IRC log for #asterisk on 20120710

00:06.31jpsharpYou have to pay for it somehow.  Either by putting up with ads or paying a per month/per minute charge.
00:18.08jgowdyYou have to pay to reach the PSTN
00:18.19jgowdyIf you want to call someone else in a peer to peer fashion
00:18.37jgowdyYou can do that for free using just Internet
00:18.48jgowdyBut that's not calling a phone number on the PSTN
00:19.09jgowdyThat's starting a SIP session with another IP address, followed by the associated RTP stream
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00:34.38gustook
00:34.44gustothis g722 is really cool
00:35.10gustoi thaught it were bad, but then i noticed, that i still had echo cancellator on and that is famous for worsen the quality
00:35.27gustobut now with disabled echo cancellator it's very good
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00:39.02gustoone can listen radio through it
00:39.36gustoi am testing codecs now that way, i have a line-out from a radio (dab+) and i am looping it back through asterisk with app_echo
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01:52.58ZenMasterSo, people who use TrixBox, use something like Vici dial to create a outbound campaign?
01:54.02ZenMasterI am just testing and learning. I have a TrixBox machine setup, and wanted to give setting up a predictive enviroment a go. Then eventually live call transfers.
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01:56.17darrenloobyIf my asterisk is running from a database. And I'm using the queues table - how do I get the AgentCalled event to trigger? It doesn't seem to be doing it.
01:56.33darrenloobyI'm in 1.8.7.1
02:16.25ZenMasterMy other question is, would a crm allow agents to dispositio calls and what not? Lead, Hungup, DNC?
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03:06.30UnixDevis there some way to use page without dahdi on asterisk 1.8? I see this has been done in asterisk 11, but I don't think its ready for prime time yet
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03:31.28UnixDevwhat version of asterisk would this commit effect? http://svnview.digium.com/svn/asterisk?view=revision&revision=358730 ?
03:37.20jpsharpAsterisk 11 eventually.  Or "trunk" if you pull the source tree from SVN.
03:41.23UnixDevahh, got it
03:41.32UnixDevany chance that can be back ported to 1.8 ?
03:51.49jpsharpLooks to be possible, but I can't guarantee it.
03:52.00ZenMasterAnyone have expierence with astercc?
03:52.42UnixDevjpsharp: I can back port it if it will go into trunk… another option is to create app_pageconf
03:53.04UnixDevjpsharp: that way it would not break existing installations using 1.8… what do you think?
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03:54.52jpsharpZenMaster: I used it a few years back and it was kinda kludgey, but that was a few years ago.  So, grain of salt and all that.
03:55.25ZenMasterI was just trying to set something up for prdictive with trix box as a test.
03:55.34ZenMasterI got it to install, and now I can't login wierd.
03:55.38jpsharpUnixDev: If you're worried about breaking existing installations, I'd go with the separate app.
03:55.40ZenMasterMight be a sign for me to leave this office.
03:55.58ZenMasterI just want to see what it looks like.
03:56.00UnixDevjpsharp: I'm just worried about getting the patch into trunk, I dislike maintaining my own patchsets
03:58.08ZenMasterGoing home.
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04:25.11linociscohi all
04:25.51linociscoi have Dlink DI-524 wifi router. Can I get asterisk firmware for that?? most said WRT54G of Linksys is compatible with asterisk firmware but I dont know about my Dlink router
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05:02.47kaldemarlinocisco: asterisk is not a firmware. if for example openwrt supports your router, you can run openwrt on it and install asterisk in openwrt.
05:03.12linociscokaldemar, how can I check?
05:03.51kaldemaropenwrt project has a device list on their web page
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05:39.49KNERDhow to get the jabber and gtalk modules to load upon start?
05:43.54kaldemarKNERD: you've done something that prevents them from loading. see that you don't have noloads in asterisk's modules.conf.
05:44.57KNERDokay..let me look..
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05:46.18KNERDkaldemar: yeah, that was it..thanks
05:46.40As001Hello, 4 of my agent channels hang in cli in state down and Application none at cli. I can not hangup them with hangup request. I restarted Asterisk and they have gone away but I wonder why would some channel get in that state ?
05:47.14As001All of them had elapsed time unbelivable big like 200 hours etc...
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06:02.45Jo_BellHi. If I set the destination of 2 audio devices to be a single SIP device URI, will asterisk mix the audio and send it in a single channel?
06:02.50AucklaAnyone awake tonight? :D
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06:11.09kaldemarJo_Bell: asterisk will only audio from two channels if you connect the channels to a conference.
06:11.30kaldemars/only/mix/
06:12.42bulkorokhi
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06:28.26AucklaCan't seem to get astercrm to allow me to login via web interface?
06:28.35AucklaAnyone have any expierence with that?
06:42.09KNERDIs there a CURRENT method to setup GV? It seems the ones a year or older just outright fail, and I cannot find one newer
06:43.49AucklaI just don't get it. :&*(
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06:53.20ChannelZWell the GV integration isn't very old relatively speaking
06:56.08KNERDwell, it has not functioned in the past 1.5 years..I am trying again...still fails
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07:06.16Jo_Bellinfboot and kaldemar: ok got the idea
07:07.25kaldemarinfobot is a bot, not a person.
07:15.20Jo_Bellkaldemar: fine. :-)
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07:23.32khurramhi every one
07:24.08khurrama newbie just need initial understanding about asterisk..
07:24.49khurramdo i have to have some telephone line connection to run asterisk server
07:27.36Samuel_CTXkhurram: no, you can use everything with SIP phones (and trunks to VOIP providers)
07:28.20khurramVOIP provider so i need to have VOIP service first from some one right ?
07:29.10khurramhow can i know if my table phone is SIP phone ? do i require to buy hardware to convert this to SIP ?
07:29.19kaldemarkhurram: no. you don't need any service or special hardware unless you want to connect to public telephony network.
07:29.59kaldemarkhurram: what is your table phone?
07:30.11khurramthe normal landline phone
07:30.39kaldemarthen it is not a VoIP phone.
07:31.17kaldemarbut you could get an ATA that speaks SIP to asterisk and lets you connect an analog telephone to it.
07:31.33khurramATA ?
07:31.42kaldemar~ata
07:31.42infobothmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
07:32.28kaldemarFXS is for connecting phones, FXO is for connecting to an analog line.
07:34.22khurram<PROTECTED>
07:34.31khurramhmmm
07:36.17khurramso if some dials my extention then i can programme Asterisk to pick the phone after 5 dial tones and play the message i have recorded and then email me or sms me about that i have missed a call on my landline from bla with recorded mesg bla bla. Am i right?
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07:39.17khurramkaldemar: thanks i watched the video and it is quite impressive but i think the turorials are missing for new developers like me from php background
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07:41.40kaldemarthe video? the tutorials?
07:43.21kaldemar~book
07:43.22infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
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08:11.38bikcmpi'm configuring asterisk 1.8 on a debian server.
08:11.40bikcmpoutgoing calls work fine.
08:11.42bikcmpincoming calls are either "nobody hears anything", or "caller hears person but can't speak"
08:11.51bikcmpi'm almost positive it's NAT.
08:12.23bikcmp<PROTECTED>
08:12.25bikcmpeout reached on transmission 6a966a664f191e641e0ed79b03f0f85c@66.54.140.46 for s
08:12.30kaldemar~sipnat
08:12.30infobot[~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet.
08:12.59bikcmpyeah.
08:13.03bikcmpkaldemar: we've been trying that.
08:13.18kaldemarwhat exactly?
08:13.26bikcmpkaldemar: give me 10 secs, i'll pull the config out on you
08:14.21darrenloobyIf my asterisk is running from a database. And I'm using the queues table - how do I get the AgentCalled event to trigger? It doesn't seem to be doing it.
08:16.15bikcmpkaldemar: okay
08:16.18bikcmpkaldemar: i've enabled nat=yes
08:16.36bikcmpkaldemar: set qualify=yes
08:16.57bikcmpkaldemar: localnet=192.168.1.1/255.255.255.0
08:16.59kaldemar~pb
08:16.59infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
08:17.04bikcmpkaldemar: that'll work then
08:17.11bikcmpkaldemar: how much you want?
08:17.13bikcmpthe whole file?
08:17.25bikcmpsip.conf, that is?
08:17.41kaldemar[general] and the peer for ipkall
08:18.15kaldemara sip debug of a call would tell more about what is wrong.
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08:20.18bikcmpkaldemar: mind if i pm it?
08:20.34bikcmpi think that i removed the secrets/passwords, but just in case.
08:21.12kaldemargo ahead
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08:22.23kaldemarexternaddr/externip is missing from your configs.
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08:23.53kaldemaralso, change the provider entry from fried to peer.
08:24.35kaldemar# is not a comment character in config files, use ; with canreinvite=no.
08:24.47ziz212hi friends, how could we mute all except one who speak in a conference call ?
08:24.53bikcmpok
08:25.19bikcmpkaldemar: ok
08:25.24bikcmpkaldemar: so what should canreinvite be?
08:25.30bikcmpnonat?
08:26.08kaldemarmore like "no" in your case.
08:26.56kaldemaralso, put your register statements under above all peer definitions or they will not work.
08:27.24bikcmpput the register statements?
08:27.30bikcmpkaldemar: what do you mean by register statements?
08:27.31kaldemarthey belong under [general]
08:27.55kaldemarthe "register => ..." lines
08:28.18bikcmpok
08:28.28bikcmpnow, do you still need a log?
08:29.12bikcmpkaldemar: where should externip be
08:29.26kaldemarexternip/externaddr goes under [general].
08:29.37kaldemarthe sample sip.conf is very helpful with these things.
