.97.97.194.165) |
| 15:25.30 | emora | Anyone here working with Adhearsion? |
| 15:26.03 | andyoutside_ | and if you park someone in the building they will hear the number they are parked at. |
| 15:26.14 | andyoutside_ | I am not sure about if it is outside thte building |
| 15:27.04 | *** join/#asterisk hrnkc (9ec15455@gateway/web/freenode/ip.158.193.84.85) |
| 15:29.00 | *** join/#asterisk hfb (~hfb@pool-96-247-114-211.lsanca.dsl-w.verizon.net) |
| 15:31.29 | hrnkc | Hi there! Would you be so kind to figure out my problem? I can't figure why is Asterisk trying to send NOTIFY every time I unREGISTER...it doesn't make sense. It should send it after i make a registration, or am I wrong? Where can I find the module, which sends the NOTIFY messages to nofity client about voicemails? Because |
| 15:32.24 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:34.18 | hrnkc | The NOTIFY message in my situation is supposed to send information about new voicemails. But it does it vice versa. Sends the NOTIFY when I unREGISTER and when I REGISTER, nothing happens. If i try to make a call, NOTIFY is sent. |
| 15:39.45 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 15:43.22 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 15:43.36 | *** join/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 15:44.27 | epart | does anybody here knows GoAutoDial |
| 15:44.53 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 15:45.15 | epart | does anybody here knows GoAutoDial please help |
| 15:46.16 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 15:46.40 | zkn | Hi, does anyone know if there a way to achieve the same as "alternateexts" in users.conf without using users.conf ? |
| 15:47.42 | zkn | users.conf now sports an optional alternateexts property, which permits allocation of additional extensions which will reach the specified user. |
| 15:49.00 | leifmadsen | sounds like a dialplan thing |
| 15:49.02 | *** join/#asterisk porche (~kursad@212.253.24.82) |
| 15:49.09 | leifmadsen | that's the traditional way to handle that |
| 15:49.20 | zkn | i want to be able to dynamically achieve this sort of feature |
| 15:49.43 | angryuser | epart, depends, speak |
| 15:50.39 | emora | Adhearsion anyone? |
| 15:51.35 | epart | just any idea.. that about goatodial.. |
| 15:53.17 | zkn | haven't been able to accompish this with regexten / regcontext in sip.conf |
| 15:56.12 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:57.01 | *** join/#asterisk InsektO (~InsektO@janus.all-kom.com.ar) |
| 16:03.40 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:08.17 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 16:08.49 | InsektO | hi, im having issues with an hardware echo cancel card from digium (VPMADT032), i get "Unable to ping the DSP" in dmesg, anyone had a similar issue? |
| 16:09.19 | leifmadsen | InsektO: hardware issues for Digium hardware should be brought up with Digium tech support |
| 16:09.28 | leifmadsen | they are the ones who are able to support those products |
| 16:12.48 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 16:12.54 | Eitan | i have just bought the book |
| 16:12.54 | Eitan | oh man |
| 16:14.33 | carrar | w00t |
| 16:14.36 | *** part/#asterisk dan__t (~dant@72.233.89.95) |
| 16:18.19 | billmania | I'm looking for some help with asterisk terminology. What is the asterisk term for the collection of all of the audio conversations associated with an inbound call from an external system? For example, an inbound call may have first been routed to a queue, then answered by one extension and then transferred to yet another extension. |
| 16:18.57 | carrar | inbound path? |
| 16:19.13 | carrar | path of a call? |
| 16:19.22 | carrar | call route |
| 16:19.35 | leifmadsen | billmania: channel |
| 16:19.50 | andyoutside | dialing plan? |
| 16:19.52 | leifmadsen | there is not really any term for the collection of audio conversations |
| 16:20.07 | billmania | carrar: That'll work for me. Does the path or route or channel then have a unique identifier, which can be used to identify all of the component CDRs? |
| 16:20.10 | carrar | bucket 'O' Channels |
| 16:20.26 | InsektO | leifmadsen, yeah, i know that, but perhaps someone had that issue, and solved it by some sort of configuring in asterisk/dahdi |
| 16:20.46 | billmania | I understand from http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql that the "uniqueid" isn't unique at all. |
| 16:21.24 | leifmadsen | billmania: it's not globally unique, no |
| 16:21.54 | billmania | Is there always a one-to-one correspondence between a "channel" and a CDR? Or, can one channel have multiple associated CDRs for the same caller ID? |
| 16:23.39 | leifmadsen | well callerid is certainly not unique |
| 16:24.02 | leifmadsen | you can do things like "forkCDR" and such that would fork the CDR at various points in time |
| 16:24.18 | billmania | leifmadsen: Caller ID for a single inbound call can change during the course of the call? |
