IRC log for #asterisk on 20110504

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00:46.26hugogeegreets all :D
00:48.16hugogeeWhat does one use to write asterisk scripts using python?
00:50.22km2what controls the timezone of times in /var/log/asterisk/cdr-csv/*.csv?
00:51.20cusco_someone earlier said something about some setting in cdr.conf
00:51.20Freeaqingmehugogee, you're probably looking for AGI
00:51.59cusco_gmttime=yes or so
00:53.36*** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
00:54.59hugogeeFreeaqingme: Thanx just found a few libs. :D  Off i go... cheers to all!
00:55.35km2thanks, i'll look
00:55.50Freeaqingmehugogee, enjoy. Feel free to hang around though ;)
00:58.28*** join/#asterisk Marvelous (~CE0@197.195.78.147)
00:58.45MarvelousHELLO
00:59.37carrarhi
00:59.44carrarWelcome to #Asterisk
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01:04.18Marvelousi need to know which  file content the configuration for Asterisk™ Configuration Engine
01:08.40p3nguinI guess that would be asterisk.conf.
01:10.59Marvelouswrong file
01:11.12Marvelous<PROTECTED>
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01:12.02seraphieMarvelous: you are talking about the Asterisk GUI. PLease see #asterisk-gui
01:12.22carrar<PROTECTED>
01:12.54Marvelouswait
01:13.01carrarno waiting
01:13.03carrarit's GO TIME
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01:16.17Marvelous:d
01:16.21Marvelousokie
01:16.24Marvelousbut the config is
01:16.27Marveloushttp.conf
01:16.40Marvelousasterisk/http.conf
01:24.17MarvelousStarting asterisk: FD 32767 exceeds the maximum size of ast_fdset!
01:25.01seraphieMarvelous: Google first.
01:25.27Marvelousokie
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02:15.42Marvelouswhere i can get  asterisk education
02:19.06FreeaqingmeMarvelous, I think digium has some training programs
02:19.20Freeaqingmeotherwise you could read ~thebook which is a great place to start
02:19.23Marvelousany online courses  i could take !!
02:19.24Freeaqingme~thebook
02:19.24infobotAsterisk: 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
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04:28.21l1nuxmanhow 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"
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04:47.09gruvfunk_afkhey l1nuxman
04:47.28gruvfunk_afkso, you are probably currently using the VoiceMailMain aplication, yes?
04:48.12gruvfunk_afkcheck out the VoiceMail application instead for your needs - create a new extension like exten => 9999,1,VoiceMail
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05:14.32gruvfunkanyone 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
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05:20.50l1nuxmangruvfunk, why VoiceMail? Whats that gonna do for me?
05:21.33gruvfunkwhat were you trying to do?
05:21.56gruvfunkleave voicemail without ringing the extension?
05:22.07l1nuxmanwithout physically using a phone
05:22.40l1nuxmanI was thinking maybe I could just 'touch a.wav'
05:22.49l1nuxmanI wanted to test voicemail attachment email
05:25.31l1nuxmangoodnight, I'll check your message tomorrow.
05:40.18gruvfunkcan AGI PHP somehow help to stream a java/flash radio station?
05:40.32gruvfunkme scratches his head
05:40.53gruvfunkscratches his head
05:41.43kaldemarices has been used to stream audio with asterisk.
05:43.08Sertysthere's no need for agi php
05:43.23gruvfunkdo tell more
05:43.54gruvfunkcan I stream this: http://player.streamtheworld.com/liveplayer.php?CALLSIGN=ARNCITY
05:44.27ChannelZif it's outputting a suitable format you could using MOH
05:44.54ChannelZthat particular URL appears to be flash though so I'm guessing not.
05:44.55Sertyswell, yeah, it's basiclly about custom MOH
05:45.21gruvfunkI've done custom MOH before using mp3 streams, m3u
05:45.24Sertysi can think that vlc might grab the rtp data itself
05:45.34gruvfunkbut a java flash?
05:45.42kaldemarthat kind of music cannot be streamed.
05:45.46kaldemaranywhere.
05:46.10Sertyskaldemar: u're talking bullshit here
05:46.11ChannelZthat bad?  Rap?
05:46.42gruvfunkthe content is not in question, but the format
05:46.57ChannelZNo, I'd say the content is pretty questionable
05:46.57Sertyswell, flash streaming is basically RTMP
05:47.12Sertyswhich is asao rtp
05:47.13kaldemarSertys: i was referring to the genre not being aline with my musical taste. :)
05:47.22Sertyskaldemar: lol
05:47.27Sertysi just tuned in to the radio
05:47.36Sertysand i kinda feel u right now
05:47.41ChannelZI guess they have morning zoo's in India too
05:47.57Sertysindia has it all :)
05:48.31ChannelZThey're inescabable
05:48.36Sertyshttp://wiki.videolan.org/Flash_Video
05:48.38ChannelZThis is why I quit listening to the radio years ago
05:48.50ChannelZbut alas
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05:51.19Sertysyeah
05:51.48kaldemarhttps://issues.asterisk.org/view.php?id=15484
05:51.52Sertysgruvfunk: u're prolly wasting precious time right now trying to figure this shit out
05:52.00gruvfunkyepper
05:52.27gruvfunksome customers want the world
05:52.43Sertysit's good if they pay for it
05:52.54Sertysi can say it's doable though
05:53.02gruvfunkyeah?
05:53.21Sertyswireshark the traffic
05:53.26Sertysget the rtmp server
05:53.35Sertysand try to stream off it with a player
05:53.45gruvfunkyeah but that's not a practical way to build a solution with 100's of radio stations
05:53.50Sertysand if vlc works, u can use it to write to stdout
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05:54.24Sertysgruvfunk: u gotta figure the basics in order to write a proper wrapper with it
05:54.40gruvfunkcurious: then what?  stdin to Asterisk..
