IRC log for #asterisk on 20120507

00:00.05[TK]D-FenderOr the WIKI on the * page
00:00.11[TK]D-Fenderor...
00:00.12[TK]D-Fender~wikis
00:00.13infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
00:00.22nfsnobodyThanks, tried all of the above before coming here :)
00:00.34Ice_StrikeDoes anyone know good VOIP provider that allow you to change any callerid without validation
00:00.39[TK]D-FenderNow you realize that virtually nobody uses it
00:00.45nfsnobodyThere is a bit of information in the asterisk wiki, but it's only very basic information.
00:00.50[TK]D-FenderIce_Strike: Most of them
00:00.55[TK]D-FenderFlowroute, voipms, etc
00:01.03[TK]D-FenderVitelity....
00:01.15Ice_StrikeThanks!
00:01.21nfsnobodyCould I ask why no-one uses it? It makes the dialplan more flexible and configurable, doesn't it? Gives you a higher level of power?
00:01.26Ice_Strikewhich one are in europe or UK?
00:03.40[TK]D-Fendernfsnobody: I don't see anything in it that qualifies it as "more powerful"
00:03.54[TK]D-FenderIce_Strike: All can terminate internationally
00:04.58*** join/#asterisk micols (~t@rlogin.dk)
00:05.07nfsnobody[TK]D-Fender - all the functionality within the standard dialplan configuration is custom coded - LUA adds the power of an entire language
00:05.12Ice_Strike[TK]D-Fender I've got a problem with CallerID on voip.ms
00:05.20nfsnobodyThat's fine, I guess I'll just have to work with what I can find - thanks for your help :)
00:05.34Ice_StrikeCallerID work well on value type but not on premium
00:05.48*** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net)
00:05.51[TK]D-Fendernfsnobody: "custom coded" ?  EVERYTHING is just code and is always "custom"
00:06.34nfsnobody... Well at some level yes, but LUA offers functionality that isn't built into the dialplan application functionality
00:06.39Ice_Strike[TK]D-Fender When I set fake CallerID and then dail a number.. the operater does not allow me to continue the call.
00:06.55WIMPynfsnobody: Like what?
00:06.59Ice_Strikework well on  value type but not on premium type
00:08.10nfsnobodyoff the top of my head, non linear style, associative arrays, classes, the ability to include external modules (of which there is an extensive library available)
00:09.20nfsnobodya lot of functionality can be built it instead of called via AGI
00:16.55*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
00:16.55*** mode/#asterisk [+o mjordan] by ChanServ
00:19.11*** join/#asterisk justdave (~dave@unaffiliated/justdave)
00:27.31*** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
00:37.24*** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net)
00:50.38*** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net)
01:02.14*** join/#asterisk sixohquad (~cody@184.65.142.249)
01:08.41*** join/#asterisk PixelBen (dcf5c8ac@gateway/web/freenode/ip.220.245.200.172)
01:11.53PixelBenI'm wondering if anyone can help - trying to figure out what's the most "current" HA project suitable for asterisk. Looking to expand my asterisk install to run HA across multiple data centres and not sure if I should be looking at Asterisk-SCF, doing my own corosync/pacemaker/drbd install, or that other thing that all the kids are doing these days that I don't yet know of :P
01:13.26ectospasmI don't think Asterisk SCF is mature enough for production
01:13.44ectospasmit may be close, but not there yet.
01:14.26PixelBenyeah I got that impression from what I was reading
01:15.14PixelBengetting somewhat lost now :P spent about a month trying to implement a DRBD solution but it just doesn't perform well across a fibre link
01:16.02ectospasmfiber or fibrechannel?
01:17.00PixelBenI guess i'm just looking for some direction on what people currently consider to be the best scenario for load balancing asterisk across geographically independent locations
01:17.57PixelBenhah - i'm going to look silly - but I'm not sure, I just know that 2 of the data centres im using said I have a dedicated fibre link between two of my servers, separate from the link i use for those servers to access the internet
01:18.45PixelBenthe idea being that i'd use that dedicated link for all my DRBD traffic, which failed abysmally, every time a packet would fall outside the tolerance range for DRBD it would fence the resources and both servers would become unavailable
01:23.35*** join/#asterisk danfromuk (~IceChat77@2.26.252.244)
01:43.53*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
01:57.26*** join/#asterisk sixohquad (~cody@184.65.142.249)
02:14.26*** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net)
02:52.19*** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.27.54)
03:12.06*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)
03:19.13*** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax)
03:23.21*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)
03:26.01*** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net)
03:29.03*** join/#asterisk jaddison (~Adium@184.66.16.213)
03:29.57jaddisonHi, has anyone successfully been able to use a Motorola VT2442 ATA with Asterisk?  If it's important, the device is an old Vonage device.
03:33.56Naikrovekprobably not, since vonage doesn't use SIP
03:34.02Naikrovekbut I could easily be incorrect
03:52.58p3nguinSince when does Vonage not use SIP?
03:53.41p3nguinIf that ATA is a SIP device and it isn't locked so you can't get into it, it'll work with Asterisk.
03:54.03*** join/#asterisk ChannelZ (channelz@burner.com)
04:09.04jaddisonThe ATA is locked.
04:09.24jaddisonand after a few hours of trying to figure out how to unlock it, I officially hate Vonage.
04:13.36Naikroveki thought there was a huge discussion in here about vonage not using sip
04:13.40Naikrovekin fact I know there was
04:13.51Naikrovekeveryone was in agreement: not sip
04:14.00Naikrovekthis is what I get, I guess.
04:14.10Naikrovekfor putting ANY faith in anything anyone tells me
04:18.31*** join/#asterisk mintos (~mvaliyav@114.143.161.111)
04:30.00*** join/#asterisk woleium (~woleium@208.53.145.169)
04:37.22*** part/#asterisk jaddison (~Adium@184.66.16.213)
04:43.57*** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net)
04:51.06*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
05:01.34*** join/#asterisk screenn (~screenn@37.46.237.217)
05:18.41*** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee)
05:30.05*** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net)
05:38.20*** join/#asterisk youjelly (~bwahahaha@39.47.148.32)
05:48.43*** part/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com)
06:05.36*** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com)
06:07.04*** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md)
06:13.38*** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at)
06:13.39*** join/#asterisk din3sh (29d4ca8b@gateway/web/freenode/ip.41.212.202.139)
06:13.40schmidtsgood morning
06:15.13din3shmrning ppl
06:15.43*** join/#asterisk Yourname` (~yourname@unaffiliated/yourname/x-837320)
06:20.27*** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105)
06:21.46*** join/#asterisk chris-NB (~chris@fw.commpany.at)
06:23.37Yourname`Hello. My VoIP trunk is sending caller ID in the "From:" header of the SIP packet. How can I extract that and use the number as caller ID?"
06:35.36kaldemarYourname`: that's where it should be. what is your problem with it?
06:35.54Yourname`kaldemar: Sorry, I meant "Contact:" field..
06:37.02kaldemarcore show func SIP_HEADER
06:40.18kaldemarSet(CALLERID(num)=${CUT(SIP_HEADER(Contact),@,1):6}) or something like that, depending how the number is set in the header value.
06:48.37*** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at)
06:54.06Yourname`kaldemar: Any idea where I can modify this just the sip trunk and not for everything?
06:56.15kaldemarYourname`: in the context that is defined for your provider in sip.conf.
06:56.33Yourname`Gotcha.
06:56.39Yourname`Thanks, kaldemar
06:58.36*** join/#asterisk bulkorok (~bulkorok@85.183.36.36)
07:08.55*** join/#asterisk ChannelZ (channelz@burner.com)
07:20.00*** join/#asterisk gavimobile1 (~user@bzq-84-108-109-1.cablep.bezeqint.net)
07:20.15gavimobile1folks, why has my nick been banned? gavimobile?
07:20.47gavimobile1(21:40:55)
07:20.47gavimobile1You have been kicked by Qwell: (Fix your connection, then message an op.)
07:20.58gavimobile1I guess that's why. can someone please unban me
07:25.15*** join/#asterisk Takapa (vegard@svanberg.no)
07:27.10gavimobile1folks what requirments do I need if I want to use multiple context in voicemail.conf. nothing but [default] seems to be working for me
07:54.12*** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no)
08:02.40*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)
08:04.51*** join/#asterisk ChannelZ (channelz@burner.com)
08:07.46*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
08:07.57*** join/#asterisk youjelly (~bwahahaha@182.177.62.248)
08:09.34*** join/#asterisk roham (~ali@31.184.187.2)
08:10.50*** join/#asterisk SamuraiGhost (~kvirc@c-76-109-247-233.hsd1.fl.comcast.net)
08:11.10*** join/#asterisk s[x] (~sx]@49.178.1.51)
08:16.46*** join/#asterisk PBXman (c335d9a4@gateway/web/freenode/ip.195.53.217.164)
08:21.22*** join/#asterisk qakhan (~qakhan@182.185.209.130)
08:23.46qakhani have Digium TDM800p 8 analog ports card. it doesnt hangup call.
08:23.49qakhanplease help
08:28.52*** join/#asterisk s[x] (~sx]@49.178.3.225)
08:29.37*** join/#asterisk din3sh_ (29d4ca8b@gateway/web/freenode/ip.41.212.202.139)
08:35.21*** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl)
08:35.29jacc0hi all! good morning!
08:36.12jacc0wat is the best/good way to loop true alle rows returned by an odbc function?
08:37.15*** join/#asterisk rebourn (~rebourn@31.176.197.84)
08:37.42rebournhello
08:38.15rebournis this the proper channel for assistance with asterisk installation?
