irclog2html for #asterisk on 20060111

00:02.15*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167048221.nb.aliant.net)
00:03.10*** join/#asterisk kippi1 (n=kippi@cpc3-hatf3-6-0-cust42.lutn.cable.ntl.com)
00:03.15kippi1hey
00:04.29*** join/#asterisk Soul (n=Soul@87-196-8-134.net.novis.pt)
00:04.29*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
00:04.40kippi1can anyone help me install chan_zap.so ?
00:05.25*** join/#asterisk Tili (i=Tili@202-133-67-33-dialup.sat.net.pk)
00:05.44*** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
00:05.46kippi1as i am getting this error http://pastebin.com/500088 so I guess its because chan_zap is not installed?
00:06.13*** join/#asterisk justnulling (i=justnull@ool-18bab443.dyn.optonline.net)
00:07.12sivanaManxPower: ping
00:08.59sivanakippi1: from the CLI> load chan_zap.so
00:09.12kippi1It just quits
00:09.16*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-109.nas28.salt-lake-city1.ut.us.da.qwest.net)
00:09.18sivanacheck your messages
00:09.29*** join/#asterisk Dovid^Away^On^V (i=dovi5988@206.sub-70-192-209.myvzw.com)
00:09.38sivanado you have some sort of PCI card?
00:09.48kippi1sivana: yeah a TE110P
00:09.54sivanadid you install zaptel?
00:10.36kippi1sivana: yeah and i do modprobe wcte11xp
00:10.42sivanano errors?
00:11.10kippi1do i need to look in the error file for errors? or would it just popup?
00:11.21sivanadid you modprobe zaptel as well?
00:11.29sivanait would list errors in /var/log/messages probably
00:11.50sivanaor /var/log/asterisk/messages
00:13.31*** join/#asterisk ZeMMaD (n=ZeMMaD@208.0.224.33)
00:14.06ZeMMaDcould some provide help with installing DigitNetworks X100P Clone Card with already installed asterisk??
00:14.35sivanacheck the wiki
00:14.37*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
00:14.38*** mode/#asterisk [+o twisted[asteria]] by ChanServ
00:14.50*** join/#asterisk cpm (n=Chip@68-66-23-191.chvlva.adelphia.net)
00:15.01moralec
00:15.24ZeMMaDwhich wiki
00:15.35sivana~docs
00:15.38jbotdocs is probably 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
00:15.46sivanahttp://www.voip-info.org/wiki-Asterisk
00:15.54*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
00:15.55*** join/#asterisk techie (n=gus@antibala.com)
00:16.05ZeMMaDi actually have the Clone from DigitNetworks.com
00:16.28sivanaya, it has instructions on how to install the X100P
00:16.42*** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167048221.nb.aliant.net)
00:16.50ZeMMaDthe thing is i have the instructions
00:16.59sivanawhat's the problem?
00:17.01ZeMMaDbut when i do a modprobe nothing happens
00:17.06sivanaok
00:17.09kippi1sivana: I only error I seem to be getting is http://pastebin.com/500097
00:17.34sivanakippi1: you working with a PRI?
00:17.42kippi1sivana: yep
00:17.50sivanapaste your zaptel.conf
00:17.54sivanain pastebin
00:18.13kippi1if I ring the number I just get a NU tome
00:18.17sivanaand zapata.conf
00:19.04ZeMMaD#
00:19.04ZeMMaD# Zaptel Configuration File
00:19.04ZeMMaD#
00:19.04ZeMMaD# This file is parsed by the Zaptel Configurator, ztcfg
00:19.04ZeMMaD#
00:19.09ZeMMaD# First come the span definitions, in the format
00:19.11ZeMMaD# span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow]
00:19.13sivananot here
00:19.13ZeMMaD#
00:19.15ZeMMaD# All T1/E1 spans generate a clock signal on their transmit side. The
00:19.17ZeMMaD# <timing source> parameter determines whether the clock signal from the far
00:19.19ZeMMaD# end of the T1/E1 is used as the master source of clock timing. If it is, our
00:19.21ZeMMaD# own clock will synchronise to it. T1/E1's connected directly or indirectly to
00:19.23ZeMMaD# a PSTN provider (telco) should generally be the first choice to sync to. The
00:19.25ZeMMaD# PSTN will never be a slave to you. You must be a slave to it.
00:19.27ZeMMaD#
00:19.29ZeMMaD# Choose 1 to make the equipment at the far end of the E1/T1 link the preferred
00:19.29NDTjesus...heh was just looking at DigitNetworks.com...they want like $400 more for a 405 card then I paid christ
00:19.31ZeMMaD# source of the master clock. Choose 2 to make it the second choice for the master
00:19.33ZeMMaD# clock, if the first choice port fails (the far end dies, a cable breaks, or
00:19.35ZeMMaD# whatever). Choose 3 to make a port the third choice, and so on. If you have, say,
00:19.37ZeMMaD# 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each
00:19.41ZeMMaD# port should be different.
00:19.43ZeMMaD#
00:19.45ZeMMaD# If you choose 0, the port will never be used as a source of timing. This is
00:19.46sivanaZeMMaD:
00:19.47ZeMMaD# appropriate when you know the far end should always be a slave to you. If the
00:19.48sivanadufus
00:19.49ZeMMaD# port is connected to a channel bank, for example, you should always be its
00:19.51ZeMMaD# master. Any number of ports can be marked as 0.
00:19.53ZeMMaD#
00:19.55ZeMMaD# Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed
00:19.56NDTwhat the hell?
00:19.57ZeMMaD# faxes, unreliable modem operation, and is a general all round bad thing.
00:19.59ZeMMaD#
00:20.01ZeMMaD# The line build-out (or LBO) is an integer, from the following table:
00:20.03ZeMMaD# 0: 0 db (CSU) / 0-133 feet (DSX-1)
00:20.05ZeMMaD# 1: 133-266 feet (DSX-1)
00:20.07ZeMMaD# 2: 266-399 feet (DSX-1)
00:20.09wunderkin~kick ZeMMaD
00:20.11jbotbugger off sod!
00:20.11NDTSTOP
00:20.11ZeMMaD# 3: 399-533 feet (DSX-1)
00:20.13ZeMMaD# 4: 533-655 feet (DSX-1)
00:20.15ZeMMaD# 5: -7.5db (CSU)
00:20.17ZeMMaD# 6: -15db (CSU)
00:20.19ZeMMaD# 7: -22.5db (CSU)
00:20.21ZeMMaD#
00:20.23ZeMMaD# The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
00:20.25ZeMMaD#
00:20.27ZeMMaD# Note: "d4" could be referred to as "sf" or "superframe"
00:20.29NDTomfg
00:20.29ZeMMaD#
00:20.31ZeMMaD# The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
00:20.33ZeMMaD#
00:20.35ZeMMaD# E1's may
00:20.35sivana~pastebin
00:20.37jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
00:20.55rob0hahaha
00:21.08kippi1http://pastebin.com/500104 and http://pastebin.com/500105 I have copied these from another server where it is working with the same lines
00:21.27rob0How To Lose Friends and Influence People
00:21.33kippi1sivana: see above ^
00:21.40sivanaya
00:21.41sivanalooking
00:22.16kippi1sivana: at one point I was getting a busy tone
00:22.17*** join/#asterisk ZeMMaD (n=ZeMMaD@208.0.224.33)
00:22.53sivanawhy do you have ccs in your span in zaptel
00:23.08NDTZeMMaD: Don't sepdn much time on irc huh bud?
00:23.11NDTlol
00:23.13sivanaheh
00:23.16NDTerr spend
00:23.21sivanakippi1: is this a 24 channel PRI
00:23.28kippi1nope 8 channel
00:23.41sivana8 + D?
00:23.48wunderkinomg.. that thread has been going on too long on users
00:23.51NDTexcuse me while I find a digital copy of Moby Dick to paste here
00:23.54ZeMMaDsivana: sorry about that i know realize what u were speaking about paste bin the link is http://pastebin.com/500113
00:24.09kippi1but so is the other box and its working, yeah
00:24.09sivanakippi1: well.. I have no idea about smaller PRIs
00:24.23kippi1hmm
00:24.25sivanakippi1: it's looks like a signalling error
00:24.37sivanaSignalling requested on channel 1 is FXO Loopstart but line is in PRI Signalling signalling
00:24.58kippi1an 10 19:24:03 ERROR[3480]: chan_zap.c:10546 setup_zap: Unknown signalling method 'pri_cpe'
00:24.59sivanasomething's not jiving between zaptel/zapata I suspect
00:25.21sivanakippi1: do you have libpri installed?
00:25.42kippi1ah maybe not, how can I check?
00:25.54sivanaI would just compile/install it
00:26.01sivananot sure how to check :)
00:26.26kippi1ok, i'll try that
00:26.27sivanaZeMMaD: what error you getting with the X100P
00:28.37kippi1trying to find a howto for libpri
00:29.30*** join/#asterisk Cyon (n=cyon@216.179.31.166)
00:29.32CyonHey
00:30.01sivanakippi1: it's part of the SVN
00:30.18CyonHum, question this time, not just idling....lol
00:30.20sivanakippi1: what version of * are you using?
00:30.50sivanaCyon: you asking to ask a question?
00:30.55CyonIf I've got a sipura 2002 sitting at a customers house, and it is registered with asterisk, and I can place calls that read the phone just fine, but outgoing calls get fast busy any time...
00:31.00CyonNo, just saying.  :-P
00:31.13kippi1sivana: cvs version
00:31.23CyonNow I've got other customers all working fine, and my own office working fine; and I took his sip context, moved it over to my outgoing dial-plan with the same problem
00:31.25sivanakippi1: svn checkout http://svn.digium.com/svn/libpri/trunk libpri.
00:31.30sivanawithout the .
00:31.33sivanasvn checkout http://svn.digium.com/svn/libpri/trunk libpri
00:31.41CyonAnd asterisk actually places the call, exactly the same as when I do...but he gets a busy instantly.
00:31.43sivanathen do make install
00:31.56*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
00:31.57CyonThis leads to my question of:  If the sip headers I'm getting from his box include an invalid Contact: entry...
00:32.29CyonSuch as:  <-- SIP read from 70.104.143.102:61517  But I have a header: Contact: aust6-17 <sip:aust6-17@192.168.1.5:5060>
00:32.53CyonHowever nat=yes is set, qualify=yes, and host=dynamic
00:32.59CyonWhat the hell is going on.  :-P
00:35.18Cyonjustinu:  Here?
00:35.26kippi1sivana: then do i need to reboot? or rebuilt zaptel?
00:35.59sivanaI dont' think so
00:36.05sivanazaptel, libpri, asterisk
00:36.11sivanayou might need to rebuild asterisk
00:36.34CyonOh, and the lovely customer swears it was working earlier in the day, doesn't have access to any configs either...
00:36.39sivanakippi1: you should ever need to reboot, it's not windows :)
00:36.55kippi1sivana: ok cool :)
00:37.04kippi1sivana: just rebuilding asterisk
00:37.32sivanakippi1: is that a BRI?
00:37.46Cyonsivana:  I've never seen you before, so I dislike bothering a new person to me, but any thoughts?
00:37.54sivanaCyon: no :)
00:37.57Cyonsivana:  lol
00:38.05Cyonsivana:  nat=yes will instantly mean Contact is ignored, yes?
00:38.18kippi1sivana: nope all PRO
00:38.19*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
00:38.22kippi1PRI
00:38.26CyonQwell!!
00:38.31Qwell[]Cyon?
00:38.39CyonI need you more than I ever have before!
00:38.41Cyonhehe
00:38.42gambolputtyIn AEL, is there a way for an iteration loop to have a label in the loop body?
00:39.17CyonQwell[]:  Nah, really annoying problem that I can't solve...I was hoping for a little advice.
00:39.31CyonQwell[]:  <insert appropriate ass-kissing here>  ;-)
00:39.52kippi1sivana: ok that loaded
00:40.10kippi1yes!!!
00:40.15sivana:)
00:40.16kippi1thank you so much
00:40.30sivananp
00:41.07litagedoes a softphone/ip phone/ata do jitter buffering, or does asterisk do it?
00:41.29blitzrageQwell[]: we have a confirmed hotel for E-Tel now
00:41.34Nuggetin my experience, neither do it and it's a big problem.  :)
00:41.43Cyonlol
00:41.58blitzrageno jitterbuffer in SIP unless you patch it
00:42.16sivanayet
00:42.22blitzrageyet*
00:42.25sivana:)
00:42.27blitzrage:)
00:42.44blitzrageyou could always patch and test it... and *gasp* add a note to the bug tracker
00:43.16blitzrageoff to cook dinner... lates :)
00:43.27Qwell[]blitzrage: So you're going then?
00:43.37Cyonblitzrage:  I'm scared to post to an active bug tracking thingy.  :-P
00:43.44blitzrageQwell[]: high probability... i"m planning on it now I think
00:43.48Qwell[]cool
00:43.56Qwell[]still trying to convince my boss(es) to get me a room
00:43.59blitzrageQwell[]: gotta find a plane ticket though
00:44.03blitzrageQwell[]: yah... do it!!
00:44.21blitzrageQwell[]: just go, and if all else fails, you can crash in our room (it'll be tight... but thats cool, I don't plan on sleeping much)
00:44.32Qwell[]heh, wanna sleep in shifts? :p
00:44.37Cyonlol
00:44.38blitzragemight have to, lol
00:44.43riddleboxcan I have asterisk ask me to enter some digits and have those entered into a script?
00:44.59blitzrageriddlebox: yep... try Read() application
00:45.11CyonQwell:  So, were you waiting for me to actually ask, cause I was waiting for you to say it was ok.  Or just ignoring me, cause if you don't want to bother right now, it's all good.  ;-)
00:45.13Qwell[]well, I'll be home in a bit
00:45.15blitzragethen pass them to AGI or whatever you want
00:45.25Qwell[]Cyon: Actually, no, I'm not of much use right now
00:45.34riddleboxblitzrage:into a python script?
00:45.44CyonQwell:  Ok.  Thanks though.  Later maybe if it's still busted.
00:45.47Qwell[]riddlebox: Sure, however
00:46.16Qwell[]Cyon: I'm sure there are plenty of people in here to help you though
00:46.20riddleboxok, so I just use read()
00:46.36blitzrageriddlebox: sure... you just pass it as an argument to your AGI call
00:47.10blitzrageriddlebox: you could use DISA too... but I think Read() makes more sense
00:47.18riddleboxok cool, one more thing, can I have it save the digits as 23:00:00
00:47.32Qwell[]riddlebox: Make your script do that...
00:47.46Ariel_the agi script should be the one that does it all
00:48.26Ariel_hello folks how is everyone doing tonight?
00:49.02xhelioxAriel1
00:49.04xhelioxAriel!
00:49.11riddleboxok, is there any examples of agi scripting?
00:49.13Ariel_xheliox, how are you ...
00:49.20Ariel_riddlebox, sure are many
00:49.22xhelioxFine, fine.. long time no talk
00:49.25Ariel_look at the agi-bin
00:49.38Ariel_also the wiki-info has lots of info on it
00:49.55riddleboxok thanks
00:49.57CyonQwell:  I tossed it out before you were here, nobody answered/sivana didn't know.
00:50.11Ariel_xheliox, yes I was moving to another city and got layed up with a fall I did when wilma came by here.
00:50.14CyonQwell:  I'll put it together in a pastebin I suppose, and try again
00:50.45xhelioxAriel: Ah, so  you're staying in Miami?
00:50.57Ariel_actually I am now in Homestead
00:51.02Ariel_less traffic
00:57.04*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
01:07.20ZeMMaDwhat would cause the following error? "Signalling requested on channel 1 is FXO Loopstart but line is in FXS Kewlstart signalling"
01:09.00Ariel_wrong settings
01:09.12*** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82)
01:09.49Ariel_ZeMMaD, do you have an red fxo module in channel 1 or a green one?
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01:23.38[av]banihttp://reentry.arc.nasa.gov/viewingforum.html
01:23.39[av]banicoooooool
01:24.19*** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com)
01:24.49masterobiwaANyone that has very good experience configuring VOIP providers and also tweaking Asterisk 1.21 ? i need emergency help!
01:24.57*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:25.47Qwell[]"tweaking" it how?
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01:28.30DrJones1can anyone help me setup sendmail?
01:28.43Qwell[]DrJones1: That's a bit out of scope of this channel...
01:28.57DrJones1im just looking for a simple smtp solution
01:29.00DrJones1i take it sendmail is not simple
01:29.07DrJones1well, in reality i just need a smtp server
01:29.10Qwell[]it can be, but...no
01:29.18DrJones1simple as can be, and i just figured put it on the redhat box
01:29.24dudesIt's not as easy as your mom, heh.
01:29.30DrJones1but if anyone knows where i can find a smtp server, that would be great :)
01:29.32Qwell[]burn
01:29.35DrJones1id only use it once a month or so
01:29.46Qwell[]DrJones1: gmail.com :p
01:29.56jbroomeDrJones1: sendmail is simple like paris hilton is a virgin
01:30.02DrJones1gmail uses pop3 huh
01:30.03dudeshaha
01:30.11DrJones1do you know the gmails smtp server?
01:30.13Qwell[]pop3 to get mail
01:30.22dudesParis Hilton was a Virgin ...
01:30.32Qwell[]DrJones1: It's in the faq, I think
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01:31.16DrJones1i dont think gmail has smtp ax
01:31.18DrJones1access
01:31.30Qwell[]I send mail from thunderbird
01:32.07DrJones1how do you start sendmail
01:32.12DrJones1maybe its already configured
01:32.18DrJones1its secure behind a firewall anyway
01:32.41*** join/#asterisk welles (n=welles@219.145.1.35)
01:32.44inv_ArpDrJones1: #Linuxhelp
01:32.57inv_ArpDrJones1: or maybe even #sendmail
01:33.01Qwell[]inv_Arp: ack, what are you doing?! :p
01:33.28inv_Arp<Qwell[]> DrJones1: That's a bit out of scope of this channel...
01:33.41Qwell[]just don't say efnet... :p
01:34.26dudesWhat's wrong /w efnet
01:34.32inv_Arplol
01:34.34Qwell[]dudes: nothing, in joke
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01:39.19Qwell[]going hom, bbl
01:39.23Qwell[]home that is
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01:40.47*** part/#asterisk CANO-1982 (n=alejandr@201.255.49.68)
01:41.57masterobiwaQwell what time are you available ?
01:42.18CyonFinally, fixed.
01:50.21*** join/#asterisk jihano (n=m@adsl-238-122-192-81.adsl.iam.net.ma)
01:50.32jihanohi
01:50.37Cyonhi
01:51.23jihanoi have 4 lines PSTN when i make a call and try to make other call it say the line busy even if the 3 other are free
01:51.37jihanohow to make asterisk if find line busy try to call other line?
01:52.03CyonUmmm, using what outgoing channel?
01:52.06*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
01:52.07CyonZap?
01:52.33jihanono gateway media
01:52.43jihanonot zap
01:52.56Cyonjihano:  Not a clue.  :-P
01:53.12*** join/#asterisk kiswanto (n=kiswanto@222.124.24.61)
01:53.30Cyonjihano:  if I had to guess, and this is a very uneducated one, either you can have sequential Dial() statements, and if the first is busy it will fall through to the next...but that is if you specify the four lines explicitly.