| 16:27.27 | fauxalliance | billmania, http://lists.digium.com/pipermail/asterisk-bugs/2010-July/082244.html |
| 16:29.43 | billmania | fauxalliance: That gives me the idea for my next question. Is there a drawing which describes the flow of call through asterisk, identifying the channels and applications and extensions? That would be most useful in reducing my ignorance. |
| 16:30.13 | carrar | Let us know when you complete that drawing so we can see i |
| 16:30.14 | carrar | t |
| 16:30.39 | billmania | carrar: That's what I thought. :-( |
| 16:31.13 | fauxalliance | ~book @ billmania |
| 16:31.25 | fauxalliance | ~thebook |
| 16:31.25 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 16:32.13 | billmania | fauxalliance: I have an electronic copy of "Asterisk: The future of telephony" 2nd edition. Am I hopelessly out of date? |
| 16:32.17 | fauxalliance | billmania, http://callflow.sourceforge.net/ just add a capture |
| 16:32.50 | fauxalliance | billmania, i'd borrow a new copy and make a small donation to the author |
| 16:32.57 | billmania | Interestingly, written by someone named Leif Madsen. |
| 16:35.09 | billmania | Prepares to join the asterisk priesthood by studying the holy book of asterisk |
| 16:37.12 | fauxalliance | billmania, fwiw, it's the only way in |
| 16:37.21 | *** join/#asterisk wonderworld (~ww@port-92-201-173-210.dynamic.qsc.de) |
| 16:37.39 | billmania | :-D Thanks. |
| 16:38.05 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 16:38.44 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:41.17 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:43.28 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:44.11 | *** join/#asterisk gruvfunk (~chatzilla@user-160uac8.cable.mindspring.com) |
| 16:44.43 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 16:48.07 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 16:50.27 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:50.36 | cj | carrar: I did? what list? |
| 16:50.50 | cj | carrar: I thought I was already famous...? |
| 16:51.26 | carrar | six list |
| 16:51.30 | carrar | of attendies |
| 16:51.42 | cj | oh, nice. Just under the wire on that one. |
| 16:52.12 | cj | now all I need is an AS number, I guess. |
| 16:52.26 | cj | oh, and a direct allocation would probably be helpful. |
| 16:54.56 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:59.40 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:00.03 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:00.44 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:04.14 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 17:04.51 | *** join/#asterisk sourcode (~code@ppp-115-87-231-86.revip4.asianet.co.th) |
| 17:05.44 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:05.57 | *** join/#asterisk hrnkc_ (9ec15455@gateway/web/freenode/ip.158.193.84.85) |
| 17:06.35 | hrnkc_ | Hi, is it possible to turn off NOTIFY messages that are sent where user has voicemail? |
| 17:08.23 | *** part/#asterisk draeath (~draeath@unaffiliated/draeath) |
| 17:08.45 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 17:09.14 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-225.mobile.uci.edu) |
| 17:10.42 | Eitan | so i just purchased "the book" and i im just verifying, if i get my t1 lines with 20 or so trumps using SIP. i am not going to require any of those digium or telefony cards correct? |
| 17:10.47 | Eitan | i would just go streight into nic? |
| 17:12.12 | leifmadsen | Eitan: well your terminology is a little all over the place -- if it is SIP, it's not a T1 |
| 17:12.22 | Eitan | <PROTECTED> |
| 17:12.23 | Eitan | lol |
| 17:12.34 | leifmadsen | you are correct in that you can get 20 channels of SIP from a provider, which runs over the network (internet) |
| 17:12.35 | Eitan | got it, t1 would then need the pri cards |
| 17:12.45 | leifmadsen | a T1 is a type of circuit |
| 17:13.15 | Eitan | ok, reading chapter 7 right now... right my understanding was that SIP could be sent over any internet connection |
| 17:13.27 | Eitan | it just so happens i was looking to get t1s to do this, cause thats the only thing avaiable in our area |
| 17:13.44 | leifmadsen | that's fine, you would be using a data T1 that would go into your router or whatnot |
| 17:14.07 | Eitan | got it :) |
| 17:14.09 | leifmadsen | then it'd just be an ethernet connection locally to your router, and asterisk would speak over that |
| 17:14.11 | Eitan | thanks for the clarification |
| 17:15.32 | Eitan | so would it be accurate to say that SIP is more up to date technologically. or perhaps where the future of VOIP is going? |
| 17:15.48 | leifmadsen | SIP is VoIP |
| 17:15.50 | leifmadsen | a T1 is not VoIP |
| 17:15.54 | Eitan | ok |
| 17:15.55 | Eitan | :) |
| 17:16.00 | leifmadsen | T1 is a traditional telephony connection |
| 17:16.04 | Eitan | i c |
| 17:17.06 | Eitan | just got confused on chapter 7 |