08:38.32jacc0yes it is
08:39.03jacc0you better start by telling what OS and asterisk version you are planning to install
08:39.08jacc0:)
08:39.33jacc0and tell us if you need any hardware configured
08:40.10rebournim just installing gentoo
08:40.30jacc0okay, I hazve no experiance with that
08:40.42rebournwhat is the best os for asterisk
08:40.52jacc0O?
08:41.28rebournwhat do you mean?
08:41.38qakhanplease help
08:41.39qakhani have Digium TDM800p 8 analog ports card. it doesnt hangup call.
08:41.43jacc0I did not know it was the best os for installing asterik
08:42.20rebournI was asking what is the recommended OS for asterisk
08:42.34rebournI also have 0 experience with gentoo
08:42.39rebournit was recommended to me
08:42.44jacc0okay, some say CentOS is the best
08:42.49jacc0for asterisk
08:42.54jacc0I prefer Debian
08:43.00jacc0or ubuntu
08:43.09rebournIm a bit more adapted to ubuntu
08:43.13kaldemarjacc0: use functions HASH, HASHKEYS and a while loop
08:43.32jacc0ty :)
08:44.37jacc0i will into that
08:44.39kaldemarrebourn: any linux distro that you're the most comfortable with will do.
08:44.47*** join/#asterisk youjelly (~bwahahaha@182.177.62.248)
08:45.06jacc0there are simple to install pagages for debian/ubuntu
08:45.11jacc0apt-get install asterisk
08:45.23jacc0apt-get install asterisk-1.8
08:45.29jacc0and you will be done :)
08:46.21qakhani have Digium TDM800p 8 analog ports card. it doesnt hangup call.
08:46.36qakhanthere is dead air after hangup
08:46.38rebournkaldemar: i will be working on a software called pbxware, and they advised me to use gentoo
08:47.02jacc0as you state: 'Im a bit more adapted to ubuntu'
08:47.13jacc0I would go for that if I was you
08:47.31rebournthe OS doesnt limit asterisk?
08:47.59jacc0nope
08:48.05rebourngood to hear
08:48.23rebournive beed at this gentoo installation for 2 days now
08:48.33rebournits killing me in many ways
08:48.37jacc0hehehe
08:48.48jacc0one more reason to go for ubuntu
08:49.09rebournim loving ubuntu so far
08:49.18rebournim not an experienced linux user
08:49.25rebournjust starting out
08:49.30jacc0one more reason to go for ubuntu
08:49.46rebournwill give it a shot
08:49.48jacc0you've got all the reasons to go for ubuntu
08:50.26jacc0just install ubuntu and open a shell and type: sudo apt-get install asterisk
08:50.35jacc0And you will have asterisk running in no time
08:51.04rebournI actually deleted my windows few days ago and have a pure ubuntu box right now
08:51.19Yourname`I've added a sip trunk, but the thing is, it needs to be registering to asterisk so it can send the calls.. how can I achieve that in peer settings?
08:54.22jacc0' it needs to be registering to asterisk so it can send the calls' != true
08:54.49jacc0you can also set an ip address in your peer settings
08:55.02jacc0host=192.100.100.1
08:55.14jacc0but if you want to register
08:56.06jacc0add the following line under [general] in sip.conf
08:56.10jacc0register => user[:secret[:authuser]]@host[:port][/extension]
08:58.07kaldemarYourname`: you can't.
08:58.11qakhani have Digium TDM800p 8 analog ports card. it doesnt hangup call.
08:58.40Yourname`Gotcha, thanks jacc0 and kaldemar
08:58.49kaldemarYourname`: the one who controls the remote end must make them register to you. or you must set an ip address or a host for the peer instead of host=dynamic.
08:59.27Yourname`kaldemar: I have it set to host=dynamic, but then I also use this same trunk to send calls TO...
08:59.42Yourname`kaldemar: This is a GSM gateway I have configured as a sip trunk..
09:00.18*** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md)
09:00.30jacc0if you have set host=dynamic one of the sides needs to register
09:00.32kaldemarYourname`: so make the gateway register to your asterisk.
09:00.58jacc0or make asterisk register to your gateway
09:01.03kaldemarYourname`: if the gateway can't register to asterisk, you cannot use host=dynamic.
09:03.35Yourname`Gonna try now..
09:18.22*** join/#asterisk acedia (~garland@unaffiliated/ffs)
09:19.23*** join/#asterisk bulkorok (~bulkorok@85.183.36.36)
09:21.01*** join/#asterisk youjelly (~bwahahaha@182.177.83.25)
09:33.57*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
09:35.34*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
09:45.55Ice_Strikehi
09:55.58qakhani have Digium TDM800p 8 analog ports card. it doesnt hangup call.
09:58.23Ice_Strikewhy Record() not working?
09:58.43Ice_StrikeI like to record the channel Dail()
09:59.07kaldemarapp Record is working. are you trying to record a call with it?
10:00.03Ice_Strikelike this?
10:00.05Ice_Strikeexten => 333,n,Dial(SIP/44XXXXXXX@peer)
10:00.06Ice_Strikeexten => 333,n,Record(/root/test:wav)
10:00.31kaldemarwhat are you expecting that to do?
10:01.07Ice_Strikekaldemar I expect to record what I hear from the channel but I do hang up after 30 seconds
10:02.09kaldemarapp Record is used to record from the CALLING channel. it is not meant for recording calls. and in your example the application will start after your call is finished.
10:02.48kaldemarwhat you want is Monitor or MixMonitor. and you should start one of those applications before the Dial command.
10:02.59Ice_StrikeAhh I see thanks.
10:03.00kaldemarcore show application Monitor, core show application MixMonitor.
10:03.59Ice_StrikeIsnt Dial() == CALLING channel ?
10:04.13Ice_StrikeFrom I understand, during the call it is calling channel
10:05.55kaldemarno.
10:06.35Ice_StrikeWhat is channel channel? I thought it mean telephone conversation
10:06.39kaldemarDial() is an application, and what is given as an argument for app Dial is the CALLED channel. calling channel would be the one that is in dialplan and makes app Dial execute.
10:07.48kaldemarchannel is not a telephony conversation, it is a channel between two endpoints. since asterisk is a B2BUA (back-to-back user agent), every call through it has two channels.
10:08.03Ice_StrikeI see, thanks
10:13.41qakhankaldemar i have Digium TDM800p 8 analog ports card. it doesnt hangup call.
10:16.02din3sh_qakhan: busydetect is on?
10:18.34qakhanyes
10:19.34qakhanwhen i hangup it shows DAHDI channel hangup, but caller phone still on call and there is dead air
10:23.43kaldemarqakhan: describe the whole scenario
10:24.07kaldemarwhat calls what and who hangs up etc.
10:24.19qakhankaldemar i have Digium TDM800p 8 analog ports card. it doesnt hangup call.
10:26.53qakhanu r caller and i m using * ext. when i hangup call then * cli shows Dahdi channel get hangup, but call is still continue on your phone and these is dead air. adter 1 min call get disconnect
10:27.53kaldemarbe elaborate. "* ext" does not mean anything.
10:28.40kaldemarand you did not say how the call comes in either.
10:29.31*** join/#asterisk woleium (~woleium@208.53.145.169)
10:29.38qakhan* ext mean asterisk extension. caller call on number and call get connected to my extension
10:29.57Ice_Strikekaldemar I can see two files /var/spool/asterisk/monitor (IN and OUT files)... I only want to record IN
10:30.08Ice_StrikeI dont want to record what I speak to them
10:31.00schmidtsIce_Strike then delete it afterwards
10:32.35kaldemarqakhan: "asterisk extension" does not mean anything either. what technology?
10:32.42*** join/#asterisk jkroon (~jkroon@dsl-244-33-172.telkomadsl.co.za)
10:32.44jkroon~book
10:32.44infobotAsterisk: 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
10:33.05qakhankaldemar sir sip extension
10:33.21qakhanlike 1002 or 1003
10:33.35Ice_StrikeIt seem Monitor() is IO Intensitive - not good for VPS
10:34.33din3sh_He means FXO calls remain on hook 1min after actual call has been disconnected
10:34.40*** join/#asterisk screenn (~screenn@37.46.237.217)
10:34.48din3sh_fxo calls to SIP extensions
10:35.27WIMPyon? off?
10:36.28kaldemardin3sh_: and how do you know that? he hasn't mentioned FXO.
10:36.39*** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au)
10:36.48din3sh_TDM800p 8 analog ports card
10:37.24din3sh_<qakhan> kaldemar i have Digium TDM800p 8 analog ports card. it doesnt hangup call.
10:37.25din3sh_lol
10:37.33din3sh_he's been pasting that for few hours now
10:37.34qakhandin3sh_ yes you are right
10:37.35kaldemardin3sh_: which might be FXO or FXS. connected to anything. he has yet to reveal the details.
10:37.45kaldemardin3sh_: for a few weeks.
10:37.50din3sh_weeks?
10:37.53qakhandin3sh_ got my point
10:37.54din3sh_:O
10:38.05qakhankaldemar all ports are fxo ports
10:38.16din3sh_qakhan
10:38.27qakhanyes din3sh_
10:38.28kaldemardin3sh_: the 16 times i said earlier was based on a log grep. :P
10:38.42din3sh_you must find out the busytones, ringtones, used by your telco in your country
10:38.47*** join/#asterisk sekil (~sekil@78.24.104.73)
10:38.55din3sh_and use accordingly in indications.conf
10:39.05qakhani have all set these settings
10:39.21qakhanbut same result
10:39.25din3sh_what country are you located?