01:53.33jihano1 moment i show you the gateway and info
01:54.10CyonIf you reference the gateway media as a whole, then that sounds not like asterisk, but the gateway media is not rolling over properly; and as soon as the first line is busy it tells asterisk "busy"
01:54.22*** join/#asterisk quadrata (n=quadrata@ool-182c2aaf.dyn.optonline.net)
01:54.55jihanohttp://www.micronet.info/Products/voip/SP5054.asp
01:55.09quadratahello
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02:02.01jihanoCyon http://pastebin.com/500208 loot at this
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02:02.29NicknaIRCIn the new "X lite" how do you edit the options such as server ip address/login/password/etc.?
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02:13.35AmbroseAnyone know if I can setup extensions.conf so a caller can press a button to ring all available extensions and whoever picks up first gets the call?
02:16.11[TK]D-Fenderyes
02:16.36dogtanianeasy to do if you install amp
02:16.39dogtanian*AMP
02:16.44jebbaAmbrose,   exten =>  123,1,Dial(IAX2/joe&IAX2/fred)        no?
02:16.49*** join/#asterisk KonfuZed (n=KonFuZeD@www.cyberground.com)
02:16.53AmbroseAh I just found queues.conf I figure I gotta send the call to a queue then set it to ring all?
02:16.56KonfuZedyo
02:17.10jebbaAmbrose, ya, you can do it with queues, but that's more complicated.
02:17.13Ambrosejebba: Then I would have to manually list every single extension that is available at the current time?
02:17.31jebbaAmbrose, ya.   If you have something more complicated, then go queues.
02:17.42jebbawith queues/agents people can log in & out.
02:18.01AmbroseI might just manually list all extensions then
02:18.12jebbaLike there is  AgentLogin()    But I don't know why there isn't  AgentLogout()  to match it...
02:18.20KonfuZedtzanger are you in montreal??
02:18.39[TK]D-Fenderjebba : You can log out with the agent login option.
02:18.54Cyonjihano:  As I said, no idea; you'll need someone that knows the grandstream.
02:18.58jebba[TK]D-Fender,   ahh, thx.  I knew I was missing something there ;)
02:20.01KonfuZedpuzzled are you in montreal??
02:20.24[av]banifender -> did you ever buy any gxp-2000's?
02:20.50[TK]D-Fender[av]bani : Nope, and not planning to!
02:21.07[TK]D-Fenderseen one, tried once, heard plenty, keeping away from!
02:21.29[TK]D-FenderKonfuZed : puzzled is n=patrick@puzzled.xs4all.nl * Patrick.  What do you think? :)
02:21.31[av]banigot a couple, you get a suprisingly lot for $80
02:21.38[av]banithat is, they sucked less than i expected
02:21.52[TK]D-Fender[av]bani : A glowing review if ever I heard one!
02:21.54[TK]D-Fender;)
02:22.13[av]baniwell, they have stuff the spa941's dont... like 7 line BLF
02:22.37[av]banidual ether
02:23.00[TK]D-Fender[av]bani : Well current firware has NASTY echo, amongst other problems.  If they were to stabilize they'd be very interesting.  However it still feels cheap....
02:23.01KonfuZedah Netherlands
02:23.13KonfuZedok so Anyone, Anyone at all in Montreal??
02:23.14[av]banilatest firmware fixed echo for us totally
02:23.21[TK]D-FenderKonfuZed : An tzanger is in LA if I recall....
02:23.25[TK]D-FenderKonfuZed : I am.
02:23.25[av]banilooks cheaper than it feels actually :)
02:23.45[av]banithe lighted display is largely a gimmick though, you can't actually _do_ anything with it
02:23.51KonfuZedoh [TK]D-Fender what are you up to at this hour ;^0
02:23.55[av]banispa-941 also sucks in that respect
02:24.02[TK]D-Fender[av]bani : muct be brand new, I checked the WIKI which is kept rather to day on this product.
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02:24.16[TK]D-Fender9:24pm?  Oh yeah.. VERY late....
02:24.25JunK-YGXP-2000, thats looks like my souvenir from astricon ! :)
02:24.49[av]baniif they'd let you program the display that would be perfect
02:24.56[TK]D-Fender[av]bani : The 941's display could do with a bit more thinking, though still better than the GXP :)  GXP's clairty and backlight are nice through.
02:25.11[av]baniBLF is super nice though
02:25.28gammacoderGXP-2000 firmware 1.0.1.13 still has echo issues on handset (speakerphone echo is largely fixed in 1.0.1.12). Grandstream Techs claim a rgeneral elease fixing echo will be out before end of January
02:25.40[av]baniwe didnt have any echo issues with 1.13...
02:25.48[av]bani.9 had echo with speakerphone, fixed in .12
02:26.11[TK]D-Fender[av]bani : Yeah I know.  GXP is a serious contender once the fine points are cleared up.
02:26.17[av]bani.12 had annoying volume level issues, which .13 fixed
02:26.23[av]baniso things are looking good
02:26.44[av]banieg, it forgot your handset volume settings everytime you hung up... :P
02:27.03[TK]D-Fender[av]bani : I *might* get one at one point.  After I hear more updates have stablilized it.
02:27.13gammacoderwith high handset volume in .13 - the user on the other end of the conversation hears themself echo - work around is to simply decrease handset volume on GXP-2000 to 3 bars at most
02:27.14[av]banifender, if they put it in a nicer case and fixed the bugs they could kill 941...
02:27.16[TK]D-FenderBut the screen is still poorly used :)
02:27.45[av]banii think i'll get an ip501 myself for home, but i can live with this
02:28.18[av]banigamma -> 3 bars in .13 is _loud_, cant see anyone wanting it louder
02:28.22[TK]D-Fender[av]bani : the 941 was a refresh.  Linksys quadrupled the lineup last week, and are aiming at small business installs.
02:28.26[av]baniunless you want to use the handset as a speakerphone...
02:28.40[TK]D-Fender[av]bani : Once you get your hands on the IP 501 you won
02:28.47[TK]D-Fender''t want to touch GS again :)
02:29.15gammacoder[av]bani -> me either, but i've got some old users
02:30.08Darwin35Poloycom ip501  web config sucks
02:30.15KonfuZed[TK]D-Fender: can you check your other irc window ;^)
02:30.35*** join/#asterisk scolsuckz (n=scolsuck@202.58.252.15)
02:30.53[av]banifender, ip501 does BLF? (not that you'd need it if you can use xml, but...)
02:31.03[TK]D-FenderKonfuZed : Which other one?
02:31.14KonfuZedi private chatted you
02:31.33KonfuZedso depends what your irc client does I suppose
02:31.42[TK]D-Fender[av]bani : Yes you can do presence on it (for 2 others max), but forget XML : the 501 doesn't have the microbrowser.
02:31.51[TK]D-FenderKonfuZed : You
02:31.57KonfuZedyes
02:32.18[TK]D-Fender're private chat failed on your side.  [av]bani seems to be able to do it just fine along with everyone else :)
02:32.25KonfuZedhm
02:32.27[av]banifender, what can you push to the ip501's display?
02:32.35KonfuZedhow be dat
02:32.36KonfuZed?
02:32.43justinunothing
02:32.50KonfuZedim at cybercafe maybe there is a blocked port but that doesnt really make sense
02:32.55[av]baniheh, the gxp2000 can do 7 ...
02:33.04[TK]D-Fender[av]bani : Can't really push much of anything ATM.  I suppose the 501/601 may become capable, but its not documented or scheduled.
02:33.12[av]baniyou abuse the LEDs of the speed dial buttons for BLF
02:33.52tzangerKonfuZed: no
02:33.53[av]baniso you'd need an aastra or cisco for that...
02:34.01[TK]D-Fender[av]bani : thats exactly what I do.  And its not really "abuse".  Keep in mind I use all IP 60x at my company.  Who the hell needs 6 line keys?! :)  So I go 3/3
02:34.15KonfuZed[TK]D-Fender so I tried again and it says waiting for acknowledgemnet
02:34.21[av]baniwell the boss wants line status so...
02:34.22[TK]D-FenderAnd then there is the attendant modules @ 14 / pre
02:34.34[TK]D-FenderKonfuZed : Your side, not mine.
02:34.38KonfuZedhmmm
02:34.39[TK]D-Fenderjust ask here
02:35.15KonfuZedhm
02:35.18[av]banifender -> its abuse if its a grandstream i think ;)
02:35.26[TK]D-Fender[av]bani : How many ext's?  And keep in mind you can't do LINE status, only EXT status.
02:35.41KonfuZedanyway rather off topic really, I find myself here in Montreal. (arrived sunday and leave tomorrow)
02:35.55[TK]D-FenderKonfuZed : whatever for?
02:36.14justinui'd like to go to montreal
02:36.18KonfuZedI was hopin to get an inside scoop on something more entertaining then my purpose for th etrip
02:36.46KonfuZedI dont expect much for a Tuesday but is there any good bars/clubs open tonight
02:36.55[TK]D-FenderKonfuZed : Its Wednesday... and late.  Check out the Comedy Nest or something.
02:37.08justinuit's tuesday
02:37.09justinuwtf?
02:37.13KonfuZedyeah
02:37.15KonfuZedI thought so
02:37.18tzangerKonfuZed: what are you in montreal for
02:37.22justinulol
02:37.24[TK]D-FenderFine, tuesday!  So I'm ahead of my time!  Sue me!
02:37.28justinulol
02:37.29KonfuZedwell to be plain an honest
02:37.33KonfuZedmy grandfathers funeral
02:37.37tzangerouch
02:37.38tzangerI'm sorry
02:37.38[TK]D-Fender*lawyer
02:37.42[av]banifender -> http://www.jackenhack.com/blog/archives/2005/11/22/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/
02:37.44KonfuZednot a big deal really
02:37.54tzangerI was gonna suggest one of montreal's fine strip clubs but yeah that just does not mix
02:37.55KonfuZedperhaps sadly so,
02:38.08KonfuZedthat was a couple days ago
02:38.25[TK]D-Fender[av]bani : Ok... now what about that link?
02:38.31[av]baniit's interesting they called it asterisk blf in the grandstream firmware
02:38.32KonfuZedI've sent my sisters back to toronto
02:38.40KonfuZedand im no longer staying with familu
02:38.51KonfuZedtime to drink somethin ;^)
02:39.25KonfuZedive already had shwarts deli
02:39.34KonfuZedsadly disapointed
02:39.46[av]banifender, 7 extensions, whatever hints * can send in sip to the phone
02:39.56KonfuZedI was 16 when I worked in the first jewish deli
02:40.02KonfuZedwell first for me anyway
02:40.07[TK]D-Fender[av]bani : Well GS seems to think that their best chance for survival is to assiciate with *.  Not a bad idea actually.  thing is that GS's quality issues and compliances tend to make the big players shun it.
02:40.44KonfuZedno dijon mustard no dark rye and no garlic pickles.
02:40.57[av]baniwell recall linksys was not exactly well regarded till the 941...
02:41.04[TK]D-Fender[av]bani : interesting "cheat" on hint to show line status.... I would love to test that.  have you yet?
02:41.17[av]banigiven the whine about past grandstream stuff, the gxp seems like a huge step up
02:41.19[TK]D-Fender[av]bani : Linksys didn't HAVE an IP phone until the 941 ;)
02:41.26[av]bani841?
02:41.37[av]baniwell
02:41.37[av]banisipura
02:41.39KonfuZed[TK]D-Fender perhaps you know Bal-Room , its across the street from this cybercafe. (virus)
02:41.52[TK]D-Fender[av]bani : A lot of the whining was baout the GXP :)  But Its close to my level of comfort and I might pick one up.
02:42.04[TK]D-FenderKonfuZed : Where?
02:42.10[av]baniwell as long as you dont expect much from $80, it's suprising you get as much as you do
02:42.13[TK]D-Fender[av]bani : Correct, Sipura :)
02:42.15Darwin35grrrrrr
02:42.23Darwin35eveyone go home
02:42.29KonfuZedon St-Laurent
02:42.29[TK]D-Fender[av]bani : yeah, its real tempting I'll admit
02:42.35[av]baniand most of the issues are just firmware so... they can be fixed if grandstream is on the ball
02:42.46[TK]D-FenderKonfuZed : I don't go downtown much.  I'm in the West Island.
02:43.06KonfuZedand Prince Arthur
02:43.20Darwin35opensource is the answer
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02:43.37KonfuZedDarwin35 do you know where I can buy one of those that works
02:43.38[TK]D-Fender[av]bani : Don't get me wrong, I know you've seen me rag on GS plenty, but it will take some turnaround to get me to advise them.  I am open to do so however.
02:43.46Darwin35then you  can choose your protocal
02:44.03[av]baniwell, i dont think theyre up to your exacting standards, but they are a good deal for $80
02:44.19Darwin35not
02:44.33KonfuZed[TK]D-Fender  I was actually in Sorel
02:44.36*** join/#asterisk ManxPowe (i=ewieling@128.sub-70-197-201.myvzw.com)
02:44.38Darwin35the gs gxp  sucks
02:44.42[TK]D-FenderDarwin35 : unfortunately the firmware sucks ass, the phones built around it look and feel like garbage.  GXP has SERIOUS "potential".
02:44.56*** join/#asterisk riddlebox (n=blah@24-171-11-166.dhcp.stls.mo.charter.com)
02:45.27[av]baniit feels like its capable of so much more though. i wish tehy'd opensource it
02:45.29[av]banithen people would go crazy
02:45.42[av]banii mean its not like they make money off the firmware.
02:45.43[TK]D-FenderDarwin35 : Yeah, as nice as protocol selection is, IAX2 was never meant to be a IP phone protocol, pretty much jsut for trunking.
02:45.59*** join/#asterisk jbarbee (n=chatzill@69.219.233.140)
02:46.06Darwin35but sip with speex
02:46.12Darwin35would be nice
02:46.23Darwin35and the pa-1688 is working on it
02:46.30[TK]D-Fender[av]bani : They may not be ABLE to.  Did they license it from someone else?  Could it be that they stand to LOSE something if they did?  Who knows...
02:46.38justinuthat's true. people would flip with an opensource hardphone
02:47.00[av]baniwell look at how many people are hacking the pa1688
02:47.06[TK]D-FenderDarwin35 : I'll wait till they put out a phone that doesn't look like a POS someone ELSE would buy at a drug-store.
02:47.50Darwin35hey I have a the yhw10 and I like the look
02:48.00[av]banifender, too bad polycom lacks the motivation to let you xml up the display
02:48.22welleshi all
02:48.31[TK]D-Fender[av]bani : You mean specifically to "push"?
02:48.40[av]banii wouldnt be suprised if the pa1688 hackers build a full html browser in ;)
02:48.43wellesis any one run  meetme on 64 bit machine?
02:48.46[av]baniyeah, well to abuse in general
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02:48.50annonimoushiya!
02:49.09Darwin35why you can inbed opera
02:49.21[av]banipush would let you do leet line status stuff
02:49.29[av]baniwhich would give the boss a woody
02:49.33[TK]D-Fender[av]bani : I *do* abuse mine :)  I have live queue stats, logo, and more on variable refresh per/phone, and even an inventory lookup screen on mine :)
02:49.54[av]baniwhat refresh?
02:49.54[TK]D-Fender[av]bani : I already havea MB script for full-company presence
02:50.13[TK]D-Fender10s for the ones that get a dynamic feed (queu stats, etc)
02:50.33[TK]D-FenderGood enough for announcement delivery, etc.
02:51.00[av]banii guess unless you have 1000's of phones even 2s wouldnt be an issue
02:52.23*** join/#asterisk Defraz (n=t0tal@103-16.69-92-cpe.cableone.net)
02:52.38[TK]D-Fender[av]bani : depends what you're feeding your phones.  My web server is another box altogether which collects queue stats with 1 lookup and just copies the resuls avross the the multiple phones accessing it.  so a max delay of 20sec for "live" data
02:53.02outtolunc20secs?
02:53.09[TK]D-Fenderactually, more like 18 or so, but close enough
02:53.09outtoluncdamn
02:53.21[TK]D-Fenderfor what they need it for, who cares?
02:53.32outtoluncmust be a shitload of db access in the background
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02:54.01[TK]D-Fenderouttolunc : not DB, AMI <- and only 1 access every 10 seconds.
02:54.29[TK]D-FenderThe webserver does that and generates a STATIC page the OTHERS load.
02:54.40outtoluncfender, note: if you don't *need* QueueMemberStatus info comment them out in app_queue.c
02:54.51outtoluncit *really* makes a difference
02:55.30outtolunc(i say comment out) because as of last month (not sure when) the ability to 'disable them from queues.conf' doesn't work
02:55.32outtolunc)
02:56.11[TK]D-FenderWhat kind of difference?  I have 4 agents.  How much of a differnce would that make?  Doesn't seem to impact my life any
02:56.22litageNugget: 2.5 hours ago when you said "in my experience, neither do it and it's a big problem.  :)" were you referring to this question of mine?:  "does a softphone/ip phone/ata do jitter buffering, or does asterisk do it?"
02:56.53justinueyebeam is a softphone with a jitter buffer
02:56.59justinuand the polycom phones have jitter buffers
02:57.06outtoluncwell in my test site (8 agents, 2 closers), there was so much traffic gen'd (they use agentlogin) that it delayed the crm screens by 10 secs or more)
02:57.39outtoluncin my situation, the crm's have a manager thread
02:57.47litagejustinu: so jitter buffers are implemented client- rather than server-side?
02:57.47outtoluncso they listen for events
02:57.50*** join/#asterisk bjohnson (n=bjohnson@i216-58-90-3.cybersurf.com)
02:58.43outtoluncand, got overwhelmed by those QueueMemberStatus messages, commented out, and back to sub-1 sec reaction times
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02:59.39outtoluncnote: the server is a dell 2800 with a single dual proc, and mysql backend
02:59.47Qwellouttolunc: just don't subscribe to events...
02:59.54outtoluncer single DC proc
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03:00.26outtoluncqwell, it's based one them, so that's not an option.. ok?
03:00.31justinulitage: it's not that simple
03:00.32outtoluncer on
03:00.45justinulitage: basically, you want the jitter buffer on the thing that converts from packetized data to an audio stream
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03:01.21[TK]D-Fenderouttolunc : I'm not using AMI "messages" :) I'm parsing CLI stuff by brute-force.  Much lighter on the scripting side.
03:01.41[TK]D-Fenderouttolunc : and at the same time check my 2 queue's VM boxs and add the count to the line.
03:01.46BrUcE_RmZhello
03:01.56outtoluncwell mine are also using my patch for sendevent so they are interactive
03:02.08outtoluncso to each his own
03:02.20[TK]D-Fenderouttolunc : how are you delivering your stats to the user?
03:02.20wellesouttolunc: do u ever run meetme on 64bit machine?
03:02.37outtoluncwelles, no i haven't, what issue are you having
03:03.04*** join/#asterisk SwK (n=SwK@12-219-147-107.client.mchsi.com)
03:03.10outtolunc(that is assuming you mean a 64bit kernel)
03:03.37wellesouttolunc: i run meetme for conferenc and i use ilbc as the codec ,it sounds well .but when i run it on 64bit machine the voice become very bad
03:03.46litagejustinu: wouldn't the "device" that converts between packetized data and an audio stream be the softphone/ata/ip phone?
03:03.56outtoluncand what timing source are you using?
03:04.12wellesboth ztdummy
03:04.15KonfuZedtzanger are you in montreal too??
03:04.15outtoluncwell
03:04.21outtoluncyou get what you pay for
03:04.26outtolunc<G>
03:05.05outtoluncyou would be better off shoving a lowend card to time off of
03:05.09riddleboxhas anyone written a AGI script in python?