08:29.54bikcmpkaldemar: yeah
08:30.06bikcmpkaldemar: just trying to double check stuff, since i've literally spent hours on this.
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08:30.25bikcmpexternaddr isn't needed if externip is given
08:30.26bikcmpright?
08:32.28kaldemarexternip is the old name for externaddr. their meaning is the same. i'd use the latter.
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08:33.45bikcmpkaldemar: ok
08:33.49bikcmpkaldemar: just modified what you told me
08:33.50bikcmpstill nothing
08:33.52bikcmpsame issue.
08:33.58bikcmpi have the sip debug logs, but god there are a lot
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08:35.15kaldemarpastebin your current configs and the output
08:35.25bikcmpsure
08:35.33bikcmpwell
08:35.34bikcmpdarn
08:35.39bikcmpkaldemar: here's the problem.
08:35.55bikcmpkaldemar: the `sip set debug` command actually put out /so much output/ that it's no longer in my backlog.
08:36.57bikcmpkaldemar: want just the config?
08:38.05*** join/#asterisk din3sh (~din3sh@41.212.203.212)
08:38.10din3shhello all
08:39.46kaldemarstart with the config. if you scrollback is not enough, use "asterisk -vvvr | tee /tmp/sip_debug.txt" for example to get it to a file.
08:40.30bikcmpkaldemar: i'll throw you the config in hopes you see the problem
08:42.42bikcmpkaldemar: pm.
08:42.57din3shanyone ever implemented click-to-call on lotus notes working with asterisk?
08:44.01bikcmpkaldemar: you get my pm's?
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08:50.20ziz212can someone hlep me in my question? Can I repost my question?
08:54.29din3shziz212: dont ask if you can ask a question dude
08:54.45din3shpost your question, if someone can help, he will
08:55.15ziz212ok i will repost it..  hi friends, how could we mute all except one who speak in a conference call ?
08:56.36din3shwhat if everybody is talking at the same time
08:56.40din3shhow will you decide which one should be talking, and which muted?
08:57.33kaldemarziz212: which conference app are you using?
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09:15.01LantiziaLo, for some reason *8 group call pickup isn't working - it just hangs up (both peers are in the same call/pickup group), if I turn on sip debug on the peer dialing *8 I see it does dial *8@<ip of pbx> in the To: line - but ultimately then checks if *8 is in from-internal context... but it's an internal function isn't it?! (yes pickupexten is set to *8 in features.conf).  Any ideas?
09:18.23LantiziaI can provide logs! Theres like nothing on the web I can find about this - this is day 4 of me asking in this channel
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09:19.50wdoekesLantizia: perhaps features.conf / res_features is not loaded properly? check the core show features (or something) and make sure it lists *8
09:20.15Lantizianah it is - i watched asterisk start up this morning ... both the app and the config files are loaded
09:20.16kaldemar"features show"
09:20.42Lantiziayeah it shows
09:20.49wdoekes(b.t.w. the relevant bit is *8@<ip> in the request-uri line (the first one, after INVITE))
09:21.18Lantiziaumm i think i may have found it - they're in call/pickup group 71 - is the max 63?
09:21.23bikcmps/53
09:21.24bikcmp4/54
09:21.30wdoekesyes it is
09:21.44Lantizia(btw I've been saying they're in 71 since thurs in here and no one has picked up on it :P)
09:21.48Lantiziawhy 63?! of all numbers
09:21.55wdoekesbits
09:22.02Lantiziais 0 allowed?
09:22.08wdoekesprobably
09:22.14Lantizianon-integers? :P
09:22.29wdoekesfloats? no
09:22.36Lantiziadamn
09:22.40wdoekes0.71
09:22.54Lantizia0.71 ?
09:22.57wdoekes;)
09:23.05Lantiziayeah a non-integer :P
09:25.26Lantiziasee our system is multi-tennant - the callgroup number is the room number
09:25.32Lantiziafrom about 1 to 200 ish rooms
09:25.44Lantiziaeach room a business - wish we could increase the number of call groups
09:27.00wdoekesLantizia: see the Pickup application and friends
09:27.21Lantiziaoh i know i could do it with dialplan - just a load of work is all
09:27.45wdoekesperhaps it's easier if you just decrease the number of rooms
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09:31.26Jo_BellAll. Where can I get 1 / a few examples on getting a number of SIP devices into a conference using Asterisk?
09:31.44Lantiziawdoekes, how about if I put each tenant in their own context? do you think it's possible to write a *8 function that'd only pick up the oldest call in that context?
09:32.46kaldemarJo_Bell: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10
09:35.02wdoekesLantizia: PICKUPMARK sounds like what you want..
09:40.52Lantiziawdoekes, err reading - not altogether following but reading
09:41.17Lantiziathis looks like it has nothing to do with contexts though
09:43.02wdoekesI didn't say that. contexts are used when the device calls.. you could set the to_be_picked_up_mark there, but you could do that using a setvar= on the device too
09:43.31Lantiziableh?!
09:43.34wdoekesyou'd still need to set the right pickupmark on incoming calls
09:44.55ziz212I am using elastix and I am trying to find out conference module in elastix? I really having problem to find out the name of it
09:45.19wdoekes~elastix
09:45.19infobotrumour has it, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org
09:46.49ziz212Ok pls let me now the conference module to try for my feature?
09:48.05ziz212Let say if I am using asterisk without any other add-ons
09:48.29Lantiziawdoekes, ok i'm reading about another idea - something about as of * 1.6.2 chan_sip uses the PickUp app by putting a hint in the context?  but it doesn't say much more
09:48.40Lantiziasomething about notifycid
09:49.01kaldemarziz212: meetme, confbridge, app_conference or app_konference. all have different syntax. ask the elastix people.
09:50.35ziz212ok thanks for the help but how could I use that unmute for active spaker and mute or others? Which module support it?
09:51.07Lantiziahe aint giving up is he? :D
09:53.39kaldemaryes, it supports it.
09:56.50wdoekesziz212: meetme has MeetmeMute etc.. manager commands
09:57.26ziz212Thanks for the help. I can try out that and see
09:58.04Lantiziawdoekes, think PickupChan may help me?
09:58.27wdoekesLantizia: if I give you a solution that I think works, why are you trying to find others?
09:59.00Lantiziai'm not sure it does - or at least I don't understand it if it does
09:59.05wdoekespickupchan might work if you your room-people share the same prefix [room71-account1]
09:59.32Lantizianah that's not always true
10:00.19Lantiziaok so pickupmark - are you saying when a call comes in and a peer starts ringing - a special pickupmark variable is set for that one instance of it ringing?
10:00.45Lantiziaand the *8 dialplan would grep all pickupmark's in the list for those starting with particular digits that correspond to that context?
10:01.29wdoekesSet(_PICKUPMARK=71) ; Dial(SIP/room71-account1)
10:01.47Lantiziathen later *8 does what?
10:01.50wdoekes*8 => Pickup(${to_be_picked_up_mark}@PICKUPMARK)
10:02.09Lantiziawhat is 'to be picked up mark' ?
10:02.13wdoekeswhere setvar=to_be_picked_up_mark=71 in [room71-account2]
10:02.56Lantiziaso pickup 71@71 ?!
10:03.13wdoekesI will leave you with 'core show application' and google
10:03.22Lantiziayeah that still makes zip all sense
10:03.39Lantiziaand whats more doesn't record how old a call is - it won't pick up the oldest one
10:05.20Lantiziais PICKUPMARK a list? thus when it picks up 71@PICKUPMARK - it finds the earliest 71 in that list and removes it when picked up?
10:05.44Lantiziaand presumably when that call ends (if not picked up with *8) it removes it from the list also?
10:06.29wdoekesso many questions
10:06.32wdoekeshave you tried it?
10:06.43Lantiziaam I at least CLOSE with that line of thinking?
10:07.01wdoekesI don't know about the pickup order, the rest looks ok
10:07.16Lantiziaso it is a list variable then?
10:07.51wdoekesPICKUPMARK in @PICKUPMARK isn't a variable, technically
10:08.03wdoekesbut you don't need to know that
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10:11.12Lantiziaso perhaps I could set accountcode in each peer to be like a 'customer number' (usually room number) and put something generic in somewhere to use accountcode as pickupmark
10:13.01*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
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10:37.41vahonenHi all. I'm running Asterisk 1.6.2.20 on CentOS 5.8, with 2.6.18-308.8.2.el5xen. When trying to load app_meetme.so, that came when installing asterisk-dahdi, I get this; "Error loading module 'app_meetme.so': /usr/lib/asterisk/modules/app_meetme.so: undefined symbol: __ast_cond_wait". Any hints...?
10:42.45din3shanyone ever implemented click-to-call on lotus notes working with asterisk?
10:48.31Jo_Bellin my /etc/asterisk, i don't see confbridge.conf. Do I have to install anything more ?
10:49.00wdoekesvahonen: then that app_meetme wasn't compiled for the right asterisk version
10:50.45kaldemarJo_Bell: what version are you using and how did you install asterisk?
10:53.16Jo_Bellkaldemar: i got the tar files for libpri, dahdi and asterisk from site and did make install on all
10:53.31Jo_Bellhow to check the version from the CLI prompt?
10:53.57Jo_Bellit shouldnt be any old version. i took the packages in the last 10 days or so.
10:56.25kaldemarcore show version
10:57.13Jo_Bellkaldemar: 1.8.13.0 is the version
10:58.04kaldemarlook under configs/ in the source package.
10:58.16kaldemar"make samples" is used to copy the sample configs under /etc/asterisk
10:58.48kaldemarsee the README
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11:00.10kaldemarthe confbridge app in asterisk 10 is more advanced than the one in 1.8.