10:40.48qakhanPakistan
10:41.21Ice_StrikePakistan goverment hate VOIP.
10:41.28qakhanand there is not setting for Pakistan in indications.conf
10:41.38qakhanIce_Strike its not true
10:42.00Ice_Strikeqakhan Come on bro, it is true. They do not like people to use VOIP.
10:42.14qakhanIce_Strike where are you from?
10:42.16The_BallI can make sip extensions work when using numbers for the sip account, however is it possible to use names for incoming sip calls? I'd like to setup an account like alex@mydomain.com instead of 201@mydomain.com
10:42.36Ice_Strikeqakhan England
10:42.42Ice_Strikebut paki
10:42.54*** join/#asterisk timahvo1 (~rogue@197.177.46.192)
10:43.19qakhanIce_Strike you should take complete name Pakistan instaed of paki
10:43.27kaldemarThe_Ball: you can use pretty much whatever you want.
10:43.52qakhanPakistan is opterating alot of call centers
10:44.08qakhanany ways din3sh_ can you help me in this
10:44.09Ice_Strikeqakhan Hmmm
10:44.21The_Ballkaldemar, that's what I tought
10:45.41din3sh_<qakhan> and there is not setting for Pakistan in indications.conf
10:46.09din3sh_<din3sh_> you must find out the busytones, ringtones, used by your telco in your country/  <din3sh_> and use accordingly in indications.conf
10:46.33din3sh_you have to modify the indications.conf with setting used in pakistan dude
10:46.52The_Balldin3sh_, does it matter if you use different indications to what is common in the country?
10:47.24din3sh_he's struggling to deal with busytone it seems
10:47.26din3sh_might help
10:48.20*** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au)
10:48.55qakhandin3sh_ not for busytone, it doesnt detect busytone after hangup but i can hear 3 busytone after 1min and then call get hangup
10:53.26*** join/#asterisk s[x] (~sx]@60-241-151-10.tpgi.com.au)
10:58.14qakhandin3sh_ should i create my own context in indications.conf? if yes then how it will takes it?
10:58.38din3sh_you can try
10:59.30qakhanbydefault which context it takes?
11:03.47*** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
11:07.44Yourname`Anyone have experience with Eurotech GSM gateway?
11:10.41*** join/#asterisk devmikey (~irc@unaffiliated/devmikey)
11:16.09*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
11:16.09*** mode/#asterisk [+o mjordan] by ChanServ
11:21.53*** join/#asterisk Blowfish-nc (~Blowfish@217.199.135.70)
11:22.02Blowfish-nchey guys
11:24.28Blowfish-nci need help
11:24.34*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)
11:28.59Ice_StrikeIs there a command to hang up the call after SendDTMF() application?
11:29.17Ice_StrikeS() option for dail application is not a solution
11:31.05jacc0hangup() ?
11:34.18Ice_StrikeNo I don't think that will work jacc0.
11:34.35Ice_StrikeAfter using SendDTMF() - Then the operator speak..
11:34.41Ice_Strikeit then start recording..
11:34.43Ice_Strikethat worked well.
11:35.00Ice_StrikeI want to hang up the call after 30 seconds.
11:35.24Ice_StrikeHang up the call after 30 seconds when SendDTMF() has been executed
11:35.33*** join/#asterisk hehol (~hehol@2001:1438:1009:200:fd99:ad8f:91a4:8cfe)
11:38.27Blowfish-ncis there any way to set up a virtual asterisk server with virtual phones for developing and testing purposes?
11:40.21jacc0if you don't mind choppie sound
11:41.03Ice_Strikejacc0 I am using VPS without any choppie sound
11:41.13Ice_StrikeMake sure you got Xen VPS
11:41.26Ice_Strikeand also decent I/O speed
11:41.29Ice_Strikeyou should be fine
11:41.50Ice_StrikeLinode or FurtureHosting I would recommend
11:42.03Ice_StrikeDon't go with OpenVZ
11:42.46jacc0the problem is the udp trafic
11:43.05jacc0of the vm is not ready and waiting for the packages they will be lost
11:43.27jacc0you could experiment with TCP maybe it helps
11:43.43Ice_StrikeA lot of people use VPS for game server
11:43.46Ice_Strikeit work fine for them
11:43.55Ice_StrikeJust make sure you dont go with cheap vps
11:45.06jacc0I'm not sure the digium hardware is supported inside a vm
11:45.49Ice_Strikejacc0 Xen will support digium hardware
11:45.52Ice_Strikebut not OpenVZ
11:46.12Ice_StrikeOpenVZ = you can't recompile kernel
11:50.39*** join/#asterisk wonderworld (~ww@dsdf-4db50155.pool.mediaWays.net)
11:51.20schmidtsIce_Strike but there is a way to use hardware within OpenVz, its a little bit tricky but possible
11:54.37*** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de)
11:54.37*** part/#asterisk ixyd_ (~denzs@carbon.gonicus.de)
11:54.39*** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de)
11:55.16Ice_StrikeHave anyone used SMS Gateway hardware with asterisk?
11:55.22Ice_StrikeI am thinking buying one.
11:55.52Ice_StrikeDoes it work the same way like making a call using VOIP?
11:55.59Ice_StrikeDail(...)
11:56.39*** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
11:59.16ixyd_hey guys, iam trying to find the reason for some of our CDRs have the disposition FAILED - i already found out Dialing a sippeer which is not registered will make the disposition FAILED - but i would like to know, if there a more possibilities to reach this disposition.... maybe some of you guys have a hint for me? :)
11:59.48*** join/#asterisk Bullmoose (~Bullmoose@70-58-133-94.bois.qwest.net)
12:00.54WIMPyIce_Strike: Use Option L to Dial.
12:04.40*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
12:05.58Ice_StrikeWIMPy Aha.
12:15.49Blowfish-ncanyone?
12:17.11WIMPyBlowfish-nc: Unless you add hardware, it's pretty virtual. What more (or less) do you want?
12:18.20Blowfish-nci need to develop an application that checks connections to other phones in the network
12:18.23Blowfish-ncproblem is
12:18.30Blowfish-ncthose phones are in another company
12:18.56Blowfish-nci followed the asterisk java tutorial
12:19.08WIMPyLike the built-in qualify option?
12:19.54[TK]D-FenderBlowfish-nc, "sip show peers" <-
12:20.34Blowfish-ncthat or disposition
12:20.43Blowfish-nci can figure that out
12:21.03Blowfish-ncbut the problem is i can't connect to the other company's phones
12:21.14Blowfish-ncso i can't test the application
12:21.24Blowfish-ncand my company does not have asterisk
12:22.04Blowfish-ncneed a virtual server+phones setup
12:22.28WIMPyAnd what's stopping you from doing so?
12:23.26Blowfish-nci don't know if it's possible to make virtual phones
12:23.39WIMPyIf yu try to run softphones on the same machine, you must make sure tey're using different ports. Bot from one another and from Asterisk.
12:23.44Blowfish-nclike dummy phones that i can call in my app
12:24.31*** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net)
12:25.02WIMPyBut I still have no idea, what that application is supposed to do.
12:25.29[TK]D-FenderBlowfish-nc, how do you expect * to even KNOW about these other phones?
12:25.57[TK]D-FenderBlowfish-nc, And without them registering to *.... you have no solution with *
12:26.07Blowfish-nci don't know fender, i'm totally new to asterisk
12:26.22*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
12:26.54Blowfish-ncthe applications needs to call other phones connected to the server and just check the line; if it's working or not, busy, etc
12:26.54[TK]D-FenderBlowfish-nc, It will not do what you've asked
12:27.21WIMPyYou don;t need any application to find that out.
12:27.26[TK]D-FenderBlowfish-nc, Phones don't randomly hand over information on their activities to just anything that asks
12:27.35WIMPyThat information is readily available in Asterisk.
12:27.47Blowfish-ncis that disposition_
12:27.48Blowfish-nc?
12:28.26WIMPyhas no idea, what "disposition" you're talking about.
12:29.23Blowfish-nchttp://www.asterisk-java.org/development/apidocs/org/asteriskjava/live/Disposition.html
12:29.30Blowfish-ncthis is basically what i need
12:30.09Blowfish-ncbut problem is
12:30.12[TK]D-FenderBlowfish-nc, that is a DIALSTATUS retun.  The phones aren't even registered to * nor are they placing calls through it
12:30.22[TK]D-FenderBlowfish-nc, Asterisk will play NO role in your solution
12:30.43[TK]D-FenderBlowfish-nc, If has no clue what activities that phone partakes in on some other server
12:30.45WIMPyYes, that is after placing a call.
12:31.42Blowfish-ncthe problem is that i can't connect to the other company's server (it's in another country)
12:31.45WIMPyYou have to ask the server the phones are using. Assuming it's only one server.
12:31.48Blowfish-ncso i need something like a local server
12:31.55Blowfish-ncwhere i can test things out
12:32.05Blowfish-ncthen send the java app to the other company
12:32.18*** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net)
12:32.23WIMPyThere is nothing to test as there's no need for any application.
12:32.45Blowfish-ncwhat?
12:32.57WIMPy>>That information is readily available in Asterisk.
12:33.23WIMPyUnless you come up with something you didn't mention so far.
12:33.29Blowfish-ncwell
12:33.58Blowfish-ncthe phone lines have to be checked every hour
12:34.14Blowfish-ncand saved to a report
12:34.28WIMPyPhones or lines? And what kind of lines?