| 17:17.20 | leifmadsen | you should find a starter book on telephony terms and how networks work. The Newton Telecom Dictionary would be useful as well. |
| 17:17.52 | Eitan | ok, i will pick that up, i was told to purchase asterisk the definitive guide to get started |
| 17:17.53 | Eitan | so i did |
| 17:18.01 | Eitan | now i know what my next purchase is |
| 17:18.36 | leifmadsen | it is the definitive guide for asterisk, but not the definitive guide for telephony :) |
| 17:18.54 | Eitan | :) |
| 17:19.05 | Eitan | i got myself a good amount of reading to do |
| 17:19.11 | Freeaqingme | Eitan, voip is not a replacement for traditional telephony |
| 17:19.41 | Freeaqingme | there will always be some sort of ATM network |
| 17:20.08 | Eitan | would it be wise though to developt a call center strictly using voip? |
| 17:20.18 | Freeaqingme | sure |
| 17:20.35 | Eitan | smiles |
| 17:20.35 | Freeaqingme | depending on your location and size it may not be the most cost effective option to use a voip trunk |
| 17:20.36 | Eitan | ok |
| 17:21.01 | Eitan | the hardware costs, as far as my understanding are much lower with voip |
| 17:21.10 | Eitan | especially because we already have the servers |
| 17:21.12 | Freeaqingme | yes, they are |
| 17:21.32 | Freeaqingme | but (at least in the netherlands), for larger setups it's cheaper to use an isdn connection than to use a voip trunk |
| 17:21.46 | Freeaqingme | but as said, that may vary and depend on your location and needs |
| 17:21.50 | Eitan | right |
| 17:22.09 | Eitan | it was slightly more expensive to go with voip than t1 per month |
| 17:22.17 | citywok | Eitan: we run a call center 100% voip, but we're putting in a T1 b/c we're tired of voip carrier reliability |
| 17:22.31 | Eitan | citywok: that was my plan |
| 17:22.34 | Eitan | i wanted to have a back up |
| 17:22.38 | Eitan | incase things went bad |
| 17:22.40 | Freeaqingme | Eitan, yeah, it's great to start |
| 17:22.43 | citywok | (small call center), we'll run t1 as primary, and anything beyond the 24 channels will go out sip |
| 17:22.50 | Eitan | got it |
| 17:23.03 | Eitan | does it make a difference that i will be running voip over a T1? |
| 17:23.03 | leifmadsen | citywok: and I've had a couple that were the opposite that started with PRI's and added SIP to expand the number of channels, and eventually just migrated off the PRIs entirely |
| 17:23.05 | citywok | a long time ago we had 4 T1's with our old pbx, then we went pure sip w/ asterisk |
| 17:23.09 | Freeaqingme | the problem with T1 is that there's no real way of getting it redundant |
| 17:23.17 | leifmadsen | Eitan: the T1 makes no difference, it's just a data connection |
| 17:23.23 | leifmadsen | the reliability of the data connection is the problem |
| 17:23.25 | citywok | leifmadsen: i like sip it's soooo much easier. but the 3 providers i use all dick me around and piss me off. |
| 17:23.34 | leifmadsen | citywok: ya I know how that is |
| 17:23.38 | Eitan | @leifmadsen: i see, the actual voip provider is the problem |
| 17:23.44 | citywok | flowroute is really pissing me off with their fraud detection |
| 17:23.52 | citywok | every time we call a new country they block us for unusual traffic |
| 17:23.54 | Freeaqingme | fraud detection? |
| 17:24.00 | Freeaqingme | lol |
| 17:24.01 | Freeaqingme | fail |
| 17:24.05 | _Corey_ | Providers themselves are usually the weak link in this chain... building redundant paths is relatively trivial |
| 17:24.11 | citywok | they've been "working on it" for months |
| 17:24.18 | Freeaqingme | call every country just once today, and you're rid of that problem :P |
| 17:24.24 | Eitan | so i shoul dbe very careful as to whom i pick as our provider |
| 17:24.34 | citywok | _Corey_: yes, but even big providers fail. bandwidth.com has failed on me a handful of times now and they claim to be 57 ways redundant |
| 17:24.37 | DaneoShiga | anyone knows a google place to find resources about phpagi? |
| 17:24.41 | Freeaqingme | if it's critical for the business: yes Eitan |
| 17:24.47 | DaneoShiga | /google/know |
| 17:24.53 | Eitan | citywok: that was one of the proivders on my shortlist, bandwidth.com |
| 17:24.59 | Freeaqingme | DaneoShiga, the source code? |
| 17:25.03 | citywok | yea they're a bunch of retards |
| 17:25.08 | Eitan | eeech |
| 17:25.12 | Eitan | they just lost a costumer |
| 17:25.18 | _Corey_ | citywok: Calling Bandwidth a "carrier" is pretty generous |
| 17:25.21 | citywok | the last time they screwed up i was down for 36 hours before i figured out what they had done wrong, it took them 45 days to get a credit. |
| 17:25.25 | Eitan | what about Cbeyond |
| 17:25.28 | Eitan | they are next in line |
| 17:25.39 | citywok | _Corey_: i said provider :P -- they're just an L3 reseller. |
| 17:25.43 | DaneoShiga | Freeaqingme: would take a long time to find out why my script don't work sometimes... |
| 17:26.00 | _Corey_ | citywok: Not anymore, they got their LEC license a couple years ago |