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11:02.05kaldemarJo_Bell: looks like the confbridge in 1.8 does not even use confbridge.conf. it is entirely different.
11:05.35Jo_Bellkaldemar: oh. for upgrading to asterisk 10, what should I do?
11:08.24Jo_Bellcan i just take the asterisk 10 tar file and do a make install?
11:08.45kaldemardo a make uninstall in the 1.8 source dir first.
11:09.55kaldemargrab the tar, exctract it, run the configure script, make menuselect to select features (if you know what you want), make, make install.
11:10.27Jo_Belli have 3 folders one for asterisk, one for dahdi, and one for libpri. i assume i have to uninstall only the asterisk..is it correct?
11:11.19kaldemaryes
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11:13.55Jo_Bellkaldemar: where to get the 10 source tar?
11:14.02Jo_Bellsite itself?
11:15.06kaldemarhttp://www.asterisk.org/downloads
11:15.24din3shis certified asterisk free?
11:15.55*** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
11:17.27kaldemardin3sh: http://www1.digium.com/en/products/asterisk/certified-asterisk
11:20.24din3shthnks
11:21.50Lantiziayes but it's kind of pointless
11:22.04Lantiziai.e. pointless if all you need is something free :)
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11:31.10vahonen"Error loading module 'app_meetme.so': /usr/lib/asterisk/modules/app_meetme.so: undefined symbol: __ast_cond_wait", solved. Asterisk was version 1.6, but meetme (asterisk-dahdi installed with yum) was 1.8. Removed a few repos and reinstalled asterisk-dahdi and now it's working.
11:31.32kaldemarLantizia: it's an LTS version with digium phone support and it goes through more testing than the mainline releases.
11:32.34Lantiziakaldemar, i already know what it is
11:32.54Lantiziai spoke at length about it with digium  - time I'll never get back
11:34.32Lantiziait's only purpose is for a completely custom production-level sytem - itsp grade - and you've got BAGS AND BAGS of cash for support
11:35.14Lantiziausing it on an every day phone system is nuts - cheaper (if you want support from digium) to use the regular LTS
11:39.00Jo_Bellkaldemar: towards the end of configure, i see that many checks are saying no starting from "checking for resample_open in -lresample", Similarly ptlib.h usability and presence returned no. checks for sqlites also returned no..
11:39.09Jo_Bellany corrective measures needed?
11:40.22kaldemarJo_Bell: at least sqlite3 development files are something you do need.
11:41.58Jo_Bellpackage name? sqlite3-src? do you know?
11:44.14kaldemarJo_Bell: what OS are you on?
11:47.07Jo_Bellubuntu.. i use apt-get
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11:47.25kaldemarJo_Bell: libsqlite3-dev
11:47.31Jo_Bellthanks
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12:30.39Jo_Bellkaldemar: when I did a fresh configure, make and make install. I am not getting any conf files in /etc/asterisk as I used to get with previous version of asterisk.
12:30.48Jo_Bellshould I do make samples to get those?
12:32.01kaldemaryou should have the old ones under /etc/asterisk unless you removed them. "make samples" will put the sample config files there, yes.
12:34.09*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
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13:01.15bulkorokcan somebody help with res_fax!? I have to send a CED in T38 but asterisk never sends it on ReceiveFax...
13:01.35s[x]Hey all, Anyone here tied a Samsung OfficeServ into an Asterisk box
13:02.35[TK]D-Fenders[x], Got another question about it?
13:02.43[TK]D-Fenders[x], Considerig just how low your odds are
13:03.34*** join/#asterisk mjordan (~mjordan@nat/digium/x-yfiscesyclrascbz)
13:03.34*** mode/#asterisk [+o mjordan] by ChanServ
13:03.35s[x][TK]D-Fender: Tell me about it, Figured id take the shot. Was purely hoping to extend to the SIP client base using a trunk with asterisk instead of direct communication with the PBX
13:03.55s[x]SIP Client Licenses are $250 a pop on the Samsung
13:04.14[TK]D-Fenders[x], Ok, that's your goal, it still isn't a question.
13:04.36s[x][TK]D-Fender: Well for starters, is it at all possible
13:04.54[TK]D-Fenders[x], They both talk SIP apparently... so "why not"?
13:04.55ChainsawWho drained all the anger out of Fender?
13:05.46s[x][TK]D-Fender: Well how would one go about Client on Asterisk box talking to Phone on PBX
13:06.12[TK]D-Fenders[x], have you read the manual for your PBX?  Start there
13:06.21[TK]D-Fenders[x], *'s isde is just like everything else
13:06.25[TK]D-Fenderside
13:08.53*** join/#asterisk sekil (~sekil@78.24.104.73)
13:12.22*** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net)
13:12.26v0lZyhello
13:12.43v0lZyi have a rather silly question i think
13:13.13*** join/#asterisk justdave (~dave@unaffiliated/justdave)
13:13.48v0lZyI have this situation where I'm using VoIP as my main line and classic ISDN as a backup line
13:14.17v0lZyHowever, this backup ISDN line is also my main phone number for incoming calls
13:14.44v0lZyNow I'm at a risk that whenever my internet drops and my users switch to dial outside our company with the ISDN provider...
13:15.11v0lZythey clog up my ISDN line with outgoing calls
13:15.23v0lZywhich then prevents incoming calls...
13:15.49*** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
13:15.53v0lZynow I know I can disable outgoing calls
13:16.10v0lZybut can I restrict the amount of channels for outgoing calls to make reservations for incoming calls?
13:16.21v0lZylike.... max 2 outgoing calls at the same time
13:16.25v0lZycan this be done?
13:16.32kaldemarv0lZy: GROUP functions
13:17.30WIMPyThat's whay call waiting is a really important feature.
13:18.02v0lZywhat does call waiting do?
13:18.28v0lZyI mean... im talking about ISDN lines which i need to keep free so people can dial in and out
13:18.32WIMPyIt will give you the incomming calls even if all channels are in use.
13:18.44v0lZyhow can it do that?
13:19.08WIMPyThe most basic feature.
13:19.52*** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br)
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13:20.20[TK]D-Fenderv0lZy, "core show function GROUP" <-------
13:20.46[TK]D-Fenderv0lZy, COUNT how many you want to allow out in that scenario and choose when to cut them off
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13:23.02v0lZyI have no idea how to use this
13:23.15v0lZyIt just give me a description of what it is
13:23.52v0lZyI imagine theres a config file i should edit?
13:24.16WIMPyYes. Your extensions.conf.
13:25.18v0lZyi imagine this is under where i define my provider?
13:26.39[TK]D-Fenderv0lZy, your DIALPLAN
13:27.58*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)
13:29.34kaldemarv0lZy: Set(GROUP()=isdn) would put a channel into a group called "isdn". NoOp(${GROUP_COUNT(isdn)}) would return the number of channels in the group. the latter can be used for example in a GotoIf to decide whether a call is allowed or not.
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13:31.03leifmadsenwdoekes: ping
13:32.35v0lZyso everytime my patteren matches, i first need to run it through this application?
13:32.44leifmadsenyes
13:32.52leifmadsenyou need to check and evaluate
13:32.54bobb_WUi cant get record to work and the CLI isn't telling me anything
13:33.30bobb_WURecord(/var/lib/asterisksounds/en/prompt:gsm)
13:33.37[TK]D-Fenderbobb_WU, And you aren't showing us anything or providing details.  Fix that and we'll see what we can do for you.
13:33.43leifmadsenbobb_WU: what version?
13:33.43bobb_WUdirectory has 777
13:34.05bobb_WU1.8 - guess i need to update this one too
13:34.19leifmadsenbobb_WU: you're using old syntax -- run, 'core show application record'
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13:34.35bobb_WUthanks very much!
13:36.42bulkorokI use receivefax and asterisk has to send a CED tone to sender, but it doesn't... I use T38 anyone knows why it's not sending!?
13:37.08kaldemarbobb_WU: ":gsm" <-- what is that supposed to be?
13:38.36coppicebulkorok: if you are using spandsp it sends the CED signal in T.38 mode. I don't know about the digium FAX module. however. most of the things you send the T.38 CED indicator to ignore it
13:39.02*** join/#asterisk bchia (~Adium@nat/digium/x-euhqiylbgeoagvtw)
13:39.22bobb_WUits just our chosen format
13:39.42[TK]D-Fenderbobb_WU, http://pastebin.com/LsjuFZvv
13:39.47[TK]D-Fenderbobb_WU, no ":"
13:40.07[TK]D-Fenderbobb_WU, Always read the instructions for your app when it doesn't do what you figured it should have
13:40.24bulkorokcoppice: I use digium fax for asterisk... and the provider from where the faxes are send wants the ced ...
13:41.40bobb_WUi'm a googler and voip-info is the main result and its super out of date apparently
13:41.46anonymouz666bulkorok: I am just curious... why did you choose digium fax for asterisk instead spandsp stuff? digium offers support for that module?
13:41.55bulkorokyeap
13:42.00[TK]D-Fenderbobb_WU, CLI instructions = your reality
13:42.21bobb_WUnow that i know how to look them up in the CLI i will check it in the future
13:42.27bulkorokspandsp and digium faxforasterisk are segfaulting asterisk... now I have digium support :-)
13:42.28bobb_WUhowever its still not working with the new format
13:42.46[TK]D-Fenderbobb_WU, And you aren't showing us...
13:43.08bobb_WUcause its on a different computer and it takes me a minute to post pastebins
13:43.27[TK]D-Fenderbobb_WU, Then save it up for us.
13:43.41*** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com)
13:45.00bobb_WUahhhh i think i just remembered something
13:45.06leifmadsenkaldemar: that :gsm stuff is the old syntax on like Asterisk 1.2
13:45.15leifmadsenI suspect it was found on voip-info or something
13:45.28bobb_WUso its playing back but not saving- i have to move the file, right?