05:54.59Sertysthen just use moh
05:55.20Sertysi might thing eagi might do the trick with pipe streaming
05:55.32Sertysbut it would eat up res
05:55.47Sertysif there're 2+ concurrent users :)
05:56.09gruvfunkyeah...
05:56.11Sertysi don't have enough time right now to test it out
05:56.38gruvfunkthe better solution is for them to figure out a media transcoder that then feeds Asterisk a friendlier format
05:57.28Sertysthat's what i'm trying to make u do
05:57.40gruvfunkright
05:57.57Sertysevery player is basically a media transcoder...
05:58.04Sertysand viceversa
05:58.31gruvfunkbut the resources would be significant for multiple callers listening in
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05:58.47gruvfunkassuming different stations
05:59.20gruvfunki'll put some more thought on it tomorrow, thanks for the leads though, gears are turning
05:59.38Sertysnp
05:59.41Sertysgtg
06:05.27WiretapSevenanyone here screwed around with a 7970 long enough to get BLF working?
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06:26.47schmidtsgood morning
06:40.15celthundermorning schmidts
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08:05.07doneiri've been goolging for an answer, but am still unsure - do the dahdi drivers (and cards i guess) support unframed e1 pri connections?
08:06.06coppiceif they are unframed, they are not PRI
08:06.24doneiri 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
08:07.31doneirhmm... unframed mode would be BRI setup then or such?
08:08.02coppiceunframed would be unframed
08:08.59doneirsorry, 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?)
08:16.07doneirokay, so bottom line unframed == BRI, framed/fractional == PRI compatable?
08:16.52doneiri had thought some juniper and cisco cards could support telephony over unframed e1 connections, though probably mixed up with network (bri) based
08:17.02coppiceunframed == HDLC/data, framed == PRI
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08:30.30ChainsawOkay, IPv6 to issues.asterisk.org is back up. Good to see.
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08:45.10sezuanShould 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.
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08:54.44asterisk-learnerhi, is there  a way to stop the execution of WaitForSilence() application while running on a channle without hanging up the channel and before timeout ?
08:56.14asterisk-learner.join freepbx
08:56.46sxpertasterisk-learner: use a shorter timeout, and use the iterations option
08:57.32sxpertah. no, can't do that...
08:58.06sxpertuse shorter timeout in a loop
08:59.07asterisk-learnersxpert: I want to wait for 5 sec of silence, i am already using : 5,1,0
08:59.13asterisk-learnerwait 5 sec 1 time, for 0 timeout
08:59.45sxpert0 timeout probably means "infinite"
09:00.50sxpertso you may want to use 5,1,5
09:00.58sxpertor something
09:01.15asterisk-learner5 ?
09:01.27asterisk-learnerwait let me explain to you my scenario better
09:01.47Pieplay>0 not 0
09:02.16sxpertbesides, if you want 5 seconds of silence, it should be (5000,1,5) or something
09:02.36sxpertdoc says "noiserequired" in milliseconds
09:02.46asterisk-learnersxpert: yeah, actually i am using 5000,1,0
09:03.28sxpertbut do explain your scenario );
09:03.33asterisk-learnerI 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)
09:04.09asterisk-learnerin thread 2 i am calling application "WaitForSilence" on the same channel, and in thread 1 i am looping over and testing channel variables
09:04.40asterisk-learneris there a way to notify thread 2 to stop executing WaitForSilence and end from thread 1 ?
09:05.16asterisk-learnerbecause as far as i know pbx_exec() is blocking and wont return before the application returns....
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09:06.04sxperthow about looping over the channel variables only. it seems the waitforsilence is unneeded
09:08.01asterisk-learnersxpert: 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
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09:08.19sxpertasterisk-learner: ah...
09:09.08sxpertI actually fail to see the need of threads in that case ;)
09:09.48N101Q: 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
09:10.32N101Did I forget something?
09:12.08asterisk-learnersxpert: I am communicating  with another server in thread1 and getting data from it, if i dont use thread2, thread1 will be blocked by WaitForSilence()...
09:12.14kaldemarN101: what kind of an account?
09:12.23sxpertasterisk-learner: ah yeah, ok, I get it now
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09:13.04asterisk-learnersxpert: :-), so is there an easy way to do it without modifying WaitForSilence() code ....
09:13.06sxpertasterisk-learner: however, you really need the 5 secs of silence
09:13.21sxpertfrom the other side
09:13.23N101A sip account by editting the sip.conf file (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts)
09:14.19asterisk-learnersxpert: ???
09:14.29kaldemarN101: 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"
09:15.03sxpertasterisk-learner: if you're using WaitForSilence, it's because you expect the line to be silent for 5 seconds
09:15.40sxpertwell, you're actually waiting for signal to stop, then 5 seconds of silence
09:17.20asterisk-learnersxpert: 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...
09:20.02sxperthmm. seems like you can't stop WaitForSilence mid-check
09:20.28sxpertyou'd have to create some loop or something
09:20.40sxpertlike waitfor silence for 1 second
09:20.51sxpertthen wait for noise with 1 second timeout in a loop
09:20.56asterisk-learnersxpert: or modify its internal for(;;) loop
09:21.17sxpertand count the seconds in a variable
09:23.06asterisk-learnersxpert: 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
09:23.08N101kaldemar: 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,
09:23.35sxpertasterisk-learner: doesn't seem to exist. guess you could add that feature though ;)
09:24.11asterisk-learnersxpert: but i feel my scenario is really weird and rare... :-P
09:24.23N101and eventually if I loop it trough my lan adress with the right credentials the client returns '' failed to register, method not allowed"
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09:25.57sxpertasterisk-learner: agreed. so the better way for you is to hack up what you need out of existing stuff
09:30.35Extenis there a log i can see why asterisk falls everytime i start an AGI script ?
09:30.41Extenit just disconnects from server
09:30.42Exten"
09:36.08kaldemarN101: what does "not much" mean? nothing at all or do you get some message?
