IRC log for #asterisk on 20100607

00:15.04*** join/#asterisk Sidnicious (~Sidney@pdpc/supporter/professional/sidney)
00:15.54SidniciousWhat kind of glue does * provide for an external application to trigger some action (e.g. dialing an extension)?
00:16.26SidniciousThe goal is actually to be able to hook up text-to-speech, so that an external application will be able to trigger a call to an extension or conference and speak some text sent by that application.
00:18.23russellbLook at the AMI
00:18.34russellbit's a socket interface for 3rd party control of asterisk
00:18.48SidniciousAwesome, thanks. I'll take a look at it.
00:19.06russellbwithin the AMI, the "originate" action is what allows you to tell asterisk to make a call, and what to connect to it
00:26.09*** join/#asterisk Benwa (~Benwa@dyn.83-228-160-169.dsl.vtx.ch)
00:26.22p3nguinSo... he says he set the PAP2T to enable DNS SRV and it magically works with VoIP.ms now.
00:31.13*** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au)
00:32.36*** join/#asterisk ltd_wk (~z@sixified.transact.net.au)
00:36.19ChannelZwacky
00:37.17p3nguinI doubt that is what made it start working because VoIP.ms does not have SRV records.
00:38.02drmessanolol
00:38.02p3nguinHe said he is also using ekiga, which he noticed is using STUN.  He's thinking that could have some bearing on it.  I just don't know how to explain it.
00:38.35drmessanoSounds like he's a tweaker and deserves death
00:39.14drmessanos/deserves death/changed some other things/
00:39.22drmessanoSorry, I do that all the time
00:39.50p3nguineasy mistake
00:39.58p3nguinThe keys are like right next to each other.
00:39.59drmessanoThank you for being so understanding
00:40.15drmessanoha
00:41.48p3nguinHow could using ekiga with a configured STUN server magically make the ATA work?  Is that even possible?
00:42.13p3nguinI have to assume they were using the same credentials registering to the same server.
00:42.22drmessanoWell
00:42.49drmessanoIf one reg replaced the other, and the host info was cached.  But that's going out on a long limb
00:43.32p3nguinI've had him kill off ekiga and power cycle the ATA.  It's still registering correctly.
00:43.53drmessanoHas he had both running?
00:44.14p3nguinat once?  Only in the very beginning when the ATA magically registered, then he shut off ekiga.
00:45.08drmessanoSounds fishy
00:45.25drmessanoThis is why I don't trust friends with ATAs.  XML FTW
00:45.38*** join/#asterisk m_tadeu (~quassel@173.191.19.95.dynamic.jazztel.es)
00:45.48p3nguinDuring testing earlier, I even had him disable registrations and I configured the peer on my system to be static, with nat/qualify.  When I would send the OPTIONS for the qualifies, the ATA would not respond.
00:46.07*** part/#asterisk linkd (~switch@unaffiliated/linkd)
00:49.34*** part/#asterisk Jaxyeh (~jax@c-69-250-52-161.hsd1.md.comcast.net)
00:52.25Naikrovekp3nguin: how's the chan_sccp thing going?  still werkin'?
00:52.31p3nguinyep
00:52.34p3nguinI use it every day.
00:52.37Naikroveknice
00:52.48p3nguinThere's only one bug that I have encountered.
00:52.59Naikrovekdo explain
00:53.17p3nguinSince there is no sccp reload, you have to module unload chan_sccp, then module load chan_sccp...
00:53.29Naikrovekoh wow
00:53.37p3nguinOh, that's not he bug.
00:53.38Naikrovekseems like that would be an easy thing to add
00:53.40p3nguinEvery once in a while, unloading the module crashes asterisk.
00:54.03p3nguinIt doesn't just kick me off the CLI, I have to actually go start asterisk again.
00:54.15Naikrovektime for some maintenance windows
00:55.15Naikrovekhmm
00:55.46p3nguinI don't often need to reload sccp anyway.  It's just that there is no other way to insert changes from sccp.conf.
00:56.19Naikrovekthat _should_ be easy enough to add
00:56.28Naikrovekis a Java developer though, not a C developer
00:59.16*** join/#asterisk nighty^ (~nighty@210.188.173.245)
00:59.40*** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk)
00:59.54*** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net)
01:01.13m_tadeuis it possible to use a "secret"(the one in sip.conf) in a database? because I want agents to be able to replace it easily
01:03.13p3nguinI think that's what realtime is for.
01:04.34m_tadeuufff....more stuff to read :) thanx
01:06.39*** join/#asterisk m_tadeu (~quassel@173.191.19.95.dynamic.jazztel.es)
01:25.07*** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru)
01:33.15*** join/#asterisk Sidnicious|Home (~Sidney@pdpc/supporter/professional/sidney)
01:33.30*** part/#asterisk Sidnicious|Home (~Sidney@pdpc/supporter/professional/sidney)
01:49.36*** join/#asterisk coppice (~chatzilla@m121-202-89-212.smartone-vodafone.com)
01:55.09*** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)
01:56.39*** join/#asterisk m_tadeu (~quassel@173.191.19.95.dynamic.jazztel.es)
02:14.41*** join/#asterisk coppice (~chatzilla@m121-202-75-150.smartone-vodafone.com)
02:22.17booduI'm always working to install cisco hardphone, and I have a problem (again) to put a choice of ring. The message "Ring List Unavailable".
02:22.28booduI'm sure of my Ringlist.xml
02:23.37booduI made ln -s of this file : RingList.xml, DistinctiveRingList.xml
02:24.27booduand also  edit my RINGLIST.DAT
02:24.57booduthe file sound is on the root of tftp server : /tftboot
02:25.23booduIt's the same problem with 7975g and 7906g
02:28.19seanjohnwhat is the o extension in asterisk?
02:28.34seanjohnoperator?
02:30.32seanjohn[anydialplan] exten => o,1,whatextensionamIon
02:31.50*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
02:48.19p3nguinoperator extension from I think the directory and from voicemail only.
02:49.08p3nguinboodu: Are you using SIP on the phones?
02:53.13p3nguinboodu: The 7960/7940 with SIP use RINGLIST.DAT.  With SCCP, they use RINGLIST.XML and/or DISTINCTIVERINGLIST.XML.  I've also been told that newer phones using SCCP want Ringlist.xml.  The file names ARE CASE SENSITIVE.
02:53.30p3nguinCheck the tftpd log to see what files are being requested.
02:55.29*** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com)
02:59.10*** join/#asterisk m_tadeu (~quassel@173.191.19.95.dynamic.jazztel.es)
03:00.57*** join/#asterisk xheliox (~jeff@i216-58-41-253.cybersurf.com)
03:01.01*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
03:01.35*** join/#asterisk boodu (~antoine@175.158.129.128)
03:03.10*** join/#asterisk Mango (~iMango@d154-20-107-65.bchsia.telus.net)
03:11.02boodup3nguin, in RINGLIST.XML with this case ? :O
03:11.25p3nguinwhat?
03:11.59booduyou say the filename in uppercase
03:12.14p3nguinRight.  All file names are case sensitive.
03:13.04seanjohnif I want to Record() to slin audio, do I just put slin as the extension or is it a different extension?
03:13.25p3nguinprobably sln
03:13.36*** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com)
03:15.03boodup3nguin, yes I know I must becarfull at the case sensitive and it's right
03:15.26p3nguinShow me the log for the tftpd where it is trying to load the files.
03:15.42booduI don't have tftpd log :O
03:16.04p3nguinI guess you had better get one if you want to solve the problem.
03:16.14booduhow to ativate the log ?
03:16.27p3nguinWhat tftpd are you using?
03:18.04booduin.tftpd on elastix system
03:18.21seanjohnexten => h,3,system(mime-construct --to ${email} --subject "Telephone Reminder" --file /var/lib/asterisk/sounds/custom/${APPTTIME}.${APPTDT}.${APPTPHONE}.sln --type audio/gsm
03:18.31seanjohnis --type ok?
03:19.18seanjohnwill any media player play sln?
03:19.20p3nguinCheck /var/log/ for various log files where tftpd can write the log.  On my system, it is found at /var/log/daemon.log.
03:20.05seanjohnI have several uncustomized, unlocked ATA's I need to do TFTPD to but I ASSUME the admin pass alone is enough to secure them, right?
03:20.22boodusure but /var/log/daemon does  not exist on elastix :O
03:20.34*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
03:20.42p3nguinThen you should learn your way around and find the file where it DOES log the info.
03:20.54seanjohnp3nguin: is the --type ok for mime construct?
03:21.07*** join/#asterisk Kumbang (~dsp@rusnas.paume.itb.ac.id)
03:21.10seanjohnwill any audio player play sln?
03:21.33booduyes it's on message :)
03:21.40*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73)
03:21.45booduin /var/log/message
03:22.22p3nguinGreat.  Now run "tail -F /var/log/message" and then reset the phone.
03:23.25seanjohnp3nquin, question 1: will any audio player play sln and question 2: is the admin password enough to secure several ATA's I have distributed?
03:25.00boodup3nguin, there are nothing
03:25.23drmessanoLinksys ATAs?
03:26.52*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
03:27.32boodup3nguin, background image choice run and there nothing on the log
03:28.50p3nguinIf there's no tftpd logging in that file, what made you decide that was the right file for the logging?
03:30.47booduI can read my restart of xinetd
03:30.52booduJun  7 08:17:46 elastix xinetd[5284]: START: tftp pid=633 from=192.168.1.13
03:31.12booduand that Jun  7 14:17:22 elastix in.tftpd[4119]: Usage: in.tftpd [-vcl][-a address][-m mappings][-u user][-t inetd_timeout][-T pkt_timeout][-r option...] [-s] [directory ...]
03:31.45UberDuperSo I'm about to setup asterisk on an openwrt router and use a linksys SPA3102. I've never setup asterisk before so I was wondering if any of you knew of a decent howto for setting it up with the SPA3102 ?
03:32.03p3nguinIf the usage appears in the log, that makes me think it has not been invoked correctly and is probably not running.
03:33.57boodup3nguin, it's running sure. I have put the option -v on my xinetd/tftpd option :)
03:34.12booduit's ok i have the log
03:34.55*** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net)
03:38.14booduhttp://pastebin.com/BqbMet4Y
03:38.17boodu:D
03:40.37p3nguinOkay, so you know what file it is looking for.
03:40.42p3nguinRight?
03:41.19booduit's ok thanks man !!!!!!
03:41.27*** join/#asterisk dzup (dzup@unaffiliated/dzup)
03:42.03booduln -s RingList.xml distinctiveringlist.xml
03:42.18booduit's curious the doc say Ringlist.xml :O
03:42.45p3nguinJun  7 03:42:28 asterisk in.tftpd[32522]: RRQ from 192.168.192.26 filename RINGLIST.XML
03:42.52p3nguinThat's what mine does.
03:45.04ChannelZmine fetches ringringringringringringringbananaphone.xml
03:47.08*** join/#asterisk soman (~somnath@118.102.130.6)
03:48.54NuggetI have a ringtone that's a GT3 starting up and accelerating away.  I use it to make sure I get up early enough on track days.
03:52.33*** join/#asterisk pacmanfan (~pacmanfan@72-160-169-131.dyn.centurytel.net)
03:55.22*** join/#asterisk Danig (~fubard@34.79-160-20.customer.lyse.net)
03:56.30Dandangt3?
03:56.39jblackA type of car
03:57.13p3nguinPorsche 911 GT3, I would imagine.
03:57.45Dandanyeah, I imagine
03:58.08pacmanfanok, this ain't really an asterisk-specific problem, but i don't know a better place to ask...
03:58.22*** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus)
03:58.49pacmanfanmy problem is that incoming audio on a SIP trunk works on inbound calls, but not outbound.
03:59.27Dandannat/rtp?
03:59.29Nuggetyeah
03:59.35pacmanfani'm using SIP registration for inbound and outbound; using Flowroute as a provider on both. behind a NAT router, dynamic IP.
04:00.20Dandanregistration is needed only for inbound
04:00.42Dandanoutbound does not require registration (most often)
04:00.45pacmanfani had everything working fine for almost a week. i shut down my AsteriskNOW machine, along with my router and DSL modem. i was assigned a new IP when i fired everything back up, and that's when i started having this problem.
04:00.46pacmanfani see.
04:01.05Dandanah! did you check your extenip?
04:01.14Dandanin sip.conf?
04:01.20pacmanfani am forwarding TCP/UDP 5060 and UDP 10000-20000 to my AsteriskNow machine.
04:01.26p3nguins/exten/extern/
04:01.34pacmanfanno, externip is set?
04:01.36pacmanfanhmm
04:01.37Dandanright
04:01.43*** join/#asterisk jony123 (jony123@unaffiliated/jony123)
04:01.49p3nguin~sipnat
04:01.49infobot[~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
04:01.55p3nguinIf you're using NAT, you have to set it up.
04:02.01pacmanfanthat would make sense, because i couldn't explain it otherwise.
04:02.14pacmanfandoes AsteriskNOW set it automatically, out of the box?
04:02.17p3nguinexternip or externhost, as well as localnet.
04:02.47pacmanfanmaybe i set externip when i was sleep-deprived, and forgot i did so...
