IRC log for #asterisk on 20100117

00:00.04Baylinkcarrar: you do realise that that answer's worse than useless, right?  :-)
00:00.07*** join/#asterisk NateHB (n=noone@static-108-0-194-65.lsanca.dsl-w.verizon.net)
00:02.25NateHBAny aastra gurus around here, ive got a few 57is, I've got them all plugged into cisco router, all those ports are tagged native VLAN1000, some reason, 1 aastra phone with VLAN enabled, LANID set to 1000 works fine gets its ip from DHCP, the rest all cant get an IL address, i can plug my laptop into the same port and its gets a correct address, any ideas?
00:04.18Guggedont enable vlan on the phones
00:04.42NateHBthe only phone that actually works has VLAN enabled
00:04.48NateHBthats what gets me
00:04.52paulc[Jan 16 15:35:29] NOTICE[19663]: utils.c:1074 ast_wait_for_output: Timed out trying to write
00:04.53Guggeand your notebook doesnt ... but it works
00:05.07paulcuh, ignore that.. bloody right mouse click
00:06.37NateHBmaybe i need to restart the switch
00:06.53NateHBfucking POS cisco catylst
00:07.05NateHBi meant frelling
00:07.18NateHBsorry, i relize theres probably alot of kids around here
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00:08.48voipmonkits not the cat's fault
00:10.56NateHBthe cat?
00:11.21NateHBvoipmonk, you think it might be the cabling?
00:11.33paulc<mx:Button x="46" y="187" label="Refresh" click="phpPlayerData.send()"/>
00:11.49paulccurses the mouse (as opposed to the cat)
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00:11.56paulcsorry for the spam/copy'n'paste junk
00:13.53GuggeNateHB: i think i would look at the operator before looking at the cables or the cat ... when a notebook works
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00:37.34drmessanoSooooo
00:37.39paulcSo?
00:38.18drmessanoSo I try to download some firmware and provisioning apps from Cisco's site
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00:38.59paulcAnd how's that going for you?
00:39.54drmessanoand can't figure out what to do with the files I actually downloaded.. Each of the ZIP and EXE archives opens in 7-ZIP and contains .text .rdata and .data files
00:40.05drmessanoI'm guessing there's some backend issue on the site..
00:40.32paulc..or are those the firmware files you're meant to push to the router somehow?
00:40.38drmessanoNope
00:40.38paulc<-- not a Cisco expert, just guessing
00:41.11drmessanoThere should be .bin files in the ZIP archives and the EXE's should actually execute in windows, and they don't
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01:07.20hardwiremeh
01:08.32ChannelZfeh
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02:03.09p3nguindrmessano: What were you trying to download and install?
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02:06.40Diffen2evning. i have a really stupid question. when i call from my asterisk to my pstn gw i dont have any headers at all if i look at the call in wireshark. strange thing is that the call gets through so it works. have i missed out anything in the sip.conf?
02:13.16drmessanop3nguin: I figured it out
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02:19.15*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #aster
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02:24.24kam187hi
02:24.37kam187does asterisk support vad for detecting answer?
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03:04.32Kattyhi.
03:05.21ChannelZahoyhoy
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04:21.06Kattyi'm am irritated at our company.
04:21.20Kattythey are talking about relationships between white and black people to be 'crossbreeding'
04:21.25Kattyi find it harsh, and untasteful.
04:21.33Kattycranky.
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05:07.24kam187hmm how do i stop progress from a sip channel being passed back
05:07.31kam187i want to control it in the extensions.conf
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05:25.18Kattyeppigy: ping
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05:26.22manxpowerany music geeks around
05:26.27ChannelZIsn't crossbreeding between species?
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05:26.36linuxvieweranyone using  Cisco 7940 phone with Asterisk?
05:26.45ChannelZI am a geek that likes music
05:27.44manxpowerWhat does F with a subscript 4 mean?
05:28.54kam187hmm
05:35.49ChannelZin what context?
05:36.37manxpowerAs in the note F subscript 4 is 349.23 Hz
05:36.50manxpowerF in the 4th Octave?
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05:37.31ChannelZwell yeah then it'd be the F above 'middle C'
05:37.49ChannelZI don't think the superscript has particular significance
05:38.11ChannelZwhich is why I asked as maybe I had no idea what you were talking about
05:38.17kam187hmm how do i overide the progress between two calls?
05:38.23ChannelZhang up
05:38.51kam187:-/
05:38.55manxpowerA USA dialtone is more or less F above middle C (350) + A above middle C (440)
05:39.15kam187i dont want the answer on one side to go through to the other, i want to control it with answer()
05:39.34ChannelZhuh?
05:39.47manxpowerkam187: Until you answer the call, all progress is generated by the local SIP phone.
05:40.00kam187i have a h323 call come in, and in the extensions it dials a sip provider out
05:40.06kam187but that providers answers right away
05:40.36manxpowerkam187: not much you can do about that other than get the provider to not answer or get a different provider.
05:40.43kam187even while its ringing, so i want to do some kind of ring detection using   background detec or something
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05:40.56manxpowerkam187: I wish you the BEST of luck.
05:41.18kam187well i could modify sip.c to not pass the progress pretty easily but i'd rather not
05:43.11kam187ok so u dont know of any way to overide progress between the two legs?
05:43.14ChannelZthen how do you decide when the other end picks up?
05:44.03kam187by monitoring the audio either using dsp.c, speech() or backgrounddetect()
05:44.29ChannelZand what is the point of doing this?
05:44.57kam187call progress detection using vad
05:46.16ChannelZ...and what is the point of doing this?
05:46.32kam187fo billing purposes
05:46.38kam187most voip routers support this
05:46.39kam187google it
05:47.18kam187for example chan_dahdi has line detection built in
05:49.04manxpowerYes, and as it says in the config file it's experimental and buggy.
05:49.11ChannelZmanxpower: so are you wanting to play dialtones on the piano or.. :)
05:49.47manxpowerChannelZ: I'm trying to come up with a dialtone different from the default, but still is still pleasing to the hear.
05:50.08kam187manxpower: yeah thats ok i'm just saying i'm not asking for some wierd wako bizare thing that no one else on the earth would ever want
05:50.10manxpowerSo I tried CE, didn't like it, trying FC now.
05:50.59manxpowerkam187: I don't know what part of the world you are from but any carrier that answered all calls is a POS carrier.  there is NO reason for a carrier to do this.
05:51.20manxpowerAnd it, in fact, illegal in the USA, AFIK.
05:51.36ChannelZkam187: I didn't say you were, just wondering what the end game was
05:51.37kam187yeah true, but u have the same problem when u connect to analog phone lines
05:51.54manxpowerkam187: Your carrier is using analog lines?
05:51.56kam187which is probably why only the analog channel drivers support it
05:52.05kam187tbh i dont really care about this carrier
05:52.26kam187i'm more interested to get the functionality working on askterisk so i know its an option
05:52.31manxpowerWell if by "support" you mean "randomly disconnect calls when enabled" then yes it is "supported" on DAHDI/Zaptel
05:53.10manxpowerkam187: get out some books on DSP programming and let us know when you have a patch. 8-)  If it was easy to do someone would have done it already.
05:53.13kam187lol is it that bad?
05:53.36kam187i know how DSPs work :P
05:54.03manxpowerkam187: there is anecdotal evidence it likes to think higher pitched voices (women) or loud voices (men) are "BUSY" tones.
05:54.32kam187ok
05:54.37manxpowerIt's a great feature to enable when a husband and wife are arguing.
05:54.42kam187haha
05:54.46ChannelZHIII...YOOOUUU...HAAAAVVE...REEEAACHED...BOOOOB
05:55.35ChannelZ(speaks monotone)
06:00.54ChannelZHmm.  Not far from where I work, there is one single block that conatins 4 bail-bonds places, 2 criminal attorneys, a behavioral health center, and a post office.