| 17:26.02 | citywok | and i had to ask them 3 times to give me a credit, to which they kept saying "it takes 30 days" -- no kidding, it's been 40... |
| 17:26.29 | Eitan | i wonder what voip provider in my area is very reliable... |
| 17:26.32 | Freeaqingme | DaneoShiga, I'm busy on developing my own agi component for the zend framework, I could give you a link if you're interested |
| 17:26.43 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 17:26.58 | _Corey_ | citywok: I feel your pain... we fought with them on a $30k disconnect charge on some numbers they never delivered for more than a year |
| 17:27.29 | citywok | lol, when we signed up they charged us flatrate, and then they charged us for the long distance. we ended up with a $5k/mo bill instead of 2k. this went on for 6 months. |
| 17:27.39 | citywok | every month they would "fix" it |
| 17:27.55 | DaneoShiga | Freeaqingme: would be great, but i don't think boss is interested in changing things now =/ |
| 17:27.57 | citywok | every month it happened again. idiots. |
| 17:28.11 | _Corey_ | yeah, Global Crossing or Level3 direct would be my recommendation if you can get away with it |
| 17:28.26 | _Corey_ | at least their service doesn't suck |
| 17:28.29 | DaneoShiga | Freeaqingme: but do you know how a script should really act after a Dial command? i need the code after it to be executed, but sometimes it don't happen |
| 17:28.31 | citywok | _Corey_: we don't do 1M/mo |
| 17:28.35 | Eitan | corey: they pretty good? |
| 17:28.40 | citywok | so we don't meet the min commit to even talk to them |
| 17:28.42 | leifmadsen | DaneoShiga: you use the 'h' extension for that |
| 17:28.50 | Freeaqingme | DaneoShiga, what do you mean? |
| 17:28.57 | leifmadsen | ya, I'm a bit confused too |
| 17:29.01 | Freeaqingme | you want dtmf digits that are entered after firing the dial() command? |
| 17:29.25 | _Corey_ | citywok: Depends on which group you're talking about... they have a regular enterprise group that would sell you a T1 and a bunch of call paths if you wanted |
| 17:29.58 | DaneoShiga | Freeaqingme: no, i have a script on one server that Dials to certain queues on others servers based on dtmf and some comparisons... |
| 17:30.13 | citywok | _Corey_: ah, okay. if i'm going T1 i'd rather just go Qwest since we already spend a boatload with them and they treat us fairly well. |
| 17:30.21 | _Corey_ | Eitan: In my experience they do SIP really well, though your mileage may vary |
| 17:30.22 | Freeaqingme | Code is here btw DaneoShiga, it's a WIP: https://github.com/Freeaqingme/zf2/tree/asterisk/library/Zend/Telephony/Asterisk |
| 17:30.56 | DaneoShiga | gonna take a look ^^ |
| 17:31.59 | DaneoShiga | let me pastebin my script, one moment |
| 17:34.00 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:35.04 | DaneoShiga | http://pastebin.com/4ht7pPXC |
| 17:35.11 | ruben23 | hi guys buy default- h.323 are not packgae or bundle by asterisk - it should be separately installed or compile..? |
| 17:35.42 | DaneoShiga | Freeaqingme: http://pastebin.com/4ht7pPXC i need code after line 196 to always execute, but it sometimes doesn't... |
| 17:37.25 | Freeaqingme | DaneoShiga, also if the caller has hung up already? |
| 17:38.11 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:39.04 | DaneoShiga | Freeaqingme: if possible, yes |
| 17:39.25 | Freeaqingme | then you could look at dead agi |
| 17:39.37 | Freeaqingme | but I'm not sure that allows to dial if the caller has hung up |
| 17:39.46 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 17:39.54 | Eitan | what about ATT's voip service |
| 17:40.10 | gruvfunk | what voip service, didn't they call it quits on CallVantage? |
| 17:40.46 | Eitan | na, doesnt look like it according to their site |
| 17:40.52 | Eitan | on the phone with their buisness people now |
| 17:40.54 | Eitan | lets see what they sau |
| 17:40.56 | Eitan | say |
| 17:40.57 | DaneoShiga | Freeaqingme: well, my biggest problem is when the code after dial don't work... |
| 17:42.24 | hrnkc_ | Hi. Does anyone know if is it possible to turn off NOTIFY messages that are sent where user has voicemail? |
| 17:42.47 | DaneoShiga | hmm, it is using dead agi... |
| 17:43.03 | _Corey_ | jrnkc_: Set mailbox= to blank on their sip config |
| 17:43.10 | leifmadsen | or don't set it at all |
| 17:43.14 | hrnkc_ | Corey> thanks a lot |
| 17:45.21 | hrnkc_ | Ive got default asterisk configuration. The weird thing is that everytime i make a call, asterisk sends NOTIFY message with number of voicemails... |
| 17:45.54 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:47.45 | cusco_ | hi... |
| 17:48.13 | gruvfunk | hey hey |
| 17:48.15 | m4xx | lol "Hacking indications.conf for Fun and Profit" |
| 17:49.17 | cusco_ | when calling isdn I would like to know wich telco does the destination belong to |
| 17:49.17 | m4xx | is that a common saying or is that paying homage to the "smashing the stack for fun and proffit" |