04:02.48DandanI have no idea about *NOW
04:02.55pacmanfank
04:03.10pacmanfanso what does one do with a dynamic IP?
04:03.26p3nguinexternhost
04:03.32p3nguinand dynamic hostname
04:03.48pacmanfank
04:04.15pacmanfangreat, i gotta fix ddns in mikrotik now :)
04:04.15boodup3nguin, the problem is partially resolved because I can set a custom default ring
04:04.18pacmanfanthanks guys!!
04:04.41boodubut i see nothing on the log on a file for default ring file
04:04.43p3nguinIt's essentially the same as externip, but it does a lookup of the hostname and sets the IP address based on the DNS response.
04:05.34p3nguinboodu: As far as I know, there is no default ring file... it's built in.
04:07.13p3nguinYou only have to put non-default ring tone files' names in the ringlist (as well as provide the files for download by the phone).
04:07.53booduwhen i go on default ring i have always ring list unavailable
04:08.35*** join/#asterisk coppice (~chatzilla@m121-203-231-242.smartone-vodafone.com)
04:09.15boodui wil found the problem i think is juste a bad name
04:09.22booduciao
04:09.32boodutiux and thx p3nguin
04:10.17UberDuperWow that's a lot of settings on the spa3102
04:11.29p3nguinyup
04:12.03DanigHey guys, I've successfully set up my switchvox server and can recieve calls. All outgoing calls however fails. I've enabled Outgoing call diagnostics and it goes through my outgoing rule ok. then it rings for a few seconds and gives me a forbidden error. I recieve a BUSY Cause code: 21. Call was hung up by the pbx. (Unknown account) Twisting my head for hours now. Please help
04:14.27UberDuperI thought this was going to be more of a dumb device.
04:16.14jony123was about to ask a question but i fixed it.  So nvm
04:20.12jony123Just wanna say thanks to all who have helped me this far on this chan and on the internet.  ANd btw i just rang pizza hit from my asterisk system
04:23.04DanigIn my log i also get a: Got SIP response 603 "Declined"
04:23.20UberDuperIs there a generic asterisk getting started guide?
04:25.31p3nguin~book
04:25.32infobot[~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
04:25.50p3nguin~answers
04:25.51infobot[~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt
04:27.50p3nguinThe SPA-3102 is fairly complex -- it's a router, a dhcp server, an FXO/FXS/VoIP gateway, and probably some other interesting things.
04:35.15UberDuperp3nguin: Thanks for the links.
04:35.43UberDuperI turned off the router bits. I'm just going to use it as an FXO FXS.
04:36.28*** join/#asterisk Defraz (~t0tal@173-116-139-40.pools.spcsdns.net)
04:40.22*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ifjxwhmucnienadp)
04:56.34drmessanoYou won't get something like an SPA-3102 without the router bits'
04:59.03drmessanoThey're marketed as branch office devices providing PSTN failover or passthru.  You won't find a market for an external single FXO device
04:59.57*** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com)
05:17.12UberDuperhuh?
05:20.55p3nguinI think what he means is that you cannot turn off the router in it.
05:21.08p3nguinYou can choose to not use it as a router, but it's still a router.
05:21.52UberDuperIP router?
05:21.58p3nguinyes
05:22.07p3nguinit is an Ethernet router.
05:22.13UberDuperDisable dhcp on the lan and set it to bridged mode. It's no longer a router.
05:22.38p3nguinThere's a way to set to bridged mode?  What software version are you using?
05:22.53UberDuper5.1.10
05:23.06p3nguinOh, I use 3.something and I don't remember seeing such an option.
05:23.19UberDuperRouter | Lan Setup | Networking Service: Bridge
05:23.25p3nguinI'll go look right now.
05:23.32p3nguinis interested
05:27.05p3nguinIt's there, alright.
05:27.33p3nguinSince I don't use the device as an Ethernet router, I guess I didn't have any reason to notice that before.
05:28.13p3nguinIf you go into bridge mode, does it simply ignore all the LAN stuff on that page?
05:29.40p3nguinSoftware Version:3.3.6(GW)
05:31.01UberDuperHaven't checked.
05:31.31p3nguinI'm not where the device is, so I'm not going to go changing random networking settings.
05:33.26p3nguinAlso, if you go to bridge mode, is it a true bridge, where it makes no difference which Ethernet port you connect your cable to?
05:34.08*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
05:35.42UberDuperI haven't tested. But that would be my assumption.
05:36.37p3nguinI'm going to have to experiment with that later.
05:47.43*** join/#asterisk Martinblr (~Miranda@61.12.17.170)
05:52.05*** join/#asterisk ming_zym (~ming_zym@121.0.29.237)
05:54.47*** join/#asterisk bluOxigen (ssf@unaffiliated/bluOxigen)
05:54.57bluOxigenhi
05:57.40*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
06:01.20*** join/#asterisk theBruno (~theBruno@128-177-27-78.ip.openhosting.com)
06:06.23*** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806)
06:07.49*** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net)
06:30.10*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
06:33.08*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
06:33.12*** join/#asterisk Faustov (user@gentoo/user/faustov)
06:41.43*** join/#asterisk ruben23 (~unit41@119.93.217.243)
06:43.10*** join/#asterisk gavimobile (~gavimobil@bzq-84-108-29-62.cablep.bezeqint.net)
06:43.14*** part/#asterisk gavimobile (~gavimobil@bzq-84-108-29-62.cablep.bezeqint.net)
06:44.50ruben23hi guys are there any chnaces i can convert my PRI lines to voip, without any hardware being used..?
06:45.20ChannelZ"convert"?
06:46.07ruben23ChannelZ:i mean used as voip my PRI lines
06:46.54ChannelZlike route incoming calls over voip someplace else?
06:48.55Dandan:)
06:50.00*** join/#asterisk J4zen (~j4zen@95-36-106-201.dsl.alice.nl)
06:50.04ChannelZI'm clearly missing the subtlety of the question
06:50.10*** join/#asterisk af_ (~getsmart@78.134.22.35)
06:50.17Dandanso am I
06:50.27mianosme too
06:50.30Dandanbut I am excused :P it is almost 3am :D
06:51.31mianosif you want to use asterisk you need a PRI card, like the 'TE' range
06:54.50J4zenAs i understand from browsing through the revision board and open tickets it is currently not possible to preserve the callerId on an attended transfer, correct? To elaborate, i present the following situation: Receptionist(100) receives inbound call from external # '0123456789'. Receptionist wishes to forward this call to sales; Receptionist places caller on hold and dials Sales(200); He/She announces the callers name and wishes to forward the c
06:54.55J4zenSorry for the wall of text.
06:56.35Dandan"to forward the c
06:56.36ChannelZThat's sort of how it's designed... because you're the one calling the other person on an attended transfer, not the person you're transferring.
06:57.05Dandanyou might do some magic with astdb and putting people in parking...
06:57.18Dandanbut that is not straightforward
06:57.34J4zenYes i realise its completely by-design, but at the same time not the most efficient. You always want to see the person you're in conversation with, not who forwarded the call (maybe as a sidenote)
06:57.54seanjohnanyone use swift?
06:57.59J4zenanalogue phone systems have been doing this since day one, so a lot of my clients complain about a 'lack' of this feature
06:58.20J4zenlet me rephrase, traditional phone systems*
06:58.38seanjohnpstn, J4zen
06:58.48J4zenthank you ;)
06:59.15seanjohnanyone use swift and know how to keep it from deleting the file after playback?
06:59.21ruben23ChannelZ: sorry got disconnected..
06:59.25ruben23yes your right
06:59.33ruben23can i used PRI lines for voip..?
07:00.03DandanJ4zen: hm, maybe write some code and distribute as one-for-all
07:00.15Dandanor send in as a request for enhancement
07:00.18ChannelZruben23: well yes, that's the whole point of Asterisk..
07:00.29Dandan:)
07:00.35*** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-056-202.pools.arcor-ip.net)
07:00.45ChannelZruben23: but not 'with no additional hardware', you have to get your lines into Asterisk *somehow*
07:01.22J4zenit's been a added feature request for little over 2 years now, yet no-one has stepped up thusfar
07:01.30ChannelZSo you do it
07:01.51ruben23ChannelZ:what hardware do i need..? to let my pri lines be used for voip..?
07:01.54DandanJ4zen: try to write a procedure, or - how you would like it to behave and I am sure someone can get it going for you
07:01.59Dandanwith a little bounty :>
07:02.01Dandanhint, hint
07:02.26Dandanruben23: your description is lacking general direction: what are you trying to accomplish?
07:02.47J4zenUnfortunatly i'm not skilled enough to modify the core, most SIP phones already support it via Alert-info/reinvite headers; problem is Asterisk doesnt sent a reinvite with the original CID upon an attended transfer. Hence the phones won't update their CID
07:03.37ruben23<PROTECTED>
07:04.07Dandanah!
07:04.20Dandanyou need a pri card - most likely a quad one
07:04.26Dandanwith echo canceller or not...
07:04.37Dandanthen - it is all voip, inside the asterisk :)
07:05.23ruben23what quad one..?
07:05.24Dandanunfortunately can't be done without hardware, you *need* to interface those PRIs somehow with asterisk...
07:05.43Dandanguys, what is the newest quad pri card from digium?
07:05.54Dandanruben23: digium makes those, sangoma does too...
07:06.50ChannelZruben23: TE2xx depending on your interface desires
07:06.59ChannelZgo to digium.com and look
07:07.16ChannelZhttp://www.digium.com/en/products/digital/
07:07.20Dandanah, right, they make dual cards now too...
07:07.30Dandani still have some tormentas :-]
07:07.42Dandanand  sangoma s514s :>
07:08.11ruben23thanks guys
07:18.13*** join/#asterisk tamiel (~tamiel@213.30.183.226)
07:20.17*** part/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
07:20.52*** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-056-202.pools.arcor-ip.net)
07:23.03*** join/#asterisk spenguin[work] (~penguin@59.162.86.164)
07:23.37spenguin[work]hey, if a user is logged in from multiple phones into the same extension - is there anyway to have all of them ring?
07:26.56*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
07:27.20ChannelZI think you want SLA but having multiple devices using the same SIP name is not deisreable
07:27.40ChannelZ(which is what I'm assuming you mean when you said "multiple phones into the same extension")
07:27.46*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
07:28.51*** join/#asterisk sekil (~sekil@80.93.247.26)
07:32.48*** join/#asterisk mpe (~mpe@94.127.49.1)
07:34.56*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
07:35.01*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
07:35.24*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
07:36.38*** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com)
07:37.16p3nguinOne extension can certainly dial multiple phones, but multiple phones cannot use the same sip account and work like you might expect.
07:38.12*** join/#asterisk war9407 (war@liquidswords.org)
07:39.19*** join/#asterisk linuxplatform (~Linux@174.80-202-175.nextgentel.com)
07:39.27*** join/#asterisk Faithful (~Faithful@202.6.145.116)
07:40.16spenguin[work]ChannelZ,p3nguin so it would be sensible to have different extensions
07:40.26p3nguinno
07:40.44spenguin[work]p3nguin: ah
07:40.57spenguin[work]different sip accounts, same exten can ring all
07:41.04p3nguincorrect
07:41.14spenguin[work]ok
07:41.18*** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net)
07:42.05p3nguinexten => 1234,1,Dial(SIP/jack&SIP/jill&SIP/jane)
07:42.18p3nguinNow extension 1234 calls all three phones.
07:43.03*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)
07:44.22p3nguinYour other choice is to use a SIP proxy that allows multiple registrations with the same account credentials.
07:45.19*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
07:47.28spenguin[work]p3nguin: ok, thanks
07:49.00*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)
07:49.38*** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-oadjyjfdxlkkxhqa)
07:53.19*** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-056-202.pools.arcor-ip.net)
07:59.53*** join/#asterisk soman (~somnath@118.102.130.6)
08:08.00jony123ok.  I have added the extention using freePBX.  Im using twinkle to try and connect and register.  But now i keep getting 'jony2, registration failed: 403 Forbidden (Bad auth)'.  I allready have a extention configured and that works everytime. ANd theres no differnce between the configs of both.
08:08.07jony123WHat do i do?
08:12.53jony123ahh what the fuk...
08:13.08jony123ops
08:13.10jony123wrong chat sorry
08:24.46*** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com)
08:32.28*** join/#asterisk mikkel (~mikkel@130.226.36.170)
08:44.11*** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr)
08:50.24*** join/#asterisk hipitihop (~denis@203.132.229.236)
08:51.22*** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net)
09:01.28*** join/#asterisk Benwa (~Benwa@dyn.144-85-187-036.dsl.vtx.ch)
09:03.38*** join/#asterisk OrNix (~ornix@host89-251-107-43.hnet.ru)
09:14.35*** join/#asterisk sekil (~sekil@80.93.247.26)
09:18.40*** join/#asterisk gavimobile (~gavimobil@bzq-84-108-29-62.cablep.bezeqint.net)
09:19.14gavimobilei just attempted to install my first installation of asterisk on ubuntu 9.04. i am trying to edit the manager.conf file but my /etc/asaterisk/ directory is empty
09:20.32*** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net)
09:20.38henkgavimobile: try /etc/asterisk/
09:21.44*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
09:21.54gavimobilehenk thats what i ment
09:22.19henkah ok
09:22.49gavimobileany other suggestions?