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06:18.46Kattyhugs ChannelZ
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09:12.41tzafrirkam187, with SIP, as with any decent VoIP protocol (and with ISDN, in DAHDI) you get proper out-of-band signalling as to the state of the call
09:12.51tzafrirNo need to guess it by listening to the line
09:13.40tzafrirCan you be more specific about where calls don't properly disconnect with SIP?
09:17.05ChannelZI don't think his issue is disconnect; he wants to know when the remote end actually picks up the phone for accurate billing
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09:22.45tzafrirThis should also be reported. Your call only starts at that point
09:24.59ChannelZwell doesn't it depend on the channel (and perhaps the carrier if you're using an ITSP)?
09:25.45ChannelZLike on analog lines DAHDI reports the channel as "ANSWERED" as soon as it's done dialing
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09:39.51djabbourAre there any good metrics on how much transfer (in GB say), asterisk traffic uses on average per call volume?
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11:27.16benngardnow i am lost...
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11:27.37benngardfrom cli: channel originate SIP/0317998985 extension 0317998975@inputinterior.se <- works
11:28.01benngardfrom ami:
11:28.01benngardAction: Originate\r\nChannel: SIP/0317998985\r\nExtension: 0317998975\r\nContext: inputinterior.se\r\nPriority: 1\r\nTimeout: 15000\r\nCallerID: Magnus Benngard<0317998975\r\n\r\n>
11:28.18benngard== Starting SIP/0317998985-0000003b at inputinterior.se,,1 failed so falling back to exten 's'
11:28.59benngardi cant se the difference, and why ",,"
11:29.39benngardend the end is ofc: <0317998975>\r\n\r\n
11:35.11tzafrirbenngard, obviously the 'extension' did not get through
11:35.31benngardsure but i cant se what i am doing wrong
11:36.08benngardexten => 0317998975,1,ExecIf($[${DB_EXISTS(CFIM/0317998975)}]?Goto(${DB(CFIM/0317998975)},1) exists in context inputinterior.se
11:36.10tzafriruse 'Exten'
11:36.16tzafrirnot 'Extension'
11:36.21benngardlets try
11:38.24benngardthat worked, thx
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12:50.44k-manare the siemens cordless voip phones any good?
12:50.52k-manlike this one: http://www.ryda.com.au/Siemens-C470IP-Voip-Cordless-Phone-p/c470ip.htm
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13:46.21benngardi have some: http://gigaset.com/shc/0,1935,se_sv_0_152411_rArNrNrNrN,00.html really good sound and very easy to configure with *
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14:09.01benngardif i know an extension and wanna know the the name of the channel connected to that extension, how do i do that (in an easy way)?
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14:53.33sun28moin \o/
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16:13.32_abc_hello
16:14.04_abc_is there a way to search the ekiga.net phonebook ? they claim to support ldap but i can't seem to go through ?!
16:14.21_abc_of course i have an ekiga account
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16:17.39scuniziI just loaded asterisk and gastman.. running gastman asks for a host name.. the only one I have is a free ekiga account.. but with that it won't connect.. any hints to get past the main screen?
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16:20.46jayteeyou need to put in the hostname of your asterisk server for gastman and gastman is soooooo old and not supported
16:21.12scunizijaytee: ah.. is there an alternative?
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16:23.01jayteescunizi, what are you looking for? a graphical management interface for asterisk?
16:23.07scunizijaytee: yes.
16:23.45scunizijaytee: and a tutorial for cli management ... most of what I've found is pretty old
16:23.56jayteethen you probably want to look at installing AsteriskNOW 1.5 or whatever the latest version is and choosing the Freepbx interface.
16:24.51jayteebut anything with Freepbx on it is going to limit your flexibility in making Asterisk do things "outside the box"
16:25.36scunizijaytee: understandable..
16:26.19jayteefor CLI management just type help for a list of commands and then if you want more info on a particular CLI command type help "command name" without the quotes
16:26.39jayteefor example help sip will show you all the CLI commands related to sip
16:27.15jayteeand while the last version of the book only covers up to version 1.4 it's still mostly relevant
16:27.20jaytee~book
16:27.21infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
16:27.34jaytee~botsnack
16:27.34infobotjaytee: aw, gee
16:28.04scunizijaytee: that's great.. thanks for the tips and link.
16:28.11jayteeyw
16:30.36scunizijaytee: you running asterisk on WinXX?  just curious because of the reference on cli "help <command>"
16:32.29jayteescunizi, WinXX? I run Asterisk on RHEL 5.2 and CentOS 5.2. While there is a "port" of Asterisk for Windows, it's garbage and should be avoided at all costs.
16:35.20jayteescunizi, I can manage my asterisk servers from a Windows workstation by running Putty (Windows version) to connect via ssh to a secure terminal session on the asterisk server and then I run asterisk -vvvvvvr to connect a remote asterisk console. Asterisk runs as a service on all my asterisk servers.
16:36.06*** join/#asterisk klochan (n=klochan@95-27-105-153.broadband.corbina.ru)
16:38.04scunizijaytee: ok.. I was curious because "help <command>" on my linux box doesn't do anything..
16:39.02jayteescunizi, help is not a linux command, it's an asterisk command typed in the asterisk CLI, not the linux command prompt.
16:42.37scunizijaytee: ah! you can see how new I am to this.. how do I get to the asterisk cli?
16:43.47*** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net)
16:43.53jayteescunizi, type asterisk -vvvvvvr from a linux command prompt
16:44.36jayteeif you get an error message about asterisk.ctl not being found then asterisk IS NOT running as a service and you'd have to type asterisk -vvvvvvvc instead
16:45.19jayteescunizi, what linux distro are you running?
16:45.43*** join/#asterisk phdpeabody (n=peabody@77-21-14-170-dynip.superkabel.de)
16:45.48*** join/#asterisk cesar_CR (n=cesar@201.199.168.170)
16:46.09scunizijaytee: you might laugh but ubuntu 8.04 with a VM running 9.10 kubuntu.. I installed asterisk in the vm
16:47.25jayteescunizi, that shouldn't matter unless you need TDM hardware to connect to the PSTN in which case a VM will not be the right solution.
16:47.32phdpeabodyOK, I was reading the wiki but I've got a retarded question.. if I install an asterisk server, what do I need to do to make/receive calls?
16:47.51jayteephdpeabody, read the book
16:47.54jaytee~book
16:47.54infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
16:49.37scunizijaytee: this is mostly for experimentation.. I sell an IP-PBX that has a linux kernel but that's where it stops resembling anything else.. good system but got my curiosity up about asterisk.. I'd also like to implement it at home for my other business.  With a sip trunk and muliple channel capability..
16:49.59phdpeabodyok I'll read the book if you can yes/no two questions.. 1. Do I need to have some special provider access to make/receive international calls 2. Is there a low-cost module plug that I can connect a standard phone to VOIP?
16:51.34jayteephdpeabody, you'd need an account with an ITSP to provide a sip "trunk" to make and receive calls outside your own network. many offer international plans.
16:52.01jaytee~itsplist-eu
16:52.16jaytee~itsplist
16:52.42jayteehmmm, can't remember how to get the ITSP list for Europe
16:53.07jaytee~itsplist-germany
16:53.10phdpeabodyheh I don't suppose it's on the website?
16:53.29jayteevoip-info.org? might be, not sure
16:53.48phdpeabodyI was thinking asterisk.org but I'll check that one too
16:53.49jayteeor you can google ITSP and look for a provider in germany
16:54.12hardwiregives phdpeabody a treat
16:54.43jayteeand for an analog phone you'll need an ATA adapter like a Linksys SPA2102 or PAP2T-NA
16:56.48phdpeabodyI think the linksys is discontinued, but cool thanks
16:57.21jayteeno, the SPA2102 is still sold in the US, your market might be different
16:58.41Kobazhmm
16:58.52Kobazwhat would cause echo on an ip to ip call
16:59.29Kobazi have a sip phone, going to asterisk, and then an iax trunk to another asterisk box which sends the call to another sip phone
16:59.38jayteemic volume set too high? is one of the sets on speakerphone?