03:05.22wellesouttolunc: what 's wrong? is there any special configure on 64 machine?
03:05.39KonfuZed[TK]D-Fender ya know I can travel like to another part of town where there might be  a good club if you know one
03:05.41[TK]D-Fenderouttolunc : Just to make sure we're comparing relative apples-to-apples, how are you feeding this data to the user?
03:05.43outtoluncwell you could compile zaptel as i586 as a test
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03:06.01KonfuZedwhat aout the Vatican, ever go there?
03:06.12wellesouttolunc: how to do? can u say in detail
03:06.16outtoluncfender: from a cgi lookup off a java app
03:06.37[TK]D-FenderKonfuZed : Nothing much I can sugget really.  I don't club or anything.  I go out for live music, but thats only locally, and to play pool.  Most of that avail somewhere around where you are.
03:06.40outtoluncwelles, look in /usr/src/zaptel/Makefile for 586
03:06.46wellesok
03:06.48tzangerKonfuZed: no I'm in Ontario
03:06.52KonfuZedah
03:07.01litagetzanger: Toronto?
03:07.12[TK]D-Fenderouttolunc : Ah, thats the difference.  Mine appears every 10 seconds ON THE PHONE'S SCREEN.
03:07.13KonfuZedwell, i'll just check out whats around then
03:07.14wellesouttolunc:thanks
03:07.34tzangerlitage: about 90min from
03:07.47[TK]D-Fendertzanger : Can never remember the name of your town....
03:08.47litagetzanger: that must be ~Hamilton if west, or...ack i can't remember what's east of Toronto  =P
03:08.58tzanger[TK]D-Fender: Listowel
03:11.02[TK]D-Fendersounds like just west of nowhere :)  but pop of +/- 30,000 isn't too bad.
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03:13.54tzangerexten => s,1,NoOp(BEFORE: CallerID is "${CALLERID(number)}")
03:14.11tzanger<PROTECTED>
03:14.26tzangerok, so why the hell is GotoIf returning 1?
03:14.43tzangerexten => s,n,GotoIf($[${LEN(CALLERID(number))} != 0]?s,gotcid)
03:15.10Qwelltzanger: $[${LEN(${CALLERID(number)})} != 0]
03:15.24tzangerQwell: dammit
03:15.26tzangerthank you
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03:15.49tzangerI love asterisk's lack of syntax checking
03:15.50tzangerhonestly I do
03:16.02QwellCALLERID(number) is a string
03:16.10QwellIt doesn't/can't know that you meant the CALLERID function
03:17.16[TK]D-Fendertzanger : that should be CALLERID(num), not number
03:17.25Qwellboth work
03:18.06[TK]D-FenderQwell : I've done the exact same kind of NoOp on my side and its worked...
03:18.18tzangerugh
03:18.36tzangergoddamnint asterlink fix your SIP boxes
03:19.12[TK]D-Fenderexten => 5144264825,2,NoOp(CallerID "${CALLERID(name)}" <${CALLERID(number)}>)
03:19.22[TK]D-Fenderhmmm, I use number there too!
03:19.23[TK]D-Fenderhmmm
03:22.26NicknaIRCWhy does asterisk always say Unable to open channel of type IAX2
03:22.43NicknaIRCI'm using Idefisk and it is making calls fine I can't answer DID calls on it
03:22.44QwellNicknaIRC: I don't know, what does the rest of the error say?
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03:24.12NicknaIRCI don't know let me re do it and get it again
03:27.55xachenFILES a bug
03:28.09xachenwhats asterlink doing tonight?
03:29.37riddleboxdoes anyone program AGI in python?
03:31.14blitzrageriddlebox: what are you looking for?
03:31.34riddleboxI am not sure I am doing this right
03:32.10riddleboxI thought I had everything setup but when I call the exten for the script it doesnt work
03:34.28riddleboxblitzrage:could you help me?
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03:38.34blitzragesorry... I haven't done much in python...
03:38.59blitzragehave you found the documentation on asterisk docs re: python?
03:39.00riddleboxhrmm I guess I need to do more research
03:39.12riddleboxyeah thats what I am going off of :)
03:39.16blitzrageheh :)
03:39.20blitzrageis it helping at all?
03:39.42riddleboxyeah it is, but as usual I am sure I am missing some real small piece :)
03:39.52BrUcE_RmZHey... could somebody help out?: I have a Audiocodes MP104 Fxo as gateway, some Snom IP phones And an Asterisk as proxy server... I ve been having trouble with DTMF.
03:39.52blitzrageyah... thats always the problem eh?
03:40.01riddleboxyeah
03:40.37blitzrageKatty: I heard you might be showing up at E-Tel
03:40.42riddleboxI think I got most of the code written but I am sure I am missing one small thing
03:41.00wunderkinkatty in public? where do i sign up?
03:41.10blitzragewww.oreilly.com
03:41.11blitzrage:)
03:41.39riddleboxyeah what I need is a book for doing AGI scripting with python lol
03:41.47blitzrageheh :)
03:42.14*** join/#asterisk Tili (i=Tili@202-133-67-82-dialup.sat.net.pk)
03:44.08riddleboxblitzrage:at least I see the script is being started, it's my script that seems to be the problem lol
03:44.32blitzragehrmmm.... take it right back to basics and just try to get something very simple to work, then built from that
03:44.46blitzragethats what I usually do
03:44.55riddleboxyeah I will have to
03:44.55blitzrageI'm not a great programmer... just a hacker
03:45.21riddleboxthats all I am, I am trying to get this to ask me a few questions and then use the answers to setup a mythtv recording
03:45.31*** join/#asterisk file (n=joshnet@mctnnbsa24w-142167048221.nb.aliant.net)
03:45.40blitzragewell, sounds neat at least :)
03:46.02riddleboxonly if it work
03:46.07riddleboxwork/works
03:46.16blitzrageyou'll get it to work I'm sure
03:46.28*** join/#asterisk viLeR (i=1000@66.128.47.232)
03:46.31fileblitzrage: how did your adventure go?
03:46.35riddleboxthis is my first night at doing the AGI stuff so I am not expecting it to work right away
03:50.08*** join/#asterisk bmg505 (n=leon@c1-29-12.rndf.isadsl.co.za)
03:51.17riddleboxblitzrage:http://home.cogeco.ca/~camstuff/agi.html
03:51.32riddleboxthere is a python section, can you take a look at it for a sec?
03:53.19[TK]D-FenderBrUcE_RmZ : Do you snom's DTMF work fine in * apps like voicemail, etc?
03:54.18justinui never had problems with my snom and *
03:54.47BrUcE_RmZok
03:54.53BrUcE_RmZlook
03:55.20BrUcE_RmZIm using g7231
03:55.35BrUcE_RmZthe phones work fine between them...
03:55.43dudesBut once you try and trancode
03:55.48dudesit's dead air?>
03:56.01justinuyou need g723 licenses
03:56.02BrUcE_RmZbut when try to operate with the AC..
03:56.07Qwellg723 licenses?
03:56.18justinui thought you could buy them
03:56.19QwellI thought * didn't support g723, except in passthrough?
03:56.23BrUcE_RmZFor the DTMF?
03:56.28BrUcE_RmZQwell
03:56.28justinui have a g723 codec for * ;)
03:56.35dudesyou can download the codec
03:56.37BrUcE_RmZive installed a codec already
03:56.37Qwelljustinu: a legal one? ;]
03:56.39dudesif you can find it
03:56.51justinuqwell it's in a country without patents ;)
03:57.23BrUcE_RmZOk... look... the DTMF some times work and other do no, when calling the AC
03:57.52BrUcE_RmZI mean... call the AC and it gives me the tone...
03:58.08BrUcE_RmZbut when trying to call some number from outside...
03:58.32justinuwhat's "ac"?
03:58.32BrUcE_RmZ... some times it works and other times does not
03:58.37BrUcE_RmZsorry
03:58.39[TK]D-FenderBrUcE_RmZ : Are your phones local to the rest of the PBX?
03:58.41BrUcE_RmZAudiocodes
03:58.44justinuoh
03:58.52justinuoh, you need to enable rfc2833
03:59.01justinuyou can't run dtmf inband on g723
03:59.08justinuit'll work... sometimes
03:59.13BrUcE_RmZOk...
03:59.39BrUcE_RmZD-fe... Yes
03:59.42*** join/#asterisk jacoyle (n=chatzill@208.4.153.208)
04:00.04BrUcE_RmZOk...
04:00.45BrUcE_RmZHas somebody ever config the Audiocodes MP FXO??
04:00.58blitzragefile: got distracted again... dinner and such
04:01.05blitzragefile: maybe tomorrow when I'm more away
04:01.08BrUcE_RmZIve tried to enable the RFC
04:01.13[TK]D-FenderBrUcE_RmZ : You should be running G.711u across the board then, and either rfc2833 or INFO for DTMF (rfc preferred)
04:01.40BrUcE_RmZOk...
04:01.59BrUcE_RmZSo... Locally I shoud use the G711?
04:02.34justinuthen you're transcoding
04:02.42[TK]D-FenderBrUcE_RmZ : yes
04:02.47justinuif you need g723 for pstn
04:03.07BrUcE_RmZJustinu: yes
04:03.13[TK]D-Fenderjustinu : its an AudioCodes... I'm sure it support G.711
04:03.26BrUcE_RmZActually... the quality is not good
04:03.46BrUcE_RmZJustinu:Yes it does
04:03.51justinuyeah, switch to 711 then
04:03.59BrUcE_RmZOk...
04:04.11justinuare phones connected via ethernet?
04:04.12BrUcE_RmZAnd what you think its better?
04:04.26justinug711 > g723
04:04.26BrUcE_RmZIn-band or RFC?
04:04.31justinuRFC
04:05.06BrUcE_RmZjustinu: Ok... yes... the phones are conected thru ethernet
04:05.15justinuok, then g711 should work well
04:05.37BrUcE_RmZOk... Ill try it now... Ill be back. Thx
04:05.42justinugood luck
04:05.44[TK]D-FenderG.711 + RFC = good
04:06.19BrUcE_RmZok
04:06.24BrUcE_RmZother one...
04:06.34BrUcE_RmZa-law or u-law?
04:06.44Qwelluse ulaw if you're in the US
04:07.01BrUcE_RmZIm in mexico
04:07.07jacoyleulaw
04:07.10Qwellyeah
04:07.14BrUcE_RmZok...
04:07.18BrUcE_RmZthx a lot
04:07.21[TK]D-Fenderulaw
04:08.07*** join/#asterisk mdawson (n=chatzill@cpe-65-26-178-240.indy.res.rr.com)
04:09.41riddleboxcan someone help me with this AGI script? http://pastebin.com/500325
04:09.42*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
04:11.20[av]banitoo bad hardly anyone supports speex
04:11.50[av]baniwould be nice to have something better than g711u for at least internal phones
04:12.04[TK]D-Fender[av]bani : whats the strong point of speex?
04:12.13justinuit's open source
04:12.17Qwell[TK]D-Fender: non 8khz, I think
04:12.21justinuti's bassed on ogg vorbis
04:12.27[av]baniyes, non 8khz and non 8bit
04:12.32[TK]D-Fender[av]bani : Well.... the GS's aren't exactly renowned for their audio quality :)  Poly/Cisco sure...
04:12.41[av]baniwell, you can only do so much with 8khz
04:13.02[av]banino matter what you do, it's going to be weedy
04:13.13[TK]D-Fender[av]bani : just think of the constant transcoding though... everything except direct calls would place a load on your server.
04:13.19jacoyleHey- I love open source but speex needs a lot of work before I will use it
04:13.22[av]banihence "internal phones"
04:13.31justinuso what's wrong with speex at this point?
04:13.45[av]banihardly anyone supports it, thats whats wrong :/
04:14.03jacoyleit sounds like ilbc
04:14.04[TK]D-Fender[av]bani : The  point is internal calls aren't the problem, its the TRANSCODING for everything else.  Not the bandwidth even, to processor load.
04:14.13justinumaybe it's just the rate you tried
04:14.14[av]banithere wouldnt be any transcoding...
04:14.19justinuhow about 16000 speex?
04:14.23[av]baniinternal calls, just routing packets
04:14.53jacoylehavent tried 16000
04:15.15[TK]D-Fender[av]bani : again, internal calls don't get transcoded if the codecs match regardless.  so the load is created playing back any default GSM * recording, or any time to terminate to anything at all .
04:15.30[TK]D-Fender(PSTN, etc)
04:16.58[av]baniwell, speex also isnt exactly a cpu killer
04:17.17justinuthe speex codecs on asterisk are the highest cost ones
04:17.47mogormanyup
04:17.54mogormanspeex is like shitty 729
04:17.58mogormanbut its free
04:18.06[TK]D-FenderHmmm new intel iMacs : The first Apple Intel machine will be the iMac: it features the same design as the latest iMac G5, including Front Row, iSight camera, the same screensizes, and the same price-- however, 2 to 3 times faster, using an Intel Duo processor, shipping today;
04:18.24[TK]D-FenderSo much for G5 = superior.....
04:18.42Qwell[TK]D-Fender: it was superior
04:18.46QwellTo their bottom line
04:18.47Qwellnow intel is
04:18.56[TK]D-FenderI'd use this to beat in my old marketing managers head who got us loaded up on Mac's :|
04:19.00*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
04:19.03[av]banippc was never competetive with x86...
04:19.05[TK]D-FenderQwell : well put :)
04:19.14[av]baniand talk about a brandamaged architecture...
04:19.17*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)
04:19.18filedamn you Telus, give me my bill
04:19.20mogormanlol
04:19.39justinuwhat was so wrong with ppc arch?
04:19.41mogormanit might have been better proccessor but no one ever wrote anything big optimized for it
04:19.45mogormanits nifty though
04:19.58[TK]D-FenderI'm mixed on OS-X as well though.  Half love, half hate...
04:20.07[av]baniwell, it had no OOE for a long time, lagged behind in FSB architecture while everyone else flew by
04:20.20alephcomfile:  Why would you want a bill from Telus?  Their bills can be nasty.
04:20.36[av]banialso, the load/store turned out to not be such a big win as they predicted
04:20.43filealephcom: I want to see if they charged me for data service
04:20.48[av]banippc assumes ram wouldn't get much faster
04:20.50[av]banibut it did
04:20.53[av]baniso they lost
04:21.12[av]banippc made a bunch of assumptions about PC architecture which turned out to not be true
04:21.15alephcomuh,huh, ok
04:21.30*** join/#asterisk Entegrity (n=Entegrit@c-65-96-118-47.hsd1.ma.comcast.net)
04:21.31justinuavbani: interesting
04:21.32[TK]D-FenderOh wow, and the laptops now! : http://www.apple.com/macbookpro/
04:21.55[av]banithey also thought they would be able to scale to much higher mhz than they did
04:22.14[av]banitbh, intel assumed the same with some of their designs
04:22.20[av]banibut ppc assumed it across the board
04:22.25jacoylelaptops - for ordering only -- not  shipping yet
04:22.32mogormanbooo
04:22.40mogormanso who wants to get me a new laptop?
04:22.55*** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au)
04:23.24[av]banithe large register file of ppc turned out to not be such a big advantage (and on sparc, it actually turned into a liability)
04:24.24[av]banithe fixed instruction size of ppc ended up being another non advantage
04:25.20[av]baniintel took all the major mistakes of ppc and amplified them 100x with itanium though ;)
04:25.34mogormanyeah that was just hillarious
04:25.46justinuheh
04:25.50mogormanwe had an sgi box with like 8 of them
04:25.54[av]baniitanium assumed compilers would get much better -- they didnt
04:26.02mogormanand it did as much g729 as my box
04:26.05[av]baniitanium assumed memory wouldnt get much faster -- it did
04:26.10mogormanthat i worked off of
04:27.09[av]baniintel got so distracted with ia64 they got totally pwned by x86_64 (didnt help that they were pooh-poohing it all along)
04:27.11riddleboxdo you need anythin special installed for python to work with asterisk?
04:27.34inv_Arpriddlebox: python
04:27.46[av]baniitanium also had pisspoor backwards compat, at least ppc had that
04:27.50[av]banihuge mistake imo
04:28.07*** join/#asterisk brockj49464 (n=brockj49@63.87.56.159)
04:28.08*** part/#asterisk santiago (n=santiago@208.195.215.97)
04:28.12mogormanpyst is all you need
04:28.31justinui wonder if the people who pushed the itanium project are still working at intel
04:29.11[av]baniprobably, ia64 is the kind of project you get by pure academia
04:29.28[av]banidriven by pie in the sky idealism, not grounded reality
04:30.37riddlebox[av]bani, I am trying to run a python script and it never works?
04:30.42[av]banior rather, what you'd get if euro-demo-coders designed a cpu
04:30.52[av]banibeautiful design, but utterly useless ;)
04:31.20riddlebox[av]bani, do you have a minute to look at it? http://pastebin.com/500325
04:32.24[av]banisorry, i'm not a python person
04:32.47justinueuro-demo-coders, lol
04:33.04riddleboxok, well I got it from http://home.cogeco.ca/~camstuff/agi.html
04:33.07*** join/#asterisk Math` (n=Math_@modemcable148.4-81-70.mc.videotron.ca)
04:33.10Corydon76-homePretty much any language or CPU is originally like that, though
04:33.18*** part/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:33.23Corydon76-homeProgrammers just learn new tricks
04:33.58jacoyleThis is cool and probably illegal in the US-   http://www.dailyphreak.com/?p=28
04:34.30NicknaIRCIn asterisk how do you make it not go to the -- Hungup 'IAX2/voicepulse-in-0   command
04:34.35NicknaIRCto suspend the call without hanging up
04:36.09*** join/#asterisk linlin (i=linlin@c-67-184-231-233.hsd1.il.comcast.net)
04:36.42*** join/#asterisk xachen (i=justin@magnum.thisgeek.com)
04:36.45jacoyleDepends on where you want the call to go
04:41.20alephcomWell, who'll be the first to post a project on asteriskhelpdesk.com?  It looks like a good idea to me.  What do you others think?
04:44.37*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)
04:46.00*** join/#asterisk techie (n=gus@antibala.com)
04:46.30*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:46.59NicknaIRCjacoyle it is going out to POTS
04:47.44justinujacoyle: i don't see how it's illegal... it's just a network test script gone haywire. "i'm sorry, it'll never happen again"
04:51.09*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:51.59*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)
04:52.28mogormanhttp://bugs.digium.com/view.php?id=6082
04:52.34mogormanoops
04:52.36mogormanwrong channel
04:52.51tuxinator_linuxmogorman, shame on you
04:53.10mogormanhey we are at 243 bugs
04:53.17mogormanneed to keep em movin
04:54.03*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
04:55.28*** mode/#asterisk [+o mogorman] by kram
04:55.35Qwellw00t
04:55.36filekram: awww how nice
04:55.37*** part/#asterisk ecronin (n=root@widget.gizmolabs.org)
04:55.46mogormanyay!
04:55.53mogormani have the set n0ow
04:56.21drumkillakram: you're my hero, just fyi
04:56.52*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
04:56.56mogormanhey drumkilla can you put me in chanserv as marko didnt :(
04:57.03drumkillano
04:57.11mogormanboo....
04:57.37Corydon76-homeI already asked... ;-)
04:58.08mogorman?
04:58.19Corydon76-homeIf I could get ops on #asterisk
04:58.26Corydon76-homeThe answer was no...