09:37.17N101yes quite
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09:54.47ferdnahow do you restart asterisk?
09:54.51ferdnaasterisk -r
09:54.58ferdnaCLI>restart now
09:54.59ferdna?
09:57.17kaldemarcore restart now in CLI or with init script.
09:58.19ferdnakaldemar, thanks
10:13.12ferdnaregister => <phonenumber>@sip.broadvoice.com:<password>:<phonenumber>@sip.broadvoice.com/<extension>
10:13.19ferdnais this correct?
10:13.34ferdnaisnt phonenumber:password@domain.com
10:13.36ferdna?
10:14.11ferdnaregister => username:password@my.service-provider.tld
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11:00.19ixyd_hi guys, is it possible to use Queue()`s gosub parameter with arguments, like  Dial()`s U() parameter?
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11:15.02devil_evoxxxhi guys
11:17.05devil_evoxxxi'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 ).
11:17.20devil_evoxxxAsterisk run as single process or it have thread?
11:17.52asterisk-learnerdevil_evoxxx: asterisk is multi threaded
11:18.48devil_evoxxxthere is anyway to reduce the load of the machine ? Or some tricks to have better performance?
11:20.29devil_evoxxxThe machine is a dual Core Xeon HT 3.4 Ghz , 4 GB ram
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11:35.08WiretapSevendevil_evoxxx, more cores
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11:43.45qCakehi
11:44.34qCakeanybody here?
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11:55.42tuxx-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.
11:56.58kaldemartuxx-: what version are you using?
11:57.19tuxx-the config is from an asterisk 1.4
11:57.37tuxx-dont have the actual asterisk running though, the box is long gone. Just trying to figure out what these options are for.
11:57.55tuxx-but afaict they dont do anything :P
11:58.50kaldemaroutgoinglimit does not exist in 1.4, 1.6.2 nor 1.8.
12:03.09tuxx-right, tnx :)
12:05.56qCakei'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,
12:05.57qCakeMusic 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?
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12:14.55qCakei'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,
12:14.56qCakeMusic 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?
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12:31.08qCakei'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,
12:31.09qCakeMusic 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?
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12:34.54rymkushi everybody!
12:35.37rymkusI'm use next dialplan construction
12:35.37rymkus<PROTECTED>
12:35.37rymkus<PROTECTED>
12:37.12rymkusCan anyone tell me how can I stop dialplan execution on the first priority if pickup was successfull?
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12:40.29ixyd_@rymkus i would expect asterisk to auto stop on successful pickup?
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12:48.02rymkus@ixyd_ yes, you understood my poor english correctly =)
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12:49.28ixyd_@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 :)
12:49.37wolfeixyd_: this is IRC, not twitter :)
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12:49.57Sertyswolfe: lol
12:50.05Sertysg point @wolfe :)
12:51.03ixyd_hehe ah sure ;)
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12:51.15rymkusixyd_  1.8.3.3 with pickup deadlock diff applied
12:51.16wolfesilly Twatter users
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12:52.32qCakeanybody: 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
12:52.32qCakeAttendants, 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?
12:52.47leifmadsenon IRC using an @ in the front signifies a channel op :)
12:53.22rymkushadn't used IRC for couple years =)
12:53.34ixyd_dito :)
12:54.52rymkusback to my problem - does pickup application return something that can be used in expression for gotoif func?
12:55.49rymkusI tried to use the i priority, but no luck
12:57.43ixyd_@rymkus my setup is different, using ael and asterisk 1.4 ... :-/
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13:03.08rymkusanyone?
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13:18.16malcolmdrymkus: not that i see, no.
13:18.32malcolmdcertainly, it's not documented...https://wiki.asterisk.org/wiki/display/AST/Application_Pickup
13:25.41qCakeanybody: 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
13:25.41qCakeAttendants, 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
13:25.42qCakenotebook it also works fine)
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13:26.24Doviddoes a dual processor machine help asterisk or it only uses one CPU ?
13:27.05sunfoneDovid: IMO by the time you need a second processor you should probably be thinking about another box
13:27.42kaldemarrymkus: 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.
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13:28.54Dovidsunfone: 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
13:28.56IamTryingDoes Asterisk supports: 1) Audio: G722, Video: H.264, Presentation: H.239 2) All over SIP ?
13:29.58DovidIamTrying: 1) Yes 2) Don't recall but supports one of em, dont think supports H.239
13:30.08Dovidone of the vid codecs
13:31.25IamTryingDovid, so no H.239 supported for SIP?
13:31.38sunfoneDovid:  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
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13:33.15leifmadsenno H.239
13:33.36leifmadsenH.264 and H.263 yes
13:34.06sunfoneAnyone using cacti to monitor call volume?  Point me to some templates?
13:34.18leifmadsensunfone: no, just use opennms for that
13:34.28coppiceH.239 is the multi-camera protocol, isn't it?
13:34.35sunfonehmm haven't heard of it... will look at it thanks
13:34.40IamTryingleifmadsen, what else can be a solution to do get h.239 working? any idea or thoughts/tips..
13:35.20leifmadsenIamTrying: you could write a patch to provide the functionality?
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13:36.28leifmadsensunfone: I wrote about it in the monitoring chapter of Asterisk the definitive guide as well -- works quite well
13:36.30Dovidsunfone: using proprietary software that has agi's on the box.
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13:39.50IamTryingleifmadsen, if bypass media is used, is it really necessary to have h.239 supports?
13:40.01sunfoneDovid: bummer :)  But perhaps your AGI's, since they are spawned as separate processes, will take advantage of your extra CPUs
13:40.12leifmadsenIamTrying: I know nothing about h.239
13:40.19sunfoneleifmadsen: I have yet to checkout your book - keep meaning to do so :)
13:40.37IamTryingleifmadsen, its all about SDP file, bypass media should do already those in peer 2 peer
13:40.37sunfoneLooking at the OpenNMS demo now... pretty impressive!