16:59.49*** join/#asterisk xuser (n=xuser@unaffiliated/xuser)
17:00.19Kobazheadset
17:01.14Kobazit's like asterisk is echoing back whatever audio it gets back to the caller
17:01.33scunizijaytee: trying to start with asterisk vvvvvvr or c results in errors for a non-existant asterisk.conf and extconfig.conf .. as well as permission issues.. should asterisk be started as root?
17:01.44*** part/#asterisk phdpeabody (n=peabody@77-21-14-170-dynip.superkabel.de)
17:02.42jayteescunizi, yes it should be started as root and there are configuration steps you have to do. now would be a good time to download and start reading the book.
17:03.27scunizijaytee: did.. do the download.. and yes.. I'll read.. I figured these basic questions would/should be answered in there..
17:04.16jayteethe book also contains a section for running asterisk as non-root but it's best to start with the simple things first. You'll also need to edit some conf files and there's a section of the book that gets you up and running fairly quickly.
17:04.46Kobazph
17:04.47Kobazoh
17:04.47Kobazhmm
17:04.50Kobazi think it's the soft phone
17:05.03scunizicool.. thanks.  I'll be afk for a while.. atleast until after the chargers game :)
17:05.13Kobazthe sip phone on the other end of the asterisk iax link is a sip soft phone (twinkle)
17:05.19Kobazit's creating echo
17:05.53*** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl)
17:05.56Kobazit's really annoying
17:06.09manxpowerKobaz: get better heassets
17:06.16manxpowerand headsets too
17:06.18Kobazit's the soft phone
17:06.30manxpowerthat is very unlikely
17:06.46Kobazwell, as unlikely as it is... that's what it is
17:06.54manxpowerSucks to be you.
17:06.56Kobazi don't have a mic on the pc with the soft phone
17:07.10Kobazif i mute all the channels in the mixer, the echo goes away
17:07.44jayteetry turning the mic volume in the mixer down or if this is ALSA try lowering your MUX setting
17:07.52Kobazmic volume is zero
17:07.54manxpowerKobaz: Does the issue go away when you switch to a different softphone?
17:08.10Kobazyeah, zoiper works no problem
17:08.31manxpowerOdd.
17:08.37p3nguinZoiper has echo cancellation... does the other phone you're using?
17:08.52*** join/#asterisk cesar_CR (i=cesar@201.201.41.242)
17:09.36Kobaznot sure
17:10.26Kobazit's not checked
17:10.29Kobazheh
17:10.31Kobazhmm
17:10.41Kobazi didn't think you needed echo cancellation with ip to ip calls
17:10.49*** join/#asterisk cesar_CR (n=cesar@201.199.168.170)
17:11.36Kobazi turned on echo cancellation in twinkle and now the echo is much less, still a very low volume echo
17:12.55*** join/#asterisk hluesea (n=hulusika@88.247.127.66)
17:14.00manxpowerecho cancelation on IP phones cancel the speaker/mic/speakerphone echo, not far end analog loop echo.
17:15.05manxpowerPolycom phones are the same way.  You can't cancel the far end analog loop echo in the IP phone because the latencies are far too high, it has to be done at the PSTN/VoIP conversion point.
17:15.35manxpowerBut with a headset you should not get much echo.
17:16.23Kobazyeah the polycom handles it fine
17:17.40*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
17:17.47*** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au)
17:18.04Kobazsince there's no moh since it never got the request
17:18.07Kobazer, wrong window
17:20.35*** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano)
17:28.25*** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk)
17:28.53*** join/#asterisk TimeRider (n=steve@78.32.26.1)
17:41.59*** join/#asterisk NateHB (n=noone@static-71-116-246-221.lsanca.dsl-w.verizon.net)
17:43.11NateHBhey guys, my asterisk server is multihomed, is there any way to set one program to use a certain outside gateway, and let asterisk use its interface?
17:43.51NateHBI've got openfire installed on the same box as asterisk, and I want to make sure openfire doesnt use the same bandwith as asterisk
17:46.48manxpowerNateHB: What interface is used is up to the OS.
17:49.09KobazNateHB: #networking
17:50.15*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano)
18:02.14*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
18:08.22*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
18:10.24*** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano)
18:14.40*** join/#asterisk Nephfl (n=none@rrcs-67-78-149-118.se.biz.rr.com)
18:16.34Nephflhello, I have always used polycom ip phones, i now want to connect some phones to a hosted asterisk pbx from behind a nat firewall and dynamic ip, so I imagine a sip phone with stun would do the job, what do yuo guys recommend?
18:18.31Nephflis this thing on?
18:19.27Nephflis anyone here?
18:26.30ChannelZnope
18:30.57*** join/#asterisk xuser (n=xuser@unaffiliated/xuser)
18:31.26p3nguinDoes a NoOp() stop a PlayTones(), or do I need to explicitly use StopPlayTones() or some other sound?
18:32.32p3nguinnephfl: Asterisk can handle NAT pretty well most of the time, so there shouldn't be any problem.  Just follow the guide for setting up NAT with SIP.
18:32.38p3nguin~sipnat
18:32.38infobot[~sipnat] Quick guide on configuring * + 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
18:36.04*** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:c5f:79bd:a077:acde)
18:36.05cuscohi
18:36.57cuscoIm trying to make a php script connect into asterisk's manager with fsockopen. Has anybody done that?
18:38.44cuscohttp://paste.debian.net/56949/
18:38.48ChannelZyes but not lately
18:39.13ChannelZyou need \r\n for one
18:39.18cuscowell im not being able to read anything
18:39.23cuscoouch
18:39.28cusco:/ ok
18:40.05cusconot enough
18:40.14cuscoI can echo the first gets($socket)
18:40.34cuscoif I try to echo it a second time (right down the bottom)
18:40.47cuscothe script hangs, probably because it has nothing to echo/read
18:41.03cusco(I added the \r)
18:41.22cuscoit only echos "Asterisk Call Manager/1.1"
18:41.48*** part/#asterisk dgilmore (n=dgilmore@fedora/dgilmore)
18:45.02ChannelZit might not be parsing the line breaks right, try using fread instead
18:45.05Nephflany suggestions for wholesale sip terminaton providers without volume requirements and no channel limits?
18:46.01cuscowith fread I need to specify a number of bytes. I can do so but it errors out anyway
18:46.18ChannelZre-paste what you have so far
18:46.31cuscoactually it hangs
18:46.32cuscook
18:47.20cuscohttp://paste.debian.net/56951/
18:48.14cuscoit echoes the first fgets() but hangs on the fread()
18:49.29cuscoow!!
18:49.42cuscoI replaced the '\r\n' with "\r\n"
18:49.45cuscoit works!
18:49.52cuscoResponse: Success
18:49.53cuscoMessage: Authentication accepted
18:50.00cuscodang
18:50.44ChannelZyah was just going to say that, \n and other escapes aren't evaluated in single quotes
18:51.19hlueseagreg
18:53.20cuscook thanks
19:02.51*** join/#asterisk Cain` (n=Geek@unaffiliated/cain)
19:05.34cuscoand don't need \r
19:05.35ChannelZNephfl: Flowroute?
19:11.01dlynesNephfl, if you're in Canada, Navigata is the best I've used
19:11.40*** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com)
19:11.42p3nguinVoIP.ms also has a presense in Canada, so they could be another option.
19:13.25*** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com)
19:15.10*** join/#asterisk Mango (n=Mango@d154-20-97-118.bchsia.telus.net)
19:15.41MangoHello.  Can someone please remind me what pbx_config.so does?