04:58.30mogormanlol
04:58.44mogormanitd be like the redbull wars
04:58.51jacoylejustinu - Good theory
04:59.01Corydon76-homeOh, good lord, is that still going?
04:59.07mogormanno its over
04:59.09mogormanno one has ops
04:59.16Corydon76-homelol
04:59.27h3xat least 10 of the people in here has a mdeop script running probably
04:59.27h3xheh
04:59.40Corydon76-homeDid that get started at IP4IT?
04:59.48mogormani cant remeber
04:59.49trixterh3x: 11 I just loaded one in case
04:59.50trixter:P
04:59.50mogormanit was fun though
05:00.04sivanaisis this a "catch-all"?  exten => _.,1,Congestion()
05:00.05Corydon76-hometwisted bought a boatload of RB
05:00.33Corydon76-homesivana: yes; it's also ill-advised
05:00.41sivanawhat's recommended then
05:00.43mogormanyeah we have tons of it at office
05:00.45Corydon76-home_X.
05:00.48mogormanyou should come by and get some
05:00.48sivanaok
05:00.59Corydon76-homemogorman: it's only 2 hours out of my way
05:01.13mogormani thought you lived in decatur or something?
05:01.18Corydon76-homeNashville
05:01.32mogormanohhhhhh
05:01.43mogormanthat explains why you dont stop by that often....
05:01.45Corydon76-homeI haven't even been to Digium's new office
05:02.40Corydon76-homeI have been to 2 of Digium's previous offices, though
05:02.51mogormanwe are expanding
05:03.01mogormanalmost have eaten all offices on the first floor now
05:03.27Corydon76-homeHeh
05:03.43Corydon76-homeWhere's my office?  :-P
05:03.51mogormancome on down
05:03.57mogormaneveryone doubles up
05:04.05mogormanyou can have a seat in engineering
05:04.24filemogorman: meeeeeeep
05:05.11mogormanwe have space for you too file....
05:05.14mogormanand anyone else
05:05.21mogormanwe are pretty fun place to hang out
05:05.47Corydon76-homeOh, I know...
05:06.13justinudo you get paid anything besides redbull?
05:06.33mogormanwhat is there other than redbull?
05:06.51justinumoney
05:07.03Nivexmoney is evil
05:07.03justinuit's useful for goods AND services
05:07.07mogormanhmm i do not know what this is
05:07.09fileooh services
05:07.12mogormani know i have redbull
05:07.13justinuthat's a problem
05:07.14mogormanand food
05:07.16mogormanand a bed
05:07.23fileyou have a bed?!?
05:07.24mogormanwhat else would i want
05:07.26mogormanor need
05:07.31mogormanyeah its pretty bad ass file
05:07.31fileinternet access :P
05:07.33mogormanyou should see it
05:07.38mogormanwell i obviously have that as well
05:07.40mogormanand power
05:07.44fileare you inviting me into your bed?
05:07.53SkramXAnyone ever used AstLinux?
05:07.55mogormani dont think my lady would like that
05:07.58SkramXis it anygood
05:08.03justinui dunno, i have expensive hobbies
05:08.06justinui need money
05:08.06filenever know, could get some threesome coding going on there
05:08.13mogormanlol
05:08.19mogormanshe hates asterisk though.....
05:08.22Corydon76-homefile: you can come code in my bed...
05:08.28mogormanit is so annoying
05:08.29fileyou touch my /dev I touch yours, that sorta thing
05:08.39mogormanlol
05:08.57fileI crack myself up sometimes
05:09.20rob0So anyway, today my son, 8, was talking about ancient Egyptians. He said they worshipped cats as gods. I said, "Dyslexics worship the dog!"
05:09.34rob0Thank you, I'll be performing here all week.
05:09.41mogormanwow rob0
05:09.51filerob0: really? I'll be sure to not be here then
05:10.06rob0mogorman: in your first official act as op, you can kick me :)
05:10.13mogormanlol you sure
05:10.14mogorman?
05:10.27rob0surely that was worth a kick?
05:10.29*** kick/#asterisk [rob0!n=mogorman@user-24-236-84-48.knology.net] by mogorman (he asked for it.....)
05:10.33mogormanheh
05:10.37mogormanw00t
05:10.40[TK]D-Fenderrob0 : Ever heard about the dyslexic agnostic insomniac?  He sits up all night wondering if there is a Dog :)
05:10.54mogormanaww sorry tk he will be back
05:11.27[TK]D-Fendermogorman : Already you succumb to the dark side of +o
05:11.46mogormanwell i dont think mark put me in chanserv
05:11.47*** join/#asterisk rob0 (i=1007@sorry.no-ip-here.net)
05:11.56mogormanso i only have it till i log out
05:11.57rob0:)
05:11.59[TK]D-Fenderrob0 : Ever heard about the dyslexic agnostic insomniac?  He sits up all night wondering if there is a Dog :)
05:12.04rob0yes!!!
05:14.10sivanawhen you forward a voicemail to another mailbox, shouldn't it delete it from yours?
05:14.35sivanado we have a deleteonfwd flag? :)
05:14.58mogormanno we dont
05:15.03mogormannot that im aware of
05:15.08mogormani dont think it will delete
05:15.12sivanadoes anyone think that would be useful?
05:15.16sivanano, right now it doesn't
05:15.17mogormani do
05:15.24sivanayou have to do 8 then 7
05:15.36sivanadeleteonforward=yes
05:15.53sivanaok.. that's my goal this weekend :)
05:17.15mogormanyay!
05:17.58[TK]D-Fenderdelete on forward would prevent you from forwarding to multiple people....
05:18.07[TK]D-Fenderunless you are planning a BIG patch....
05:18.09sivanaoh
05:18.13sivanaya.. that might suck
05:18.29sivanatoo bad you can't specify more than 1 mailbox while in the "forward" section
05:18.38[TK]D-FenderWhich would gain it a "lose" vote.  frankly for the # of times a person should ahve to forward messages I don't mind the extra step esp if you can do it blind
05:19.03justinufender:  dormez-vous jamais?
05:19.04sivanaI could start by making the forward section accept multiple boxes
05:19.37[TK]D-Fendersivana : pas trop tard encore!  J-m-ens va bien-tot
05:19.55sivanaoui, mois aussi
05:19.56[TK]D-Fendererrr, justinu!
05:20.11JunK-Ytk is a machine!
05:20.41justinuan impressive machine :P
05:20.52[TK]D-FenderYup, a carbon-based yet still mostly water machine :)
05:21.06JunK-Yjustinu: thats what his gf said :P
05:21.16[TK]D-Fender(working on becoming caffeine-based as per NASA specs)
05:21.20sivanaheh
05:21.28justinuJunK-Y: how would you know?
05:21.33fileugh
05:21.36fileI drank pepsi... what a mistake
05:21.48justinudiet pepsi rules
05:21.51JunK-Yjustinu: leave ur imagination here :P
05:21.54JunK-Ycoke rocks.
05:21.57justinuthe regular pepsi is meh
05:22.06fileit's not agreeing with me
05:22.16fileI drink water now all the time, so pepsi is rather... harsh
05:22.22[TK]D-FenderI grew up on diet soda so the "normal" stuff all seems too gloopy for me...
05:22.36justinuyeah, it's like drinking 4 tablespoons of sugar, or something
05:22.50rob0"gloopy" is a Russian word
05:23.07*** part/#asterisk kiswanto (n=kiswanto@222.124.24.61)
05:23.35*** join/#asterisk thrash__ (n=bcrochet@cpe-069-134-063-230.carolina.res.rr.com)
05:24.17thrash__I'm trying to build the zaptel-modules package from zaptel-source on debian sid, and the path for the modules is funky. Anyone else run into this?
05:24.30thrash__Zaptel 1.2.1
05:26.10*** join/#asterisk enemy^x (n=null@85.196.70.98)
05:27.19enemy^xI`m having problems getting hints to work. I get "No application 'hint' for extension" in the logs (running 1.2.1
05:28.03justinui can only get about 1.3mBytes/sec thruput out of asterisk
05:28.08justinukinda disapointing
05:28.38mogorman?
05:28.53justinubefore i max out my hardware
05:29.18mogorman1.3mBytes of what
05:29.27justinuRTP
05:30.05[TK]D-Fenderenemy^x :paste a sample of one of your hint entries....
05:30.28*** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com)
05:31.05justinuif I go much past 1.3mB/sec, asterisk starts to ignore BYEs
05:31.11justinutakes a few retries for them to get thru
05:31.58*** join/#asterisk pengyong (n=lala@222.188.138.144)
05:32.00*** join/#asterisk xtr-II (n=01928375@S0106000c41ed11e1.vf.shawcable.net)
05:32.12dudesthat's not too bad
05:32.18justinuload average is currently at 17
05:32.31dudes17.0?
05:32.32justinuit's behaving properly tho
05:32.40justinuyeah actually 20.0 now
05:32.40dudeswhat's your idle
05:32.46justinu13%
05:32.54dudesthat sounds about right
05:32.58dudesSIP calls?
05:33.02justinuyeah
05:33.15dudeswhat's your b/w up/down
05:33.29justinuit's on 100mbit ethernet
05:33.34justinudoing local lan testing right now
05:33.47dudessimulations
05:33.52justinuright
05:34.11dudeswhat kind of box
05:34.15justinui can get about 80 channels on G720 on this machine
05:34.18justinuit's a dell sc1425
05:34.21justinusingle xeon 3.0
05:34.33justinus/G720/G729/
05:34.44dudes729 to 711
05:35.06justinuright
05:35.26dudesthat's not bad
05:35.37justinuit's stable at this capacity
05:35.56justinuno more tho
05:35.56dudeshow many ulaws
05:36.00justinu160
05:36.27dudesIt should be able to do more than that
05:36.33justinuthat's what I'm thinking
05:37.05dudesCause I've had a Dual Xeon 3Ghz 1GB ram doing 280 SIP to Zaps
05:37.12dudesso almost 600 channels
05:37.22dudesand it could do more
05:37.25justinug711 to zap?
05:37.29dudesyea
05:37.41justinucool
05:37.56justinui think context switches is the bottleneck
05:37.57dudesthere was 5 TE410P's so 20 T1's
05:38.28*** part/#asterisk thrash__ (n=bcrochet@cpe-069-134-063-230.carolina.res.rr.com)
05:38.55justinuonce the system gets near 19,000  cs/sec, load average starts to get very high
05:39.00dudesAll it did was called to a * box from a dialogic dialer to a xo/qwest SIP trunk
05:39.19justinucool
05:39.27mogorman5 te410ps all working?
05:39.28mogormanwow
05:39.37dudesyea all worked in one sever
05:39.37mogormanthats an awesome pc
05:39.40dudeserr server
05:40.01dudesit's cool when * starts and it'd sync up
05:40.39mogormanwow
05:40.42mogormanthats cool
05:41.23*** join/#asterisk techie (i=gus@antibala.com)
05:41.36dudesonly had 450 SIP lines though (and I don't think I ever seen it much over 380ish active bridges).
05:43.36[TK]D-FenderOk,  I'm fried, 'nite all...
05:43.47enemy^x[TK]D-Fender: sorry about that, I figured it out, hint seems to be case sensitive
05:44.24enemy^x[TK]D-Fender: seems to work better now, but it doesnt change any states under "show hints" probably some lame thing I didnt check yet
05:44.31justinua bientot
05:45.57[TK]D-Fendersalut
05:54.46*** join/#asterisk Insanity5 (n=feaw@ip68-111-5-23.sv.om.cox.net)
05:54.57Insanity5What's a good cheap ip phone for home use?
05:55.14Insanity5I'm sick of no ata / features iwth this hacked vonage pap2 :P
05:55.17fileone that works.
05:59.49mogorman~oej
05:59.55mogormanor is it
05:59.55Corydon76-homeGrandstream
05:59.57mogorman~seen oej
06:00.07jbotoej <n=oej@apollo.webway.se> was last seen on IRC in channel #asterisk, 18h 38m 37s ago, saying: 'nfiermes: File a bug report then. A lot has happened since then.'.
06:00.07enemy^xhow can I have various limits for incoming and outgoing? lets say, 1 for incoming to sip and unlimited for outgoing from sip.
06:00.30Corydon76-homeOr any of the PA168-based phones
06:01.48*** join/#asterisk Qorky (n=spam@202.173.160.26)
06:02.29Qorkywhen i do a sip show channels im seeing heaps of calls that arnt hungup properly.
06:02.39Qorkylike 10.0.10.137      00          3c26700a186  00109/00003  unkn  No       Tx: NOTIFY
06:02.59Qorkyit looks to be after transfering from that 00 extension the calls kindof stay up.
06:03.15Qorkyexten => 03,1,Dial(SIP/03,15)
06:03.15Qorkyexten => 03,2,Voicemail(u03)
06:03.15Qorkyexten => 03,3,Voicemail(b03)
06:03.16Qorkyexten => 03,4,Hangup
06:03.16Qorkyexten => 03,hint,SIP/03
06:05.43Qorkyis that the right way to have an extension ?
06:05.56Qorkylike if it gets through shouldnt it handup the call?
06:06.19Qwelltip: voice pings do not work
06:06.33Corydon76-homeWow, you want to get two voicemails?
06:06.55Qorkyisnt one if busy and one if unavailable.. ?
06:07.20Corydon76-homeDrop the if on both, and I'll agree with you
06:07.20mogormanyes but thats not how extension logic works for busy
06:07.37mogormango read the sample doc for example
06:07.39file[laptop]Qwell: did you miss House tonight?
06:07.44Qwellfile[laptop]: yep :)
06:07.45*** join/#asterisk Jzalae (n=sk@dsl-66-63-110-48.gwi.net)
06:07.48*** join/#asterisk kc5cqm_ (n=kc5cqm@cpe-68-206-112-195.stx.res.rr.com)
06:07.52file[laptop]Qwell: I messaged you earlier in the day :P
06:07.54Qwellmeetme > house
06:07.55Corydon76-homefile[laptop]: homeless chick?
06:07.57QwellI saw
06:08.08file[laptop]Corydon76-home: no
06:08.10file[laptop]brand new episode
06:08.34Corydon76-homeOh, House got moved to Thursday for us
06:09.43kc5cqm_quick question:  What can be possibly causing this "unable to create channel of type SIP" error using the verbatim examples from onramp's site?    The funny thing is it was working perfectly...could this possibly be something other than a config issue? I can't even get my 2 local phones to talk to each other.
06:10.15sivanakc5cqm_: is the SIP device registered?
06:10.54kc5cqm_they get dial tones
06:11.37kc5cqm_as far as I can tell, they are.  Is there a CLI command to show registered sip devices?
06:12.31kc5cqm_"sip show channels" shows nothing though
06:12.34*** join/#asterisk alk (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
06:13.09kc5cqm_err...wait a sec...it's on the dta
06:13.23kc5cqm_nevermind...I had twiddled around and disabled the 'send registration request'
06:13.25kc5cqm_I bet that's it
06:14.12*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)
06:15.01kc5cqm_it was it ;-)
06:15.15*** join/#asterisk Pegger (n=peg@pool-68-163-134-103.bos.east.verizon.net)
06:15.28Peggercan asterisk input cdr into mysql yet?
06:15.45*** join/#asterisk shido6 (n=bleh@i216-58-29-215.cybersurf.com)
06:15.59sivanaPegger: look in the addons dir
06:17.36Peggerin my instilation
06:18.23sivanain the source
06:19.53*** join/#asterisk masterobiwa (n=master@201.199.76.194)
06:20.24kc5cqm_is it possible to specify multiple ports in iax.conf?
06:20.51kc5cqm_voicepulse specifies 5036, and fwd I think uses the standard iax2 ports
06:21.00kc5cqm_or can I just stick this in each context
06:21.13shido6kc5cqm_ stop limiting yourself JUST to asterisk
06:21.17shido6what about iptables
06:21.18shido6:)
06:21.21shido6port forwarding
06:21.23shido6yeesh
06:21.30shido6think OUTSIDE of the box :) its linux!
06:21.46kc5cqm_good point...
06:25.06*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)
06:30.44kc5cqm_shido6, so there's no problem with putting port 5036 right onto 4569, mixing iax and iax2?
06:37.25shido6nope
06:37.29shido6iax and iax2?
06:37.33shido6err
06:37.38shido6why are you using iax?
06:37.44shido6and not iax2 ?
06:37.45shido6:)
06:38.06shido6ive never had a prob with using one or the other port with iax2
06:38.10kc5cqm_voicepulse is using iax
06:39.10drumkillawell, they may say "iax", but they mean "iax2"
06:40.07kc5cqm_they also specify port 5036 in their example iax.conf
06:41.05drumkillawell that has to be way old ...
06:41.10drumkillathey can't still be using iax ...
06:41.15drumkillathat wouldn't compatible with anything
06:41.23drumkillawouldn't be*
06:44.25kc5cqm_could very well be an old doc
06:45.18*** join/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com)
06:48.00*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
06:48.57AmbroseI want to have 2 outbound dialing groups. So in zapata.conf I should put group = 1 and then channel = 1, channel = 2, etc for each context? Same goes for the second group?
06:49.06AmbroseI'm trying to get the syntax figured out
06:49.42silentfuryanyone here familiar with Pingtel's sipX?
06:50.48mogormannot very
06:50.56mogormanwe are generally asterisk folk here
06:51.05Qwelltip: Push to Talk on iaxcomm == good
06:51.25silentfuryah, #sipfoundry is empty :(
06:51.49mogormanthere is a shocker....
06:51.55alephcomThey must be smarter than us.
06:52.16mogormanlol
06:53.25silentfuryprobably makes more sense I ask when I'm in front of the box. we're having problems getting the Polycom 301's to register with the exchange.
06:55.54*** join/#asterisk CANO-1982 (n=alejandr@201.255.54.167)
06:56.37*** part/#asterisk CANO-1982 (n=alejandr@201.255.54.167)
07:03.29mogormangnite
07:03.35mogormansleep well people
07:03.39kc5cqm_g'nite
07:03.41kc5cqm_bbl
07:04.11welleshi mogorman
07:06.04*** join/#asterisk jyukes (n=jameshot@pool-138-89-211-251.atc.east.verizon.net)
07:06.17*** join/#asterisk ThaZZa_Work (n=me@203.80.44.200)
07:07.33ThaZZa_WorkHey all.
07:07.41alephcomgreetings
07:08.09ThaZZa_WorkAnyone be able to help someone with a little iax to iax extention dial-plan?
07:09.33*** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk)
07:09.50*** join/#asterisk kamuix (n=kamuix@195.78.4.174)
07:10.09*** join/#asterisk VeNoMouS_ (n=venom@60-234-209-199.bitstream.orcon.net.nz)
07:10.26VeNoMouS_any 1 alive?
07:10.49VeNoMouS_why does this example of dialing to another sip not work
07:10.50VeNoMouS_exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
07:11.08VeNoMouS_it thinks num@host is host
07:11.14VeNoMouS_it thinks num@host is a host
07:14.19alephcomexten => _42X.,1,Dial(SIP/user:password@otherprovider.net/@${EXTEN},30,rT)
07:14.25alephcomme thinks but I'm half asleep
07:14.38*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
07:14.55ThaZZa_Worki was thinking something along the same lines alephcom
07:15.59*** join/#asterisk yce19 (n=frijrijs@200.163.1.214)
07:16.08yce19yo! :)
07:16.16yce19anyone can help me plx?