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13:44.34sunfoneI just wish it wasn't java based (sigh).  Java seems to make everything all that more complicated and takes so much underlying resources
13:46.22leifmadsensunfone: 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
13:46.40leifmadsensunfone: ya, I run it inside a VM to try and limit the amount of resources it can use
13:47.09sunfoneTo date all of our monitoring is hand-tooled mrtg :)
13:47.43sunfoneJust 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
13:47.54sunfoneAnd it hardly uses any resources
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13:48.15rymkuskaldemar: isn't COLP  QSIG and PRI related feature? I use only SIP and IAX lines.
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13:49.26ChainsawI 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?
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13:54.09nite613Hi 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
13:54.22nite613I've considered crafting and injecting the SIP packet, but checking for an easier way ;)
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13:57.42leifmadsennite613: soft hangup?
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14:00.35nite613leifmadsen: what do you mean?
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14:02.16leifmadsennite613: there is an asterisk CLI command called 'soft hangup' in 1.4 that you could use to request the channel be hun gup
14:03.00nite613Woohoo! :) Thank you, that seems to be exactly what I was looking for!
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14:04.54_Corey_nite613: you might also look into 'rtptimeout'
14:07.34nite613Corey, don't talk to me about rtptimeout ;) That's a whole 'nother issue I'm fighting
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14:08.16_Corey_lol
14:08.20nite613I'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
14:08.30leifmadsennite613: you can probably access that command (or a similar hangup command) from the Asterisk manager interface as well
14:08.56nite613Thanks again, leifmadsen, the CLI command will work fine for me
14:09.22leifmadsensounds good
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14:33.48Aut0ExeChi guys anyone here use voip.ms ?
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14:35.20psilikonAut0ExeC, yeah I use them
14:35.50Extenyip
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14:54.16m4xxwhen compiling on freebsd i'm getting http://paste2.org/p/1398829
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14:58.11Aut0ExeCpsilikon: nice... thats like prepaid right?
14:58.20Aut0ExeCpsilikon: with roll over?
14:58.38m4xxi've been using didforsale.com, seems ok so far
14:59.44psilikonAut0ExeC, yes
15:02.02m4xxooops perhaps i should install dahdi ;[
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15:15.54Aut0ExeCpsilikon: thanks bro
15:16.15Aut0ExeCpsilikon: only thing is I see there is I think a 25 dolloar minimum
15:16.35Aut0ExeCpsilikon: but considering that you can roll over year after year... I guess I will never loose it
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15:21.49andyoutsideHas  anyone noticed in 1.8.3.3 that when you park someone you do not hear wherethey are parked to
15:22.46gruvfunkhey Aut0ExeC did you finish out that setup?
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15:25.30emoraAnyone here working with Adhearsion?
15:26.03andyoutside_and if you park someone in the building they will hear the number they are parked at.
15:26.14andyoutside_I am not sure about if it is outside thte building
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15:31.29hrnkcHi 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
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15:34.18hrnkcThe 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.
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15:44.27epartdoes anybody here knows GoAutoDial
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15:45.15epartdoes anybody here knows GoAutoDial please help
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15:46.40zknHi, does anyone know if there a way to achieve the same as "alternateexts" in users.conf without using users.conf ?
15:47.42zknusers.conf now sports an optional alternateexts property, which permits allocation of additional extensions which will reach the specified user.
15:49.00leifmadsensounds like a dialplan thing
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15:49.09leifmadsenthat's the traditional way to handle that
15:49.20zkni want to be able to dynamically achieve this sort of feature
15:49.43angryuserepart, depends, speak
15:50.39emoraAdhearsion anyone?
15:51.35epartjust any idea.. that about goatodial..
15:53.17zknhaven't been able to accompish this with regexten / regcontext in sip.conf
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16:08.49InsektOhi, 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?
16:09.19leifmadsenInsektO: hardware issues for Digium hardware should be brought up with Digium tech support
16:09.28leifmadsenthey are the ones who are able to support those products
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16:12.54Eitani have just bought the book
16:12.54Eitanoh man
16:14.33carrarw00t
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16:18.19billmaniaI'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.
16:18.57carrarinbound path?
16:19.13carrarpath of a call?
16:19.22carrarcall route
16:19.35leifmadsenbillmania: channel
16:19.50andyoutsidedialing plan?
16:19.52leifmadsenthere is not really any term for the collection of audio conversations
16:20.07billmaniacarrar: 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?
16:20.10carrarbucket 'O' Channels
16:20.26InsektOleifmadsen, yeah, i know that, but perhaps someone had that issue, and solved it by some sort of configuring in asterisk/dahdi
16:20.46billmaniaI understand from http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql that the "uniqueid" isn't unique at all.
16:21.24leifmadsenbillmania: it's not globally unique, no
16:21.54billmaniaIs 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?
16:23.39leifmadsenwell callerid is certainly not unique
16:24.02leifmadsenyou can do things like "forkCDR" and such that would fork the CDR at various points in time
16:24.18billmanialeifmadsen: Caller ID for a single inbound call can change during the course of the call?
16:27.27fauxalliancebillmania, http://lists.digium.com/pipermail/asterisk-bugs/2010-July/082244.html
16:29.43billmaniafauxalliance: 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.
16:30.13carrarLet us know when you complete that drawing so we can see i
16:30.14carrart
16:30.39billmaniacarrar: That's what I thought. :-(
16:31.13fauxalliance~book @ billmania
16:31.25fauxalliance~thebook
16:31.25infobotAsterisk: 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
16:32.13billmaniafauxalliance: I have an electronic copy of "Asterisk: The future of telephony" 2nd edition. Am I hopelessly out of date?
16:32.17fauxalliancebillmania, http://callflow.sourceforge.net/  just add a capture
16:32.50fauxalliancebillmania, i'd borrow a new copy and make a small donation to the author
16:32.57billmaniaInterestingly, written by someone named Leif Madsen.