19:16.08KobazMango: loads your dialpla
19:16.09Kobazn
19:16.09benngardi wrote a small app that (by a simple html/php/ami) could transfer a call that i have answered, but i had i hard time to find the channel my extension was connected to, did  use "Action: CoreShowChannels" and looked for my extension and "BridgedChannel", that worked but was not fun at all to parse, is there a better way to do it?
19:16.22MangoKobaz: Thanks
19:16.43p3nguinIs _doVoiceMail something specific to OderlyQ, or to Asterisk in general?
19:17.41*** join/#asterisk Katty (n=User@adsl-70-253-161-187.dsl.stlsmo.swbell.net)
19:17.47Katty:>>>
19:17.54Katty4 new breeds of bird are in the yard this morning!
19:18.13Kobazit's raining!
19:18.15KobazRAIN!
19:18.17Kobazin january
19:18.18MangoMy wife would like to know which ones :)
19:18.19p3nguinYour milkshake brings all the birds to the yard.
19:18.21Kobazall my snow is going away
19:19.18benngardi i wish my snow could go away
19:19.31benngardjust a brown mess outside
19:19.53Kobazit's gotta snow some more before the end of winter
19:20.05Kattya white breasted nuthatch, a red bellied woodpecker, a tufted titmouse, and a american goldfinch
19:20.06Kobazi still have a season ski pass, and i haven't gone enough for it to pay for itself yet
19:20.22MangoKatty: Yay =)
19:20.28Katty:>
19:20.35Mangois married to an ornithologist
19:20.48dlynesbird lover
19:21.06benngardi am swedish, but i do (roughly) understand what kind of bird it is
19:21.16KattyMango: you should share my crittercam with her.
19:21.30dlynesmice have tits?
19:21.35Kobazof course
19:21.49Kattythey are mammals. they do have live young.
19:21.55dlyneswhy are their tits tufted, though?
19:22.00Kattywhat?
19:22.18dlynestufted mice tits
19:22.21dlyneserm
19:22.24dlynestufted titmouse
19:22.30dlynessnickers.
19:22.31Kattydlynes: http://upload.wikimedia.org/wikipedia/commons/d/d7/Tufted_titmouse_perching_2006-11-23.jpg
19:22.48Kobazthat's not a mouse
19:22.58dlynesno kidding
19:23.01dlynesripped off
19:23.01Kattyno, it's a titmouse.
19:23.14dlynesI thought i was going to get to see some titties
19:23.17MangoKobaz: Wow!  You must be an ornithologist too =)
19:23.52Kobaztotally
19:23.55MangoKatty: What country do you live in?
19:24.05dlynesnot to mention that breast on those nuts
19:24.12dlynesand that woody pecker
19:24.16KattyMango: Central USA
19:24.21dlynesKatty's got a dirty mind
19:24.32ChannelZBORING - if you type 'tits' into google it doesn't auto-suggest anything
19:24.46benngardcensur!
19:25.15dlynesbut i guess his belly's red from getting poked with a woody pecker all the time
19:25.33*** join/#asterisk titter` (n=titter@c-76-101-240-142.hsd1.fl.comcast.net)
19:25.46dlynesand titter`'s here just in time
19:25.51benngardChannelZ: type breast into google and se what u get as first ;)
19:26.20dlynesbenngard, scarless breast enlargements?
19:26.28dlyneserm implants, that is?
19:26.39titter`titter means to giggle, no clue whatcha talking about
19:27.02titter`http://pastebin.com/m25010808 -- this is flooding my console, and my pri lines are down
19:27.05dlynestitter`, oh...we were just talking about mice tits
19:27.07titter`anyone know what the deal is with this
19:27.21ChannelZWikipedia.
19:27.29ChannelZI'm sick of Wikipedia
19:27.32Kobaztitter`: have you tried reloading dahdi?
19:27.35dlynesChannelZ, why's that?
19:27.44Kobaztitter`: anything good in your system log?
19:27.46titter`I am going to, but I am curious what would cause this error
19:28.00dlynesa filled up scheduler
19:28.00ChannelZI don't know.
19:28.14dlynesdon't ask me what the scheduler is, though
19:28.14*** join/#asterisk DelphiWorld (n=Miranda@41.104.103.211)
19:28.16DelphiWorldhi
19:28.22titter`It just happened my boss called me freaking ... they fired the old asterisk guy after his less than stellar setup lead to a box getting rooted and a very expensive six digit hack with intl calls to cuba
19:28.23DelphiWorldany iristel customer here?
19:28.35ChannelZ(although 'breast' on wikipedia has some wikiporn)
19:28.37dlynestitter`, beauty
19:28.50titter`dahdi_tool isn't even installed -.-
19:28.54titter`brb.
19:29.05dlynesDelphiWorld, iristel?
19:29.27Kobazooo wikiporn
19:30.08DelphiWorlddlynes: http://www.iristel.ca ;)
19:30.36DelphiWorlddlynes: dyrect lines, give me one! ;)
19:30.51dlynesKobaz, it even has some pictures of that oddity that's become popular recently with the mcdonald's diet.....man titties
19:31.18dlynesDelphiWorld, wow...Canadian...never heard of them, though
19:31.21Kobazheh
19:32.00DelphiWorlddlynes: register with it;)
19:32.06dlynesah...based out of markham, on
19:32.28*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
19:32.40dlynesDelphiWorld, they any good, or do they suck?
19:32.51DelphiWorlddlynes: very good;)
19:32.53titter`installed dahdi-tools and dahdi_tool didn't install -.-
19:33.05dlynescoverage areas across canada, including bc
19:33.13Kobazhttp://www.urbandictionary.com/define.php?term=McGurgles
19:33.20dlynesI wonder why I never heard of them in BC, and I'm in the telecom industry there
19:33.43ChannelZDo you eventually get the McRuns?
19:34.01Kobazi would think so... after eating the McMountain of food
19:34.46MangoI've never heard of them either...in BC as well.
19:35.25*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:35.31*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
19:36.02MangoLooks expensive
19:36.08*** join/#asterisk joako (n=ston3d@opensuse/member/joak0)
19:36.28DelphiWorldMango: google for algeria connect
19:36.38*** join/#asterisk darkrift_lp (n=dark@65.92.170.72)
19:37.12DelphiWorldiristel is providing algeria did
19:37.30DelphiWorldbrb
19:37.46darkrift_lpsomeone have an idea if it's a good idea to have a TDM card and an asterisk installation through a virtual machine ? Is it reliable for the communication between the card and the asterisk serv, or there's actually some issue with that ?
19:38.20Kobazdarkrift_lp: your first challange is getting the tdm card actually working in a virtual machine
19:38.43Kobazdarkrift_lp: and your second challange is making sure your virtual machine gets very frequent and accurately timed execution timeslices
19:39.09darkrift_lpyeah, that'S the problem
19:39.27Kobazasterisk is very time-sensitive, for example there's certain things that must be sent out every 20ms
19:39.43Kobazjust buy a $299 quad core dell server
19:40.44darkrift_lpwell if I need to buy a new one I'll do that, I just didn't want to have another server in the server room because we wanted to have limited number of servers ... (not enough space and cost saving)
19:42.42*** join/#asterisk wepy (n=wepy@ip70-179-126-55.dc.dc.cox.net)
19:42.44wepyhello
19:43.19titter`hmm, dahdi_tool doesn't install when I compile dahdi-tools
19:43.45wepywhat is the service called where I get a phone number that's forwarded to my asterisk box?
19:43.47darkrift_lpthanks for the info Kobaz
19:43.55wepy(but I don't lease the phone line, i only handle IP)
19:44.01Kobazwelcome
19:44.18wepyheh
19:44.43Kobazhmm
19:45.44*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
19:47.18wepyIs there a kind of service where I get an incoming number and the ability to dial out to traditional phones?