09:23.02henkgavimobile: on debian (at least stable) asterisk is split in several packages. asterisk-config is the one containing the files in /etc/asterisk/
09:23.06henkis that installed?
09:23.39gavimobilehenk: the /etc/asterisk/ directory is empty
09:23.48gavimobilewhich was my originall question
09:24.17gavimobilehenk: not even sure if my installation is good. however i didnt see any errors when i compiled
09:24.34henkgavimobile: you installed asterisk NOT using the packages your distribution already gives you?
09:25.01gavimobilei did NOT use synaptic
09:25.33*** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-236.canterbury.ac.uk)
09:25.38gavimobilei did however use this to guide me through the installation http://www.experts-exchange.com/articles/Networking/Telecommunications/IP_Telephony/Asterisk_/How-to-Install-Asterisk-on-Ubuntu-9-04-9-10-from-Scratch.html
09:25.54gavimobileplease note this is a test machine!
09:26.44henkah, good luck then...
09:27.14gavimobilehenk: have i dont something wrong? can u help me any further in the matter? if not, why?
09:27.58henkgavimobile: i don't have the time to debug manual installation. imho one should use the packages included in the installed distribution if there are any.
09:28.33henkor rather: i am not motivated to spend my time for that atm :)
09:28.37*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
09:28.59gavimobilehenk, could u possibly tell me how to remove any previous instlations? this was i can install from synaptics
09:30.07henkgavimobile: if you are lucky there is 'make uninstall'. if not, you probably have to manually remove anything put there by the install script. but you can also stay with the manual installation if you like it that way...
09:30.43gavimobilehenk, am i better off reinstalling my os?
09:31.07henki don't know
09:32.20kaldemarthere is an uninstall target in asterisk's makefile. so, "make uninstall" will install if for you. make config on the other hand will make you the sample configurations in /etc/asterisk.
09:32.26kaldemargavimobile: ^^
09:34.09gavimobilekaldemar: thanks.. ill give it a try
09:38.23*** join/#asterisk DJ_HaMsTa (~carlos@unaffiliated/dj-hamsta/x-2342346)
09:42.00*** part/#asterisk sekil (~sekil@80.93.247.26)
09:45.55*** join/#asterisk soman (~somnath@118.102.130.6)
09:46.48UberDuperShould I even bother trying to use a gui to set this up?
09:47.50kaldemarUberDuper: if you're planning to use asterisk without a gui, no.
09:48.21UberDuperIt's just for a (what I hope to be) simple home system for blacklisting callers.
09:49.06*** join/#asterisk sulex (~sulex@office.blindata.ch)
09:50.07*** join/#asterisk hesco (~hesco@c-24-99-160-121.hsd1.ga.comcast.net)
09:53.27*** join/#asterisk BANSAL (~bansal@117.199.117.143)
09:55.09*** join/#asterisk mikkel (~mikkel@130.226.36.170)
09:56.10hescoI'm working w/ colleague and now have his softphone (3cx) registered to my server from outside my local firewall.  We can make one another's phone ring through the pbx.  But we hear no audio when the call is answered.  The warning reads:  "chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission"
09:56.27hescoany clues how I might proceed to make this work?
09:57.56kaldemar~sipnat
09:58.07infobot[~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
10:09.47*** join/#asterisk UQlev (~yuriy@212.50.99.8)
10:20.09hescothank you sir.  checking that out now.
10:22.49*** join/#asterisk pinoyskull (~pinoyskul@121.54.32.140)
10:25.49*** join/#asterisk joobie (~joobie@CPE-124-181-130-239.vic.bigpond.net.au)
10:30.08*** join/#asterisk BANSAL (~bansal@117.199.121.196)
10:30.55*** join/#asterisk Benwa (~Benwa@dyn.144-85-187-036.dsl.vtx.ch)
10:37.53*** join/#asterisk gazzerh (~garryh@host81-143-121-49.in-addr.btopenworld.com)
10:40.37*** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br)
10:49.04hescoAs I think through this sipnat issue, I'm realizing that my router is actually a locked vonage ATA (motorola vt2442).  Would it interfere somehow with my NAT'ing my sip and rtp ports to my server inside this LAN?
10:50.27ChainsawIt wouldn't surprise me if they "accidentally" broke that, yes.
10:51.35*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73)
10:52.08hescoyes, might not be time yet to buy a new router, but it is certainly time to stop trying to figure out my missing audio issue and get some rest.
10:56.16*** join/#asterisk Ole_ (ole@54b.pl)
10:57.31*** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net)
10:59.10*** join/#asterisk Hoi (~hoiung@client-80-3-100-202.bsh-bng-012.adsl.virginmedia.net)
11:04.20*** join/#asterisk Cain (~Geek@unaffiliated/cain)
11:06.30*** join/#asterisk ming_zym (~ming_zym@121.0.29.237)
11:15.21*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
11:35.59*** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73)
11:40.52*** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69)
11:45.53*** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp)
11:49.42*** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-ymdjegtydseukyvq)
11:49.50*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
11:57.40*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)
11:58.23*** join/#asterisk saftsack (~oliver@p579DD0C5.dip.t-dialin.net)
12:00.39pigpenHi all, quick question.  I am setting up an Asterisk box to handle voicemail for a Nortel Norstar MICS.  However the analog ports are not sending the asterisk box "hangup" signal.
12:00.53pigpenAny ideas to possibly overcome this other than an absolute timeout?
12:01.03[TK]D-Fenderpigpen: "callprogress=yes" <- and say a prayer
12:01.18pigpenHey [TK]D-Fender.
12:01.28pigpenYeah, Did, prayed.  No dice.
12:02.00pigpenI have read that depending on the module on the Norstar, that "Disconnect Supervision" can be enabled....so I guess that is my last best bet.
12:02.10pigpenThanks for confirming....
12:02.18*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
12:03.47k-mani seem to keep getting this problem with my vsp, i get asterisk working ok, then all of a sudden it stops registering, then i change the register line to register with anther one of their servers. then again, that works ok for a while and then stop
12:03.51k-mans
12:04.06*** join/#asterisk coppice (~chatzilla@m121-202-3-16.smartone-vodafone.com)
12:07.37*** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net)
12:07.49*** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net)
12:08.51*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
12:12.51*** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br)
12:20.23tzafrir_laptoppigpen, otherwise, you need "busydetect"
12:26.15*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-126-104.mia.bellsouth.net)
12:26.51*** join/#asterisk muiro (~muiro@unaffiliated/muiro)
12:27.36*** part/#asterisk muiro (~muiro@unaffiliated/muiro)
12:32.40*** join/#asterisk bent_screwdriver (~socain00@74.255.249.66)
12:40.19pigpenYeah, got busydetect=yes and busycount=5.  I wonder if I lower the busycount.
12:44.05*** join/#asterisk houms (~houms@wsip-70-167-244-115.dc.dc.cox.net)
12:44.19houmscan anyone help me troubleshoot siptapi? when I try to dial from outlook I get the following output in asterisk cli
12:44.23houmsusing trixbox
12:44.29houmshttp://pastebin.com/wDwDLA8h
12:46.24[TK]D-Fenderhoums: you have no 110 in your system
12:46.56houmsbut that is my extension on the system
12:47.06[TK]D-Fenderhoums: Your call is coming in.  *'s job is done.  Go configure FreePBX to handle that number.  It is NOT supoprted here.
12:47.08[TK]D-Fender~freepbx
12:47.31infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
12:47.33[TK]D-Fenderhoums:   -- Executing [110@from-sip-external:1] <--- you call is not landing in [from-internal]
12:47.44[TK]D-Fenderhoums: Go set the context right then
12:47.58[TK]D-Fenderhoums: Because that other one doesn't include internal extensions.
12:48.32houmsmy question is where do i set the context correctly?
12:48.46[TK]D-Fenderhoums: GUI issue.  #freepbx <-----------
12:48.57houmsthanks for the info.
12:52.14*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
12:52.18*** part/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
12:52.22*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
12:57.09tzafrir_laptoppigpen, the lower the busycount the higher the chance for false detection
12:59.48*** join/#asterisk coppice (~chatzilla@202.64.176.19)
13:03.25*** join/#asterisk infobot (~infobot@rikers.org)
13:03.25*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.8 (2010/06/01), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.32 (2010/06/01), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.1 (2010/06/02) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs
13:10.09*** join/#asterisk brad_mssw (~brad@shop.monetra.com)
13:17.21*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)
13:18.58*** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca)
13:19.48*** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net)
13:21.34*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
13:22.12*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:22.12*** mode/#asterisk [+o leifmadsen] by ChanServ
13:24.19*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
13:26.50*** join/#asterisk Benwa (~Benwa@dyn.144-85-187-036.dsl.vtx.ch)
13:28.19*** join/#asterisk bsaxon (~bsaxon@12.107.149.61)
13:28.39*** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca)
13:29.03*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
13:30.03*** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca)
13:30.25*** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831)
13:30.52*** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
13:32.17*** join/#asterisk houms (~houms@wsip-70-167-244-115.dc.dc.cox.net)
13:32.58*** join/#asterisk turneralex (~turnerale@203.206.217.243)
13:33.01turneralexHeyhey
13:33.13*** join/#asterisk fox_dewd (~r@41-133-163-224.dsl.mweb.co.za)
13:33.25fox_dewdgood day ppl
13:33.57fox_dewdI would like to know how do I test a 32 channel echo cancelation card
13:34.09*** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com)
13:34.34turneralexquick question. If i run an asterisk server and purchase 1000 Sip lines from a sip provider. Can I resell those lines to end users, using ATA devices? In essence building my own VOIP provider.. Is that possible with Asterisk?
13:35.34[TK]D-Fenderfox_dewd: Place calls
13:36.11[TK]D-Fenderturneralex: * doesn't care what you intend to bill people
13:36.25turneralexAh yes, forgot about that bit :/
13:37.15[TK]D-Fenderturneralex: * can use ITSPs and ATAs.  Your only difference is the fact you want to bill someone/
13:37.37*** join/#asterisk UQlev (~yuriy@212.50.99.8)
13:37.44turneralexAny decent radius based billing applications out?
13:37.46fox_dewdwe are a reseller for digium products so I need to know how I can test the echo cancelation card to its full extend to be able to say that its working or not
13:38.14[TK]D-Fenderturneralex: * doesn't do RADIUS for that last I heard.  CDR <-
13:38.34[TK]D-Fenderfox_dewd: Use it <-
13:38.41turneralexFrustrating
13:39.07[TK]D-Fenderturneralex: Make a conversion script
13:39.52turneralexis thinking i should just hire someone :P
13:40.18turneralexhttp://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
13:40.40fox_dewd[TK]D-Fender: If i do not have a line that has an echo how can I test it?
13:40.41*** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br)
13:41.05*** join/#asterisk hugorebelo (~hugo@200.171.132.124)
13:41.12[TK]D-Fenderfox_dewd: What is a etst that doesn't place a challange for ti to handle?
13:41.15[TK]D-Fendertest*
13:42.01[TK]D-Fenderturneralex: if this little bit scares you.. then you'd better.  And also better consider your business model.
13:42.13turneralexWell see Im a networking guy
13:42.20turneralexAnd we've deployed this flash network in Sydney
13:42.26turneralexand were considering VoIP
13:42.33turneralexI was just curious as to what was involved
13:43.32[TK]D-Fenderturneralex: * isn't what would eb commonly used for such an enterprise anways.
13:43.47fox_dewd[TK]D-Fender: I do not want to waste time with testing when I do not need to
13:43.47[TK]D-Fenderturneralex: Most ITSPs that use * at all use it in the back-end as an app server, etc
13:44.01[TK]D-Fenderfox_dewd: If you don't need to.. then why are you doing it?
13:44.14[TK]D-Fenderstares at chickens & eggs for a bit...
13:44.29fox_dewd[TK]D-Fender: its my job to test it, not to waste time
13:44.42[TK]D-Fenderfox_dewd: So what would be a waste of time?
13:45.04fox_dewdoh please
13:45.20Dandanlol
13:45.21[TK]D-Fendermoron
13:45.21Dandan:>
13:46.10[TK]D-Fender"Hi I want to know if my car works, but I don't want to start the engine and take it for a drive.  So how will I KNOW that it's working?"
13:46.26[TK]D-FenderCorrection...
13:46.33[TK]D-FenderFUCKING MORON
13:47.01Dandan[TK]D-Fender: you can probably inspect it visually :>
13:47.54[TK]D-FenderDandan: Yes... I inspected the text fox_dewd wrote visually and determined that he is a fucking moron :p
13:48.09Dandan:>
13:48.13*** join/#asterisk guax (~guax@unaffiliated/guaxinim)
13:48.58guaxleifmadsen, sorry to bother but any news on the recording calls with transfered calls from the originator?
13:49.33leifmadsenguax: I know what you're talking about, but I know of no current work on changing any behaviour there
13:49.39Dandanoh, question, I am thinking about replacing our old (as in OLD!) hylafax installation with asterisk, but need the php interface to send faxes. Is there a such thing or should I start writing something now?