13:45.47bobb_WUyeah that was VOIP-info and it said for 1.2
13:46.18bobb_WUnvm - i will use the k flag
13:46.41[TK]D-Fenderbobb_WU, No, you don't hav to move the file
13:46.49bulkorokif I would at least understand the source of res_fax.c... it seems like with T38 there will never be a ced tone generated...
13:47.07[TK]D-Fenderbobb_WU, You can record wherever your * user has rights to.  You can playback any file * can erad and has rights to.
13:47.14[TK]D-Fenderread*
13:47.36coppicebulkorok: the CED signal comes from the actual FAX engine, not from res_fax
13:48.38bobb_WUwell my script is still broken but the recording ability is working again.  thanks guys :)
13:48.47bulkorokcoppice: I see the ced stuff only in res_fax.c ....
13:48.59Kattyhello my asterisk does not work at all how to fix plz.
13:49.51Faustovkill it with fire
13:50.19[TK]D-Fender^^^
13:50.22[TK]D-FenderCorrect answer
13:50.37kaldemaroh beautiful. my polycom hung up a call when i sent it a BYE from a different machine from my asterisk. which asterisk did not react to and left the channel up.
13:50.43Kattyi don't like that answer, i want a new one
13:51.01coppicebulkorok: oh, the are fudging in an actual audio tone when going into T.38 mode
13:51.20kaldemaranother call with malformed tags put the polycom to a state that it responds to any message with server error and does not hang up the call even when it's supposed to.
13:53.38bulkorokcoppice: this is how it should be I think... but the ced is not send...
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14:01.53asteriskATmarmuDhi guys, I can't find the mistake in this one "$AGI->exec("Dial", "SIP/101");" why isn't a call placed? the agi hangs there and nothing happens. any help is highly appreciated
14:02.34*** join/#asterisk _zoom_ (~Eissa@196.1.219.122)
14:02.37_zoom_hi,
14:02.56_zoom_am looking for a commercial Contact center based on asterisk
14:02.57_zoom_?
14:03.12*** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan)
14:04.05_Corey__zoom_: I can help you with that...  I'll send you a PM
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14:05.47n3hxsKatty, you must take that crank like thing and insert it into the front of your PBX, then wind it up. It must have run down overnight.
14:08.20[TK]D-FenderasteriskATmarmuD, perhaps you should show us the full call with AGI debug....
14:09.03[TK]D-FenderasteriskATmarmuD, As well as your AGI
14:09.42asteriskATmarmuD<PROTECTED>
14:10.10[TK]D-Fendermaybe... I also don't know if you even declared the var in that call...
14:10.44asteriskATmarmuD<PROTECTED>
14:11.06[TK]D-FenderasteriskATmarmuD, Start showing us.  We aren't going to take your word for it...
14:11.30asteriskATmarmuD<PROTECTED>
14:13.10Kattyn3hxs: i like it :>
14:13.21Kattyi also like caffeine.
14:13.32KattyALL THE CAFFEINES.
14:13.35asteriskATmarmuD[TK]D-Fender: here is the AGI: http://pastebin.com/XGwT9tia
14:14.26asteriskATmarmuD[TK]D-Fender: line 185 should initiate the call
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14:17.30carrarI gots the caffeinesers
14:17.42Kattyglomps carrar
14:18.09carraryou glomper you!
14:18.35carrarI roasted 3 cups of coffee beans yesterday too
14:19.05carrarthats the best way to drink coffee
14:19.11carrarfresh roasted beans
14:20.13ChainsawWe have pre-roasted beans in the office. But the grinding & brewing happens as you request it.
14:21.18carrarShould put a roaster next to it
14:21.29carrarsmoke up the whole office with coffee bean smells
14:21.43carrarI did it in my office (room at home) once
14:21.51carrarsmelt like a coffee roaster for 2 week
14:21.53carrars
14:22.04carrarwas awesome
14:25.00leifmadsenKatty: plz 2 b helping! my astrix system crashy!
14:25.11Kattypamples it.
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14:27.03asteriskATmarmuDok, a general question. If I want to automatically call a list of numbers, may I use AGI and call the perl-agi-script directly from the command line?
14:29.28[TK]D-Fender"help channel originate" <-
14:29.53[TK]D-FenderKatty, PAMPLEMOUSE!
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14:58.52bulkorokcoppice: I tried with res_fax_spandsp... this on is sending the ced tone!
14:59.49coppicea lot of people move from the digium FAX to spandsp to fix problems :-)
15:00.19bulkorok:)
15:00.26coppiceI've also helped people who were paying for digium FAX, but actually using spandsp :-\
15:01.06bulkorokthis is my situation now... but maybe I can use SendFax with digium fax and spandsp for receiving
15:01.26coppicewhy use digium FAX at all?
15:02.02*** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e)
15:02.41bulkorokasterisk segfaults when I use spandsp... so I hoped that it doesn't when using digium fax... but asterisk is segfaulting with it too :-(
15:03.09bulkorokbut... I opend a ticket to digium and mybe it's fixed soon... for bith techs!
15:03.12bulkorokboth
15:03.44ChannelZI've used FFA, not extensively, but haven't had any problems
15:04.53bulkorokwe will receive about 2.000 - 3.000 faxes per month with this... so I have to test very hard...
15:04.56coppiceChannelZ: how many 100's of k of FAXes do you handle each day?
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15:07.59bulkorokk guys... got to go... I opened another ticket to digium... we will see waht the y say about that :-)
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15:38.38din3shanyone ever implemented click-to-call on lotus notes working with asterisk?
15:43.05[TK]D-Fenderdin3sh, Told you yesterday, TAP is GENERIC and there links ont he WIKI for tools for this
15:43.08[TK]D-FenderTAPI*
15:43.10[TK]D-Fender~wikis
15:43.11infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
15:44.18din3shalready looked stuff up on it
15:44.39din3shi actually wanted to find somebody who already implemented it via lotusscript
15:44.41leifmadsenI suspect near-zero people here have done lotus notes integration
15:44.55din3shi guess too
15:45.10leifmadsenlotus notes is a fairly specialized app that not a lot of people use
15:45.12coppicehalf of them haven't even heard of Lotus Notes :-)
15:45.20din3shlol
15:45.21leifmadsenexactly :)
15:45.27*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
15:45.33leifmadsenI only know one person who has ever used it, and they do it all backwards in Sarnia
15:46.37*** join/#asterisk RubyRails (~jaiken@209.33.214.243)
15:46.53*** part/#asterisk RubyRails (~jaiken@209.33.214.243)
15:47.28din3sha tricky project, they want their lotus notes to be integrated with pbx, through click-to-dial, and also missed call status on their mail
15:49.00leifmadsensounds like you have some reading up to do on lotus notes
15:49.15ChainsawYou could always hire an IBM contractor/integrator.
15:49.23din3shtrying to install a trial version
15:49.51din3shChainsaw:yeah would have to consider that too
15:50.18din3shaparently avaya pbx already has some of tool working with lotus
15:52.02din3shirrespective of lotus mail, how to implement missed call alert on mail?
15:52.32*** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23)
15:52.46Chainsawdin3sh: Asterisk supports IMAP as a voicemail backend, you may be able to do something with that.
15:55.13din3shcan i use ${DIALSTATUS} - NOANSWER to trigger some kind of send mail application?
15:55.22nnyhave a strange issue, not sure what to look at explicitly yet. We have an office with some phones that are remote (over VPN). They can't use the phone's conference feature. - When they hit join it dies. The local phones perform this function normally. I am setting up a test now, not sure if i'll see anything since the "join" is on the phone's side. I have directmedia=no if that helps. I think the problem is the phones can't handle the media streams effectively
15:55.34Chainsawdin3sh: Sure.
15:56.01nnydin3sh: you can send an email during a call using System or Shell (for example)
15:56.15nnywe have used sendmail to send texts from the cli before screwing around
15:57.33din3shwill try that
15:59.12*** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e)
15:59.17din3shusers also want to logoff/lock their phones when they lock their computers (CTRL+ALT+DEL) in windows
15:59.31din3shany idea on how to go about implementing such a feature?
16:00.00nnydin3sh: look into asterisk management interface
16:00.28nnydin3sh: everything you're trying to do will require a connection to it to handle those types of things. You could have windows log off also log an agent off, for example
16:01.18*** join/#asterisk General_Z0D (~z0d@216.255.243.34)
16:01.24nnydin3sh: lockign the phone itself is however a feature of the phone and I don't think many phones have an API to interpret remote calls to do so. Agent log off would render the phone useless unless someone logged in, however. Just shooting from the hip
16:01.29[TK]D-Fender<din3sh> a tricky project, they want their lotus notes to be integrated with pbx, through click-to-dial, and also missed call status on their mail <- missedcall via e-mail is 1 trival line of dialplan.
16:01.39General_Z0Dcany anyone recommend a good working open source billing system for asterisk
16:01.59[TK]D-FenderGeneral_Z0D, a2billing
16:02.47nnydin3sh: you also may look at something like this http://www-10.lotus.com/ldd/nd85forum.nsf/Customer/6900587585d8221105257677001ec69a?OpenDocument
16:02.52General_Z0Dthanks fender
16:03.05General_Z0Daint that star2billing now?
16:03.18din3sh[TK]D-Fender: missedcall via e-mail is 1 trival line of dialplan. how!?
16:03.38[TK]D-Fenderdin3sh, "man sendmail"
16:04.13din3shhow to send details about the calling number/etc?