16:35.09billmaniaPrepares to join the asterisk priesthood by studying the holy book of asterisk
16:37.12fauxalliancebillmania, fwiw, it's the only way in
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16:37.39billmania:-D Thanks.
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16:50.36cjcarrar: I did?  what list?
16:50.50cjcarrar: I thought I was already famous...?
16:51.26carrarsix list
16:51.30carrarof attendies
16:51.42cjoh, nice.  Just under the wire on that one.
16:52.12cjnow all I need is an AS number, I guess.
16:52.26cjoh, and a direct allocation would probably be helpful.
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17:06.35hrnkc_Hi, is it possible to turn off NOTIFY messages that are sent where user has voicemail?
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17:10.42Eitanso 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?
17:10.47Eitani would just go streight into nic?
17:12.12leifmadsenEitan: well your terminology is a little all over the place -- if it is SIP, it's not a T1
17:12.22Eitan<PROTECTED>
17:12.23Eitanlol
17:12.34leifmadsenyou are correct in that you can get 20 channels of SIP from a provider, which runs over the network (internet)
17:12.35Eitangot it, t1 would then need the pri cards
17:12.45leifmadsena T1 is a type of circuit
17:13.15Eitanok, reading chapter 7 right now... right my understanding was that SIP could be sent over any internet connection
17:13.27Eitanit just so happens i was looking to get t1s to do this, cause thats the only thing avaiable in our area
17:13.44leifmadsenthat's fine, you would be using a data T1 that would go into your router or whatnot
17:14.07Eitangot it :)
17:14.09leifmadsenthen it'd just be an ethernet connection locally to your router, and asterisk would speak over that
17:14.11Eitanthanks for the clarification
17:15.32Eitanso would it be accurate to say that SIP is more up to date technologically. or perhaps where the future of VOIP is going?
17:15.48leifmadsenSIP is VoIP
17:15.50leifmadsena T1 is not VoIP
17:15.54Eitanok
17:15.55Eitan:)
17:16.00leifmadsenT1 is a traditional telephony connection
17:16.04Eitani c
17:17.06Eitanjust got confused on chapter 7
17:17.20leifmadsenyou should find a starter book on telephony terms and how networks work. The Newton Telecom Dictionary would be useful as well.
17:17.52Eitanok, i will pick that up, i was told to purchase asterisk the definitive guide to get started
17:17.53Eitanso i did
17:18.01Eitannow i know what my next purchase is
17:18.36leifmadsenit is the definitive guide for asterisk, but not the definitive guide for telephony :)
17:18.54Eitan:)
17:19.05Eitani got myself a good amount of reading to do
17:19.11FreeaqingmeEitan, voip is not a replacement for traditional telephony
17:19.41Freeaqingmethere will always be some sort of ATM network
17:20.08Eitanwould it be wise though to developt a call center strictly using voip?
17:20.18Freeaqingmesure
17:20.35Eitansmiles
17:20.35Freeaqingmedepending on your location and size it may not be the most cost effective option to use a voip trunk
17:20.36Eitanok
17:21.01Eitanthe hardware costs, as far as my understanding are much lower with voip
17:21.10Eitanespecially because we already have the servers
17:21.12Freeaqingmeyes, they are
17:21.32Freeaqingmebut (at least in the netherlands), for larger setups it's cheaper to use an isdn connection than to use a voip trunk
17:21.46Freeaqingmebut as said, that may vary and depend on your location and needs
17:21.50Eitanright
17:22.09Eitanit was slightly more expensive to go with voip than t1 per month
17:22.17citywokEitan: we run a call center 100% voip, but we're putting in a T1 b/c we're tired of voip carrier reliability
17:22.31Eitancitywok: that was my plan
17:22.34Eitani wanted to have a back up
17:22.38Eitanincase things went bad
17:22.40FreeaqingmeEitan, yeah, it's great to start
17:22.43citywok(small call center), we'll run t1 as primary, and anything beyond the 24 channels will go out sip
17:22.50Eitangot it
17:23.03Eitandoes it make a difference that i will be running voip over a T1?
17:23.03leifmadsencitywok: 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
17:23.05citywoka long time ago we had 4 T1's with our old pbx, then we went pure sip w/ asterisk
17:23.09Freeaqingmethe problem with T1 is that there's no real way of getting it redundant
17:23.17leifmadsenEitan: the T1 makes no difference, it's just a data connection
17:23.23leifmadsenthe reliability of the data connection is the problem
17:23.25citywokleifmadsen: i like sip it's soooo much easier. but the 3 providers i use all dick me around and piss me off.
17:23.34leifmadsencitywok: ya I know how that is
17:23.38Eitan@leifmadsen: i see, the actual voip provider is the problem
17:23.44citywokflowroute is really pissing me off with their fraud detection
17:23.52citywokevery time we call a new country they block us for unusual traffic
17:23.54Freeaqingmefraud detection?
17:24.00Freeaqingmelol
17:24.01Freeaqingmefail
17:24.05_Corey_Providers themselves are usually the weak link in this chain... building redundant paths is relatively trivial
17:24.11citywokthey've been "working on it" for months
17:24.18Freeaqingmecall every country just once today, and you're rid of that problem :P
17:24.24Eitanso i shoul dbe very careful as to whom i pick as our provider
17:24.34citywok_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
17:24.37DaneoShigaanyone knows a google place to find resources about phpagi?
17:24.41Freeaqingmeif it's critical for the business: yes Eitan
17:24.47DaneoShiga/google/know
17:24.53Eitancitywok: that was one of the proivders on my shortlist, bandwidth.com
17:24.59FreeaqingmeDaneoShiga, the source code?
17:25.03citywokyea they're a bunch of retards
17:25.08Eitaneeech
17:25.12Eitanthey just lost a costumer
17:25.18_Corey_citywok: Calling Bandwidth a "carrier" is pretty generous
17:25.21citywokthe 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.