| 17:49.21 | cusco_ | how can I know that |
| 17:49.28 | leifmadsen | m4xx: it's just something we made up :) |
| 17:49.32 | malcolmd | ruben23: you should do chan_ooh323 from the addons directory in make menuselect |
| 17:49.33 | DaneoShiga | Freeaqingme: it's using dead agi, i think the code doesn't go on when the caller hangs up, and goes ok when the agent hangs up... |
| 17:49.33 | cusco_ | I was reading about q.931 |
| 17:49.44 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:49.53 | Freeaqingme | DaneoShiga, afaik the idea of deadagi is that it continues running after it's hung up |
| 17:50.05 | Freeaqingme | but you'd have to check if it doesnt error if you then try to dial out without a caller |
| 17:50.09 | Freeaqingme | (I think it does/will) |
| 17:50.21 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-fkvoaaafpjlhajdv) |
| 17:50.34 | m4xx | so no homage to: http://www.phrack.org/issues.html?id=14&issue=49 ? ;[ |
| 17:50.41 | m4xx | it would have only made it cooler ;x |
| 17:50.53 | DaneoShiga | Freeaqingme: I see... gonna take a look at it, thanks :) |
| 17:51.08 | Freeaqingme | yw |
| 17:56.44 | *** join/#asterisk porche (~kursad@212.253.24.82) |
| 17:56.57 | porche | Hi |
| 17:57.18 | porche | I am looking for a number validation service that runs over SS7 |
| 17:57.48 | porche | does any one know one? |
| 18:01.19 | ruben23 | malcolmd: i see xxx_chan_ooh323 im using asterisk 1.6 |
| 18:05.11 | Freeaqingme | porche, I think no such service exists |
| 18:05.21 | Freeaqingme | that could be considered to be one of the weak points of ss7 |
| 18:07.49 | ruben23 | malcolmd:? it means its not package with asterisk 1.6..? |
| 18:08.53 | porche | Thank you Free |
| 18:09.05 | porche | any other option than ss7? |
| 18:10.13 | *** join/#asterisk frawd (~francois@242.Red-81-38-28.dynamicIP.rima-tde.net) |
| 18:11.43 | cusco_ | i |
| 18:11.45 | cusco_ | hi |
| 18:11.49 | cusco_ | regarding moh |
| 18:12.01 | cusco_ | how do I choose the order of files in a class? |
| 18:12.11 | Eitan | whats the call capacity on the free version for simultaneous calls? is there one? |
| 18:12.28 | cusco_ | in musiconhold.conf I have mode=files and directory=/var/lib/asterisk/moh/Lusomundo |
| 18:12.36 | Kobaz | free version of asterisk? no limits |
| 18:12.49 | cusco_ | now Im adding files in alphabetic order say moh1.al moh2.al moh3.al |
| 18:13.02 | cusco_ | but then the 3 goes on top when I do 'moh show files' |
| 18:13.33 | Eitan | thats what i thought |
| 18:13.37 | Eitan | cbeyond rep is trying to sell me some shit |
| 18:13.41 | Eitan | saying im gonna have to pay for licences |
| 18:13.43 | Eitan | i was like no |
| 18:14.02 | _Corey_ | hmmmm |
| 18:14.34 | _Corey_ | Normally SIP trunking is sold based on call paths, so isn't that what they mean? |
| 18:15.11 | Eitan | they are so confused |
| 18:15.13 | Eitan | and they are confusing me |
| 18:16.04 | cusco_ | http://paste.debian.net/116005/ |
| 18:16.06 | cusco_ | :| |
| 18:17.17 | m4xx | eh |
| 18:17.21 | m4xx | [May 4 14:16:45] WARNING[20781]: res_monitor.c:499 __ast_monitor_stop: Execute of ( nice -n 19 sox -m "/var/spool/asterisk/monitor/wtf-in.wav" "/var/spool/asterisk/monitor/wtf-out.wav" "/var/spool/asterisk/monitor/wtf.wav" && rm -f "/var/spool/asterisk/monitor/wtf-"* ) & failed. |
| 18:17.32 | m4xx | any ideas? |
| 18:18.07 | cusco_ | why not use MixMonitor() ? |
| 18:18.15 | m4xx | because that didn't work at all |
| 18:18.23 | cusco_ | works here... |
| 18:18.27 | m4xx | i'm using freebsd |
| 18:18.34 | m4xx | not sure if it makes a huge difference |
| 18:18.58 | m4xx | mixmonitor doesn't produce any errors, but it doesn't create any file either |
| 18:22.05 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 18:23.51 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:26.12 | m4xx | is there anything that would cause mixmonitor to not work on freebsd? |
| 18:26.30 | m4xx | it doesn't work with either absolute path or relative path |
| 18:27.00 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 18:31.26 | Kobaz | mixmonitor should be used in all cases unless you specifically need to have individual legs of the call being recorded |
| 18:31.43 | Eitan | so $800 bucks for 30 call paths |
| 18:31.54 | cusco_ | how do I set order of several moh audio files?? |
| 18:31.57 | Kobaz | Monitor() is part of the audio bridging thread... and if you have high disk io, your call quality will suffer |
| 18:32.41 | m4xx | http://paste2.org/p/1399169 |
| 18:39.30 | JonathanRose | m4xx: I'll have a quick look to see if that dialplan works as it should on my platform now. |
| 18:39.49 | m4xx | JonathanRose: thank you =] |
| 18:40.50 | cusco_ | please.. its getting on my nerves |
| 18:40.55 | cusco_ | moh class file order |
| 18:41.13 | cusco_ | http://paste.debian.net/116007/ |
| 18:42.19 | m4xx | it's a hack, but have you tried prepending a number to the file names |
| 18:42.31 | m4xx | 1-fileblah.wav 2-filebalh.wav |