07:17.15alephcomWhat's your question.  No guarantees I can help
07:19.17*** join/#asterisk Entegrity (n=Entegrit@c-65-96-118-47.hsd1.ma.comcast.net)
07:19.50ThaZZa_WorkStill no takers for anyone who know the switch command?
07:20.29alephcomI wish I had taken time to learn it....   I always was going to but just never got around to it.
07:21.17ThaZZa_Workalephcom: The switch command?
07:21.43alephcomyeah, it was on my list to use it between my boxes but it still hasn't happenend.
07:21.45yce19<PROTECTED>
07:21.45yce19yce19 Jan 11 05:19:57 NOTICE[26912]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
07:21.55VeNoMouS_well
07:21.58VeNoMouS_if that is the case
07:22.00*** join/#asterisk Pegger (n=peg@pool-68-163-139-134.bos.east.verizon.net)
07:22.04VeNoMouS_some 1 better change the default sample config's
07:22.07VeNoMouS_cause thats where its @
07:22.19VeNoMouS_i'll try it the other way
07:22.28ThaZZa_Workalephcom: It is very very easy to setup and get working.. I am just trying to adjust it a little. and mix in with some contexts. and having issues.
07:23.09ThaZZa_Workalephcom: I am trying to make it so if you dial 20X it will go to the switch context and send the full extention 20X.
07:23.17yce19any brazilian? vono user?
07:24.04ThaZZa_Workyce19: Looks to me liek you are only providing part of the SIP/IAX hostname. should there not be more than just vono?
07:24.27*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-225.claranet.co.uk)
07:24.32VeNoMouS_alephcom and thazza neg, it thinks its still part of the host oh i missed a /@
07:24.37ThaZZa_Workyce19: For example sip.vono.net
07:25.15VeNoMouS_yea neg
07:25.21VeNoMouS_even with /@
07:25.28VeNoMouS_it things the whole thing is part of the host
07:25.31VeNoMouS_after the first @
07:25.50VeNoMouS_s/things/thinks
07:26.32yce19i call to my voip no, but it doesn't rings!
07:28.29yce19i cant answer the call
07:28.51yce19and cant dial
07:29.59yce19yce19 look
07:29.59yce19yce19 extensions.conf
07:29.59yce19yce19 [default]
07:30.00yce19yce19 [vono]
07:30.00yce19yce19 exten => 1,1,Dial(SIP/vono,60,Ttr)
07:30.00yce19yce19 exten => 1,2,Hangup
07:30.04yce19yce19 sip.conf
07:30.06yce19yce19 [general]
07:30.08yce19yce19 register = user:pass@vono.net.br/1
07:30.10yce19yce19 [vono]
07:30.13yce19yce19 type=friend
07:30.16yce19yce19 username=user
07:30.18yce19yce19 fromuser=user
07:30.20yce19yce19 secret=pass
07:30.22yce19yce19 host=vono.net.br
07:30.24yce19yce19 fromdomain=vono.net.br
07:30.26ThaZZa_Work~pastebin
07:30.27jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
07:30.27yce19yce19 insecure=very
07:30.28yce19yce19 disallow=all
07:30.30yce19yce19 allow=g729
07:30.30alephcompastebin.ca is your friend
07:30.30rikstaidiot
07:30.32yce19yce19 context=vono
07:30.47yce19just need help
07:31.34ThaZZa_Workyce19: Please do not dump loads of data to everyones screen.. please use pastebin.ca that way people will be more likely to want to help you.
07:31.44yce19thx! :)
07:31.51enemy^xis it possible to limit the amount of incomming calls to 1 while allowing several outgoing calls?
07:32.20*** join/#asterisk EriSan (n=erisan@151.8.109.117)
07:32.40yce19enemy^x, i call my voip no, it rings but asterisk dont do nuthin!
07:33.37yce19<PROTECTED>
07:34.05*** join/#asterisk jpk (n=jpk@p54A776DC.dip.t-dialin.net)
07:34.18jpkHi guys.
07:35.08jpkI was directed here by drumkilla. Can anyone point me to the right direction for custom cdr values please? I have not found anything useful on voip-wiki.
07:35.31jpkI would like to put the AOC-E values in a CDR field. And I cannot use the userfield since this is already used otherwise.
07:35.34jpkAny clues?
07:35.37*** part/#asterisk yce19 (n=frijrijs@200.163.1.214)
07:36.01*** join/#asterisk Astar (n=astar@ANantes-154-1-61-100.w81-53.abo.wanadoo.fr)
07:44.18welleshi all
07:44.22*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
07:45.58welleshow to compile zaptel as i586 ?
07:47.16tzafrir_laptopwelles, for what CPU?
07:47.37tzafrir_laptopwelles, basically, point it to your kernel's config
07:48.34wellesamd64
07:49.15tzafrir_laptopit is a module that needs to be loaded to a specific kernel
07:49.30*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
07:49.44wellestzafrir_laptop: can u say more detail?
07:49.58*** join/#asterisk ChrisDE (n=ChrisDE@80.187.134.251)
07:50.31tzafrir_laptopdo you want to build it for an ia32 kernel or for a amd64?
07:50.46wellesamd64
07:52.14wellestzafrir_laptop: now the problem is that : on 32bit machine the meetme works fine when using ilbc as the codec, and when on 64bit machine the sound becomes bad
07:52.47tzafrir_laptopsorry, GTG. Anyway, doesn't sound like a zaptel problem
07:53.02tzafrir_laptopmore in the domain of asterisk
07:53.52jpkcan anyone enlighten me on codecpriority option in iax.conf?
07:53.58wellestzafrir_laptop: is any hint to solve such a problem?
07:54.06jpkI have set codecpriority=caller but still the server seems to determine the codec.
07:54.52jpk> requested format = ilbc,
07:54.52jpk> requested prefs = (),
07:54.53jpk> actual format = alaw,
07:54.54jpk> host prefs = (alaw|ulaw|g729|g726|ilbc|gsm),
07:54.55jpk> priority = mine
07:55.06jpkThis is e.g. with idefisk as a client.
07:59.35*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
08:04.01zoait says priority=mine
08:04.03zoawhich is weird
08:04.16*** join/#asterisk Soul (n=Soul@87-196-8-134.net.novis.pt)
08:05.05*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)
08:06.07*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)
08:06.43ThaZZa_WorkWhooo.. Worked it out.
08:06.56*** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk)
08:07.01ChrisDEhi *. is there any change in handling variables in asterisk 1.2.x? I have the problem that I have to define the accountcode this way: exten => _X.,1,SetAccount(${ACCOUNTCODE}). If I don't do this my radius-script won't be able to get the Variable ACCOUNTCODE. ???
08:08.16ChrisDEthe radius-script is accessing the manager api...
08:08.29jpkoopmannzoa: my point.
08:08.54jpkoopmannzoa: I though codecpriority=caller should change just that.
08:09.10zoayes true
08:09.20zoaso that sounds like not working
08:09.32*** join/#asterisk wellng (n=welles@222.90.196.81)
08:09.38jpkoopmannBefore being bashed around on mantis for reporting something that is not a bug, I want to make sure this is one.
08:09.45jpkoopmann+again
08:10.12jpkoopmannzoa: Can you by any chance try to reproduce this?
08:10.28jpkoopmannzoa: codecpriority is a global iax.conf option, is it not?
08:10.30zoai dont really have a lot of time for it now :(
08:10.40jpkoopmannunderstootd. :-)
08:10.42zoai dont know, really
08:10.45fourcheezejpkoopmann: don't worry about being bashed on mantis - it gives other people the chance to feel superior ;-)
08:11.11jpkoopmannfourcheeze: *g* Gives me bad karma though! :-)
08:11.26*** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net)
08:11.36*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:11.55alephcomChriseDe:  The replaced SetAccount with "Set"
08:12.00fourcheezeI tend to take the view that all bug reports are helpful, if only to point out the deficiency in documentation that makes one think it is  a bug
08:12.09fourcheezebut then I have nothing to do with bug reports and *
08:13.25fourcheezeanyone know of any * conferences coming up in western europe?
08:14.14fourcheezelooking at astricon already
08:14.59shido6what do you use instead of digitittimeout?
08:15.32ChrisDEalephcom: setaccount with set?
08:15.41zoaastricon is going to be soon
08:15.57Qwellzoa: like Oct, isn't it?
08:16.39zoai dont really know
08:16.42zoai think its unknown
08:16.51zoaoct is in the US i think
08:17.16*** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net)
08:17.16alephcomChrisDE:  Yes, the exact command slips my mind.  It's on the wiki though.
08:17.19*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
08:17.25MozartsGhost-:- MozartsGhost  is identified to services
08:17.25MozartsGhost: idle     : 117 hours 3 mins 44 secs (signon: Fri Jan  6 12:53:33 2006)
08:17.29MozartsGhostooh, leet.
08:17.32MozartsGhostmwehaea.
08:17.36MozartsGhosthi all.
08:17.37zoaanybody here using idefisk ?
08:18.39ChrisDEalephcom: no the problem ist that I have to do a setaccount - if I don't do this the asterisk manager doesn't get the variable
08:19.55jyukesyo anyone using CommPartners DID product?
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08:30.06koperniqshi
08:34.55Qwellman...time goes FAST in meetme confs
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08:37.15koperniqsis there a way to get the Useragent info for all peers? eg added to sip show peers ?
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08:47.13kippihow do you do menu driven queues, where you can say press 1 for x press 2 for y ?
08:57.18{zombie}kippi: that's called an IVR (interactive voice response)
08:57.20{zombie}http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu
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08:58.50ChrisDEhi again. In asterisk v 1.0.x  in the manager there are notices like
08:58.51ChrisDEEvent: Newstate
08:58.52ChrisDEChannel: SIP/21100000448-70d2
08:58.52ChrisDEState: Up
08:58.52ChrisDECallerID: 0
08:58.52ChrisDEDNID: 3461234567
08:58.52ChrisDEUniqueid: 1136969584.4553
08:59.08ChrisDEin the new asterisk version 1.2.x the DNID is missing. How do I get it?
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09:02.07wellnghi mogorman
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09:08.18shido6they are finally selling them!
09:08.18shido6http://store.apple.com/1-800-MY-APPLE/WebObjects/AppleStore.woa/72804/wo/uG5f604HuQmZ2EKNWXkNCBgqZNw/0.SLID?nclm=MacBookPro&mco=E27B7429
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09:09.57coppicewhat's an Intel Core Duo processor? :-\
09:10.52drraydual core cpu
09:11.09drraysupposed to be the same as 2 cpu's
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09:11.51wellnghi drray
09:11.53coppicewell if they mean the Yonah, that isn't supposed to be released yet
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09:12.23coppicetypical Mac - they only use the slowest versions of the chip
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09:13.58koperniqsis there a way to get the Useragent info for all peers? eg added to sip show peers ?
09:14.16drrayit's funny that these intels are 2-3 times faster than g5 macs
09:14.24drraykind of puts an end to the intel vs ppc thing
09:15.53n3c8the g5 is  a very old chip in escence... and I don't care what people say, I love the ppc motorola goodness, reminds me of my old amiga
09:15.55n3c8good times
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09:16.30drrayapple uses very old G5's in their macs?
09:16.44coppicethe ppc chips for apple have seldom come from motorola.
09:17.05alrsnot since Apple screwed over Motorola, at least.
09:17.38n3c8yes but deep down they will always be motorola's
09:17.43coppicemotorola would *love* to have supplied all the CPUs, but seldom got their act together
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09:18.11coppicen3c8: the PPC is IBM through and through, in everything except AltiVec
09:18.21alrsMy understanding is that Motorola put PPC on the backburner after they spent so much money preparing to roll out a PReP box and got rat-fucked by Jobs.
09:19.39coppicealrs: I dount that made any difference. Motorola Semiconductors and Motorols Computers are more likely to be happy seeing each other suffer
09:20.21tzafrir_laptopIIRC motorola is not much the power behind ppc
09:20.26tzafrir_laptop(lately)
09:20.29drrayI'm sad to see the powerbook line go away
09:21.04coppicenever were. I worked at Motorola Semis when the PPC was first getting going, and their plans were totally brain dead
09:21.06n3c8i think that's a little unfair... i thought it was more  a case of competitive intel pricing driving motorola away from the ppc
09:21.07wellngtzafrir_laptop: can u tell me how to set rtc to 1024Hz?
09:21.34tzafrir_laptopwellng, you mean HZ?
09:21.36drraywelling - do you mean app Milliwatt?
09:21.50tzafrir_laptopnot sure.
09:21.56wellngyes
09:22.39wellngtzafrir_laptop: it 's said the default value is 100Hz.
09:23.10zoawellng,
09:23.11tzafrir_laptopwellng, it is set at kernel compile time
09:23.13zoamake menuconfig
09:23.18zoaprocessor options
09:23.25zoatiming frequency or so
09:23.54tzafrir_laptopwill ztdummy work if HZ != 1000?
09:24.10tzafrir_laptops/work/build/
09:24.14wellngzoa: i try in this way ,but i can not find the option
09:24.48wellngtzafrir_laptop:i donot know
09:25.07zoaztdummy builds if its not 1000
09:25.31tzafrir_laptophow can it provide timely ticks?
09:25.38zoait cant
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09:33.28enemy^xif someone calls a direct number, and then gets busy, then I push the caller into a queue for that person. But for some reason, it calls on the persons phone after adding him into the queue (even though he is busy?).... How can I stop the queue from hammerin people which are busy on the phone.
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09:34.19kippihi,
09:34.36jpkfn~zoa: Cannot reach you via query due to irc-bouncer problem.
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09:35.25kippihow can you do menus with queues, for like press one for x press 2 for y
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09:40.29zoahey ho jpk
09:40.35zoastrange
09:40.47zoajoin #asteriskguru then
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09:51.46dpryojpk: You probably have to register with freenode first ;)
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09:52.02jpkI know. But my nick is taken.
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09:56.26welleszoa: i can any option to modify the Hz on menuconfig
09:56.52welleszoa: i can not find any option to modify the Hz on menuconfig
09:56.59zoawhat kernel are you using ?
09:57.05zoait was introduced in 2.6.12 i think
09:57.06welles2.6
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09:59.00welles2.6.9-22. i c thanks
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10:01.00RoyKintel dual-core powerbooks......
10:01.12darkskiezbattery life is not mentioned
10:01.21darkskiezprobably shit
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10:04.03coppicewhen notebooks mention battery life its usually a meaningless figure
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10:07.42korkibongoanyone here have experience with tunneling ISDN over ip?
10:08.31jpkoopmannanyone here can point me to a "custom-cdr field" documentation?
10:09.15farhantry www.voip-info.org wifi
10:09.27jpkoopmannNothing there. Nothing I found.
10:09.46jpkoopmannThere is a userfield in CDR but that is not what I want.
10:09.58farhanalright, what are u looking for then?
10:10.03farhancan u program in C?
10:10.10jpkoopmannI wanted to have an additional CDR field for the AOC-E values and drumkill told me to ask here.
10:10.17jpkoopmannnot really.
10:10.19farhanok
10:10.29*** part/#asterisk ChrisDE (n=ChrisDE@80.187.134.251)
10:10.29farhanfirst, u need switch to mysql cdrs
10:10.34jpkoopmannalready done.
10:10.34farhannot csv
10:10.50farhanok, then get to command line mysql, and add the field
10:10.57farhanwhere do u want to set this field from?
10:11.06jpkoopmannActually: chan_zap.c
10:11.16farhanwhat do u want to put into it?
10:11.23jpkoopmannAOC-E is reported by ISDN and contains the cost of the call.
10:11.31farhanoh
10:11.44jpkoopmannAnd I entered a feature request to have this in the standard distribution.
10:11.50jpkoopmannIt is quite important in Germany.
10:11.50farhanalright, this requires hacking chan-zap.c
10:12.08jpkoopmannI know. I have someone who would hack it (kapejod who is doing bristuffed)
10:12.15jpkoopmannHis comment: Where do I put it? :-)
10:12.23farhanalright, i will tell u where
10:12.43farhanin the pvt part of the ast_channel datastructure
10:12.50jpkoopmannTherefore the feature request which was closed by drumkill. He thinks this should be a custom-cdr value and I think this should be part of the standard *.
10:13.41zoajpkoopmann: we do custom development if you want
10:14.22jpkoopmannfn~zoa: That's the point. I think this is a valuable feature for all ISDN customers in Germany. This should not be a custom developed solution for a single company.
10:14.25zoai also think it should be part of the standard
10:14.40jpkoopmannThen do me a great favour and add your comment to my bug report.
10:14.43zoaand if its part of the isdn standard, i think you want to convince matt
10:14.53zoaurl ?
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10:15.33jpkoopmannhttp://bugs2.digium.com/view.php?id=6152
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10:15.45jpkoopmannmatt?
10:15.50gnom[eq
10:15.54gnomõóéëî
10:16.10jpkoopmannfn~zoa: matt who? remember: I am new here. :-)
10:16.30gnomáëÿ ÿ æå â ðîñèè
10:17.02zoahe is called cresl1n here
10:17.29gnomûûûû
10:17.32jpkoopmannIC. And he is doing exactly what in * that I want to convince him?
10:18.43*** part/#asterisk gnom (n=smirnoff@81.222.176.137)
10:18.53zoahe is the author of chan_zap
10:19.01jpkoopmannAh. That makes sense! *g*
10:20.28jpkoopmannNow he could make the change in chan_zap (as could kapejod from bristuffed), still someone needs to decide on the additional CDR field.
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10:21.00zoaits better if he would, ask kapejod would probably not disclaim the patch
10:22.08zoai made a comment and closed it out
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10:22.37zoai would recommend to try alternative ways of opening the discussion, e.g. by putting 1) a patch on mantis or 2) opening a discussion on asterisk-dev
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10:31.07jpkoopmannfn~zoa: I see. Uups. I just reopened it. A simple patch will not do. In order to do this cleanly we need a new AST_CONTROL frame.
10:32.43jpkoopmannAsterisk-devel? Mailing-List or IRC?
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10:33.25kippiwhere can I download festival from?
10:33.32zoamailinglist
10:33.39jpkoopmannIc.
10:34.08*** join/#asterisk gdh (i=foobar@bum.net)
10:34.56gdhDoes anyone have a WAV of a UK phone ringing tone handy? :)
10:36.25coppiceit goes ring ring....... ring ring....... ring ring
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10:39.19kink0hello
10:39.34cfhwhere can i config the languages messages?
10:40.00kink0is ok ( even not RFC compilant ) that Asterisk sends Max-Forwards fields in SIP responses ?
10:40.36kink0cfh: my experiments with voice messages ( spanish ) I got a very bad recorderd messages,
10:40.53fenlandergdh: somewhere, yes :) I can look it out if you want
10:41.13kink0is a more beautyfull voice the english one ... may be some day there available a good voices for spanish
10:41.16cfhok but i found http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international this
10:41.29gdhcoppice: Thanks - helpful ;)
10:41.40gdhfenlander: please :)
10:41.44cfhand if i call the digital receptionist works whti my language it
10:41.56kink0cfh: yes, and you can get the voice recorderd messages for other languages and install it, just use the /fr /es or so directory
10:42.13coppicegdh: are you actually looking for a wavefile, or to know what it looks like?
10:42.16cfhif i call the voicemail from internal astersik set the languages on en
10:42.25cfhwhere can i change this set
10:43.12gdhcoppice: I have some cheap phones with preset american ringtones - all the other tones provided are musical - I just want a UK ringing  sound :)
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10:43.46kink0; Default language
10:43.46kink0;language=en
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10:45.14cfhkink0 where is the config file on asterisk @ home?