12:34.48Blowfish-ncmaybe i'm using bad terminology
12:35.12[TK]D-FenderBlowfish-nc, "core show channels"
12:35.34WIMPyYes, sometimes asking the right question is the key :-)
12:35.42Blowfish-nclet's say there are 10 phones in the network, i need to check if they are all connected and working
12:35.50WIMPySo tell us exactely whay you need.
12:36.06WIMPyOk. As said: Asterisk will do that on its own.
12:36.19Blowfish-ncbut how do i save that information to a report
12:36.38*** join/#asterisk timahvo1 (~rogue@197.178.175.70)
12:36.45[TK]D-FenderBlowfish-nc, I've already given you the 2 commands for this.
12:36.47[TK]D-FenderBlowfish-nc, "core show channels"
12:36.52[TK]D-FenderBlowfish-nc, "sip show peers" <-
12:37.37WIMPyYou can get it from the log file, if enabled, you can listen on AMI for events (in realtime) or you can check the current status as [TK]D-Fender said.
12:39.14Blowfish-ncsip show peers seems to be useful for me, thanks
12:39.34Blowfish-ncbut i still need to make a java application that they can run
12:39.51Blowfish-ncthat application needs to connect to the server and run sip show peers
12:40.10WIMPyOh, and as mentioned at the beginning, the qualify option needs to be enabled in sip.conf.
12:40.44Blowfish-nci will do that
12:40.54*** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com)
12:41.15[TK]D-FenderBlowfish-nc, AMI <-
12:41.18[TK]D-Fender~book
12:41.18infobotAsterisk: 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
12:41.19[TK]D-Fender^^^
12:42.11Blowfish-ncbut i need to have an environment to test the app that i'm developing
12:42.38Ice_StrikeHow do I check the ping speed between UK server (asterisk installed) and voip provider located in USA?
12:42.41Blowfish-nchow do i test my app without having an asterisk server running
12:42.49WIMPyInstall Asterisk and some soft phones.
12:42.50*** part/#asterisk Bullmoose (~Bullmoose@70-58-133-94.bois.qwest.net)
12:42.53[TK]D-FenderIce_Strike, "man ping"
12:43.20Ice_StrikeI know about that ping
12:43.29Ice_Strikei meant check the speed from asterisk?
12:43.53[TK]D-FenderIce_Strike, *'s qualify timeout it not a usable metric for performance
12:44.07[TK]D-FenderIce_Strike, and "sip show peers" lists it to you
12:44.12[TK]D-FenderIce_Strike, If you've enabled it
12:44.26Ice_Strikeoute/611xxxxx         216.115.69.xxx              5060     Unmonitored
12:44.34[TK]D-FenderWhich clearly you haven't
12:44.45[TK]D-Fender"Unmonitored"
12:45.01Ice_StrikeOn the voipms, it say  OK (2 ms)
12:57.24*** join/#asterisk ghghz (~ton@kluonis.kvb.lt)
12:57.50ghghzhey. how to implement simpliest way dial and when answered play a mp3 file?
12:58.01ghghzbut dial should be originated from asterisk console
12:59.10WIMPychannel originate ... application playback ...
12:59.42*** join/#asterisk serafie (~erin@nat/digium/x-rwufhlmduivdmsan)
13:02.09*** join/#asterisk mjordan (~mjordan@nat/digium/x-npmjbbtjchjgdwiu)
13:02.09*** mode/#asterisk [+o mjordan] by ChanServ
13:04.20jacc0how can I check if a file exists in dialplan?
13:05.08leifmadsenjacc0: STAT()
13:06.10*** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net)
13:09.14Ice_StrikeIf I want to make a lot of regular call to UK, is it recommended to look for VOIP provider in Europe?
13:09.26Ice_Strikeand get a server in Europe
13:09.58Ice_Striketo improve performance and avoid choppy sound
13:10.22*** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au)
13:11.48[TK]D-Fenderthe hop to think about is between your server and provider
13:14.12leifmadsenIce_Strike: if you still have to send the data from NA to Europe, that won't really matter
13:14.33Ice_StrikeNA?
13:14.37leifmadsenadding another server there, if anything, could potentially even add latency (however minor)
13:14.40leifmadsenNA == North America
13:14.45Ice_Strikeoh
13:15.19Ice_Strike@leifmadsen My VOIP with flowroute and my server in London
13:15.44leifmadsenIce_Strike: not sure how that helps
13:15.55leifmadsenwhere are your phones located?
13:16.17Ice_Strike@leifmadsen Phone is located in UK
13:16.25Ice_StrikePhone connect to London Server.
13:16.39Ice_StrikeLondon server is connected to Voip provider in USA
13:16.50leifmadsenoh ya, if everything is physically residing in London, then anything you can do to drop the hops is better
13:17.08leifmadsenyou could ask flowroute if they have an EU presence
13:17.22Ice_StrikeThey don't I have checked.
13:17.44Ice_Strike@leifmadsen I also have a server in USA
13:17.55Ice_Strikewould that improve any performance?
13:17.59leifmadsenthat doesn't help you any
13:18.02leifmadsenwhy would it?
13:18.27Ice_StrikeBecause less ping between usa server and voip provider in usa?
13:18.31Ice_Strikeabout 30ms
13:18.45leifmadsenok... but how does that help with the stuff located in the UK?
13:18.55leifmadsenyou still have to traverse to the USA server
13:19.07leifmadsenyou have to look at the whole route to determine the latency
13:19.20leifmadsenadding a server in the US does not instantiate magic
13:19.46*** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net)
13:20.55Ice_Strikeoh yea but I wont be using the hardware/soft phone to make call.. In extensions.conf I will be adding loop to make regular call and save recording files.
13:21.06ghghzhow to set callerid(num) in using originate channel?
13:21.32jpsharpIn my experience, latency doesn't become really noticable until you pass 200ms.
13:21.44jpsharpAnd even then, its workable until you pass about 750ms.
13:21.52[TK]D-Fenderghghz, Uoui can't set callerid from CLI originate, only AMI Originate ro CAll Files
13:22.01[TK]D-FenderYou*
13:23.39Ice_Strike@leifmadsen In the extensions.conf there will be a loop to make a lot of call by itself and record it for 20 seconds...  I wont be using hardware phone and software phone....
13:23.48Ice_Strikewell software phone just for testing.
13:23.58leifmadsenok, well it seems that every answer I give comes against a lack of information unfortunately
13:25.22Ice_Strike@leifmadsen It may sound choppy when it has to traverse to the USA server but the recording via asterisk should be affected?
13:25.33leifmadsenwhy would it sound choppy?
13:25.38leifmadsenthat is jitter, not latency
13:25.54leifmadsenand you can compensate for that with a jitterbuffer
13:26.18[TK]D-FenderWhatever is between * and your provider is what will change the outcome of your recording
13:31.09*** join/#asterisk _Corey_ (~chatzilla@64.215.11.114)
13:32.32*** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt)
13:32.56*** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134)
13:33.45*** join/#asterisk BeeBuu (0ed6e2a1@gateway/web/freenode/ip.14.214.226.161)
13:33.54BeeBuuhello,all
13:34.19[sr]hi
13:34.28*** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan)
13:36.08BeeBuui got a problem,please help me.I had made a programm to catch the AMI event,it was worked fine when calling 20 channels,but when 40channels, the Asterisk disconnect my programm connection. Any help please?
13:40.14*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
13:40.14*** mode/#asterisk [+o putnopvut] by ChanServ
13:40.49fpriorHi, I've one client (Zoiper) in LAN that is registering as sip:21@206.55.127.159:62399 instead of sip:26@192.168.0.40:5060; nat=no in sip.conf; anothers clients are OK. Any idea ?
13:43.36Ice_StrikeIs it possible to dial the numbers on the server without using soft/hardware phone?
13:43.46Ice_StrikeIt is only going to be used for recording.
13:43.56[TK]D-FenderIce_Strike, AMI / CLI Originate, Call Files
13:43.58[TK]D-Fender~book
13:43.58infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
13:43.59[TK]D-Fender^^^
13:44.11*** join/#asterisk teloniusz (goldie@inferno.hell.pl)
13:44.18Ice_Strike[TK]D-Fender Ah thanks.
13:44.46*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
13:45.29*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
13:49.04*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)
13:49.39teloniuszhi, can I ask a question concerning AGI and a2billing here? there is virtually no people on #a2billing
13:51.31jpsharpAsk the question and see who/if anyone can answer.
13:51.39teloniuszanyways... I have a problem with a2billing - it reports me that every connection has 0 seconds. I know the connections are established, I can talk through it
13:52.25[TK]D-Fenderteloniusz, this is no 2nd level support for a2billing
13:52.32[TK]D-Fendernot*
13:52.38teloniuszI'm trying to debug AGI with 'agi set debug on' and I see that the a2billing application starts DIAL app
13:52.52BeeBuucan anyone help?
13:52.59*** join/#asterisk fhmiv (~fhmiv@c-68-47-6-72.hsd1.ga.comcast.net)
13:53.26teloniuszbut DIAL app returns only when the connection is finished, not whet it's established.
13:53.39*** join/#asterisk m_tadeu (~quassel@89.180.88.192)
13:54.02teloniusz[TK]D-Fender: this is, in fact, AGI question, not a2billing question.
13:54.08[TK]D-Fenderteloniusz, Like EVERY APP.
13:54.38[TK]D-FenderDialplan processing is linear, not "background"
13:54.42*** join/#asterisk frawd (~francois@92.Red-88-11-98.dynamicIP.rima-tde.net)
13:54.46*** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net)
13:55.13teloniusz[TK]D-Fender: I know, I'm new to AGI, but I've written several scripts.
13:56.31teloniusz[TK]D-Fender: so, there's no difference with AGI, it cannot do asynchronous dial, right?