| 18:43.16 | cusco_ | yes |
| 18:43.24 | cusco_ | I tried prepending and sufixxing |
| 18:44.20 | cusco_ | look: http://paste.debian.net/116009/ |
| 18:44.23 | cusco_ | I don't understand |
| 18:45.20 | _Corey_ | cusco_: If it's important, you could always merge the files together |
| 18:45.32 | cusco_ | I think I will |
| 18:45.36 | cusco_ | it is important yes lol |
| 18:45.55 | JonathanRose | m4xx: Yeah, doesn't work in any build. |
| 18:46.15 | cusco_ | ? |
| 18:46.19 | m4xx | so am i doing something wrong or is it broken? |
| 18:46.37 | JonathanRose | I think the channel needs to connect to something first. |
| 18:46.48 | m4xx | so answer then mixmonitor? |
| 18:46.52 | JonathanRose | No |
| 18:47.02 | cusco_ | no |
| 18:47.06 | cusco_ | mixmonitor first |
| 18:47.07 | cusco_ | then answer |
| 18:47.10 | cusco_ | o.O ? |
| 18:47.16 | JonathanRose | It's just that there isn't anything listening, so there isn't an audiohook for it to bind to. |
| 18:47.27 | cusco_ | yes I thought about the same |
| 18:47.28 | m4xx | right now i'm using originate, then passing it off to that dialplan |
| 18:47.31 | cusco_ | try playback () |
| 18:47.47 | JonathanRose | If it doesn't bind to anything, it doesn't start the filestream. |
| 18:47.59 | m4xx | i'm sorry, i'm still learning what do you mean by that? |
| 18:48.04 | cusco_ | no audio |
| 18:48.12 | cusco_ | thus no need to record it |
| 18:48.20 | m4xx | but i know there's audio |
| 18:48.22 | cusco_ | try dial() some peer |
| 18:48.26 | m4xx | if i use the monitor |
| 18:48.30 | m4xx | there is audio |
| 18:48.30 | cusco_ | ow... |
| 18:48.52 | JonathanRose | The channel isn't sending or receiving audio. |
| 18:48.52 | *** join/#asterisk kdas (8898be1b@gateway/web/freenode/ip.136.152.190.27) |
| 18:49.00 | JonathanRose | And monitor works a little differently. |
| 18:49.15 | m4xx | i'm positive there's audio being received, if i do Monitor(wav,wtf,m) i do get the in and out legs |
| 18:49.24 | m4xx | the in leg does contain audio |
| 18:50.12 | m4xx | cusco_, that pastebin was your musiconhold.conf ? |
| 18:50.20 | kdas | hi, i am using my cell to connect to asterisk however there is a problem when some one calls me and i am not connected to asterisk then later connect i dont get a missed call notification. how can i setup this functionality? |
| 18:50.54 | kdas | my cell is using a sip client by the way |
| 18:51.26 | cusco_ | m4xx: was moh show files |
| 18:51.30 | cusco_ | cli command |
| 18:51.35 | m4xx | ah |
| 18:52.07 | cusco_ | kdas: on dialplan on hangup get DIALSTATUS - if NOANSWER do something |
| 18:52.20 | cusco_ | like writting to a text file or sending a sms :p |
| 18:52.22 | cusco_ | or email |
| 18:52.30 | JonathanRose | m4xx: I think that's because monitor establishes the connection at that point. MixMonitor doesn't do that. |
| 18:53.09 | JonathanRose | Tell you what though, if you want MixMonitor to work without any other behavior just so that you can make a recording, play an empty sound effect. |
| 18:53.14 | JonathanRose | So MixMonitor(...) |
| 18:53.16 | JonathanRose | Answer() |
| 18:53.22 | cusco_ | JonathanRose: actually mixmonitor() has option b |
| 18:53.32 | cusco_ | to only record while channel is bridged |
| 18:53.36 | JonathanRose | That just makes it wait until the call is bridged. |
| 18:53.38 | cusco_ | if he's not using that ... |
| 18:53.42 | JonathanRose | He isn't. |
| 18:53.53 | cusco_ | it should record anyway.. right? |
| 18:53.56 | m4xx | if i use originate, then answer() in the dialplan, isn't that bridged? |
| 18:54.08 | cusco_ | I guess so |
| 18:54.11 | kdas | cuscco_: umm cant i just get a missed call notification when sip client connects? is there some kind of sip message? |
| 18:54.23 | kdas | writing to a file seems pointless |
| 18:54.34 | JonathanRose | Not really. |
| 18:54.51 | JonathanRose | The bridge option is more for stuff like where you have a menu that'll connect you to another line... |
| 18:54.54 | cusco_ | kdas: once the call is over there is no more processing of that. you can do somehting when the call is over like write to log file |
| 18:55.09 | JonathanRose | Then it won't start monitoring until the phone is picked up on the other end... |
| 18:55.11 | cusco_ | kdas: don't write to file, send an e-mail then |
| 18:55.39 | kdas | not what i want but i get it |
| 18:55.50 | m4xx | i thought with originate it wouldnt get to the dialplan untill the call was picked up |
| 18:56.19 | m4xx | at least not priority 1 |
| 18:56.42 | JonathanRose | As I was saying, try MixMonitor(), Answer(), and then Background(silence/1). That should make it work. |
| 18:56.51 | kdas | another question. i have jabber setup and my sip client supports sip iming (simple) i am guessing. is there a way to recieve send sip messages in asterisk? |
| 18:56.55 | m4xx | will do |
| 18:57.03 | Eitan | alright i just ordered 30 trunks with cbeyond |
| 18:57.07 | Eitan | lets hiope they know what they are doing |