10:45.20kink0cfh: edit your /etc/asterisk/"channel".conf
10:45.42kink0where channel may be oss. alsa. zaptel or what you are ussing and wanna to se a different language
10:46.04kink0also you can select language in extensions.conf if you wanna support languages based on extensions
10:46.05cfhok
10:46.07cfhthanks
10:46.13RoyK.... ..
10:46.51cfhin zapata default languages is it
10:47.14RoyKit defines in what language it should ring
10:47.15RoyK:P
10:47.41cfhfrom an internall call the language is en
10:48.33Mimmusdoes anyone use 802.1p in his/her VoIP LANs?
10:48.37kippihas anyone installed festival and willing to help me out?
10:49.48Mimmuskippi: festival is not a simple beast but usually works out of the box
10:49.48cfhfrom an external call the language is it
10:49.52cfhwhy?
10:50.12kippiMimmus: does it work well once installed?
10:51.08Mimmuskippi: I think so, for Asterisk you can use Festival() app or an external agi script
10:52.01kink0anyone has suffered problems while response SIP messages from Asterisk sent the Max-Forwards field ?
10:52.54kink0I have a problem connecting nexton -> asterisk, apparentely because asterisk responses SIP headers sent Max-Forwards info , that appears is not RCF compilant
10:53.16kink0RFC
10:56.34kippiMimmus: Just found the rpm :D
10:56.43Mimmuskippi: good
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11:08.26HamNoIanyone knows of best predictive dialers around?
11:09.15MimmusI read yesterday about VICIDIAL http://astguiclient.sourceforge.net/vicidial.html
11:09.18trixtermost predictive dialers are bundled with crm solutions too
11:09.42trixterso 'best' is really really subjective
11:10.19HamNoItrixter: how is gnudialer?
11:12.44trixterhavent used it
11:13.05trixterit does integrate with crm stuff so if you want to use it with crm you should look at that part to make a decision
11:14.02trixterpersonally I didnt find any that did what I want how I wanted it so I ended up writing my own
11:14.02HamNoItrixter: I only need outbound call detection of faxes, busy tones and stuffs
11:14.51trixterif you just need a little bit of detection it shouldnt be hard to write one
11:15.24trixterthere is even an article on nerd-vittles about 'teleyapper' on exactly how to do just that
11:15.33trixtercame out like a week ago or something
11:15.37HamNoItrixter: gnudiar sounds like what I need but it doesn't work with the latest version of *
11:16.25HamNoItrixter: I see. It's interesting to do one for my own then
11:16.36trixterI would have no idea try #gnudialer, if anyone is awake they should be more able to help
11:16.40Astarztdummy: Unknown symbol rtc_register ; i "ve this error when loading zdtummy
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11:17.40puzzledmorning
11:17.45trixtersoundsl ike you dont have rtc enabled in your kernel, but I could be wrong
11:18.35trixterhi puzzler
11:19.55drray_http://dallas.craigslist.org/mis/124257388.html
11:19.57drray_oops
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11:21.44trixterthey need to make a law banning showing 250 pound women in thongs on tv
11:22.02Astari saw it was using usb timing no ?
11:22.26drray_so you want tv to only represent 2% of the women out there?
11:22.48trixteryeah why not?
11:24.55d-techyou mean 98% of woman are 250lb and wear THONGS?!
11:26.47Astari don't find rtc timer in my kernel
11:26.56drray_I'm sure about the thong part of that actually
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11:29.50kippihi
11:29.51kippiexten => 4,1,Answer()
11:29.51kippi<PROTECTED>
11:29.52kippi<PROTECTED>
11:29.52drray_I wonder how long before someone puts windows on the new MacBook
11:30.04kippiit just seems to hang me up, anyideas why?
11:30.26trixteris festival installed *and* configured
11:31.35kippiguess not, getting this error Jan 11 06:30:15 WARNING[3354]: app_festival.c:376 festival_exec: festival_client: connect to server failed
11:32.24fulgaswhgat's the voltage for and E100p ?
11:32.26fulgas5 ?
11:33.39darkskiezMessage type: DISCONNECT (69) Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Network beyond the interworking point (10)  Ext: 1  Cause: Normal, unspecified (31), class = Normal Event (1) ]
11:33.43darkskiezWhat does that mean?
11:34.34Mimmuskippi: I never tried with Festival daemon, I used festival-script.pl AGI script that generates an audio file on-the-fly
11:36.40kippiMimmus: with the daemon how do i get it to create a file?
11:37.00Mimmuskippi: I don't even to start the daemon!
11:37.14kippiah ok
11:37.15kippisorry
11:37.19darkskiezI've a problem with calls to bangkok getting disconnected after 30 seconds, does that message mean anything helpful?
11:37.29trixterI am happy that the sacramento asterisk users group is getting a free copy of libisup to do a ss7 demo ...  that should be fun :)
11:39.30Mimmuskippi: it is simple, for instance: exten => *60,1,Answer, exten=> *60,2,AGI(festival-script.pl|Your phone number is ${CALLERIDNUM}.)
11:40.33kippiMimmus: where do you get the festival script from
11:40.41*** join/#asterisk zotz (n=zotz@24.231.47.175)
11:40.52Mimmuskippi: festival-script.pl is a really simple perl scipt included in AsteriskAtHome distribution
11:41.04kippinot using at home
11:41.16Mimmuskippi: I can also mail it to you, if you like
11:41.26*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com)
11:41.27kippiyeah that would be good thanks
11:43.23jalsothi
11:43.34jalsotanybody experiencing asterisk crash on loading speex compiled with SSE?
11:46.10tzafrir_laptopJabroni, what is your CPU?
11:48.15jalsottzafrir_laptop: did you want to ask me? :)
11:48.47tzafrir_laptopjalsot, yes. Did it crash with SIGILL? (illegal instruction)
11:49.18jalsotI have: Intel(R) Pentium(R) 4 CPU 3.00GHz (...fxsr sse sse2 ss...)
11:50.09jalsotI think not, checking...
11:52.25kippianyone know how to start festival as a server?
11:52.56jalsotI got: Asterisk exited on signal 11
11:53.30jalsotbut while safe_asterisk didn't drop me a coredump, I started with -vvvcg and what is crazy, this way it didn't crash
11:53.58*** join/#asterisk aNaSTaCia_geBeri (n=Haydar@85.108.150.190)
11:54.02jalsotcan it be a problem with access rights? normally I run asterisk as a normal user, not as toor
11:54.05jalsotroot
11:58.42*** join/#asterisk benve (n=benvenut@sei.yacme.com)
12:00.24*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
12:02.21*** part/#asterisk benve (n=benvenut@sei.yacme.com)
12:05.35drray_oh noes, jalsot knows my root password!
12:10.01*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com)
12:10.53jalsotdrray_: :D
12:13.52kink0I have a problem connecting nexton -> asterisk, apparentely because asterisk responses SIP headers sent Max-Forwards info , that appears is not RFC compilant
12:14.57*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
12:15.05sivanamorning
12:23.32*** join/#asterisk dcoulson_ (n=dcoulson@wilbur.geekcolony.net)
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12:26.33jalsottzafrir_laptop: I found out something :) When I run asterisk with -c, speex with SSE works fine, however when I start with safe_asterisk, it crashes. Any idea?
12:28.17tzafrir_laptopjalsot, no
12:29.03*** join/#asterisk tuxinator_linuxM (n=tuxinato@70-32-106-248.ontrca.adelphia.net)
12:34.32sivanadoes anyone know if Teliax allows you to pass in CID name/number?
12:35.16tzafrir_laptoptrixter, here?
12:35.33kippican anyone help me with a festival error i am getting?
12:35.41trixterwhats up?
12:36.07*** part/#asterisk drray_ (i=drray@dsl254-011-243.sea1.dsl.speakeasy.net)
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13:01.52cj-rmHey ppl.... I'm about to start using realtime * and I'm wondering if the extensions defined in a particular context get overridden by the switch statement and those stored in the database.  I'm guessing you can have some extensions defined in your dialplan and others in the DB.  Is that right?
13:03.02*** join/#asterisk nextime_ (n=nextime@213-140-6-103.ip.fastwebnet.it)
13:03.13cj-rmkippi: Whats the error?
13:05.47*** join/#asterisk tempy (n=slacker@c220-239-92-6.rochd1.qld.optusnet.com.au)
13:07.26*** join/#asterisk cpm (n=Chip@199.227.4.34)
13:08.10cj-rmWith realtime asterisk can you specify some parts of a dialplan context in the DB and others in extensions.conf?
13:09.49tuxinator_linuxMjpkoopmann, man of many names
13:10.39tempyhi, is anyone aware of a reliable asterisk-compatible hardware vendor in Australia?
13:11.50RoyKzoa: ping
13:14.01*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)
13:14.19zoapong
13:14.57thazzaping
13:15.14zoapang
13:15.44RoyKzoa: it seems the jb doesn't want to be adaptive
13:15.52RoyKit says 'fixed' whatever i do in the config
13:16.36thazzawang
13:16.41tuxinator_linuxMwong
13:16.54RoyKtwang!
13:16.59tuxinator_linuxMtwong
13:17.18thazzawong
13:17.35tuxinator_linuxMIt's 5 AM here, what's your excuse for acting silly?
13:17.36thazzabong
13:17.40tuxinator_linuxMbing ?
13:17.47thazzasing?
13:18.06thazzatuxinator_linuxM: Acting silly? This is me normally.
13:18.17jalsottzafrir_laptop: FYI: I had an older init script which had 'export LD_ASSUME_KERNEL=2.4.1'. It seems, this caused this amazing problem :)
13:18.31zoaadaptive should be the one by stevekann
13:18.32jalsottzafrir_laptop: so problem solved, thanks for help
13:18.38zoawe will have a look at it
13:18.55zoait only works on the global config
13:19.41zoajb-impl=adaptive
13:21.16zoayou need to reload asterisk
13:21.22zoa<PROTECTED>
13:22.31zoayou probably need to change it in zapata.conf and not in sip.conf
13:23.41zoaif you are doing sip to zap that is
13:26.04*** join/#asterisk Ahrimanes (n=michael@aronsen.dk)
13:27.00*** join/#asterisk Clkio (n=clkio@217.165.233.238)
13:27.28Clkiohello? is it ok to ask general telephony/voip questions?
13:28.45sivanacan I include another file in iax.conf?
13:28.59sivanaClkio: no, this is a baking channel
13:29.25tuxinator_linuxMClkio, put the word * in there somewhere, and we won't notice it doesn't have anything to do with *
13:29.27benjksivana: want my Guinness sourdough bread recipe?
13:29.32Mimmussivana: yes, of course, as usual include ...
13:29.38Clkiook i guess i can then, some ppl are anal, if i dont ask specific asterisk question i get the boot
13:29.51sivanaMimmus: #include outsidefile.ext ?
13:30.05Mimmussivana: si, ah sorry... yes
13:30.12sivanaty
13:30.14Mimmus'si' is italian!
13:30.35sivanaI got it :)
13:30.51tzangeryeah sivana is kind of uncultured
13:31.12sivanahaha... you live with the menonites
13:31.49Clkiocompany A in country A need to call regular phones in Country B through Company B in Country B, the only thing we wanna pay for is the local charges from company B to local telephones in area, how do you achieve such a thing?
13:32.26tzangerClkio: you need PSTN access in country B
13:32.32tzangerand PSTN access generally is not free
13:32.50tzangeryou pay for either analogue phone lines or you pay for digitial (ISDN) lines to the telco
13:32.52benjk[asterisk-A] ====IAX-trunk====> [asterisk-B] -----> PSTN of country B
13:33.04tzangerexactly
13:33.35tzangerthen in company A's dialplan you only route calls that would be local to Company B to Company B
13:33.38Clkiotzanger well I have 24 lines in country B to a phone call from country A -- internet -- country B server -- use any of the 24 lines
13:33.42*** part/#asterisk cfh (n=luca@82.193.23.6)
13:34.10tzangerClkio: then what benjk just posted is exactly correct.  AsteriskB would have (likely) a T1 card
13:34.47Clkiois there dedicated hardware i dont think anyone would be able to set up asterisk there
13:34.54Clkiotakes me back to my first post :)
13:35.13tzangeryes there are T1 gateways, or you could just build one with a cheap system and asterisk and just bolt it to their wall.
13:35.29sivanaahem
13:36.44*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:36.48sivanatzanger: can you have Mark msg me when you talk to him next?
13:37.18tzangeryeah I think so
13:37.43Clkiothat these things are expensive http://www.voipsupply.com/index.php?cPath=94_128
13:38.01tzangerindeed they are
13:38.09tzangerI could supply you with one ofr half the price
13:38.17tzangerClkio: or you could do it yourself for quite a bit less
13:38.18koperniqsis there a way to get the Useragent info for all peers? eg added to sip show peers ?
13:38.22Clkioasterisk alternative?
13:38.34tzangerno, it'd be a preinstalled asterisk system
13:39.09tuxinator_linuxMClkio, ewww, dell's
13:39.21tzangerDell's what?
13:39.35Clkiowould throwing two of these do the job ? http://www.voipsupply.com/product_info.php?products_id=796
13:40.02tzangerdo you have 24 POTS lines or a T1?
13:40.18ClkioDSL
13:40.40tzangerthe 24 lines you have, are they analog lines or a single t1?
13:40.46Clkioanalog
13:40.47tzangerDSL is not a valid answer
13:40.54ClkioADSL 512
13:41.09tzangerok so you have 24 regular phone lines on a wall you need to plug in to a machine
13:41.10tuxinator_linuxMtzanger, not a fan of dells
13:41.22tzangertuxinator_linuxM: yes but you use 's which is possessive.  :-)
13:41.32tzangerI'm just being a grammar nazi, don't worry
13:41.45tzanger~google bobtheangryflower apostrophe
13:41.47Clkioits actually adsl2+
13:41.50Clkio256k
13:41.53tzangerClkio: again that does not matter
13:42.12tzangerAlthough that will be awfully tight for getting 24 voice calls through without gsm or g729
13:42.23tuxinator_linuxMtzanger, I sincerely appricate you pointing my grammer error to me, I don't want to give the wrong impression of my feels toward dell's again
13:42.41tzangeryou used it wrong again :-)
13:42.50tzangerdell's  == something belonging to dell
13:43.00tzangerdells == multiple 'dell'
13:43.23thazzadell's == a place where you purchase meat.. Oh sorry thats a deli. lol
13:43.44coppiceDingley Dell == something from a Monty Python sketch, and what Dell always reminds me of :-)
13:43.56tzangerhttp://www.angryflower.com/bobsqu.gif
13:44.16tzangerClkio: actually 256k is not enough for 24 gsm voice channels
13:44.21coppiceI came home from deli only this morning. Oh, no. That was Delhi.
13:44.24tzangernot even with iax2 trunking but I'd have to check again
13:44.25tzangercoppice: haha
13:45.01tuxinator_linuxMDid I use my apostrphe correctly?
13:45.10tzangertuxinator_linuxM: nope
13:45.15tzangeryou said you dislike dell's
13:45.37tzangerwhich is not correct.  you dislike dell's _what_  (dell's means something belonging to dell)
13:45.43*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
13:45.51Clkiotzanger how will it work so that employees in company A can just dial the phone number of regular people in country B? the gateway on both end with take care of mapping?
13:45.54tzangeranyway I didn't mean for this to get so out of hand :-)
13:45.56*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
13:46.22tzangerClkio: as I said you will not get 24 channels across a 256kbps link with gsm.  g729 maybe, I have to run the numbers
13:46.46sivanaClkio: the dialplna is however way you set it up in the extensions.conf of both boxes
13:46.46Clkiodont worry about bandwidth they either upgrade line or reduce ports
13:46.48Clkioi guss
13:46.56sivanadialplan
13:47.07*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
13:47.20coppicethe thing I dislike most about Dell is the way the rip off Asians
13:47.36Clkiobut i cant use asterisk because it requires hackers not some random employee who gets lost
13:48.10tzanger24 calls over IAX2 nontrunked using gsm will be 612kbps.  trunked will get that down to 327kbps
13:48.43tzanger24 calls over IAX2 nontrunked using g729 will be 492kbps, trunked gets you to 206kbps
13:48.46tzangerbut that's awful tight
13:49.06tzangernot to mention $240 per side for g729 licensing
13:49.24tzangerand heavier hardware to be able to actually do a decent job with g729
13:49.51jpkoopmannwhat about ilbc? How much bandwidth does that take and is it that much worse than g.729?
13:50.15tzangerilbc sounds worse than gsm and takes more bandwidth than gsm
13:50.34sivana<PROTECTED>
13:50.36tzangeryou oculd use g723-1 but I don't know if asterisk has that :-)
13:50.53tzangeryou could fit it in using LPC10 easily :_)
13:51.00jpkoopmannit does? I just fooled around with ilbc between two * and was not able to notice that much difference to alaw.
13:51.02tzangersivana: who let you out of the business meeting?
13:51.08sivanahehe
13:51.19tzangerjpkoopmann: I seem to notice and any time I switch from gsm to ilbc everyone in the office complains
13:51.47jpkoopmannthe only thing I noticed was meetme problems. Other than that it was fine. On the other hand that was just one user.
13:52.07jpkoopmannDoes it really take more bandwidth than gsm? That's strange. I would have expected less.
13:52.28jpkoopmannIs there a good bandwidth comparison for the supported codecs?
13:53.20tzangergsm = 13kbps, ilbc = 15kbps
13:53.22*** join/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)
13:53.41jpkoopmannwhat about g.729?
13:53.46tzanger8kbps
13:53.49sivanawav?
13:54.02tzangersivana: don't make me smack you with a business plan
13:54.07jpkoopmannIC. But will sound better (if your PC can cope with it).
13:54.21tzangerjpkoopmann: no, it does not sound better in my testing
13:54.32jpkoopmannNot?
13:54.45tzangerdoes not.  ilbc sounds worse than gsm in my (and my coworker's) ears
13:55.05PrivalCall waiting... How can I enable it by default? Aastra 9133i and 480i do not have callwaiting enabled by default so when line1 is busy on those phone and call waiting is not enabled, Asterisk sees the phone as busy. Any hints appreciated.
13:55.05jpkoopmannI was referring to g.729 in comparison to ilbc/gsm.
13:55.23*** join/#asterisk javar (n=javar@69.79.51.8)
13:55.28tzangerg729 sounds about the same as gsm in my comparison tests... much worse for on-hold music but that's not exactly what it was designed for
13:55.56jpkoopmannagreed. But if it "only" sounds like gsm, it is not really worth the hassle.
13:56.03jpkoopmannAt least not for * <-> * scenarios.
13:56.20tzangerjpkoopmann: depends on available bandwidth or if you're going to be handing off to something else, yes.
13:56.38jpkoopmannThat's what I meant.
13:57.03jpkoopmann8 to 13 is not that much a difference. Not like gsm to alaw e.g.
13:57.25jpkoopmannBut since we are so used to ISDN here I am currently working with alaw only if somehow possible.
13:57.27Kattyblitzrage: yes, i may be at E-Tel
13:57.32Kattyblitzrage: my room is already booked.
13:57.48Kattyblitzrage: just waiting for billing to give me my plane ticket.
13:58.01*** join/#asterisk elephantMan (n=elephant@197.205.103-84.rev.gaoland.net)
13:59.52MimmusPrival: callwaiting=yes ?
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14:00.38*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
14:00.59sivanaanyone here using parked calls?