13:49.46Dandanleifmadsen: how is TFOT3 going? :>
13:49.47leifmadsenguax: there would be no news to report afaik
13:50.02leifmadsenDandan: oh not too shabby -- just finished one of the chapters on Friday (new chapter)
13:50.19Dandanhe spelled behaviour with a *U*, shame on you leif!
13:50.42leifmadsenDandan: I still use hylafax with Asterisk, so not sure you really need to replace it entirely
13:50.47guaxleifmadsen, im actually considering the digium support subscription as a way to solve that but im afraid that woudnt be enough.
13:50.51leifmadsenDandan: I'm Canadian -- I use British style spelling :)
13:51.14Dandanleifmadsen: would like to just add that feature, hylafax is old, works with 32 analog modems and just shows its age :-/
13:51.15leifmadsenguax: I'm not sure if that's the point of the support subscriptions, but I'd check with Digium sales to verify that
13:51.28Dandanthat's why php2fax would be a welcome addition
13:52.10leifmadsenguax: if you can clearly demonstrate the issue, you can open a bug in the issue tracker and I can better review it to determine if that is a bug, or just "how it works"
13:52.18Dandanbut I can't find anything clean/easy/useful
13:54.29[TK]D-Fenderleifmadsen: We speak English.. not "American" :)
13:54.36guaxleifmadsen, i will, thanks. Another issue is related to CDR but im pretty sure thats a recurrent subject and i bet just CEL in late 1.8 will address that.
13:54.58*** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net)
13:55.02leifmadsenguax: ya, changing CDR is a hard thing to do and unless it is a regression, is unlikely to be changed
13:55.17leifmadsen[TK]D-Fender: :)
13:56.21Dandanoh come on people :) Digium is American, so we speak Americano!
13:56.31guaxi see "and engineering support (bug fixing)." on the support page, but i dont think 2k will make digium fix the cdr on 1.4 =P
13:57.32*** join/#asterisk Martinblr (~Miranda@61.12.17.170)
13:59.24*** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com)
14:01.43*** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca)
14:04.57*** join/#asterisk sekil (~sekil@80.93.247.26)
14:05.43*** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br)
14:08.11*** join/#asterisk Bladerunner05 (~Bladerunn@81-174-56-54.static.ngi.it)
14:08.30Bladerunner05hello
14:09.03*** join/#asterisk soman (~somnath@118.102.130.6)
14:11.33*** join/#asterisk Professional (~Pro@unaffiliated/shani)
14:12.03*** join/#asterisk Researcher (~user@unaffiliated/unafilliate)
14:14.14Bladerunner05I'm using tdm410p and if I call using  _0.,1,Dial(DAHDI/G1/${EXTEN}) all works fine, but If I do  _0.,1,Dial(DAHDI/1/${EXTEN})  _1.,1,Dial(DAHDI/2/${EXTEN:1}) to choose the channel for outgoing call it return this error: app_dial.c:1275 dial_exec_full: unable to create channel of type 'DAHDI' cuase 0: unknown
14:14.21guaxleifmadsen, https://issues.asterisk.org/view.php?id=17484
14:15.41*** join/#asterisk Janos (~cramos@190.10.52.113)
14:16.03tzafrir_laptopBladerunner05, and those channels are actually available? You see them on the output of:  dahdi show channels     ?
14:16.36*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
14:16.52*** join/#asterisk Professional (~Pro@unaffiliated/shani)
14:17.35Bladerunner05tzafrir_laptop: if I run the command I get: pseudo 1 did_trunk_1 default and same for psudo 2, pseudo3, pseudo 4
14:17.41*** join/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr)
14:17.45hurdmanhi
14:18.02hurdmanis there a way to exit from a conference without HangUp ?
14:18.19hurdmanto be able to go into an over conference ?
14:18.21leifmadsenhurdman: there is an option for that I believe
14:18.34tzafrir_laptopBladerunner05, hmmm... could you please pastebin the exact output?
14:18.37leifmadsenhurdman: some DTMF key to exit the conference in combination with continuing in the dialplan
14:18.48leifmadsenhurdman: might be two different flags
14:18.52hurdmanleifmadsen: ok, i should re-read the manual
14:18.56hurdmanthanks
14:18.58leifmadsencore show application meetme
14:19.09leifmadsenmay or may not be possible in 1.4
14:19.09McBoingboI keep getting "WARNING[20488] chan_sip.c: Got 200 OK on REGISTER that isn't a register" in Asterisks messages log, everything seems to be working fine, is this a problem connecting to our SIP provider?
14:19.56Bladerunner05tzafrir_laptop: take a look http://pastebin.ca/1878423
14:20.21*** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net)
14:20.26WIMPyMcBoingbo: I used to see that when I had problems with packet loss.
14:20.28hurdmanleifmadsen: i found, p['<keys>']
14:20.34hurdmanleifmadsen: that's great, thanks
14:21.21tzafrir_laptopBladerunner05, what device is it?
14:21.31tzafrir_laptopWhat version of DAHDI?
14:22.20Bladerunner05tzafrir: tdm410p , and how can I ook the dahdi version ?
14:23.18*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
14:23.18*** mode/#asterisk [+o putnopvut] by ChanServ
14:24.14McBoingbohow can I troubleshoot the "WARNING[20488] chan_sip.c: Got 200 OK on REGISTER that isn't a register" problem then?
14:25.47*** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br)
14:26.02*** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net)
14:26.02*** mode/#asterisk [+o Deeewayne] by ChanServ
14:27.46*** join/#asterisk raj-darkmystery (~suraj@122.169.7.126)
14:27.57raj-darkmysterykaldemar: r u there?
14:28.15raj-darkmysterykaldemar: need a li'l help with asterisk
14:29.34*** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.163.227.mtnl.net.in)
14:31.36ArsenickHi all, I'm trying to find out if it's possible to notify asterisk users that their voicemail if full ?? Like and email or something like that ?
14:31.41*** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com)
14:32.16raj-darkmysteryafter reboot redfone is showing red alarm
14:32.37*** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com)
14:32.38Janoshey there, i have a asterisk server who's load is going really high in a very random way, what would be the best way to diagnose a load problem ?, logs don't say anything
14:32.55raj-darkmysteryhere is the outut of fonulator status : fonulator dead but subsys locked
14:33.41ArsenickJanos, did you try to monitor the server ? I mean using sar or other tool like that ? to see if there's a certain time of the day or other similarity between the times you have problems ?
14:34.11tzafrir_laptopBladerunner05, cat /sys/module/dahdi/version
14:35.13raj-darkmysterytzafrir_laptop: 2.1.0.4
14:35.19*** join/#asterisk g0z (~gauze@entropy.tmok.com)
14:35.21*** join/#asterisk nova911 (~Adium@122.182.0.38)
14:36.21tzafrir_laptopraj-darkmystery, is that an answer to my question to Bladerunner05 ?
14:36.49raj-darkmysterytzafrir_laptop: ohh i thought i was askin to me :-/
14:37.01[TK]D-FenderArsenick: Either run a process to check your boxes on an interval, or whenever you call Voicemail() to leave one.
14:37.15[TK]D-FenderArsenick: But it won't be * sending the e-mail.. its up to you
14:37.28raj-darkmysteryguys need some help with asterisk, unable to call, redfone showing red alarm
14:38.11tzafrir_laptopraj-darkmystery, I'm not well-familiar with redfone. But red alarm basically means: no bits are flowing
14:38.22tzafrir_laptopDid it work before ?
14:38.48nova911is there way to get simultaneous ring on multiple sip enabled soft & hard phones
14:39.40[TK]D-Fendernova911: what is "simultaneous"?
14:40.33[TK]D-Fendernova911: "core show application dial" <- call multiple devices.
14:40.33JanosArsenick, yes setup a monitoring tool last time it happened, two weeks ago, but like i said it's totally random, can't find a patter yet, but what i was really wondering is what should i check in this cases, assuming i'm able to log in, because top only shows the asterisk process consuming 100% cpu
14:40.44raj-darkmysterytzafrir_laptop: yes it was working till machine got reboot due to power failure
14:40.55nova911my manager want all his sip enabled phones to be ringing when he receives a call
14:41.19tzafrir_laptopIs the remote device up?
14:41.46[TK]D-Fendernova911: then dial them all
14:41.49raj-darkmysterytzafrir_laptop: remote device in the sense? are you asking about clients?
14:42.45tzafrir_laptopraj-darkmystery, the adapter
14:42.48nova911sip extension configured is same on all the devices
14:42.57[TK]D-Fendernova911: that is NOT happening with *
14:43.02tzafrir_laptopanyway, I don't know it well enough, so I give up at this point
14:43.03nova911hahah
14:43.07raj-darkmysterytzafrir_laptop: yes it is up
14:43.12[TK]D-Fendernova911: You can't have multiple devices fighting over a registration
14:43.20[TK]D-Fendernova911: The last one wins.  End of story
14:43.24nova911yes i thought the same
14:43.31[TK]D-Fendernova911: Register as different users
14:43.32Bladerunner05tzafrir_laptop: the version : 2.2.0.2
14:43.40*** join/#asterisk pyite (~dschreibe@unaffiliated/pyite)
14:43.45nova911u mean diff extensions
14:44.02raj-darkmysterytzafrir_laptop: i frequently face this problem.. as soon as machine gets reboot, everytime i need to troubleshoot that.. previously i was restarting fonulator and trick was working
14:44.20[TK]D-Fendernova911: No, and extension is a line in extensions.conf.  I'm saying register as a different eprr in sip.conf
14:44.28[TK]D-Fendernova911: And dial them all.
14:45.05Janosso anyone has any idea of what to look for when your asterisk is eating up your cpu ?
14:45.30[TK]D-FenderJanos: What version?
14:45.52Arsenick[TK]D-Fender, pl Thanks!
14:46.04*** join/#asterisk AsteriskNoob (~AsteriskN@host86-128-21-197.range86-128.btcentralplus.com)
14:46.28raj-darkmysterytzafrir_laptop: here is the debug output : app_dial.c:1296 dial_exec_full: Unable to creat
14:46.31raj-darkmysterye channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
14:46.55AsteriskNoobHello. I was hoping someone could help me get Asterisk 1.6 to send fake ringtones to callers. When people call, they hear only silence.
14:47.02tzafrir_laptopraj-darkmystery, if the span is in alarm, you can't get calls over it
14:47.11ArsenickJanos, I never had such problem with asterisk, but I'm pretty sure if you activate the log you should wat least see what's happenning when the process hang
14:47.14nova911Thanks  [TK]D-Fender
14:47.32*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
14:48.24Janos[TK]D-Fender, 1.4.21
14:48.39[TK]D-FenderJanos: Ancient.  update time.
14:49.05[TK]D-FenderAsteriskNoob: Ringing() <- Core show application Dial() <-  take your pick
14:49.39raj-darkmysterytzafrir_laptop: any guess? :-/
14:50.26*** join/#asterisk tftech (~tftech@63.214.236.150)
14:51.08tftechGood morning room
14:51.22tftechI have a silly question, but having an issue figuring it out
14:51.36McBoingboAny ideas how I can solve/troubleshoot "Got 200 OK on REGISTER that isn't a register" errors?
14:51.59AsteriskNoob[TK]D-Fender: Ringing.. haven't tried that one...
14:52.10tftechI am trying to send only the last 14 digits of my DNIS
14:52.25[TK]D-Fendertftech: To where?
14:52.50tftechFender, to the termination gateway
14:53.02tftechI am taking the call in, and resending it to a cisco
14:53.16tftechI need to add 4 digits to the end of the ANI
14:53.26tftechand only send a total of 14 digits as the DNIS
14:53.36Janos[TK]D-Fender, well not really, only 2 years old, besides this king of problem is something i would really like to learn how to diagnose, so any idea how to ask asterisk "what are you doing ???" :P
14:53.36tftechI got it all, but the keeping it to 14
14:53.39raj-darkmysteryredfone showing red alarm, debug showing cause 34 - Circuit/channel congestion
14:55.22tftechexten => XXXXXXXXXXX,1,Dial(SIP/${CALLERID(num)}3401@XXXX)
14:55.35tftechjust need to send the 3401 and only 10 digits of the ANI
14:56.05AsteriskNoob[TK]D-Fender: Ringing() didn't work... and the Dial ...,r also does not work. Any other ideas to get ringing to work?
14:56.15guaxleifmadsen, thanks for the fast answer
14:56.44*** join/#asterisk ariel_ (~chatzilla@63.214.236.169)
14:57.00tftechariel - Is that you? -thor
14:57.14*** join/#asterisk Corydon76-lap (~Corydon76@nat/digium/x-bawhidmmzhvpcmpd)
14:57.15*** mode/#asterisk [+o Corydon76-lap] by ChanServ
14:57.32ariel_yes
14:57.35tftechlol
14:58.55ariel_fancy meeting you hear today.
14:59.01tftechyea
14:59.22tftechI need to get in here more to learn
14:59.30raj-darkmysteryguys need help, aterisk redfone showing red alarm after reboot, unable to make calls, debugging shows cause 34
14:59.34ariel_Ahh
14:59.51tftechI ment to come by with my issue, but we are in a meeting
15:01.00[TK]D-FenderAsteriskNoob: I have no proof that you've done this in a sane way...