16:05.34din3shnny: thnks , already checked that srcipt, trying to download lotus notes to test it
16:07.45[TK]D-Fenderdin3sh, Include the CALLERID obviously
16:10.02din3shwill try it
16:11.47*** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210)
16:12.59General_Z0Ddoes 10.5 of * have realtime support like in 1.4 it has been some timesince I used it
16:15.14*** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com)
16:15.16nnyoops
16:15.32nnydin3sh: did you get my last message? "not sure if TAPI is supported natively in lotus notes, but it's the common phone system API. If it isn't native, you're better off using scripting to call the AMI instead"
16:18.55[TK]D-FenderGeneral_Z0D, * doesn't tend to LOSE functionality over time...
16:19.22[TK]D-FenderActually you're better off with AMI altogether
16:24.08*** join/#asterisk navaismo (~navaismo@189.144.118.69)
16:25.31din3shnny: thanks, looking into that
16:27.07ThinkGNU-Hitting an interesting issue: if I set timeout for more than 20 seconds when dialing a local channel I can't get the call to go through the rest of the dialplan rules for that dialplan
16:28.42nnyteliax.. thoughts?
16:30.15[TK]D-FenderThinkGNU-, Show us
16:31.15[TK]D-Fendernny, They are (were) OK, but not the strongest on any one side.Flakey on rarer occasions, but otherwise decent enough...
16:31.47ThinkGNU-Okay [TK]
16:40.25ThinkGNU-I'm trying to narrow down the conditions under which I'm able to reproduce the problem and then get the logs from that
16:44.27*** join/#asterisk coppice (~chatzilla@123203240103.ctinets.com)
16:46.07ThinkGNU-http://pastebin.com/z68JcVsd
16:46.18ThinkGNU-If my timeout is 25 or greater I seem to have this problem
16:47.23Qwellwhat timeout?
16:47.37ThinkGNU-The timeout on the Dial application that calls the local channels
16:47.56QwellI see no such timeout in your log
16:48.12ThinkGNU-I agree
16:48.23ThinkGNU-If I set the timeout to 23 seconds for example then the timeout works
16:48.35ThinkGNU-If set to 25 or greater then the call just ends
16:48.41*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
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16:49.57*** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
16:50.34[TK]D-FenderThinkGNU-, Dialplan please....
16:50.38*** part/#asterisk PaybackTony (~Adium@75-148-84-182-Oregon.hfc.comcastbusiness.net)
16:51.24ThinkGNU-Sure
16:51.53carrarShouldn't ThinkGNU be using sipwitch?
16:52.00carrar:)
16:52.29[TK]D-FenderAnd I don't see that call that CALLS the Local channel
16:52.39[TK]D-FenderI certainly don't trust that at all
16:52.49[TK]D-Fender[Jul 10 10:41:41]     -- Executing [channel_2@office-reception:1] Verbose("Local/channel_2@office-reception-bc
16:53.01[TK]D-FenderFirst line shows I'm already missing the greatest limiting factors
16:53.39ThinkGNU-http://pastebin.com/NAZjnx5Z
16:54.59[TK]D-Fender[Jul 10 10:42:05]   == Spawn extension (open, office-day, 2) exited non-zero on
16:55.22ThinkGNU-What's causing that to happen just by changing the timeout number
16:55.23[TK]D-FenderCall starts at 41, ends at 05. that's ... 25 seconds <-------------
16:55.23ThinkGNU-?
16:55.28[TK]D-Fendersame => n,Dial(Local/channel_1@office-reception&Local/channel_2@office-reception/n,25,Tk)
16:55.34[TK]D-FenderYOUR 25 second ilmit was reached
16:55.37[TK]D-Fenderlimit*
16:55.58ThinkGNU-right but why won't it move along to the next priority in the dialplan?
16:56.16*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
16:56.38*** join/#asterisk oej_ (~olle@h87-96-134-129.dynamic.se.alltele.net)
16:57.04[TK]D-FenderI see limits logs there.  I don't what else you've cut off...
16:57.52*** join/#asterisk nny (~Scott@174.107.223.14)
16:57.53[TK]D-FenderI also only see the nested channels die, not the main
16:58.00[TK]D-FenderMy trust level is shinking fast
16:58.12carrarthrows TK a rope
16:58.25carrartied it to the timeout!!
16:58.28[TK]D-Fenderhangs ThinkGNU- with it
16:58.31[TK]D-Fender:D
16:58.33ThinkGNU-Thanks
16:58.34nny[TK]D-Fender: thanks. Looking for a secondary provider, striking out here and there. If you have any other suggestions, feel free. I think i've gone through all the ones the channel suggests
16:58.55ThinkGNU-Sorry I'm obviously missing what's going on here and I don't mean to agitate you guys
16:59.04[TK]D-Fendernny, vitelity/voip.ms pretty solid, les.net
17:00.03carrarThinkGNU, at the time time, does it jump to the t priority?
17:00.10carrarThinkGNU, at the timeout time, does it jump to the t priority?
17:00.18carrarerr
17:00.22carrart extension I mean
17:00.23[TK]D-FenderOf course not.
17:00.33[TK]D-Fenderthis ian app timeout, not an IVR
17:00.37carraroh
17:00.39carrarI missed that
17:00.40nny[TK]D-Fender: thanks
17:00.44ThinkGNU-yeah, it's just ending
17:01.05ThinkGNU-I have a log where I set the app timeout to 23 seconds and everything executes beautifully
17:01.10ThinkGNU-24 works fine
17:01.14ThinkGNU-25 and greater no luck
17:01.16[TK]D-FenderI still don't see the ENTIRE call
17:01.31ThinkGNU-I'll get an entire call, sorry TK
17:01.41[TK]D-FenderEntire call + MORE
17:01.44[TK]D-Fenderboth sides
17:01.53*** join/#asterisk ectospasm (~ectospasm@66.172.33.249)
17:01.57[TK]D-FenderI want to make sure that things have a chance to die out
17:02.17ThinkGNU-okay, gimme just a second and I'll have that for you
17:07.20ThinkGNU-http://pastebin.com/vz0L1X5A
17:07.28ThinkGNU-That's everything it logged for that call TK
17:09.10*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
17:09.34[TK]D-FenderLocal/channel_1@offic
17:09.34[TK]D-Fendere-reception <-- missing the /n on it
17:09.38[TK]D-Fender2nd one had it
17:09.51[TK]D-Fender[Jul 10 11:02:35]     -- Called channel_1@office-reception
17:09.52[TK]D-Fender[Jul 10 11:02:35]     -- Called channel_2@office-reception/n
17:10.27ThinkGNU-lemme fix that real quick
17:11.14[TK]D-FenderJul 10 11:02:59]   == Spawn extension (open, office-day, 2) exited non-zero on 'SIP/wyomingcom1-0000021f'
17:11.21[TK]D-Fenderthere's the outer channel dying
17:12.12ThinkGNU-exiting non-zero isn't what it should be doing right?
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17:12.58[TK]D-FenderThinkGNU-, One serious consideration is that you are getting a call from an ITSP .... THEY may lok at this and go "Hey, MY client isn't responding" ... and GIVE UP
17:13.08[TK]D-FenderThinkGNU-, You should probably ANSWER the call first <-
17:13.24ThinkGNU-That's a good point
17:13.52ThinkGNU-I knew I was overlooking something simple here
17:18.28ThinkGNU-Let me make some adjustments and see what happens. Thank you for helping out guys. Obviously I still have a lot to learn here but I am determined to learn.
17:18.50[TK]D-FenderThinkGNU-, Well you just got a good lesson in tracing things back to the start
17:19.03[TK]D-FenderThinkGNU-, And isolating each breaking point
17:19.03ThinkGNU-Yes I did and thank you so much
17:19.23ThinkGNU-This is the kind of experience I can definitly use.
17:21.53ThinkGNU-[TK]D-Fender, that did it. I ANSWER the call first and then no problems. Thank you again for all your help guys. One day, when I don't suck as bad I hope to be able to contribute here in the channel and answer questions.
17:25.53[TK]D-FenderThinkGNU-, You're welcome.
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17:28.21bobb_WUhow do you reload just the dialplan?  and is there a good doc for asterisk CLI commands?
17:29.24outtolunchelp     and     help dialplan
17:30.37nnywhat is the suggested method for using one touch recording (automon) with meetme? I see options for dial and queue, but not for the application. Do i have to use Local?
17:31.30bobb_WUok and how do i execute a System() command after the user hangs up after a recording?  is there a trick for that?  i'll pastebin my current code if need be
17:31.54ectospasmbobb_WU: put System() in the 'h' extension
17:32.25ectospasm'h' executes after the call has hung up.
17:32.28bobb_WUso i can move that to behind the Hangup() one too?
17:33.00ectospasmwhat do you mean,"behind?"
17:34.47bobb_WUi have a hangup in all my extensions, i thought maybe i could add a backend action after that executes
17:36.07[TK]D-Fenderno
17:36.11bobb_WUpastebin.com/KbHaiyPw
17:37.35bobb_WUreturnCall.sh writes a few lines to a {exten}.call file then moves it to /var/spool/asterisk/outgoing
17:37.47bobb_WUit works fine from bash and the message is there to playback
17:38.19bobb_WUrunning 1.8 on this server
17:38.44ectospasmHangup() isn't necessary unless you want to force a hangup in every context/extension.  The 'h' extension executes *after* Hangup(), or the far end hangs up, so you can do your call postprocessing there.
17:39.06bobb_WU(that's what i meant by behind)  thanks
17:39.20ectospasmexten => h,n,...  ; put your post prcessing here.
17:40.21bobb_WUso not exten = > 8051,h ?