| 18:58.20 | m4xx | wow, that worked |
| 18:59.37 | m4xx | should mixmonitor(), answer(), sendtdtmf() work the same? |
| 18:59.49 | m4xx | *senddtmf |
| 18:59.56 | JonathanRose | If it involves sound that would be put onto the calling line, it'll work. |
| 19:01.12 | leifmadsen | heh, good luck ;) |
| 19:01.12 | leifmadsen | when they work, they seem to be fine.... |
| 19:01.12 | leifmadsen | have a customer using them, and don't hear from him too much anymore |
| 19:01.33 | m4xx | this is my original dialplan: http://paste2.org/p/1399210 |
| 19:02.07 | m4xx | which didn't work, doesnt the senddtmf send audio over the line? |
| 19:04.29 | JonathanRose | I'll check real fast. |
| 19:06.38 | JonathanRose | Well, apparently I'm wrong in my assumption then. |
| 19:06.52 | JonathanRose | It does send audio. I'm not sure if it's just different because it's DTMF or what. |
| 19:07.55 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 19:08.58 | m4xx | having to add the play silence seems really hacky to me =[ |
| 19:09.29 | JonathanRose | You usually won't have to though. |
| 19:09.34 | *** join/#asterisk kdas (8898be77@gateway/web/freenode/ip.136.152.190.119) |
| 19:09.57 | kdas | anyone have ideas how to implement sip ims with asterisk? |
| 19:10.58 | JonathanRose | To be honest, I'm still unsure what you are trying to do with this. I don't believe I can think of any case where you'd want someone to call a number and just start recording them without any kind of prompt. |
| 19:11.17 | m4xx | i've got 2 projects |
| 19:11.20 | cusco_ | another question... |
| 19:11.29 | m4xx | one is to do something similar to checking voice mail off site |
| 19:11.37 | cusco_ | can I make it so taht periodic-announce for a queue only plays once ? |
| 19:11.43 | cusco_ | not every 60 seconds |
| 19:11.52 | cusco_ | but only the first time |
| 19:12.05 | kdas | <PROTECTED> |
| 19:12.07 | m4xx | calling a number, entering some digets, hanging up |
| 19:12.20 | m4xx | i just want to monitor so i can verify that things are working |
| 19:12.23 | leifmadsen | cusco_: some info should be here: http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html#ACD_id288932 |
| 19:12.54 | m4xx | i'm also making a message delivery service, call a customer, play a message and hangup, also want ot monitor to make sure everything is working out ok |
| 19:13.31 | ruben23 | hi is chan_ooh323 bundle ot included on what version of asterisk..? |
| 19:13.49 | leifmadsen | ruben23: 1.6.2 and later, but the good changes are in 1.8 and later |
| 19:13.52 | leifmadsen | we already had this discussion |
| 19:15.23 | m4xx | would it not make sence to (mix)monitor those two type of applications? |
| 19:15.40 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 19:22.33 | ruben23 | leifmadsen: because when i do make menu select --> xxx chan_ooh323 , i see like this i cant select it to install, any idea..? |
| 19:23.02 | Qwell | ruben23: You don't have prerequisites installed... Look at the bottom of the screen. |
| 19:24.09 | ruben23 | ok |
| 19:27.53 | ruben23 | <PROTECTED> |
| 19:28.16 | Qwell | What does it say at the bottom of the screen? |
| 19:34.20 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:35.08 | WiretapSeven | leifmadsen, quick question before I run out the door if you don't mind |
| 19:35.31 | ruben23 | Qwell: just a second |
| 19:35.32 | WiretapSeven | how long is likely before this is in a release? https://issues.asterisk.org/view.php?id=13996 |
| 19:37.55 | ruben23 | Qwell: i see this---------------->The NuFone Network's OpenH323 Channel Driver Depends on: openh323(E) |
| 19:38.42 | JonathanRose | m4xx: Looking over a few things in app_mixmonitor.c, it seems that just using senddtmf doesn't actually send any audio over the audio-hook. I guess DTMF works a little differently from regular two-way audio. |
| 19:39.07 | ruben23 | this si for chan_h323, but other guy here on the forum mention this--> chan_ooh323 , what could be the differetnce between the two..? |
| 19:39.34 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 19:39.35 | JonathanRose | I'm thinking that mixmonitor actually just plain will not capture the DTMF as a result. |
| 19:39.49 | JonathanRose | Which might be the intent actually. |
| 19:40.12 | WiretapSeven | and I'm gone :/, oh well, if there's a response I'll see it tonight |
| 19:41.32 | m4xx | JonathanRose, why wouldn't it work if only audio was received? |
| 19:42.08 | JonathanRose | Because the audio never gets into the audiohook. |
| 19:42.14 | *** join/#asterisk Corydon76-home (grey@c-69-137-80-31.hsd1.tn.comcast.net) |
| 19:42.14 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 19:42.41 | JonathanRose | If the audio isn't in the audiohook, mixmonitor isn't going to see it. |
| 19:42.58 | JonathanRose | And DTMF doesn't get into the audiohook. It's weird. |
| 19:43.11 | leifmadsen | WiretapSeven: I have no answer for you |