17:25.25Eitanwhat about Cbeyond
17:25.28Eitanthey are next in line
17:25.39citywok_Corey_: i said provider :P -- they're just an L3 reseller.
17:25.43DaneoShigaFreeaqingme: 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
17:26.02citywokand 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...
17:26.29Eitani wonder what voip provider in my area is very reliable...
17:26.32FreeaqingmeDaneoShiga, I'm busy on developing my own agi component for the zend framework, I could give you a link if you're interested
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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
17:27.29citywoklol, 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.
17:27.39citywokevery month they would "fix" it
17:27.55DaneoShigaFreeaqingme: would be great, but i don't think boss is interested in changing things now =/
17:27.57citywokevery 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
17:28.29DaneoShigaFreeaqingme: 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
17:28.31citywok_Corey_: we don't do 1M/mo
17:28.35Eitancorey: they pretty good?
17:28.40citywokso we don't meet the min commit to even talk to them
17:28.42leifmadsenDaneoShiga: you use the 'h' extension for that
17:28.50FreeaqingmeDaneoShiga, what do you mean?
17:28.57leifmadsenya, I'm a bit confused too
17:29.01Freeaqingmeyou 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
17:29.58DaneoShigaFreeaqingme: no, i have a script on one server that Dials to certain queues on others servers based on dtmf and some comparisons...
17:30.13citywok_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
17:30.22FreeaqingmeCode is here btw DaneoShiga, it's a WIP: https://github.com/Freeaqingme/zf2/tree/asterisk/library/Zend/Telephony/Asterisk
17:30.56DaneoShigagonna take a look ^^
17:31.59DaneoShigalet me pastebin my script, one moment
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17:35.04DaneoShigahttp://pastebin.com/4ht7pPXC
17:35.11ruben23hi guys buy default- h.323 are not packgae or bundle by asterisk - it should be separately installed or compile..?
17:35.42DaneoShigaFreeaqingme: http://pastebin.com/4ht7pPXC i need code after line 196 to always execute, but it sometimes doesn't...
17:37.25FreeaqingmeDaneoShiga, also if the caller has hung up already?
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17:39.04DaneoShigaFreeaqingme: if possible, yes
17:39.25Freeaqingmethen you could look at dead agi
17:39.37Freeaqingmebut I'm not sure that allows to dial if the caller has hung up
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17:39.54Eitanwhat about ATT's voip service
17:40.10gruvfunkwhat voip service, didn't they call it quits on CallVantage?
17:40.46Eitanna, doesnt look like it according to their site
17:40.52Eitanon the phone with their buisness people now
17:40.54Eitanlets see what they sau
17:40.56Eitansay
17:40.57DaneoShigaFreeaqingme: well, my biggest problem is when the code after dial don't work...
17:42.24hrnkc_Hi. Does anyone know if is it possible to turn off NOTIFY messages that are sent where user has voicemail?
17:42.47DaneoShigahmm, it is using dead agi...
17:43.03_Corey_jrnkc_: Set mailbox= to blank on their sip config
17:43.10leifmadsenor don't set it at all
17:43.14hrnkc_Corey> thanks a lot
17:45.21hrnkc_Ive got default asterisk configuration. The weird thing is that everytime i make a call, asterisk sends NOTIFY message with number of voicemails...
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17:47.45cusco_hi...
17:48.13gruvfunkhey hey
17:48.15m4xxlol "Hacking indications.conf for Fun and Profit"
17:49.17cusco_when calling isdn I would like to know wich telco does the destination belong to
17:49.17m4xxis that a common saying or is that paying homage to the "smashing the stack for fun and proffit"
17:49.21cusco_how can I know that
17:49.28leifmadsenm4xx: it's just something we made up :)
17:49.32malcolmdruben23:  you should do chan_ooh323 from the addons directory in make menuselect
17:49.33DaneoShigaFreeaqingme: 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...
17:49.33cusco_I was reading about q.931
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17:49.53FreeaqingmeDaneoShiga, afaik the idea of deadagi is that it  continues running after it's hung up
17:50.05Freeaqingmebut you'd have to check if it doesnt error if you then try to dial out without a caller
17:50.09Freeaqingme(I think it does/will)
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17:50.34m4xxso no homage to: http://www.phrack.org/issues.html?id=14&issue=49 ? ;[
17:50.41m4xxit would have only made it cooler ;x
17:50.53DaneoShigaFreeaqingme: I see... gonna take a look at it, thanks :)
17:51.08Freeaqingmeyw
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17:56.57porcheHi
17:57.18porcheI am looking for a number validation service that runs over SS7
17:57.48porchedoes any one know one?
18:01.19ruben23malcolmd: i see xxx_chan_ooh323  im using asterisk 1.6
18:05.11Freeaqingmeporche, I think no such service exists
18:05.21Freeaqingmethat could be considered to be one of the weak points of ss7
18:07.49ruben23malcolmd:? it means its not package with asterisk 1.6..?
18:08.53porcheThank you Free
18:09.05porcheany other option than ss7?
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18:11.43cusco_i
18:11.45cusco_hi
18:11.49cusco_regarding moh
18:12.01cusco_how do I choose the order of files in a class?
18:12.11Eitanwhats the call capacity on the free version for simultaneous calls? is there one?
18:12.28cusco_in musiconhold.conf I have mode=files and directory=/var/lib/asterisk/moh/Lusomundo
18:12.36Kobazfree version of asterisk? no limits
18:12.49cusco_now Im adding files in alphabetic order say moh1.al moh2.al moh3.al
18:13.02cusco_but then the 3 goes on top when I do 'moh show files'
18:13.33Eitanthats what i thought
18:13.37Eitancbeyond rep is trying to sell me some shit
18:13.41Eitansaying im gonna have to pay for licences
18:13.43Eitani 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?