15:01.26AsteriskNoob[TK]D-Fender: OK.. Where can I show you?
15:01.34WIMPyraj-darkmystery: Either you don't have a line connected or your redfone is telling rubbish.
15:01.46[TK]D-FenderAsteriskNoob: PASTEBIN.  Where else?
15:01.48*** join/#asterisk yahh (~root@122.170.40.59)
15:01.48*** join/#asterisk d00gster (~dt@94.96.17.23)
15:02.15yahhguy's i am facing a problem
15:02.41yahhi have T4XXP and T2XXP on my server
15:03.14raj-darkmysteryWIMPy: i think redfone is telling rubbish, but i need to troubleshoot that, i cant convence redfone for working fine :(
15:03.17raj-darkmysteryWIMPy: line is connected, but still it is showing red alarm
15:03.30raj-darkmysteryi am a newbie in asterisk
15:03.30yahhnow if i dial a call to the number then it is giving error - Ring requested on unconfigured channel 0/1 span 2
15:03.36yahhand call got hangup
15:03.43AsteriskNoob[TK]D-Fender http://pastebin.com/snB5DX6V
15:03.53yahhwhen i see pri show spans it is showing following
15:03.59yahhPRI span 1/0: Provisioned, Up, Active
15:04.00yahhPRI span 2/0: Provisioned, Up, Active
15:04.00yahhPRI span 2/1: Provisioned, Up, Standby
15:04.00yahhPRI span 5/0: Provisioned, Up, Active
15:04.00yahhPRI span 6/0: Provisioned, Up, Active
15:04.21yahhso here i can see that call is going to standby span
15:04.43yahhnow how to make that span Active
15:04.43yahh?
15:05.09[TK]D-FenderAsteriskNoob: Show an actual complete call
15:05.17[TK]D-Fenderyahh: PASTEBIN, do not flood in here
15:05.19[TK]D-Fender~pb
15:05.20infobot[~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.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.
15:05.21[TK]D-Fender^^^^^^^^^66
15:05.22WIMPyraj-darkmystery: Thats not an Asterisk issue you have there, but one with the redfone. There are probably not too many ppl around with redfone experioences.
15:05.30yahhi am sorry
15:05.40yahhI will take care now onward
15:05.50[TK]D-Fenderraj-darkmystery: And infinitely fewer with positive ones
15:06.18bluOxigenif i have more then one asterisk folder in /etc like astersk asterisk-old asterik-abc how can i find out from which folder the asterisk is runniing
15:06.20Bladerunner05tzafrir_laptop: dahdi version is 2.2.0.2
15:06.29*** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.135.216.mtnl.net.in)
15:06.31Dandani swapped redphone for adtran
15:06.39Dandanredphone and their TDMoE sux...
15:06.44raj-darkmysteryWIMPy: ohh.. but if possible can you ask to your expertise friends as well... coz i think i am in big trouble :(
15:06.58AsteriskNoob[TK]D-Fender: http://pastebin.com/dD9gNwkk
15:07.03DandanBladerunner05: try # lsof
15:07.04Dandan:)
15:07.11*** join/#asterisk mpe (~mpe@94.127.49.1)
15:08.03tzafrir_laptopBladerunner05, just looked again at your dial strings. Note you have ${EXTEN} in one and ${EXTEN:1} in the other
15:08.27*** join/#asterisk raj-darkmystery (~suraj@122.169.7.126)
15:08.29WIMPyraj-darkmystery: Try calling redfone.
15:08.48WIMPyraj-darkmystery: Try calling redfone.
15:08.51Bladerunner05tzafrir_laptop: is to use 2 before the number to use che second channel ..
15:09.11Bladerunner05tazafrir_laptiop: and with 0 use the first available channel
15:09.19tzafrir_laptopBladerunner05, just make sure first you dial exactly the same number to the same line
15:09.29[TK]D-FenderAsteriskNoob: So you answer the call play audio and get no ringing calling out to handsets?
15:10.13AsteriskNoob[TK]D-Fender: The audio plays fine, and the wait is fine, then the phones ring, but the caller hears only silence... I'm using 1.6.2.7 if that helps.
15:10.19McBoingboIs the "Got 200 OK on REGISTER that isn't a register" problem a vague one, dont seem to be getting a lot of feedback from you guys, which is odd :)
15:10.46[TK]D-FenderAsteriskNoob: Verify taht you have a proper indications.conf file
15:10.46bluOxigenlooking for p3nguin
15:10.54[TK]D-FenderAsteriskNoob: If youa re unsure swap yours for teh sample config
15:11.42Bladerunner05tzafrir_laptop: this is a part of my dialplan http://pastebin.ca/1878452 and use 2,4,6,8 for channel 1,2,3,4 and 0 to use the first available channel
15:11.46tzafrir_laptopBladerunner05, you can also test lines directly using:  originate DAHDI/1/123456 application Playback demo-instruct
15:12.08tzafrir_laptopthis dials the number 123456 to channel 1
15:12.40AsteriskNoob[TK]D-Fender: It looks fine, but I'll paste in...
15:12.51[TK]D-FenderAsteriskNoob: No, don't paste.  Just replace
15:13.27AsteriskNoob[TK]D-Fender: I meant do a pastebin so you can see the UK bit.
15:14.19*** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk)
15:14.50Bladerunner05tzafrir: with originate works fine..
15:15.18*** part/#asterisk eject_ck (~eject_ck@85.223.182.86)
15:15.41AsteriskNoob[TK]D-Fender http://pastebin.com/FAdYLEdS
15:16.31[TK]D-FenderAsteriskNoob: I'm at a bit of a loss then... go look at sip debug and see if the phones report ringing.
15:16.56[TK]D-FenderAsteriskNoob: and try with the Dial option instead
15:17.46AsteriskNoob[TK]D-Fender:  -- SIP/lsa_14-0000026c is ringing is what the asterisk -r console shows, and I've tried the ,r option in Dial() and that fails too.
15:18.24AsteriskNoob[TK]D-Fender: This worked just fine on an AsteriskNow system running 1.4.xx
15:18.35[TK]D-FenderastWhat ver preciesly?
15:19.08raj-darkmysteryWIMPy: thanks for your help but i can not call them up, they have ticketing system and i do not have all the details available required for registering a ticket
15:19.47raj-darkmysteryhey guys anyone knows how to taggle with redfone red alarm, asterisk debug output shows cause 34
15:20.52p3nguinbluoxigen: What'd I do?!
15:21.10AsteriskNoob[TK]D-Fender: Not sure what version it was exactly... It was an AsteriskNow I downloaded from Digium.com about 2-3 months ago (in March). Now they show 1.5...
15:21.43[TK]D-FenderAsteriskNoob: ... you're at CLI on your machine... go LOOK
15:22.03AsteriskNoob[TK]D-Fender: The current one is 1.6.2.7
15:22.06*** join/#asterisk BANSAL (~bansal@117.207.80.49)
15:23.11AsteriskNoobIt's not an AsteriskNow install, but a CentoOS yum install.
15:23.25AsteriskNoob(not that that should make a diff)
15:23.30*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
15:24.54[TK]D-FenderAsteriskNoob: it could.  Package vs source tarball
15:25.40AsteriskNoob[TK]D-Fender: Oh.. It was a yum install Asterisk type install using the repos given on the Digium website.
15:26.11AsteriskNoobSo rpms.
15:29.44*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
15:29.49bluOxigenp3nguin : aah good u here, HI,.... i have a redhat bbox, with asterisk installed and running on it, its my friends machine, now under /etc there are more then one asterisk folders, how can i find out which asterisk is the one running
15:31.11[TK]D-FenderbluOxigen: do a reload and see which etc folder is being referenced
15:31.56[TK]D-FenderbluOxigen: And that's hardly a question you need to target at just one person
15:32.32AsteriskNoob[TK]D-Fender: I might have a conflict. I have asterisk 1.4 and 1.6 installed according to yum...
15:32.55bluOxigen:) thx for the advice  [TK]D-Fender],
15:33.07bluOxigenwen i do reload
15:33.11bluOxigeni do not see any thing
15:33.25[TK]D-FenderbluOxigen: core set verbose 10
15:34.10AsteriskNoob[TK]D-Fender: Nevermind. I misread the info..
15:35.02bluOxigen[TK]D-Fender], still just telling me the port range allocated and that event and queue logger restarted
15:35.33[TK]D-FenderbluOxigen: PASTEBIN <-
15:41.04*** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com)
15:42.13*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
15:43.42*** join/#asterisk Belgarath (~Belgarath@banda.pl)
15:45.06*** join/#asterisk CunningPike (~CunningPi@204.239.8.157)
15:45.32yahhwhy PRI span shows "Standby" ?
15:48.18bluOxigen[TK]D-Fender - i am using VNC viewer for redhat, how can i do pastebin
15:48.45[TK]D-FenderbluOxigen: SSH to the box instead
15:49.13p3nguinor open a browser window after you've connected the VNC.
15:50.22bluOxigenhttp://pastebin/Z7C12372
15:50.27bluOxigenhttp://pastebin.com/Z7C12372
15:51.08p3nguinreload?  What is this, Asterisk 1.2?
15:51.19yahhwhy PRI span shows "Standby" ?
15:51.25p3nguinYou already said that.
15:51.50*** join/#asterisk jrz (~jrz@a190165.upc-a.chello.nl)
15:52.23*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
15:56.03*** join/#asterisk btsteve (~tstevens@204.10.23.130)
16:01.20*** part/#asterisk tftech (~tftech@63.214.236.150)
16:02.32bluOxigenhttp://pastebin.com/uei0GG0y
16:03.01[TK]D-FenderbluOxigen: Stop * entirely, then start it manually
16:03.16Dandan(with a -c at the end :>)
16:03.22p3nguinAsterisk 1.6.0.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
16:03.28p3nguinTHere's your clue.
16:03.38bluOxigen?
16:03.57p3nguinYou wanted to know what version you were using, right?
16:04.08bluOxigennop
16:04.11Dandansaturn*CLI> show version
16:04.11DandanAsterisk CVS-04/27/04-18:17:59 built by root@saturn on a i686 running Linux
16:04.13bluOxigenu asked me
16:04.13Dandan:->
16:04.31Dandanthat is in production! :)
16:04.32russellbDandan: awesome :-)
16:04.36Dandanand has been for the last 6 yrs :)
16:04.42Dandanhey Russ :)
16:04.45russellbwaves
16:04.58p3nguinhrm... I asked you what version?  I don't remember that.  I could have sworn you were trying to figure out what asterisk you were using.
16:05.02bluOxigen[TK]D-Fender : stop now ....right ?
16:05.15Dandanhaven't been drining with you since... Arizona :)
16:05.51bluOxigen[TK]D-Fender : asterisk -vvvvvvvgc ? is that wat i be doing
16:07.11[TK]D-Fenderys
16:09.12bluOxigeni think i have found the problem
16:09.31bluOxigenhttp://pastebin.com/yjxingfu
16:09.39bluOxigenno modules loaded
16:10.04*** join/#asterisk The-Bat (~The-Bat@59.162.86.164)
16:10.06bluOxigenthe folder /etc/asterisk is empty no files in that folder
16:11.48[TK]D-FenderbluOxigen: Go track down what binary that is calling.  it will use /etc/asterisk by default unless you specified otherwise.  Check your startup script if you were using one. Restarting manually may not have been a great idea in that case
16:15.24*** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net)
16:17.01*** join/#asterisk Yon (Yon@c80-217-244-16.bredband.comhem.se)
16:19.13bluOxigen<[TK]D-Fender><p3nguin>  thx mate, it worked,  since the /etc/asterisk folder was empty i copyied files from the backfolder and it all worked
16:19.31bluOxigenthx for your cooperation and support
16:19.49bluOxigen<PROTECTED>
16:20.13p3nguinI think the chronic changer has departed.
16:20.23*** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se)
16:21.28*** part/#asterisk AsteriskNoob (~AsteriskN@host86-128-21-197.range86-128.btcentralplus.com)
16:22.03yahhWhy PRI span shows "Standby"  status?
16:22.21*** join/#asterisk Knightfal (~j@mailer.1callres.com)
16:23.12*** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com)
16:28.54KnightfalHow can I keep Asterisk from honoring blocked callerid ID. I have a customer that sends me calls that have a 4 digit CID If the poriginal caller has their CID blocked I get anonymous in my endpoints CID and I need to pass the 4 digits alond to them.
16:29.29Knightfal....lots of typos, its early.
16:29.34*** join/#asterisk Tim_Toady (~moi@193.92.244.148.dsl.dyn.forthnet.gr)
16:30.30russellbit's
16:31.47[TK]D-FenderKnightfal: "show application setcallerpres"
16:32.32*** join/#asterisk diegomad (~mad@190.146.200.120)
16:32.53*** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen)
16:34.39*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
16:35.48*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)
16:37.35*** join/#asterisk mcrownover (~markcrown@remote.gawest.com)
16:38.11*** join/#asterisk korcan (~kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net)
16:38.55*** join/#asterisk kannan (~service@122.165.66.32)
16:45.01guaxleifmadsen, do you think paid support in what level would justify a review of the transfer case?