19:47.28drmessano^wepy: That would be an ITSP
19:47.46Kobazsomething very strange is going on with originate and musiconhold over iax
19:47.50p3nguinwepy: a DID, maybe?
19:47.51drmessano^wepy: Theres no "forwarding" here, this is a native transport, no more or less than analog or PRI
19:48.00drmessano^~itsp
19:48.00infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
19:48.00wepyah thanks
19:48.27wepydo most ITSP providers handle hunt-modes and voicemail?
19:48.35Kobazno
19:48.38drmessano^no
19:48.41wepyor could i just have all calls routed to my asterisk system?
19:48.59wepyhm
19:49.09Kobazthat's generally how it is, yes
19:49.33wepyactually i want to have a single number, where, if I'm at work, it rings my work phone, but if i'm elsewhere, it dials my cell phone
19:49.46wepybut it seems like i'd need to least traditional phone service for that ;\
19:49.54drmessano^Why?
19:49.54wepys/least/lease
19:50.00drmessano^Didnt we just go over this
19:50.21darkrift_lp~itsplist-ca
19:50.21infobot[~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca
19:50.37wepyif I used ITSP, it's call -> ITSP -> my asterisk -> ITSP -> cell phone|work
19:50.39p3nguinIf you have Asterisk, you can use FollowMe or even basic dialplan with a sequence of Dial() commands.
19:50.57wepyhm
19:51.26wepywould latency be an issue if it all had to pass through the ITSP twice and my asterisk box once?
19:51.58drmessano^And if you have Ma Bell, it's call > price inflated pass through ma bells switch > Phone or cell
19:52.00p3nguinNot really.  Network latency is the only problem if you pass in and out of networks.
19:52.29drmessano^$20 a month vs $75 a month is a no brainer
19:52.31wepywas hoping to save $ ;)
19:52.37wepyhah
19:52.54drmessano^How are you gonna save money using Analog?
19:53.20wepycan't..
19:53.33*** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00)
19:53.53dennis00so, where's Obelix?
19:54.18wepyhow do you get service from bell?
19:54.21wepyor the like
19:54.53drmessano^Same way you have for the last 100 years.. call them
19:55.20drmessano^They probably have a hotline/priority queue just for new analog signups
19:55.28wepyso the phone company would perform ITSP services?
19:55.33drmessano^No
19:55.37wepyi thought they were all analog ;)
19:55.49wepyoh
19:56.43drmessano^The last thing you want is any sort of VoIP services from your traditional phone company
19:57.37drmessano^Their single biggest expense is maintaining all that horrid 100+ year old infrastructure, and you get to share in that with all the saps still using analog lins
19:57.39drmessano^lines*
19:58.03drmessano^Find a nice ITSP and forget about that hole in your wall
19:58.11wepyheh
19:58.56wepyhow do you know which ITSP's are good?
19:59.10wepyand do they usually let you test latency?
19:59.24drmessano^Google is your friend.. try <name of provider>+sucks or <name of provider>+problem
19:59.30wepyhaha
19:59.33drmessano^Test how?
19:59.42drmessano^Just ping their gateway
19:59.49wepynot sure, actually so much of it might have to do with my home ISP
20:00.14wepydrmessano^: so do you think I even need my own asterisk system?
20:00.19drmessano^if you see one that looks good, try it.. if it doesnt work, dump em
20:00.34drmessano^wepy: Only you can answer that
20:00.36Kobazhmmmmm
20:00.39Kobazpokes [TK]D-Fender
20:01.26dlynesdrmessano^, actually...not quite true
20:01.33dlynesdrmessano^, we save money by going analogue
20:01.58drmessano^How so?
20:02.12dlynesWe provide voip to the demarc room
20:02.23dlynesAnd then provide analog from there to the individual units
20:02.41drmessano^What does that have to do with getting an analog line from Ma bell?
20:02.46dlynesWay cheaper to use existing cat 3 infrastructure than to run all new cat 5 cabling
20:02.56wepyhm
20:02.58*** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com)
20:02.58dlynesyou said analog...you didn't say it had to be from ma bell
20:03.10drmessano^The entire conversation was about PROVIDERS
20:03.15drmessano^Now HOW YOU WIRE THE INSIDE
20:03.19drmessano^Not*
20:03.19dlyneswe're a provider :)
20:03.32dlynesand we provide analog to the customer
20:03.33wepyhas anyone heard of a good ITSP for the northeast USA?
20:03.41p3nguinsure
20:03.49dlynesthey could care less whether it's voip before it gets to them, and for that matter, they don't even know
20:04.03p3nguinVoIP.ms and Flowroute are both fine and they serve all of USA.
20:04.10wepythanks :D
20:04.12drmessano^dlynes: and obviously you dont use ANALOG from ma bell, so what I said _IS_ entirely true
20:04.29drmessano^dlynes: great, still out of context
20:04.32dlynesnah...but we've used analog from telus and bell telephone
20:04.47dlynesand they can still be cost effective...depends on your needs
20:04.57drmessano^great, and I used to have an orange bicycle
20:05.03dlyneslol
20:05.06dlynesbut seriously
20:05.16dlynesi wouldn't provide a business all voip lines
20:05.26drmessano^He's one person wanting a line
20:05.28dlynesI'd use voip for their overflow and long distance
20:05.54ChannelZ'cept flowroute seems not to be taking on new customers
20:05.55dlynesdoesn't take much to get you riled up, does it? =)
20:06.03drmessano^Maybe we should sell him on the merits of PRI too
20:06.20dlynesChannelZ, vitelity's decent
20:06.35dlynesChannelZ, I'm in Canada, and they're still not bad from here (they're in Denver)
20:06.49ChannelZhmm thanks, I'm in Denver :)
20:06.50wepycan you use ipsec with any of the ITSPs?
20:06.54ChannelZI knew of teliax
20:07.03drmessano^dlynes: Well, when some people decide to jump in so they can make some point entirely out of the context of the conversation so they can look smart, it's annoying.  Just sayin
20:07.12dlyneslol
20:07.17p3nguinIf you're going to use Vitelity, you might as well go VoIP.ms since they seem to be giving lower rates and they resell Vitelity.
20:07.41Kobazp3nguin: you can get lower rates if you go direct with vitelity
20:07.55drmessano^Flowroute isnt taking on new customers?
20:08.05dlynesp3nguin, i'm getting pretty damned cheap rates from vitelity...can't see how you can get much lower than I already have
20:08.08p3nguinWhat do you mean by "go direct" with them?  You pay, they give you service.
20:08.17Kobaz0.8 cents a minute incoming
20:08.21ChannelZdrmessano^: well not through the website anyway
20:08.27Kobaz1.5 cents with voip.ms, isn't it?
20:08.31drmessano^Oh nice
20:08.48drmessano^That doesnt sound good
20:08.49titter`[Jan 17 02:33:28] NOTICE[31198] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
20:08.56Kobazgo direct instead of through a reseller
20:09.04titter`After that, the error went crazy
20:09.11titter`every 3 seconds
20:09.15dlynestitter`, is your d channel down?
20:09.24titter`it's up now
20:09.25Kobaztitter`: you don't have a rhino card by any chance?
20:09.28titter`no
20:09.31Kobazthat's good
20:09.51titter`it's a TE122
20:10.04dlynestitter`, your d chan is up now, and it's still giving you those errors?
20:10.12titter`no error now
20:10.16titter`stoped asterisk
20:10.19titter`and restarted dahdi
20:10.21titter`and all is well
20:10.24dlynestitter`, ah
20:10.31titter`but want to know what caused it
20:10.34dlynestitter`, but given time, it happens again?