18:15.11Eitanthey are so confused
18:15.13Eitanand they are confusing me
18:16.04cusco_http://paste.debian.net/116005/
18:16.06cusco_:|
18:17.17m4xxeh
18:17.21m4xx[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.
18:17.32m4xxany ideas?
18:18.07cusco_why not use MixMonitor() ?
18:18.15m4xxbecause that didn't work at all
18:18.23cusco_works here...
18:18.27m4xxi'm using freebsd
18:18.34m4xxnot sure if it makes a huge difference
18:18.58m4xxmixmonitor doesn't produce any errors, but it doesn't create any file either
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18:26.12m4xxis there anything that would cause mixmonitor to not work on freebsd?
18:26.30m4xxit doesn't work with either absolute path or relative path
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18:31.26Kobazmixmonitor should be used in all cases unless you specifically need to have individual legs of the call being recorded
18:31.43Eitanso $800 bucks for 30 call paths
18:31.54cusco_how do I set order of several moh audio files??
18:31.57KobazMonitor() is part of the audio bridging thread... and if you have high disk io, your call quality will suffer
18:32.41m4xxhttp://paste2.org/p/1399169
18:39.30JonathanRosem4xx:  I'll have a quick look to see if that dialplan works as it should on my platform now.
18:39.49m4xxJonathanRose: thank you =]
18:40.50cusco_please.. its getting on my nerves
18:40.55cusco_moh class file order
18:41.13cusco_http://paste.debian.net/116007/
18:42.19m4xxit's a hack, but have you tried prepending a number to the file names
18:42.31m4xx1-fileblah.wav 2-filebalh.wav
18:43.16cusco_yes
18:43.24cusco_I tried prepending and sufixxing
18:44.20cusco_look: http://paste.debian.net/116009/
18:44.23cusco_I don't understand
18:45.20_Corey_cusco_: If it's important, you could always merge the files together
18:45.32cusco_I think I will
18:45.36cusco_it is important yes lol
18:45.55JonathanRosem4xx:  Yeah, doesn't work in any build.
18:46.15cusco_?
18:46.19m4xxso am i doing something wrong or is it broken?
18:46.37JonathanRoseI  think the channel needs to connect to something first.
18:46.48m4xxso answer then mixmonitor?
18:46.52JonathanRoseNo
18:47.02cusco_no
18:47.06cusco_mixmonitor first
18:47.07cusco_then answer
18:47.10cusco_o.O ?
18:47.16JonathanRoseIt's just that there isn't anything listening, so there isn't an audiohook for it to bind to.
18:47.27cusco_yes I thought about the same
18:47.28m4xxright now i'm using originate, then passing it off to that dialplan
18:47.31cusco_try playback ()
18:47.47JonathanRoseIf it doesn't bind to anything, it doesn't start the filestream.
18:47.59m4xxi'm sorry, i'm still learning what do you mean by that?
18:48.04cusco_no audio
18:48.12cusco_thus no need to record it
18:48.20m4xxbut i know there's audio
18:48.22cusco_try dial() some peer
18:48.26m4xxif i use the monitor
18:48.30m4xxthere is audio
18:48.30cusco_ow...
18:48.52JonathanRoseThe channel isn't sending or receiving audio.
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18:49.00JonathanRoseAnd monitor works a little differently.
18:49.15m4xxi'm positive there's audio being received, if i do Monitor(wav,wtf,m) i do get the in and out legs
18:49.24m4xxthe in leg does contain audio
18:50.12m4xxcusco_, that pastebin was your musiconhold.conf ?
18:50.20kdashi, 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?
18:50.54kdasmy cell is using a sip client by the way
18:51.26cusco_m4xx: was moh show files
18:51.30cusco_cli command
18:51.35m4xxah
18:52.07cusco_kdas: on dialplan on hangup get DIALSTATUS - if NOANSWER do something
18:52.20cusco_like writting to a text file or sending a sms :p
18:52.22cusco_or email
18:52.30JonathanRosem4xx:  I think that's because monitor establishes the connection at that point.  MixMonitor doesn't do that.
18:53.09JonathanRoseTell 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.
18:53.14JonathanRoseSo MixMonitor(...)
18:53.16JonathanRoseAnswer()
18:53.22cusco_JonathanRose: actually mixmonitor() has option b
18:53.32cusco_to only record while channel is bridged
18:53.36JonathanRoseThat just makes it wait until the call is bridged.
18:53.38cusco_if he's not using that ...
18:53.42JonathanRoseHe isn't.
18:53.53cusco_it should record anyway.. right?
18:53.56m4xxif i use originate, then answer() in the dialplan, isn't that bridged?
18:54.08cusco_I guess so
18:54.11kdascuscco_: umm cant i just get a missed call notification when sip client connects? is there some kind of sip message?
18:54.23kdaswriting to a file seems pointless
18:54.34JonathanRoseNot really.
18:54.51JonathanRoseThe bridge option is more for stuff like where you have a menu that'll connect you to another line...
18:54.54cusco_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
18:55.09JonathanRoseThen it won't start monitoring until the phone is picked up on the other end...
18:55.11cusco_kdas: don't write to file, send an e-mail then
18:55.39kdasnot what i want but i get it
18:55.50m4xxi thought with originate it wouldnt get to the dialplan untill the call was picked up
18:56.19m4xxat least not priority 1
18:56.42JonathanRoseAs I was saying, try MixMonitor(), Answer(), and then Background(silence/1).  That should make it work.
18:56.51kdasanother 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?
18:56.55m4xxwill do
18:57.03Eitanalright i just ordered 30 trunks with cbeyond
18:57.07Eitanlets hiope they know what they are doing
18:58.20m4xxwow, that worked
18:59.37m4xxshould mixmonitor(), answer(), sendtdtmf() work the same?
18:59.49m4xx*senddtmf
18:59.56JonathanRoseIf it involves sound that would be put onto the calling line, it'll work.