14:01.21Kattyblitzrage: http://www.webcon.net/~izaah/gallery/d/1062-1/c2w_00001621.jpg and so you can find me (=
14:01.25tzangeryes
14:01.43sivanawhat would you put in the context?
14:01.48Clkiotzanger which hardware would i need for that scenario if i were to use asterisk and my connection between the two countries is adsl2+ 1mbps http://www.asterisk.org/hardware
14:01.50tzangerKatty: at least give him the butt shot, jeez you need to think like a guy
14:02.03Kattytzanger: oh hush, you silly rabbit.
14:02.18tzangerClkio: as I said, your bandwidth connection has absolutely *zero* to do with your PSTN interface.
14:02.40tzangeryou have 24 analogue phone lines to hook up.  You can use a TE110P+FXO channel bank, or the new TDM2400P
14:02.43tuxinator_linuxMKatty, Found you !
14:02.57tzangeror six TDM404P cards but that's kind of silly :-)
14:03.04*** join/#asterisk endre (i=endre@vlan2-v6-gw.otthon.urbnet.hu)
14:03.09Kattytuxinator_linuxM: the last conference i went to, no one talked to me )=
14:03.43tzangermy suggestion is likely the TDM2400PE (24 ports+echo can) and a Sangoma S518 ADSL card
14:03.59tzangerthat'll fit in a 1U server
14:04.04Clkiotzanger two of these offcourse?
14:04.08tuxinator_linuxMKatty, come to VON, I hang with oyu
14:04.11tzangerboot from USB, done.
14:04.14Clkiopci full length? does that plug into standard pc?
14:04.30Kattytuxinator_linuxM: (=
14:04.33tzangerClkio: you need a case that is large enough, of course.
14:04.48tzangerkatty needs to come to Listowel, we can go cow tipping
14:04.48Clkiotzanger on that tdm card, where do the phones plug? silly heh
14:04.57*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
14:05.00tzangerClkio: you really need an asterisk consultant.
14:05.04Kattytzanger: i do not tip cows. i'll hug them though.
14:05.13Kattytzanger: and feed them lettuces.
14:05.20tzangerKatty: cows don't eat lettuch
14:05.20iDunnowhat would they do with a fiver anyways?
14:05.21tzangerer lettuce
14:05.21benjkpoor lettuces
14:05.29benjkhow cruel
14:05.32tzangeriDunno: you'd be surprised
14:05.32Kattyyes.
14:05.36Kattyand next i'll hug /you/
14:05.39Kattyvery cruel indeed.
14:05.55Clkioi need a smack
14:06.11tzangershe'll hug me tightly around the neck with her hands, I'm sure.  :-)
14:06.18tzangerHomer Simpson style
14:06.22*** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net)
14:06.24sivanaonce you park a call on let's say extension 702, how do you retrieve it?
14:06.28*** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com)
14:06.29Kattythat's what you think.
14:06.31tzangersivana: you dial 702 of course
14:06.44sivanaI see
14:06.49Kattyi...don't think i hugged anyone at the last conference.
14:06.58tzangerand you of course need 'parkedcalls' to be included in your context
14:06.59Beirdotzanger, "why you little!"
14:07.04tzangerBeirdo: exactly.
14:07.05*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
14:07.10benjkhug the lettuce
14:07.10Kattywunderkin: hi.
14:07.16Kattybenjk: k
14:07.23wunderkinhi.. kitty katty
14:07.31sivanatzanger: I don't the context thing, what would you put in there?
14:07.35sivanaunderstand
14:07.40jbalcombCan anyone recommend a method to have Asterisk to log individual state changes on a call?
14:07.43Clkiowhats the adsl card for if u have an adsl modem with ethernet interface?
14:07.49tzangersivana: in your context, you make sure you have include => parkedcalls
14:08.01tzangerClkio: you get rid of the external ADSL modem, of course
14:08.08*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:08.09sivanatzanger: in order to have access to the 7xx extensions?
14:08.12benjkClkio: you can avoid using the ADSL modem
14:08.13Clkiodont have to?
14:08.17tzangerI have had *nothing* but trouble with them as you approach your bandwidth limits
14:08.30tzangersivana: correct.  7xx is the default, you edit that in features.conf
14:08.40sivanaya, I get it now
14:08.43Clkiowhy would i?
14:08.44tzangersivana: I have to admit I'm getting a little nervous that I'm peering traffic with you right now :-)
14:08.50sivanalol
14:08.53Beirdotzanger, never had an issue here.  traffic shaping is your friend
14:08.57benjkClkio: more control
14:09.01*** join/#asterisk [chico] (n=chico@p54912824.dip0.t-ipconnect.de)
14:09.16tzangerBeirdo: every ADSL modem I've tried has started "blocking" my upstream at 1/2 my limit
14:09.24Beirdowow
14:09.30tzangerso to get proper upstream performance I have to rate limit at 50% of my upstream which is 800k
14:09.30Beirdomine doesn't
14:09.33tzangerwhich is terrible
14:09.44tzangeruse the Sangoma S518 and I can rate limit at 800kbps and actually use 800kbps
14:09.48BeirdoI rate limit at 700k on an 800k upstream
14:09.51Beirdonever had an issue
14:09.55tzangerBeirdo: whose modems are you using?
14:10.02BeirdoSpeedstream
14:10.11tzangerBell Canada gives out some shitty speedstream crap
14:10.24BeirdoI didn't get it from Bell
14:10.25Beirdohehe
14:10.26tzangerused lucent though too which are much better
14:10.40tzangerpersonaly I like having it all in the case and under kernel control
14:10.40BeirdoI actually wanted to buy an Alcatel one
14:10.50Beirdoew
14:10.58Kattyalcatel?
14:10.59BeirdoI don't like internal modems, never have
14:11.15tzangerI never used to but I kind of grew out of it
14:11.25Beirdoand I run OpenBSD on the firewall
14:11.26Beirdoso...
14:11.42tzangerthe original plan was to start playing with the internal driver code so that it could send prioritised ATM cells
14:11.48Beirdoahhh
14:12.10Beirdonow if anything on the other end would use it, sending voice over AAL2 would be sweet
14:12.33sivanabut the anti-matter containment field overloaded and short circuited the photon converters
14:12.34tzangerso that when a 1500 byte packet got fragmented and put in the outbound queue and a little VOIP packet came in next the VOIP ATM frames would get sent ahead of the bigger cell "cluster"
14:12.52tzangersivana: I'm serious, don't make me send you to a conference
14:12.56sivanaheh
14:13.12sivanasend me to E-Tel :p
14:13.14Beirdoideally, I'd run voice over ATM.  screw over IP.  but the DSL providers don't do AAL2
14:13.25tzangeryou just want to hug katty
14:13.29*** join/#asterisk [chico] (n=chico@p54912824.dip0.t-ipconnect.de)
14:13.33*** part/#asterisk [chico] (n=chico@p54912824.dip0.t-ipconnect.de)
14:13.36sivanaI don't like to screw over IP.. too much jitter
14:13.44Beirdohehe
14:13.46tzangerjitter's where all the feeling is
14:13.54*** join/#asterisk bmg505 (n=leon@c1-56-4.rndf.isadsl.co.za)
14:13.59coppiceah, fragmentation is such a problem. my children are a fragment asian and a fragment european, and they are a huge problem :-)
14:14.13*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
14:14.17iDunnoheh
14:14.23KattyiDunno: boo.
14:14.28parylg'morning guys
14:14.33iDunnohiya Katty :)
14:14.34tzangerheh
14:14.46iDunnoat last, today is looking up :)
14:14.51Mimmuswho is involved in libpri development?
14:14.59tzangeriDunno: days can only look up when you're in the gutter :-)
14:15.30coppicetzanger: a guy in a gutter can always be washed down a drain
14:15.43tzangeryou must have *really* big drains in .hk
14:16.15brad_msswcoppice: still no headway on bureaucracy for bug 5090 ?
14:16.17iDunnotzanger: it's been a bad day, I woke up late, more or less ran in to work to get here 2 mins late, then things just didn't work all morning... I've not got things working so it's looking up :)
14:16.19coppicestorm drains are plenty big enough
14:16.32wunderkinhe speaks from experience
14:17.00*** join/#asterisk [chico] (n=chico@p54912824.dip0.t-ipconnect.de)
14:17.16Beirdobig storm drains or little people in the gutters, I guess
14:17.31coppicebrad_mssw: i am no longer interested in what happens to bug 5090
14:17.45koperniqshow can i get useragen info for all peers in one list
14:17.46brad_msswcoppice: that's a shame
14:17.55RoyK~seen wasim
14:17.58jbotwasim is currently on #asterisk (4h 15m 55s), last said: 'hehe'.
14:18.23RoyKcoppice: you know, there _is_ a bounty for it....
14:18.40brad_msswcoppice: any interest in openpbx perhaps, since it's supposedly a fork of asterisk ?
14:18.51brad_mssw(not that I have ever really looked into it)
14:19.00*** join/#asterisk simulated (i=user@adsl-070-155-044-220.sip.bct.bellsouth.net)
14:19.06RoyKcoppice: and that can be increased
14:19.09coppicethe T.38 stuff is in openpbx. needs a lot more testing though
14:19.22brad_msswoh, didn't realize that
14:20.54*** join/#asterisk ManxPower (i=ewieling@18.sub-70-210-30.myvzw.com)
14:21.01brad_msswcoppice: are you working on support for openpbx, or are you dropping your efforts all together ?
14:21.02pifanyone using isdn phones connected to a chan_capi device?
14:21.04Kattywe have this 'box' which takes a t1 and turns it into analog lines....and also our broadband.
14:21.27ManxPowerI manged to outsmart a cat (those of you with cats know how impressive this is)
14:21.31Kattyand the lines from there go into asterisk. but we have to powercycle that chunk of hardware a /lot/ ...is this normal?
14:21.35zoapif: i think that is not possible as afaik NT mode is not support on chan_capi
14:21.41Kattyonce every 2 days.
14:21.53pifzoa: I'm doing it, so it's possible
14:22.05zoahmm i need to fix my documentation then
14:22.23pifonly, I don't get ring of dialtone , althtough the call gets through
14:22.27ManxPowerKatty, sounds like the box is a channelbank
14:22.44KattyManxPower: it might be.
14:22.54ManxPowerKatty, it is NOT normal.
14:22.56*** join/#asterisk pengyong (n=lala@218.93.102.142)
14:23.16paryli've been seeing voicemails getting dropped ever since i went to asterisk... now i think i've figured out the problem, but i don't know what to do about it.  it appears that the SIP phones are getting notified of a waiting message before the message has finished recording.  if the user logs into VM to get their message and begins playback before the caller hangs up, the message simply gets deleted
14:23.18KattyManxPower: how often (roughly) would be an acceptable powercycle ammount?
14:23.27ManxPowerYou can expect an Adtran channelbank to have uptimes in years.
14:23.44ManxPowerKatty, We have never had to powercycle any of the 30 or so boxes we have.
14:24.07Kattyhmm.
14:24.16pifzoa: this is a public forum, stuff can be of interest to others
14:24.22Kattythis doesn't make me happy.
14:24.37zoai dont see how my email addy can be of interest to others :)
14:24.39ManxPowerKatty, what brand is your box?
14:24.41zoaim just asking for you config files
14:24.48zoanot asking for your help for the rest :)
14:25.00zoai dont have problems, just want to add documentation for others
14:25.01zoa:)
14:25.28KattyManxPower: VINA
14:25.37ManxPowerOH GOD NO!!!
14:25.43MimmusI'm getting "Unknown IE 26 (cs6, Unknown Information Element)" messages at console
14:25.52*** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
14:25.53cj-rmWith realtime asterisk can you specify some parts of a dialplan context in the DB and others in extensions.conf?
14:26.04Katty...
14:26.17ManxPowerKatty, I don't actually know anything about VINA other than the fact that Bellsouth tried to put one in at one of our offices.
14:26.28Kattyi see.
14:26.34tzangerwtf's a vina?
14:26.45ManxPowerThey are used for integrated voice/data.  You can assume it's a channelbank.
14:26.52tzangerahh
14:26.55ManxPowertzanger, an integrated channel bank and router thingy
14:27.16cj-rmDoes anyone here use realtime asterisk?
14:27.22ManxPowerBell South tried to put one in, hand us an ethernet port off of it and say that it was frame relay.
14:28.18javarhi
14:28.20ManxPowerWe laughed at them, told them that their drug dealer sold them some bad product and then called our account rep.
14:28.40javarsomebody can explain me how install a patch for asterisk?
14:29.00ManxPowerKatty, call the carrier and insist the box is replaced.
14:29.06KattyManxPower: no
14:29.09ManxPowerThe VINA should not crash.
14:29.27KattyManxPower: they won't do it unless i have /evidence/ of it not working properly. and saying i have to powercycle it isn't enough.
14:29.28tzangerManxPower: heh
14:29.46KattyManxPower: the phone company is quite incompitent though.
14:29.51ManxPowerKatty, Well, you can always just replace the box yourself.
14:29.52KattyManxPower: moreover, i wasn't asking for help.
14:29.55tzangerer
14:29.56tzangerKatty: take an axe to it
14:29.56KattyManxPower: just information.
14:30.25KattyManxPower: thank you for trying to Fix me though.
14:30.33*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
14:30.49*** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net)
14:31.36paryld'oh... disconnected... did anyone happen to answer my question?
14:31.48ManxPowerKatty, What phone company do you have?  If it's BellSouth I can give you the name of our sales.droid (who is suprizingly good)
14:31.58KattyManxPower: it's a little one you've never heard of before.
14:32.05ManxPowerKatty, Ah, OK.
14:32.59*** join/#asterisk `lyme (n=Lyme@manufacturerstransportation.com)
14:33.00ManxPowerKatty, you can always make the phone compayn dispatch a tech to come out and reset the box.  Eventually they will replaceit.
14:33.07RoyK~seen wasim
14:33.10jbotwasim is currently on #asterisk (4h 31m 7s), last said: 'hehe'.
14:33.20KattyManxPower: oh, i get it.
14:33.25KattyManxPower: you're trying to fix me again, right?
14:33.25tuxinator_linuxMKatty, I had a provider that made me powercycle the box a bunch of times, and they replaced it a few times, I and it always had problems, I ended up changing providers
14:33.38zoapif, but i understand the annoyance of people here contacting others in private, im just not like that
14:34.00tuxinator_linuxMKatty, not typing well today, I am
14:34.07ManxPowerKatty, No, I just enjoy helping people torture their phone company.
14:34.13zoaso but please send me your configs that i can fix this nasty bug on asteriskguru, i didnt know chan_capi does NT mode
14:34.14KattyManxPower: i see.
14:34.27*** join/#asterisk welles (n=welles@61.150.11.163)
14:34.39Kattytuxinator_linuxM: sadly, we are not going to be changing providers.
14:36.08tuxinator_linuxMKatty, what is the brand of your box that needs restarting?
14:36.15Kattytuxinator_linuxM: VINA
14:36.19tzangersivana: first pass, seems okay.  not sure what I'm looking for though
14:36.57sivanajust approval :)
14:37.07koperniqshow can i get useragent info for all peers in one list? eg as additional information sip show peers
14:37.08tzangerhaha
14:37.28sivanatzanger: is there a way to query the BIOS heat reports from within linux?
14:37.48tzangerlmsensors is about as good as you can get
14:37.59*** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net)
14:38.09sivanais that a slack pack?
14:38.30zoapif: this is the tutorial i want to change: http://www.asteriskguru.com/tutorials/bri.html
14:38.33coppicetzanger: configuring lmsensors is a PITA
14:38.45tuxinator_linuxMKatty, my trouble was with an adtran
14:38.45iDunnolmsensors are a PITA
14:38.56javarsomebody can explain me how install a patch for asterisk?
14:39.15KattyiDunno: pita bread.
14:39.21iDunnoget source, apply patch, compile, pray.
14:39.32iDunnoKatty: hmmmm pita bread is good :)
14:39.37KattyiDunno: tis.
14:39.40fourcheezeiDunno: you missed out "sacrifice goat"
14:39.49darkskiezand 'install' :)
14:39.50Kattyfourcheeze: let's leave the goats out of this.
14:39.55iDunnofourcheeze: dammit! I knew there was something that I was missing.
14:40.19fourcheezeI've never known a patch work without that
14:40.42*** join/#asterisk mmlj4 (n=looseduk@ip70-171-92-106.no.no.cox.net)
14:40.42darkskiezi've got quite fond of goat curry because of it
14:40.45Kattywow, i'm just not waking up today.
14:41.05*** join/#asterisk secure75 (n=mic@ppp-62-245-162-105.mnet-online.de)
14:41.26iDunnoKatty: know that feeling, and I've been at work since 9am (GMT/UTC)
14:41.31RoyKzoa: Jan 11 14:42:20 WARNING[19215]: chan_iax2.c:707 jb_warning_output: Resyncing the jb. last_delay 0, this delay -21784626, threshold 1000, new offset 21784626
14:41.42RoyKthat's one hell of a delay
14:42.24RoyKespecially for iax2, while doing a sip/zap call
14:42.28KattyiDunno: i've just been kinda sitting here drinking grape juice for 30 minutes.
14:42.41RoyKKatty: sounds like a sour experience
14:42.59iDunnoahh - I've fought a big Java CMS system, Oracle, and Tomcat so far today.
14:43.03KattyRoyK: not really. grape juice is sweet.
14:43.11tzangercoppice: I never had all that much trouble
14:43.15Hmmhesaysnevada nevada nevada here I come
14:43.23RoyKKatty: perhaps the one with a ton of sugar
14:43.28KattyHmmhesays: and missouri on the way back?
14:43.32RoyKKatty: but not the one I usually get
14:43.37HmmhesaysHey Katty, naw
14:43.40Hmmhesaysjust back to msp
14:43.46KattyHmmhesays: you make me sad.
14:44.03HmmhesaysKatty: lots of things make me sad
14:44.09zoaroyk, thats why it is resynching
14:44.11coppicetzanger: if your motherboatrd doesn't match the standard config, and you don't have a config file from the motherboard maker, trying to figure out the arcane config file is where the fun starts
14:44.16KattyHmmhesays: are you having a sad week?
14:44.21zoaits because the damn phone sends fucked up timestamps
14:44.27RoyKzoa: no idea...
14:44.28Ariel_argh  I hate MS Windows XP and MS Office....
14:44.31Hmmhesaysits been rough, between work and the engaged girlfriend
14:44.32Ariel_hello everyone
14:44.39KattyHmmhesays: woah, backup.
14:44.40HmmhesaysAriel_, long time how are you?
14:44.42zoathe resynching is to make sure the delay is NOT there
14:44.42zoa:)
14:44.44KattyHmmhesays: i thought you split it with her.
14:44.44RoyKzoa: note this was from chan_iax2 in a sip/zxap call
14:44.50RoyKs/zx/z/
14:44.53Ariel_Hmmhesays, fine hope your doing fine as well
14:44.58Kattyhey Ariel_ (=
14:45.00HmmhesaysKatty: i more just realized my place in that situation
14:45.01tzangercoppice: I typically just look at the chips on the mobo and fuck around until I find it :-)
14:45.06tzangersivana: check linuxpackages.net
14:45.06RoyKjbot: thanks
14:45.06jbotpas de quoi, RoyK
14:45.08zoahmm strange
14:45.10KattyHmmhesays: spill.
14:45.37HmmhesaysAriel_ been a rough couple weeks, but vacation starts tonight
14:45.40Ariel_ok I have not work with argh M$ for a long time now. But I am at a customer does anyone know why Office could take a long time to access the directory to save files?