20:10.42titter`first time i've seen this lol
20:10.42dlynestitter`, usually caused by your d chan going down
20:11.04titter`it causes our pri to ring busy obviously, and my CIO flipped out
20:11.08dlynestitter`, but i've never had it spam me
20:11.24titter`Last time something like this happened, the server was hacked
20:11.30dlynestitter`, i've only ever had it happen sporadically, and only maximum three times in a row
20:11.36drmessano^Voip.ms' termination is cheaper than Vitelity, as is origination for this area
20:11.39titter`That is when they let go the old PBX guy
20:12.25titter`dlynes: thanks. still trying to figure out what the hell that scheduler error is
20:12.44titter`dylnes: it started with the d chan going down, then it spammed that scheduler error over and over
20:12.50dlynestitter`, anyways...d chan going down is usually a problem on the CO end
20:13.11dlynestitter`, but if you're getting a scheduler error at the same time, I would think it's probably an error on your end
20:13.18wepywhat's d chan?
20:13.19dlynestitter`, might even be a configuration error
20:13.42*** join/#asterisk DelphiWorld (n=Miranda@41.104.103.211)
20:13.43dlyneswepy, it's where your error correction, did information, ... all comes in on for PRIs and BRIs
20:13.47DelphiWorlddlynes: i am back
20:13.54dlynesDelphiWorld, congratulations
20:14.03DelphiWorlddlynes: for what? lol
20:14.08dlynesshurg
20:14.13dlynesJust being facetious :)
20:14.13titter`dlynes: i'll tear down dahdi, and reinstall it ... it looks like this was converted from zaptel config files
20:14.41DelphiWorlddlynes: do you will by around tomorow?
20:14.44titter`dlynes: any suggestions for this ... any locations that dahdi lives besides /etc/dahdi
20:14.52dlynesDelphiWorld, possibly
20:15.04Kobazman
20:15.08dlynestitter`, no idea...last time I worked with a PRI was when dahdi didn't exist
20:15.10Kobazthese asterisk bugs are getting weirder and weirder
20:15.24dlynesKobaz, aren't they already weird enough?
20:15.31drmessano^Kobaz: Aren't all bugs weird?
20:15.34Kobazno, this is really weird
20:15.40Kobazsome bugs are straightforward
20:15.53*** join/#asterisk klochan (n=klochan@78-106-111-127.broadband.corbina.ru)
20:16.13dlynesKobaz, it's complaining about some problem with windows on a linux box?
20:16.15*** join/#asterisk darkrift_lp (n=dark@65.92.170.72)
20:16.35Kobazheh
20:16.35Kobazwhat?
20:16.45wepywhy isn't more of the world using sip@blah for dialing?
20:16.46drmessano^I would think the general concept of "It shouldnt be doing that" would be classified as weird
20:17.00wepyinstead of analog phone nubmers..
20:17.03dlynesKobaz, you said the bugs were weird...so i guess they're not that weird, after all :)
20:17.06Kobazheh
20:17.11drmessano^wepy: Because the old get old, and the young get stronger.. it may take a week, and it may take longer
20:17.24drmessano^They got the guns, but we got the numbers
20:17.31dlyneswepy, because that would just be completely obtuse?
20:17.36wepyis there a way for vonage customers or skype users to dial sip@ "numbers"?
20:18.00DelphiWorldwepy: see skype for sip
20:18.04*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:18.18Kobazall i've been doing all week is submitting bug reports
20:18.19drmessano^Better yet, see Skype for Asterisk
20:18.41Kobazhttps://issues.asterisk.org/view.php?id=16627
20:19.53wepymaybe more companies using voip should announce their sip@ numbers
20:20.02carraror maybenot
20:20.06wepywhat do you call sip@ numbers? that saying feels old/dumb
20:20.21wepysip addresses?
20:21.26drmessano^SIP URI dialing
20:21.32*** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com)
20:23.18Kobazwhat i find, is everything is all fine and good if you're doing simple stuff in asterisk
20:23.31Kobazor stuff that you've done before, and it all works fine
20:23.34titter`well on a sidenote, I successfully got my asterisk to trunk to shoretel ... shoretel is pretty misinformed on asterisk, and the kb they provided was wrong
20:23.39Kobazas soon as i try something new... i run into a million bugs
20:23.52Kobazdrmessano^: does that happen to you?
20:24.06wepythanks..
20:24.08carrartitter, yeah it would be even better it Shortel phones would work with Asterisk
20:24.13carrarit=if
20:24.20Kobazevery new project I start... or even any time i add a new feature to an existing project... i wind up submitting like 5-10 asterisk bugs
20:24.32carrarShortel makes nice phones
20:24.35wepymaybe just 1 more question:  I noticed that some ITSPs have DIDs in some countries, and others do not..
20:25.01dlynesKobaz, try doing blf and/or sla...you'll hit a whole raft of bugs there, too
20:25.02wepyis there a way to acquire a DID from one ITSP, then have incoming calls on that DID forwarded to an ITSP/asterisk box near home?
20:25.11wepylike combining DIDs from multiple ITSPs
20:25.12titter`Shoretel is nice, hardware is nice, some of the features are nice ... but the cost ... bleh. License per extension, per voicemail box, per sip trunk, on and on
20:25.13dlynesKobaz, blf and sla were so broken for us, we had to abandon the project
20:25.21Kobazyeah
20:25.26Kobazi've had lots of problems with blf
20:25.44dlynesKobaz, but it wasn't just asterisk that was broken
20:25.44drmessano^Kobaz: I find a bug here and there, and sometimes something seldom used will get broken as hell, but overall I wouldnt say that anything complex is usually broken.. I will say that in the past there were far more issues with regression, but that's gotten much better over time
20:25.57dlynesKobaz, our phone's firmware was broken w.r.t. blf, too
20:25.59titter`carrar: do you have a asterisk -> shoretel setup right now?
20:26.05carrarno
20:26.10carrarYou doing it over SIP?
20:26.18dennis00Can I set a callerid with noop?
20:26.18dlynesKobaz, and the company moves like snails on bugs...and half the time don't admit it's an issue on their end
20:26.21titter`yep
20:26.25Kobazheh
20:26.40Kobazi need to hire a full time c coder whose job it is to fix asterisk bugs
20:26.42carrartitter, should be easy, I hear Shortel doesn't support authentication
20:26.55*** join/#asterisk Deiz (n=swh@unaffiliated/deiz)
20:26.58dlynesKobaz, digium's way better about fixing bugs than any hardphone manufacturers are about fixing firmware bugs
20:27.05titter`I setup a * server next to the shoretel switches so no nat is involed ... then use sip from asterisk to shoretel on the lan ... and iax to asterisk over the wan for the rest of my asterisk boxes
20:27.08*** join/#asterisk DelphiWorld (n=Miranda@41.104.101.60)
20:27.10Kobazpolycom does a good job fixing firmware bugs
20:27.18drmessano^Digium is better about fixing bugs than Microsoft
20:27.22Corydon76-digdlynes: Hush, we need all the help we can get
20:27.27Kobazwho wants to move to central pa, and fix asterisk bugs for me
20:27.28dlynesKobaz, aastra's good about fixing bugs that they admit exist
20:27.38dlynesKobaz, but a lot of the time, they don't admit the problems exist
20:27.41p3nguindennis00: No, but you can set Caller ID with CALLERID(num)=
20:27.47dlynesCorydon76-dig, ?
20:28.10Corydon76-digwould love it if 100 companies each hired coders to do nothing but work on and post patches for bugs in Asterisk
20:28.17KobazCorydon76-dig: i'm going to do that
20:28.24KobazCorydon76-dig: i'll be one of those companies
20:28.36dlynesCorydon76-dig, yeah..it would be nice...but at the same time, digium's still doing a good job with what they've got
20:28.39drmessano^Seriously, lets count the number of bugs in Windows that show up as updates on patch Tuesday that go back to >>> earliest supported version vs those of Asterisk..