16:45.12KnightfalThx TK
16:49.53p3nguinMust be a four-wheel-drive if it has a transfer case.
16:50.53Dandanwith an LSD!
16:51.23p3nguinonly in the rear.
16:51.38p3nguin(air locker on the front)
16:51.54bent_screwdriver[TK]D-Fender: Knightfal: wow that is weird i was just about to post a question about why my polycom endpoints anonymous sometimes, even when i always pre-append the caller id name with some text. this fixes that as well?
16:52.25DandanHm, not a bad design...
16:53.47*** join/#asterisk cusco (~trilili@213.63.137.210)
16:54.05cuscowhat can cause "chan_iax2.c: Max retries exceeded to host 192.168.2.7 on IAX2/gateway-iax-670 (type = 6, subclass = 2, ts=1411269, seqno=170)"
16:54.08cusco?
16:56.10cuscoand manager connection keeps getting "broken pipe"
16:56.13cuscocan't figure why
16:56.51[TK]D-Fenderbent_screwdriver: Presentation flag follows
16:57.54yahhplease someone can help me....
16:57.56yahhWhy PRI span shows "Standby"  status?
16:58.35cusco[TK]D-Fender: by the way the re-invite is working flawlessly now :-)
17:00.13*** join/#asterisk gazzerh (~garryh@93-97-187-150.zone5.bethere.co.uk)
17:03.11bent_screwdriver[TK]D-Fender: so if i set SetCallerPres(allowed) before I pre-append cid(name) it will then let it pass to the phone? this is on incoming pri calls.
17:03.49[TK]D-Fenderbent_screwdriver: did you LOOK and TRY?
17:03.55Kattyherroes
17:04.25*** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com)
17:04.32carrarHARRO
17:05.55Kattycarrar: shall we...
17:05.55Kattycarrar: hug?
17:06.40*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
17:06.51bent_screwdriver[TK]D-Fender: looked, but didn't try yet as i don't really understand what the different options do, and not much out there as far as exapmples. at least for what i am trying to do. i guess i'll do some testing and see if i can get it to work. thx tk
17:07.46Kattyanyone know of a good dvd to avi software for winders
17:07.54carraryeah yeah
17:08.00carrarhuggles kattyrooo
17:08.10Kattyhugs carrar
17:09.45beekbeek hugs katty
17:10.39*** join/#asterisk t_dot_zilla (~t_dot_zil@firewall-a.buf.ny.i-evolve.net)
17:11.26Kattyhugs beek
17:11.27ChannelZWhat does 'sip set history' do?
17:11.35Dandananyone following Steve the Black-Turtleneck-And-Jeans guy?
17:12.35ChannelZor rather, how do I get * to forget about a peer showing up in 'sip show peers' ?
17:12.53Kattyyou mean steve jobs?
17:12.54Kattyat apple?
17:12.56pacmanfansip reload maybe?
17:12.58ChannelZI was playing around with defaultusername and such and now it's being remembered by asterisk even though it's no longer in the configs at all
17:13.04kannanhello, i am having trouble with a Sangoma e1 A102 (2 span E1 Card) installation, the install seems OK, then the server freezes up. The line works OK on a Diigium E1 Card tho
17:13.05DandanKatty, yeas and his ongoing show
17:13.07Dandan(ATM)
17:13.10Kattyno
17:13.17ChannelZNot doing it.. I deleted the entry out of the database too, but it still shows up
17:13.27DandanChannelZ: did you reload sip channel?
17:13.43kannani used ubuntu 8.04 server edition, and wanpipe 3.5.11
17:13.46Kattyhe's a looney
17:14.15DandanKatty: indeed
17:14.16spiceycurryBased on these settings, could someone give me an example of an extensions.conf to test a simple dial?  Using Dahdi/g1 Here are my conf files.  system.conf - http://pastebin.com/3RdkM7Aq, dahdi-channels.conf - http://pastebin.com/UdYYgRSL, chan_dahdi.conf - http://pastebin.com/QVahTnNP, sip.conf - http://pastebin.com/defMgqMT
17:14.19Kattybeek: http://www.recipezaar.com/recipe/Simply-Sour-Cream-Chicken-Enchiladas-129926 <- what do you think?
17:14.34ChannelZDandan: I restarted asterisk entirely... about to try again in a sec once someone gets off a call
17:15.02Dandankannan: hard to tell based on your short description... did you try sangoma tech support?
17:15.05pacmanfanis suddenly hungry for Thai red curry... or yellow curry... or green curry... or Masaman curry. heck, i'll take any Thai curry.
17:15.06Dandanthey are great
17:15.22beekKatty: Looks tasty
17:15.23kannanDandan , ok thanks i just sent them an email,
17:15.29Dandanpacmanfan: go play: www.google.com/pacman :P
17:15.39pacmanfanalready have. :D
17:15.48pacmanfanold as the hills, yo.
17:15.58Dandango play some more :P
17:16.10kannanDandan, just wanted to kmnow if anyone faced similar issue, and if so, how they may have resolved it
17:16.11pacmanfank, bbl
17:16.36Dandankannan: I have much older wanpipe and no, I had no isses with it...
17:16.41ChannelZOk now it's finally gone.  Sheesh.
17:16.42Dandanso yes, it is unusual...
17:17.03kannani had previously installed 3 other cards with no issues at all...
17:17.06ChannelZI think it was pulling out of the astdb before, I didn't clear everything in the right order
17:19.03Dandankannan: that would be your answer: it might be a defective card, try switching PCI slots?
17:19.32Dandan(unfortunately, Sangomas are known for being very good players with PCIs and IRQs)
17:19.49*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
17:21.42kannanDandan, yes i tried switching the slots.. after the servers freezes , i have to cold reboot, then when i try to upgrade wanpipe setup, it says no Card detected. But if i remove the card and reboot , and then do a proper shutdown and fix the card, the wanpipe gets installed fine. The Card LED always REd, and then the server freezes up again.
17:22.11Dandando you have any spare card?
17:22.21Dandans/any/a/
17:22.52kannanyes, but it is a PCI , not a pci-E
17:23.14*** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com)
17:23.52kannanthe PCI card (Digium te122) works fine
17:24.31*** join/#asterisk Ad-Hoc (~nimbus@62.1.137.237.dsl.dyn.forthnet.gr)
17:24.36kannanDandan , like you said, Sangoma seems to have greta support (they have sent a detail mail to me already)...
17:24.43Dandan:)
17:24.49DandanI am glad it is working for you
17:25.06DandanI think I still know a few people working there
17:25.11leifmadsenguax: I am not familiar with digium's open source subscription model enough to know if that would advance the case or not. I believe that is really a fundamental way of how Asteriks works
17:25.16Dandanincluding the guy responsible for writing wanpipe :)
17:25.21Dandansoftware that is
17:25.55russellbbut of course you'd rather buy from the company that makes asterisk
17:26.02[TK]D-FenderYeah I've dealt with Nenad.... guy is a God
17:26.25guaxleifmadsen, ok, thanks again.
17:26.33Corydon76-lapwould be nice if the guys behind wanpipe would code up to standards, so that the driver could be integrated
17:26.43*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
17:27.00Corydon76-lapAs of right now, you cannot have a Sangoma card and a Digium card in the same machine and expect both to work fine
17:28.13kannanrussellb, i so far got only digium cards, but on twice had a problem with HDLC aborts when used as a PRI NET to connect to a legacy pbx
17:28.16*** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca)
17:28.43kannansince that was the scenario, i now tried sangoma, (used 3 before with no probs) but stucjk suddenly here
17:29.08russellbkannan: If you run into a problem, we guarantee that it will be fixed.  :-)
17:29.47kannanCorydon76-lap, really, i did not know that, however , i re-installed from the OS , and removed the Digium card now , so i am OK, but good learning
17:30.50Dandaninteresting, did not know, I have a sangoma s514 (T-1) and a few tormentas together
17:30.52kannanrussellb, i know ... you guys atDigium were great so far
17:30.54Dandanafair
17:30.58*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)
17:31.14Corydon76-lapDandan: which tormenta card?
17:31.17Dandan2
17:31.23DandanTormenta2
17:31.50Corydon76-lapDandan: that's a fairly old card; the current cards actually run quite a bit faster
17:32.06Dandanno need to have it run faster
17:32.14Dandanit does its job just fine :)
17:32.31Corydon76-lapI think someone said that the card took 1/7th of the time when it switched from bus slaving to bus mastering
17:32.37Dandanpeople irc as root? sheeeesh...
17:32.42cuscocan IAX2 use some equivalent of SIP's re-invite to establish direct comunication for audio between end points?
17:32.50*** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net)
17:33.03Corydon76-lapcusco: yes
17:33.27Dandandisables iax2 along with skinny, mgcp and h323 first thing after install :-D
17:33.28cuscoCorydon76-lap: is dere a directive to set it?
17:33.31p3nguindandan: Yes, there are stupid people on IRC, contrary to popular belief.
17:34.28Corydon76-lapcusco: transfer=mediaonly
17:34.29Dandanheh :-/
17:34.32[TK]D-Fendercusco: TRANSFER=YES
17:34.49Corydon76-lap[TK]D-Fender: that transfers control streams, as well
17:35.10iscariohi, i encountered trouble to compile dahdi-tools on my openbsd... (got a missing ldhc somewhere) Do you know if it is possible to compile it on openbsd ? i would have liked meetMe that's why i would have needed it.
17:35.40[TK]D-Fenderiscario: use ConfBridge instead.... make your life easier
17:36.01Corydon76-lapiscario: it probably would need to be ported.  The work was done for FreeBSD, so it's decently close
17:36.05*** join/#asterisk fors1 (~forsen@pat-tdc.opera.com)
17:37.00iscario[TK]D-Fender: i'll have a look then
17:37.19Dandandon't forget to give back to the community :)
17:37.22p3nguinYay! My 2TB disk just arrived!
17:38.00iscarioCorydon76-lap: mmmh i'll have a look if confbridge doesn't save me :D
17:38.38Dandanp3nguin: I have nothing but problems with my controllers and 2TB disks :/
17:38.50p3nguinick
17:40.54*** join/#asterisk fors1 (~forsen@pat-tdc.opera.com)
17:40.56DandanI am wondering how much manual configuration would I have to do to get pre-1.0 .conf files to work with 1.6 :>
17:41.32Corydon76-lapProbably quite a lot
17:41.45Dandanyou are not helping here Corydon76-lap :-/
17:42.08Corydon76-lapThere are a large number of apps which have been converted to dialplan functions, setvar has been renamed to set, application delimiters are now comma alone
17:42.58*** join/#asterisk zyphlar (~z@wsip-70-182-59-230.ph.ph.cox.net)
17:43.42Dandanmeh, I got also a custom announce_queue application as a hack in the source code :/
17:43.48zyphlarhello fellow technologists!
17:44.04Dandanwhat would you want from us?
17:44.11zyphlaroh just comradery
17:44.34Dandangood, then sit down, have a beer (or two)
17:44.38zyphlarand a question about asterisknow's web interface not working from anything but localhost even though iptables is stopped :)
17:44.48zyphlaroh man a beer sounds good
17:46.18Dandanzyphlar: check if apache listens on 0.0.0.0 or ::?
17:46.23Dandannetstat -anep |grep apache
17:47.48zyphlarthanks; had to use | grep apache and it's reporting tcp 0 0 :::80 :::* LISTEN 0 7642 2461/httpd
17:47.56zyphlarso looks like there is a listening issue
17:48.04zyphlarsilly network config scripts
17:48.14Dandananything in /var/log/apache2/error.log?
17:48.18Dandanor iptables -v -L
17:48.18Dandan?
17:48.48zyphlarlogs and iptables has nothing
17:48.57zyphlari stopped iptables though
17:49.03Dandanmaybe your provider blocks :80?
17:49.35zyphlarnah it's all local
17:49.41zyphlaris :::80 correct?
17:49.57Dandanyeah, awkward though
17:50.13Dandan[::]:80 is preferred I thought...
17:50.42zyphlarthe initial setup was slightly more kludgy than the guide makes it seem
17:51.42zyphlaris there something i should be doing after setup, with the config menus and such?
17:52.10zyphlaror maybe i gotta do an MD5 check on this iso
17:54.01*** join/#asterisk btsteve (~tstevens@nat/google/x-hribaftraardkrge)
17:54.22DandanI've never seen *now in my life
17:54.37DandanI consider it to be a... no-product :)
17:55.01zyphlarfunny how "linux made simple" ends up being like that ;)
17:55.45zyphlarno! you must deal with the minute variations between package versions and *nix flavors before you can get a SIP box! bwahaha
17:55.46DandanI would start tcpdump on port 80 and see if you actually get any requests from your worksation because I think it is your workstation that might have a problem
17:57.26zyphlarbtw asterisknow is based on centos 5.5
17:58.13zyphlari notice you're using ubuntu-ish filenames so the centos difference might be doing things (though i'm pretty sure i got rid of selinux)
17:58.51*** join/#asterisk lucasb (~lbussey@24.67.33.55)
17:59.05*** join/#asterisk Alagar (~Administr@122.164.38.246)
17:59.09*** join/#asterisk sourcode (~code@ppp-58-8-85-193.revip2.asianet.co.th)
17:59.28zyphlari am seeing packets on port 80 arriving and being sent back, might be an httpd issue
18:04.32*** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org)
18:05.43zyphlarwould tcpdump show those packets going back and forth if there were a firewall or routing issue? it's only sending back a single tcp packet or two
18:05.56Dandanyou would not see a response
18:06.00Dandanusually just a drop?