13:56.41[TK]D-FenderCorrect
13:57.42teloniuszSo... asterisk CDRs don't report any ANSWERED call, yet debug says:     -- SIP/10.40.1.21-0000008f answered SIP/7284633182-0000008e
13:58.10[TK]D-FenderHas the call ENDED?
13:58.41teloniusz[TK]D-Fender: yes, after all that I hung up one side and the call ended.
14:00.37teloniuszSIP debug shows BYE packet, then OK.
14:01.28teloniuszwaaait
14:02.04teloniuszOK, disregard this. Problem is stranger than that, it started to bill properly some connections and I didn't change anything :>
14:02.24teloniuszanyways, thanks :>
14:02.42teloniuszI'm going to isolate the problem.
14:02.45jpsharpHeisenbugs, yay!
14:03.23Ice_StrikeWhat the minimum RAM do I need for 15 simultaneous calls (SIP/ulaw) with recording at the same time?
14:03.55jpsharpEnough to boot the OS.  Asterisk isn't especially heavy on memory.
14:05.42Ice_StrikeCool
14:06.00Ice_StrikeLet say 1GB Ram
14:06.57[TK]D-Fenderfar more than enough.
14:07.10[TK]D-FenderYou could get away with 256meg easily enough
14:08.10Ice_Strikeeven with recording?
14:08.24Ice_Strikewow...
14:08.31*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
14:09.12jpsharpNot unless you're recording to a ramdisk.
14:09.20[TK]D-FenderRecording doesn't waste memory... it gets written to DISK
14:09.40Ice_StrikeYea it does.
14:11.03*** join/#asterisk Defraz (~Defraz@69.20.176.131)
14:12.48*** join/#asterisk danfromuk (~IceChat77@2.26.252.244)
14:13.46*** join/#asterisk krotos (~d3v1l@host192-95-dynamic.1-79-r.retail.telecomitalia.it)
14:13.48krotoshi all
14:17.46Ice_Strikehi
14:21.25*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
14:21.25*** mode/#asterisk [+o sruffell] by ChanServ
14:25.30krotosthere is any way to make DMTF over RFC2833
14:25.36krotosa little bit too long?
14:25.57*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
14:26.40*** join/#asterisk sekil (~sekil@78.24.104.73)
14:39.58*** join/#asterisk sereal-work (~sereal@unaffiliated/sereal)
14:40.23m_tadeuhi all...does anyone know if gtalk video support is available for asterisk?
14:43.58Qwellheh
14:44.04Qwellm_tadeu: How experimental would you like to get?
14:44.22filewhistles
14:44.46m_tadeuQwell: As experimental as it gets ;)
14:44.47fileI haven't finished writing it yet
14:44.50Qwellfile: pfft
14:45.08tzafrirgrrr.... sending a fax to somewhere does not work. It's a specific company to which we need to send a fax (not call. Not email. Fax).
14:45.38tzafrirAnd the funny thing is that a normal phone call gets through to them but a fax gives a fast busy immediately
14:45.58tzafrirWhich made me realise how easy it is to do this intentionally
14:46.07WIMPytzafrir: The good old BC thing?
14:46.43tzafrirIf you want to have a line that will never accept faxes but will accept test phone calls, so you can always blame the others for not trying good enough
14:47.00tzafrirWIMPy, BE / BI, I guess
14:47.31WIMPywas referring to CHANNEL(transfercapability)
14:48.43WIMPyBut if a phone call works, it must be them having some issue.
14:49.14tzafrirWIMPy, the fax call was sent as SPEECH
14:49.29WIMPyWell, that's bad.
14:50.43WIMPyErr, forget the line befor that.
15:02.32*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
15:22.15*** join/#asterisk hehol (~hehol@217.9.101.222)
15:23.36*** join/#asterisk Ad-Hoc (~nimbus@athedsl-377652.home.otenet.gr)
15:29.13*** join/#asterisk retentiveboy (~retentive@72.54.144.26)
15:31.47Ice_StrikeI want to make 10 current call from 1 user, is it possible to specify each call name?
15:32.11Ice_StrikeExample: CallRobot1, CallRobot2 from 1 user.
15:35.01*** join/#asterisk brdude (~brdude@12.155.183.30)
15:39.15*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
15:40.18*** join/#asterisk vinhdizzo (~vinh@dhcp-v030-210.mobile.uci.edu)
15:40.53[TK]D-FenderIce_Strike, what is this "user"?  How are the calls "placed"?  Where does any call have a "name">
15:42.46Ice_Strikeuser is a phone that you specify in sip.conf
15:42.53Ice_Strikeex: [phone1]
15:43.18[TK]D-FenderWell every call is just a call... so what does "named" mean?
15:45.09*** join/#asterisk Bullmoose (~Bullmoose@70-58-133-94.bois.qwest.net)
15:46.50leifmadsenIce_Strike: you mean the callerID?
15:47.03leifmadsendefine "user"
15:49.22*** part/#asterisk ixyd_ (~denzs@carbon.gonicus.de)
15:52.13*** join/#asterisk gusto (~gusto@88.128.172.42)
15:54.45gustotha, is that 2-5 or 5-2 that version there?
15:55.00gustoah may
15:55.21gustook, i see, so recently a new version of asterisk came out, thats good
15:55.41QwellThere's always recently a new version of Asterisk.
15:55.49gustoreally?
15:56.06QwellReleases are typically done monthly.
15:56.14leifmadsen4-6 weeks
15:56.19leifmadsenmajor releases yearly
15:57.25gustocool
15:59.32gustobut 1.8 does only have security and bugfixes, right?
15:59.38Qwellno
15:59.38leifmadsenright
15:59.43Qwellerr
15:59.43leifmadsenno new features
15:59.47gustoso new functionality is only in 10, the new one
15:59.48QwellI totally missed the bugfixes part
15:59.51Qwellno
15:59.54leifmadsengusto: also no
15:59.55Qwell~asterisk versions
15:59.59leifmadsennew features in trunk
15:59.59Qwell~asteriskversions
16:00.04leifmadsen~asteriskversioning
16:00.04infobotAsterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
16:00.09Qwellinfobot: I hate you so much sometimes.
16:00.09infobotYou hate you so much sometimes.?
16:00.14Qwellyes.  that too
16:00.22leifmadseninfobot: I hate infobot a lot sometimes
16:00.23infobotYou hate infobot a lot sometimes?
16:00.25leifmadsenyes
16:00.25Qwellinfobot: asterisk versions is <reply> asteriskversioning
16:00.25infobotQwell: okay
16:00.28Qwell~asterisk versions
16:00.29infobotasteriskversioning
16:00.34leifmadsenheh
16:00.34gustoah, i did not even know, that there is a differnce between trunk and 10, i thaught that 10 would be the trunk
16:00.38Qwellinfobot: asterisk versions is see asteriskversioning
16:00.38infobot...but asterisk versions is already something else...
16:00.41leifmadsenQwell: I think <asteriskversioning>
16:00.42Qwellinfobot: no, asterisk versions is see asteriskversioning
16:00.42infobotokay, Qwell
16:00.46Qwell~asterisk versions
16:00.46infobothmm... asterisk versions is see asteriskversioning
16:00.51Qwellp3nguin: FIX IT
16:00.59Qwellinfobot: no, asterisk versions is ~asteriskversioning
16:00.59infobotokay, Qwell
16:01.01Qwell~asterisk versions
16:01.02infobotasterisk versions is probably ~asteriskversioning
16:01.08Qwellinfobot: forget asterisk versions
16:01.08infobotQwell: i forgot asterisk versions
16:01.16leifmadseninfobot: no, asterisk versions is reply asteriskversioning
16:01.16infobotleifmadsen: okay
16:01.21leifmadsen~asterisk versions
16:01.21infoboti guess asterisk versions is reply asteriskversioning
16:01.24leifmadsenbah
16:01.32QwellI totally just watched p3nguin do this.
16:01.35leifmadsenI always forget that one :)
16:01.40gustobtw. why is p3nguin's nick p3nguin? does he look like a penguin?
16:01.41leifmadsenya I know it's possible
16:01.42*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
16:01.49Qwellgusto: He looks more like a p3nguin.
16:02.05leifmadsenpthreenguin
16:02.23gustolike three penguins?
16:02.36*** join/#asterisk tomwish (~Tom@75-172-177-253.phnx.qwest.net)
16:03.08gustoi have a devil soft toy on my desk, that one is looking good
16:03.11leifmadsenno, three guins
16:03.40Qwellleifmadsen: GAH, you're gonna love this.
16:03.47Qwellinfobot: no, asterisk versions is <reply> see asteriskversioning
16:03.47infobotQwell: okay
16:03.52Qwell~asterisk versions
16:03.52infobotAsterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
16:03.55QwellJERK
16:03.58Qwellinfobot: no, asteriskversions is <reply> see asteriskversioning
16:03.59infobotokay, Qwell
16:04.03Qwell~asteriskversions
16:04.03infobotAsterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
16:04.17Qwellleifmadsen: protip: /msg infobot help redirection
16:06.17gustowow asterisk 11 will be LTS? and comming this year? that's cool, so i can upgrade from 1.8 to 11 then ;-)
16:06.39Qwelllots of cleanup goodies in 11
16:06.46gustohm?
16:06.50gustowhat kind of?
16:07.04QwellStuff to make life easier, more stable, etc
16:07.21gustoah
16:07.29gustoso you mean sourcecode cleanup?