20:28.43dennis00p3nguin: in sip.conf or extensions.conf? not like this? exten => _X.,1,NoOp,${CALLERIDNAME}
20:28.43Corydon76-digKobaz: sweet, the community appreciates your assistance
20:29.09KobazCorydon76-dig: i appreciate the good access to the developers that's available.. it helps things move along nicely
20:29.25Kobazbut watch out... i'm gonna have you guys on speed dial soon
20:29.29p3nguindennis00: In extensions.conf, Set(CALLERID(num)=12345)
20:29.33Corydon76-digFor 5 years, I worked for a reseller, fixing bugs and contributing features to Asterisk
20:29.44wepyi worked at a place that used shoretel
20:29.50dlynesCorydon76-dig, did you interpret what i said as digium bashing, or something?  It wasn't, in case you misunderstood
20:29.52p3nguindennis00: NoOp() does NOTHING.
20:30.01wepyit was alright, but we had lots of problems with cisco gear internally
20:30.03Kobazp3nguin: sure it doesn't
20:30.18Kobazp3nguin: NoOp prints stuff to the console output if you're in verbose mode
20:30.19dlynesp3nguin, well...not quite nothing
20:30.19wepymuch less reliable than pots
20:30.22titter`Corydon76-dig: have you ever seen an error like this before http://pastebin.com/m25010808
20:30.30dennis00p3nguin: thank you.
20:30.32dlynesp3nguin, it allows you to insert meaningful status and/or error messages in your logs
20:30.33Corydon76-digdlynes: I hear you, but I want to encourage, not discourage, outside companies to help work on Asterisk
20:30.42dlynesCorydon76-dig, ah
20:30.44p3nguin<PROTECTED>
20:30.45p3nguin[Synopsis]
20:30.45p3nguinDo Nothing
20:30.45p3nguin[Description]
20:30.45p3nguin<PROTECTED>
20:30.48dlynesCorydon76-dig, didn't realize i was discouraging them
20:30.50drmessano^I found a random Avaya phone at work once.  I sprayed the edge of the trash can with Lysol after disposing of it because the phone accidentally hit the rim on the way in.
20:30.52Kobazp3nguin: that's a lie
20:30.59p3nguinkobaz: Then file a bug on it.
20:31.10Kobazi think that's my job now
20:31.15Kobazall i do is file asterisk bugs
20:31.22Corydon76-digtitter`: I'm just here momentarily.  I have laundry to do today
20:31.28p3nguindlynes: You should be using Verbose() for that.
20:31.37Corydon76-digBig, massive piles of laundry
20:31.42*** part/#asterisk DelphiWorld (n=Miranda@41.104.101.60)
20:31.43titter`Corydon76-dig: no hurry, it happened after this [Jan 17 02:33:28] NOTICE[31198] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
20:31.57titter`It spammed over a million lines in my log file lol
20:32.26dlynesp3nguin, verbose is a dialplan command?
20:32.32p3nguincorrect
20:32.32Kobazapplication
20:32.54p3nguinVerbose(1,some stuff you want to see)
20:33.11tzafrirtitter`, some lower-level error. e.g. bad line
20:33.22dlynesp3nguin, i.e. Verbose(Call coming in for the DID ${ARG1} for the customer ${ARG2} from ${ARG3})?
20:33.22p3nguinor Verbose(3,some stuff you don't want to see until verbose level is 3)
20:33.27Kobazokay so, hmm
20:33.30Kobazwhat else am i going to do today
20:33.35dlynesp3nguin, oh...so even better than noop then
20:33.37p3nguindlynes: exactly
20:33.43dlynesp3nguin, I've been using noop for years :0
20:33.58dlynesI guess verbose is a somewhat new dialplan application?
20:34.00drmessano^NoOp is teh win
20:34.06tzafrirVerbose(3,...) is effectively NoOp, right ?
20:34.19*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
20:34.38Kobazdlynes: it's "new" as of 2004
20:34.39Kobazr3581 | citats | 2004-08-05 22:12:54 -0400 (Thu, 05 Aug 2004) | 2 lines
20:34.39KobazAdd app_verbose.c to cvs (bug 2212)
20:34.53dlynesah
20:35.03EmleyMoorIs there any way to get Zoiper to show whether a contact on my box is on the phone or not?
20:35.06dlynesso it probably didn't exist in asterisk when i first started using it, then
20:35.08wepywhat's a SIP trunk?
20:35.12EmleyMoor(IAX2 account)
20:35.13p3nguinwepy: nothing
20:35.16Kobaz~siptrunk
20:35.17infobotwell, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk
20:35.19dlynesbut it's still pretty old
20:35.20drmessano^NoOp is the ultimate man application in Asterisk.. "Sure, you can come over, but don't expect me to commit to anything or care"
20:35.33wepycool
20:35.39*** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk)
20:36.00tittertzafrir: thanks, could you elbaorate a little more ... I have seen a forum post that Digium has said this was a bug (Oct. 09), and it was fixed with a patch to libpri
20:36.01Kobazwepy: when people say 'sip trunking' they mean, handling a bunch of calls over sip
20:36.11drmessano^Woman: "I love you"  -- Man: "NoOp()"
20:36.28*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
20:36.32dennis00I see noop all the time.
20:36.36tittertzafrir: http://forums.digium.com/viewtopic.php?p=136339&sid=2c8ef807d7c32dc6a3ee5b21c9cad646#p138057
20:36.42tzafrirtitter, in most cases it's caused by a lower-level error (below libpri)
20:36.47dennis00But maybe they just meant noob.
20:36.49tzafrirwhat version of libpri do you have?
20:36.58drmessano^"Danny, did you break that??!!!??!!"   "Um, NoOp()"
20:37.17titter1.4.10.1
20:37.18p3nguindennis00: Just because you see it often does not make it the best application for the job.
20:37.36dennis00p3nguin: I have changed it, I want my new sim card to test it.
20:37.52tzafrirHmm... so we now also have Shawn Bright on -users. For a moment I thought it was Sean
20:38.03Kobazwiggity
20:38.17wepy~iax
20:38.18infobothmm... iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for  Inter-Asterisk Exchange
20:38.42dlynes~iax2
20:38.43infobotwell, iax2 is http://www.voip-info.org/wiki-IAX
20:38.43p3nguinI call it "eye axe"
20:38.47tzafrirtitter, that's a different error
20:38.59drmessano^p3nguin: It's EEKS
20:39.02*** join/#asterisk LemensTS (n=customgt@71.86.32.146)
20:39.04wepywhat's a good SIP-capable cordless phone for a home?
20:39.15wepymaybe witha few nice features, but most importantly, bug free
20:39.16EmleyMoorp3nguin: Isn't that a Dutch soccer team?
20:39.19EmleyMoor<g>
20:39.35LemensTS[Jan 17 14:32:16] WARNING[30560]: app_dial.c:1272 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown)    <--thats normal if a sip phone is offline and you try to call it right?
20:39.36dlyneswepy, try siemens...my coworker's pretty happy with his
20:39.38tittertzafrir: looks the same to me, minus the line number -- http://pastebin.com/m25010808 http://pastebin.com/d57e37e5b
20:39.48p3nguinwepy: You can use any cordless phone you want... with an ATA hooked to it.
20:39.54wepyok thanks
20:40.03Kattyyawns
20:40.05wepy~ata
20:40.06infobothmm... ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
20:40.09*** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net)
20:40.09p3nguinwepy: An ATA will turn any phone into a VoIP phone.
20:40.10wepyeffing acronyms :)
20:40.21*** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no)
20:40.26wepyi don't have a regular phone ;)
20:40.30wepyi'm all IP at home
20:40.40Kobazeyepeeee
20:40.48p3nguinSo you want a cordless with a SIP base.  Okay.
20:41.06wepysip base..