18:06.04zyphlarmaking me think httpd isn't set up right, but it looks fine to me
18:06.19zyphlarrunning tcpdump on the server side, i can run it on client side as well
18:21.00*** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com)
18:22.01*** join/#asterisk btsteve (~tstevens@204.10.23.130)
18:23.29zyphlaryeah i'm not receiving any packets from asterisk
18:23.45zyphlarbut the connection is recognized in netstat, status is SYN_RECV
18:23.55zyphlarfor httpd and sshd
18:24.18*** join/#asterisk SonOfGods (~asd@77-85-130-177.btc-net.bg)
18:25.42*** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca)
18:25.43SonOfGodscan someone help me
18:25.45SonOfGodsplease
18:25.53Skeeter-ariel_, hey
18:26.00SonOfGodsI'm new at asterix :P
18:26.12ariel_hi Skeeter-
18:26.14zyphlari am too, Dandan was helping me but he seems afk
18:26.22beek~ask
18:26.45infobotQuestions 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.
18:26.45*** join/#asterisk txwikinger2 (~quassel@sblug/member/txwikinger)
18:26.45Skeeter-ariel_, i have a little issie with the Spectralinks
18:26.46[TK]D-FenderSonOfGods: First, it's "Asterisk", second this i the suppotr channel for FreePBX.  And help to tell us WHAT you need help with specifically.
18:26.46zyphlarmy problem is with asterisknow not responding to any outside packets
18:27.01beekWow -- the 'bot is really slow today.
18:27.03[TK]D-FenderOOP
18:27.06[TK]D-Fenderdangit.
18:27.12Skeeter-ariel_, 1 of my clients get the voicemail icon about 24 hours after the voicemail gets into the inbox, but on the 550 and 650, i gets on the screen automaticly
18:27.15[TK]D-FenderSystem spaz and a channel junmp
18:27.24[TK]D-FenderSonOfGods: Ok, 2nd part still applies!
18:27.53[TK]D-FenderSonOfGods: So... whats the problem?
18:27.54ariel_Skeeter-: seems phone is then not correctly registered to the Asterisk system
18:28.07ariel_Asterisk only sends the notification when the vm comes in
18:28.32Skeeter-ariel_, any ways to send a push notification or sth
18:28.37ariel_Spectralinks are passive they don't poll for vm
18:28.42Skeeter-sees [TK]D-Fender
18:28.58Skeeter-ariel_, do you have any ideas?
18:29.11*** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty)
18:29.11ariel_Skeeter-: you can setup the other vm system in asterisk to a push
18:29.17Skeeter-missed his emote
18:29.44SonOfGodsI need call center that calls to thousand numbers and hang-up after 1 signal
18:29.44Skeeter-ariel_, that is out of my range capability
18:29.47*** join/#asterisk guilhermebr (~Guilherme@189.27.61.94.dynamic.adsl.gvt.net.br)
18:29.55SonOfGodsthen recieve calls and play a message and hang-up again
18:29.57SonOfGodsis this possible?
18:30.04ariel_Skeeter-: only real way is using a push system
18:30.09[TK]D-FendersonsYes
18:30.17[TK]D-FenderSonOfGods: Yes
18:30.20Skeeter-ariel_, have you ever encounter that problem?
18:30.26SonOfGodsexcellent!
18:30.27SonOfGods:))
18:30.38SonOfGodsI'll download the source now
18:30.42Skeeter-SonOfGods, sounds like spamming
18:30.48ariel_yes I have, everyone that has a passive phone will run into this issue
18:31.00Skeeter-ariel_, what do you tell them
18:31.09SonOfGodsno, it's only to my clients
18:31.16SonOfGodsthey're informed about the call
18:31.28ariel_switch to a different setup and asterisk does have a new vm system you can configure
18:32.45Skeeter-ariel_,  i cant tell them that
18:33.39ariel_Skeeter-: I am on a support call lets talk about this later.  I have no real response other then you can set it up via normal asterisk.
18:34.01*** join/#asterisk rocksfrow (~kyle@pool-96-235-66-170.atl01.dsl-w.verizon.net)
18:34.07ariel_it's a bit harder to set this up via Freepbx
18:34.14*** part/#asterisk rocksfrow (~kyle@pool-96-235-66-170.atl01.dsl-w.verizon.net)
18:34.36Skeeter-ariel_, that again i was afraid
18:35.05zyphlarattention: i am an idiot! my subnetting was wrong, one too many .255's means the server got my packets but couldn't send any back...
18:36.51[TK]D-Fenderzyphlar: 11 steps to go :p
18:37.14zyphlarof my 12-step self help program?
18:37.28SonOfGods[TK]D-Fender: there is no version of Asterisk for Windows, right?
18:37.35zyphlarhehehe
18:37.38SonOfGods:D
18:37.43[TK]D-FenderSonOfGods: Officially no.
18:38.03*** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com)
18:38.07SonOfGodsokay, I'll search, maybe there is something
18:38.57*** join/#asterisk Hellmark[S10e] (~hellmark@75-121-89-69.dyn.centurytel.net)
18:38.57zyphlarSonOfGods: you're probably going to find it easier to use AsteriskNow or Trixbox than a kludgy setup on Windows or in a VM.
18:39.46SonOfGodsthanks!!
18:39.47[TK]D-FenderSonOfGods: Expect no support for whatever you find
18:40.06*** join/#asterisk txwikinger2 (~quassel@sblug/member/txwikinger)
18:40.18SonOfGodsbut AsteriskNOW is official version I think :))
18:40.25KavanSSonOfGods, if you are truly the son of god, thank you for single-handedly porting asterisk to DOS 6.2
18:40.38Hellmark[S10e]Hey. Thinking about getting an asterisk setup with chansccpb, and was curious about extension management.
18:40.51SonOfGodshehe, that's not my nickname :))
18:41.40zyphlarHristoTorbov: if you want official paid support, check out: http://www.digium.com/en/supportcenter/asterisk.php
18:42.00*** join/#asterisk korcan (~kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net)
18:42.05Skeeter-iPhone 4 is amazing!
18:42.06zyphlarit's well worth the money if you're planning to run your business off an Asterisk box
18:42.33Hellmark[S10e]We process a lot of phones here, testing and refurbing them, and currently we have cisco routers with a batch script to clear the extensions every so often, but was wondering if something like that would be possible with asterisk, or just have asterisk keep assigning extensions and reusing them on its own
18:42.51KavanSHellmark[S10e], I'm sure you could automate something like that
18:43.27Hellmark[S10e]Just was wondering how I could go about doing something like that.
18:45.09Hellmark[S10e]my experience with asterisk is somewhat limited.
18:46.12*** join/#asterisk JT (~j@unaffiliated/jt)
18:49.54leifmadsenAsterisk 1.6.2.9-rc2 is now available. See the release announcement for more information: http://www.asterisk.org/node/51365
18:51.59Hellmark[S10e]Or at least, as is, how can I manually clear what phone is assigned what extension?
18:52.58*** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl)
18:55.13cuscohi... this is worrying
18:55.29cuscoasterisk cli shows "BROKEN PIPE" on manager connections
18:55.47cuscowe have a java software that monitores manager events to show queue/peers etc
18:56.13cuscoand we have about 10 of them connected for more than a year
18:56.30cuscoocasionally we would se some "broken pipe" messages on asterisk CLI
18:57.00cuscobut now, its like 300 messages.. softphone dialing shows "establishing call" and waits some seconds before those warnings finish printing
18:58.17cuscohttp://paste.debian.net/76525/
19:05.31*** join/#asterisk fofware (~fabian@190.7.25.160)
19:09.18ChannelZyour manager app either quit or quit reading from it's stdin anyway
19:10.35Dandanhow do you graph realtime rrd? simply? do I need to write some manager interface?
19:12.00*** part/#asterisk kannan (~service@122.165.66.32)
19:12.12*** join/#asterisk Beave (~beave@DCC.SEND.startkeylogger.000.telephreak.org)
19:12.24*** join/#asterisk jstapleton (~jstapleto@173.15.197.73)
19:16.38ChannelZs/from it's stdin/from its stdin or socket/
19:16.38*** join/#asterisk imox1234 (~imox1234@p4FC5C509.dip0.t-ipconnect.de)
19:16.41imox1234hello, how can i manage multi call with a asterisk calling card ?
19:16.58ChannelZWhoa, they're selling asterisk calling cards now?
19:17.22zyphlarand asterisk jelly beans
19:17.38imox1234dont sell :D
19:17.39zyphlarnow with FXO, FXS, and DAHDI flavors!
19:18.30[TK]D-FenderThere is no "Asterisk Calling Card".
19:18.40cuscoChannelZ: sorry I did not understand if you were persuading me to change the beaviour of out app
19:19.04zyphlarimox1234: I'm just a noob, but I think you need to use better and more technically-accurate English in order for us to help you
19:19.32imox1234ok
19:19.38imox1234moment
19:20.02cuscowe are using the asterisk-java.org class and its also having some warnings  http://paste.debian.net/76528/
19:20.03zyphlarshocked, that might have worked
19:20.27*** join/#asterisk imox1234 (~imox1234@p4FC5C509.dip0.t-ipconnect.de)
19:21.13*** join/#asterisk dkirker-openmo-1 (~dkirker@openmobl/ceo/dkirker)
19:21.23cuscoguess he borught his translator :P
19:21.30imox1234hello everyone.. I am searching for a easy to use prepaid solution for asterisk which enables multiple calls on the same callingcard
19:21.47cuscowhat do you mean by calling card?
19:21.55zyphlarimox1234: so you want to buy VOIP service from someone like Vonage?
19:21.57[TK]D-Fenderimox1234: * has nothing to do with calling cards
19:22.09[TK]D-Fenderimox1234: You can call out any device or conenction you set up.
19:22.37[TK]D-Fenderconnection*
19:23.30imox1234each of my "user" can get one ore more peers, so it should be enabled that he can make simultaneous calls with the same"calling card", i found a solution which gives each user a calling card, but blocks it if a call is started (I think because it is much more complex to think of a solution with multiple calls)
19:26.20*** join/#asterisk Z_God (~julius@wlan226184.mobiel.utwente.nl)
19:27.00*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
19:27.17zyphlarcan i quit my job and do imox1234's job instead? i wonder if he'll let me drink his beer
19:29.35*** part/#asterisk nova911 (~Adium@122.182.0.38)
19:29.53cuscohow about "no more connections allowed" ?? http://paste.debian.net/76529/
19:42.46*** join/#asterisk rrb3942 (~rbullock@208.34.105.161)
19:45.59*** join/#asterisk sysop1 (~gerrym@gw.futuremetals.com)
19:46.14*** part/#asterisk sysop1 (~gerrym@gw.futuremetals.com)
19:46.50*** join/#asterisk knot (yiffstar66@unaffiliated/devemo)
19:48.15*** join/#asterisk Faithful (~Faithful@202.6.145.116)
19:50.47Skeeter-ariel_, still on support call?
19:51.31*** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net)
19:52.18*** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se)
19:56.55*** join/#asterisk QaDeS (~mklaus@p54A1B176.dip0.t-ipconnect.de)
19:57.28ChannelZcusco: these all sound like self-explanatory error messages to me.  What does 'manager show connected' say?
20:01.32*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
20:04.33*** join/#asterisk t_dot_zilla (~t_dot_zil@firewall-a.buf.ny.i-evolve.net)
20:04.42cuscohttp://paste.debian.net/76534/ help!!
20:04.58cuscoChannelZ: 0 users connected.
20:05.07cuscoI disabled manager
20:05.11cusco(I guess)
20:06.18cuscoI have 0 users connected
20:07.17cuscoim going to enable manager agai
20:07.18cusco:/
20:08.28UberDuperIs there any way to see a list of applications that are available in my installation?
20:08.43russellb*CLI> core show applications
20:09.00UberDuperExcellent. Thank you.
20:09.01russellb*CLI> core show application <foo>
20:10.48UberDuper49 applications in this installation.
20:11.00UberDuperI'm using some 1.6.2.6 packages on an openwrt router.
20:16.07*** join/#asterisk Mw3_ (mw3@mw3.hu)
20:21.39UberDuperHrmm. I don't see any kind of blacklist or filter application in this install.
20:24.54*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw)
20:29.23*** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com)
20:30.30wdoekes2UberDuper: the applications you see are lower-level than that
20:33.02UberDuperI guess I just have to build modules for this platform.
20:33.33Hoianyone here?
20:33.51Hoianyone tell me if theres something wrong with asterisk.org forum sign up?