16:07.32Qwellpretty sure there's a list of the goals on the wiki
16:07.38Qwellit's more than cleanup
16:07.45*** join/#asterisk serafie (~erin@nat/digium/x-oxcnpqohmpbwnkcd)
16:08.18Qwellhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects
16:08.57gustohttps://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects
16:09.00gustoah
16:09.08gustoi found that too
16:09.38leifmadsen~asterisk11
16:10.08*** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman)
16:10.39verywisemanwhich is better in analog telephony card Digium or sangoma?
16:10.42leifmadseninfobot: asterisk11 <reply> Asterisk 11 is slated to be an LTS (long term support) release. You can find the projects scheduled to be included in Asterisk 11 at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects
16:10.42infobotleifmadsen: okay
16:10.45leifmadsen~asterisk11
16:10.58leifmadsenfail
16:11.00Qwellhuh
16:11.08leifmadsenoh , I think I forget the 'is'
16:11.14Qwellbut he said ok
16:11.21Qwell~asterisk11
16:11.31Qwelloh, haha
16:11.37Qwell~asterisk11 <reply> Asterisk 11
16:11.38infobotextra, extra, read all about it, asterisk11 <reply> asterisk 11 is slated to be an LTS (long term support) release. You can find the projects scheduled to be included in Asterisk 11 at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects
16:11.43Qwellinfobot: forget asterisk11 <reply> Asterisk 11
16:11.43infoboti forgot asterisk11 <reply> asterisk 11, Qwell
16:11.44leifmadsenfuuuu
16:12.29Qwellinfobot: asterisk11 is <reply> Asterisk 11 is slated to be an LTS (long term support) release. You can find the projects scheduled to be included in Asterisk 11 at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects
16:12.29infobotokay, Qwell
16:12.33Qwell~asterisk11
16:12.33infobotAsterisk 11 is slated to be an LTS (long term support) release. You can find the projects scheduled to be included in Asterisk 11 at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects
16:12.36leifmadsennice
16:12.40leifmadsenforgot the 'is'
16:12.59QwellI wonder how many weird ass entries like that one exist in his db
16:13.04leifmadsenheh
16:13.06leifmadsenindeed
16:13.14leifmadsen~asterisk10
16:13.14infobotAsterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
16:13.50pabelanger~asteriskX
16:14.10Ice_StrikeWhen using a softphone - I want to set maximum 10 concurrents calls. For each concurrents channel to have unique callerID?
16:14.16Ice_Strikenot sure if im making any sense!
16:14.20leifmadseninfobot: asterisk10 is also a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15.
16:14.20infobotleifmadsen: okay
16:14.23WIMPyiAsterisk?
16:14.33leifmadsenpabelanger: you're trying to open pandoras box
16:14.35leifmadsen~asterisk10
16:14.35infobotAsterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/, or a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15.
16:20.22*** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net)
16:21.13*** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt)
16:21.14cuscohey
16:21.20cuscoto store audio
16:21.28cuscowhat is a good format to use, to save disk space?
16:21.43Qwellit depends
16:21.45Ice_Strikemp3
16:21.49cuscoew
16:21.50QwellGod no
16:21.57QwellAnything but mp3.
16:22.02cuscoim storing in gsm right now
16:22.04cuscoquality is ok
16:22.07Ice_Strikehe said save disk space
16:22.10Qwellcusco: stick with gsm then
16:22.23cuscook
16:22.28cusconow some recordings
16:22.37cuscoI listen them in asterisk
16:22.48cuscowhile I do, I use sox to convert them to al
16:22.54cuscosince we're using g711 a
16:23.04cuscobut.. it takes a huge amount of space
16:23.16cuscoand I will get no better quality than I had in gsm
16:23.25cuscoso I should actually keep it in gsm, right?
16:23.41cuscoIm storing those as cache, so I don't have to copy them again
16:23.44cuscoand its eating disk
16:23.47cuscoheh
16:26.44p3nguinWhen barging in on a call with ChanSpy(), I want to play a tone or other notification sound that indicates I have barged in.  How can I play some form of notification into the channel(s) I am attaching to?  If I was Dial()ing, I would use a subroutine and play a sound file into the called channel, but I can't determine a way to do it when using ChanSpy().
16:27.09Qwelloh, sure, now he shows up
16:27.43QwellI'm surprised there's no option for that
16:31.16p3nguinFrom what I can tell, the beep sound that I hear when I do not use option q is only played to me and not to the channel I am attaching.  I'd like for that beep sound to be heard by the channels I am going to spy on.
16:31.32QwellYou are correct.
16:31.33*** join/#asterisk paulc (~paulc@unaffiliated/paulc)
16:36.28*** join/#asterisk BMJ (~bjohns@nat/digium/x-uohinsznlvteteax)
16:36.29*** mode/#asterisk [+o BMJ] by ChanServ
16:37.56*** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net)
16:39.02Ice_Strike201 exten is already on the call, how do I listen that call when I am on different phone?
16:39.36p3nguinExtenSpy(201)
16:39.40Ice_StrikeThanks!
16:43.23*** join/#asterisk timahvo1 (~rogue@197.176.13.81)
16:46.57talntidhm, what would happen if you did exten => 201,1,ExtenSpy(201)
16:50.03*** join/#asterisk fhmiv (~fhmiv@c-68-47-6-72.hsd1.ga.comcast.net)
16:50.19p3nguinYou might see the fatal error about a channel not being able to connect to itself, and then the call would end.
16:50.21cuscosox is cpu intensive in gsm :(
16:50.51*** join/#asterisk bluregard (~matt@c-71-201-99-216.hsd1.il.comcast.net)
16:55.21Ice_StrikeLet say I have 3 phones, all of them connected to User1 (sip.conf)
16:55.57Ice_Strikeif phone 1 dialed exen 501
16:56.06Qwell~book
16:56.07infobotAsterisk: 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
16:56.19QwellThe questions you've been asking suggests that you've not read that.  You should read it.
16:56.22Ice_Strikeand phone2 also exten 501, how to prevent that?
16:57.15p3nguinYou can't register three different phones to one peer account at the same time.
16:57.49*** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
16:57.56Ice_StrikeOh
16:58.11p3nguinBut if you meant something different... you can prevent more than one call to an extension by using GROUP and GROUP_COUNT.
16:59.40Ice_StrikeLet do this way.. 3 phones registered to different accounts (User1,User2,User3)... If phone1 dialed exten 501.. and then phone2 dialed exten 501 - it should prevent it.
16:59.53p3nguinGROUP and GROUP_COUNT
16:59.58Ice_StrikeGreat! :)
17:03.24[TK]D-Fender"core show application chanisavail"
17:03.35[TK]D-Fender"core show function DEVICE_STATE"
17:03.37[TK]D-Fender^^^^
17:03.42*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
17:04.10p3nguinice_strike: Something like this:  http://pastebin.com/2wNu8gZk
17:04.43Ice_StrikeI think that is right
17:04.47p3nguinHe asked for limiting calls to an extension.  He said nothing about devices or channels used by devices.
17:07.03Ice_Strikeexten => _5XX,1,NoOp();
17:07.07Ice_StrikeThat should work I think
17:08.02krotos\quit
17:08.23talntidare there any expert level asterisk freelancers who I can hire? I already have a full setup, but I'm sure it could use some tuning, and I have some feature "requests" to be done.. ?
17:08.39*** join/#asterisk Beave (~champ@bundy.vistech.net)
17:08.53talntidjust wondering if there is a well known one, or a site you guys recommend to find someone
17:09.21mjordantalntid: try the asterisk-biz mailing list
17:09.31talntidroger that, thanks :)
17:09.40p3nguinIf you don't need it done right away, I'd be interested.  I have a few other things I'm dealing with right now, so it might be a little while before I could start on a new project.
17:10.01talntidI'm in no hurry
17:10.04BeaveI have a strange problem.  I can load my wctdm module (load fine),  see in it lspci (no problems/no IRQ conflicts) but dahdi_hardware doesnt see it.
17:10.18QwellBeave: What card?
17:10.56BeaveQwell: Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules)
17:11.25talntidp3nguin, pm?
17:11.39p3nguinYeah, go for it.
17:12.02BeaveIt might be a older card,  cant remember when we got it.  Pulled from a working production box.
17:12.27cuscoQwell: my problem with gsm is using sox, sox will eat cpu and asterisk will degrade audio
17:12.35cusconice won't cut it
17:12.36cusco:|
17:13.49Qwellcusco: save it as the native format then
17:16.27*** join/#asterisk Psychoroxx (~psycho@p57B09430.dip0.t-ipconnect.de)
17:18.05p3nguinkobaz: Even using the workaround with ChanSpy and Wait in a loop, it still blocked and held that channel permanently.
17:19.34BeaveQwell: whats strange is everything else works (dahdi_tool, dahdi_cfg, etc).. should i just not worry about the dahdi_hardware?
17:20.33*** join/#asterisk m_tadeu (~quassel@89.180.88.192)
17:21.46sruffellBeave: What is the output of "lspci -d e169:* "  and what version of dahdi-tools are you using?
17:21.57sruffellI mean "lspci -d e159:*"
17:22.03*** join/#asterisk wonderworld (~ww@dsdf-4db504c8.pool.mediaWays.net)
17:22.17Beavenothing for the lspci -d command.. checking the tool version
17:22.45Beavedahdi-tools-2.5.0
17:22.53sruffellnothing for lspci -d d159:* ?
17:23.02sruffell<sigh>
17:23.08sruffellI mean lspci -d e159:*
17:23.25BeaveAsterisk-JAX dahdi-tools # lspci -d e159:*
17:23.25Beave01:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
17:23.47sruffellWhat about "lspci -d e159:* -vv -nn"
17:24.16Beavea bunch,  anything in particular should i paste?