17:40.40bobb_WUi only need this number to postprocess
17:41.31[TK]D-FenderNO
17:41.34[TK]D-Fenderther is no "chaining this
17:41.47[TK]D-FenderYou need to integrate this into it
17:42.10nnyi'm just gonna use the local channel for recording meetme on demand, let me know if that's improper :)
17:43.33nnyhmm maybe not, seems using local to call meetme supercedes the admin and user menu :S
17:43.48nnyrecording worked, however.. lol
17:46.07bobb_WU"ther is no "chaining this" can you clarify?  its still not running the script on hangup
17:46.37bobb_WUi tried exten = > 8051,h,System() and exten => h,n,System()
17:46.59[TK]D-Fender"h" is NOT a priority
17:47.26[TK]D-Fenderand exten => h,n,System() doesn't tell me you have a priority 1 in the first place
17:47.46[TK]D-Fenderand from what you're showing you also seem to be suggesting calling that with no parameters
17:48.12bobb_WUit has parameters
17:48.16bobb_WUdid you see the pastebin?
17:48.51nnyso I guess my actual question to channel is "can I use local to record meetme (with the W option) and still pass key pressed to Meetme for the admin/user functions?
17:49.02*** join/#asterisk brdude (~brdude@12.155.183.30)
17:49.06[TK]D-FenderYour pastebin shows your trying to use "h" as a PRIORITY
17:49.14[TK]D-Fenderit is not
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17:58.15leifmadsenbobb_WU: uhhh, note that 8051,h,Foo() is invalid
17:58.38bobb_WUyeah i think i understand
17:59.05bobb_WUits h,1,Foo() but that means i force an action on every exten in the context
17:59.13bobb_WUso i'm giving it its own context
17:59.32*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
18:00.23bobb_WUand it worked
18:00.30bobb_WUish
18:01.35*** join/#asterisk newtonr (~newtonr@nat/digium/x-erwsntpvgzxuxjnx)
18:01.56*** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net)
18:05.23bobb_WUSystem(/asteriskScripts/returnCall.sh ${CALLERID(num)})  doesn't seem to be working.  its being called with the correct data as seen by logging output
18:06.23[TK]D-FenderWe aren't seeing anything
18:06.34leifmadsenwhat isn't working?
18:06.46bobb_WUits not calling the script
18:06.54[TK]D-FenderWe aren't seeing anything
18:07.22leifmadsendoes somethign simple like, System("touch /tmp/this_worked") work?
18:10.51bobb_WUok http://pastebin.com/95eBpZwk
18:11.28bobb_WUi'm not seeing yep and there is nothing under ./calls/
18:11.33bobb_WU*Yep
18:12.10[TK]D-Fenderempty pastebin
18:12.55bobb_WUhttp://pastebin.com/95eBpZWK
18:13.05bobb_WUits case sensitive- hmmm
18:14.11[TK]D-FenderYou never EVER use "cp" to put a call file in the spool folder
18:14.31bobb_WUwhat should i use instead?
18:14.33outtoluncmv
18:15.01[TK]D-FenderAnd I don't see proof that * can read that script you are colling.  that the paths are right.. that * is defaulting to the FOLDER yout hinit it is since yuo're creating files relative to where you ARE
18:15.04bobb_WUk
18:15.20[TK]D-FenderThis is a massive failure in debugging practice
18:16.10[TK]D-Fender"you think"*
18:16.25[TK]D-Fendergah, typing going down the tubes again
18:17.21bobb_WUwell how about it.  its working correctly!
18:17.55bobb_WUi made the paths absolute and changed cp to mv
18:18.02*** join/#asterisk justdave (~dave@unaffiliated/justdave)
18:28.54*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
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18:30.49UnixDevwould it be possible to move on this request? (https://issues.asterisk.org/jira/browse/ASTERISK-18068) .. its already been done in trunk (https://reviewboard.asterisk.org/r/1754/) and looks pretty trivial to back port
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18:43.38Kattyhello my asterisk does not work at all how to fix plz.
18:43.56Kattyit makes a gurgle noise.
18:44.01*** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e)
18:44.05drmessanoYou make a gurgle noise
18:44.11drmessanoIts hawt too
18:44.14Kattythat's a drinking game.
18:44.24Kattyi saw it at hastings the other night.
18:44.34drmessanoThats also AFTER a drinking game
18:44.35*** join/#asterisk rcaskey (~Rob@dumbledore.athenshousing.org)
18:44.47_Corey_Katty: sounds like a plumbing issue...  pour Drano into system
18:44.55Kattyoh wait, no. it's called "Gargle"
18:45.07Katty_Corey_: *hee*
18:45.11malcolmdyou mean the scary things on top of old buildings?
18:45.12drmessanolol
18:45.17Kattyadds to list.
18:46.09Kattyi'm knitting a fabulious little thing today.
18:46.14Kattyand by little i mean a 28 hour project.
18:46.50[TK]D-Fender28 HOURS LATER
18:46.58KattyYES.
18:47.00[TK]D-Fenderpictures zombies with scarves
18:47.21Kattyi have some smaller projects in the making too.
18:47.24KattyBACON STRIPS.
18:47.25slav3_kittenKatty, trade you :D i've got roughly 2 or 3 days left on the ddrescue of my ipod :(
18:47.34drmessanoThe only thing I ever spent 28 hours on was my best friends mom
18:47.50slav3_kittendrmessano, was she worth it
18:47.57Kattyhow does any knitting project the size of an ipod take 2 to 3 days?
18:48.17drmessanoslav3_kitten: I had to start giving her CPR after 14 hours, so not really
18:48.22Kattyoh. ddrescue.
18:48.24[TK]D-FenderKatty, AND JD.  That's SMART.
18:48.28slav3_kitten:(
18:48.30Kattyi totally just skipped that word.
18:48.31slav3_kittenthat sucks
18:48.41Katty[TK]D-Fender: that's a terrible idea.
18:48.58slav3_kittenlol @ Katty
18:49.18slav3_kittenyea when i was on vacation last month mister ipod started making funny noises an losing bits of songs
18:49.38[TK]D-FenderKatty, The EMT guys are locals here... one of their runs was on a Taco Bell that's within walking distance of my place
18:49.38Katty:<
18:50.05KattyI guess they call them Volcano Tacos for a reason.
18:50.22Kattyslav3_kitten: i hope it completes ok :<
18:50.44drmessanoI thought they called them Volcano tacos because of what happens 24 to 36 hours after
18:50.45slav3_kittenme too, or at least lets me recover file names
18:51.02slav3_kittenthen i can spend a week having dad search the attic for my CD collection
18:51.30Kattyyour dad has better things to do.
18:51.32Kattylike your mom.
18:51.36Kattysearch for it yourself!
18:51.36drmessanoHA LOL
18:51.54slav3_kittenKatty, you figure out how to climb a ladder one legged and on crutches an i'll give you a cookie
18:52.11drmessanoOne legged AND on crutches?
18:52.13Kattyask your girlfriend.
18:52.21Kattyaccepts cookie.
18:52.24slav3_kittenwhat girlfriend
18:52.25drmessanolol
18:52.36sp00kzhop
18:52.38slav3_kittenyou see the channel we are all currently in?
18:52.47Kattyyes.
18:52.49Kattydanny is married.
18:52.50slav3_kittenwhat part of that makes you think i have a girlfriend or boyfriend for that matter
18:52.54Kattyand i have a gentleman friend.
18:53.00drmessanoI think Katty meant to say your mom has a peg leg with a kickstand
18:53.04Kattyi refute your logic and replace it with my own.
18:53.08slav3_kittenmeh
18:53.21slav3_kitteni've been single for like *does math* 10 years
18:53.31drmessanoYes, Katty has a special gentleman friend
18:53.31Kattywhy.
18:53.35Kattydrmessano: yes.
18:53.36drmessanoand she's got a beverage here
18:53.41KattyYES.
18:53.45Kattyit's a lovely beverage.
18:54.06Katty...you can't have it. either of them.
18:54.09slav3_kittenKatty, before i got disabled 5 years ago i was regularly dating to find a woman into computers, and electronics
18:54.14drmessanoslav3_kitten: Theres no reason to be single with lonely married women and zaftig girls all over the internet
18:54.24slav3_kittennow i can't even get dates
18:54.32Kattyoh i see.
18:54.36drmessanoIts like this
18:54.53QwellAren't you married?  You aren't allowed to give dating advice. :p
18:54.54Kattyhave you been /trying/?
18:54.59drmessano"Hi"  "Are you married?"  "No?"  "Are you Fat?"  "Yes?  W00t"
18:55.06slav3_kittenKatty, i have
18:55.12Kattywell keep trying dear.
18:55.21drmessanoThat should be a macro in your favorite IM client
18:55.29Kattya good friend of mine is paralyized from the waist down due to a mountain biking accident where he flipped over the bars.
18:55.34slav3_kitteneh, lately i've decided my money would be better spent on computer and electronics bits
18:55.43slav3_kitteni'm terribly sorry to hear that
18:55.46Kattyand he has a lovely lady in his life now thanks to him being persistant.
18:55.49drmessanoCheck out myspace, Meetme.com, and Yahoo.  Look for anyone married or fat.  Easy targets
18:55.50slav3_kitteni broke my spine a little bit at work
18:56.00QwellKatty: persistent ~= creepy
18:56.06Kattyno not in a creepy way.
18:56.08Kattyjust kept looking
18:56.14Kattydidn't give up.
18:56.19Kattynot all stalkery over a person or anything.
18:56.34Qwellstalks Katty
18:56.43drmessanoJust keep asking the Doctor Messano 2 key questions and you will find love
18:56.46Qwell(in a non-creepy way)
18:56.48slav3_kittenis his lovely lady from a slovic nation that has good prices an low shipping?
18:57.01drmessano"Are you married?"  "Are you fat?"  <-- All you ever need to know
18:57.02Kattyno.