| 19:43.25 | leifmadsen | it's a feature, and is probably very low on the list of items to get resolved |
| 19:43.44 | m4xx | why not provision your cisco phone with dhcp and tftp? |
| 19:44.41 | m4xx | n/m i read wrong |
| 19:51.58 | cusco_ | leifmadsen: so basically I set a limit in the queue at 60 seconds, I playback the announcement, raise QUEUE_PRIO and queue it again |
| 19:52.13 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:53.57 | gruvfunk | cusco_ how about just a normal Playback prior to entering the Queue? |
| 19:54.13 | gruvfunk | and skip periodic-announce |
| 19:54.20 | cusco_ | nope |
| 19:54.25 | cusco_ | I need it at 55s |
| 19:55.11 | cusco_ | Im hoping QUEUE_PRIO works accordingly |
| 19:56.40 | cusco_ | queue is suposed to last for 3 minutes |
| 19:57.14 | cusco_ | and I have a audio informing the client that he is watinf for 55 secs and he may press 8 to get a call back in the next 48h |
| 19:57.21 | cusco_ | to leave a contact |
| 19:57.33 | cusco_ | and I wish to play that only once |
| 19:59.28 | *** join/#asterisk etfonhomey (~chatzilla@65.166.171.74) |
| 20:04.33 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:14.04 | *** join/#asterisk Corydon76-home (yellow@c-69-137-80-31.hsd1.tn.comcast.net) |
| 20:14.04 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 20:18.35 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:29.09 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v009-087.mobile.uci.edu) |
| 20:40.56 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:45.30 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 20:49.19 | *** join/#asterisk sskyles (~sskyles@60.105.91.184.cfl.res.rr.com) |
| 20:52.04 | sskyles | Is there a PDF or text file anywhere that fully documents Asterisk configuration in more detail than just looking at the config files themselves? |
| 20:52.28 | Freeaqingme | ~thebook |
| 20:52.28 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:52.30 | Freeaqingme | sskyles, ^^ |
| 20:52.34 | gruvfunk | :) |
| 20:52.48 | sskyles | Great! |
| 20:52.50 | Freeaqingme | whatever your question is: ~thebook usually suffices as an asnwer |
| 20:53.06 | gruvfunk | if not, google, or ask in here |
| 20:54.51 | sskyles | Well, for example; I don't know what dundi is or even if I need something like that for home use. I'd probably disable it. I'm sure any good documentation will tell me all about it, how it's used or how to disable it. |
| 20:56.51 | leifmadsen | sskyles: yes there is a discussion abut dundi and what it is and how to use it in that book as referenced |
| 20:57.07 | leifmadsen | for the most part that book should answer any general questions like that, if not, let me know! |
| 20:57.27 | sskyles | It looks like exactly what I need, thanks again. |
| 20:59.20 | *** part/#asterisk sskyles (~sskyles@60.105.91.184.cfl.res.rr.com) |
| 21:02.15 | *** join/#asterisk karen_m (~karen@d50-99-60-236.abhsia.telus.net) |
| 21:02.21 | karen_m | !newbook |
| 21:02.29 | karen_m | ~newbook |
| 21:02.29 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 21:02.29 | karen_m | .newbook |
| 21:02.32 | karen_m | ~thebook |
| 21:02.32 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 21:03.07 | karen_m | is there a pdf for trhat book available? |
| 21:03.21 | leifmadsen | karen_m: for purchase, yes |
| 21:03.31 | karen_m | reading online lol :) |
| 21:03.32 | simplydrew | hmm. I should download that to my iPad for some reading at some point |
| 21:03.34 | simplydrew | makes a note |
| 21:04.15 | leifmadsen | there are kindle, etc... editions as well |
| 21:21.11 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:21.28 | *** join/#asterisk Miriup (~dirk@pdpc/supporter/active/miriup) |
| 21:39.52 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 21:45.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:45.37 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:45.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:46.21 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:52.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:54.08 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 21:54.17 | *** part/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 22:11.34 | *** join/#asterisk espent (~espent@46.80-203-219.nextgentel.com) |
| 22:17.00 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 22:33.09 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 22:36.28 | *** join/#asterisk wonderworld (~ww@port-92-201-173-210.dynamic.qsc.de) |
| 22:46.34 | *** join/#asterisk gnutsy (~mmj069@ifg.fttp.xmission.com) |
| 23:07.14 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:09.16 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 23:25.09 | *** join/#asterisk philippel_mac (~p_lindhei@184-195-204-37.pools.spcsdns.net) |
| 23:36.34 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 23:44.05 | sezuan | Has Packet2Packet bridging been remove since Asterisk 1.4? |
| 23:57.54 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |