irclog2html for #asterisk on 20060320

00:00.02nextimefor example, for many people freepbx is good and work very well
00:00.12nextimefor others it suck
00:01.00Ubeyguywhere can i download freepbx  ?
00:01.20nextimefreepbx.org?
00:01.43brookshiresf.net/ampportal
00:01.45brookshirelol
00:01.48brookshirei think
00:04.53*** join/#asterisk CrippsFX (n=CrippsFX@Kitchener-HSE-ppp3568787.sympatico.ca)
00:09.38CrashHDhow can I make calls authenticated (iax) based on ip?
00:09.46CrashHDI'm already using host=IP
00:09.54CrashHDbut it doesn't seem to mark the calls as authenticated
00:11.27*** join/#asterisk _Paulo_ (n=pirch@201-13-17-36.dsl.telesp.net.br)
00:11.36*** join/#asterisk Vazir (i=anton@82.198.21.17)
00:11.40_Paulo_~seen coppice
00:12.03jbotcoppice <n=chatzill@53.162.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 18h 2m 51s ago, saying: 'its wrong that short DTMF is produced when you press keys on a cellphone. well controlled lengths are produced. for most makes of infrastructure the lengths are very long. they are never too short, though'.
00:12.04VazirDoes anyone used H323 with Asterisk?
00:12.04VazirHi Folks :)
00:12.30_Paulo_Vazir, I tried but open323 makes my server freeze
00:13.13VazirYeah, I have some luck, but I need stable solution :)
00:13.31*** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net)
00:13.38VazirMaybe anyone knows stable combination of openh323/pwlib for example
00:14.03asterboyAnyone know a TFTP server for Linux, (non-rpm)?
00:14.10asterboycan'
00:14.20Qwellasterboy: there are several
00:14.28asterboyI've been looking to no avail.
00:14.31Vazirany linux have tftpd daemon
00:14.41asterboynot if your an LFS type like me.
00:14.44_Paulo_asterboy, look at http://freshmeat.net
00:14.58asterboysourforge.net
00:15.01Vazirthan use debian
00:15.01Qwellselinux-tftpd, tftp-hpa, netkit-tftp, atftp, linksys-tftp
00:15.11Qwellat least 3 of those are tftpd's
00:15.45Vazirany chan_ss7 experience guys?
00:16.01Qwellasterboy: I use the hpa one
00:16.29asterboylooking...
00:16.35nextimeVazir : for the moment the only one stable solution for h323 with asterisk in my opinion is chan_oh323, isn't the most performant, but is the most stable, the second one is chan_woomera, the third is chan_ooh323 that i like but is really young, and the latest for stability but good for performance is chan_h323
00:17.07asterboythought sourceforge and freshmeat were the same.
00:17.16nextimeanyway, i'm testing hard ooh323 but i'm using only oh323 in production at the moment
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00:17.35Vazir:) funny - but for me chan_h323 was most stable - and passed 10000 calls before crash
00:17.42*** join/#asterisk Skumling (n=skumling@fw.sg12.dk)
00:17.44brookshireaster: same people, different content
00:17.45justinuthat's stable?
00:17.46justinulol
00:18.01nextimeVazir : 10000 calls before crash isn't stable for me :)
00:18.13Vazircan you share any experience for your oh323
00:18.14nextimestable is "don't crash at all"
00:18.18*** part/#asterisk CrippsFX (n=CrippsFX@Kitchener-HSE-ppp3568787.sympatico.ca)
00:18.22VazirGlad to find someone H323 !
00:18.43Vazircombination of openh323/pwlib/asterisk versions?
00:19.13nextimeAsterisk SVN-trunk-r7230 built by root @ voip01 on a i686 running Linux on 2005-12-18 12:45:23 UTC
00:19.24nextimeasterisk-oh323-0.7.3
00:19.37nextimeopenh323_Mimas_patch2
00:19.41nextimepwlib_Mimas_patch2
00:19.55shmaltznextime, then you should try Avaya their Parner system will not crash by 10000 calls
00:19.56Vaziruhum... what distro do you use?
00:20.09nextimeVazir : debian
00:20.22VazirThe same as me
00:20.35VazirI did tried both on stable and testing. What yours>
00:20.36Vazir?
00:20.38nextimeshmaltz : this combination of * and oh323 is really stable for me
00:20.58shmaltznextime, good for you
00:21.09Vazir<PROTECTED>
00:21.31nextimei'm generating a about 200000 calls/day and is running good from 4 m without problem
00:21.55shmaltznextime, for what type of business?
00:21.59nextime( before i was using a oldest version of asterisk )
00:22.12VazirI do use SIP/H323 convertor for now and would try your combination today!... Hm strange if I did not try
00:22.14Vaziryet
00:22.17nextimeshmaltz : traffic reselling
00:22.35shmaltzwhy did you add oh323 to the mix?
00:22.49VazirI do use MVTS as a softswitch and want to have an Asterisk as a gateway to PSTN
00:23.02shmaltznextime, you doing it in VoIP, not TDM, right?
00:23.05asterboythx for the tftp-hpa
00:23.21nextimeshmaltz : because some telco support *only* h323
00:23.34asterboylooks good.  Will save me from running my Polycom init files at a Windows box.
00:23.35VazirI would say MOST telcos...
00:23.38nextimeshmaltz : voip AND tdm
00:23.44VazirH323 is in commercial world
00:23.46shmaltzso you have telco giving you traffic over h323?
00:23.49asterboyStrange that Sourceforge has no tftp servers.
00:23.59shmaltzVazir, not here in the US
00:24.02nextimeshmaltz : no, i give h323 traffic to telco :)
00:24.10UbeyguyAnyone here use MetaSwitch?
00:24.17Frogzooasterboy: sourceforge has atftpd dude iirc
00:24.31Vazirshmaltz, maybe for sure :)
00:25.52alephco1I have a provider using it....  I wish I had one.
00:25.52VazirBut even with US partners I'm offered H323 first :)
00:25.52shmaltzVazir, maybe for sure what?
00:25.52nextimeshmaltz : anyway, i'm using sip and iax2 too
00:25.52Ubeyguymy company has a metaswitch 4/5 class
00:25.52nextimebut in commercial world h323 is the most requested and most time is the only one solution in voip
00:25.52asterboymust have typed in my search wrong, couldn't find squat.
00:25.52shmaltzin any case if asterisk is configured correctly, you  shoulnd't have a problem getting 10000 calls
00:26.12CrashHDhow do I force an incoming iax2 call into a certain context?
00:26.23shmaltzCrashHD, iax.conf
00:26.25nextimeshmaltz : so, my config is correct :)
00:26.40shmaltznextime, I didn't see your config, I don't know
00:26.55CrashHDshmaltz: that I know, but it doesn't seem to be working
00:27.10nextimeshmaltz : trust me, my config is working perfectly for my needs :)
00:27.13shmaltzCrashHD, you did a reload in asterisk CLI?
00:27.17CrashHDya
00:27.21shmaltznextime, I got no clue
00:27.37CrashHDOTICE[415]: chan_iax2.c:6799 socket_read: Rejected connect attempt from 71.16.179.149, request '6023571734@default' does not exist
00:27.40shmaltzCrashHD, and how do  you know it's going to the wrong context?
00:27.47CrashHDthose notices
00:28.13shmaltzdoes shoe dialplan default show you an extension for 6023571734?
00:28.20shmaltzshoe==show
00:28.25CrashHDthere is no default in the dialplan
00:28.34CrashHDhttp://pastebin.com/611597
00:28.37CrashHDis my iax context
00:28.37shmaltzCrashHD, then you answered your quesiotn
00:28.47CrashHDit shouldn't be going to the default
00:28.58CrashHDI have context set to inbound
00:29.07shmaltzCrashHD, but does your extension.conf have a inbound context?
00:29.11CrashHDyes
00:29.23shmaltzCrashHD, pb it
00:29.50shmaltzplus, pb the follwoing command from the CLI:
00:29.52shmaltzshow dialplan inbound
00:31.05CrashHDhttp://pastebin.com/611601
00:31.08CrashHDsimple simon stuff
00:31.15CrashHDit's just not going to the right context
00:31.43*** join/#asterisk Ridgeback (n=jircii@104.243.8.67.cfl.res.rr.com)
00:32.13Ridgebackanyone know if you can push SIP prescence status through an IAX trunk?
00:33.07CrashHDshmaltz any ideas?
00:33.39Ridgebackmy buddy watch/status works fine locally. just would like it to span over the trunks
00:33.40Frogzooasterboy: I lie: I got it from the ubuntu repos, but the copyright says: ftp://ftp.mamalinux.com/pub/atftp
00:33.57shmaltzCrashHD, looks like you are requesting to dial with the @default at the end
00:34.11shmaltzRedgeback, I doubt it
00:34.24Ridgebackshmaltz: ok thanks
00:34.36CrashHDso the inbound call is directing to the default context?
00:35.11CrashHDI'm pretty sure the dial is just DIAL(IAX2/${PEERNAME}/${EXTEN})
00:35.19CrashHDany reason why it would want to veer toward default?
00:36.16CrashHDit is currently hitting the last context that is set in the iax.conf
00:36.18CrashHDwhy would that be?
00:36.41asterboyI'm trying to run it but it just exits without any message.
00:36.46asterboycompiled nice
00:36.53asterboytried tftpd -l
00:37.14asterboybut I had to copy it from the compile directory, make install did not put it in /usr/bin
00:37.26asterboywhat do you start it with? inetd?
00:40.47Frogzooasterboy: either/or
00:40.51shmaltzCrashHD, try including the inbound context in the default context, see if the call goes thru
00:41.06Frogzooasterboy: you remembered ./configure?
00:41.14CrashHDshmaltz: I think I figured out, the call is coming in as unauthenticated
00:41.16asterboyhmm, not sure why it won't start, strange
00:41.35CrashHDshmaltz: but why that is I'm unsure. I have the host=HOSTIP setup
00:41.35asterboyohya, did the ./configure; make; make install
00:41.44Frogzooasterboy: os depending, may need root perm to start
00:41.46shmaltzCrashHD, exactly  thats why I told you to do the last test
00:41.47asterboywhere does it put the log output?
00:42.01asterboyya I should check the README again.
00:42.23Frogzooasterboy: some OSs won't let you access ports < 1024 without root
00:42.34asterboyI'm ok there.
00:42.44CrashHDshmaltz: ok hold one sec
00:44.34CrashHDshmaltz: yes the calls go through at that point
00:44.49CrashHDshmaltz: so how can I get this call to auth?
00:45.05shmaltzI don't know, what is your setup?
00:45.09shmaltztry register first
00:45.28CrashHDshmaltz: the call is inbound from a voip provider (they don't register, just send the call)
00:45.47shmaltzCrashHD, then you have to register
00:46.02*** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F3734.dip0.t-ipconnect.de)
00:46.16CrashHDshmaltz: isn't there a way to do it only based on ip?
00:49.04*** join/#asterisk razu_ (n=razu@80-235-91-173-dsl.prn.estpak.ee)
00:50.55*** join/#asterisk Snake-Eyes (n=blog@202.168.41.172)
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01:02.33Ubeyguyhi is there a asterisk@home room?
01:05.34Abydos313not really, what's your question
01:06.22Ubeyguyok well i have a VoIP account and the only info i have is Username password and server
01:06.46Ubeyguynow when i try to add a trunk theres a out section a in section and i dont know what to put ( Using AMP)
01:07.08Abydos313most providers have a howto on setting up with asterisk.
01:07.15Abydos313is it an iax account or sip
01:07.23UbeyguySIP
01:07.47Abydos313who is the provider. a quick search will probably show a howto.
01:08.22Ubeyguywell my company set up there own VoIP metaswitch and there is really no one i can talk to that manages it
01:09.20*** join/#asterisk theorem_ (n=theorem@pool-71-127-251-111.nwrknj.fios.verizon.net)
01:09.21Abydos313outgoing sounds like the place to put the info
01:09.22theorem_wee
01:09.57Ubeyguywhat about Incoming Settings?
01:09.58theorem_ok, so if I just installed asterisk onto debian ... what should I do next ?
01:10.32Ubeyguytheorem_:  format hd
01:10.35Abydos313i've never done it thru asterisk@home . only put entries in sip.conf or iax.conf
01:10.41theorem_dang
01:10.54UbeyguyAbydos313: can you give me a sample of sip.conf you use
01:10.58*** join/#asterisk exten123 (n=exten@60.49.6.190)
01:11.47Abydos313they are all over voip-info.org
01:13.40*** join/#asterisk Mw3_ (n=mw3@national.t-error.hu)
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01:17.45*** mode/#asterisk [+o russellb] by ChanServ
01:20.22CrashHDwhy is it that if no secret or user is given (or included in context) on an iax context and only host= is used the call still shows as unauthenticated?
01:24.31russellbif you provide no way to authenticate a user, it will always say that
01:24.42orlockHmm..
01:24.42russellbit's just telling you that the call was accepted without any authentication
01:24.50orlockI've got some grandstreams here, which can dial each other
01:25.01orlockbut i'm having problems dialling out using a sip account
01:29.49*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
01:30.18theorem_orlock - firewall problems ?
01:32.57*** join/#asterisk |ryan| (n=foo@c-67-174-157-188.hsd1.ca.comcast.net)
01:33.22|ryan|does anyone know of a cheap (less then $100) 3.3V PCI FXO card?
01:33.40mogormanx100p
01:34.00CrashHDhow come my inbound iax calls are showing unauthenticated
01:34.41CrashHDI have host=10.0.4.243
01:34.47CrashHDand secret=secretkeyhere
01:34.55CrashHDstill shows as unauthenticated
01:35.15|ryan|mogorman: no, the X100P is 5 volt
01:35.36|ryan|will not work in a 3.3 volt PCI socket.
01:35.50|ryan|something USB would also do
01:35.58|ryan|I have a soekris net4801
01:36.13*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
01:41.03CrashHD$20 bucks via paypal to the first person to help me fix this damn iax problem
01:41.14tsumeheh
01:41.17CrashHDheh
01:41.21CrashHDdesperation
01:41.35CrashHDso it'll take you 5 minutes to solve
01:41.40CrashHDI'm just a big dummy
01:41.58tsumeChotaire: what is the problem?
01:42.06tsumeoops
01:42.08CrashHDheh
01:42.10tsumeCrashHD: what is the problem?
01:42.24CrashHDiax calls are not hitting the proper contexts'
01:42.32CrashHDand are showing unauthenticated
01:42.48tsumeI have rules, I must see code ;)
01:42.54tsumeI don't have ESP
01:43.07CrashHDsure, let me get it all up on pastebin
01:43.07CrashHDbrb
01:43.47CrashHDjust to give you some background
01:43.56CrashHDI have inbound traffic from a voip provider
01:43.58CrashHDvia iax
01:44.12*** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net)
01:44.14CrashHDwhich goes to a * gateway machine (voip_gw_1)
01:44.25CrashHDsome calls get passed to another * machine (voip_gw_2)
01:44.30CrashHDsome get handled or passed else where
01:44.48CrashHD(voip_gw_2) also needs to dial out through (voip_gw_1)
01:48.29CrashHDcheck query
01:48.54CrashHDa call from gw_1 to gw_2 shows as authenticated
01:49.21CrashHDcalls from [ipcomms] to gw_1 show unauth as well as from gw_2 to gw_1 (show unauth)
01:49.49CrashHDconsequently inbound calls on iax fall into [inbound] because it is the last context= defined in the iax.conf
01:50.34asterboyfinallyl got the tftp-hpa working.
01:50.52asterboyfound an entry about user name in /var/log/sys.log
01:51.13asterboylooks like you need to invoke the command with a -u username.
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01:51.23asterboyalso, added entries into /etc/services.
01:51.31asterboywould be nice if that was in the README.
01:51.46asterboytypical linux
01:52.02NuggetLinux is poo.
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01:55.11asterboypoo? as in corn and poo?
01:55.48FrogzooNugget: :X
01:55.51asterboyas in - dries on bum hair, makes itchy, make for smelly finger?
01:56.13asterboyas in - bacon strip
01:56.25Frogzooatftpd works for me - just install the package & go
01:56.50asterboylfs can be a litle picky
01:57.08asterboyusually missing some library the standard distros dump everywhere.
01:57.26asterboyI like my box tight. :->
01:57.52asterboyas in - clamped around my finger
01:58.47tsumeWindows is poo
01:58.51tsumelinux is the savior
01:59.02tsumebut if you want a better system than both.. use MacOSX :)
01:59.21asterboyThey all make me money one way or the other.
01:59.43asterboyI like em all, but for Servers I love Linux.
01:59.51adelashas anyone heard of ACN here?
01:59.56asterboy~acn
02:00.13adelasdo you think the companies any good?
02:00.16asterboynot even jbot knows...what does it stand for?
02:00.27adelasACN = phone company
02:00.43adelasa digital phone, analog phone, and dsl provider company
02:00.53asterboyDo they have technical support that directs you to India?
02:02.09asterboyFunny how the automated atendant never has an east indian accent.
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02:26.09orlockHmm, i have inbound calls working, but outbound i;m getting a 404
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02:32.53Kumbangguys, how can set flash hook time, my old PBX seems to reject it
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02:40.34[hC]I hate to ask this, but does anyone have, or us, the Zyxel P2000W wifi phone?
02:40.41[hC]us=use
02:41.59DjeliNot me, sorry
02:44.56X-Rob[hC], apparently they suck really really badly.
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03:04.34Vazirhi men! Anyone using OH323 in production?
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03:19.57CrashHD-- Attempting native bridge of IAX2/ipcomms149-12 and IAX2/voip_gw_2-13
03:19.57CrashHD<PROTECTED>
03:19.57CrashHD<PROTECTED>
03:20.00CrashHDwhat does that mean?
03:20.04CrashHDdoes it everytime
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03:22.40rpmsounds like you are using incompatible codecs.
03:22.47rpmls
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03:30.26babyhuey<PROTECTED>
03:30.26babyhuey<PROTECTED>
03:30.26babyhuey<PROTECTED>
03:30.26babyhuey<PROTECTED>
03:30.39babyhueyi get that every time i call in, then i cant do anything with the system, and i have to reboot
03:30.44babyhueyit scrolls that over and over
03:30.52I-MOD~pb
03:30.57jbotpb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:31.16babyhueysorry for that
03:31.35|ryan|does anyone know of a cheap (less then $100) 3.3V PCI FXO card? I've got a X100P, but it needs a 5v PCI socket :(
03:35.18babyhueyhttp://pastebin.com/611808
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03:43.46ambrientobabyhuey, what about some extensions.conf lines?
03:44.28babyhueywell, im using amp to set it up, dont know if that helps
03:44.40babyhueystill want me to post it?
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03:45.48whatisthatHi
03:46.04whatisthatcan I use database to store sip user in Asterisk
03:47.00ambrientosure babyhuey, why not? :)
03:47.25whatisthatwhich database can be used to store sip user
03:47.32ambrientowhatisthat, how is that?
03:47.47I-MODwhatisthat: yes, sql, voip-info.org
03:47.50ambriento:)
03:48.08whatisthatI want to use database to store user and password instead of using sip.conf file
03:48.24babyhueyambriento: http://pastebin.com/611818
03:48.59whatisthatis there any guide for this
03:50.17whatisthatI-MOD: can you tell me where to find document guide for this
03:50.50I-MODhttp://voip-info.org
03:51.04I-MODits on there somewher
03:51.05I-MODe
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03:55.11ambrientothats a lot of stuff babyhuey
03:55.15babyhueyyea, i know
03:55.19babyhueyits all generated by amp
03:55.54I-MODactually, i think amp support issues are directed toward #amportal
03:56.05babyhueyi dont think its an amp problem
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03:57.22ambrientowhich verbose level are you using at CLI's?
03:58.23babyhueyi usually just hit it 4 times
03:59.17X-Robwhatisthat, you're looking for 'realtime'
03:59.31X-Robthat lets you store stuff in databases, rather than config files. See voip-info.org
03:59.47X-RobI-MOD, #freepbx now
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04:17.41whatisthatcan you tell me some usb phone can be used with asterisk
04:20.38exten123can we give FXS port a channel name like FXO port?
04:20.59exten123I mean by digium hardware
04:27.15Octothorpe[away]CrashHD:  Did you get your problem fixed?
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04:33.23TheCops[TK]D-Fender, hey :)
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04:38.03CrashHDhow can you check t1/pri clocking?
04:38.13CrashHDOctothorpe[away] sort of
04:38.33CrashHDOctothorpe[away] I worked around the issue. I still have some questions about how * deals with iax authentication
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04:40.17nayyareswhere can i find the a2billing installation howto?
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04:41.13BugKhamHi, I've got a quick newbie question. How do I know that the ISDN-PRI is connected properly?
04:41.25I-MODzttool
04:41.30BugKhamassuming that all configurations are right
04:41.52BugKhamztcfg?
04:42.04I-MODyou can also use pri show span 1 on the asterisk console
04:42.28SwK_bugkham pri show span ${SPAN_NUMBER}
04:42.39SwK_so like "pri show span 1"
04:42.41*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
04:42.43BugKhamI-MOD: I normally 'cat /proc/zaptel/*'
04:42.54BugKhamSwK_: k, thanks
04:43.09SwK_zttool will only show you if the T1 has carrier pri show span will show you if the D is up
04:43.19SwK_omg is blitzrage
04:44.03blitzrageomg! :)
04:44.11brookshirewhat a nub
04:44.19blitzragebrookshire: OH NO YOU DIDN'T
04:44.29blitzragebrookshire: I need you to fix something :)
04:44.45blitzragenevermind -- you already fixed it
04:44.58brookshireblitz: lame lame lame
04:45.31blitzrageLlama's Ate My Eggo's
04:45.31ambrientobabyhuey, I'm sorry but I got stuck in some other stuff over here
04:45.57babyhueyi got it working
04:45.59ambrientobesides, its almost 2am and I have to got up really soon
04:46.09babyhueythanks for the help
04:46.09babyhuey2am?
04:46.09babyhueywhere are you?
04:46.09babyhueyits only 11.45 here
04:46.26ambrientoI'm in brazil
04:46.31babyhueycool
04:46.33ambriento:)
04:46.43babyhueyalright, well, ill talk to you later then
04:46.47ambrientoAnd no need to thanks me, I didn't do anything :)
04:46.57ambrientobe my guest babyhuey. :)
04:48.14*** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net)
04:58.22BugKhammy status says, Status: Provisioned, In Alarm, Down, Active
04:58.44Math`heh I got that too, but no D-Channel is config'd on the other side so I understand
04:59.03BugKhamso, the link is down, am I correct?
04:59.29Math`it is
04:59.54BugKhamMath`: hehe
05:01.23BugKhamMath`: do u have to put crc parm in your zaptel.conf?
05:02.00Math`I just closed the pri's other end and its still displaying that status
05:02.01Math`I didnt
05:02.30BugKhamMath`: this is the first time so I don't know if the crc parm is required
05:07.03asterboyInteresting that you can bypass password for the reset function of a polycom phone.
05:08.02Math`by unplugging the power and putting it back?
05:08.30asterboylol
05:08.48Math`it sure bypasses the password
05:08.54asterboywhy do they put that function in a password protected area?
05:08.59Math`I have no idea
05:09.16Math`well when I open a polycom which isnt provisioned yet, its not passwd-protected
05:09.24*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)
05:09.27Math`by in my config it is (config which I took from the Wiki(tm))
05:09.30asterboyI just hold done the [Vol-] [Vol+] [Mic Mute] [Messages] keys for >3secs
05:09.41Math`ah ya thats documented too
05:10.02asterboypage 17 of user guide
05:10.07Math`yup
05:10.19asterboywhen debugging it sure makes it faster to reboot
05:10.43Math`indeed
05:13.30Math`you can also send the phone a certain kind of sip message to make it reboot
05:16.09*** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net)
05:18.52Math`justinu: I finally tested the async rtp branch, I need to Background(silence/1|n) then do a Ringing(); for it to ring on the other end
05:27.55*** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
05:40.30*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:42.22*** join/#asterisk julien[re] (n=julien[r@AStDenis-103-1-11-206.w80-8.abo.wanadoo.fr)
05:42.28julien[re]hi there
05:43.08asterboyWhen setting up a dial plan where I just want a line pickup to immediately goto another SIP extension, I have this: exten => 1,2,Dial(SIP/Upstairs)...must be missing something cause it doesn't work.  Any suggestions?
05:43.33orlockHmm
05:43.40julien[re]exten => s,1,Dial()
05:43.44orlocki can dial in to my asterisk system, but trying to call out i get a 404
05:43.52orlocki'm setting it up using SAIL
05:44.21asterboyjulien, how will it know where to Dial with just dial()?
05:44.36julien[re]i mean Dial(SIP/Upstairs) ;)
05:44.57asterboyoh ok, the s,
05:45.00julien[re]that's what I have on one of my DID and it works
05:45.04mishehuanybody here using iaxmodem to send faxes?  I'm having a problem where iaxmodem doesn't seem to want to connect to the proper context (when dialing, it send 1234567@ )
05:45.08julien[re]the first item is for the DID
05:45.24asterboytrying, thanx
05:45.25julien[re](as far as i've understood)
05:46.05julien[re]btw, i've been wondering how to have such a SIP URI working: http://www.voip-info.org/tiki-index.php?page=Phone+Numbers
05:46.10julien[re]any1 could help?
05:46.29julien[re]i mean, i'd like to have this URI for my *
05:47.47justinumath: i used Playtones(ringing)
05:48.41Math`oh
05:48.52Math`now I Queue() so I can listen to moh when ringing :)
05:50.21justinudial option m does that too, i think
05:51.10asterboyjulien[re], I think that the SIP URI is a function of either a soft phone or a hardware phone where you can cut&paste the URI...some phones like my Polycom allow you to use the alpha keys to punch it in or you could do some fancy asterisk programming with IVR and do it with a regular phone via FXS adaption.
05:51.43Math`justinu: it does! is that a new option?
05:52.02justinudunno
05:52.58asterboyAnyone here setup Polycom phones so you can dial another Polycom phone by hitting a line key.  I want to just dial from one extension to another via two Polycom phones.
05:53.03asterboy?
05:53.18Qwellasterboy: Do they have speeddial support?
05:53.26asterboyyes
05:53.29asterboyIP 600
05:53.29Qwellthen sure
05:53.46asterboyah, ok..I was thinking I had to register the lines.
05:53.48Qwellor are the speeddials not on the line keys?
05:53.56asterboyThen setup a dial plan to call one or the other.
05:54.17asterboyI'm sure I can setup the speed dials on any of the line keys.
05:54.38asterboySo do the SIP URI thing?
05:55.29Math`you can override any keys in the configs
05:55.44Math`even arrows.... tho this is not recommanded (for obvious reasons)
05:56.04justinui had a frustrating experience, my polycoms wouldn't use their custom ring tone
05:56.05asterboylol, even the arrows...that is neat.
05:56.15justinueven tho it downloaded the ring tone, and I could set it manually
05:56.23justinuit would not pick the setting up from the config file
05:56.42*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
05:57.04justinuyep, no matter what I set in the xml config, it would chose 2
05:57.41asterboyjustinu, I just had some of that frustration.
05:57.54justinuno kidding...
05:57.55asterboyrealized it was a rogue "/>"
05:57.57asterboyend tag
05:58.02justinuyeah - i thought that too
05:58.08asterboyin the wrong spot due to my cutting and pasting.
05:58.09justinubut i'm pretty sure my xml is ok
05:58.37asterboyThe line I setup would just not show up.
05:58.50asterboytotally ignored the changes.
05:58.57justinui'll crank of the debug log level, see if the stupid thing complains about invalid xml
05:59.10justinus/of/up/
05:59.33asterboycool, didn't jbot did that.
06:00.10asterboys/didn't/didn't know/
06:00.39asterboyeven pickup up the "'"
06:00.39*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-43-154.cybersurf.com)
06:00.39*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
06:00.39*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
06:01.16*** join/#asterisk Snake-Eyes (n=blog@202.168.41.172)
06:02.33julien[re]any1 could try a SIP URI for me?
06:03.07asterboyonce I learn how to program my speed dials...sure
06:03.12julien[re];)
06:06.11*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
06:07.20gigglesanybody know the format of the xml config file on a mediatrix 2102?
06:07.34Qwellgiggles: I'm just guessing here, but...
06:07.39Qwellcould it maybe be in...xml format?
06:08.09asterboylol
06:08.22giggleshar har har
06:08.38asterboyspeaking of...just found Polycom's speed dial instructions on page 34 of Admin Guide.
06:08.49mattodudeQwell: no, it's in encoded binary XML format!
06:08.55Qwelleww
06:09.18asterboy<Ethernet address>-directory.xml
06:09.28asterboyhate having to reboot all the time.
06:09.33*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:10.53*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
06:11.20gigglescan it be that nobody knows how to program this device?
06:11.34gigglesor is it that no one wants to tell?
06:11.48asterboynot here anyway...did you try clusty.org?
06:15.31gigglesok something new, a lot of hits
06:15.45gigglesnot sure if there is anything useful, but thanks
06:16.36*** join/#asterisk Foxamemnon (n=elrond@71.195.202.240)
06:16.44x86anyone have sip extensions defined in a mysql database working correctly?
06:17.16x86i'm using asterisk 1.2.4 on my Gentoo system (which takes the source and applies a few patches)
06:18.19x86i have mysql in my USE flags, and I have CDR with mysql working (using cdr_addons_mysql)
06:18.24x86CDR works great
06:20.59FoxamemnonHello.  I'm looking for some help getting Asterisk to record video with voicemail.
06:21.24*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
06:21.31asterboyX86, I'd like to do this also...can't help much no since its new to me.
06:21.43asterboys/no/now/
06:21.50FoxamemnonWhen I make the connection, I get errors from Asterisk:  "No translator path from unknown to unknown" and "Unable to translate to format h263, source format unknown"
06:22.08asterboyfor my CDR, I just wrote a little shell script.
06:22.41FoxamemnonBut I don't know what's wrong.  My understanding is that Asterisk will just save the h261/h263 stream to disk.  That's fine by me and Asterisk wouldn't really need to know the internals of the format.
06:23.47asterboyI had that error when trying to use a SIP connection without a license for G729 codec from Digium.
06:24.09*** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au)
06:24.33FoxamemnonHmmm... in this case, the connection is GSM audio and h263 video.  Shouldn't need any licenses for that, right?
06:24.52asterboywell I'm thinking that your missing a codec or something.
06:25.03asterboyliscense or no.
06:25.22asterboycan you test each codec individually?
06:25.53asterboycause it looks like it want to convert from one to the other but it's missing the codec or the config to do it.
06:26.06asterboys/want/wants/
06:26.41littleballhello, who can give me a suggestion about setting up a minimum cost of asterisk system? basically, it collects to analog phone in one end and the other end it connects my main asterisk system through internet.
06:27.01littleballs/collect/connect/
06:27.16FoxamemnonThat's what I thought too.  But I'm not sure how I can test the doces properly.  I checked the source distribution (I'm using the Debian packages) and h263 is part of the core of Asterisk.
06:27.58Foxamemnondoces->codecs
06:28.41*** join/#asterisk Assid (n=assid@59.183.13.48)
06:28.47Assidheya
06:28.53AssidMar 20 11:55:44 mercury kernel: zaptel: disagrees about version of symbol struct_module
06:29.04Assidi cant manage to get the zaptel working
06:29.22asterboylittleball, $25 FXO adaptor from ebay, $10 PII 233MHz, 96Mb Ram machine, Your time ...priceless.
06:29.38Assidwithout struct i get those RTC errors
06:30.40asterboyyuk, I hate compile errors...either the hardware or the software...and software is tought to tell...I always have a spare box with a fresh  install handy so I can eliminate the possibilities...otherwise goto clusty.org
06:30.46Assidanyone know how do i fix it?
06:30.55asterboys/tought/tough/
06:32.54asterboyFox, I'm not up to par on those codecs, so can't help much, but I'd start by creating a test case...h263<->h263.
06:33.32Assidasterboy: i get that error when i boot up and zaptel doesnt wanna load
06:33.42asterboyah
06:33.57asterboyDid you try different pci slots?
06:34.05Assidpci slots?
06:34.24trixterdid you compile zaptel with the same kernel headers that were used to build your running kernel?
06:34.27Assidi dont have any zaptel device
06:34.28asterboyzaptel is pci hardware no?
06:34.42Assidnah.. using ztdummy
06:34.44Assidtrixter: yes
06:34.45asterboyah
06:34.51Assidlemme recompile again
06:34.54trixterif you did then that error shouldnt be present
06:35.06trixterthat error indicates that there is a version mismatch :/
06:35.24Assidwithout that.. i normally get some RTC error of 1024Hz
06:37.17asterboyIf anyone wants a simple CDR script to save your eyes:
06:37.18asterboycat /var/log/asterisk/cdr-csv/Master.csv |grep incoming |cut -d"," -f2,10,12 |sed -e "s/\"//g" |grep
06:37.21asterboy<PROTECTED>
06:37.45io_errorasterboy: ah, that script hurt my eyes!
06:37.49asterboybeen using that until I get SQL online
06:37.52asterboylol
06:38.02asterboyits only temporary
06:38.07io_error:)
06:38.47*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
06:40.04Assidas soon as i load ztdummy : i get rtc: lost some interrupts at 1024Hz.
06:41.27Assidtrixter: any clue whats to be done?
06:42.37asterboyrtc = real time clock, all I can add.
06:43.17Assidi thin i had a similar issue with another box before
06:43.24Assidbut it was related to USB
06:43.30asterboywhy it's having an interrupt issue at some specific cycle...dunno.
06:45.13Assidtrixter?
06:45.37FoxamemnonWell, I tried narrowing it down.  The program I'm using supports only h261 for video.  So I changed Asterisk's sip.conf to allow only gsm and h261.
06:46.07FoxamemnonI get the same error messages when trying to record a voicemail.  It seems Asterisk keeps wanting the video stream to be h263 even though it's not.
06:48.08asterboytry a full reboot, (unecessary usually, but it will make sure your config sticks).
06:48.43asterboyI did a lot of changes that did stick often cause I was not changing the right file or reloading properly.
06:49.10asterboys/t did/didn't/
06:49.23asterboylol
06:49.26io_errorhe
06:49.30io_errors/he/heh
06:49.35io_errors/he/heh/g
06:49.41io_errorLOL
06:49.46asterboy:P
06:50.21asterboyit's getting late hear...me spelling is drifting.
06:50.30asterboys/hear/here/
06:50.46io_errorio_error meant: the final / is generally optional, you stupid bot
06:51.02Assidi have a ohci usb
06:51.10Assidi even have the drivers in the kernel
06:52.07Assidi got USB Controller: Broadcom CSB6 OHCI USB Controller
06:52.25asterboys/got/have/
06:52.58FoxamemnonOkay, I restarted the machine.  But it's still doing the same thing.
06:53.39asterboyAssid, try taking the card out for now...see if there is any change.
06:53.47Assiderr.. onboard
06:53.56asterboydisable in bios
06:54.26Assidlemme quickly disable the ohci usb
06:54.55asterboyFoxamemnon, where did you make the change to tell * to use your codec.
06:55.10Assidfrom the kernel
06:55.15Assidmaking it into module
06:55.28io_errorAssid: what do you have set in the kernel config for CONFIG_HZ ?
06:55.29asterboyAssid, may not help but it can eliminate the possibility
06:55.38Foxamemnonasterboy, In sip.conf.  I set disallow=all, then allow=gsm and allow=h261
06:56.00asterboymaybe do a pastebin
06:56.17FoxamemnonIt does seem to have some effect.  If I set allow=h263 instead of h261, then I get no video support, which is correct.
06:56.32asterboyok that helps
06:56.45Assid# CONFIG_HZ_100 is not set
06:56.46AssidCONFIG_HZ_250=y
06:56.46Assid# CONFIG_HZ_1000 is not set
06:56.46AssidCONFIG_HZ=250
06:57.17asterboyAssid, that may help to play with those settings.
06:57.23io_errorlooks pretty default to me.
06:57.25asterboyguessing though
06:57.29io_errorTry it at 1000
06:58.44asterboyFoxamemnon, what does "show codecs" at CLI give ya?
06:58.55io_errorAssid: do you just see the error message once, or does it repeat?
06:58.57Assiderr.. i had a similar issue on another box.. checking those
06:59.01Assidio_error: repeat
06:59.11Assidthats the same CONFIG_HZ there
06:59.12io_errorAssid: what CPU and mobo?
06:59.32Assidp4 2.4  dellpowereddge 600sc
06:59.41Assiderr.. lemme get the mobo
06:59.59io_erroryeah, change the kernel to 1000 HZ and try again
07:00.00Assidno clue which mobo.. but its an intel from what i can see
07:00.16asterboyaha! see that's what you get for going to the darkside! Dell/Intel
07:00.38Assiderr..
07:00.47Assidits a dedicated server..
07:00.58io_erroroh... hmmm.
07:01.08io_errorAssid: dedicated, or VPS?
07:01.11*** join/#asterisk edwar64896 (n=edwar648@72.83.233.220.exetel.com.au)
07:01.23Assiddedicated....
07:01.38FoxamemnonOkay, here's the "show codecs" output:  http://pastebin.ca/46317
07:01.52io_errorAssid: what distro?
07:02.03Assiddebian
07:02.07Assiderr.. this is not good : Unable to find swap-space signature
07:02.38edwar64896join #asterisk-bugs
07:02.41edwar64896oops
07:02.42edwar64896soz
07:02.53io_errorAssid: I'm starting to suspect hardware problems
07:03.31io_erroryow...building a kernel for debian is a complete pain :(
07:03.51*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
07:07.50x86ok, i think i got sip extensions from mysql working properly...
07:08.02asterboyFoxamemnon, that is exactly what I have.  Looks in order.
07:08.29x86is there a script i can use to convert all of my existing extensions in the sip.conf flat-file into the mysql database?
07:09.41niZonanyone know how to enable the http server on a cisco ip phone (sccp firmware)?
07:10.24asterboyx86, shouldn' be too hard to wip up a script, imagine someone has done it though and there may be something floating in interspace.
07:10.57QwellniZon: I don't know if you can disable it
07:11.24asterboyFoxamemnon, wish I could get this going with ya, (be doing some of the same soon), anyway, gotta go to sleep ...have a good one.
07:11.37Foxamemnonasterboy, Thanks for your help tonight.  Guess I'll work on it some more tomorrow.  Goodnight.
07:12.10asterboyya, I'll be on...like to do more on it.
07:12.16asterboynight all
07:13.17niZonQwell: well my 7970 doesn't accept connections on port 80
07:16.55BugKhamwhat does this imply? -> "Detected alarm on channel 31: Yellow Alarm"
07:18.29BugKhamchan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 19
07:20.15AssidMar 20 12:48:22 mercury kernel: rtc: lost some interrupts at 1024Hz.
07:20.15AssidMar 20 12:48:22 mercury last message repeated 218 times
07:20.20Assidstill cant get rid of it
07:20.28Assidhad the usb disabled from bios too
07:21.07io_errorAssid: hm, I think you should have the USB enabled, ztdummy will want it I think
07:21.54Assiderr.. you just said to disable it
07:22.09io_errorer, I didn't say anything about disabling USB
07:22.15Assidyou didnt?
07:22.22io_errorno, that was asterboy
07:22.27io_errorwho has left
07:22.33Assiddammit
07:22.43Assidthe tech at the DC is gonna go nuts
07:22.56Assidive made him go to my box 4-5 times in the past 2 hrs
07:26.39*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
07:27.01Assidthig is .. last time this happened.. to another box
07:27.04kmilitzerMorning everyone ...
07:27.07Assidi dont know what i did to fix it
07:28.02Assidthat time it was because of : USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #1
07:28.13Assidthe 82801 usb uhci
07:28.34io_errorAssid: does this box have IDE disks?
07:29.24Assidwhich one ? the one that doesnt work? yes.. IDE .. but it works through serverworks
07:29.45io_errorAssid: try this: hdparm -u 0 /dev/hda
07:30.53io_errorer, excuse me
07:30.57io_errorAssid: try this: hdparm -u 1 /dev/hda
07:31.20Assid/dev/hda:
07:31.21Assid<PROTECTED>
07:31.21Assid<PROTECTED>
07:31.26io_errornow try ztdummy :)
07:31.33Assidyou kidding me?
07:31.37io_errorno
07:31.55Assidnah
07:31.57Assiddidnt work
07:32.07io_errorhm, you're losing interrupts somewhere else then
07:32.12Assidbut i made ehci and uhci into modules
07:32.20io_errorA nice comment in the kernel source said that was probably the cause :)
07:34.06io_errorAssid: do you have CONFIG_PREEMPT_* anything set in the kernel config?
07:34.32AssidCONFIG_PREEMPT_NONE=y
07:34.32Assid# CONFIG_PREEMPT_VOLUNTARY is not set
07:34.54io_errorhm, my working system has CONFIG_PREEMPT_VOLUNTARY=y :)
07:35.13io_errorand # CONFIG_PREEMPT_NONE is not set
07:35.25*** join/#asterisk justnulling2 (n=justnull@ool-18bcc906.dyn.optonline.net)
07:35.40Assidthis is what i have in my other workin:
07:35.41Assid# CONFIG_PREEMPT_NONE is not set
07:35.41Assid# CONFIG_PREEMPT_VOLUNTARY is not set
07:35.41AssidCONFIG_PREEMPT_BKL=y
07:35.47io_errorhm, either way
07:35.59io_errorbut CONFIG_PREEMPT_NONE is probably a bad idea to run *
07:36.06justnulling2i get "No application 'MeetMe' for extension" how can i fix it and i have ztdummy loaded?
07:36.09Assidwhats it for?
07:36.59io_errorrealtime response
07:37.17*** join/#asterisk g0mb0 (n=g0mb0@external.micom.mng.net)
07:37.42Assidwheres that in menuconfig?
07:38.09io_errorCONFIG_PREEMPT - This option reduces the latency of the kernel when reacting to real-time or interactive events by allowing a low priority process to be preempted even if it is in kernel mode executing a system call.
07:38.35io_errorAssid: Processor type and features
07:39.50Assid<PROTECTED>
07:39.56Assid( ) Voluntary Kernel Preemption (Desktop)
07:40.01Assid( ) Preemptible Kernel (Low-Latency Desktop)
07:40.55io_errorI like this one ( ) Preemptible Kernel (Low-Latency Desktop)
07:41.05io_errorBut either of the two should be fine
07:41.41Assidshould i still change the Timer frequency (250 HZ)  --->
07:41.49io_erroryes
07:41.55io_error1000 HZ
07:42.24Assidweird
07:42.30Assidnone of the other boxes are running at that
07:42.41io_errorSame hardware?
07:42.59Assidnah..
07:43.06Assidactually
07:43.07io_errorheh, well make the changes, if it doesn't work you can chagne it back :)
07:43.18Assidi got another box.. which used to run 2.6.11 and asterisk
07:43.21Assidsame hardware
07:43.39Assidand the same settings i brought here
07:44.37*** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it)
07:45.01*** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au)
07:46.15Assidhrmm
07:46.18Assidsomething went nuts
07:46.28Assidits redoing thr whole kernel compilation all over again
07:47.09x86that's what it's supposed to do
07:47.12x86you want that ;)
07:48.28|ryan|does anyone know of a cheap (less then $100) 3.3V PCI FXO card? I've got a X100P, but it needs a 5v PCI socket :(
07:48.30Assidman.. im supposed to be working . instead im sitting and figuring this crap out since 5 days
07:48.37Assidsomething or another always blowing up
07:48.43io_errorAssid: hehehhe :)
07:48.58Assidfirst.. kernel doesnt wanna upgrade coz of devfs
07:49.01Assidlost 2-3 days there
07:49.11Assidthen... grub blows up
07:49.16x862-3 days for devfs?
07:49.17Assidnot zaptel
07:49.22x86you must be a linux newbie ;)
07:49.36Assidx86: its a redhat box converted to debian remotely
07:49.38io_errorx86: he's running debian, of course he's a newbie :)
07:50.12Assidthat too.. the old debian (woody).. [at the time of installation]
07:50.32iDunnoouch ;)
07:50.47*** join/#asterisk linxroute (n=linxrout@58.187.123.87)
07:50.49x86could be worse...
07:50.54*** join/#asterisk medusaXX (n=medusaxx@p54A983D5.dip0.t-ipconnect.de)
07:50.55x86could be potato ;)
07:51.00Assidthen..... i finally just asked that tech to punch it into another debian box.. imaged the drive into this one.. and started it with sarge
07:51.08*** join/#asterisk twisla (i=twisla@lutin.jard.in)
07:51.25Assidthat just managed to get over.. and i started playing with zaptel.. when grub got messed up.. cause of the RTC..
07:51.41Assidi was trying to grub-install .. when the system locked up cause of that rtc issue
07:51.53io_errorAssid: yeah, it's sounding more and more like bad hardware
07:52.06Assidso the device map got corrupted. i said screw it and i just left  it for the night..
07:52.09*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
07:52.19justnulling2fixed meetme but just got /usr/sbin/safe_asterisk: line 42:  9373 Segmentation fault      (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY}
07:52.27Assidtoday i fixed that.. fixed that swap issue.. and am HOPEFUL of the zaptel
07:52.29*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
07:52.44*** join/#asterisk kilobit (n=seth@210.193.58.33)
07:53.12*** join/#asterisk psk (n=psk@golia.caltanet.it)
07:54.28*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
07:54.30Zeeekwake up and smell the nespresso (tm)
07:54.36kilobithi, im getting alot of "WARNING[18268]: chan_sip.c:4826 check_auth: Stale nonce received from" message... sip users can't register... any ideas?
07:54.56Assidnespresso?
07:55.59Assidgonnna tryu and install asterisk on the other box with EXACT same hardware
07:56.01Assid02:55:43 (1.17 MB/s) - `asterisk-1.2.5.tar.gz' saved [10546846/10546846]
07:57.40linxroutesorry if may i ask , does anyone here using the g729 "open source "
07:57.52Qwelllinxroute: no, it isn't legal to use it
07:58.02edwar64896yeah I used it for a while
07:58.09edwar64896legal smeagal ;-)
07:58.24linxrouteyou meant for a business or production box
07:58.35linxroutehow about just for testing
07:58.38edwar64896nope. my use was a home box.
07:58.51linxroutecos i was kind of wondering
07:58.56edwar64896Be warned, its a real pain in the ass.
07:58.59linxrouteif there any ..
07:59.04edwar64896you have to go through hoops with intel to get the license
07:59.12edwar64896you might as well just pay Digium for a couple of channels.
07:59.16linxroutecompatibale issue
07:59.19edwar64896it's a lot less hassle and it works straight away.
07:59.34linxroutecos i find the quality seems to be not very good
07:59.48linxroutei dont know how about the digium codec
07:59.52edwar64896its probably the best lossy codec avaialble at the moment.
07:59.52Assiddammit
07:59.53linxroutedoes it better ?
07:59.53AssidMar 20 02:59:20 box kernel: Zapata Telephony Interface Registered on major 196
07:59.53AssidMar 20 02:59:20 box kernel: Zaptel Version: 1.2.4 Echo Canceller: KB1
07:59.53AssidMar 20 02:59:20 box kernel: Registered tone zone 0 (United States / North America)
07:59.57Assidthis box works fine
08:00.08linxroutethanks edward
08:00.21linxroutehave u ever tried the " original" codec
08:00.22AssidEXACT same hardware
08:00.26linxroutefrom Digium ?
08:00.31edwar64896[mark] no problem
08:01.26linxroutehello
08:01.27Assidio_error: that works fine on another box .. with the EXACT same hardware
08:01.28io_errorAssid: Yep, hardware, figures
08:02.37*** join/#asterisk atif_ (n=atif@202.92.16.30)
08:03.03atif_hello there, can some one help me regarding OPTIONS message support in Asterisk....
08:03.25atif_asterisk is replying at udp src port instead of contact information provided in VIA headers
08:03.26*** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder)
08:03.33xbmodder_lappywhat are the downsides to ilbc?
08:04.31Assidhow the hell do i get those guys to give me another box?
08:05.06io_errorAssid: tell them the motherboard in that box is fried and to replace it.
08:05.26Assidwell... accoridng to these guys.. if the machine boots up.. mobo is fine
08:05.39io_errorAssid: cancel your contract for cause
08:05.40Assidone sec..
08:05.51Assidim regetting the kernel .config from that one
08:05.54Assidand recreating it
08:06.15Assidwithout modifications
08:06.33io_errorAssid: you've got a broken APIC or something probably, in any case if it works on another identical (hw/sw) box, then it's hardware
08:07.45Assidyep
08:07.56Assidthats why i took the .config from the other box and am recompiling
08:08.09Assidif it works.. it works
08:08.13Assidif it doesnt
08:08.14io_errorAssid: who is this server hosting company?
08:08.20Assidim gonna get the hard drives exchanged
08:08.25Assidev1server
08:08.30Assids
08:08.31io_errorah, that explains it
08:08.40io_errorNever, ever do business with ev1servers
08:08.43QwellI use ztdummy on my ev1 server
08:09.05Qwellio_error: there is nothing wrong with ev1
08:09.07Assidyeah.. my other box works fine too
08:09.21io_errorQwell: it sure sounds like there is.
08:10.45Assidwhile thats compiling im gonna go get ready
08:12.03io_errorThe best hardware is your own hardware. :)
08:12.27Assidi know people who have issues with that
08:12.38Assidthat intel one i showed you earlier.. uhci usb
08:12.43Assidthat had a hell lotta problems too
08:13.06*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:13.22*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
08:13.58Zeeekuse loudspeaker
08:14.23tsumeio_error: and what is your own hardware called? :)
08:14.57*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-161.claranet.co.uk)
08:16.46BugKhamdoes it take long to start an E1?
08:18.02io_errortsume: Which one? The one on my desk, the one under my desk, or the one next to my desk?
08:19.33*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
08:19.59tsumeio_error: any?
08:20.27*** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com)
08:20.44io_errortsume: and what do you want to know about it?
08:23.29*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
08:25.45tsumeio_error: you make them?
08:28.24io_errorsometimes.
08:30.47io_errorI need a new headset.
08:31.11Zeeekplantronics
08:31.21io_errorheh :)
08:31.32io_errorit's 2:30 in the morning, where am I going to get one of those at this time of day?
08:31.51Zeeekfedex
08:31.56io_errorsheesh
08:32.41*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
08:36.03io_errorah, everything seems to be working.
08:36.15*** join/#asterisk Heartwich (n=Miranda@130.228.38.63)
08:37.22*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
08:37.32io_errorI love it when a plan comes together.
08:37.59Zeeekalways have two headsets
08:38.11*** join/#asterisk ribbo (n=riaan@196.36.161.235)
08:38.11Zeeekgreat for debugging, too
08:38.18io_errorZeeek: heh, it's the mic, I have to have it almost IN my mouth for it to pick anything up
08:38.28Zeeekyes that's the first to go
08:38.52Zeeekbut look, I had a sound problem a while back, change heaset and it was ok. Then I changed back and it was still ok!
08:39.05Zeeekso it wasn't the headset but the connector on the PC!
08:39.19io_error393612
08:39.24io_errorer
08:39.38io_errorwrong number :)
08:39.54Zeeek303 is FWD? 612 is the old minneapolis area code
08:40.01Zeeek393
08:40.05io_error393 = fwd
08:40.08Zeeek612 was thre echo test?
08:40.08io_errorat least on my *
08:40.15io_error613 is the echo test
08:40.15Zeeekor the time
08:40.24Zeeekdate/time
08:40.34Zeeek666 is the echo test on mine
08:40.45io_errorHm, ok taht's a little better.
08:40.57io_errorthe mic needed to be jacked up in the mixer
08:41.41tsume99999999999999999999999999999999 is mine
08:41.43tsumej/k :)
08:41.46io_errorheh :)
08:41.50Assidalso get this:  end_request: I/O error, dev fd0, sector 0
08:41.50Assidend_request: I/O error, dev fd0, sector 0
08:42.05io_errorAssid: that means ther'es no floppy disk in the floppy disk drive, it's harmless :)
08:42.23io_errorunless you DO have a floppy disk in the drive, then it's a problem :)
08:42.26Assidjust saying i got wayyyyyy too many errors taking place
08:42.30Zeeekheh
08:43.07Assidwish i could reboot my life
08:43.09Assidmake things easier
08:43.41Assidits like anything possible that can go wrong is going wrong
08:43.41ZeeekAssid just don't re-format first
08:45.50trixterreformatting is fun
08:45.54trixterI do it several times a day
08:46.06Zeeekwhy keep running windows then?
08:46.14trixternever said I was
08:46.16Zeeeknyuk, nyuk
08:46.18io_errordd if=/dev/urandom of=/dev/hda
08:46.38trixteradd a bs=1024 or something to make it work better
08:46.51trixteractually doing 1M chunks would prolly be better than 1k
08:46.53Assidrtc: lost some interrupts at 1024Hz.
08:46.57Assidstill getting it :(
08:49.33ZeeekMontpellier
08:50.48Delvarhi, anyone got teh URL for the digium g729 .so's and the register app?
08:51.17trixterftp://ftp.digium.com ?
08:52.20trixterftp://ftp.digium.com/pub/asterisk/g729/  more specifically
08:52.31Delvaryeah thanks i go tit :)
08:52.35Delvargot it*
08:52.38brookshireit's under documentation on g729 page
08:52.39brookshirehttp://www.digium.com/en/supportcenter/documentation/viewdocs/G729
08:53.04Delvarther used to be links form teh digium site, but they dont seem to be there any more
08:53.36Zeeek<PROTECTED>
08:54.31brookshirehttp://kb.digium.com/entry/17/5/
08:54.33brookshirethere is another one
08:55.25Zeeek<PROTECTED>
08:56.19Zeeekyou get there from digium products -> G.729 codec -> ralated information, Documentation -> Readme
09:01.05*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
09:03.54*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@80.178.85.79.adsl.012.net.il)
09:05.18Heartwichhow is the best way to get a status for the extensions in asterix?
09:05.42Zeeekyou mean?
09:06.03*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
09:06.14Heartwichi want to list the status for the phones connected to asterisk. is they ringing, busy aso.
09:06.24Heartwichfor use in php.
09:06.38*** join/#asterisk astra^^ (n=muhajir_@59.145.104.74)
09:06.39Zeeekuse the manager interface
09:07.17Heartwichby making an socket connection to it?
09:07.26Zeeekhttp://www.google.fr/search?q=asterisk+manager+interface&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official
09:07.35Zeeekread a few of these
09:07.56io_errorHeartwich: show channels ?
09:08.05Zeeekyou can just use php on apache or another http server
09:08.23Zeeekor use FOP which does exactly what you said
09:08.28io_erroryeah, FOP is nice :)
09:08.34ZeeekFOP uses the manager interface
09:08.46Heartwichwell, i want to the same that fop does, just without flash.
09:09.08Zeeekstudying how FOP works might help
09:11.15*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)
09:17.14Assidio_error: i interchanged the harddrive
09:17.21Assidbut the one that was working
09:17.24Assidis still working
09:21.11Assiddamn
09:21.14Assidhe just had to leave
09:21.38BugKhamwhy do I keep getting this -- B-channel 0/xx successfully restarted on span 1"?
09:22.08edwar64896[Bugkham] this is your PRI resetting.
09:22.17edwar64896you can change the default timeout for PRI resets in your zapata.conf file.
09:22.26edwar64896its perfectly normal behaviour
09:23.28BugKhamedwar64896: so it has no effect on the current connections/calls?
09:23.34edwar64896nope
09:23.56edwar64896you'll notice that any b channel in use doesn't get touched
09:24.51*** join/#asterisk y-man (n=y-man@udp115909uds.hawaiiantel.net)
09:24.54BugKhamedwar64896: okay
09:31.53*** join/#asterisk sundancer (n=marko@shyana.perkmandlc.org)
09:32.59sundancerHi.. can anyone tell me what software other than bristuff also supports HFC (cologne) PCI cards
09:33.23sundancerBecause in bristuff i cant assign MSN number i want to use with asterisk
09:33.39*** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
09:33.42sundancerI have two MSN numbers but HFC card takes over both of them
09:33.45sundancerAnd i dont want that
09:33.51astra^^how will i make * accept calls from an ser sip server and reroute to an outbound proxy
09:34.58Greek-Boywhere can one find recorded sounds?
09:35.55x86<PROTECTED>
09:36.03Greek-Boylol
09:36.03Greek-Boyk
09:36.38*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:36.40Greek-Boyis it a bad idea to use synthesis to create voice files. (text 2 speech type of thing)
09:38.50BugKhamerr with 1.2.5, there's no applications "Cut" or "SubString"
09:39.20BugKhamwhat do you guys use? executing a system 'cut"?
09:40.16Juggieyou just use the variable name
09:40.37Aurs(Deprecated, use ${variable:a:b} instead)
09:40.37Juggie${EXTEN:2:5}
09:40.47BugKhamohh right
09:40.57BugKhamthanks
09:41.03Aursshow application substring
09:41.32BugKhamAurs: -> Your application(s) is (are) not registered
09:41.40Aursok. hehe
09:41.48Aurson 1.0.9 i get this: (Deprecated, use ${variable:a:b} instead)
09:42.03Aursamong other things
09:42.24*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
09:43.02*** part/#asterisk edwar64896 (n=edwar648@72.83.233.220.exetel.com.au)
09:43.27Zeeeksubstring and x:y do not replace CUT
09:43.49luckyduckhi, is it possible to pass arguments to an agi script which i call inside of an callfile?
09:43.55luckyducki cant find the right syntax
09:44.09*** join/#asterisk exten123 (n=exten@60.49.6.190)
09:44.40luckyducki can call the agi app using the Application keyword
09:44.55luckyduckthe name of the agi script can be passed using the data keyword
09:45.05*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
09:45.16astra^^how will i make * accept calls from an ser sip server and reroute to an outbound proxy
09:45.27luckyducki cant find a way to pass arguments to the agi-script
09:45.28luckyduckany hints?
09:47.03Aursluckyduck: show application agi
09:47.12*** join/#asterisk oej (n=oej@bkkb-gw.bitcon.no)
09:49.20*** join/#asterisk xterminus (n=cmauch@00104bc8bd59.click-network.com)
09:51.05luckyduckAurs: thx, will try to do it that way
09:51.21Aurs<PROTECTED>
09:51.32Aursnp
09:52.39*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
09:57.15*** join/#asterisk snip3r (n=sniper@195.246.199.136)
09:57.26snip3rhi all
09:57.46snip3rZeeek: I wonder if you remember my problem
09:58.08snip3rdo you have a couple of minutes to figure out what happened?
09:58.14Zeeekgod no, what was it?
09:58.19snip3r:D
09:58.48snip3rthe point is that if a client is behind NAT, * doesn't even TRY to send the RTP
09:59.14snip3rif it's outside NAT, everything is OK
09:59.43luckyduckAurs: yeah, i saw it. didnt noticed the show command before. works like charm, thanks
10:00.13snip3rI've checked this with tcpdump and saw this errrr... odd behavior :)
10:01.04ZeeekI'm afraid I don't have a clue for you at all
10:01.16snip3ruh
10:01.18*** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl)
10:01.28*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
10:01.34snip3rthe issue is that NATed hosts never get audio
10:01.56ZeeekI haven't seen that here
10:02.06snip3rthe usual fixes (nat=yes etc) are setup
10:02.21*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
10:02.49snip3rI've posted a call setup trace at cozmo.hu/log.txt
10:03.12snip3rit's a bit old, however
10:03.14snip3r:)
10:03.29snip3rbut the situation hasn't changed since
10:04.34snip3rso: if I try to connect with a client behind NAT, I get no audio
10:05.00snip3rif I call from a public IP, works like a charm
10:05.01Zeeeksometimes it's a matter of the NAT router just not working right
10:05.11snip3rthat's the point
10:05.21Zeeekin which case nothing to do but replace it - what is the router?
10:05.44snip3rI've tcpdump'd a call to a NAT device and * doesn't try to send the RTP at all!
10:06.00Zeeekanswer?
10:06.02snip3rthe router is a cheapo X-Micro one
10:06.11snip3rdoing port restricted cone
10:06.36Zeeeklook no further - change the router and don't look back
10:06.44snip3r:)
10:07.10snip3rI've got a Linksys WRT54GP2 at hand
10:07.31Zeeekuse that
10:07.47snip3rbut the really weird part is that I managed to use * behind this router
10:08.47snip3rand now I'm confused 'cause I did very little config to * to achive full NAT penetration a week ago
10:09.20snip3rand now it doesn't want to to the same after several days of debugging
10:09.30snip3rto do the same, sry
10:09.58snip3ranyway, trying to change the router
10:11.20Zeeekk
10:11.20*** join/#asterisk stefan (i=stefan@home.stefan.id.au)
10:19.21RoyK~nickometer snip3r
10:19.25RoyK:)
10:20.30dpryo~nickometer RoyK
10:20.33*** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)
10:20.41trixterthere you have it
10:20.44CurusIs it possible to ask a queue how many people are currently waiting?
10:20.47trixteran authoritative rsponse
10:21.35*** join/#asterisk backblue (n=igor@82.102.1.42)
10:22.43trixterCurus: yes its possible, depending on various factors
10:22.54Zeeekshit, curl actually works!
10:23.12Curustrixter: How?
10:23.32CurusPreferably something I can look at in extensions.conf
10:23.48trixterI never said that was one of the factors
10:26.59astra^^do we have to buy g723 codec for givin routes to g723..
10:27.09astra^^* supports g723 right?
10:28.43RoyKpassthrough
10:28.44RoyKonly
10:28.56astra^^ok do we get to buy g723
10:28.59RoyKor perhaps with that transcoder card that came out
10:29.09RoyKthere is no software solution for g.723.1
10:29.11RoyKfor asterisk
10:29.15astra^^ok
10:29.18RoyKbut why do you need it?
10:29.23RoyKg.729a does the job
10:29.38*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
10:29.39RoyKdifference is only like 10% more data
10:29.42RoyKwhen conting overhead
10:29.51RoyKcounting.....
10:30.45astra^^i have g729 al so
10:30.45RoyKg.729a is 8kbps + 16kbps overhead. g.723.1 is 5.3kbps + 16kbps overhead.....
10:30.58RoyKrather patch up asterisk to use larger packets than the usual 20ms
10:31.02RoyKthere is a patch
10:31.03RoyKsomewhere
10:31.08astra^^ok .. .
10:33.31Zeeekcan | be escaped in a function ?
10:36.45Zeeekworse yet, set(name=Hey|Now) will set it to Hey
10:36.59ZeeekMoral of the story, don'(t use | as a delimiter
10:37.39*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
10:38.17RoyKZeeek: usually set(Zeeek=quite\|stupid) will work :P
10:38.32Zeeekyou've shown yourself amply on the ML
10:38.51RoyKque?
10:39.20X-RobRoyK, I think he hates you.
10:39.26*** join/#asterisk bmg505 (n=leon@dsl-165-157-13.telkomadsl.co.za)
10:39.42Zeeekbed time in .au, no?
10:39.50RoyK.no
10:39.51X-Rob8:30pm, it's only early.
10:39.58X-Robhint: /ctcp x-rob time
10:40.03Zeeekthat's past my bedtime
10:40.11snip3r8:30 PM?
10:40.14Zeeekyep
10:40.17snip3r:D
10:40.26ZeeekI'm only 5
10:40.31snip3rhuh
10:40.39snip3rthat's weird
10:40.48ZeeekHarvard has accepted me for next year
10:40.51snip3rhe's 5 and does * better than me
10:40.56ZeeekI'll be 6 then
10:41.03Zeeekmuhahah
10:41.11snip3rZeeek: what is your IQ?
10:41.16Zeeek42
10:41.24snip3r*10^3?
10:41.33Zeeekor ASC('@') + 2;
10:41.37snip3ruhh
10:41.44Zeeekoopos no good
10:41.54snip3r'xcuse me?
10:42.35Zeeekasc('@') is 64 in hex, 40
10:42.45snip3r:)
10:42.53*** join/#asterisk g0mb0 (n=g0mb0@external.micom.mng.net)
10:42.56Zeeekso what about this router
10:42.56snip3rthat's sg I already know :)
10:43.05snip3rchanging it soon
10:43.25snip3rbut first I started an * install on one of my linux boxes
10:43.26g0mb0Can Asterisk support R2 digital compelled signaling with Sangoma card?
10:45.52MavvieOur PABX supporter says that the Alcatel 4400 takes time from the network. Anybody here knows how that works?
10:46.00Mavviethe phone network, not the computer netowkr
10:46.22snip3rwhat is your question?
10:46.37snip3rthe 4400 is pretty big
10:46.47Assidumm . is 726 enabled in asterisk by default
10:47.00snip3rcheck sip.conf
10:47.17snip3rand look for the row: 'allow='
10:47.52snip3rif I recall correctly, pcmu, pcma and gsm are enabled by default
10:48.05Assidshow translation shows it
10:48.07Assidthats why
10:48.08snip3rbut if you get your hands on a CLI
10:48.15snip3ryep
10:48.17snip3r:)
10:48.20Zeeekshow translation is just a chart
10:48.27snip3rshow codecs?
10:48.57ambrientog0mb0, yes it can.
10:49.03g0mb0Hi all, Can Asterisk support R2 digital compelled signaling with Sangoma card?
10:49.04Zeeeksame thing, a chart
10:49.11ambrientog0mb0, yes it can.
10:49.23Zeeekthe little pbx that could
10:49.25Assidi have allow all
10:49.34Assidfor my sip in my context
10:49.37RoyKg0mb0: see http://soft-switch.org/
10:49.50g0mb0I see
10:49.54Assidbut cant seem to make a call with 726-24
10:50.54snip3rAssid: are you sure that both endpoints support .726?
10:51.07snip3rLinksys/Sipura stuff?
10:51.26Assidsnip3r: yes.. linksys pap2
10:51.32snip3rboth?
10:51.40Assiderr.. connecting to asterisk
10:52.01snip3ranyway, you should issue a 'sip debug peer [peername]' command
10:52.16snip3rasterisk -r on the same machine
10:53.48snip3rAssid: check this out: http://www.voip-info.org/wiki-ITU+G.726
10:54.09snip3r^^ this site rocks
10:56.47*** join/#asterisk snip3r (n=sniper@195.246.199.136)
10:57.32RoyKAssid: iirc asterisk only supports g.726-32
10:58.47Assidokay heres something interesting
10:58.48AssidCapabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)
10:58.57Assidpeer supports ilbc?
10:59.27Assidi dont see ilbc in the list
11:00.07snip3rin which list?
11:00.22Assidin the pap2 web interface
11:00.38Assidokay now time to test a few calls
11:01.00Assidanyone got free calling?
11:01.48snip3ryep, that's odd
11:02.39Assidhow the hell does one conference in this pap2?
11:07.36*** join/#asterisk fulgas (n=fulgas@209.8.233.252)
11:07.52*** join/#asterisk michael-i (i=user@141.41.38.185)
11:09.43BugKhamany reasons why would my ISDN PRI-E1 does not support  "Playback(file,noanswer)?
11:10.02BugKhamerr no "would"
11:15.48*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
11:19.06*** join/#asterisk Qorky (n=spam@202.173.160.26)
11:20.19CurusBugKham: Did you remember to answer the call before trying to play?
11:23.47twislaare the application names in extensions.conf CaseSensitive ?
11:26.04*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
11:26.36Qorkywheres a howto install the cvs-head please?
11:28.01kaldemarhttp://www.asterisk.org/download <-- take a look under "SVN repository"
11:30.32QorkyI must be behind the times. I dont have svn
11:33.18FrogzooQorky: subversion - lets you check in a bunch of files under one change
11:33.35X-RobBugKham, 'Playback(file,noanswer)' isn't valid
11:34.07X-Robespecailly on an E1
11:36.20*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
11:36.30BugKhamX-Rob: so, it's valid on a T1?
11:36.35X-RobNo
11:36.46X-Robit's valid on a SIP or IAX device with the 'r' flag on it
11:36.55X-Robjust about everything else you have to Answer first
11:37.26*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)
11:38.15BugKhamX-Rob: I was about to put an announcement to a caller without having them paid for the call
11:38.22X-RobCan't do it
11:44.21CurusSome providers accept it
11:44.28CurusMost don't
11:46.36*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
11:46.56BugKhamCurus: okay
11:47.57BugKhamCurus: I'm wondering if it's the same way as putting truetones for mobile phones
11:54.32*** join/#asterisk in-side (n=bsd-desk@213.58.69.127)
11:54.34in-sideHI
11:54.41in-sideI have a strange problem here
11:54.51in-sideI would like to know if anybody has a glue for it :(
11:55.15*** join/#asterisk linstar (n=achu@220.225.191.18)
11:55.17in-sidewhat happens is my * box in a normal call send byes ok to my sip gw
11:55.19linstarhi
11:55.44linstaris there is any good documentation about connecting two asterisk servers using sip?
11:55.46in-sidebut when I try to use the autohangup in function dial()
11:55.59*** join/#asterisk psk (n=psk@golia.caltanet.it)
11:56.10in-sideit gimme a delay sending the bye or simple ignores it
11:56.32astra^^WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4)
11:56.37astra^^y do i get this msg
11:56.48linstaris there is any good documentation about connecting two asterisk servers using sip?
11:57.09in-sidedoes any one has a clue why when a uac hangup I can receive every bye and not receive when asterisk does the hangup ?
11:57.36in-sidelinstar: you mean in reduntancy ?
11:58.03*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
11:58.23linstarin-side : I found some documentation about connecting two asterisk servers I mean peer connection
11:58.56linstarin-side: Asterisk - dual servers
11:59.03linstarconfiguration
12:00.47astra^^WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4)
12:00.55linstar<PROTECTED>
12:01.03astra^^wht mgt b the problem
12:03.00xterminusis it possible for iax phones (idefisk,etc) to make sip address calls (eg sip:extension@domain.name) through asterisk?
12:05.57xterminuswhen i try - asterisk doesn't seem to even try to make the call... in logs all i see is "Mar 20 04:04:46 NOTICE[22966]: chan_iax2.c:7053 socket_process: Rejected connect attempt from 10.0.0.254, who was trying to reach '9586111@mutual.bcwireless.net'" for example
12:06.15*** join/#asterisk lorinc (n=ang@caracas-2449.adsl.interware.hu)
12:07.54*** join/#asterisk chris_ast (n=Administ@59.93.56.163)
12:08.22chris_astanyone there
12:08.38RoyK<PROTECTED>
12:08.52chris_asthi RoyK
12:09.13*** join/#asterisk core-ix (n=ivo@pirus.securax.be)
12:09.26linstardo anybody familiar with Asterisk - dual servers?
12:09.59chris_astRoyK, I have a doubt in Astersik AGI and PHP, can you help me
12:11.57linstaris  anybody familiar with Asterisk - dual servers?
12:13.12xterminuslinstar, ask your question (i dont run dual servers - but i'm pretty good at asterisk in general)
12:14.39RoyKchris_ast: i don§t use php with agi. sorry
12:15.12RoyKlinstar: what do you want to do?
12:15.20RoyKfailover? load balance?
12:15.21*** join/#asterisk eliel (n=eliel@200.123.183.89)
12:15.28RoyKsandwitch the servers?
12:15.39xterminusagi + perl/bash here - dont use php for much of anything
12:17.10*** join/#asterisk saftsack (n=saftsack@p54A7FDDF.dip.t-dialin.net)
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12:19.26in-sidelinstar: sorry I wasn't here
12:19.41in-sidelinstar: what is your problem ?
12:20.57michael-ianyone have a quick introduction to writing/compiling your own channel driver?  If I finish mine up and put it into /channels will it be compiled?
12:22.58*** join/#asterisk sanee (n=sanee@82.117.210.45)
12:23.43xterminusanyone know of a website/something where you can get it to call an arbitrary sip number (for testing incoming calls?)
12:25.14Curusxterminus: Your favourite phone should do that, if it supports IP dialing
12:26.14linstarin-side : I have created two asterisk servers and now want to connect them
12:26.24*** join/#asterisk fourcheeze (n=rich@82.153.215.21)
12:26.38fourcheezeanyone know of an ATA that does SIP and Skype ?
12:26.41xterminusCurus, yah - but if its on the inside of a nat network, it really cant test for incoming nat traversal very well
12:27.02*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
12:27.19linstarin-side  : If I dial one extension of the 2nd server from the 1st server I want it to work
12:27.39linstarin-side : Is it possible with sip configuration?
12:32.35*** join/#asterisk basti__ (n=basti@dsl-220-253-65-42.NSW.netspace.net.au)
12:33.05xterminuslinstar: with asterisk, use iax.conf to define a context for the remote asterisk server
12:34.55astra^^WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4)
12:34.57xterminuslinstar, then dial those numbers in extensions.conf like Dial(IAX2/hostname/{EXTEN}@remotecontext)
12:35.31chris_astxterminus, I have a very small AGI in PHP and I need to know how can I read a argument passed to it?
12:37.16linstarxterminus : I have no hardware installed
12:37.35linstar<PROTECTED>
12:38.41linstarxterminus : I mean I have configured the asterisk servers with sip only
12:40.04xterminuslinstar, concept is the same
12:40.26xterminuslilo, dialing command will be a little different is all
12:40.28Zeeek"asterisk is now deprecated. We recommend you purchase a Nortel pbx and quit fooling around"
12:41.19xterminuschris_ast, not sure - my experience is limited to arguments with websites - not php (looking)
12:41.37linstarxterminus : can I configure it same in sip.conf and extensions.conf?
12:41.58linstarxterminus : Can you give me good URL for reference?
12:42.01Zeeekchris_ast aren'(t all args passed on stdin ?
12:42.42astra^^WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4)
12:43.39xterminuslinstar, i dont have a url on hand other than the asterisk bible (voip-info.org)
12:44.16kaldemarlinstar: http://www.voip-info.org/wiki-Asterisk+-+dual+servers
12:44.23*** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1)
12:45.23ZeeekI never figured out how switch works
12:45.40Zeeekor rather how to limit it
12:46.01Zeeekbut calling the other server as a "user" works fine
12:47.15xterminusZeeek, it looks like with php, you have to establish handes to deal with STDIN
12:47.28xterminus,
12:47.28xterminusSTDOUT
12:47.28xterminus, and STDERR - easist way to deal with those variables would be to read them into a hash probably
12:47.28xterminuswhoops
12:47.48xterminusxchat wierdness
12:48.00Zeeekyeah there is an example of almost every language (well three or four)
12:48.26Zeeekyou could also use ENV type args
12:49.22*** join/#asterisk coppice (n=chatzill@91.203.17.210.dyn.pacific.net.hk)
12:49.35xterminusbah - hashes are better (of course - this is coming from a perl fan =)
12:53.27*** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net)
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13:01.37tzafrirhow do I authenticate SIP calls by the IP of the calling gateway?
13:02.00tzafrirlet's assume I can trust the IP address
13:02.08tzafrir(not to be spoofed)
13:02.31Zeeekif it was spoofed, they'd never get an answer :)
13:03.03*** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br)
13:03.48tzafrirI can change the default context for all non-identified calls
13:04.56tzafrirBut then I can't tell between two specific gateways
13:05.16*** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F0A92.dip0.t-ipconnect.de)
13:05.17_Paulo_coppice, how are you?
13:05.24coppiceOK
13:05.41Zeeekgood
13:05.43*** join/#asterisk medusaXX (i=proxy@p54A9A5E7.dip0.t-ipconnect.de)
13:06.01[ProB]CrazyMancoppice: if my faxes look like : http://www.roterschnee.org/failure.tif is this an libtiff issue ?
13:06.10_Paulo_coppice, I think need some help with libmfcr2
13:06.31*** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua)
13:07.48_Paulo_coppice, I was trying to send a clear back right after answer, but could not figure out where should I insert this.
13:08.33_Paulo_[ProB]CrazyMan, this is with rx_fax or tx_fax?
13:08.55[ProB]CrazyMan_Paulo_: rx_fax
13:09.00coppiceif you want to clear back after answer just drop the call
13:09.24*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:09.30_Paulo_coppice, there is a provision in some pbx in brazil
13:09.54_Paulo_coppice, to avoid collect calls they send a clearback after answer
13:10.27_Paulo_I want to send the clearback without drop the call
13:11.17coppiceyeah, I know about that trick
13:11.17_Paulo_[ProB]CrazyMan, what version of libspandsp are you using?
13:12.11_Paulo_coppice, I tried to drop the cal inside chan_unicall, but I cant re-answer, because the call is in incorrect state
13:12.13coppice[ProB]Crazyman: rxfax didn't produce that TIFF file
13:12.17[ProB]CrazyMan_Paulo_: spandsp-0.0.2pre25
13:13.36[ProB]CrazyMancoppice: not?
13:14.10coppiceSoftware (305) ASCII (2) 27<Adobe Photoshop CS Windo ...>
13:14.23[ProB]CrazyMancoppice: yes thats right ..
13:14.31[ProB]CrazyManits not the orginal tif file ..
13:14.35_Paulo_coppice, so, is libmfcr2 the right place to do this?
13:14.46coppiceso its worthless crap, that is wasting my time
13:14.53[ProB]CrazyManwait ..
13:17.08coppice_Paulo_ I think mfcr2.c needs to go into a special timed release-that-is-not-really-a-release state. You can easily make the current mfcr2 code tolerate these short  clears, by adjusting a timer who's name I have forgotten. I haven't yet provided a means to generate them
13:17.33*** join/#asterisk brookshire (n=mbrooks@gateway.digium.com)
13:18.31*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
13:18.31_Paulo_coppice, I've done that, tolerate these short clears... with mfcr2->clear_back_persistence_check = 1000
13:18.54_Paulo_coppice, What I wanto to do now is generate these myself.
13:19.39coppiceI'll get around to doing it at some point, but I'm ripping the unicall stuff apart at the moment to restructure things
13:21.42*** join/#asterisk fugitivo (n=ajf@201.255.177.90)
13:21.43_Paulo_coppice, I wold like to help you
13:21.50fugitivohello
13:21.52coppice[ProB]CrazyMan: that looks like things fell apart due to timing slips
13:22.01[ProB]CrazyManhm
13:22.20_Paulo_[ProB]CrazyMan, use the last IAXmodem stuff
13:22.24[ProB]CrazyMancoppice: so what could I do against timig slips ?
13:22.40coppicefix them. it is nothing to do with me
13:23.14[ProB]CrazyMantiming slips -> bristuff related ?
13:25.01_Paulo_[ProB]CrazyMan, timing slips can be hardware related.
13:25.31_Paulo_[ProB]CrazyMan, bad network interface cards
13:25.38_Paulo_[ProB]CrazyMan, bad zaptel hardware
13:25.50*** join/#asterisk illuy (n=assdf@85-65-123-85.barak-online.net)
13:25.54_Paulo_[ProB]CrazyMan, shared interrupts
13:26.23_Paulo_[ProB]CrazyMan, too much interrupts
13:26.36[ProB]CrazyMan_Paulo_: there is just one quadbri card inside
13:26.59_Paulo_[ProB]CrazyMan, By the way, app_rxfax works fine for me with a digium TE110P
13:27.54_Paulo_[ProB]CrazyMan, I had timing problems with app_txfax, but I'm using IAXmodem+hylafax now and it works fine.
13:29.47_Paulo_[ProB]CrazyMan, for my needs, iaxmodem+hylafax proved to be a better fit then app_txfax, anyway, because I dont need to reinvent the wheel (the error handling).
13:30.50*** join/#asterisk Skymarshal (n=Skymarsc@p54AF2C85.dip0.t-ipconnect.de)
13:31.13_Paulo_coppice, what "set_mf_signal()" does?
13:31.55coppiceif you haven't worked that out, I don't think you are going to get very far with solving your problem
13:31.57*** part/#asterisk linstar (n=achu@220.225.191.18)
13:32.21SkymarshalI have an AGI problem. I want to use "STREAM FILE marryme 15000 3" but it doesn't work (CLI says that it is playing the file but I can not hear it). I can use "EXEC Playback marryme" but not the "STREAM FILE" command. Any idea why?
13:32.34_Paulo_coppice, I tried to put a set_mf_signal(uc, ch, mfcr2->back_abcd_clear_back) after the set_abcd_signal(uc, ch, mfcr2->back_abcd_answer)
13:33.27coppiceseems like you don't understand how MFC/R2 works
13:33.57_Paulo_coppice, Yes, you are right
13:34.15coppicewell, that makes it kinda hard to do development :-)
13:35.05coppiceclue: once the call is answered, the MF signals are not used any more
13:35.52_Paulo_coppice, where should I be looking?
13:37.16coppiceafter
13:37.18coppice<PROTECTED>
13:37.19coppiceyou need a timer to wait for a few hundred ms, then you need
13:37.21coppice<PROTECTED>
13:37.22coppiceand another few hundred ms wait, then
13:37.24coppice<PROTECTED>
13:37.44_Paulo_coppice, thanks.
13:43.09*** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F11B7.dip0.t-ipconnect.de)
13:44.18*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
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13:55.50_Paulo_coppice, Is uc_schedule_event the rigth way to wait a few ms inside libmfcr2?
13:56.29coppiceyep. if you look at all the other timers, they use that
13:56.49*** join/#asterisk Mauro__ (n=mauro@oliver.altascumbres.cl)
13:56.54Mauro__Hi
13:57.04Mauro__I have a noob question :P
13:57.32Mauro__what is the easiest way to develop a web app with asterisk?
13:57.44Zeeekusing a bounty
13:57.50MikeJ__HEH!!
13:58.26Mauro__:P
13:58.40_Paulo_coppice, thanks again.
13:59.03MikeJ__Mauro__, pick your fav web dev language.... develop
13:59.30*** join/#asterisk L|NUX (n=linux@202.5.145.58)
13:59.43Mauro__php is fine?
14:01.17*** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe)
14:01.35*** join/#asterisk apardo (n=apardo@87.218.45.153)
14:02.02MikeJ__sure
14:02.10MikeJ__depends on what you need to do
14:02.19MikeJ__there is a php asterisk mod somewhere
14:02.22*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
14:02.29MikeJ__ast_php or somthing like that
14:02.31*** join/#asterisk sergeus (n=s@195.112.98.13)
14:02.31MikeJ__never used it..
14:06.14astra^^WARNING[25769]: chan_sip.c:2530 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4)
14:06.35kardecallanI have MFC-5C class, I want to integrate with asterisk. Is there anybody that can help me?
14:06.47astra^^i get tis error wen a call is placed frm a ser .. to * we are using g729 at both ends
14:07.05*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
14:07.27*** join/#asterisk digg10 (n=john@206-248-152-244.dsl.teksavvy.com)
14:09.33_Paulo_coppice, I looked at all the other timers, all set the callback function. Is it mandatory?
14:10.11_Paulo_(my brain hurts from reading C code)
14:10.51Mauro__where can I read the asterisk api documentation?
14:11.41*** join/#asterisk PumpkinPie (n=a@unaffiliated/PumpkinPie)
14:11.44PumpkinPieHello
14:11.48xterminusis it possible for iax phones (idefisk,etc) to make sip address calls (eg sip:extension@domain.name) through asterisk?
14:12.41*** join/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net)
14:13.12PumpkinPieI have voicepulse service but not 'connect voicepulse' service. Can I still use asterisk ?
14:13.51kaldemarMauro__: you can find some at voip-info.org
14:14.42Mauro__thanks
14:15.07_Paulo_kardecallan, coppice wrote libunicall
14:15.18MikeJ__Mauro__, fore real api, there is doxygen in the code, it's onthe asterisk.org site
14:15.34*** join/#asterisk zotz (n=zotz@24.231.32.85)
14:17.40Mauro__the real api is written in C?
14:17.54MikeJ__y
14:23.40PumpkinPiecan I replace hardware based voip products with software?
14:24.17*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
14:24.21_Paulo_PumpkinPie, software runs in hardware.
14:24.30*** part/#asterisk babyhuey (n=justin@ip-131-123-81-11.housing.res.kent.edu)
14:24.54_Paulo_PumpkinPie, you can replace almost any appliance with a PC
14:25.39kardecallan_Paulo_ I need to write it?
14:26.24_Paulo_kardecallan, are you into #asteriskbrasil.org?
14:26.45PumpkinPiePaulo: im talking about asterisk to repalce my sipura device specifically
14:27.18ZeeekPumpkinPie yes
14:27.58Creperumare there some apps for using * as a telemarketing tool or CATI?
14:28.14PumpkinPieZeeek: I think voicepulse gathers the serial number off of my sipura device so it knows which is my account? etc?
14:28.17_Paulo_PumpkinPie, replace a "no mobile parts" appliance with a PC has some tradeofs
14:28.41PumpkinPiehow do I configure the serial number with asterisk
14:29.04*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:29.04Zeeekuse voicepulse connect and IAX
14:29.18PumpkinPiei dont want to use 'voicepulse connect'
14:29.40PumpkinPietheres no point in me asking if it can replace my hardware if I use 'voicepulse connect'
14:29.43*** join/#asterisk Emrah (n=user@adslgva0491.worldcom.ch)
14:29.53EmrahHello!
14:30.19ZeeekI don't use vp SIP so I don't know about serial numbers. MOst SIP providers don't do that AFAIK
14:30.25EmrahI'm wondering, do you know how it is possible to generate a call from asterisk and starting an application when the remote party peak up the phone?
14:30.37Zeeekbut just so you understand, one can use an IAX phone with voicepulse connect so there was nothing funny about my answer
14:30.47EmrahI mean from a way or another, I'd like to receive a call, like a call back application, and be able to access my voicemail
14:31.49PumpkinPiei dont care what you can or cant use with 'voicepulse connect'
14:31.49PumpkinPiethats not what I have
14:31.49kmilitzerCan someone tell me how I can find out which codecs are actually installed and usable in my asterisk?
14:31.58ZeeekI do,'t know any reason why voicepulse normal service wouldn't work
14:32.08Zeeekbut then I don't have that service
14:32.37Emrahkmilitzer: show codecs
14:32.46Emrahor something like that let me try
14:32.54Creperumppl, my client wants to build a callcentre for outgoing calls - he wants to call his clients periodically - how can it be done with *? are there some scripts?
14:33.08PumpkinPieit identifies my sipura unit with its serial number yes? or with what? and how do I setup that serial number with asterisk?
14:33.21kmilitzerThis shows G.729A which I haven't installed (that's the version where u need the license for, right?)
14:33.28jsharpCreperum:  Yes and yes.
14:33.46Zeeekshow codecs just shows a table of them - not nec installed
14:33.46Emrahkmilitzer: in your * cli ti sow codecs
14:33.55Creperumjsharp: ok, where they can be found?
14:34.09Zeeekin modules maybe?
14:34.13PumpkinPiehow do I make asterisk appear/work as a replacement of my sipura unit ?
14:34.20digg10hello
14:34.39RoyK~seen zoa
14:34.41jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 4d 21h 51m 24s ago, saying: 'it looks kinda suspicious :p'.
14:34.43Emrahsorry for my wrong answer if it is one
14:35.05jsharpCreperum:  Depends on how complex you want the system to be.  Asterisk has some basic autodial out functions.  But if you want something like a predictive dialer, you'll need some additional software.
14:35.08EmrahAnyone has an idea about a way to make Asterisk calling a number from the cli or the manager and then starting an application?
14:35.30ZeeekEmrah if it's voicemail, just put the voicemail number in the call file
14:35.32*** join/#asterisk coppice (n=chatzill@21.202.17.210.dyn.pacific.net.hk)
14:35.47EmrahWhat do you mean the call file?
14:36.02Creperumjsharp: i know only about .call files... is there more complex soft?
14:36.24ZeeekEmrah I must have mixed up your question with someone lese's
14:36.27jsharpYes.  There are companies who have build full-scale telemarketing level predictive dialers.
14:36.39jsharpAnd call centre packages.
14:36.43jsharpFor Asterisk.
14:37.16Creperumjsharp: of course, it's proprietary soft..
14:37.40jsharpI think there's a GPL package.
14:37.58*** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
14:38.07Creperumjsharp: please, can you name any?
14:38.17digg10how can i put some conditions in the dialplan?
14:38.26*** join/#asterisk austinnichols101 (n=austinni@70.46.69.131)
14:38.45digg10i also need to query other databases while in the dialplan
14:38.57*** join/#asterisk exten123 (n=exten@202.133.101.88)
14:39.14*** join/#asterisk cuco (n=diego@local.xorcom.com)
14:39.22Tilihey is there any idea of enabling H323 channel to let Asterisk act as 3G gateway
14:40.51HamYaIis FC5 released today?
14:42.20exten123ya
14:42.35io_errorI'm already running it :)
14:43.14exten123io_error, any thing that need to change in order run asterisk? or just like install in core 4?
14:43.46io_errorShould be the same as always.
14:44.10exten123io_error, what advantages u give after using fc5?
14:44.23io_errorOh, it's nice :)
14:44.39PumpkinPiehow do I make asterisk appear/work as a replacement of my sipura unit ?
14:45.13*** join/#asterisk core-ix (n=ivo@pirus.securax.be)
14:45.59exten123PumpkinPie, buy single channel fxo port card. and install in 1 of ur pc that run linux and install asterisk inside that pc
14:46.00HamYaIio_error: I can't see in the download site, where did you get it from?
14:46.52PumpkinPieexten123 cant I not use hardware?
14:47.22PumpkinPieI want to replace my sipura with software
14:47.42*** join/#asterisk tmjb (n=tmjb@www.grappoloin.com)
14:47.48io_errorPumpkinPie: just set up a SIP trunk to your provider
14:47.55*** join/#asterisk k31th (n=keith@87.117.194.66)
14:48.11snip3rZeeek: are you around?
14:48.20Zeeekjust leavinjg
14:48.22snip3r:)
14:48.23k31thbit of a longshot this but does anyone use speex ?
14:48.28Zeeekchanged router?
14:48.43snip3rI've installed * on another box and it works like a charm
14:49.00Zeeekhmmmmm odd. But good in a way
14:49.02tmjbhello,could some one recomend me what cisco phone is the best for Asterix PBX tnx
14:49.09Skid7940/60 ?
14:49.12Skidthey both work well for me
14:49.15snip3rtried the previous one with the Linksys router, but without any luck
14:49.18ZeeekSipura 940
14:49.26snip3rLOL
14:49.28snip3r:)
14:49.34*** join/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net)
14:49.40snip3rSipura is Cisco, in a certain sense
14:49.50Skidya, *cough*
14:50.17backblueSipura is cisco? *g*
14:50.31tmjbSipura is Cisco ??
14:50.35snip3r...
14:50.38Skidno.
14:50.42snip3rLinksys acquired Sipura
14:50.51k31thsipura / linksys / cisco
14:50.53Skidanyway, imo go with 7940/60 with sip 7.4
14:50.54snip3rLinksys is a division of Cisco
14:50.57Skid7.5 == buggy
14:51.04PumpkinPiewhere can I find info on 'sip trunk' setup ?
14:51.10backblueits not the same
14:51.11snip3r7960 has a nice huge screen
14:51.15k31ththe sipura phones look the same as the cisco ones work the same less features basically
14:51.17backblueas linksys its not cisco
14:51.20Skidi've heard there's a 7970 sip image too
14:51.23Skidwhicih i need to get hold of
14:51.28exten123any one got any idea make asterisk have a future like live call monitoring another party conversations? does u guy thing meetme can dun this, or actually got others existing function work for that.
14:51.29backblueeven if they are all in the same company
14:51.30Skidust cant be arsed :)
14:51.48io_errorHamYaI: download.fedora.redhat.com
14:51.51snip3rhold of what?
14:52.01snip3r^^ Skid?
14:52.13Skid7970 sip
14:52.16snip3rIC
14:52.22Skidsomeone mentioned it in here two weeks back
14:52.26snip3rum
14:52.50Skidone thing that really annoys me, about cisco phones are the logo displays tend to work on and off
14:52.59Skidif i reboot, it'll be unable to locate http server
14:53.02Skidreboot again, fine
14:53.07Skidreboot later, same :P
14:53.08Skidetc etc
14:53.20snip3rI need something like a 48V adapter or a PoE converter for the 7960
14:54.16snip3rdo you know an on-line shop that sells Cisco PSU's?
14:54.21Skiderm
14:54.25Skidthere's some chap on ebay that sells them new
14:54.30Skidi bought a few from him
14:54.36snip3rgreat
14:54.38Skidlike 15 gbp
14:54.56Skiddepends on how many phones you have, rather than going for a poe switch i guess
14:55.03snip3rI've got a pair of these nice phones, but need some adapters
14:55.27snip3rhardly can wait to get my hands on 'em
14:55.40Skid:)
14:56.08Skidi need to sort out an alternative provider now, the morons who i currently have support is a waste of time
14:56.17Skidtook me 2 weeks to get 3 new lo-call numbers
14:56.25Skidand now they wont sodding work with their shitty control panel
14:56.49SkidI peer with about 3 i think, so i must try them first for the amount of waffling my girlfriend does heh
14:57.27tmjbis cisco 7970 or some other 100% compatible with asterisk?
14:57.43Skid7940/60's are
14:57.45Skid(SIP)
14:57.49Skid7912 too
14:57.55Skidwe've usd all those on our asterisk box
14:58.05Skidright from 1.0.7
14:58.09austinnichols101snip3r: voipsupply.com
14:58.25tmjbtnx Skid
14:58.25*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
14:58.29*** join/#asterisk oej (n=oej@bkkb-gw.bitcon.no)
14:58.29Skidnp
14:58.32Skid12's are cheap
14:58.35Skid(obv)
14:58.43Skid40's are cheaper than 60's bu tonly have 2 lines
14:58.55Skidsuppose it depends what your needs are though
14:59.08austinnichols101having 2 lines may not be an issue because you only use one for multiple calls in a basic setup
14:59.12Skidagain, there's people that sell 10/100 lots on ebay refurb's
15:02.14*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
15:02.16astra^^<PROTECTED>
15:02.16astra^^<PROTECTED>
15:02.16astra^^<PROTECTED>
15:03.20kmilitzerastra^^: What should that tell us?
15:03.49*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
15:04.02rpmbrrr...
15:04.22Skidkmilitzer: can't you read? :P the sky is yellow is means
15:05.43MRH2Hi anyone know how long it normally takes for patches to be reviewed   (not my patch just a bug i would like to have a fix for) :)
15:05.50astra^^kmilitzer: :)
15:05.50kmilitzerSkid: Sorry, not enough coffee, now that you say it, I can see it too ;)
15:05.53astra^^done
15:06.00astra^^i forgot to put {
15:06.03astra^^heheheh
15:06.11astra^^Thanx anyway
15:10.11*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
15:16.41Nivexoooh!  http://lwn.net/Articles/167897/   Will that deprecate ztdummy for things like MeetMe?
15:18.38*** join/#asterisk miztic (n=gerard@rarcoa.com)
15:19.07kmilitzerAny idea how I can measure the jitter of a call/connection?
15:19.35*** join/#asterisk azzie (n=az@azzie.net)
15:20.04blitzragekmilitzer: iax2 jb debug
15:20.46kmilitzerblitzrage: ... for SIP calls? ;)
15:20.59*** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net)
15:21.41vgsterdoes anyone run chan_fax?
15:22.31exten123does digium analog card support FSK caller id?
15:23.21MikeJ[Laptop]vgster, we run chan_fax
15:23.27MikeJ[Laptop]exten123, yes
15:23.46vgsteri get an error when i build it
15:23.48*** join/#asterisk sergeus (n=s@195.112.98.13)
15:24.03blitzragekmilitzer: not sure for sip -- I don't see anything for it
15:24.04MikeJ[Laptop]your the second one who has told me that, and I can't replicate it
15:24.17MikeJ[Laptop]what kind of box, what asterisk version, what error?
15:25.06MikeJ[Laptop]I think a change in trunk asterisk broke it...
15:25.11*** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
15:25.24vgsterMikeJ[Laptop] > did you get any errors when you built it?
15:25.37MikeJ[Laptop]no, but have not done against truck
15:25.39MikeJ[Laptop]trunk
15:25.46kmilitzerblitzrage: it would be really cool to meassure the jitter between two SIP-Systems ...
15:25.47MikeJ[Laptop]give me the info....
15:25.53tzangerblitzrage: actually iax2 show netstats
15:25.55blitzrageMikeJ[Laptop]: do you happen to know?
15:25.56MikeJ[Laptop]OS, asterisk version, error...
15:25.58tzangeriax2 jb debug is a little more... verbose
15:26.03blitzragetzanger: right -- that too, I forgot about that
15:26.07MikeJ[Laptop]know what?
15:26.08tzangerMikeJ[Laptop]: what problem is this?
15:26.18MikeJ[Laptop]huh?
15:26.24MikeJ[Laptop]I'm lost
15:26.26blitzrageMikeJ[Laptop]: if there is a "netstats" style thing for SIP?
15:26.32blitzragetzanger: you'd probably know
15:26.33*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
15:26.36MikeJ[Laptop]oh, the rtcp stuff
15:26.42MikeJ[Laptop]did that ever get commited?
15:26.48MikeJ[Laptop]that
15:26.53exten123MikeJ[Laptop], it auto support?
15:26.56MikeJ[Laptop]that's where it is.. not sure if it got in or not
15:26.57tzangerno netstats for sip yet
15:27.01tzangerat least none that I"m aware of
15:27.02blitzragenot that I'm aware of ... but yah, that'd be required to measure the jb in SIP right?
15:27.05exten123MikeJ[Laptop], do I need configure any thing?
15:27.29MikeJ[Laptop]exten123, don't even know if it got committed... it was in the bug tracker a long time ago
15:27.35MikeJ[Laptop]search for RTCP
15:27.41MikeJ[Laptop]see what happened to it
15:27.52blitzrageI'm going to find it now
15:28.05JuggieMikeJ[Laptop], it still in progress so far as i know
15:28.09Juggieoej still wants it
15:28.13blitzragehttp://bugs.digium.com/view.php?id=2863
15:28.17blitzragestill open
15:28.28MikeJ[Laptop]wow.. old bug
15:28.35blitzrageaye
15:28.37oejAnd a branch to test :-)
15:29.05Juggiehe speaks
15:29.06blitzrageoej made the last post on March 9th
15:29.10blitzrageall hail oej
15:29.14blitzragehail!
15:29.17*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
15:29.20oejAnd it's part of test-this-branch
15:29.26oejWhich all of you tested during the weekend!
15:29.29oejRight?
15:29.46blitzrageoej: I didn't sleep at all -- I was up with test-this-branch all night... he just wouldn't sleep
15:29.48oejI guess I have to code the missing part of RTCp
15:30.02oejWhere we take some data into the dialplan
15:30.06Juggiewe dont pay you for nothing :)
15:30.10oejOr a cdr variable
15:30.23blitzrageoej: Sweden got a mention this morning in a report I was watchin on CPAC about the Canadian economy, and apparently Sweden figured something out about money :)
15:30.24*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfist.dialup.mindspring.com)
15:30.26oejOh, you don't pay me nothing for nothing?
15:30.40blitzrageI prefer to pay nothing for the world
15:30.45oejOh, if we figured out something about money in Sweden, I missed it since I am in Norway
15:30.46Juggieoej, i told you, setup a paypal account, i'll send beer.
15:30.55tsumeheh
15:30.59oejAnd these guys figured out something about oil...
15:31.14tsumeand 75 for the people in town ;)
15:31.24blitzragetsume: you don't charge enough :D
15:31.34tsumeblitzrage: I know ;(
15:31.44tsumeblitzrage: I have contracts, so it doesn't matter :)
15:31.45blitzragetsume: I'm the same way though
15:31.50blitzragetsume: same
15:33.23*** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net)
15:34.40*** part/#asterisk cuco (n=diego@local.xorcom.com)
15:36.54Zeeekno matter how much I charge they always want to help
15:37.03Zeeekwhich makes it that much harder
15:37.04*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
15:37.40PumpkinPietsume: can I replace my sipura hardware with asterisk, without buying any more hardware?
15:37.58*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:37.58*** mode/#asterisk [+o anthm] by ChanServ
15:38.22ZeeekPumpkinPie what is your sipura hardwireare btw? an ATA?
15:38.54PumpkinPieI think so yes
15:38.56PumpkinPiespa-2000
15:39.35Zeeekand is there some kind of restriction in your vp agreement saying you can only use their hardware or sipura ATA?
15:39.36MikeJ[Laptop]asterisk does not have a magic software fxo port to plug your lines int
15:39.38MikeJ[Laptop]into
15:40.03MikeJ[Laptop]so you can not replace an ata with asterisk without using hardware...
15:40.16PumpkinPieI dont need to plug my phone into it at all
15:40.23MikeJ[Laptop]ummm
15:40.30MikeJ[Laptop]so you don't use the ATA then?
15:40.41PumpkinPieI dont have to, no
15:41.01PumpkinPietheoretically it shouldn't be hard to make software appear as the spa-2000 unit
15:41.10ZeeekPumpkinPie maybe tell us what exact setup you'd want if you could use asterisk and vp? I'm not sure I understand? I thought you just had a phone you wanted to replace
15:41.21PumpkinPieim not sure why you need a fxo
15:41.28PumpkinPienot my phone
15:41.30MikeJ[Laptop]ok.. totally lost me
15:41.33PumpkinPieI dont need the phone
15:41.43PumpkinPieI want to replace the spa-2000 unit with a software solution
15:41.49MikeJ[Laptop]you have an ata.. that you don't use, that you want to replace with asterisk, to do nothign?
15:41.54Zeeekwhat's connected to the sipura?
15:41.55*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfist.dialup.mindspring.com)
15:42.03in-sideHi
15:42.06Skidw 14
15:42.10PumpkinPiemy phone right now
15:42.17in-sidedoes anyone here had worked with agi setcallback ?
15:42.28MikeJ[Laptop]PumpkinPie, well.. you can't do that without hardware
15:42.31Zeeekso you want to plug your sipura into asterisk and have asterisk talk to vp ?
15:42.45PumpkinPieno I dont want to plug my phone into it
15:42.48in-sideany one here are used to deal with perl agi ?
15:42.50PumpkinPieI dont need a phone
15:42.56MikeJ[Laptop]PumpkinPie, so what is it going to do
15:43.00MikeJ[Laptop]just voip to voip?
15:43.05MikeJ[Laptop]or voip to ivr
15:43.12MikeJ[Laptop]no analog phone?
15:43.15PumpkinPieanswer the call and play a sound.. hang up
15:43.15Zeeekthe point is, I think vp will let asterisk connect
15:43.21PumpkinPieno analog or digital phone
15:43.24Zeeekfor whatever you want to do with that
15:43.26Hmmhesaysgod I hate this place
15:43.29MikeJ[Laptop]PumpkinPie, yes, with a voip provider, you can probably do that
15:43.32in-sideI can get setcallback working in perl agi
15:43.43in-sideit result 1 no error
15:43.46MikeJ[Laptop]Hmmhesays, but you love me, right?
15:43.47PumpkinPieI want to replace the spa-2000 unit with a software solution...
15:43.51in-sidebut doesn't execute the sub
15:44.01in-sidedoes anyone have an ideia why that is happening ?
15:44.09MikeJ[Laptop]PumpkinPie, yes, most likely you can.
15:44.10Zeeekwhat is the functionality you want to replace with asterisk?
15:44.23MikeJ[Laptop]do you have passwords and such for your voip account
15:44.31MikeJ[Laptop]Zeeek, he said, play sound, hangup
15:44.39PumpkinPiehow do I configure asterisk to replace my spa-2000 unit?
15:44.54MikeJ[Laptop]PumpkinPie, you read the stuff on the wiki
15:44.55Zeeekyeah you can do that
15:45.00PumpkinPiei dont care if it plays a sound or not.. its just an example
15:45.02MikeJ[Laptop]set up your sip config
15:45.02PumpkinPiehow do I configure asterisk to replace my spa-2000 unit?
15:45.02in-side:S
15:45.09MikeJ[Laptop]~docs
15:45.11jbot[docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
15:45.13Zeeekhave you built asterisk yet PumpkinPie
15:45.15MikeJ[Laptop]you read taht
15:45.22PumpkinPieI looked at the sip config and I see lots of crap I dont have
15:45.35MikeJ[Laptop]PumpkinPie, who is the voip provider
15:45.35Zeeeknah
15:45.37*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
15:45.41Zeeekvoicepulse
15:45.49in-sidegreat no perl guys here?
15:45.52MikeJ[Laptop]they have sample asterisk configs on their site
15:46.02MikeJ[Laptop]in-side, more of an opal guy
15:46.03Zeeekunless they have a weird proxy system or a license agreement, it'll be possible
15:46.12in-sideopal?
15:46.16MikeJ[Laptop]voicepulse is asterisk friendly
15:46.30Zeeekwell, connect is, I'm not so sure about VP vanilla
15:46.32MikeJ[Laptop]pearl, opal... insert other semi precious stone here
15:46.39PumpkinPieMikeJ: that is for their service specifically designed for it... called 'voicepulse connect'
15:46.40MikeJ[Laptop]hmm
15:46.42MikeJ[Laptop]true
15:46.43in-siderotlfl
15:46.43PumpkinPiethats not what I have...
15:46.49in-sidedamn...
15:46.54MikeJ[Laptop]dunno then
15:46.58ZeeekPumpkinPie still, it should be easy to do
15:47.05MikeJ[Laptop]in-side, sure there is somwhere
15:47.08MikeJ[Laptop]what's the prob
15:47.11Zeeekusually the phone/ATA has more shit in the config than asterisk needs
15:47.12in-sidestupid dcumentation
15:47.14MikeJ[Laptop]I can hobble my way a bit
15:47.15*** join/#asterisk epablo (n=epablo@200.109.73.215)
15:47.29in-sideI can't figure out why setcallback simple refuses to work :S
15:47.33epablohi people.. how's it going?
15:47.35PumpkinPiehow do I get login information etc?
15:47.36*** part/#asterisk chris_ast (n=Administ@59.93.56.163)
15:47.45Zeeekby looking at the ATA web interface
15:47.53in-sidePumpkinPie: please go read the documentation man :(
15:48.14PumpkinPiein-side when I clicked on the documentation link it just went to some website asking me to buy stuff
15:48.18*** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net)
15:48.50Zeeek*you have to read the doc for your sipura and find the web interface. Look at that stuff and you'll be ok
15:49.02in-sidePumpkinPie: welcome to internet...
15:49.29*** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154)
15:49.40gbodemantvhi all
15:49.44epabloI need to use the manager to capture SIP peer login info: when a user logs in,logs out or timesout. What event should I suscribe?
15:49.57gbodemantvanyone in here use REALTIME configs
15:50.16gbodemantvtrying to figure out if I can user calling groups, AMP, and FOP with it
15:50.27twisted[asteria]if you mean realtime as in DB driven, yeah.  if you mean realtime as in bill mahr, hell no
15:50.59MikeJ[Laptop]twisted[asteria], liar.. you love the bill mahr... cmon.. admit it
15:51.10twisted[asteria]MikeJ[Laptop], lol
15:51.14MikeJ[Laptop]morning
15:51.20twisted[asteria]or something like it.
15:51.28gbodemantvtwisted: :) yeah the REALTIME DB
15:51.31twisted[asteria]it's the first day of spring, and it's nasty out.
15:52.12Zeeektomorrow is fiorst day, no?
15:52.24in-sideperl agi is so damn well documented :( (NOT!)
15:52.30asterboywe killed a whale and are feasting on whale blubber in our igloo.
15:52.49twisted[asteria]Zeeek, nope.
15:52.51twisted[asteria]Zeeek, http://www.holidaysmart.com/seasons.htm
15:52.52twisted[asteria];)
15:53.05twisted[asteria]today is the spring equinox, and thus, the first day of spring
15:53.13gbodemantvany suggestions on where to start
15:53.23ZeeekAll my life I believed it was the 21st <(
15:53.25MikeJ[Laptop]in-side, use mod_perl
15:53.48in-sideMikeJ[Laptop]: what for?
15:53.56in-sideuse mod_perl with asterisk ?
15:53.57MikeJ[Laptop]the perl agi stuff is...
15:53.59MikeJ[Laptop]yeah
15:54.03in-sidereally?
15:54.09MikeJ[Laptop]one sec
15:54.27in-sideok
15:54.40PumpkinPieall I need is the serial number and the mac address off the sipura so I can emulate the device?
15:54.40in-sidedoes it fast as the other?
15:54.42MikeJ[Laptop]errr
15:54.45MikeJ[Laptop]res_perl
15:54.50MikeJ[Laptop]sorry.. wrong software
15:55.04epabloperl agi has a decent man
15:55.21mishehuhmm...  anybody know of any good way to bill calltime for faxes sent thru hylafax using iaxmodem when using one instance of iaxmodem for multiple entities?
15:55.24in-sideepablo: really?
15:55.28in-siderotfl..
15:55.54MikeJ[Laptop]http://www.pbxfreeware.org/archives/2005/06/res_perl_welcom.html
15:55.56in-sideI just stacked up here with setcallback
15:56.04_Paulo_mishehu, use the hylafax logs instead of *
15:56.11MikeJ[Laptop]I was never a big agi fan..
15:56.17in-sideit is not supposed it to be executed after call ends ?
15:56.32in-sideI need to execute a db query after asterisk set over the call
15:56.36epabloin-side: i've managed to make my stuff work.. ;)
15:56.52in-sideepablo: well I not called it well documentated :)
15:56.54in-sideanyway
15:56.58in-sidequestion is...
15:57.10in-sideI need to execute a query after call ends
15:57.26in-sidebut I have to be shure that asterisk has already handle with all bye stuff
15:57.31in-sidehow can i do it?
15:57.40in-sideI was supposing i had to use setcallback
15:58.23in-sideMikeJ[Laptop]: thanks but I will check it later
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15:58.35epabloi did that with perl agi.
15:58.47in-sideok I'm using perl agi too
15:58.54in-sideproblem is that perl is executing the query
15:58.59mishehu_Paulo_: was hoping to streamline it to one single database if possible.
15:58.59in-sideafter the first bye leg
15:59.02ZeeekI'm sleepy and hungry. Can asterisk help ?
15:59.07in-sidenot after the leg two
15:59.31in-sidehow can I be shure to only execute a command in asterisk after a bye have been sent ?
15:59.54gbodemantvtwisted:
16:00.08_Paulo_mishehu, I have a patch so hylafax autenticates via mysql
16:00.10mishehuespecially since hylafax logs are flat text files
16:00.21gbodemantvany resource you can recommend for REALTIME config
16:00.23asterboyI have a SIP line registered on my Polycom phone and a speed dial using "sip:extension@localhost", but only get the fast beep when trying to dial it.  What is the format for a sip URI call?
16:00.26kmilitzerWhat's the planned release date of asterisk 1.4 ?
16:00.34mishehu_Paulo_: for hylafax 4.2.x ?  got a url for hte patch?
16:00.37epabloin-side:are you trying to implemente a retry or something similar'
16:00.55in-sideepablo: no I doing the handle of accounting to ser
16:01.05in-sideyap but it is similar
16:01.18_Paulo_mishehu, I wrote it. e-mail me and I send it to you.
16:01.26in-sideproblem is I have to be shure to execute it only when the last leg had dealed with bye
16:01.30in-sideat asterisk side
16:01.30asterboythought it was, "sip:exten@server"
16:02.11asterboyAnyone make a SIP to SIP call with Polycom phones?
16:02.13epabloin-side: so you wan't to run it on hangup of both channels?
16:02.23astra^^Mar 20 10:01:39 NOTICE[25958]: chan_sip.c:6278 check_auth: stale nonce received from '1001 <sip:1001@64.246.52.52>'
16:02.28in-sideMikeJ[Laptop]: thanks for the tip bbut res_perl is worse documented stuff I saw in my life
16:02.39asterboyOR how do you setup extension to extension calling?
16:02.39in-sideepablo: exaclty
16:02.50astra^^am gettin an error what does tat mean
16:02.52astra^^Mar 20 10:01:39 NOTICE[25958]: chan_sip.c:6278 check_auth: stale nonce received from '1001 <sip:1001@64.246.52.52>'
16:02.52*** join/#asterisk Jedirl (n=hhgds4@154.Red-217-127-168.staticIP.rima-tde.net)
16:02.53in-sideafter all legs been down on that transaction
16:02.55JedirlHello
16:03.04JedirlI'm having problems with my AGI extensions
16:03.06in-sideif possible keep state of some inside vars of call
16:03.12in-sidelike sip call id
16:03.31epabloin-side: I think you can do it when receving the exit code for the Dial..
16:03.45asterboyAnyone have a Polycom to Polycom setup?
16:03.54in-sidehmm
16:03.57Jedirlasterisk doesn't keep executing all the flow in the extension when the channel is hung-up
16:04.00in-sidecan you dram me it ;)
16:04.07in-sideI think I already had it
16:04.10mishehu_Paulo_: You've Got Mail<tm>
16:04.14in-sideI was trying to use setcallback
16:04.21epabloin-side: setup an AGI.. inside just do the dial.. and catch the return code.. Then do your stufff
16:04.21in-sidewatching callstatus also
16:04.23astra^^what does stale nonce mean?
16:04.55in-sideat this moment all stuff is working except the catch code
16:05.01Jedirlepablo: that's what I've done but my post-dial stuff never gets executed
16:05.14in-sideshould it should be something like...
16:05.24in-sideagi(script1)
16:05.29in-sideagi(scrip2) ?
16:05.35fulgashey in-side :)
16:05.39asterboyCan anyone please verify for me the SIP URI format?
16:05.41in-sidehey fulgas
16:06.23epabloin-side: it could work.. but I was thinking of doing it all inside script1
16:06.43in-sideat this moment I have all inside one script
16:06.54in-sideproblem is the asterisk execute it after the first bye
16:06.56asterboycan I use say, "sip:upstairs@192.168.1.2", (where upstairs is a registered line)?
16:06.56in-side:S
16:07.15in-sideand problem is i need bith :(
16:07.17in-sideboth
16:07.22epabloJedirl: I really haven't tried it.. but it should work
16:07.40epablo<PROTECTED>
16:07.40asterboyholy fuck, must be on ignore.
16:08.15Jedirlepablo: it doesn't
16:08.20asterboynobody here uses sip or polycom?
16:08.26Zeeekyes, both
16:08.31asterboyjbot, do I exist?
16:08.32jbotACTION does I exist.
16:08.39in-sideya maybe it is abetter ideia
16:08.39Zeeekbut not polycom-polycom which you specifically requested
16:08.42in-sideusing two
16:08.47*** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982)
16:08.49in-sideit will force it run
16:08.55in-sidejsut after hangup
16:09.01*** part/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982)
16:09.35asterboyZeek, how do you talk from one phone to the other?
16:09.41*** join/#asterisk fulgas (n=fulgas@209.8.233.252)
16:09.43asterboylike exten to exten
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16:11.58Zeeekasterboy thru asterisk - that's one of the things it does well
16:13.07asterboyYes, I'm trying to do it through *.
16:13.07Zeeekour phones are in different locations behind different NAT so there's no reason to try
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16:13.14RoyKops
16:13.14in-sidecool...
16:13.16ZeeekI just dial say 2002 and asterisk handles it
16:13.16Jedirluhm
16:13.16JedirlI'm lost
16:13.50mishehuthat was a nice little split there.
16:13.50Jedirlwhy my extension flow doesn't keep executing when the phone hangups?
16:13.50epabloin-side: did it work?
16:13.50JedirlI can't catch the ANSWERED and DIALSTATUS and all that
16:13.50asterboySo you register one of the lines as 2002 and dial it from another phone...should be the same as what I'm trying to do.
16:14.04in-sidedidn't test it yet
16:14.04in-sideI will drop a feedback
16:14.04in-sidein a moment
16:14.04mishehujsharp: heees?
16:14.30Zeeekasterboy this stuff is documented in a great artcile here: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
16:14.34asterboyI'll test with numbers instead of Letters.
16:14.47asterboythanks Zeeek, I'll digest that.
16:15.13Jedirlanyone could take a look at my extensions.conf? http://pastebin.com/612559. When Dial ends with a hangup, the rest of the extension doesn't get executed
16:15.35jsharpmishehu:  Laughs.  It just made me snicker.
16:15.37JuggieJedirl, its not supposed to.
16:15.44Jedirluh?
16:15.57Juggiewell its hangup
16:16.00Jedirlso how can I get ANSWEREDTIME, DIALSTATUS and that?
16:16.01Juggiewhat are you expecting
16:16.11Juggiewhen a hangup occurs it goes to the h context.
16:16.14iCEBrkryo yo yo
16:16.30Jedirluhm?
16:16.56mishehujsharp: I didn't realize heee was a way to laugh, but sure!
16:17.00JedirlJuggie: where can I see an example of what you are saying?
16:17.24*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
16:17.30JuggieJedirl, add 'exten=> h,1,Noop(Hello i was hungup)'
16:17.36Juggieto your context
16:17.40Juggieand you'll see what i mean
16:17.43JuggieALSO
16:17.55Juggiethere is another option, where you can have dial continue after a hangup.
16:18.13Juggiebut i would only use that if you intend to do another subsequent dial
16:18.35Jedirluhmmm
16:18.37JedirlOK
16:19.07Juggiedid you see it trap the hangup?
16:19.22asterboyZeeek, not much at that link.  Falling back on the * manual here: http://www.digium.com/handbook-draft.pdf
16:19.26Jedirllet me check
16:19.47Zeeekasterboy well I thought it explained how to dial a number and connect to a phone ;)
16:20.17Jedirlgreat Juggie, that's what I need
16:20.22asterboyZeeek, it just shows a summary for me.
16:20.22Juggieyou can use dial with option 'g' if you want to continue without triggering a hangup
16:20.24Zeeekyou just need extensions that make the connection as in 2000,1,Dial(SIP/FUBad,45,t)
16:20.39Juggiebut jerdirl, i woudnt use it unless absolutely necesairy
16:20.40Zeeekwith the vmail logic if you want it
16:20.42*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
16:20.42JedirlJuggie, the vars i've set are the same in the "h" extension?
16:20.46Juggieeg stacking dials together
16:20.49JuggieJedirl, yes
16:20.54Jedirlthen I'll use it
16:21.06Zeeekyou can also have alpha extensions like Sales,1,Dial(SIP/Sales123)
16:21.08Juggietry Noop(${DIALSTATUS})
16:21.13Jedirl:D
16:21.14JedirlGREAT
16:21.16*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
16:21.19Juggiein exten=>h,1...
16:21.20asterboyZeeek, that example helps...give's me a bone to chew on....giving it a try
16:21.21Jedirlit's easier than I though
16:21.35Juggieyep, you seem to understand programming so youl'l be fine
16:21.39Juggiecheck out www.voip-info.org
16:21.43Juggielots of information there.
16:21.45Zeeekasterboy look up macro-oneline on the net
16:22.02*** join/#asterisk mattodude (n=matt@gateway.digium.com)
16:22.11Zeeekhttp://www.google.com/search?q=asterisk+macro-oneline&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official
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16:24.55asterboyFound an example: exten => 2030,1,Macro(oneline,SIP/2030)
16:26.45asterboyThis exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@sipphone)
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16:27.20asterboyat least tells me my Dial( format is not "sip:${EXTEN..."
16:27.46asterboylooks like it should just be "${EXTEN}@sipphone"
16:28.07ZeeekI think I mùentioned to you about 9 months ago that ou should read the docs on the dialplan
16:28.54Zeeekthe format for Dial is explained in show appication dial, in the asteriskdocs.org site and on the wiki
16:32.11asterboyya the docs do give very good examples...been through the a dozen times...must not be intuative enough for me.
16:32.35Zeeekhow intuitive is this: TECHNOLOGY/channel ?
16:32.48asterboydoesn't tell me much
16:32.58asterboyI need specifics.
16:32.59Zeeekthen you need a dictionary as well :)
16:33.23Zeeekyou need ti deduce the specifics frm the generalities
16:33.38Zeeekso you probably know that SIP is a technology?
16:33.39asterboythat I am not very good at.
16:34.22asterboyyes, however, what is the cormat for channel?  I've seen sip:exten@server.
16:34.22asterboybut that does not work
16:34.23JedirlJuggie
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16:34.27JedirlI need the 'g' option but it doesn't seem to work
16:34.44asterboyI've seen SIP/exten@server, but for me that does not work.
16:34.51asterboythere is something missing, not sure what.
16:34.53Zeeekasterboy you've never seent hat in a dial command have you?
16:35.05Zeeeksip: is a SIP URL
16:35.06Jedirlwell I think I need the 'g' option because I retry with another dialstring if the first's dialstatus was CONGESTION or CHANUNAVAIL
16:35.12asterboyURI
16:35.36asterboybut when dealing with general contexts, I'm left to try it when the other does not wok.
16:35.42Zeeekwhatever, it won't go in the dial
16:35.45asterboys/wok/work/
16:35.57JuggieJedirl, sounds like you probally do, if you dial w the g option it should go back into the second dial?
16:36.31asterboyIt's very frustrating...if it did work I wouldn't be asking so I can understand what replaces general assumptions.
16:36.56Zeeekasterboy I understand that English may not be your mother tongue but you really need to read or re-read the docs mentioned because you're too far away from undertsanding the most basic principles of it all. And type show application dial at the cli and read and study that
16:37.02*** join/#asterisk thieumS (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net)
16:37.23JedirlI dial Zap/g1/962510XYZ|30|g
16:37.31Jedirlbut it still goes to the 'h' extension
16:37.31Jedirl:??
16:37.43asterboyTo my thinking, if I have a registered SIP line, why can't I dial extension@server?
16:37.54asterboyshould be that simple...but obviously I'm missing something.
16:38.42thieumShi, what 's happening if trunk=yes on A and nothing on B on a friendy IAX2  A-B interco ?
16:39.08JuggieJedirl, paste bin the output for me.
16:39.24Octothorpe~pb
16:39.29jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
16:39.57jsharpthieumS:  If you have trunk=yes on one side, but no the other, stuff breaks in ugly ways.
16:40.21thieumSthat's what i thought
16:41.07*** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com)
16:41.55*** join/#asterisk gaspiz (n=gaspiz@86.34.6.164)
16:41.55sevardI was looking at installing ARI.  I'm running * on my wrt, problem is there is no PHP for WhiteRussian.  I'm trying to think of a solution but coming up with nothing.
16:41.56*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
16:42.33gaspizhi does anyone of you know an issue with phpagi and newer php versions?
16:43.00JedirlJuggie I've pasted it to you by /msg because it contains my phone number... :D thanks
16:43.03gaspizI installed a newer version of php and deadagi is now crashing
16:44.10Jedirlgaspiz: php 5?
16:44.47iCEBrkrsevard: fastAGI?
16:45.24iCEBrkrsevard: oh, nevermind me. oops
16:45.28sevard:)
16:45.37sevardI'm attempting to think of another solution
16:45.38gaspizjedirl: yes
16:45.40elielhello oej
16:45.41sevardattempting to think
16:45.43sevardkey words :S
16:45.57oejHello
16:46.39elieloej: what do you think about pach (6735)?
16:46.56Jedirlgaspiz: try enabling Zend 1 compatibility in php.ini; anyway PHP5 is WAY different from PHP4, many scripts aren't compatible
16:46.59oejeliel: Sorry, no time right now
16:47.09elieloej: ok no problem! :-)
16:47.31*** part/#asterisk epablo (n=epablo@200.109.73.215)
16:48.00sevardiCEBrkr: :(
16:52.23*** part/#asterisk thieumS (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net)
16:52.32mutilatorhttp://time3.livejournal.com/62990.html
16:53.46*** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net)
16:56.30gaspizjedril: I tried using a newer php 4.4.x and still having problems
16:56.39*** join/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net)
16:57.02Jedirlgaspiz: check which problems... "having problems" is not a very descriptive error
16:58.48*** part/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net)
16:59.41backbluedo you guys dont use iax2 trunks with domains suport?
17:01.57*** join/#asterisk stack_ (n=stack@63.239.190.202)
17:03.40*** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no)
17:03.47stack_So my company decided to get a PRI for our telephone lines with asterisk as the PBX.  I don't know too much about PRI and the company that is setting it up for us what's us to pick the options (Start Signal, Signal Protocol, Outpulse, etc...) for our line, which I know nothing about.  We are getting a Digium TE110P.  What options should I be looking for?
17:04.17jsharpAre you in the US?
17:04.20stack_yes
17:04.43jsharpTell em you want a PRI that follows NI-2 ISDN.
17:04.44_Paulo_mishehu, I sent a mail to you
17:04.53*** join/#asterisk unmanaged (n=unmanage@64.89.118.139)
17:05.05jsharpIf they stare blankly at you, get a new telco.
17:05.20*** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no)
17:05.30Hmmhesaysni2 5ess, dms, who cares
17:05.47gaspizjedril: running Deadagi... script.php returned 0
17:05.51stack_jsharp, ok that's just one of the options on the page. For example "Line Coding: D4/AMI or B8ZS"
17:06.01jsharpB8ZS
17:06.07Hmmhesaysesf
17:06.07gaspizjedril: actually not calling the php script
17:06.13stack_Framing: SF or ESF
17:06.16jsharpESF
17:06.25stack_Wiring 2 wire or 4 wire?
17:06.29jsharp4
17:06.57stack_Start Signal: Loop, Ground or E&M?
17:07.02Hmmhesaysnone
17:07.11jsharpUh.  That's not a PRI if they're asking for that kind of signalling.
17:07.18Hmmhesaysthats for a cas/robbed bit t1
17:07.24jsharpYeah, what he said.
17:07.27stack_Sorry, that's down in the Trunk Group Information
17:08.06RoyK~seen zoa
17:08.31jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 5d 25m 14s ago, saying: 'it looks kinda suspicious :p'.
17:08.41stack_what's SS7? Sorry, my boss is doing the ordering, he just has me do the setup...
17:08.56jsharp~ss7
17:08.58jbotss7 is probably can be used in conjunction with ss7box.com - see the website.
17:09.34jsharpDon't worry about ss7.  Unless you're going to be a carrier, you'll not need it.
17:09.44stack_k
17:10.14Jedirlss7 with asterisk
17:10.17Jedirlanyone? :D
17:10.21stack_Jack Type: Smartjack or RJ48C
17:10.30jsharpI'd recommend Smartjack.
17:10.43jsharpSo they can do loop testing in the event the circuit goes down.
17:10.50stack_k
17:11.13stack_Under ISDN Details: Primary D channel: NFAS or FAS
17:11.14Jedirlanyone made chan_ss7 work?
17:11.26jsharpAre you just getting 1 PRI?
17:11.29*** join/#asterisk Holos (n=asdf@204.101.26.106)
17:11.34*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
17:11.35a1fahm
17:11.37jsharpOr will you grow to lots more?
17:11.37a1fathis is so fucked up
17:11.40a1faTotal SIP Message Sent:    19
17:11.47a1fai dont see any SIP msgs reach the server
17:11.50a1fawhich makes me believe
17:11.55a1fathe ISP is blocking SIP traffic
17:12.05a1fai have sip debug ip
17:12.08a1faenabled
17:12.12a1faand i dont see any
17:12.13stack_jsharp: we'll be using about half of it now and plan to grow completely into it with a year
17:12.17a1fasip msgs coming in
17:12.21jsharpOkay.  Go with FAS, then.
17:12.31jsharpYou'd go with NFAS if you were bringing in a bunch of PRIs.
17:12.42stack_ok
17:12.45a1faTotal SIP Message Sent:    56
17:12.47a1fawow
17:12.50a1fathis is insane
17:12.51HolosI have asterisk-sounds-1.2.1 and vm-options.gsm says "press 4 to change your password" but 4 is to enter a temporary greeting. I checked sounds-1.2.5 and it seems to be the same sound file. Anyone have a corrected file?
17:13.08stack_So I imagine this is PRI over T1, do I need to worry about these T1 options?
17:13.36jsharpProbably not.
17:14.10a1faanybody ever experienced same problems?
17:14.25stack_jsharp: awesome, thanks... I'll probably have more quetions but this should be all I need for now
17:14.38jsharpK
17:15.02a1fathis is insane
17:15.08a1fagod damn GRANDSTREAM
17:15.13a1faand their nasty ass interface
17:16.40a1faso much for that
17:16.44a1fa<-- idiot
17:16.56a1fai downed a wan interface on the router
17:17.00a1faoh well
17:17.04jsharpWhups.
17:17.08a1faupsy :P
17:17.20a1fafuck
17:17.31a1faits not coming back up
17:17.33a1faoh well
17:18.03austinnichols101a1fa: potty mouth :)
17:18.24a1faah yes
17:18.29a1faaustinnichols101 : i am still having a same issue
17:18.35a1fait works for 5 minutes
17:18.39a1fathen asterisk stops recieving packets
17:18.40jbalcombanyone know why i see (oui Unknown) so much in my tcpdump of phone traffic?
17:19.02*** join/#asterisk chrismog_ (n=chrismog@mog.traxtech.net)
17:19.57HolosHow do I disable my temporary message in Voicemail?
17:20.23*** join/#asterisk r_evolution (i=_evoluti@208.251.203.246)
17:21.06jbalcombHolos: go into the menu to record a temporary greeting and press 2 to erase it
17:21.39*** join/#asterisk nDuff (n=ccd@64.128.31.220)
17:21.55jbalcombHolos: -> VM -> 0 -> 4 -> 2
17:22.31Holosjbalcomb: For some reason the newer sounds didn't get installed and I'm missing all the prompts for that. Thanks,
17:22.52*** join/#asterisk Trazz (n=traderz@65.114.86.29)
17:22.53Trazzi am calling out and my call is cut off before i can get to the voice mail.. what setting do i need to change to allow the call to ring longer?
17:23.18jbalcombHolos thats odd. im running sound 1.2.1
17:23.27Hmmhesaysthe timeout of your dial command
17:23.59nDuffIs there a conventional way to set caller ID strings for folks with non-DID numbers? I'm trying to figure out how to format the caller ID string in sip.conf for users who are accessible only via extension. I'm currently pondering something like ("Person Name" <2345678901+EXTN>)
17:23.59Holosjbalcomb: Me too.. I just did a make install in the sounds-1.2.1 file and it's still missing. No worries, I'm switching to my spare server (1.2.5) this week anyways, I just copied over the missing files.
17:24.12asterboyDial(Technology/resource[&Tech2/resource2...][|timeout][|options])
17:24.55jbalcombHolos right on. i'm working on setting up a 1.2.5 system right now. got my new dell 2850. :)
17:25.49a1fafuck
17:25.51a1fa;(
17:25.53a1fanot to self
17:25.57a1fadont bring down wan interface
17:26.14*** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com)
17:26.24austinnichols101a1fa: want to run through it again?
17:26.33a1fai cant
17:26.36a1fai am cut off
17:26.45a1fai have to wait a few hours
17:26.46Winkiesup gents, i'm having problems tracking a call that goes into a queue, the manager events returned hold no reference to what actual call is ringing the agents
17:26.55Winkieanyone got a clue how to better track queue events?
17:28.23a1faFlash Operator, Winkie
17:29.52Winkiea1fa: i'm sorry?
17:30.08Winkieah right i see
17:30.25Winkiethat's only going to pull information from the manager interface though, and there's a critical piece of information we don't get that i need
17:32.36Trazzwhere can i change the timeout on the dial out command?
17:32.49backblueanyone know anykind of alarms in zaptel? i need notification if one span its down. anyone knows how?
17:32.58backbluelayer1 and layer2 notifications
17:33.00Zeeektrazz in the Dial() parameters
17:33.04*** join/#asterisk mogorman (n=mogorman@gateway.digium.com)
17:33.46backbluehi mogorman
17:34.02mogormanhi blackblue
17:34.40Trazzthanks
17:35.35tuxinator_linuxmog has so many names
17:35.48mog_workjust mog
17:36.55asterboynever forget the value of SIP DEBUG.
17:37.03Winkiebackblue: i think you're just looking for yellow / red alarms?
17:37.43asterboymy sip dial()s are not working cause of a 407 Proxy authentication required.
17:37.51asterboynow that does not show in the manual.
17:38.10*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
17:39.08austinnichols101freePBX is really starting to look nice!
17:40.06RoyK.... ..
17:40.37backblueWinkie: i dont get yellow / red alarms when one span its down.
17:40.58backblueaustinnichols101: why?
17:42.13RoyKwhy?
17:42.36austinnichols101backblue: the modules stuff.  They just added a disa module between 2.0.0 and 2.0.1.  It's very nice to have disa as a destination and it's very cool to see how it can be extended
17:43.28austinnichols101I expect to see a lot of add-in modules
17:43.38*** part/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net)
17:44.40backblueaustinnichols101: pbx without disa, its not pbx! :o
17:44.59austinnichols101I had done my own, but it's nice to be able to just set them up from the GUI
17:45.20austinnichols101one less thing to bother with
17:45.49austinnichols101disa + nextel 'free' incoming = sticking it to the man, bigtime!
17:47.14a1faincoming minutes should always be free
17:47.18a1fayou crazy amerikanz!
17:48.02austinnichols101agreed
17:49.34a1fai love european prepaid gsm cellphones
17:49.41a1fafree incoming calls
17:49.59a1fac0.05/minute
17:50.04a1faawezome!
17:50.09austinnichols101a1fa: at least we can keep our voip phones online for more than 5 minutes at a time :)
17:50.16a1faLOL
17:50.17a1faf0X u!
17:50.30austinnichols101lol
17:50.37Jedirluhm
17:50.42Jedirlin USA you pay for incoming calls?
17:51.11austinnichols101yup
17:51.14austinnichols101most providers
17:51.18Juggieon cell phone yes, depending on the provider
17:51.33austinnichols101you actually have to sign up for a 'free' incoming plan
17:51.37*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
17:51.48austinnichols101instead of 'caller pays', we're on 'everyone pays'
17:52.40Jedirluhmmm
17:52.41Jedirljejeje
17:52.48justinuwelcome to the USA
17:52.54justinuwhere everyone gets fucked
17:53.06austinnichols101justinu: welcome to the USA, please bend over
17:53.29LoRezit's typical of capitalism.  you don't vote with your dollars so they keep bending you over.
17:53.50asterboylike brokeback mountain?
17:53.57asterboyexcept no spit
17:54.13austinnichols101yes, just like brokeback but with cellphones
17:54.27*** join/#asterisk jpm_SD (n=jpm@207-40-115-38.sugardog.com)
17:54.53justinu<PROTECTED>
17:54.55Jedirlhehehehe
17:55.25*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
17:55.33generalhanwhats going on everyone !?
17:56.10iDunnowell, th earth is rotating at approximately a 24 hour period around it's axis...
17:56.18justinuLorez: you're right. what's important here is what ringtone you have, or whether your phone has enough cameras
17:56.21iDunnoand a 1 year period around the sun
17:56.23justinunot price/service
17:56.27generalhanapporx
17:56.35generalhanlol
17:56.55*** join/#asterisk ToTo (n=ToTo@host62-142.pool874.interbusiness.it)
17:57.09austinnichols101'friends and family' and 'IN' plans must look ridiculous from outside the US
17:57.14LoRezjustinu: I'm all too familiar with the situation :)  luckily I haven't had to pay for cell service for the past 11 years.
17:57.27justinulorez: my work has been taking care of my bill for a while also
17:57.33generalhanmy boss is looking for a wireless conference room phone to accomidate 12 users around the table; anyone had any good experiences with any ? or heard anything good about some ?
17:57.46justinuwireless conference room phone? never heard of such a thing
17:57.51generalhanbah
17:57.53justinui can suggest the polycom soundstations tho
17:57.58austinnichols101polycom
17:58.02justinuperhaps w/ an ATA + Wifi Bridge
17:58.06justinu*shrug*
17:58.09generalhanhmm
17:58.51stack_jsharp: you still around?
17:58.58justinulorez: in the US, they call people "consumers"
17:59.02justinunot customers
17:59.05justinuit's pretty damn sad
17:59.11justinuno one cares tho
17:59.25*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
17:59.31generalhanjustinu: http://www.officeworld.com/Worlds-Biggest-Selection/PCY220007880001/05Q4/ something like this ?
17:59.46salviadudjustinu
17:59.48austinnichols101generalhan: polycom makes a voip conference phone
17:59.51salviaduddid you like Fight Club?
17:59.52generalhani just want to be sure that the quality isnt gonna suck on that phone because of the wireless vs wired
17:59.53justinuinteresting, didn't know they made a wireless product
17:59.54*** join/#asterisk Zeeek_ (n=IceChat7@80.125.80.38)
17:59.56austinnichols101just add wireless bridge and stir
17:59.59justinusalviadud: yeah, great movie
18:00.14justinugeneralhan: polycom makes great stuff, you can't go wrong
18:00.19salviadudyou know how i can take out some audio samples from a DVD?
18:00.27salviadudcould mplayer do such a thing?
18:00.35*** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com)
18:00.36*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
18:00.36*** mode/#asterisk [+o anthm] by ChanServ
18:00.38justinui dunno... you need to rip the audio tracks off the disk
18:00.46justinubut i'm not up on the latest techniques
18:01.16stack_We are getting a PRI over T1 setup and they are asking how we want the T1 setup.  The last option is "Dialtone" and it can be either Precise, SCC or None... anyone know the answer :)
18:01.26justinunever heard of that
18:01.36justinuwhy the hell would you need dialtone over a PRI?
18:01.41justinui'd opt for none
18:02.01stack_Oh, sorry, not the T1, this is the trunk group...
18:02.10austinnichols101stack: fyi: my carrier could only deliver callerid NUMBER + NAME over NI2.  NI1 could only give callerID number
18:02.12justinustill
18:02.31justinui don't know why you'd need dialtone... unless you were doing centrex or something
18:03.24stack_the other options are for Signal Protocol, Outpulse and Start Signal, if that helps (you'll have to pardon me, I'm pretty green when it comes to Telco stuff, I'm just the UNIX need here)
18:03.46justinuall that crap isn't applicable to a PRI
18:03.51justinuit's for CAS T1s
18:03.52Luke-JriDunno: the Earth does not rotate...
18:04.21*** part/#asterisk Zeeek_ (n=IceChat7@80.125.80.38)
18:04.24*** join/#asterisk Zeeek_ (n=IceChat7@80.125.80.38)
18:04.59stack_justinu: Im just a little confused... what's the point of the dialtone on the app?  wouldn't we need one?
18:05.48*** join/#asterisk ToTo (n=ToTo@host62-142.pool874.interbusiness.it)
18:05.48justinuno
18:06.03justinuno need for dialtone on a PRI
18:06.03stack_justinu: why is that?
18:06.07stack_ok
18:06.12justinubecause it uses enbloc dialing
18:06.18justinuand inband tones are legacy on ISDN
18:06.31asterboyholy crap, asterisk docs search won't allow "sip proxy" cause its shorter than 5 characters
18:06.41stack_justinu: ok, gotcha... this stuff make my head hurt a little but i'm getting it, thanks
18:07.32*** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net)
18:07.37cripitohi
18:09.03*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
18:09.06Luke-Jrhi
18:12.01justinustack: the telco isn't helping the situation by asking you for settings that don't apply to your circuit type!
18:12.56*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
18:16.15*** join/#asterisk stoffell (n=stoffell@d51A4D1F8.access.telenet.be)
18:18.02jpm_SDI thought that is what the Telco was for - asking irrelevant questions?  at least that is what Sprint does.
18:18.58Luke-Jrthey're always looking for some excuse to drop or charge for support
18:19.39Luke-Jrmy latest one was "oh, we don't support Asterisk" followed soon after by "we don't support a manually configured PAP2 either"
18:20.01Luke-Jrof course, I was never getting a single packet for calls, so my end didn't matter at all
18:20.23Luke-Jrin fact, I'm still not... iConnectHere/deltaThree hasn't managed to fix my account yet
18:20.39Winkiedamnit, no queue name is ever passed back
18:20.42Luke-Jr(needless to say, I'm planning to port my number away from them)
18:20.51Winkiei guess nobody making app_queue ever thought anyone would use > 1 queue >:(
18:21.07Luke-Jrso modify it
18:21.11Luke-Jryou've got source
18:21.16Winkiei am doing, but it's still very annoying
18:21.53Luke-Jrlast time I went to modify something like that, I found I didn't need to: it already had support for what I wanted, just completely undocumented
18:22.00Winkiehaha what was that?
18:22.04*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
18:22.07*** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net)
18:22.28Luke-JrWinkie: 'jump' statement in AEL
18:23.11jpm_SDLuke-Jr -- Making the World a better place.
18:23.17Luke-Jrlol
18:23.46Luke-Jrwell, I could have gone farther and fixed the bugs in AEL instead of just using old extensions.conf for the two macros that the bugs broke... =p
18:24.13Winkiei haven't found a need to use AEL yet, but i'm finding it hard to work with asterisk's call reporting, tracking calls is quite tricky
18:24.23Luke-JrAEL isn't a need; it's a want =p
18:24.23*** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net)
18:24.41SexyKenHey guys -- Is call blocking (IE:  *67) possible with VOIP
18:24.42Winkieat the moment to get the queue name i have to check up on QueueMemberStatus and AgentCalleds, when AgentCalled should say the queue name surely
18:24.54*** part/#asterisk rjt69 (n=rjt@wsip-68-15-224-183.om.om.cox.net)
18:25.33mutilatorwith a name like SexyKen whats it matter?
18:25.58stoffellLuke-Jr, you mean AEL is good/better to create complete dialplans, or only partially ?
18:26.11Luke-Jrstoffell: when it works, it's better
18:26.37jpm_SDThe question would then be -- How often does it "work"
18:26.58stoffellLuke-Jr, okay, looking into AEL then :)
18:26.59Luke-Jrstoffell: nice not to have to repeat the extention every line and keep track of priorities
18:26.59SexyKenMy name shouldn't matter.  But some people are obviously lacking a certain ability to respond, I suppose.
18:27.17Luke-Jrnot to mention switch  statements and the like
18:27.18justinulol
18:27.35stoffelloh, i see Luke-Jr, good tip
18:27.41justinuwith statements like that... those who can respond probably won't
18:27.48Luke-Jrjpm_SD: for-loops are broken in my installed version
18:27.50stoffelljustinu, lol
18:28.40Luke-Jrand functions need to use | instead of commas
18:29.07*** join/#asterisk sssk (n=sssk@s55935276.adsl.wanadoo.nl)
18:29.15Luke-Jrthe only stuff that broke for me really was my str_replace and iterator functions
18:29.22stoffellLuke-Jr, it seems very well documented on voip-info, nice
18:29.49Luke-Jrs/functions/macros
18:30.16SexyKenIs caller ID blocking possible with Asterisk?
18:30.19Winkiei wonder how many people actually use asterisk's CDR for billing
18:30.32Luke-JrWinkie: what else would you use?
18:30.35NuggetJust about anything is possible with Asterisk.
18:30.36Luke-JrSexyKen: yes
18:30.48Luke-JrSexyKen: voice obfuscation is probably possible too if you want that
18:31.51mutilatormake SexyKen's voice sound sexy?
18:31.51SexyKenLuke-Jr, what would the process be?  I know how to set my CID< and I know if I dont set the CID that my provider will use theirs
18:31.51Nuggetrigging up asterisk to punch people in the face if they ask questions that are covered by the faq is even possible.  just a Simple Matter of Programming(tm).
18:31.52Luke-JrSexyKen: don't use such a provider, or set a fake CID
18:31.52WinkieLuke-Jr: well CDR provides virtually no information, so i'm using the manager interface + CDR events, but even then the information provides is woefully inadequate
18:31.53Luke-Jrto actually block it, there's some setting for it
18:31.57Winkiethere's no proper way to track calls and to be honest the syntax is a little iffy
18:31.58justinuor set the presentation indicator to privacy
18:32.08justinuasterisk CDRs suck
18:32.09Winkieplus who knows i could nicely crash asterisk by deadlocking the manager interface D:
18:32.21Luke-JrWinkie: CDR provides account, destination number and seconds connected-- all the info needed
18:32.23stoffellWinkie, we also use manager to track calls in an own cdr
18:32.25Winkiejustinu: totally, we're trying to hack stuff up with a bunch of ResetCDRs and ForkCDRs etc
18:32.31justinuyep, it's a big joke
18:32.37WinkieLuke-Jr: if you think that's all the info needed you've never worked anywhere, sry :(
18:32.42justinuagreed
18:32.50Luke-Jrno telcos anyway
18:32.59Winkienowhere with more than 2 phones
18:33.10Winkiei'm having to hack things in to do basic tracking of calls in queues
18:33.11justinulol
18:33.15Luke-Jrjust because a place has phones doesn't mean people pay for using them =p
18:33.16medusaXXdoes asterisk support video with sip?
18:33.21Winkieyes
18:33.30medusaXXnice
18:33.35medusaXXdo you know some clients for windows?
18:33.38WinkieLuke-Jr: tell me how account, destination number and seconds connected is enough for tracking transfers :)
18:33.44Winkiei don't use windows i'm afraid
18:34.05justinueyebeam is a SIP client w/ video
18:34.06Luke-JrWinkie: ah, didn't consider transfers; maybe call #?
18:34.19WinkieLuke-Jr: queues
18:34.23Winkieagents
18:34.24Winkieetc
18:34.26WinkieI can go on and on
18:34.39justinufreeswitch will have much better CDRs
18:34.42Winkiethere's no guarenteed way to check on how Agent/1001 was transformed into Local/whatever@wherever etc
18:34.44justinusince it tracks the entire history of a call
18:34.56Winkiemeh
18:35.02Winkiei'm more interested in asterisk fixing theirs
18:35.09justinumeh - don't hold your breath
18:35.24*** join/#asterisk noky (n=Noky@200.69.211.18)
18:35.26Luke-JrWinkie: sounds like a hack.. Agent/1001 should just be connected to the call, not transformed
18:35.31nokyhi
18:35.31*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
18:35.35Winkiethere's a fair amount more coders now though
18:35.45WinkieLuke-Jr: welcome to chan_agent and AgentCallbackLogin
18:35.58justinuthere's too many political problems, imo
18:36.19Winkiepossibly
18:36.29WinkieI need to submit a couple of patches but i'm too lazy to figure out how to do it yet
18:36.32justinui'm not sure they'd even admit that the CDR structure is lacking
18:36.46*** join/#asterisk MGSsancho (n=user@adsl-67-127-164-145.dsl.irvnca.pacbell.net)
18:36.51*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
18:37.02Luke-JrWinkie: IIRC, first you need to sign a copyright release form or such
18:37.17Luke-Jrhence why I avoid edting * code =p
18:37.25justinuluke-jr: agreed
18:37.30WinkieLuke-Jr: any idea why? digium retains copyright on your code or something?
18:37.37Luke-JrWinkie: yeah, I think so
18:37.40justinusign away your first born
18:37.42Winkiewell isn't that infuriating
18:37.43mog_workdigium does not
18:37.49Nuggetdigium needs the ability to release your code under their commercial license, not the gpl.
18:37.53Winkieah i see
18:37.58mog_workyou grant digium a license
18:38.01mog_workto use your code
18:38.01Nuggetthey don't require you to hand over copyright, just licensing flexibility.
18:38.02*** join/#asterisk joe4319 (n=yjoe@65.222.176.9)
18:38.04Luke-JrNugget: maybe I don't approve of non-GPL'd usage?
18:38.05mog_workyou retain copyright
18:38.06Winkiethat's fine then
18:38.18WinkieLuke-Jr: then that's your problem
18:38.18NuggetLuke-Jr: then don't contribute to asterisk.
18:38.18Winkie(and you're an idiot ;) )
18:38.24Luke-JrWinkie: not at all, it's *'s problem
18:38.29justinuheh
18:38.31WinkieLuke-Jr: no it's not
18:38.35Winkiethey have picked their license
18:38.36stoffellLuke-Jr, if everyone would think that way.. :(
18:38.42jpm_SDThis could go on all day...
18:38.50Luke-Jrstoffell: now I know why FreePBX was forked, I think
18:38.50mog_workyeah
18:38.51Winkieit could but it won't because i'll just start insulting everyone 8)
18:38.54mog_workit tends to as well
18:38.57mog_workits openpbx
18:38.58stoffellindeed, on to some practical stuff? dundi experts here? ;)
18:39.00stoffellLuke-Jr, indeed..
18:39.01mog_workfreepbx is amp
18:39.02justinuopenpbx != freepbx
18:39.04Luke-Jrah
18:39.14mog_workand its not even openpbx
18:39.18mog_workas that is a perl pbx
18:39.21mog_workits openpbx.org
18:39.25tsumeflash is dumb, heh
18:39.31mog_workbut i digress
18:39.34Luke-Jrso does openpbx actually have development?
18:39.40mog_workwhat do you need to know about dundi stoffell
18:39.41justinua bit
18:39.47mog_workthey have mostly kept up with bugs
18:39.47*** join/#asterisk MGSsancho (n=user@adsl-67-127-164-145.dsl.irvnca.pacbell.net)
18:39.55mog_workbut there are a few things there that are not in asterisk
18:39.57mog_workand vice versa
18:40.07Luke-Jrcan't Asterisk changes be merged directly?
18:40.13stoffellmog_work, i'm just reading up on it, want to get a test-setup working this week
18:40.14mog_workno
18:40.19mog_workas they changed api names
18:40.20mog_worketc
18:40.23Luke-Jroh
18:40.30Luke-Jrthat was a bit dumb :\
18:40.31stoffellmog_work, any good references besides TFOT and dundi.com ?
18:40.31brodiemis there ANY way I can get firmware updates from Polycom (IP301)? It seems crazy that I have to become a certified reseller to get them
18:40.36justinuthat was a big mistake
18:40.37mog_worknot that i know of
18:40.43mog_worki have learned from doing
18:40.47stoffellbrodiem, through the guys that selled them to you..
18:40.50mog_workvoip-info has an example as well
18:40.50justinubrodiem: talk to whomever sold you the phone
18:40.56*** join/#asterisk bsdfreak (n=alex@breakbeats.okkernoot.net)
18:40.56joe4319Can anyone help with a problem dialing out on a TDM400P?  I get out 50% of the time.
18:41.06mog_workadd some waits jo
18:41.11mog_workjoe4319,
18:41.29mog_workinstead of dial(zap/g1/${EXTEN})
18:41.31jpm_SDjoe4319, what happens the other 50%?
18:41.40mog_workdo dial(zap/g1/www${EXTEN})
18:41.45joe4319I get a Verizon message that says enter a calling card
18:41.47*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
18:41.48stoffellmog_work, i can understand, but do the phones still 'need' their own fixed server then?
18:41.49mog_workthat will allow for 300 ms wait before it dials
18:41.52ian_kDo we sell the octasic cards yet?
18:42.00mog_workno there are ways around
18:42.15brodiemthe guys I bought it from (antonline.com) just referred me to Polycom's resource center page.. they said they are just "online" resellers
18:42.26justinulol
18:42.26*** part/#asterisk ian_k (n=ian@gateway.digium.com)
18:42.39justinuyou could complain to polycom
18:42.43justinuabout them not being authorized reseller
18:42.49brodiemtalking to polycom is like talking to a wall
18:43.03jpm_SDbrodiem, but less responsive.
18:43.04brodiembut anyway, they told me that their resllers weren't allowed to give our the firmware either
18:43.10justinuthat's not true
18:43.14Luke-Jrjustinu: um... you don't need to be authorized to resell stuff =p
18:44.01asterboyjust look at ebay
18:44.17brodiemanyone know who sells them at a nice price and will offer the firmware?
18:44.24brodiemI need about 20 of them
18:44.34Luke-JrI got my PAP2-NA fourth-hand
18:44.35brodiemantonline sells them w/ POE cable for $117
18:44.36stoffellmog_work, you have any tips on recources on the complete "cloud-alike" way to setup *, dundi and sip phones?
18:44.53brodiemthat's the IP301 btw
18:45.04mog_workstoffell, dundi + iax +regexten + roundrobin dns = HAPPY
18:45.09joe4319Thx mog_work.  I think that did the trick.
18:45.13mog_workbut aside from that i dont have any real tips
18:45.15*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
18:45.30stoffellmog_work, okay, it's start! thanks!
18:45.30Luke-Jrmog_work: does anyone actually use dundi worth setting it up? =p
18:45.33mog_workyup joe4319 there is no way for it to know if it has gotten dial tone from telco yet so it just starts dialing
18:45.49mog_workumm yes Luke-Jr like you wouldnt even know
18:46.01Luke-Jrmog_work: I wouldn't. I'm not part of the dundi network
18:46.08joe4319Why can't it detect a dial tone?
18:46.15mog_workthere is much more to dundi than e164 network
18:46.15stoffelli also believe 'many many' people are using dundi.. (and not only public.....)
18:46.23Luke-JrI just know that eg, enum had virtually no benefit
18:46.23mog_workexactly stoffell
18:46.33mog_workthats where it is huge
18:46.44*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
18:46.44hardwiremogmogmog
18:46.46stoffellcorrect me if I'm wrong, but the possibilities this has in an enterprise ... oh boy..
18:46.51*** join/#asterisk epablo (n=epablo@200.109.73.215)
18:46.53mog_workjoe4319, it just doesnt attempt to , you could probably write a soft dsp to do it
18:46.58mog_workyeah stoffell has the idea
18:47.06mog_workhardwirehardwirehardwire
18:47.19stoffelllol
18:47.37epabloHow to i get the asterisk manager to showme all events?  or to sendme all events'
18:47.42a1fa;(
18:47.45stoffellavaya can only "drea" of 1 virtual pbx :D
18:47.49stoffell"dream"
18:48.06mog_workepablo, you can set the events manager sees in manager.conf
18:48.10brodiemok how about this, can someone give me the sip firmware for polycom phones? :)
18:48.18nokyhi, i'm trying to set asterisk's cdr with MySQL database, i installed the asterisk-addon and I configurate cdr_mysql.conf. I create a mysql database with the table 'cdr'... But it doesn't work...
18:48.23noky2006-03-20 15:14:28 ERROR[4995]: cdr_addon_mysql.c:438 my_load_module: Failed to connect to mysql database asterisk on 127.0.0.1.
18:48.26nokyany ideA?
18:48.33stoffellbrodiem, buy the 501, i've got a release of that one :)
18:48.34jsharpIs mysql started?
18:48.39stoffellbrodiem, doesn't the phone has sip installed already?
18:48.48epablomog_work: I've been looking at it.. but I cant find how to make the call
18:48.48jsharpCan you connect to it with the mysql client?
18:48.52mog_workwell it obviously isnt connecting noky you proably have it misconfigured
18:48.55nokyyes
18:49.03nokyi have the table sip_buddies working OK
18:49.04mog_workepablo, ?
18:49.07brodiemstoffell I have 1.6.2 now, but I wanted to update it and have the ability to keep it up to date
18:49.09mog_workoriginate?
18:49.16nokymysql is started
18:49.20*** join/#asterisk livesNbox (n=livesNbo@68-76-129-3.ded.ameritech.net)
18:49.31mog_workyou can turn on debug noky to see connect attempt
18:49.37mog_workbut if i had to guess its auth error
18:49.48epablomog_work:  I set up a user.. and can connect with telnet, but I need to se events related to SIP register
18:50.15livesNboxHey guys -- I'm trying to figure out how to let agents log into the queues -- I am using the AgentCallbackLogin command and it asks me for my agent number followed by # sign -- so I put it in.. and then nothing happens.. about 10 seconds later, the line hangs up.
18:50.37mog_workits there epablo you just need to parse
18:50.45noky2006-03-20 15:18:19 WARNING[4995]: res_config_mysql.c:553 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default.
18:51.00nokymy mysql.sock is in /tmp/mysql.sock :(
18:51.07epablomog_work:  but it doesn't come out on the screen there is nothing to parse :S
18:51.48mog_workwell sip registration might not have a manager event associated with it
18:51.53mog_workto which you will need to make on
18:51.54mog_worke
18:51.57mog_workbut i thought it did
18:51.57*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
18:53.05*** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu)
18:54.00epablook.. let me test other events
18:54.22epablomog_work: thanks
18:55.23mog_workno prob
18:55.30*** part/#asterisk epablo (n=epablo@200.109.73.215)
18:59.32*** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com)
18:59.54SexyKenHey guys -- how would I change this:  http://pastebin.ca/46357  to include a *67 ability to block caller id on the call?
18:59.58a1fakeep-allive setting to :5s
19:01.41tzangerSexyKen: you'd have
19:02.05tzangerexten => *67,1,SetDB(BLOCKCID=true) and *67,2,Goto(s,1)
19:02.31tzangerthen change s,2, to gotoif(GetDB(BLOCKCID)=true?s,4)
19:02.35tzangerand make the current 2 a 3
19:02.48tzangeryou'd have to put a little magic in there though to erase the db key afterward
19:02.51SexyKenWell -- I dont mean to sound...stupid...but I have no idea what you just said?
19:03.25tzangerSexyKen: or have something like _*67.,1,SetDb(BLOCKCID=true) and continue on, erasing the key afterward
19:03.50*** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de)
19:03.54SexyKenAny chance you could through that into a pastebin the way it should look?  I'm sorry if I'm difficult, I am speaking a foreign language with asterisk tho :-(
19:04.24saftsackhi, im searching for moh music. does anyone of you know a good url?
19:06.06*** join/#asterisk pb_ (n=pb@cpc1-cmbg6-0-0-cust434.cmbg.cable.ntl.com)
19:06.32Kattyis there a way to make VI spit out a certain line of a document?
19:06.39Nuggetgoogle for "royalty free music"
19:06.49Kattyor cat.
19:07.08*** join/#asterisk Broom (n=none@12.174.235.14)
19:07.28Broomhello all, i was wondering if anyone could guide me to information on how to turn up the volume of tue MOH
19:07.29Broom?
19:07.29*** join/#asterisk epablo (n=epablo@200.109.73.215)
19:07.31I-MODKatty: you could use a combination of tail an dhead
19:07.37*** part/#asterisk epablo (n=epablo@200.109.73.215)
19:07.41I-MODs/an dhead/and head
19:07.46I-MODs/an dhead/and head/
19:07.50I-MOD....
19:08.06Kattynevermind
19:08.12Kattyit's :n
19:08.14*** part/#asterisk Peaceful (n=Peaceful@70.98.162.62)
19:08.44KattyI-MOD: and what on earth are you talking about?
19:09.34Kattyjustinu: hiya.
19:10.39justinucapitalizing vi is blashphemy :P
19:10.43I-MODnvm, i'm just 10 different kinds of retarded all the time
19:10.49justinublasphemy
19:10.50saftsackNugget, didnt find any good sources :(
19:11.57*** join/#asterisk twisla (i=twisla@lutin.jard.in)
19:13.40*** join/#asterisk epablo (n=epablo@200.109.73.215)
19:14.11epablois there a way of getting asterisk to save the local tree on a DB?
19:14.23justinukatty: try this command: sed -n {45p} file.txt
19:14.27justinuwould print line 45 only
19:15.21epabloHas anyone done a asterisk - ser, setup?
19:16.13epabloi want to transperantlly setup SER to loadbalance my asterisk servers.  Is this posible?
19:16.15justinu~seen r_evolution
19:16.21jbotr_evolution is currently on #asterisk (1h 55m 58s), last said: 'but soon!'.
19:18.03*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:18.30*** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net)
19:19.41*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)
19:20.18sundancerNoobish question here.. how to configure dialplan if to give me external line via Zap interface when i dial 9 and then external number
19:20.45iCEBrkrsundancer: Dial(Zap...)
19:20.58sundancerWhere do i specify external number?
19:21.02*** join/#asterisk MattH (n=MattH@63.174.244.195)
19:21.15iCEBrkrsundancer: Use ${EXTEN}
19:21.19MattHHi... can anyone explain why _X.,1,Zapateller is not catching inbound calls that don't have a route?
19:21.25Netgeeksexten => _9.,1,Dial(Zap/G1/${EXTEN:1})
19:21.32SexyKentzanger any luck?
19:21.32iCEBrkrsundancer: It's like dialing another SIP phone, but you're using the zap device instead
19:21.38Netgeeksassuming you have your zap channels set up as group=1
19:21.56iCEBrkrMattH: ZapTeller has nothing to do with 'route'  or whatever you're talking about :P
19:22.05BroomNetgeeks: doesn't it needs to be _9.| so it strips off the 9?
19:22.18sundancerYup, thanx.. but i dont understand what is ${EXTEN} variable here
19:22.32Netgeeks${EXTEN:1} stips off the 9
19:22.36Broomperfect
19:22.39sundancerAhh i get it
19:22.56sundancer${EXTEN:10} would cut first 10 digits ?
19:23.03MattHiCEBrkr, right understood :)   is there a reason you (or anyoen else) can think of that _X. wouldn't match a call coming in?
19:23.16epablo<PROTECTED>
19:23.18Netgeeks${EXTEN} contains the full number that was matched in that extension line....  Yes, ${EXTEN:10} would strip the left 10 digits
19:23.20sundancerThanx!
19:23.26Broomcool
19:23.36NetgeeksMattH: yes
19:23.40iCEBrkrMattH: Yea, cuz nothing matches it.
19:23.42BroomNet: you happen to know how to spike up the volume of the MOH?
19:23.52Netgeeksno number comes in, therefore asterisk would look for the s extenstion
19:24.03MattHyeah I guess so...  bah not getting DNIS
19:24.06iCEBrkrNetgeeks: Not always true :P
19:24.23Netgeeksone number comes in... for example 1 wouldn't match _X. because _X. says there is at least two numbers in the string..
19:24.53iCEBrkrIt depends on your provider.   For instance, VoicePulse delivers calls with my full number, while my friend delivers only the last 4 digits if my phone number...
19:25.54NetgeeksI run a provider that delivers a random length of your number sometimes from the left side, sometimes from the right, just to keep you on your toes  ;)
19:26.00iCEBrkrLOL
19:26.04Qwell[]Netgeeks: See msg. :)
19:26.05NetgeeksWe don't have any customers.. they leave right away
19:26.11MattHI'm using asterlink
19:26.16jsharpMakes customer support easy.
19:26.18MattHthey claim to deliver the number, but I'm not seeing it
19:26.33*** part/#asterisk epablo (n=epablo@200.109.73.215)
19:26.35iCEBrkrMattH: You should see what's going on with your inbound calls via the CLI
19:26.39iCEBrkrMattH: 'set verbose 9'
19:27.17MattHyeah I do see. .and I see no inbound DNIS
19:27.45iCEBrkrAsterisk doesn't complain about not knowing what to do with XXX-XXX-XXXX?
19:27.50iCEBrkror whatever the error message is?
19:27.51MattHyes it does :)
19:27.56MattHand it doesn't match on _X.
19:28.09iCEBrkrYou're still not understanding
19:29.14*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
19:29.29MattHI SEE s@from-trunk doesn't exist
19:30.01iCEBrkryea, ok.. So it's looking for extension 's' in context 'from-trunk'
19:30.23iCEBrkrSo it'll never 'match' _X.
19:30.44MattHright... so that's on the provider's end? why would it try to send it to s@from-trunk?!  everyone else sends it to just from-trunk
19:30.52*** join/#asterisk kend (n=chatzill@host-64-65-199-187.man.choiceone.net)
19:30.57*** join/#asterisk nshm (n=shmyrev@b.gz.ru)
19:31.01MattHcan I do s,1,include=>from-pstn ?
19:32.06kendReading the astGUI install notes, and they mention using the "o" flag in extensions.conf, and then show a line line this: exten => _901144XXXXXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},55,tTo) -- where do I even look up what the "tTo" on the end means?
19:32.23*** join/#asterisk fifer (n=20f04395@c-24-20-155-56.hsd1.wa.comcast.net)
19:32.37iCEBrkrkend: On the Wiki
19:32.40iCEBrkr~wiki
19:32.41iCEBrkr~docs
19:32.43jbotit has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
19:32.50jsharptT = transfer allows, o says "set outbound callerid to the same as inbound callerid"
19:32.53kendThanks -- but what do I look for?  I mean, I have no idea what it *is*.
19:32.54iCEBrkrkend: Look up the Dial() application
19:33.01kendAh!  Thanks.
19:33.17iCEBrkrkend: Each flag is documented in the on the Dial() page
19:33.27fiferin * 1.2.x how are lines in a single context in the dial plan handeled when they are the same exten and priority, is ony the first one taken or the last?
19:33.37kendHadn't realized it was a Dial() thang. Makes sense, now.  *feels dumb*
19:34.01iCEBrkr,1,Dial() should have tipped ya off :D
19:34.19fiferI know this question makes no sense with hand written dialplan, but I'm actualy trying to get AMP to do something and want to override something it controles in a context
19:34.25iCEBrkrfifer: Why would you have the same priority?
19:34.36kendYeah, yeah... I'm tired.  It's Monday.  for i in `cat /usr/local/excuses`; do echo $i; done
19:34.45iCEBrkrfifer: Waste of time man.. AMP will overwrite it later.
19:34.52iCEBrkrkend: :)
19:35.00*** part/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br)
19:35.11fiferGood question, because I'm including lines in an included custom file
19:35.21iCEBrkr~amp
19:35.26jbotwell, amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
19:35.26*** join/#asterisk sfChrisJacob_ (n=sfChrisJ@sf-nat.sourcefire.com)
19:35.26iCEBrkrerr hrrm.
19:35.27fiferthey are not overwritten, just dont know how they are handled.
19:35.37MattHblah there we go
19:35.39iCEBrkrfifer: But the main file will be over-written
19:36.04iCEBrkrif Amp over looks includes (which I doubt) you'll be fine.
19:36.15fiferI know, I know, the question is about * not amp
19:36.19sfChrisJacob_Hey all, anyone know of a wireless headset (with remote answer) that will work with the Polycom IP 500?
19:36.22*** join/#asterisk Lino` (n=Lino@i577BF45F.versanet.de)
19:36.23fiferIt is about how the extensions.conf is parsed
19:36.52fiferThese are includes AMP has in there to allow you to customize things.
19:37.11nokyhi
19:37.20fiferBotom line, I was just wondering how similure lines were dealt with.
19:37.20sfChrisJacob_Polycom had a pdf fow awhile with compatible units, but I cant find it anymore... and google is not proving to be as helpful as it could be...
19:37.21*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
19:39.56a1faMar 20 19:26:20 NOTICE[10294]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE!  Last qualify: 168
19:40.00a1fagod fucking damn it
19:40.04a1fait was reachable for an hour
19:40.10*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
19:40.30a1faTotal RTP Packet Loss:    127
19:40.35a1faRegistered:    Yes
19:40.41a1fathis stupid shit still think its registred
19:41.10*** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)
19:41.14Kattywhat's a /nice/ song with a good solo?
19:41.18Kattya good geetar solo
19:41.27NetgeeksHiya Katty!
19:41.27austinnichols101stairway to heaven
19:41.29a1fabasement jaxxx
19:41.34a1fafuck bitches
19:41.42Kattyhiya Netgeeks!
19:41.47Kattyaustinnichols101: i've already got that one
19:42.09austinnichols101peter frampton - do you feel like I do
19:42.09NetgeeksAre you looking for acoustical guitar or electric?  what flavor of music?
19:42.18xbmodder_lappy?
19:42.44xbmodder_lappyAnglea?
19:42.52a1faKatty : got a license :L
19:42.53vuudKatty: anything from G3
19:43.17xbmodder_lappymissouri, that sucks
19:43.20xbmodder_lappy_sucks_
19:43.36_Paulo_Katty, Bad to The Bone, ZZ Top
19:43.41a1falol
19:43.50*** join/#asterisk pb_ (n=pb@cpc1-cmbg6-0-0-cust434.cmbg.cable.ntl.com)
19:43.57*** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu)
19:43.59a1fa_Paulo_ she is not running a strip joint
19:44.11a1fabad to the bone.. lol
19:44.16[TK]D-FenderKatty : Final Countdown (Europe), Here I Go Again ; Still oF the Night (Whitesnake)
19:44.19*** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net)
19:44.28austinnichols101D-Fender: the 80s are over
19:44.32[TK]D-FenderLIES!
19:44.46*** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de)
19:44.50austinnichols101I've got some 'Winger' here for you
19:44.56saftsackhi i couldnt find any moh files in google :(
19:45.03sfChrisJacob_no love on the Polycom headsets huh?
19:45.07[TK]D-Fenderaustinnichols101 : I already play "Easy Come Easy Go", and "Seventeen" :)
19:45.19austinnichols101nice
19:45.52stoffellsfChrisJacob_, plantronics works nice with polycom
19:46.09saftsack[TK]D-Fender, do you have moh files?
19:46.23[TK]D-Fendersaftsack : Like most everyone else, yes.
19:46.23sfChrisJacob_do you know if polycom/plantronics allow for remote answer?
19:46.46saftsackdo you know a good source for them?
19:46.55stoffellsfChrisJacob_, you mean hitting 'answer' on the headset or wire ?
19:46.56sfChrisJacob_I am looking for wireless, and I seem to think this is possible without the lifting contraption...
19:46.57[TK]D-FendersfChrisJacob_ : Yes & no, thats in SIP-B to be implemented in * for this summer (expected)
19:47.10sfChrisJacob_hitting answer on the headset
19:47.19[TK]D-FendersfChrisJacob_ : Oh that... No....
19:47.33[TK]D-FenderI have the plantronics lifters for my CSR's.. PITA...
19:47.40astra^^any open source billing software for*?
19:48.16sfChrisJacob_[TK] so the hand set needs to be physically lifted in order to answer?
19:48.34stoffellsfChrisJacob_, i only use the "wired ones" but they don't have remote answer
19:48.56sfChrisJacob_stoffell, I see... thanks for the info
19:49.03[TK]D-FendersfChrisJacob_ : for a headset no, but I extended your meaning a little.  You just need to press the headst button.  But that means you have to be in range of the phone
19:49.05saftsack[TK]D-Fender, can you give me a file? :)
19:49.24[TK]D-Fendersaftsack : * comes with 3-4 samples already....
19:49.35saftsackwhere to find these samples?
19:50.08[TK]D-Fendersaftsack " its in the "add-ons" file I believe
19:50.23saftsackin the source directory?
19:50.36[TK]D-FenderJust go look....
19:50.54sfChrisJacob_[TK]d-fender, Yeah... so say the receptionist is in the next room making coffee and she hears the phone ring... she can press a button on the wireless headset and get the call....
19:51.39a1faanybody else having broblems with CrapPhone100
19:51.51a1fathis nat is bothering me soo much
19:52.01[TK]D-FendersfChrisJacob_ : nope....
19:52.34sfChrisJacob_damn.... ok... got it... need to press the answer button on the phone...
19:52.42a1fait seems, that everytime it unregisters
19:52.47a1fait cant re-register
19:52.49[TK]D-FendersfChrisJacob_ : you need either a lifter or to be next to the phone.  I would suggest you get an ATA and a plantronics headset phone and ring them in parallel
19:52.51a1fastupid netgear
19:53.11a1faUNREACHABLE
19:53.15a1fai hate this word
19:54.30*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
19:57.44*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
19:59.07r_evolutionman... everyone went strangely quiet
19:59.34xbmodder_lappyno
19:59.37xbmodder_lappynot meh
20:01.44jsharpThen I realized I'd make more money just broadcasting the revolution.
20:01.51r_evolution...
20:01.53r_evolutionhar har
20:02.25*** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net)
20:04.30NetgeeksDarn, Katty is MIA and i found her the best guitar solo
20:04.58*** join/#asterisk Inkubot (n=inkubot@200.74.170.218)
20:05.03Inkubothi
20:05.06r_evolutionyou guys are just mean...
20:05.10xbmodder_lappyi'd like to die...
20:05.22xbmodder_lappyInkubot, it ryhmes!
20:05.23xbmodder_lappy:-D
20:05.32Inkuboti've got some SpeedTouch devices (thomson).
20:05.35*** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com)
20:05.44InkubotAsterisk doesn't recognize the FLASH key
20:05.55Inkubotit thinks that is a Hung UP
20:05.57Inkubotor something
20:06.23xbmodder_lappyInkubot, what ATA?
20:06.23a1fayup yuypyppy
20:06.24a1fay
20:06.33a1fano body buy BUDGETPHONE
20:06.36a1fafuck grandstream]
20:06.39a1faomfg
20:06.44Inkubotxbmodder_lappy ST-190 and a ST-780(wl)
20:06.48jbalcombi thought it was budge not budget?
20:07.10xbmodder_lappyInkubot, what brand?
20:07.31Inkubotdamn.. what is brand!
20:08.00Inkubotsorry, English it is not my language
20:08.02xbmodder_lappyoh
20:08.03xbmodder_lappylol
20:08.06Inkubot:P
20:08.06xbmodder_lappySpeedTouch
20:08.09xbmodder_lappyis the brand
20:08.12Inkubotyeps
20:08.13xbmodder_lappyI don't know about them.
20:08.32xbmodder_lappyI know that grandstream doesn't support FLASH
20:08.34a1fajbalcomb : assbite, you got the idea
20:08.44xbmodder_lappyand that sipura does...
20:08.57xbmodder_lappythats my limited knowledge of these evil things called ATAs
20:09.00Inkubotemmm.. i also have grandstream devices.. it's works fine
20:10.23Abydos313man i ordered my spa3k from digitnetworks.com and they charged my card on the 15th and just sent device now. what a buch of fuktards. i'll never use them again
20:10.30*** join/#asterisk starlein (i=star@fo0bar.de)
20:11.01tzangeryeah a whole 3 business days, gosh you must really be in a bind
20:11.03Abydos313anyoe else ever use them? is their regular business practice for them?
20:11.35*** part/#asterisk sfChrisJacob_ (n=sfChrisJ@sf-nat.sourcefire.com)
20:11.39Abydos313no bind, it's just bullshit to charge a card and not send product out to customer for 5 days
20:11.40starleindoes anyone know why i get with enabled qualify in iax2 show peer ... Status: UNKNOWN
20:11.49tzangerI could see you upset if it happened habitually but it's a first time you ordered from them and there was a discontinuity between when they charged and shippped... jeez
20:12.03justinuIBM put an auth on my credit card for a laptop they won't ship for another two weeks
20:12.16Abydos313my customers would never put up with that in my business
20:12.19r_evolutionkill them justin.
20:12.22Abydos313haha
20:12.26justinui just want the fucking laptop! :)
20:12.32tzangerAbydos313: sometimes that stuff happens.  As I said if it is a habitual practise of a company I would avoid them, but sometimes shit just happens.  No need to be a prick about it
20:12.32r_evolutionok well
20:12.35r_evolutionget the fucking laptop
20:12.36r_evolutionTHEN
20:12.38r_evolutionkill them justin.
20:12.47justinuheh
20:12.55Abydos313how is asking if someone else has issues with them being a prick?
20:12.55r_evolution;)
20:13.46r_evolutionyou bastards... distracted me...
20:14.02r_evolutionput the wrong IP in for connection ;x
20:14.20tzangerAbydos313: are you obtuse?
20:14.22tzanger"what a buch of fuktards. i'll never use them again
20:14.23tzanger"
20:14.26tzangeris not being a prick?
20:14.37Abydos313ok that was alittle over the top..heh
20:15.06r_evolutionyou know... any time anyone asks about someone else being obtuse
20:15.10r_evolutioni cant help but think of triangles
20:15.21generalhanlol
20:15.54r_evolutionwonder why that is... :-\
20:16.23justinumy triangle is obscene
20:16.23tzangerr_evolution: :-)
20:16.25justinunot obtuse
20:17.04r_evolutionoh my godness gracious!
20:17.05r_evolutionhah
20:17.10r_evolutionthrowback expressions
20:17.21*** join/#asterisk razu_ (n=razu@80-235-91-173-dsl.prn.estpak.ee)
20:17.32justinu.ee?
20:17.50r_evolutionO_o
20:17.51r_evolutionare you making noises at me justin?
20:17.54a1faaiiight
20:17.57a1faeverybody stop for a second
20:18.02justinu(12:17:26) razu_ [n=razu@80-235-91-173-dsl.prn.estpak.ee] entered the room.
20:18.04justinuwhat's .ee?
20:18.05a1fanat=yes;qualify=2000
20:18.08r_evolutionoh
20:18.14a1fakeep-alive=10 (on the phone)
20:18.16justinuestonia?
20:18.20a1faregister-every=3600s
20:18.21r_evolutionmaybe?
20:18.30a1faand it goes unavailable
20:18.35a1faUNREACHABLE
20:18.57justinuconnectivity problem then
20:19.02starleindoes anyone know why i get with enabled qualify settings in "iax2 show peer ..."  Status: UNKNOWN ???
20:19.12tzangeralephcom: 3600s is 1h is it not?
20:19.12a1fajustinu : i can get to it via http://
20:19.15a1fai can get to the phone
20:19.19a1fai was thinking dns issues
20:19.22tzangeryour NAT holes would close up far sooner than that
20:19.25a1faso i used static IPs
20:19.26tzangertry 60s not 3600ds
20:19.29tzangerer 3600s
20:19.30juanjocHas anybody been experiencing memory corruption problems when using the latest app_txfax with spandsp?
20:19.45a1fatzanger : i tried that too.. didnt help.. what happens, it goes unreachable, and never re-registers
20:20.02tzangerhmm it's like the phone is convinced it doesn't have to re-register
20:20.17a1fai have the newest firmware
20:20.20justinui suppose it could be DNS
20:20.23a1fai guess the most stable
20:20.30a1fadoes anybody know a public dns?
20:20.32justinurule it out by switching to numeric IP address
20:20.32a1fathat i can use?
20:20.34*** join/#asterisk dja (n=alden@zaphod.math.ohio-state.edu)
20:20.35justinu4.2.2.2
20:20.36r_evolutioni know in the case of some of our customers... they STAY unknown...
20:20.40a1fajustinu : i did it.. it didnt work
20:20.46justinunot DNS then
20:20.48a1fajustinu : i will do it again
20:20.52jbalcomba1fa haha.. assbite, that's a good one.
20:20.55justinuyou can use 4.2.2.2
20:21.01jbalcombi always use 4.2.2.2
20:21.06r_evolutionbecause the firewall that they happen to be located behind isn't very friendly :)
20:21.15jbalcombjustinu i don't tell anyone about it though...
20:21.18a1fa4.2.2.2?
20:21.19justinulol
20:21.35Abydos313that number is used by alot of techs.. cisco owns it i think
20:21.35a1fadnsauth1.sys.gtei.net.
20:21.39jbalcombapple owns 4.x.x.x/8
20:21.39razu_justinu : ee is estonia :)
20:21.50justinuok
20:21.51a1fai want to own
20:21.55justinui thought 4.2.2.2 was verizon
20:22.01a1fa3.1.33.7
20:22.05jbalcombi'm pretty sure its apple
20:22.07a1faits my lifes dream
20:22.10Abydos313we could look it up
20:22.11jbalcomb4.2.2.1 and 4.2.2.2
20:22.15a1fawhois 4.2.2.2
20:22.24a1faOrgName:    Level 3 Communications, Inc.
20:22.26jbalcombAbydos313 you do it, cause i don't know how
20:22.40a1falevel3 owns it
20:22.42Abydos313dnsstuff.com
20:22.46a1faCIDR:       4.0.0.0/8
20:22.58jbalcombwhich one does MIT own?
20:23.05jbalcombthey have a /8 as well
20:23.08*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
20:25.34a1fathis is insane
20:25.42a1famaybe the ISP is randomly dropping packets
20:25.59justinuset up a ping
20:26.33a1fayah
20:26.42a1fawhat sip runs via tcp?
20:26.50justinuyou can run sip over TCP
20:26.56justinunot directly to asterisk tho, without patches
20:27.01a1fasucks
20:27.21justinua lot of times tcp can be worse than udp
20:27.26a1fayeah
20:27.28a1fasure is
20:27.40justinunow SCTP is what we need
20:30.45a1fathe router has gone mad now
20:30.50a1fanetgear
20:30.55a1fashe cant ping it anymore
20:31.17justinuanyone ever have issues with mixmonitor recording one side of the conversation much louder than the other?
20:31.24justinuplaying with the gains doesn't help
20:31.50justinueven tho the gain adjustments are working, the local side of the conversation in the monitored calls is always very loud
20:32.09jbalcombjustinu could be attenuation
20:32.37jbalcombjustinu can you set an option in the mixmonitor to do it's own adjustments?
20:32.45justinui've had some really bizzare experiences w/ this install
20:32.50justinuztmonitor also shows very high TX levels
20:33.05justinuhowever, if I set txgain -50, you can't hear shit on the calls, but ztmonitor still shows high TX levels
20:33.17jbalcombjustinu as does mine. i assume for now that mine is because of the gxp-2000 super gain mic.
20:33.47jbalcombjustinu now that is wierd. are you seeing any difference at all in ztmonitor?
20:34.02justinuno
20:34.09jbalcombjustinu do you have one of those 1ghz test tone numbers?
20:34.15justinuthis is a sangoma system... i need to talk to them about this
20:34.19justinusomething is screwy
20:34.23*** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no)
20:34.27jbalcombjustinu any chance you are monitoring the wrong channel?
20:34.30justinuit didn't act like this when I originally set it uop
20:34.32justinuno chance
20:34.55justinuhowever, I did update everything recently... asterisk 1.2.5, zaptel 1.2.4, wanpipe 2.3.3 current
20:34.57RoyK~seen zoa
20:35.16jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 5d 3h 51m 59s ago, saying: 'it looks kinda suspicious :p'.
20:35.17jbalcombjustinu how about a make clean ; make install on zaptel to see if it's jacked somehow?
20:35.17justinubut i've since reverted back to all the original software revs
20:35.17jbalcombjustinut i don't know wanpipe
20:35.17justinuand the behavior is still there
20:35.25*** part/#asterisk eliel (n=eliel@200.123.183.89)
20:35.56jbalcombjustinu maybe the upgrade/revert busted some lib or config?
20:36.01justinupossible
20:36.17*** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net)
20:36.18*** join/#asterisk mattodude (n=matt@gateway.digium.com)
20:36.36jbalcombjustinu have you looked around for file or links being used that point to 1.2.5 stuff?
20:36.49*** join/#asterisk Assid (n=assid@59.183.55.15)
20:36.51a1faok
20:36.53justinunot specifically... i'm mostly interested in the kernel modules
20:36.54Assidheya
20:36.59a1fajustinu : i think i know whats the problem
20:37.00a1faSPI
20:37.01justinui should look thru there
20:37.04a1faStateful Packet Inspection
20:37.07Assidum. is there a way to relaxdtmf for incoming iax calls?
20:37.08a1faon that netgear router
20:37.11justinuah, could very well be alfa
20:37.11a1fahahhaha
20:37.20a1faits actually simple flood protection
20:40.37*** join/#asterisk nsillik (n=nsillik@avenger.cis.temple.edu)
20:41.26nsillikhas anyone here had much luck getting outbound calling to landlines working with Gizmo from asterisk? I can call other gizmo numbers, but not the pstn
20:43.23*** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz)
20:44.55*** join/#asterisk Flauto (n=zhao@adsl-75-3-130-186.dsl.chcgil.sbcglobal.net)
20:45.09FlautoMar 20 14:26:01 WARNING[30440]: chan_sip.c:6840 check_via: 'Server217' is not a valid host
20:45.13Flautowhat is this for?
20:45.17LostFrogIs there any way to get the status of a sip peer from manager? Like we do from sip through hints?
20:46.45Flautoany idea?
20:46.57DelmarFlauto, check sip.conf for the section relating to Server217
20:47.13Flautohmm...
20:47.38DelmarFlauto, perhaps u have typo'd the host=
20:48.05DelmarFlauto, after u fix it do "sip reload" at console
20:48.13Flautothanks
20:48.19Flautolet me see
20:48.21justinulostfrog: there are peerstatus change events, in the manager, afaik
20:49.21Kattythere's a linux terminal program that will take an ip and tell me what ports it's listening on
20:49.31Kattysomeone refresh my memory as to the name of it
20:49.34Netgeeksnmap
20:49.37Kattythanks.
20:51.32Kattymy brain just sucks sometimes.
20:51.56tsumeHey hot babe! :D
20:52.07tsumeKatta-kun1! oi! :D
20:52.07Kattytake that comment and shove it
20:52.48tsume*slapped
20:54.46LostFrogjustinu: So, I have to listen to Evens.. :(
20:55.46justinu?
20:56.34HmmhesaysKatty: mine to
20:56.36Hmmhesays*too
20:57.29Hmmhesaysoh therapy can you please fill the void, am I retarded or am I just overjoyed
21:00.34*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
21:03.37a1falol
21:04.10a1faso shitty
21:04.20a1fanow i cant get a WAN IP via Cable Modem
21:04.28a1faso i cant do anything
21:04.43justinucable modems suck
21:04.47a1fasure does
21:07.38tsumeKatty: will you bear my children?
21:07.59a1facable is cool
21:08.08a1fadid you know you can get 30mibts down that crazy cable?
21:08.14tsumecable is definitely cool, you can uncap the modems
21:08.38a1fatsume : how do you uncap a modem?
21:08.39a1fa:P
21:08.51LostFrogillegaly.
21:09.03Inkubotdamn FLASH key
21:09.04Inkubot:\
21:09.08tsumea1fa: all cable implementations available use docsis
21:09.08Inkubotand damn ATA
21:09.22tsumealephcom: you don't need to know how ;) its for specials like I to know ;)
21:10.11KattyHmmhesays: bwha?
21:10.18KattyHmmhesays: what's going on with you? therapy?
21:12.17*** join/#asterisk zotz (n=zotz@24.231.32.85)
21:13.20HmmhesaysKatty well I got really drunk this weekend and tore FZ a new one
21:13.42Hmmhesaysi suppose that could be considered therapy
21:13.44Inkubotcan someone giveme an account on an Asterisk BOX to test a thing ?
21:13.51Hmmhesaystesting what
21:13.56Inkubotan ATA
21:14.02Hmmhesayspaypal me a 10
21:14.06*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
21:14.06*** mode/#asterisk [+o russellb] by ChanServ
21:14.11Inkubotpfff...
21:14.15a1fakeep talking
21:14.16a1fatsume
21:14.20a1fadocsis
21:14.27tsumedocsis :) yes
21:14.39tsumea1fa: you don't need to know about it ;)
21:14.40Hmmhesaysaight paypal me 5 bucks and one is all yours
21:15.03jbalcombInkubot i'll do it for $.99
21:15.08a1fatsume : what you got?
21:15.09Hmmhesayshahaa
21:15.21Inkuboteiiiiiiii guys i want to try just a thing.. no more than a minute
21:15.22a1fahow do you connect to it
21:15.22astra^^what is the difference betwenn g729 and g729a
21:15.30tsumea1fa: its a trade secret ;)
21:15.34Hmmhesaysg729 is annex A
21:15.40Hmmhesaysg729a even
21:15.57astra^^hw do i knw which g729 i am using
21:16.16*** part/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net)
21:16.26astra^^i bgt a g729 licence y'day
21:16.39astra^^is it g729a or g729
21:16.43Hmmhesayswouldn't matter to you anyway
21:16.48a1fatsume : i am sure the telco would know you uncapped your modem instantly
21:17.31Hmmhesaysg729a is less complex than g729
21:17.34tsumenope
21:17.43tsumethey can only tell if they are able to connect to the modem
21:18.09astra^^so wen i get g729 it supports both?
21:18.33a1faso it would raise supicions if they cant connect
21:18.36Hmmhesaysg729 a and b are stream compatible with g729
21:18.43a1faand they can immediatley assume, you fuxerod the modem
21:18.47Hmmhesays~google
21:18.49jbotit has been said that google is a search engine found at http://www.google.com/
21:19.19astra^^ok
21:19.23*** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr)
21:19.43iqastra^^: did you find out the number of legs it covers :) ?
21:19.46astra^^but tellme i jst bgt a new g729 frm digium
21:20.01Hmmhesayswhy do you care?
21:20.05Hmmhesays;)
21:20.18astra^^and weather it is g729a or g729
21:20.20iqastra^^: call Digium. They have good support
21:20.28astra^^please....
21:20.31astra^^:(
21:20.38russellbYou do not get g729b from Digium.
21:20.47iqastra^^: did you pay for it ;) ?
21:20.55KattyHmmhesays: FZ? tore? new one? mew?!
21:21.11astra^^yes i did
21:21.13KattyHmmhesays: speak kat you freak!
21:21.16astra^^10$/channel
21:21.21Hmmhesaysastra^^ you know what is really irritating?
21:21.30astra^^am sorry
21:21.35Hmmhesayshttp://www.digium.com/en/products/voice/g729codec.php <--
21:21.49Katty(=
21:21.55Hmmhesayswe're all n00bs at some point, but come on man, read a little
21:22.25astra^^ok what do i do to get g729a
21:22.48iDunno# come on let me hold you
21:22.51iDunno# touch you
21:22.54iDunno# feel you
21:22.57iDunno# always
21:22.58iDunno:)
21:23.02iq:)
21:23.07iqlol
21:23.09Hmmhesaysfeel the lurve bi@atch
21:23.16Hmmhesayslol
21:23.23Hmmhesayshot
21:23.52Hmmhesayswas it good for you?
21:24.35*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
21:24.37Hmmhesayslol
21:24.58*** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de)
21:25.47a1fawhats'up with all these describes
21:29.38*** join/#asterisk sanee (n=sanee@82.117.210.45)
21:32.08*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
21:35.03*** join/#asterisk Seldon1975 (n=someone@199.243.101.131)
21:35.20Seldon1975Q:what's red and invisible?
21:35.57iDunnonothing.
21:36.03Seldon1975A: no tomatoes
21:37.07tuxinator_linuxaren't we all
21:37.53X-Robdo some coding, you lazy buggers.
21:38.00Netgeeks!!
21:38.09Seldon1975hno point: our project was cancelled
21:38.28tuxinator_linuxX-Rob: I am.. doing that... coding
21:38.37Netgeeksfunction hello_world();
21:38.47Seldon1975while(true);
21:38.54jsharpprofit();
21:38.56jsharpendwhile()
21:39.16tuxinator_linuxyou know... coding is not as fun when you do it on a schedule
21:39.18Seldon197510 PRINT("I AM COOL");
21:39.20Seldon197520 GOTO 10
21:39.32Seldon1975tuxinator_linux: amen
21:40.01justinu>              v v  ,,,,,"Hello"< >48*,          v v,,,,,,"World!"< >25*,@
21:40.25tuxinator_linuxI used to go into computer stores... back in the day.... and quickly write programs in qbasic.... 'You've been hacked'... or something to that affect.... and then watch them try to figure it out.
21:40.52Seldon1975haha that was a lot of fun
21:40.55Netgeeksby the way, you didn't need that ; at the end of line 10, Seldon...
21:41.20Seldon1975Netgeeks: sorry, I sold my C64 about 20 years ago...
21:41.44tuxinator_linuxmost of us were born about 20 years ago
21:41.44*** join/#asterisk MattH (n=MattH@63.174.244.195)
21:41.45Netgeekshehe, yeah, I know, I'm even trying to remember if you needed the ().  I don't think so
21:41.53Seldon1975now I just use an emulator occasionally to play Defender of the Crown
21:42.03MattHHi... why would iax2 show channels show 99ms of jitter, when a ping between me and the other iax server comes up with 13ms?  is there really that much packet loss?
21:42.15justinujitter != latency or packet loss
21:42.18MattH00006/00017  00021/00020  00127ms  0122ms  0162ms  ulaw
21:42.34tuxinator_linuxback to work for me
21:42.49MattHjustinu, understood... but my ping times are a solid 13ms.. not going up and down huge amounts
21:43.36justinui'm not convinced those stats can be trusted either, matt
21:43.43MattHahh hrmm ok..
21:43.49ManxPowerjustinu, is that RPG-II?
21:43.50ManxPowerit's been 15 years since I did anything in RPG-II
21:43.58justinuManxPower: befunge ;)
21:44.08MattHwell my audio is pretty nasty to this host... to whem I have pretty good solid 13ms pings.. no drops... no jumps... yet the audio is bad.. and asterisk is showing the above
21:44.11MattHany thoughts?
21:44.21justinumatth: increase the frequency/size of the pings
21:44.30MattHjustinu, I did a ping -f!
21:44.30justinui use ping -i0.02 -s180 -c 5000
21:44.37MattHand then all came back fine.. will try larger ones
21:44.55justinuthat's pretty close to the kind of traffic an RTP stream is
21:45.04MattHk
21:45.31MattH221 packets transmitted, 219 received, 0% packet loss, time 5472ms
21:45.31MattHrtt min/avg/max/mdev = 13.061/13.998/16.703/0.688 ms, pipe 2
21:45.50justinulooks good, your problem must be somewhere else
21:46.11MattHyeah.. why would asterisk show high jitter issues, yet ping times would be fine?  that's wher eI am drawing a blank
21:46.23MattH1745 packets transmitted, 1743 received, 0% packet loss, time 43174ms
21:46.23r_evolutionjust != jitter O_O
21:46.23MattHrtt min/avg/max/mdev = 13.017/13.912/29.220/0.811 ms, pipe 3
21:46.25justinufucked up QoS?
21:46.30r_evolutionjustin !=jitter even
21:46.36ManxPowerMattH, Well, RTP is not ICMP for one thing.
21:46.53justinumaybe something is deliberately causing jitter on RTP or IAX packets
21:46.54*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
21:46.55MattHI'd think QoS.. but I have other iax streaming going to external hosts and they are ok
21:47.06ManxPowerAlso Asterisk's jittter buffer (what little there is of it) has known issues.
21:47.08tmccraryMar 20 16:43:45 WARNING[17480]: chan_zap.c:10868 setup_zap: Ignoring signalling
21:47.08tmccraryMar 20 16:43:45 WARNING[17480]: chan_zap.c:10868 setup_zap: Ignoring switchtype
21:47.12MattHcould be... but again... other iax2 connections sound fine
21:47.20tmccraryis that normal for a PRI connection with asterisk on a TE110P?
21:47.21ManxPowerMattH, what happens if your turn OFF the jitter buffer
21:47.25justinui'd use ethereal to analyze the RTP stream
21:47.27MattHI dunno.. let me try
21:47.36justinutmccrary: it's normal when you do a reload chan_zap.so
21:47.42tmccraryoh ok, thanks
21:48.16MattHinterestingly.... it sounds better!
21:48.31MattHwow that was odd
21:48.32justinuheh
21:48.39MattHit's POTS quality
21:48.39ManxPowertmccrary, you cannot change those things when doing a reload, you can only change them my stopping and starting Asterisk or by unloading and reloading chan_zap.so
21:49.22MattHwell now that's interesting
21:49.27MattHso it's a jitterbuffer bug?
21:49.28*** join/#asterisk alexis101 (n=alex@70.54.204.92)
21:49.37alexis101hello all
21:50.05alexis101I was wondering if any of you know how to implement *67
21:50.07ManxPowerMattH, pretty common, actually.
21:50.17ManxPowerMy satillite connection is going up and down more often than linda lovelace.
21:50.26a1faManxPower :lol
21:50.30MattHhehe interesting
21:50.41a1faManxPower : how does that work? do you dial-in somewhere?
21:50.51tzangeralexis101: I explained this earlier this aft I thought
21:51.10ManxPowerDamn rain
21:51.24alexis101ok i will check the logs
21:51.25jsharpmmm Ku band rain fade
21:51.26tmccraryIs there like a zap debug command?
21:51.28ManxPowera1fa, Um, no.  I use a dish
21:51.31*** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no)
21:51.42ManxPowertmccrary, what information are you trying to see?
21:51.51NuggetI hear ManxPower is quite a dish.
21:51.55tmccraryIs the channel is busy or what
21:52.00ManxPowerjsharp, I shall celebrate the say I can get a T-1 in here.
21:52.06ManxPowersay == day
21:52.40jsharpJust a T1?  No T3?  OC-3?
21:53.09a1fai dont have those problems
21:53.11jsharpTo the house?
21:53.19tmccrarytmccrary has an OC-256
21:53.40ManxPowerjsharp, My DirectTV has a signal strength of 93 right now, where the DirecWay dish has a signal strength of 41
21:53.43*** join/#asterisk shimi (n=shimi@unaffiliated/shimi)
21:53.47blitzrageI have an OC-512 to my house
21:53.51ManxPowerOf course I paid the extra $50 to get the larger dish too
21:53.58tzangerexten => _NXXXXXX,1,do.what.you.need.for.all.calls.here
21:53.58tzangerexten => _NXXXXXX,n,GotoIf($[${BLOCKCID} = 1]?foo)
21:53.58tzangerexten => _NXXXXXX,n,Set(CID(name)=SOME NAME)
21:53.58tzangerexten => _NXXXXXX,n,Set(CID(num)=8012345678)
21:53.58tzangerexten => _NXXXXXX,n(foo)do.more.of.what.you.need.for.all.calls.here
21:54.00tzangerexten => _NXXXXXX,n,Dial(Tech/${EXTEN}@peer,timeout,options)
21:54.03tzangeralexis101: ^^
21:54.09jsharpDirecTV and Direcway on the same dish?
21:54.10tzangerand then simply do this
21:54.12ManxPowerjsharp, I live on the top of a mountian, 11 miles from the nearest CO
21:54.12justinuis directway 2 way over satellite, or do you have a modem?
21:54.17ManxPowerjsharp, hell no!
21:54.25jsharpDirecway is 2 way.
21:54.25ManxPowerDirecWay is 2-way
21:54.28shimiAnyone perhaps has a pointer for setting up asterisk to call back upon request, preferrably to a given number, then execute something like running DISA ?
21:54.29justinuic
21:54.37blitzragetzanger: CID == CALLERID I think right?
21:54.43jsharpSounds like the monkey who installed your direcway dish didn't point it right.
21:54.44tzangerexten => _*67.,1,Set(BLOCKCID=1)
21:54.44tzangerexten => _*67.,n,Goto(${EXTEN:3},1)
21:54.46blitzrageor is CID valid too?
21:54.49ManxPowerjsharp, 1) you cannot put both on the same dish and 2) I'm riding on the owner's DirecWay connection, but have my own DirecTV dish
21:54.49*** join/#asterisk bazz (n=nick@fw.marklogic.com)
21:54.57PoWeRKiLLhow can I create a new cdr column in the db and set it from extension.conf ?
21:55.02tzangerblitzrage: yeah, that's just off the top of my head, there may be some minor changes to make Asterisk "grok" it
21:55.24Octothorpetzanger:  pastebin is your friend
21:55.24Octothorpe~pb
21:55.25jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
21:55.26blitzragetzanger: understood -- I wish CALLERID function was just CID actually
21:55.26ManxPowerjsharp, DirecWay is VERY picky about being just perfect aim.
21:55.30tzangerOctothorpe: I'm familliar with pastebin
21:55.35tzangerblitzrage: me too :-)
21:55.36ManxPowerWe will be moving the dish to get a clearer view of the sat
21:55.53blitzragethere's nothing wrong with 4-5 lines of pasting
21:55.58tzangerPersonally I am not that keen on _*67. as a pattern match, I like my dialplans to match more strictly
21:55.59blitzrageits when its obscene that its a problem
21:56.18MattHewww.. so now I'm just getting bad lag
21:56.28tzangerso personally I'd have _*67NXXXXXX, _*67NXXNXXXXXX, and the _*670 and *671 variants :-)
21:56.37bazzwhat i'd like to do: have asterisk call my voicemail, dial a few numbers to get to the record a message prompt, have asterisk play a sound file over the line (to record it), dial a few more #'s to confirm, and hang up.  (a) can i do this (b) how hard is it (c) if i can do this, where should i begin reading to figure out how
21:56.45jsharpAh.  DirecTV and Direcway are on different sats.  I thought they were all serviced off the same satellite.
21:57.26ManxPowerjsharp, They might be on the same sats, but they are different companys, do not offer combined billing and do not offer combined dishes
21:57.29Hmmhesaysi'm thinking about getting a razr
21:57.33Hmmhesaysare they any good?
21:57.46jsharpOh.  Gotcha.
21:57.58ManxPowerHmmhesays, shaving is overrated
21:58.11justinumotorola phones suck
21:58.11Hmmhesaysi meant phone
21:58.29a1falilo
21:58.40jpm_SDHmmhesays, Verizon reports a 46 percent return rate on that phone... take that for what it's worth.
21:58.46jsharpI've used standalone direcway systems and they all made me want to stab myself with a rusty trout.
21:58.48a1falilo u crazy ass :)
21:58.54tzanger46%?  holy crap
21:58.56ManxPowerBut Jessica Simpson has that phone!
21:59.09ManxPowerjsharp, Yeah.
21:59.24a1fawhat phone?
21:59.28shimiAnyone perhaps has a pointer for setting up asterisk to call back upon request, preferrably to a number given by the caller, then when the calledback number answers, execute something like running DISA to let them dial on my expense?
21:59.35tzangerdirectway for interactive?  isn't the lag pretty decent?
21:59.46justinu1.5 seconds or so, iirc
21:59.53*** join/#asterisk digime (n=digime@ip68-101-196-93.sd.sd.cox.net)
22:00.13ManxPowerjsharp, but at this point I cannot afford the $500 - $700/month for a T-1, and there is no cable or DSL here.
22:00.13ManxPoweronce we get a few more seasonals here I'll be able to offset the cost be selling interent and phone service to them
22:00.44ManxPowerand since cell phones don't work very well up here (pretty much anything but Verizon does not work)  I'm pretty sure they will want POTS service
22:00.46jsharpThe latency bounced all around the place cause of the TDM multiplexing they use on the enduser-to-satellite connection.
22:00.57ManxPowertzafrir, ping me
22:01.01justinuthere's no existing pots service?
22:01.03jsharpIt was all sorts of jittery.
22:01.11*** join/#asterisk Dr-Linux (n=Linux@host202-147-168-130.lhr.dancom.net.pk)
22:01.27ManxPowerjustinu, there is pots service here, but nobody wants to pay $40/month for a service they will only use on the weekend.
22:01.28Hmmhesaysyeah their like on the 3rd generation of that phone htough
22:01.38Octothorpeshimi:  check nerdvittles.com
22:01.58justinumakes sense
22:02.12ManxPowerI figure $25/month for combined interent and telephone.
22:02.17Octothorpeshimi:  it is something like "phone home" or something like that, they have a whole tutorial on their site
22:02.19jpm_SDHmmhesays, as I said.. take it for what it's worth... They may be fine now, but I'd be leary.
22:03.32*** join/#asterisk maxx4life (n=maxx4lif@71-35-210-12.slkc.qwest.net)
22:03.34tmccraryI setup a PRI connection between a TE110P and a traditional phone system. But I keep getting: SIP/2.0 486 Busy here when I try to call it.
22:04.01bazzanyone?
22:05.12shimihmmm... web activated DISA... psshh :)
22:05.39*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
22:06.18tzangertmccrary: can you get it to work without the SIP
22:06.30tzangeri.e can you place calls through the PRI with Local/ or something?
22:06.31*** join/#asterisk amdtech (n=stdamd11@ab1-1-246.shsu.edu)
22:07.04tmccraryHow would I go about doing that?
22:07.21tmccraryor do you mean with a traditional phone?
22:07.32astra^^Mar 21 03:29:07 WARNING[32272]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses!
22:07.35tzafrirManxPower, here
22:07.52tmccraryI think this may be an issue with the comdial phone system, i was just curious if this is a common issue or not
22:08.19astra^^i am gettin tis error ..
22:08.21astra^^Mar 21 03:29:07 WARNING[32272]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses!
22:08.31astra^^i have a g729 licence
22:08.32justinupretty clear to me
22:08.46astra^^i hve a licence i bgt yday
22:08.54jsharpHow many licenses?  How many calls do you have going?
22:08.56[TK]D-Fenderastra^^ : Its in USE and a device is ask to use one more than you have free.
22:09.32astra^^i gt only 1 licence
22:10.45shimiOctothorpe, unfortunately it does calling via PHP script - that I already know how to do :) I need something that accepts a call from one of my DIDs (or whatever), then asks for a phone number, then waits a few seconds, dials to it, and connects that call with DISA. :)
22:11.06astra^^wid 1 licence we can place only 1 call right
22:11.06*** join/#asterisk FuriousGeorge (n=Brian@ool-43536ea8.dyn.optonline.net)
22:12.00[TK]D-Fenderastra^^ : Correct
22:12.33astra^^ok bt nw my client placed only one call frm a switch .. for testing .. and i gt tis error
22:12.56tmccraryPatents suck
22:13.13tmccraryDo you have missing buttons on your keyboard or something?
22:13.36astra^^no why did u ask that
22:13.41tmccraryOr perhaps missing fingers
22:13.45astra^^:'
22:13.49tmccrarybc joo tk lk ths
22:13.54tzangerhahahaha
22:13.58tzangerI hate when people do that
22:13.58[TK]D-Fendertmccrary : NO, just not his native language.
22:14.02astra^^hmmm... not funny
22:14.09[TK]D-Fendertmccrary : if u cn rd th tn u cn pgm n c ;)
22:14.24[TK]D-Fendertmccrary : if u cn rd ts tn u cn pgm n c ;)
22:14.28tzangerI don't know if that's because of a native language or simply because they're so used to SMSing from phones without T9
22:15.32astra^^lol very funny.. ha  ha  ha.. and 3 more ha..
22:15.48tzangerastra^^: once license = 1 transcode at a time
22:15.58tzangerastra^^: 2 licenses = 2 transcodes at the same time
22:15.58tzangeretc
22:16.12astra^^thank tou choo much
22:16.25astra^^:>>
22:16.28tmccraryastra^^: When you wrote that ha ha ha part, I pictured that kid from indiana jones and the temple of doom
22:16.41digimeanyone know where the asterlink guys are?
22:16.48tzanger#openpbx likely
22:16.55digimeoh yeah
22:17.04justinu#asterlink?
22:17.37digimeno they are not in their channel
22:17.45digimewho are people using these days for incoming DID?
22:19.21r_evolutionwell that works... werd.
22:19.42r_evolutionTK = funny guy today ;)
22:20.03digimewho are people using for incoming DID?
22:20.11denonnufone
22:20.15[TK]D-Fenderr_evolution : nice to see SOMEBODY got it ;)
22:20.39FuriousGeorgehe all
22:20.39r_evolutionoh yes... may i add how much i enjoyed the ';')
22:20.52FuriousGeorgeHEY
22:21.00ManWithYellowBat?
22:21.02tzangerhahahha
22:21.16ManWithYellowBatwhere'd that damn dirty ape go
22:21.36*** join/#asterisk ronn (n=zakforev@84-45-132-117.no-dns-yet.enta.net)
22:22.10asterboydam corn, unpredictable areodynamics
22:22.17r_evolutionew
22:22.19r_evolutionyou're gross
22:22.35FuriousGeorgelol
22:22.42xbmodder_lappyweeee
22:23.38Mauro__any good opensource softphone for linux/bsd?
22:23.50FuriousGeorgeMauro__: no
22:23.51FuriousGeorgeexiga
22:23.52FuriousGeorgeis new
22:23.58Mauro__:D
22:24.02FuriousGeorgehavent tried
22:24.03*** part/#asterisk a1fa (n=a1fa@207.210.210.202)
22:24.18*** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no)
22:24.38r_evolutionnote to self
22:24.45r_evolutiondo not listen to jungle whilst coding...
22:24.54r_evolutiontype too fast... make sloppy mistakes
22:24.56tuxinator_linuxjungle?
22:25.00r_evolutionyessir
22:25.02r_evolutiondrum and bass
22:25.04r_evolutionedm
22:25.26r_evolutionthink Roni Size... Dieselboy... Concord Dawn... Black Sun Empire... Evol Intent... etc
22:25.27*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
22:25.51*** part/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
22:26.06r_evolutionwhat in the hell...
22:26.07jpm_SDphotek
22:26.08tuxinator_linuxskin a wizard?
22:26.09r_evolutionskins a wizard?
22:26.15r_evolutionphotek is decent
22:26.19r_evolutionthat sounds dirty...
22:26.37jpm_SDI hear Wizards make good cloaks.
22:26.57*** join/#asterisk doolph (n=doolph@201.227.72.230)
22:26.58FuriousGeorgeit does sound dirty doesnt it
22:27.07FuriousGeorgei dont think it is though
22:27.16doolphsup
22:27.36doolphany astguiclient experimented user?
22:27.54r_evolutionare we experimenting on people now?
22:27.57r_evolutionsweet.
22:28.12tuxinator_linuxwhere do I sign up r_evolution for the experiments
22:28.19doolphLaughing Out Loud
22:28.37*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
22:28.45jpm_SDYou know.. you can just   LOL.. we all know what it means now.
22:28.58r_evolution...
22:29.00r_evolutionBURN!
22:29.04doolphit wasnt me
22:29.04r_evolutionyou dont sign me up
22:29.30r_evolutionthat is classic...
22:29.31r_evolution<doolph> Laughing Out Loud
22:29.31r_evolution* ManxPower has joined #asterisk
22:29.31r_evolution<jpm_SD> You know.. you can just   LOL.. we all know what it means now.
22:29.35r_evolutioni love it.
22:29.42*** part/#asterisk tmccrary (n=tmccrary@68.78.185.254)
22:29.43*** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net)
22:29.47*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:29.57doolphwas the script
22:30.01doolphit auto fillup
22:30.57*** join/#asterisk Mitja (n=Mitja@cpe2-25-116.cable.triera.net)
22:31.02*** part/#asterisk Mitja (n=Mitja@cpe2-25-116.cable.triera.net)
22:31.09r_evolutionlol
22:31.12r_evolutionhrrmmm
22:31.13*** join/#asterisk Op3r (n=op3r@202.71.189.90)
22:31.15r_evolutionLOL
22:31.17r_evolutionhrmm
22:31.18r_evolutionO_o
22:31.21r_evolutionLoL
22:31.23r_evolutionhrmmm
22:31.27Op3rHi can I ask questions regarding vicidial?
22:31.33Op3rand how it works?
22:31.41r_evolutionyou can always ask them... but it doesnt mean someone will answer them
22:31.52doolphOp3r it's predictive dialer
22:32.01Op3rdoolph: I know
22:32.03*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
22:33.19doolphthen
22:33.19Op3rbecause here is the setup
22:33.19Op3rwe connected analog phones to the quintums
22:33.19Op3rsometimes vicidial can connect to the phones
22:33.19doolphI think they use sip now
22:33.19Op3rbut sometimes if a user just hung up the phone without login off first when they try to login it wont connect to the phones anymore
22:33.19Op3rthe hardphones's protocol was sip
22:34.17Op3rI was wondering whats happening
22:34.20Op3r:(
22:34.54Op3rI just learned voip for about 2 weeks I am supposed to be the comp admin not the pbx admin L(
22:36.59PrimerAnyone know if it's possible to map a button on a 7960 to "login/logout of a support queue" on asterisk?
22:38.03Hmmhesayssure, make your voicemail button hit that extension
22:38.13Primerwell, we also need voicemail
22:38.20Primerso it can't be that button
22:38.24Mauro__*77
22:38.25Mauro__:D
22:38.36PrimerMauro__: callate
22:38.41Mauro__:)
22:39.19*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
22:39.37r_evolutionjust because i really enjoyed that
22:39.38r_evolution<doolph> Laughing Out Loud
22:39.38r_evolution<jpm_SD> You know.. you can just   LOL.. we all know what it means now.
22:39.40r_evolution:-D
22:39.43Op3rdoes anyone knows any great vicidial how to;s and tutorial?
22:39.57doolphwhat
22:45.45*** join/#asterisk ToTo (n=ToTo@host62-142.pool874.interbusiness.it)
22:46.47*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
22:46.55tmccrarywhat is the a bchannel and dchanne;?
22:47.33bweschketmccrary: b channel carries voice - d channel carries signaling for a group of b channels in a PRI group
22:47.53tmccraryah
22:47.55tmccrarythanks
22:48.30*** join/#asterisk Strom_C (i=strom@66.159.243.60)
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22:54.18*** join/#asterisk Nodren (n=nodren@64.193.95.10)
22:54.22Hmmhesaysi subscribed to guitarone today
22:54.26Hmmhesays12 bucks a year, cheap
22:55.11Nodrenwhat distro would anyone recommend for running asterisk with a digium TDM400P
22:55.25justinuwhatever you feel comfortable with
22:55.37*** join/#asterisk naturalblue (n=Administ@87.192.100.109)
22:55.41Hmmhesaysi recommend "insert *nix flavor you feel comfortable with here" ___________
22:55.55Nodrenwell i've used debian and rehat based distros before
22:56.03Nodreni've had nothing but pure trouble trying to install this on centos
22:56.12Hmmhesaysso don't use cent
22:56.17justinucentos works for me
22:56.29Nodreni mean freebsd? ubuntu? debian sarge?
22:56.39justinufbsd is not a linux disto
22:56.40justinudistro
22:56.47Hmmhesayssarge works fine
22:56.48Nodrenthen not fbsd :P
22:56.51Hmmhesaysubuntu works fine
22:56.59Nodrenubuntu works great?
22:57.01Nodreni might try that.
22:57.12Hmmhesayshell fbsd works fine for some people
22:57.35[TK]D-FenderNodren : Ubuntu is missing a LOT of packages required to build * and its components.  probably better off with a more "complete" distro
22:58.04Hmmhesaysi disagree [TK]D-Fender
22:58.25Hmmhesaysyou get apt-get install every package you need in ubuntu
23:00.24[TK]D-FenderHmmhesays : I never said you coudn't do it, rather that you'll have plenty to do before you canhope to be up and running.
23:00.53Hmmhesaysi just did one a couple days ago,  5 minutes of apt'ing and you're done
23:01.07*** part/#asterisk amdtech (n=stdamd11@ab1-1-246.shsu.edu)
23:01.22[TK]D-FenderHmmhesays : True, but thats for the more experienced user who know whre to geb everything from and what to get.
23:01.27Abydos313amportal howto has a full list of requirements before * should be installed
23:01.45[TK]D-FenderHmmhesays : just not as friendly "out of the box" as most other common ones
23:01.57Strom_CAMP is pure assrape. Don't use it.
23:02.12Abydos313you can use the howto to list the packages required
23:02.15justinulol
23:02.25Abydos313see it's good for something
23:02.39Hmmhesays[TK]D-Fender you are right, its no debian
23:03.58*** join/#asterisk VahramI (n=nospam@83.139.4.112)
23:03.58*** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk)
23:04.29*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net)
23:04.47shimiI've set up asterisk with a script that copies a call request to a /var/spool/asterisk/outgoing. console says that all the commands were executed, yet it seems that the outgoing call request is not performed. what can it be?
23:06.04VahramIcollegues, who has good practice writing cdrs by Agi script?
23:06.39mishehuhmm...  I forgot, do polycom ip 50x's get the sntp setting from dhcp or from the config files?
23:06.52MstlyHrmlsyes :-)
23:07.09[TK]D-Fendermishehu : Both
23:07.13MstlyHrmlsif it's in DHCP, it will use that, but you can also set it in the config files (IIRC, anyways. I just use DHCP)
23:07.27shimihow do I designate ANY ZAP interface when setting up an outgoing call ?
23:08.02shimiI mean I have Zap with 31 channels, how do I tell asterisk to just use a free one? Zap/what ?
23:08.19mishehu[TK]D-Fender: know what the line in the configuration files is (and which file) ?  I don't have anything in MAC-phone.xml, and the phones are hardcoded with IP information, and while in the settings menu on the phone itself it shows the ip of the ntp server, the web interface shows "clock" and the wrong offset.
23:08.35mishehushimi: define a group
23:08.44shimihey, mishehu! :)
23:08.47shimisup dude?
23:09.03mishehushimi: busy as all hell.  you still hanging around #israel?
23:09.26shimirarely
23:09.29[TK]D-Fendermishehu : its not in the phone, its in sip.cfg somewhere
23:09.34shimiI have ZAP/g0 - should I use that?
23:09.52[TK]D-Fendermishehu : just search for SNTP or GMT
23:10.06mishehushimi: *nod*
23:10.08shimiuhm, I will just try, heh :)
23:10.48mishehushimi: I'm only on #israel efnet, but I rarely even speak there these days.  hard to take 3 classes at the college, run a small biz fulltime, and still have time to waste.  ;-)
23:11.22*** join/#asterisk Lurkan (n=Lurkan@201.152.101.6)
23:11.29shimioh well :)
23:11.36shimiok, it looks better now, it's not complaining
23:11.45shimithough it says that it couldn't complete the call :\
23:13.24*** join/#asterisk heart (n=zippetto@lugbari/people/heart)
23:13.59Flautohey people
23:14.11Flautoi have a stupid question here
23:14.54Flautohow can i set up to answer a call by auto attentant and to give it only two chances of dialing an extension and then just hung up
23:15.09justinucreate a variable
23:15.11justinuincrement it
23:15.12justinutest it
23:15.34Flautoi have a variable
23:15.41Flautobut it goes in circles
23:15.41*** join/#asterisk dimmik (n=dimmik@static217244.dsl.hol.gr)
23:17.02Flautojustinu, do you mind me to show you my settings privately?
23:17.26r_evolutionINFINITE LOOP PWNS YOU!
23:17.27r_evolution;)
23:17.36Flautoright
23:17.42justinui'm not all that good at IVR stuff in the dialplan
23:17.50Flautookay
23:18.00dimmikhi everyone. I am trying to set * to use g711 for the local phones and g729 when talking to a sip trunk. I am trying to avoid transcoding. Is this possible?
23:18.00justinujust giving you a suggestion
23:18.00Flautor_evolution, would you help?
23:18.06justinuuse noop to display your variable
23:18.09justinufigure out why it's not incrementing
23:18.26Strom_Cdimmik, um, you have to transcode if you're going from 711 to 729
23:19.03dimmikwhy so? both the sip trunk and the phone are capable of using g729
23:19.15Flautono, at the end of the variable, it is waiting for the caller to dial an extension
23:19.40justinuyou need to create a counter variable
23:19.49Strom_Cdimmik...so you want the phone to talk 729 to asterisk if the call goes out over a trunk, yet 711 if the call is internal?
23:19.54Flautohow would i do it?
23:19.54justinuso you can track how many times that particular channel has been thru the menu
23:19.59dimmikbtw, I am not using Tt etc in the dial cmd
23:20.14justinusmething like Set(COUNT=1)
23:20.26justinuor Set(COUNT=${COUNT}+1)?
23:21.08dimmikStrom_C : I want the phone to talk g729 to the trunk without * touching the media
23:21.10*** join/#asterisk AlexCTI (n=alex@adsl-072-156-253-012.sip.mco.bellsouth.net)
23:21.10Flautowhere should i put it
23:21.38justinuright before it plays the menu to ask for an extension
23:21.40AlexCTIHi.
23:21.42justinuthen test it
23:21.44Strom_Cdimmik, so have the phone talk 729 to asterisk and then asterisk will function as a pass-through
23:21.45dimmikStrom_C : Basically, making * act as a sip proxy
23:21.50*** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com)
23:21.56Flautookay
23:22.00dimmikStrom_C : exactly
23:22.21Strom_Cdimmik, I wasn't asking you, I was telling you :)
23:22.21justinudimmik, you use the allow, and disallow statements in sip.conf to control what codec stuff uses
23:22.28*** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net)
23:22.46dimmikStrom_C : I tryied it without success
23:23.01Strom_Cwhat happened?
23:23.05Strom_Cdefine "without success" :)
23:23.14dimmikthat was for justinu :)
23:23.43Strom_Cthen why did you direct the statement at me?
23:23.44justinucodec negotiation is kinda lame
23:23.50justinuit might be that you can't do what you want
23:23.54dimmikwell, I set the defaults to disallow=all and allow=g729 and allow g711
23:24.09justinuin that case, if the phone supports g729, it'll choose that first
23:24.16dimmikand for the sip trunk just  disallow=all and allow=g729
23:24.20Strom_Cdimmik, why do you want g711 if you're restricting everything to g729?
23:24.37dimmikI want g711 for internal calls
23:25.06Strom_Chow many simultaneous calls are you planning on having your asterisk box handle?
23:25.36dimmikjustinu: this is correct, still I wanted to have g711 for internal calls
23:25.45justinui understand what you want
23:25.48justinui don't know the answer tho
23:25.54justinuit may not be posible
23:26.01[TK]D-FenderStrom_C : then just set the phones to G711 and te SIP peer to G729.  End of story.  * will transcode anything goin over that trunk.
23:26.02justinuor you may need some trickery in your dialplan
23:26.04dimmikStrom_C : just a few, transcoding is not possible though
23:26.06Strom_Cdimmik, how many simultaneous calls are you planning on having your asterisk box handle?
23:26.13Strom_Cdimmik, why is transcoding not possible?
23:26.33Strom_Care you too cheap to buy the licenses or something?
23:26.38dimmiki have a box with geode cpu
23:26.48[TK]D-FenderThat'd do it :)
23:27.19Strom_Cdimmik, get a real box :)
23:27.24*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
23:27.24dimmikhehe
23:27.48tsumegeode is good for sliim devices
23:27.53tsumenot for servers
23:27.55dimmikStill, codec negotiation seems wrong in *
23:28.47justinuyou'd need to dynamically set the codec basedon the destination of the call
23:28.52Strom_Cexactly
23:28.56Strom_Cand that's a bit mad
23:29.02websaeanyone have a good suggestion (cost efficient) for a company that has 9 phone lines (8 voice and 1 FAX), they have a traditional PBX system for PSTN lines, they are going to be switching over to VoIP instead of SBC...what's a good interface to bring in the VoIP lines to this PBX that they already have, any suggestions, anyone :)?
23:29.14dimmikIs this possible?
23:29.29justinui don't think so
23:29.30Strom_Cwebsae, rip out the analog lines, install a PRI.
23:29.33justinunot out of the box
23:29.49mishehuanalog lines...  cacka poo poo
23:29.58websaethey don't want a PRI
23:29.59justinuwebsae: you want something like a Lucent TNT
23:30.03dimmikthis will require a reinvite, though
23:30.10websaethey wa show...oh
23:30.17websaewhoops...sorry
23:30.21justinuor a 12 port SIP to FXO gateway
23:30.39Strom_Cjesus, don't you people know about the rule of two-wire conversion? :)
23:30.52websaeokay...
23:31.01websae12 port SIP to FXO..would work
23:31.08websaei could just wire that in right?
23:31.08Qwell[]Strom_C: When going from two-wire, to ethernet, reuse existing cables? ;)
23:31.20Strom_Cwebsae, why do they want to hang on to their legacy PBX?
23:31.46justinuwebsae: yeah, it would basically just drop right in
23:31.47Flautodid not work
23:31.58websaethey have PSTN phones (FXS)....they don't want to buy 20 new ones
23:32.02justinuan asterisk machine w/ a TDM2400 would also work
23:32.16justinuor some other high density analog telephony board
23:32.19Strom_Cwebsae, asterisk machine with a quad-span T1 card, PRI, and two channel banks
23:32.52websaejustinu...know of any good 12 port SIP to FXO?
23:33.00Qwell[]websae: Asterisk :P
23:33.04mog_workdamn
23:33.06mog_workbeat me to it Qwell
23:33.08Qwell[]mog_work: damn?
23:33.09Qwell[]oh :p
23:33.14mog_worki was gonna say 2400p and asterisk
23:33.16justinuwebsae: i have no hands on experience with such thing
23:33.19Qwell[]I'm looking at skinny
23:33.23Qwell[]ha!
23:33.27justinuum, I said asterisk and 2400
23:33.34Qwell[]It crashes my phone when I lift up the handset :P
23:33.40websaeso setup an asterisk box with a 2400
23:33.40Strom_Cwebsae, you can kludge something together or you can do it right.  It would be much better to give them a unified solution than to kludge yet another part onto their existing system
23:33.45mog_workchan_skinny Qwell ?
23:33.46Qwell[]it registers though...yay me!
23:33.49Qwell[]mog_work: of course
23:33.54mog_workgood start
23:33.56Qwell[]heh
23:33.59mog_worklots of fun stuff to  debug
23:34.05mog_workyou are pretty lucky
23:34.13Qwell[]how so?
23:34.18mog_workyou get all the fun stuff
23:34.20websaechannel banks are expensive
23:34.22mog_work^_^
23:34.25Qwell[]oh, you can do it if you'd like. ;)
23:34.27Qwell[]I'm a nice guy
23:34.27justinuyou don't need channel banks
23:34.33mog_workbye
23:34.39Qwell[]heh
23:34.59*** join/#asterisk Andr3w_ (n=Andrew@stjhnf0122w-142162049036.pppoe-dynamic.nl.aliant.net)
23:35.24websaejust asterisk and a TDM 400
23:35.25Strom_Cwebsae, if they have 20 existing stations, a channel bank is going to be as cost-effective per station as a multi-port FXS gateway
23:35.33justinuTDM400, you'll need 3 of them
23:35.41websae*2400
23:35.49justinuhe wants to keep the existing PBX
23:36.00Strom_Cscrew keeping the existing PBX
23:36.20justinuthat's his decision
23:36.23Strom_Cyou don't graft kludges onto existing hardware if you're going to do the install with any sense
23:36.24dimmikjust get 6 pap2 and get rid of it
23:36.26justinui'm not going to try and sway him
23:36.51websaeokay...get rid of exisitng PBX
23:37.00websaeand get channel banks then?
23:37.04websaewhich ones?
23:37.04*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
23:37.04Strom_Cwebsae, do this
23:37.11Strom_C- asterisk box
23:37.12justinunow you need two TDM2400s, or a dual span T1 card + channel banks
23:37.16Strom_C- quad-span T1 card
23:37.20Strom_C- channel bank
23:37.26Strom_C- replace POTS lines with PRI
23:37.36Strom_C- renegotiate long-distance contract
23:37.43Strom_Cyou will save them plenty of money
23:37.46websaei ahve termination
23:37.55websaei have sip origination adn termination
23:38.04websaei don't need PRI
23:38.06*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
23:38.15Strom_CPRI will be less expensive than nine analog lines
23:38.27websaei have a VoIP carrier
23:38.31*** part/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)
23:38.32websaethat uses SIP...
23:38.38Strom_Cyou're going voip only?  you're taking a big risk...
23:38.40justinuare you happy with their quality?
23:38.42Strom_Cwho is the carrier?
23:38.44justinubecause yes, that's risky
23:38.53websaeyes i am very happy with their quality
23:38.56harryvvI dont know why it is but at cirtain times I have to shut down asterisk then when restarting it get a errr about outch! error while writing audio data broken pipe. zaptel/wcfxo would not load then I would recompile them. this is probebly a old issue and has been corrected with the latest patches.
23:39.37websaeso i don't need PRI
23:39.44Strom_Cwebsae, WHO IS THE CARRIER?
23:39.48harryvvAlso, is it typical to see two instances of asterisk running? I have been seeing this as of late.
23:39.55*** join/#asterisk unmanaged (n=unmanage@64.89.118.139)
23:39.59websaeComSolo
23:40.29Strom_Care you installing a dedicated T1 for the voip traffic?
23:40.43websaethey have biz class roadrunner connection
23:40.48Strom_Cwhat speed?
23:40.56websae5mbs i think
23:40.59Strom_Csymmetric?
23:41.01websae1.5mb
23:41.11Strom_Cwait wait wait wait
23:41.12websaethey have aplenty of bandwidth
23:41.25harryvvhi Strom_C
23:41.26Strom_Cyou're going to run their voice AND data traffic over a 1.5 megabit connection??
23:41.31Strom_Care you mad?
23:41.33websae1.5 up
23:41.35harryvvI would not
23:41.36justinui wouldn't run shit over a roadrunner connection
23:41.37justinuthey suck
23:41.44*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
23:41.49websaetheir fiber is quite nice :)
23:41.52Strom_Cand over roadrunner no less
23:42.01unmanagedin the current SVN release of asterisk, show application voicemail says that you can put an option for gain on rec of VM but I can't seem to get it to work...
23:42.06Strom_Ccan we check this guy into the asylum now? :)
23:42.15justinuyou oughta let him dig his own grave
23:42.24Strom_Ctrue, I should
23:42.28harryvvAre there any quick test tools on the market to do a snap shot of network bandwith for medium sized networks?
23:42.35websaeso i need a asterisk box with what in it now?
23:42.42websaefor the 20 analog phones
23:42.49websae?
23:42.54Strom_CI told you already
23:42.55justinusingle span T1 card + channel bank
23:42.57justinuor TDM2400
23:43.02dimmikbtw I thing I found something regarding codec negotioation : http://bugs.digium.com/view.php?id=4825
23:43.07unmanagedt1 card and channel bank
23:43.11websaei don't need a t1 card
23:43.19unmanagederr
23:43.26harryvvweb, how many channels are you going to be using?
23:43.34websae20 analog phones
23:43.34justinuyou might be able to get the T1 card + Channel bank for less
23:43.38justinuthe TDM2400 card is like 2500 bucks
23:43.38websae9 sip channels
23:43.40unmanagedyou can get 20 ata's
23:43.52harryvvweb, comming into your office?
23:43.56Strom_Cwebsae, trust me from experience.  You are going to severely regret running all their voice and data traffic over a single cable modem connection.
23:43.59justinuTE110P is 450
23:44.11justinuchannel bank? not sure... maybe 200-300 off ebay
23:44.19websaemaybe i'll get another dsl connection
23:44.35justinuget a T1, and a router w/ good QoS policies
23:44.36harryvvStrom_C, did you recomend to him a seperate IP address and dedicated bandwith to voice traffic?
23:44.36Strom_Cwebsae, are you an employee or just the consultant?
23:45.21Strom_Charryvv, no, I'm recommending that he get a PRI for the voice traffic
23:45.30harryvvyea good idea.
23:45.38justinuthat's what I sell my customers
23:45.44websaepart owner
23:45.50harryvvpri her cost 600-1200 dollars per month.
23:46.11websaejustinu: what do you sell your customers?
23:46.18justinuPRIs and SIP phones
23:46.22Strom_Cwebsae, listen to me.  Don't be a cheapskate.  You do not want to sacrifice reliability for a few hundred dollars a month.
23:46.56harryvvyes, realability issues will sink you in no time. no one wants poor voice connection.
23:46.59*** join/#asterisk voipuser_au (n=voipuser@static-114.241.240.220.dsl.comindico.com.au)
23:47.06blitzrageanyone else seem to be able to get to ftp.digium.com?
23:47.12blitzragenevermind -- I'll just use wget and the http link
23:47.32harryvvim connected
23:47.35harryvvblitzrage
23:47.53*** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net)
23:47.57blitzragewierd... couldn't get anything from the ftp a few mins ago -- ahhh well :)
23:48.09harryvvgo for a walk
23:48.10harryvv;)
23:48.40harryvvWe are experaincing some of our nicest weather. One month ago it was a rain record. 40 days of rain almost in a row.
23:49.23tuxinator_linuxblitzrage is going crazy
23:49.32*** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe)
23:49.36Flautojustinu, did not work
23:50.04harryvvBTW, shaw digital telephone and rogers telephone are competing in this market. Its a little discouraging to sell to people when thay say thay may want to go with them. Canadians overall are conservative and go with name brands.
23:50.25*** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc)
23:50.55blitzragetuxinator_linux: going?
23:51.10blitzrageharryvv: agreed
23:51.17brc_~seen Corydon-w
23:51.30jbotcorydon-w is currently on #asterisk, last said: 'Damn fork bombs... ;-)'.
23:51.30brc_~seen Corydon76-home
23:51.31jbotcorydon76-home is currently on #asterisk (7h 38m 19s), last said: 'for unlimited data, including SMS'.
23:51.54brc_hey Corydon-w you around?
23:53.47*** join/#asterisk jasonpr2 (n=jason@64.78.192.164)
23:55.29jasonpr2anyone know of any good tutorials on setting up Manager?
23:55.47Strom_Cjasonpr2, yes, I've got the world's simplest tutorial
23:55.48harryvvblitzrage outside of digium, is there any asterisk support networks? say a company that does tier I-III support?
23:55.53Strom_C1. Delete manager
23:56.04Strom_C2. Learn how to operate config files :)
23:56.23harryvvcompany would ask...who is going to support our gear..I would. thay ask, what if your killed..then I did not have a good answer.
23:56.24jasonpr2I need to do a lot of dynamic stuff
23:56.43Strom_Cerm...by manager, you mean AMP, right?
23:57.13jasonpr2?? I'm a little new to asterisk.  I just want access to the socket
23:57.23Strom_Cnever mind then - I was wrong
23:57.28jasonpr2I looked at AMP it was pretty ugly
23:57.50jasonpr2way to many dependacies
23:58.05Strom_Charryvv, is there a directory somewhere of voip / telecom consultants?
23:58.12Kattyhihi
23:58.28blitzrageharryvv: how do you mean?
23:58.30harryvvStrom_C not that i know of. I think most are independent.
23:58.32blitzrageStrom_C: there is a listing on the wiki
23:58.40Strom_Cblitzrage, link?
23:59.15blitzrageStrom_C: www.voip-info.org
23:59.18blitzragenot sure where it is on the site
23:59.22blitzrageshould be easy to find though
23:59.35Strom_Ci know the wiki, but I didn't see a page along those lines, which is why I asked
23:59.41harryvvblitzrage say a support company that has several asterisk consultant working under one roof and at a rate the customer can afford. no way in hell is a small company going to pay 175 per hour for support..thay will go some where else.

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