20:41.18*** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org)
20:41.40wepyi'd like to be able to dial sip URI's and also regular phone numbers, but all of it would go through the ITSP i choose
20:41.56p3nguinFor example, my cordless doesn't have an RJ-14 on it for a regular phone cord, it only has an RJ-45 for an Ethernet cable.
20:42.22EmleyMoorCan presence hints from Asterisk work over IAX2?
20:42.44wepyyea
20:42.44p3nguinyes
20:42.51wepyactually, wifi would be nice here..
20:43.00wepybut cordless -> rj45 would also work
20:43.12p3nguinNow you've gone from a cordless phone to a Wi-Fi phone.
20:43.33p3nguinMost people don't like Wi-Fi and SIP together.
20:43.37EmleyMoorI have just started using Zoiper Communicator and presence working would be a definite plus point
20:43.48tittertzafrir: I will try libpri 1.4.10.2 based on this bug report https://issues.asterisk.org/view.php?id=15892
20:43.56tittertzafrir: thanks
20:44.03EmleyMooruses his N95 as a WiFi SIP phone, when it works
20:44.23p3nguinemleymoor: Just put the hint in your dialplan.   exten => 2040,hint,IAX2/2040
20:44.29wepywifi's not stable enough?
20:44.57EmleyMoorp3nguin: Hmmm... already there but seems only to work over SIP
20:45.24p3nguinexten => 1234,hint,SIP/1234
20:46.09p3nguinEither Zoiper isn't working with presence correctly or you've specified the wrong Tech in the dialplan.
20:46.46EmleyMoorI have hints for the two "user" extensions that can detect when the user is using any of their phones... but Zoiper doesn't see it
20:47.20EmleyMoor(X-Lite and recent ekiga do)
20:48.09p3nguin"core show hints"
20:48.34drmessano^wepy: get a nice ATA like a Linksys PAP2 and a $20 DECT phone from walmart
20:48.38p3nguinThe second column shows the tech/channel
20:48.50drmessano^Win, win.. When the phone tech changes, toss it and get the next $20 phone
20:48.57EmleyMoorShown truncated due to the length
20:49.14p3nguinDoes it show IAX2/abc123?
20:49.20p3nguinor SIP/
20:49.32EmleyMoorYes and yes
20:49.44p3nguinhmm
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20:56.49wepyis you use a DECT phone and ATA, how can you make the phone dial sip URI's?
20:56.58wepymaybe asterisk can translate some special numbers into URIs?
20:57.05EmleyMoorcore show hints shows that the hint changes to InUse if my partner's Zap phone is in use - but no sign of any meaningful presence readout at all in Zoiper
20:57.14[TK]D-Fenderwepy: What kind of SIP URI's are you intending on dialing?
20:57.47p3nguinemleymoor: Sounds like a Zoiper problem, then.
20:58.20EmleyMoorIs there another IAX2 softphone for Linux that can do presence?
20:58.20wepy[TK]D-Fender: not sure, but I assume some day everyone will have one :)
20:58.43p3nguinSo you wanted Zap presense, not IAX2 presense.  I guess I misunderstood that.
20:59.09p3nguinAnd I can't spell presence.
20:59.59EmleyMoorp3nguin: One hint per user covers all phones each user has
21:00.24EmleyMoor(be they IAX2, SIP, Zap)
21:00.44drmessano^Wepy: You can set up extensions in Asterisk to dial a user@host.  When the day comes that we use URIs for phones it will likely be picking a name out of an address book, much like we do on smartphones, and clicking dial.
21:00.52p3nguinIf a "user" has all those techs on his phone, Asterisk should be able to show hints for each tech.
21:00.58Kattythis salad is so awesome.
21:01.18EmleyMoor... or "phones of each of tth
21:01.22Kattyit has balogna, almonds, quacamole, and snyder's honey mustard nibblers.
21:01.27EmleyMoor... or "phones of each of those techs"?
21:01.48p3nguinI don't like balogna, so I would have to toss it.
21:01.55p3nguintosses katty's salad
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21:02.08EmleyMoorIt's not so much what it shows for as what it shows to.
21:02.55[TK]D-Fenderwepy: Stop inventing problems that don't exist and move on to something productive
21:03.29Kobazhmm
21:03.43Kobazthis problem with chanspy not working when the channel isn't generating audio is a real pain
21:04.28p3nguinIf the channel isn't generating audio, what would you listen to during the spying?
21:04.32Kobazwhisper
21:04.38EmleyMoorSilence?
21:04.39Kobazwhisper is also affected
21:05.13Kobazit's not silence, it's a lack of audio frames
21:05.25Kobazyou can transmit silence
21:05.25Kattyp3nguin: well, i was going to put chicken on it
21:05.30Kobazand that's okay
21:05.38Kobazbut if there is no audio data, whisper just flat out breaks
21:05.39Kattyp3nguin: but i nukerwaved it for too long and it tasted funny
21:06.03EmleyMoorKobaz: What are you trying to achieve?
21:06.21KobazEmleyMoor: being able to whisper into a channel that may or may not be recieving audio
21:06.44EmleyMoorKobaz: And if it's not, how do you expect to be able to?
21:06.56Kobazby fixing the bug in chanspy
21:07.23Kobazi need to find where the reads are done, and make them non-blocking
21:07.51Kobazthat's the plan anyway, probably will be more involved than that
21:11.10dennis00p3nguin: I would like my callerid spoofing to work with budggetphone, but there probably is no way?
21:11.52p3nguindennis00: If you set the CALLERID(num) on the channel when you Dial() out... if it does not work, then there is probably no way.
21:12.29dennis00Too bad.
21:13.37p3nguinThey might be like voipbuster.
21:14.10p3nguinYou have to authenticate phone numbers before you can use them as outbound CID, and then you have to select the numbers from the list in their phone app.
21:17.19[TK]D-Fender[16:11]<p3nguin>dennis00: If you set the CALLERID(num) on the channel when you Dial() out... if it does not work, then there is probably no way.No, LOTS of other things can affect this
21:17.26*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:17.53p3nguinLet him know what else he can do... I'm sure he's interested if there is another way.
21:18.13[TK]D-Fenderdennis00: pastebin your sip peer, and debug from a call attempt including changing the CID
21:19.09dennis00ok
21:31.17dennis00http://pastebin.ca/1755165
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21:33.45wepyvoip.ms has a great web site heh
21:34.02wepythey explain the services very clearly.. love it
21:34.25p3nguinI use VoIP.ms for my toll-free DID and termination.
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21:36.27[TK]D-Fenderdennis00: in [31107142866] add "sendrpid=yes" , "trustrpid=yes" and retry
21:40.29dennis00[TK]D-Fender: same thing.
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21:49.09dennis00Can anybody a provider that does not block thijs?
21:52.58Kobazhmm
21:53.19Kobazsplicing in audio with chanspy is not always nicely handled
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21:59.42dennis00I am looking for a voip provider that does not require mobile number and accepts callerid spoofing.
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22:03.24Kobazsomeone
22:03.26Kobazer
22:03.35Kobazsomeone broke originate in 1.6.0 svn :(
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22:15.31Kobazhttps://issues.asterisk.org/view.php?id=16628
22:15.37Matt_AHello, is it true that Freeswitch has a better architecture than Asterisk?
22:16.29Kobazhaha
22:17.07Kobazit depends on the definition of better
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22:26.03titterShould I recompile Asterisk after upgrading libpri
22:26.08Kobazno
22:26.37Kobazit's a dynamic library, and the ABI is the same
22:27.08titterKobaz: thanks
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22:37.32Kobazman
22:37.38Kobazall this bug reporting is making me hungry
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22:53.34dennis00Are lots of people using Voipbuster with Asterisk?
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23:38.56wepyhow much jitter is OK on a link that's about 200ms round trip?
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