17:24.20Beavedont want to flood.
17:24.41Beavei can pastebin it.
17:24.42Beavehang
17:24.43sruffellThe subsystem line
17:24.44*** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net)
17:24.51BeaveSubsystem: ATCOM Technology co., LTD. AX400P 4-port analog card [b1d9:0003]
17:25.29leifmadsenlooks like a clone
17:25.45Beavepurchased from digium..  however,  i can call and verify.
17:25.49sruffellnah..that's ok.
17:25.52QwellIt most definitely was not.
17:25.59sruffellThose cards use the TigerJet info
17:26.07leifmadsensruffell: I'm looking at the AX400P
17:26.23sruffellso…dahdi_hardware should be picking that up.  It's in the list.
17:28.36Beavejust talked to him,  "The only place i've ever ordered cards is from Digium".
17:29.00Beavebut.. that's him...
17:29.10sruffellhttps://github.com/sruffell/dahdi-tools/blob/2.5/xpp/perl_modules/Dahdi/Hardware/PCI.pm#L72 <— that's where that module should be recognized on the 2.5 branch in dahdi-linux.
17:30.56Beaveyep.
17:31.00Beavehmmm
17:31.27Beavefrom dmesg: Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules)
17:31.35BeaveDESCRIPTION => 'Wildcard TDM400P REV I'
17:31.45BeaveMan, i'd say your right.
17:31.51sruffellPerhaps check to make sure that matches what's in your /usr/lib/perlxxxx   folder for the version of perl that is on the command line.
17:33.17BeaveHardware/PCI.pm:'e159:0001/b1d9'=> { DRIVER => 'wctdm', DESCRIPTION => 'Wildcard TDM400P REV I' },
17:33.20BeaveHardware/PCI.pm:'e159:0001/b118'=> { DRIVER => 'wctdm', DESCRIPTION => 'Wildcard TDM400P REV I' },
17:33.23BeaveHardware/PCI.pm:'e159:0001/b119'=> { DRIVER => 'wctdm', DESCRIPTION => 'Wildcard TDM400P REV I' },
17:33.27leifmadsen~pb
17:33.27infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
17:33.33leifmadsenBeave: ^^
17:33.40Beavewill do ,  sorry
17:34.52Beavecard [b1d9:0003] <- yep.
17:35.02*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
17:35.48atanhttp://asterisk.pastey.net/ <- no love
17:36.33Qwellinfobot: no, pastebin is A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
17:36.33infobotQwell: okay
17:37.01Qwell~pb
17:37.01infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
17:37.04Qwellstupid bot
17:37.12Qwellinfobot: no, pb is <reply> see pastebin
17:37.12infobotQwell: okay
17:40.21p3nguinhmm
17:40.26p3nguin~pastebin
17:40.26infobotit has been said that pastebin is A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
17:40.53p3nguinForgot the <reply> part for setting ~pastebin.
17:41.06Qwellinfobot: no, pastebin is <reply> A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
17:41.06infobotokay, Qwell
17:41.09Qwellmeh
17:41.13p3nguin~pb
17:41.13infobotA "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.
17:42.11*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
17:48.54*** join/#asterisk zerohalo (~zerohalo@67.103.230.22)
18:10.13*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
18:10.24*** join/#asterisk blee (~blee__@70-119-212-189.res.bhn.net)
18:12.19bleeHey all, im running a really old version of asterisk (1.4.21), and quite randomly today i have been having issues with asterisk getting really CPU heavy when its using odbc to access the DB.  This is affecting voice quality while its happening
18:12.35bleeive tried both a local DB and remote DB, same result
18:13.52Qwell~upgradeasterisk
18:13.52infobotBefore requesting assistance, you should be running the latest version of a supported release branch.  See the channel topic for the latest versions available in currently supported branches.
18:15.35*** join/#asterisk fhmiv (~fhmiv@c-68-47-6-72.hsd1.ga.comcast.net)
18:16.23bleedoh, i figured you would say that
18:18.19ChkDigitI am terribly vexed.
18:18.54ChkDigitI have created a macro that forwards an extension to a cell...
18:19.20ChkDigitI want asterisk voicemail to pickup, so setting a Dial timeout is needed...
18:19.52ChkDigitMy macro uses Dial(Local/5551212@forwardtrunk)
18:20.00ChkDigiterr...
18:20.06ChkDigitMy macro uses Dial(Local/5551212@forwardtrunk, 16)
18:20.24*** join/#asterisk jetlag (~jetlag@pool-71-168-245-146.cmdnnj.east.verizon.net)
18:20.35ChkDigitcalling forwardtrunk causes a real Dial of
18:20.51ChkDigitDial(SIP/outline/5551212,16)
18:21.00ChkDigitBut, it rings indefinitely.
18:21.13ChkDigitAny ideas?
18:24.08[TK]D-FenderChkDigit, Idea : show us your actual diapln and the actual call
18:24.16[TK]D-Fender~pb
18:24.17infobotA "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.
18:24.18[TK]D-Fender^^^
18:24.40ChkDigitDarned pastebin...  okay...
18:27.56*** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net)
18:29.05ChkDigitnotices something in pastebin he didn't see before...
18:29.34[TK]D-Fendernaturally...
18:30.16BeaveQwell: Okay..  Figured it out.  i ran an strace on the dahdi_hardware routine,  it was having problems accessing some /proc /sys info.. turns out,  udev wasnt started..
18:30.34BeaveAsterisk-JAX ~ # dahdi_hardware
18:30.34Beavepci:0000:01:02.0     wctdm+       e159:0001 Wildcard TDM400P REV I
18:30.35Beave:)
18:31.13Beavenot the interesting part is,  the channels answer/etc,  but then network card stops functioning when the wctdm is active :(  no sharing of IRQ's,  so looking into that now.
18:31.17Beaves/not/now
18:31.35WIMPyBeave: I also fell for that one once.
18:32.19BeaveWIMPy: the dahdi_hardware thing?  Yeah..  new something was up because everything else was "working".
18:32.55*** join/#asterisk netman (netman@228.170.76.188.dynamic.jazztel.es)
18:32.57WIMPyThe need for udev.
18:33.14WIMPydoesn't like to have udev on servers.
18:34.16*** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com)
18:34.38*** join/#asterisk netman (netman@228.170.76.188.dynamic.jazztel.es)
18:38.29*** join/#asterisk FinboySlick (~shark@74.117.40.10)
18:39.22*** join/#asterisk netman (netman@228.170.76.188.dynamic.jazztel.es)
18:42.09Ice_StrikeWhy I get this error
18:42.09Ice_Strike[May  7 19:40:58] ERROR[7689]: pbx.c:3401 ast_func_write: Function Group not registered
18:42.18Ice_Strikeexten => _5XX,n,Set(Group(${EXTEN})=limit)
18:42.21[TK]D-FenderGROUP <-
18:42.46[TK]D-Fender<p3nguin> GROUP and GROUP_COUNT
18:42.53*** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net)
18:44.03*** join/#asterisk Nugget (nugget@carrera.macnugget.org)
18:46.10FinboySlickAnyone here deploying SPA2102 in an auto-provisioning setup (tftp or http)?  This isn't really on-topic but but I figured I might be forgiven on a quiet day.
18:47.16FinboySlickI have a situation where they seem to ignore option 66 (they request it) completely when taken out of the box.
18:49.42*** join/#asterisk scgm11 (~Sebastian@r186-50-105-191.dialup.adsl.anteldata.net.uy)
18:50.02leifmadsenIce_Strike: you're not using the GROUP() function correctly
18:50.05scgm11hi, I have a question regarding to 1.8 branch and busydetect
18:50.13leifmadsenyou're trying to use it like an application, and it's a function
18:50.21scgm11is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8??
18:50.32scgm11I have to detect 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)
18:50.33leifmadsenIce_Strike: Set(GROUP(category)=value)
18:50.54leifmadsenscgm11: sounds like you might need to modify indications.conf ?
18:51.03*** join/#asterisk mchou (~quassel@unaffiliated/mchou)
18:51.04leifmadsenthat's where the indications are defined
18:52.07scgm11leif
18:52.14scgm11las time I put that on the bug tracker
18:52.21scgm11cmaj told me: "indications.conf is for playing back tones, not listening for them."
18:52.49scgm11I think is bussypattern where it should be but, I don't know if this kind of pattern
18:53.04scgm11detectable
18:59.14scgm11any ideas?
18:59.39*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
19:07.21scgm11I tried anyway putting on indication.conf with no result
19:11.33*** join/#asterisk [netman] (~netman@228.170.76.188.dynamic.jazztel.es)
19:12.02*** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net)
19:15.16*** join/#asterisk netman (netman@228.170.76.188.dynamic.jazztel.es)
19:15.54*** join/#asterisk netman (netman@228.170.76.188.dynamic.jazztel.es)
19:19.40*** join/#asterisk pa (~pa@unaffiliated/pa)
19:24.11*** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net)
19:24.26Ice_StrikeThanks!
19:41.00HiveWhat does a '/' mean in a pattern matching segment? ex: https://wiki.asterisk.org/wiki/display/AST/SMS+and+extensions.conf
19:42.01kaldemarafter / you can enter a caller id number, and the pattern will only match for calls from that caller.
19:42.18Hiveoh wow that is nifty
19:46.08*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
19:47.47*** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net)
19:50.54*** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com)
19:51.50michael-iHi all. I was wondering if Asterisk 10.x can optimize the codecs being used in a call through SDP renegotiation? Internal is all 722, upstream is ulaw. I want to renegotiate the internal leg if bridged upstream so both legs use ulaw, eliminating transcoding.