18:57.08Kattyshe's an adorable little brunette with a heart of gold.
18:57.20Kattyand happens to like dr who.
18:57.23slav3_kitteni never said that mail order brides can't be high quality...
18:57.25Kattyno it's not me.
18:57.31Kattythey can be.
18:57.33Kattyi know one.
18:57.37Kattyshe just wanted to start a family.
18:57.40drmessanoOh, and if you get to step 2.. Ask them "So are you happily married or just married?"  <---- SRSLY
18:57.58slav3_kittenyea that's my main problem these days Katty
18:58.08slav3_kitteni'm near 30 and i want nothing to do with having a family or starting one
18:58.17Kattyoh that's not a problem.
18:58.26Kattyi'm getting close to 30 and want none of that family crap either.
18:58.28slav3_kittenwomen in my age group seem to want to start families and that crap
18:58.34Kattyi disagree.
18:58.37slav3_kittenso, want to get some coffee?
18:58.48Kattydrmessano: did you see that comment?
18:58.52Kattydrmessano: he has no idea.
18:59.02drmessanoYes I did
18:59.09Kattyit's amusing
18:59.22slav3_kittenwhat don't i have any idea about?
18:59.22drmessanoLet me explain the age groups...
18:59.24Kattyslav3_kitten: and this is why -> https://sphotos.xx.fbcdn.net/hphotos-ash4/c0.0.403.403/p403x403/376956_10100311924647677_407737399_n.jpg <- posted 30 minutes ago.
18:59.36Kattyslav3_kitten: i HAVE a coffee. iced coffee.
18:59.39slav3_kitten...
18:59.42slav3_kittenfreaky!
18:59.44drmessano18-25 Either concerned about school or want a family, NOW
18:59.53Kattyslav3_kitten: no it's actually quite tasty. not freaky at all.
18:59.54drmessano25-33 Happily Single
19:00.01Kattydrmessano: YES.
19:00.13drmessano33-39 Exiting first marriage, looking for good guy.  PRIME REAL ESTATE
19:00.22Kattythat's not true.
19:00.26Kattyi already did my graceful exit.
19:00.34slav3_kittenyou were married?
19:00.36drmessano39 to 45 -- Doesnt matter, they will do anything in bed tho
19:00.37Kattynearly.
19:00.40slav3_kittenoh good
19:00.48slav3_kittendodged that bullet
19:00.53drmessanoOk well
19:00.58Kattydrmessano: agreed. one of my friends really enjoys that 40-50 age group.
19:01.02drmessano33-39 is also "STILL HAPPILY SINGLE"
19:01.02slav3_kittenso what you're saying drmessano is aim for 40
19:01.14Kattyslav3_kitten: yes.
19:01.24Kattyslav3_kitten: as my friend says, they cook clean, and enjoy sex!
19:01.36jpsharpWhere does that biological clock of "I HAVE TO PROCREATE NOW" come into play?
19:01.49drmessanoNah.. You will find LOTS of very cool, awesome women who married some guy too young and now want something real in the 30 to 40 group
19:01.55Kattyjpsharp: i don't know. i've not ran into it yet.
19:02.01Qwelljpsharp: 16, if TV has taught me anything.
19:02.10Kattythis is why i love Qwell
19:02.13slav3_kittenKatty, pm?
19:02.26jpsharpNo, that's "I HAVE TO BE AN ATTENTION WHORING TEENAGER NOW"
19:02.27Kattyslav3_kitten: sure!
19:03.02Kattyjpsharp: they are annoying :<
19:03.09*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
19:03.20*** join/#asterisk PipBoy (PipBoy@ip067.216.145.100.nas.net)
19:03.24drmessanoSecond time around can be completely awesome for a lot of women.  I keep telling my best friend that.  He's 35 and wants a damn 35 yr old puritan virgin that goes to church 3 times a week OR NOTHING.  Says hes doomed to never find anyone.  I keep telling him to look for a nice second-time-around gal who is completely awesome and sick of the meathead she married after high school
19:04.03Katty...goes to church? :<
19:04.06Kattyyuck. i do not approve.
19:04.11Kattybut that's ok, i dont' have to!
19:04.25drmessanoThe whole "All the good ones are taken" thing is bullshit.  Nobody stays in bad marriages anymore, and there's lots of great gals going back into circulation
19:05.13PipBoyquick question. When a person calls in over a SIP trunk.. And pressed a digit in an IVR.. is it still considered DTMF?
19:05.29drmessanoOf course it is
19:05.59PipBoyAh, figured as much... just trying to tweak a pbx to be a little more dtmf sensitive... not picking up a lot of key presses
19:06.10PipBoythanks
19:06.34jpsharpIf its coming in over SIP, you should use something like SIP INFO or RFC to transport DTMF rather than trying to detect the tones.
19:07.06jpsharprfc2833
19:07.17KattyPipBoy: I LOVE PIPBOYS
19:07.23KattyPipBoy: in reference to fallout.
19:07.32PipBoyfer leveling n stuff
19:07.33PipBoylol
19:07.37Kattyi don't want to set the world on fire
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19:08.08PipBoyhmmm.. thanks jpsharp . I need to look into that, as I have no idea what my system is currently doing
19:10.32PipBoyBut yea, I believe my system must be doing DTMF.  You know the whole "press * to start recording" I have to constantly mash * lol... Customer is kinda annoyed :P
19:10.50jpsharpinband DTMF over SIP sucks, especially if you're running something other than G711 codec.
19:12.13Kattytell them to take their stress out on the * button!
19:12.15Kattyproblem solved!
19:12.47PipBoyyea * is the modern digital bubble wrap!
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19:13.10jpsharpIf I did that, my phone would be a shattered, broken shell of itself...lying bleeding in a corner.
19:13.46PipBoyhaving "Nam" flashbacks lol
19:13.53PipBoyYOU DONT KNOW!! YOU WERNT THERE!
19:14.05Kattyputs phone into therapy
19:14.34slav3_kittenrefrains from inserting a holodisk into PipBoy
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19:15.03PipBoylol... hey now! i dont go that way
19:16.06Kattywhy not.
19:16.12Kattyyou should try it. you might like it.
19:16.17PipBoylolol
19:16.31Kattyi bet you didn't like vegetables as a kid either!
19:16.48PipBoywalks away slowly and starts googling RFC's
19:16.51ThinkGNU-Did I switch channels somewhere down the line?
19:17.31Kattyyes.
19:17.45ThinkGNU-lol
19:18.10ThinkGNU-You know, PipBoy still might not like vegetables
19:18.20PipBoyits true
19:18.27PipBoymy diet is beer and potatos
19:18.47Kattypotato is a vegetable.
19:18.47drmessanoAre you irish?
19:18.51Kattyand beer is made from a plant.
19:18.53PipBoycanadian :P so close
19:19.06Kattytechnically you could be a vegetarian.
19:19.10ThinkGNU-cows are made from plants too I think, indirectly
19:19.13drmessano4 course meal to an irishman.. 3 beers and a potato
19:19.14Kattyunless you consider yeast an animal.
19:19.19*** join/#asterisk justdave (~dave@unaffiliated/justdave)
19:19.23QwellKatty: I could totally be a vegetarian.
19:19.35QwellJust as long as they change the classification of meat to be a vegetable
19:19.41Katty:P
19:19.42Qwellthat's my only hangup
19:20.01Kattylately i've been enjoying burgers.
19:20.23Kattyand leinenkugel
19:20.36drmessanoSomeone told me I don't eat enough greens, so I put some green M&Ms in my bacon, ham, and sausage omelette the other night
19:20.52Kattyyou should have put green onions on top
19:20.58Qwellcilantro
19:21.11Kattycilantro is evil. it makes me sneeze and turn redish.
19:21.17Qwelllame
19:21.18drmessanoI'm saving the green onions for Bulgogi
19:21.37KattyBulgogi?
19:22.06KattyBUGATTI?!
19:22.14drmessanoNo
19:22.20Kattysad :<
19:22.29Kattydanny you should get me a bugatti.
19:22.34Kattythey're only 1.5mil.
19:22.35drmessanohttp://en.wikipedia.org/wiki/Bulgogi
19:22.43Kattyi found a picture. it looks tasty.
19:23.10drmessanoI make mine with beef, sesame oil, soy sauce, and sugar.  It's amazing
19:23.35drmessanoDrop it on some rice, and top it with sesame seeds and green onion
19:23.52Kattyi think you're avoiding the bugatti topic.
19:24.07drmessanoLet me just say this
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19:24.42drmessanoThat's 1.4-ish million more than I have spent on women who I have done the naked roly poly with
19:24.50drmessanoSo.. There's qualifications here
19:24.51Kattyderik promised me a bugatti.
19:25.00Kattyas soon as he can afford to get himself one too.
19:25.23drmessanoVery nice
19:25.27drmessanoThat seems reasonable
19:25.31Kattyi agree.
19:26.13drmessanoI would be happy with a nice second hand job right now for her
19:26.25drmessanoIts either new car, or clunker
19:26.30drmessanoNo in-between
19:26.50Kattysecond hand job?
19:26.53Kattyshe doesn't work?
19:27.48drmessanoWell, "job" in the objective sense.  We need a better car.. Something second hand
19:28.36Kattyah i see.
19:29.06drmessanoBuying a second hard car is like buying software.. If you're not gonna buy the real deal, expect someone to lie to you about how awesome it is
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19:29.22ThinkGNU-You just have to know what you're looking at
19:29.22drmessano"Dude, this is JUST LIKE PHOTOSHOP... $19.99"
19:29.36Kattyi agree.
19:29.39Kattyhow much are you willing to spend?