19:01.12leifmadsenheh, good luck ;)
19:01.12leifmadsenwhen they work, they seem to be fine....
19:01.12leifmadsenhave a customer using them, and don't hear from him too much anymore
19:01.33m4xxthis is my original dialplan: http://paste2.org/p/1399210
19:02.07m4xxwhich didn't work, doesnt the senddtmf send audio over the line?
19:04.29JonathanRoseI'll check real fast.
19:06.38JonathanRoseWell, apparently I'm wrong in my assumption then.
19:06.52JonathanRoseIt does send audio.  I'm not sure if it's just different because it's DTMF or what.
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19:08.58m4xxhaving to add the play silence seems really hacky to me =[
19:09.29JonathanRoseYou usually won't have to though.
19:09.34*** join/#asterisk kdas (8898be77@gateway/web/freenode/ip.136.152.190.119)
19:09.57kdasanyone have ideas how to implement sip ims with asterisk?
19:10.58JonathanRoseTo 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.
19:11.17m4xxi've got 2 projects
19:11.20cusco_another question...
19:11.29m4xxone is to do something similar to checking voice mail off site
19:11.37cusco_can I make it so taht periodic-announce for a queue only plays once ?
19:11.43cusco_not every 60 seconds
19:11.52cusco_but only the first time
19:12.05kdas<PROTECTED>
19:12.07m4xxcalling  a number, entering some digets, hanging up
19:12.20m4xxi just want to monitor so i can verify that things are working
19:12.23leifmadsencusco_: some info should be here: http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html#ACD_id288932
19:12.54m4xxi'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
19:13.31ruben23hi is chan_ooh323 bundle ot included on what version of asterisk..?
19:13.49leifmadsenruben23: 1.6.2 and later, but the good changes are in 1.8 and later
19:13.52leifmadsenwe already had this discussion
19:15.23m4xxwould it not make sence to (mix)monitor those two type of applications?
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19:22.33ruben23leifmadsen: because when i do make menu select --> xxx chan_ooh323 , i see like this i cant select it to install, any idea..?
19:23.02Qwellruben23: You don't have prerequisites installed...  Look at the bottom of the screen.
19:24.09ruben23ok
19:27.53ruben23<PROTECTED>
19:28.16QwellWhat does it say at the bottom of the screen?
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19:35.08WiretapSevenleifmadsen, quick question before I run out the door if you don't mind
19:35.31ruben23Qwell: just a second
19:35.32WiretapSevenhow long is likely before this is in a release? https://issues.asterisk.org/view.php?id=13996
19:37.55ruben23Qwell: i see this---------------->The NuFone Network's OpenH323 Channel Driver Depends on: openh323(E)
19:38.42JonathanRosem4xx:  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.
19:39.07ruben23this si for chan_h323, but other guy here on the forum mention this--> chan_ooh323 , what could be the differetnce between the two..?
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19:39.35JonathanRoseI'm thinking that mixmonitor actually just plain will not capture the DTMF as a result.
19:39.49JonathanRoseWhich might be the intent actually.
19:40.12WiretapSevenand I'm gone :/, oh well, if there's a response I'll see it tonight
19:41.32m4xxJonathanRose, why wouldn't it work if only audio was received?
19:42.08JonathanRoseBecause the audio never gets into the audiohook.
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19:42.41JonathanRoseIf the audio isn't in the audiohook, mixmonitor isn't going to see it.
19:42.58JonathanRoseAnd DTMF doesn't get into the audiohook.  It's weird.
19:43.11leifmadsenWiretapSeven: I have no answer for you
19:43.25leifmadsenit's a feature, and is probably very low on the list of items to get resolved
19:43.44m4xxwhy not provision your cisco phone with dhcp and tftp?
19:44.41m4xxn/m i read wrong
19:51.58cusco_leifmadsen: so basically I set a limit in the queue at 60 seconds, I playback the announcement, raise QUEUE_PRIO and queue it again
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19:53.57gruvfunkcusco_ how about just a normal Playback prior to entering the Queue?
19:54.13gruvfunkand skip periodic-announce
19:54.20cusco_nope
19:54.25cusco_I need it at 55s
19:55.11cusco_Im hoping QUEUE_PRIO works accordingly
19:56.40cusco_queue is suposed to last for 3 minutes
19:57.14cusco_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
19:57.21cusco_to leave a contact
19:57.33cusco_and I wish to play that only once
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20:52.04sskylesIs there a PDF or text file anywhere that fully documents Asterisk configuration in more detail than just looking at the config files themselves?
20:52.28Freeaqingme~thebook
20:52.28infobotAsterisk: 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
20:52.30Freeaqingmesskyles, ^^
20:52.34gruvfunk:)
20:52.48sskylesGreat!
20:52.50Freeaqingmewhatever your question is: ~thebook usually suffices as an asnwer
20:53.06gruvfunkif not, google, or ask in here
20:54.51sskylesWell, 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.
20:56.51leifmadsensskyles: yes there is a discussion abut dundi and what it is and how to use it in that book as referenced
20:57.07leifmadsenfor the most part that book should answer any general questions like that, if not, let me know!
20:57.27sskylesIt looks like exactly what I need, thanks again.
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21:02.21karen_m!newbook
21:02.29karen_m~newbook
21:02.29infobotPlease see ~thebook for more information about Asterisk: The Definitive Guide
21:02.29karen_m.newbook
21:02.32karen_m~thebook
21:02.32infobotAsterisk: 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
21:03.07karen_mis there a pdf for trhat book available?
21:03.21leifmadsenkaren_m: for purchase, yes
21:03.31karen_mreading online lol :)
21:03.32simplydrewhmm. I should download that to my iPad for some reading at some point
21:03.34simplydrewmakes a note
21:04.15leifmadsenthere are kindle, etc... editions as well
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23:44.05sezuanHas Packet2Packet bridging been remove since Asterisk 1.4?
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