20:34.24Hoicos everytime i try to sign up, it doesn't send no email confirmation that its been created and nor can i login since it doesn't prompt for password when creating a new account
20:36.39Tim_ToadyUberDuper or you can just add blacklisting capabilities in dialplan, its prety simple and maybe better when you run in embeded than to load an extra module
20:38.36*** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
20:38.59*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
20:39.41*** join/#asterisk hansin (~chatzilla@c-76-120-68-209.hsd1.co.comcast.net)
20:40.51*** join/#asterisk uqlev (~yuriy@91.184.221.31)
20:41.51*** join/#asterisk boodu (~antoine@175.158.129.128)
20:42.46boodulol
20:42.51boodu*lo
20:47.21*** join/#asterisk CrazyYoss (~luther@c-98-207-228-123.hsd1.ca.comcast.net)
20:58.58*** join/#asterisk bsaxon (~bsaxon@12.107.149.61)
21:01.36*** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)
21:02.38*** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com)
21:03.20*** join/#asterisk nickaugust (~nick@rrcs-97-79-25-103.se.biz.rr.com)
21:03.22Micc_I hate that my cdrs are all messed up. in dst is usually s
21:03.32nickaugusthey can I just buy the board inside a polycom phone?  like lets say I opened one up and broke something :)  or do i have to buy the whole phone?
21:03.44Naikroveklol
21:03.47Naikroveknickaugust: whole phone probably
21:03.55Naikrovekyou trying to make a PA speaker or something?
21:04.17nickaugustNaikrovek: no just dickin around mainly :)
21:04.23Naikrovekheh
21:04.36Naikrovekfind a polycom reseller and ask them what you can do
21:11.21*** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com)
21:13.39nickaugustis it possible to power a polycom from a laptop?  i know there is the power over ethernet option but i dont think usb would provide enough power right?
21:16.59ChannelZor buy a shit one off ebay and gut it for parts
21:17.29nickaugusthm good idea
21:22.32*** join/#asterisk mandragor (~ergudicsu@70.158.116.40)
21:22.35*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
21:26.42leifmadsenMicc_: what is the extension that executed the Dial()?
21:26.51leifmadsenif you're using a Macro() to call someone, then the dst is likely to be 's'
21:27.07*** part/#asterisk rrb3942 (~rbullock@208.34.105.161)
21:27.27Micc_leifmadsen, yeah most of my dialplan is macros or subs.
21:27.41Micc_leifmadsen, is there something I can set to tell the CDR what the dst should be?
21:27.57leifmadsenMicc_: ya, execute Dial() from the extension you want it to be
21:28.07leifmadseni.e. use a Goto() which performs your Dial()
21:28.12Micc_leifmadsen, I want it to be the phone number that was called on the inbound, not the number I dial on the other side.
21:28.19*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
21:28.21leifmadsensee above
21:28.26leifmadsenasterisk doesn't care about that
21:28.38leifmadsenit only cares about what the extension was that executed the Dial()
21:29.06Micc_yes, I always use goto.
21:29.06leifmadsenin your Macro() just make a pattern match that you can Goto() which performs the Dial()
21:29.54Micc_so the extension that performs the dial is the number that shows in the CDR? Thats going to make my dialplan very messy.
21:31.00Micc_It also gets complicated because of parking. the context the dial is in is the context that gets called back when the park times out.
21:31.09Micc_Which wouldn't be a problem here I guess.
21:31.37Micc_I do a lot of stuff before and after the dial.
21:31.56Micc_would a GoSub work with a return?
21:32.12*** part/#asterisk nickaugust (~nick@rrcs-97-79-25-103.se.biz.rr.com)
21:32.49Micc_Sometimes there is no dial at all.
21:33.04Micc_I wish there was a way I could just set what the dst should be.
21:33.36leifmadsenLike I said, afaik it is the extension that was last executed that is the value of 'dst'
21:33.45leifmadsenyou may or may not be able to modify that with the CDR() function
21:33.48Micc_I'm thinking of just hacking the source to make dst writable. And only set it when I really need to.
21:34.00leifmadsenSet(CDR(dst)=foo)
21:34.03leifmadsennot idea if that works or not
21:34.07leifmadsendinner!
21:34.12Micc_right, it says read only right now.
21:34.22Micc_but even if it was writable may or may not work.
21:34.42Micc_because it probably gets reset when it sends it to the database anyways.
21:35.14Micc_I know what I need to do. I just need a custom cdr field that I can set.
21:36.12Micc_I may want that dst value for something else anyways. but I do need to know for billing what the inbound number was.
21:42.12*** join/#asterisk dkirker-openmo-1 (~dkirker@openmobl/ceo/dkirker)
21:44.29*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
21:50.19*** join/#asterisk nickaugust (~nick@rrcs-97-79-25-103.se.biz.rr.com)
21:50.22*** join/#asterisk rbd (~rbd@rrcs-98-101-33-14.midsouth.biz.rr.com)
21:51.27rbdhey guys...I have an issue where any system prompts, and any .wav prompts that are encoded as ulaw fail to play. however, prompts encoded as PCM wav at mono 8KHz play fine
21:51.33nickaugustok so if I connect my polycom to my laptop's ethernet and do internet sharing from the wirless card do you think i'll get good sound quaility?  providing the wireless internet has a good interent connection?
21:53.18*** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net)
21:54.39voxterany of you guys know of a reliable 'sip ping' tool i can use from linux commandline?
21:54.47voxterpreferrably something that does an OPTIONS packet
21:57.12*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
22:03.25cuscovoxter: sipp - its not simple tho
22:03.44cuscohttp://sipp.sourceforge.net/
22:03.45voxterya i suppose that would work.
22:03.48voxterIve used it plenty
22:03.58voxterjust a bit heavy for testing, im looking to get output data and graph it.
22:04.16cuscowrite numbers depending on stdout
22:04.19cuscolol
22:04.42cuscoI tried to fiddle with sipp in the past and wasn't very sucessfull, so I can't help you much
22:08.19*** join/#asterisk DelphiWorld (~Miranda@41.200.10.114)
22:10.24*** join/#asterisk RobH (~robh@wikimedia/RobH)
22:10.55*** part/#asterisk DelphiWorld (~Miranda@41.200.10.114)
22:14.23*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
22:17.50mandragorhello is there a way look at the asterisk logs and see what queue a call went through?
22:18.08*** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com)
22:18.25*** join/#asterisk Benwa (~Benwa@dyn.144-85-187-036.dsl.vtx.ch)
22:23.05hansinI want to get a VoIP phone number for home. I have Polycom Soundpoint 500 phone I can use. I want to be able to either runt he phone directly connected to a provider or put Asterisk running from home between the phone and the VoIP provider. I'd like to be able to do either and switch as desired. Not sure what the correct term I am looking for. I really just want a single home line. Any...
22:23.07hansin...suggestions on what I need to search for? Thanks.
22:23.48*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
22:26.53xhelioxhansin: Well those are two entirely different things. :)
22:28.21xhelioxhansin: Setting up a Polycom with a SIP provider should be a no brainer. You can use the web interface and simply configure the phone for whatever provider you use. Assuming you're behind a firewall (NAT), you might need to forward some ports and tell your SIP provider you're using NAT.
22:30.57*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
22:34.52*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
22:36.46*** join/#asterisk Whtsup (~sssi@119.63.136.50)
22:36.49Whtsuphello
22:37.10Whtsupanyone alive hea ..
22:37.11*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
22:39.06Whtsup???
22:39.10Whtsupneed help
22:39.23Trixboxeryup
22:39.28Trixboxerwhtsup
22:39.49Whtsupi cannot hear caller tune of dialled no
22:40.13Whtsupi just hear default tone of ring ring on every call
22:40.24Whtsupwhen i dialled from pc dialer
22:43.06*** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net)
22:46.59*** join/#asterisk ketema (~ketema@ketema.net)
22:49.32*** part/#asterisk ketema (~ketema@ketema.net)
22:54.14*** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33)
22:57.22*** join/#asterisk corretico (~laguilar@201.201.46.106)
23:03.05*** join/#asterisk jks (jks@193.189.93.254)
23:03.09*** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net)
23:04.26hansinxheliox: I was hoping I could get a provider where I could use the phone alone, or through Asterisk in between. So if a VoIP provider is what I need for just using the phone, what would I need to have VoIP service where I can use Asterisk. I once had * running with the X101 card to my POTS line. What would be the equivalent for VoIP. I would need some service that could provide gateways to...
23:04.26hansin...POTS and all that right? I guess I figured a provider would provide that.
23:12.56*** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com)
23:15.26*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
23:21.42carrarlooks at Katty
23:22.17zyphlarhansin: still looking for an answer?
23:23.17*** join/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net)
23:24.34hansinI think I get it now. PBX hosted service for just the phone, and SIP trunk to use with Asterisk. Though sounds like I can make * look like a phone if I get PBX hosted service.
23:25.13zyphlarI'm not sure about the details, but I'm sure it's possible
23:25.40zyphlarmany SIP services offer support for such things
23:25.48zyphlaror at least how-to guides
23:26.09hansinzyphlar: Okay, thanks. I've already learned a bunch this last half hour. I should be okay from here, at least much more informed after I do my more in-depth homework.
23:26.12k-mananyone here having problems registering with mynetfone?
23:27.22ChannelZhansin: there isn't really much difference between a phone and what you call a trunk (there really is no such thing as a "sip trunk" but that's another semantic argument)
23:29.32*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
23:29.42mianosk-man, mynetfone is up for me
23:29.50mianosyou mean connection or sip settings?
23:30.47hansinChannelZ: Yeah, that kind of makes sense. I suppose though with a trunk they just don't give you voice mail and all the rest. But otherwise the same. I could see that.
23:30.50k-manmianos: yeah, its up for me too at the moment, but i keep getting this issue where i can't register, i get no response from their servers, however if i change to a different server (sip10 instead of sip01) it works for a while, then last night that stopped too, then i changed it to sip20 (the next ip) and now thats working for the moment
23:31.23mianos[mynetfone]
23:31.24mianoshost = sip01.mynetfone.com.au
23:31.33ChannelZhansin: but even that is a function of what services you buy/setup through the ITSP
23:32.07mianosit's been up 100% for me
23:32.19mianospennytel missed a few regos
23:32.21p3nguinhansin: VoIP.ms offers all the features that asterisk would provide for your own on-site PBX.
23:32.49p3nguinThey allow you to use phones directly to them or use a PBX in between.
23:32.53k-manmianos: ok, well maybe its me or my router then
23:32.58p3nguin~itsp
23:32.59infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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.
23:32.59mianosmaybe
23:33.01k-mani wonder if my router is doing something weird
23:33.06carrarit is
23:33.10carrarI'm logged into it
23:33.15carrarhacker your cats
23:33.21carrarhacken'
23:34.20*** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl)
23:35.07*** part/#asterisk Scorcerer (scor@czlug.icis.pcz.pl)
23:35.31*** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com)
23:37.23hansinp3nguin: Thanks. Good info.
23:40.09*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
23:44.21*** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com)
23:46.07*** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek)
23:46.27*** join/#asterisk warewolf (warewolf@warewolf.org)
23:46.50warewolfevening folks
23:47.42*** join/#asterisk Ta^3 (~tacvbo@189.146.184.125)
23:48.31warewolfI've got a TDM410P card w/ two each FXO and FXS cards; analog phones plugged into the FXO ports get dialtone, but as soon as I press any number on the phone I get a congestion tone
23:48.53warewolfI think something is jacked up in my dialplan, but I can't figure out how to get enough debugging info out of asterisk
23:49.07warewolfI see simple switch, then hungup in the debug logs
23:51.05ChannelZcore set verbose 4
23:51.50ChannelZand make sure you have 'notice' turned on in logger.conf for 'console'
23:51.54*** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net)
23:52.04*** join/#asterisk kam187 (~kam187@81-179-8-102.dsl.pipex.com)
23:52.08kam187hey guys
23:52.23kam187is there an option for dial() on how it chooses a line in a group?
23:52.27kam187eg round robin etc
23:52.29ChannelZbut yes it sounds like your dialplan is not good (check your context on those two channels)
23:53.06warewolfChannelZ: ooh, pointing out notice/etc in console helps
23:53.27ChannelZg1 goes forward through the group, G1 goes backwards.  I don't think there is RR...
23:53.43ChannelZOh wait, there is!
23:53.46ChannelZr1, R1, etc
23:54.17ChannelZ(r being forwards, R being backwards)
23:54.50ChannelZlearn sumfin new every day..
23:54.54*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
23:55.29kam187great :)
23:56.09*** part/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net)
23:56.48kam187hmm it didnt like r1
23:57.14ChannelZwell are your channels grouped right?
23:57.23ChannelZ(define "didn't like")
23:57.34kam187yeah, they are.. g1 works fine
23:57.55kam187<PROTECTED>
23:58.03kam187its treating it as a normal channel name
23:58.42kam187exten => _12345644X.,1,Dial(test/r1/0${EXTEN:8},60)
23:59.07ChannelZWhat ver of asterisk?
23:59.17kam1871.4.29
23:59.25kam187maybe its new for 1.6?
23:59.31ChannelZwhat is 'test'?
23:59.48kam187my channel driver
23:59.59ChannelZso it's probably a limitation of your channel driver then

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