19:58.50*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
20:01.35leifmadsenmichael-i: don't believe that is currently possible
20:03.45michael-ileifmadsen: figured, not an everyday need.. but perhaps more and more useful as g722 catches on
20:04.40WIMPywould like that as well.
20:07.58verywisemanwhen i execute `/etc/init.d/dahdi start` this error apeear http://fpaste.org/hV4A/, why?
20:09.22justdaveI keep getting this in my logs:
20:09.22justdave[May  7 13:08:39] ERROR[10843]: utils.c:1211 ast_careful_fwrite: fwrite() returned error: Broken pipe
20:09.41justdavebumping up the debug level doesn't seem to give any more clue than that though
20:09.47justdaveany way to find out what it's having trouble writing?
20:10.01leifmadsenjustdave: you can likely ignore that since it's usually just asterisk sending or receiving audio frames for a channel that went away
20:10.17leifmadsenI get that if I hangup a call sometimes and always just ignore it
20:10.51justdavehuh.  okay.  I'm only noticing it on one server (out of 6) so I was curious if I had something broken on that one
20:11.21justdaveit does look like it's right after hangups now that you mention it
20:17.07pabelangerjustdave: or you need to fix your AMI scripts
20:18.40*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
20:19.21*** part/#asterisk scgm11 (~Sebastian@r186-50-105-191.dialup.adsl.anteldata.net.uy)
20:29.51*** join/#asterisk f0ner00t (f0ner00t@79.133.201.88)
20:30.39f0ner00tHello.. I have a PRI with oneway transmission on outbound calls. It seems to change who is experiencing who is sending the progress message. Anybody hear of this issue?
20:31.56WIMPyNo, but there were errors parsing your text.
20:32.43f0ner00tWhat do you mean Whimpy?
20:33.31f0ner00tOh sorry The one way transmission seems to change depending who sends the Progress message on the PRI
20:35.37WIMPyAre the calls ever answered?
20:38.43f0ner00tyes.. Depending on who sends the progress message the other person can hear the outbound call
20:39.30WIMPyIf it's an outbound call it seems pretty obvious who could send a progress message.
20:44.53*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
20:46.22*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
20:47.24*** join/#asterisk din3sh (2988f577@gateway/web/freenode/ip.41.136.245.119)
20:49.30Hivehas anyone messed around with receiving SMS messages with MessageSend() in asterisk 1.10.x?
20:49.47Hiveerr i guess messagesend is for sending messages so just remove that part from the sentence lol
20:52.13*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
20:53.24*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
20:54.50*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
20:54.59*** join/#asterisk wonderworld (~ww@dsdf-4db504c8.pool.mediaWays.net)
21:01.21*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
21:08.10*** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net)
21:14.56*** join/#asterisk troyt (~troyt@2001:1938:240:3000::3)
21:20.07f0ner00tWimpy: I can send a progress message the customer does by default. If I turn my progress message on I can hear the person they cannot hear me. If i turn my progress message off and the progress message comes from the pbx than they customer can hear em but i can't hear them
21:20.57WIMPyYou surely dont send a progress on an outbound call.
21:21.06Ice_StrikeHmm
21:21.10Ice_StrikeWhy this dont work
21:21.11Ice_Strikeexten => _5XX,n,Set(GROUP(${EXTEN})=limit)
21:21.11Ice_Strikeexten => _5XX,n,GotoIf($[ ${GROUP_COUNT(limit)} > 1 ]?overlimit);
21:21.15Ice_StrikeIt work to limit to 1
21:21.17WIMPyAnd the situation still sounds like the callmay not be answered.
21:21.22Ice_Strikeexample, 501 in call
21:21.29Ice_Strikewhen I do 502 - it wont let me
21:21.38Ice_Strikebecause 501 already in call?
21:25.23*** join/#asterisk Skaag (~Skaag@67.133.201.134)
21:25.44Skaagwhere do I setup a sip trunk if I want my soft phones to be able to dial through that trunk?
21:25.48Skaag(in which file)
21:26.07*** join/#asterisk gusto (~gusto@80.187.242.131)
21:27.14WIMPyThere is no such thing as a sip trunk, but sip accounts are configured in sip.conf and your dialplan in extensions.conf.
21:27.22gustohey
21:27.39talntidlooking to make a .call file out, but basically I want to send it to... agi(blah.agi)
21:27.45gustomy slovak pbx is online again, i am in better mood
21:33.38Ice_StrikeAhh got it fixed
21:33.38Ice_Strikeexten => _5XX,n,GotoIf($[ ${GROUP_COUNT(limit@${EXTEN})} > 1 ]?overlimit);
21:36.58f0ner00tWimpy we are seeing a progress message from the asterisk box... Even when the progress message is turned off
21:39.58f0ner00tWhimpy And you are correct the progress message is suppose to be comming from the central office / switch.
21:49.27f0ner00tWIMPy: ?
21:51.41*** part/#asterisk mjordan (~mjordan@nat/digium/x-npmjbbtjchjgdwiu)
22:04.28*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
22:05.25*** join/#asterisk blee (~blee__@adsl-065-083-173-085.sip.mco.bellsouth.net)
22:14.12WIMPySo you are saying you see Asterisk sending a progress on an outgoing call?
22:18.57*** join/#asterisk Bullmoose (~Bullmoose@71-37-169-46.bois.qwest.net)
22:19.35*** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134)
22:21.40*** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net)
22:31.05*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
22:34.00*** join/#asterisk mattjfox (~Crakore@ip72-219-183-40.oc.oc.cox.net)
22:34.17*** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-mwvmfmqjclwlohml)
22:39.14*** join/#asterisk bongfrog (~winston@c-50-134-135-61.hsd1.co.comcast.net)
22:41.26bongfroghas anyone else had an issue where they were not able to delete a dynamic member from a queue.?  CLI reports 'Unable to remove interface 'SIP/98324' from queue '9000': Not there'
22:43.28talntidhttp://pastebin.com/pTtXUnc3
22:43.38talntidI'v been playing with that, and can't get it to loop back correctly. advice?
22:44.07bongfrogwhen I do a queue show 9000 i see SIP/98324 followed by white space then (dynamic)etc.   I wonder if an agent typed a control character or the like.
22:46.20talntidnvm i got it
22:50.01f0ner00tWimpy... correct.
22:50.48*** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf)
22:50.52WIMPyInteresting.
22:51.12WIMPyDid I mention today that I don't trust libpri?
22:51.26WIMPyCan you paste such a call?
22:51.34michael-iIs there a combination of applications which will allow me to playback a file to a phone while simultaneously recording the input from that phone? Background() and Playback() both block, not allowing Record() to do its job
22:51.55talntidconference it
22:51.56talntid:)
22:54.13fpriorHi, I'm testing hot-desking ( from * CookBook); all CDR registry refers to device and not to extension. In future I need extract from CDR table extension's stats. Is possible resolve this situation ?
23:26.59mattjfoxcan I ask newb questions?
23:27.19WIMPy~ask
23:27.19infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
23:30.39*** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com)
23:32.03mattjfoxJust wondering if Digium is the only company that has released open source versions of asterisk
23:32.19mattjfoxI know there are different * wrappers
23:32.44mattjfoxI know it was originally developed by Mark Spencer owner of Digium
23:33.58mattjfoxthe book im reading Asterisk future of telephony is outdated
23:34.10WIMPy~book
23:34.10infobotAsterisk: 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
23:34.23fileanyone can take the source code, change it, and then release it... but to do it in totality is a *huge* time investment
23:34.37filethere have been projects that have done so, and they didn't last
23:35.24mattjfoxthanks for the book reference
23:35.56*** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer)
23:36.28JerJerhow does one turn on verbose compile mode?
23:36.42filemake NOISY_BUILD=yes
23:37.30JerJerok - any ideas why autoconf finds openssl-devel headers, but then linker doesn't add an -L for it ?
23:37.47JerJeror  i see -lssl and -lcrypto  but still get undef references
23:38.05JerJerundefined reference to `SSL_library_init'    and centos 6.2 has openssl-devel installed
23:38.21JerJer1.8.12.0.rc1  - trying newer ver now
23:39.27fileunknown, have not heard of anyone experiencing that issue
23:39.56JerJerundefined reference to `SSL_library_init'
23:40.01*** part/#asterisk mattjfox (~Crakore@ip72-219-183-40.oc.oc.cox.net)
23:40.09*** join/#asterisk mattjfox (~Crakore@ip72-219-183-40.oc.oc.cox.net)
23:40.33JerJeron the last step to create the asterisk executable
23:40.55michael-i32-bit vs 64-bit problems maybe?
23:41.29JerJeri see
23:41.35JerJeropenssl-1.0.0-20.el6_2.4.x86_64
23:41.36JerJeropenssl-1.0.0-20.el6_2.4.i686
23:41.36JerJeropenssl-devel-1.0.0-20.el6_2.4.x86_64
23:41.52JerJerthis is a x86_64
23:42.33JerJerIntel(R) Xeon(R) CPU           L5320  @ 1.86GHz
23:45.46gustoha?
23:45.49gustowhat happend?
23:46.20*** join/#asterisk kayfox (~kayfox@xheotris.zerda.net)
23:47.01gustowell, when someone is trying to compile something the -devel packages are important
23:50.00JerJergusto  openssl-devel is installed and detected by configure (since i can add res_crypto and others that depend on openssl)
23:58.26talntidhuh. the speech to text thing is pretty damn cool.

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.