19:30.00drmessanoI do know what I am looking at.. The problem is the misrepresentation and the time consumption test driving crap
19:30.38Kattytell me your price range, dagnabbit.
19:30.47drmessanoProbably 6-8K
19:30.58Kattyand are there any requirements?
19:31.02Kattyautomatic... 4 doors
19:31.09Kattymust have backseat.
19:31.15drmessanoAutomatic, 4 doors.. all must work
19:31.39Kattyhave you considered salvage?
19:32.21drmessanoWe spent a 2 weeks looking for the POS she drives now.. Can't tell you how many $200 cars there are out there with $2000 price tags
19:32.39drmessano"You didnt specify you needed a transmission with it"
19:33.27ThinkGNU-who needs those?
19:33.35Kattyya get a rotobox ;)
19:33.41ThinkGNU-I agree with you on the overpriced used cars though
19:33.55Kattyi'll sell you mine for that much!
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19:34.09Kattythere's nothing wrong with it, other than it's not as shiny as a 350z
19:34.31*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
19:34.31*** mode/#asterisk [+o sruffell] by ChanServ
19:35.57drmessanoI had a guy try to sell me a 2000 KIA Sephia.. It looked like it spent 3 months underwater.  Tires were bald.. He wanted $1200 for it.
19:36.14drmessanoOh, and it was in the bulletproof vest section of town
19:47.23rcaskeydrmessano, here in the states I highly recommend if you can at all afford it sucking it up and buying a new Elantra, Sonata, or Optima. Used cars are overpriced now.
19:48.14Kattyi disagree.
19:48.39Kattyi do agree that some people don't know how to properly price a used car.
19:49.05rcaskeyIt's not that, it's that if they do they will sell it for more than you should pay
19:49.14rcaskeybecause there are too many broke folks that have to buy a used car
19:50.15drmessanoThats true
19:50.52rcaskeydrmessano, the trick is to find a base Hyundai model - you don't want the GLS
19:51.34drmessanoWell, if we buy new, it's going to be for me.. and she'll get my car.  That being said, I would need an SUV or large crossover..
19:51.49rcaskeydrmessano, kids or tools or dogs or what?
19:52.03drmessanoTools, equipment, dead bodies
19:52.13rcaskeyno truck?
19:52.53drmessanoNeed it covered.. Dont want to deal with a pickup with some add-on cover... though I had considered it once
19:53.17rcaskeywould you want a white panel van? they are cheap
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19:54.13drmessanoNope.. I dont need a work vehicle.. I need a car that can hold my EMS and Fire gear, tools, and occasionally large pieces of equipment
19:54.49rcaskeyhow large? would a small SUV be big enough?
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19:55.04drmessanoYeah, like I said.. SUV or large crossover
19:55.32rcaskeyI'd look at hte Kia Sorento and the Hyundai Santa Fe
19:56.19drmessanoI was thinking about a Sportage or a Sorento
19:56.22rcaskeyif I was buying for my wife however I'd buy the Murano because it's pretty
19:57.23rcaskeyif you can go that small you might see if you could do with an Elantra Hatchback, the seats fold down
19:57.39drmessanoShe's getting a Rondo in the deal.. Which kinda stinks.. She doesn't need that kind of space.  She really needs some little thing she can drive back and forth to work and the mailbox.  Just a pain to find something that isnt crap
19:58.18drmessanoWe BOTH don't want to buy her a new car.. But almost have no choice but to give her the Rondo
19:58.33rcaskeyif shes doing low MPG a 15 year old crown vic in good condition will probably keep runnin
19:58.39jpsharpA minivan.  I love our Dodge Caravan.
19:59.07rcaskeyNot touching a Dodge
20:02.22*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)
20:02.29drmessanoI thought about buying an older camry, but yeah, the pricing is insane
20:02.52drmessano"Blue book is $4000, but I put a new head gasket on there, so I want $6500"
20:02.59ThinkGNU-I have over 200k on my 2003 Chryser(dodge) minivan
20:03.04[TK]D-Fenderrondo is actually pretty shitty on storage...
20:03.10ThinkGNU-Chrysler* I meant
20:03.18drmessanoI have fit a LOT in my Rondo
20:03.23*** join/#asterisk acidjazz (acidjazz@notchill.com)
20:03.31[TK]D-Fenderimpractical width, etc.
20:03.39drmessanoWell, yes
20:03.41acidjazzwhat is the best service these days out there for bulk sms/phone calls (automated not people)
20:03.49[TK]D-Fendermaybe witht he rear seat folded out, but base... blows
20:03.49drmessanoIt's defintely not EASY to pack, at all
20:04.27[TK]D-FenderBecause of all the musical gear I bring around I needed to be abe to fit my keyboard hidden well away in the trunk.
20:05.05drmessanoI wont let my wife pack it up when we go on trips.. She leaves it all outside and I make it all fit.  Just used to how to maneuver crap in there
20:05.06[TK]D-FenderEnded getting a 2004 Altima.  a decade worth of that family was well designed for port access size.
20:05.13ThinkGNU-Ford is building some really good vehicles now
20:05.18drmessanoNice
20:06.11UnixDevford: fix or repair daily
20:06.22ThinkGNU-old thought there UnixDev
20:06.33ThinkGNU-Ford's stuff, as well as Chrysler's stuff has gotten really good recently
20:06.37jayteeford: first on race day
20:06.51UnixDevfound on road dead
20:07.44ThinkGNU-the new ecoboosts are pretty badass
20:08.18UnixDevall new cars a nice.. i just don't personally like ford.. but you are right about them getting better recently
20:08.41acidjazzhey all what is the best service these days out there for bulk sms/phone calls (automated not people)
20:08.47ThinkGNU-I didn't like fords before but their stuff is improving so much
20:08.54*** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net)
20:10.08citywokIs there a way to set SQL server options using res_odbc/cdr_adaptive_odbc?  like SET ANSI_NULLS, etc.
20:14.58*** join/#asterisk e1722162 (52a8ca64@gateway/web/freenode/ip.82.168.202.100)
20:15.53*** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.6.0 (2012/07/10), 1.8.14.0 (2012/07/10), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
20:20.46*** join/#asterisk Nivex (~kjotte@atlantis.home.nivex.net)
20:21.51NivexETA for new packages on http://packages.asterisk.org/ from today's release? (looks like pabelanger is the maintainer?)
20:22.51mjordanNivex: given that I just hit the "send e-mail" button, some patience would be appreciated :-)
20:23.25Nivexmjordan: most impressive :) though the package that's in there is still 1.8.11.x, so...
20:23.44mjordanfor debian, yes.  pabelanger does not maintain the debian/ubuntu stuff for Asterisk
20:24.22Nivexah, so he's been framed :P  (looking at the Maintainer field in Packages)
20:24.55mjordanNivex: that should be updated :-)
20:25.40mjordanNivex: what link are you looking at that shows that?
20:25.47Nivexhttp://packages.asterisk.org/deb/dists/squeeze/main/binary-i386/Packages
20:25.54mjordanah
20:26.02mjordanwell, that is accurate as of 1.8.11.1
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20:41.09pabelangerNivex, Ya, Digium no longer signs my paychecks, so I've had to let them slide.  However, if you are experienced in maintaining them, I'm sure access could be provided.  There are some technical hurtles but nothing serious
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20:51.33[TK]D-Fenderpabelanger, How long have been out?
20:52.02*** join/#asterisk ks3 (~ks3@74.115.41.6)
20:52.30pabelanger3months?
21:01.00[TK]D-Fenderah
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21:28.22Sean-DerWhat is a good way to dump incoming traffic? I having a server set up in a DNAT, I have my externip set, nat=yes, qualify=yes but I can't seem to get anything to happen
21:28.50Sean-DerI tried to use tcpdump to dump all trafic, but I don't see any requests from a softphone coming in
21:29.44Sean-DerIs there a better way to debug this? I have a feeling that my traffic is not even making it to the server, it would just help my process if I could be sure I was at least getting the invites coming in
21:29.59Sean-DerI also have localnet set
21:36.00Sean-DerI also have sip debug set on
21:36.08Sean-DerAnd not seeing anything?
21:38.01[TK]D-FenderYou settings hardly matter for seeing nothing.  that means packets just aren't reaching you at all
21:46.12Sean-Der[TK]D-Fender: Am I doing it correctly though? I was doing a tcpdump on 5060-5070, will that catch the traffic. I have sip bound to 5060 and my softphone sflphone has just the host and auth details
21:48.36[TK]D-FenderAll you should need is "sip set debug on
21:49.18Sean-DerYep thats what I did. Thanks for the help [TK]D-Fender I am gonna mess with my router some more I am just learning as I go
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22:03.19m_tadeuhi everyone...I'd like to know something about the bridge event for the manager...is the callerid1 always the caller and the callerid2 always the receiver?
22:04.36mjordanthey're not the caller ID
22:04.43mjordaner, nm
22:04.44mjordan:-P
22:06.13mjordanm_tadeu: that isn't anything that enforces that relationship
22:06.59mjordanin general, as its currently implemented, callerid1 will refer to the caller, and callerid2 will refer to the callee
22:07.34mjordanbut it might be dangerous to make that assumption, as that's currently an implementation decision.  What you do know is that two channels have been bridged, who those channels are, and their caller IDs
22:09.24m_tadeumjordan: thanx for the explanation. My problem here is that I need to know who is receiving the call. is there a way to know this on a manager?
22:10.25mjordanI'd listen for a Dial event in conjunction with a Bridge event
22:10.36mjordanthe Dial event will guarantee who the caller is in that relationship.
22:10.53mjordanWhen the Channel in a Dial event is in a Bridge event, you know that the other channel was the callee.
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22:14.00m_tadeuI was hoping for a more explicit way to do it :) but that will do the job
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