14:45.57*** part/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
14:46.09Ariel_Hmmhesays, sorry to hear it's been rough. I hope the vacation goes well.
14:46.10HmmhesaysKatty: not much to spill, i'm the *other* guy. I'm cool with that
14:46.36RoyKzoa: http://pastebin.com/500825
14:46.47KattyHmmhesays: she's still getting married to whatshisface?
14:46.51RoyKzoa: it's at the start of the call
14:47.01Hmmhesaysthey are on a *break* for awhile
14:47.05Kattyoh.
14:47.08Kattyi see.
14:47.17Kattyand you're the /other/ guy
14:47.18iDunnoAriel_: you're saving to a network share and samba sucks donkey bollocks?
14:47.18Hmmhesaysnot getting along and such.. I can see why, he's a jackass
14:47.22*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
14:47.36RoyKzoa: or from another call http://pastebin.com/500828
14:47.37Kattymorning fender.
14:47.39jbalcombCan anyone recommend a method to have Asterisk to log individual state changes on a call?
14:47.45HmmhesaysKatty: yeah, he went rummaging through her phone monday night looking for my number, she kicked him out for that
14:47.54tzangerjbalcomb: not offhand
14:47.59Kattyhaha
14:48.01[TK]D-FenderKatty: mew.
14:48.17KattyHmmhesays: good luck (=
14:48.20Ariel_idpromnut, actually it's not saving to samba it's to an ms 2000 server... argh give me my CentOS system back please. I like OpenOffice it works better.
14:48.26HmmhesaysKatty: i don't need luck
14:48.28jbalcombtzanger: that is a shame. does this strike you as a uncommon desire?
14:48.28Hmmhesaysbut thanks
14:48.32KattyHmmhesays: enjoy it while it lasts....i'm still trying to find a female.
14:48.46tzangerjbalcomb: nah I had wanted it a few times for debugging
14:49.01tzangeryou'd have to meter out channel.c, the channel state changes all pass through a function in there
14:49.22tzangerAriel_: OO is nice but sucks goat balls for graphing
14:49.25iDunnoAriel_: erm - but saving to a remote filesystem? CIFS/Samba? both of which suck ;)
14:49.28jbalcombtzanger: ok, we are trying to track billing and also display the call path on the end users screen so they can feel what to expect.
14:49.34tzangerwhich is what I use Excel for 99% of the time :-)
14:49.39KattyiDunno: i'd rather have samba than nothing at all :P
14:49.50tzangerjbalcomb: so meter it out to a special log, tail the log and have some fun with it
14:49.59iDunnoKatty: I'd rather have scp or rsync than samba ;)
14:50.07jbalcombtzanger: meter?
14:50.11Ariel_tzanger, its' excel in this case yes. But takes about 2 minutes to go through the directory to save.
14:50.13tzangerI like OO Writer beter than Word though, it has some funny stuff for importing graphics and stuff though
14:50.18*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
14:50.22HmmhesaysKatty, thats the plan
14:50.29tzangerAriel_: that's odd, can you see what samba's doing?  The network?
14:50.43KattyHmmhesays: good man.
14:50.44tzangerjbalcomb: I mean add calls to a logging function whenever the channel state changes
14:51.22Ariel_tzanger, no samba.  it's all argh M$ at this customers site... I am here to setup an asterisk box. But can't save some files I have for them to print doc's out.
14:51.44tzangerhmm.  put it on a usb key and sneaker-net it?  that blows.
14:52.04jbalcombtzanger: ok, that sounds like my plan B thinking. the question then becomes, how to trigger a reaction when the state changes? perhaps simple add it at each junction in the call plan?
14:52.07*** join/#asterisk mkrufky (n=mk@68.160.103.77)
14:52.12Hmmhesayssneakernet is the greatest of the *nets
14:52.13*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
14:52.13Ariel_actually I pulled my laptop out and I am connecting it to there laser printer.
14:52.15idpromnutAriel_: eh? :)
14:53.00Hmmhesaysi hate that name
14:53.03cj-rmI'm using realtime asterisk with mysql, but I keep getting the following error:  Realtime mapping for 'realtime_ext' found to engine 'mysql', but the engine is not available.  Anyone got any ideas????
14:53.04Hmmhesaysgood os
14:53.07Hmmhesaysbut I hate the name
14:53.11*** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net)
14:53.15Ariel_yes it's an ugly name but it works
14:53.31mog_workcj-rm, mysql isnt loaded
14:53.34mog_workor connected
14:53.43tzangerjbalcomb: uh, do you want channel state changes or dialplan stuff?  The two aren't the same
14:53.53coppiceHmmhesays: all the good URLs have been taken. ubuntu was the best they could do :-)
14:53.53cj-rmmog_work: how do you mean?
14:54.17tzangerman a hot chocolate goes down FAST.
14:54.24jbalcombtzanger: good question. probably both really.
14:54.27Ariel_do you know what  buntu means in tagalog.....
14:54.48tzangerjbalcomb: well you can do the dialplan stuff really easily with system() or an AGI.  metering out channel.c is of course a little more involved but certainly doable
14:54.53puzzledanyone know if the TE210P card is available with the onboard echo can module? the Digium website does not mention it
14:54.59tzangerI wonder if the manager interface will already do it for you
14:55.05RoyKwtf is the te210? 2-port?
14:55.09puzzledyes
14:55.13coppiceAriel_: dunno. should I ask the neighbors? :-)
14:55.33tzangerRoyK: howso?
14:55.35tzangerhonestly
14:55.45cj-rmmog_work: I've got res_config_mysql.so copied into my asterisk modules dir.  And as far as I can tell the mysql settings specified in res_mysql.conf are correct
14:55.46tzangeryes sangoma cards are good but I have not found them to be exceptionally better than digium
14:55.53tzangerwtf is asterisk listening on port 2000 for?
14:55.58puzzledlol
14:56.03Kattytzanger: it's trying to find its mother.
14:56.13coppicetzanger: people seem to have rather less interrupt troubles with sangoma
14:56.23RoyKtzanger: i really dislike having to ship the board to the US to upgrade firmware...
14:56.36tzangercoppice: I heard that at the start but as time wore on I have heard and seen simialr interrupt troubles.
14:56.49wunderkinRoyK, you don't have to as long as it is 2nd gen
14:56.51tzangerRoyK: you don't have to; if you have a JTAG interface you can get the updated firmware :-)
14:56.54coppiceRoyK: digium fixed that
14:56.55iCEBrkr[TK]D-Fender: dude.
14:56.57Ariel_coppice, it's a sexual part actually slang word for dick
14:57.05RoyKok
14:57.17puzzledif I want to test back to back 2 boxes with a T210P in each I need an ISDN cross cable right?
14:57.23RoyKso all i have to do is send my boards to digium to upgrade to an upgradable firmware?
14:57.35RoyK:P
14:57.45wunderkinpuzzled, yes
14:57.46cj-rmmog_work: And I know mysql is up and running
14:57.51puzzledwunderkin: thanks
14:57.56tuxinator_linuxMAriel_, and you know this why?
14:58.13Kattytuxinator_linuxM: Ariel's smrt.
14:58.18Ariel_tuxinator_linuxM, I know some tagalog
14:58.25mog_workno i mean the module cj-rm
14:58.29*** join/#asterisk brimston1 (n=brimston@pcp01534724pcs.huntsv01.al.comcast.net)
14:58.33Ariel_Katty, thanks
14:58.39jbalcombtzanger: I'll give it a go and find some place nice to post my results. Thanks for the help.
14:58.44Hmmhesaysi hope to get my wireless monitors next week
14:58.47tuxinator_linuxMAriel_, Tagalog is a tough language
14:58.49KattyRoyK: http://www.bekkoame.ne.jp/~s_ita/port/port2000-2099.html
14:59.04KattyHmmhesays: i've secretly changed their delivery address to my work.
14:59.09Ariel_got to go now driving to the City.  argh I don't like having to drive into the city it's over an hour away.
14:59.25HmmhesaysKatty they are coming out of kansas city mo
14:59.25iDunnooops.
14:59.34cj-rmmog_work: so how do I load it?  res_config_mysql.so exists in /var/lib/asterisk/modules
14:59.40iDunnosomething is naughty and doesn't tidy up it's tmp files.
14:59.51wasimmoved there to see montana play for the chiefs
14:59.56mog_workdo a show modules
15:00.00mog_workor if i were you
15:00.01cj-rmoh wait up... I have to specify it in a config file don't I? :)
15:00.06mog_workid restart asterisk with the debug on
15:00.17mog_workand see if it trys to connect and fail
15:00.17mog_works
15:00.41cj-rmyou're right the module isn't loaded...
15:00.55Kattyput your right module in...
15:01.16*** join/#asterisk jsharp (n=jsharp@65.88.255.245)
15:01.39mog_worknow all you need to do is find out whay
15:01.40Kattyput your right module out
15:01.59mog_workid stare asterisk -vcd and with all options on in logger.conf
15:02.00Drukenput it back in and shake it all about?
15:02.02mog_workand see what happens
15:02.33cj-rmmog_work: ahh yeah... I think I need a cdr_mysql.conf in /etc/asterisk
15:02.44rob0wasim ME TOO
15:02.45tzangerjbalcomb: you may want to see if the manager interface gives you enough before you go digging too deep
15:03.24wasimrob0: he didn't do too well though
15:03.29RoyKKatty: why me?
15:03.56jbalcombtzanger: i was not aware of any manager interface. is it native to asterisk or an add-on?
15:03.57rob0oh no not the Montana thing. I was born there.
15:04.22tzanger~manager
15:04.24jbotmanager is, like, a thing that should be killed
15:04.25rob0Len Dawson :)
15:04.27tzangerhaha
15:04.34tzangergoogle for the asterisk manager interface
15:05.10*** join/#asterisk Broom (n=none@12.174.245.227)
15:05.12Broomhello all
15:05.33Broomi recently installed Asterisk@Home and all of the sudden it does not detect any of my Zap Cards
15:05.39Broomany command I can run to verify?
15:05.53mog_worklspci to see if they exist
15:05.57KattyRoyK: hmm?
15:06.29KattyRoyK: do you need a hug?
15:07.00Broomcould it be this one: Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
15:07.01Broom?
15:07.23iDunnoargh! a machine that has editor set as nano not vim!
15:07.26brimstoneBroom: looks like a digium card
15:07.28*** join/#asterisk oej (n=oej@apollo.webway.se)
15:07.32iDunnothat caught me out for a couple of seconds.
15:08.33jbalcombtzanger: will do. thanks again.
15:08.42tzangerhahaha
15:08.48tzangersounds like my craven underling has been there
15:09.22sivanais ulaw file format called .ul?
15:09.41Kattysivana: i don't think so.
15:09.43Kattysivana: i think it's .wav
15:09.50Kattysivana: just encoded differently.
15:10.09Kattysivana: you can even use sound recorder in windows to save a .wav in ulaw
15:10.20sivanaI want to switch to native moh instead of mpg123, but I want ulaw format to avoid any transcoding
15:10.27Ahrimanesis there a way to disable announcements on Queue's?
15:10.33sivanaKatty: I see
15:10.33iDunnowav has many formats... ;)
15:10.42KattyiDunno: i'll format your wav in a minute.
15:10.44Broomhumm.. i'm getting this error: line 0: Unable to open master device '/dev/zap/ctl'
15:10.44RoyKKatty: perhaps...
15:10.44iDunnoit's more or less a container format.
15:10.49KattyRoyK: k
15:10.51iDunnoKatty: ohhh, please do :)
15:10.54Broomwhen i run ztcfg -vvv
15:10.56RoyKKatty: but i just wondered
15:10.57KattyiDunno: with pleasure ^_^
15:10.58RoyK<
15:10.58RoyK>
15:11.49Broomi have a /dev/zapctl but not /dev/zap/ctl
15:12.48brimstoneor "make mknod" (i think) in the zaptel source directory
15:13.04brimstonenope, "make devices"
15:13.19Broomok, thanks
15:13.43Kattyman? what's man?
15:14.01RoyKBroom: or read the udev readme if using udev
15:14.02KattyHmmhesays: stop that.
15:14.18KattyHmmhesays: you're going to mess that lovely nose up.
15:14.20BroomThanks alot
15:14.29Hmmhesaysha! it's already got a hump in it
15:14.39KattyHmmhesays: i don't honestly remember your nose.
15:14.48MimmusI'm having an absurd problem: Asterisk doesn't detect answer for some (rare) numbers, essentially automatic responders I think
15:15.59sivanaso an mp3 file could have ulaw format?
15:16.00Hmmhesaysheh
15:16.10Mimmusline is PRI E1. Where can I indagate? Driver (sangoma)? Asterisk?
15:16.25Kattysivana: i don't really know, but i don't think ulaw is quite /that/ compressed.
15:16.26tzangerindagate?
15:16.46cj-rmmog_work: Now I'm getting "MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info."  Any idea why?  As far as I can tell the settings I've specified are correct
15:17.00Mimmustzanger: investigate. you are my everyday english teacher. Thanks!
15:17.08tzangerMimmus: no I just didn't know what you meant
15:17.18mog_workrun with debug
15:17.21tzangersangoma and digium cards both work fine
15:17.23mog_workit will make it a lot easier
15:17.36mog_workbut digium cards give mog that extra boost....
15:17.44brimstonethat and redbull
15:17.51mog_workmmmmmmmmmmmmmmmmmmmmmmmmm redbull
15:17.53Kattytaurine :<<
15:18.01*** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net)
15:18.10hugo-v6redbull? with vodka?
15:18.16GoshenWhat is the voicemail app in 1.2?
15:18.19BroomRoy: once I add the lines to the permissions file for udev, what do I do next?
15:18.22Mimmustzanger: my english is very bad :-(
15:18.37mog_workjust redbull
15:18.43lunklol
15:18.45sivanaKatty: I'm just trying to avoid any transcoding
15:18.59sivanaI want ulaw all the way
15:19.00sivanaheh
15:19.02tzangeryes digium cards help out Asterisk
15:19.03Mimmuseven 'pri intense debug span ...' gives me non useful info
15:19.04GoshenI am getting this error...
15:19.04GoshenJan 11 08:16:15 WARNING[9669]: file.c:508 ast_openstream_full: File vm-intro does not exist in any format
15:19.05GoshenJan 11 08:16:15 WARNING[9669]: file.c:820 ast_streamfile: Unable to open vm-intro (format ilbc): No such file or directory
15:19.10cj-rmmog_work: the only other thing I get is Mysql Realtime: Couldn't establish connection
15:19.22KattyMimmus: accent?
15:19.39*** join/#asterisk Prival (i=user69@MTL-HSE-ppp207971.qc.sympatico.ca)
15:19.39Goshendo I need to enable transcoding somewhere or something?
15:19.43*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
15:19.48mog_worktell you what
15:19.48hugo-v6hmmm got vodka but no redbull how sad.
15:19.51mog_workgo to logger.conf
15:19.59mog_workand turn on all options for console
15:20.00Mimmustzanger: I have a Digium card too but I tried also Sangoma
15:20.13mog_workand then start asterisk like this asterisk -vvvvdc
15:20.16ms345~stripdigit
15:20.21mog_workand put the out put to pastebin.ca
15:20.23mog_workill find it for ya
15:20.35PrivalCall parking... I can park a call that was received on a particular phone. But I can not park a call from the phone that initiated the call. Any hints?
15:20.39Hmmhesaysi play guitar a lot better when i drink energy drinks
15:20.40cj-rmmog_work: cheers dude
15:20.40MimmusKatty: uh?
15:20.41*** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com)
15:20.46*** join/#asterisk javar (n=javar@69.79.51.8)
15:20.48KattyMimmus: you said your english was bad.
15:20.58KattyMimmus: is that because of a heavy accent?
15:21.20MimmusKatty: very bad. tzanger doesn't even understand me
15:21.26silentfuryi realize this is an asterisk channel
15:21.38silentfurybut I'm working with Pingtel's sipXchange
15:21.42silentfuryand I need serious help.
15:21.53Goshenis it voicemailmain now in asterisk 1.2?
15:22.08silentfuryI have 2 polycom SIP 301 phones that won't register
15:22.25*** join/#asterisk Utah_Dave (n=boucha@0-1pool139-119.nas28.salt-lake-city1.ut.us.da.qwest.net)
15:22.32mog_workim sorry silentfury
15:22.37mog_workwhy not move to asterisk?
15:23.08silentfurygood question. i'll have to ask my boss sometime.
15:23.13Hmmhesaysbecause he's using sipXchange
15:23.17*** join/#asterisk SplasPood (i=jwb@206.252.198.100)
15:23.27mog_workyeah but obviously its not working
15:23.32Kattytuxinator_linuxM: lucky.
15:23.32mog_workand no one to help him
15:23.39mog_workwe can help him with asterisk
15:23.56*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:23.56*** mode/#asterisk [+o anthm] by ChanServ
15:23.56*** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
15:24.02Hmmhesaysdoesn't sipXchange use vxml?
15:24.09silentfurycan somenoe clear up a few things about the polycom IP phones at least?
15:24.19RoyKMimmus: i hardly understand myself, so it's ok
15:24.33sivanacan sip1 register on box A, have voicemail reside on box B, and still get WMI?
15:24.37PrivalCall parking... I can park a call that was received on a particular phone. But I can not park a call from the phone that initiated the call. Any hints anyone?
15:25.00mog_workyeah i might be able to do that silentfury
15:25.12*** join/#asterisk dippo_ (n=cwage@quietlife.net)
15:25.26dippo_is there a way to tell if there's an active call on a zaptel channel?
15:25.30Mimmusit would be beautiful if someone could point me to a consultant able to diagnose PRI problerms
15:25.34silentfurythe IP Gateway setting on the phones under network settings.. that's the Audiocodes gateway I have or the box that's hosting the exchange?
15:25.44Mimmusdippo_: show channel ... ?
15:26.19dippo_i don't see anything that indicates that there
15:26.23PrivalMimmus: Just did a PRI install the ther day using Sangoma. Might be able to help a bit...
15:27.15Goshenso the vm-intro is in the sounds directory as vm-intro.gsm.dist and needs to be renamed to vm-intro.gsm to work....
15:27.18Goshenthanks but no thanks
15:27.28MimmusPrival: thanks but it seems a subtle problem: Asterisk doesn't detect answer for some rare numbers
15:27.49mog_workid point it at exchange
15:27.58Broomsorry but another question, is WCT1 supported by Asterisk@home?
15:28.04*** join/#asterisk SERGEUS (n=s@195.112.98.13)
15:28.06mog_workyes
15:28.11PrivalMimmus: Ok... probably out of my league... :-)
15:28.19mog_workyes it does
15:28.24mog_workit has zaptel on it
15:28.32MimmusPrival: thanks anyway
15:28.33tzangerdamn what happened to the graphical manager interfaces
15:28.36tzangergastman is dead
15:28.39tzangerastman doesn't exist
15:28.48mog_workastman does...
15:28.49gammacoderI've got a couple Asterisk installs in small businesses with PRIs to the PSTN, now I've got a client who has a current contract with SBC for Centrex - can you use Centrex service with an Asterisk system?
15:28.53mog_workits in util
15:28.58mog_workneed to have libnewt installed
15:29.10mog_worki used it 2 days ago tzanger
15:29.22dippo_man the difference between ulaw and gsm quality is pretty drastic
15:29.26dippo_too bad the bandwidth used is too
15:29.29tzangerdippo_: well duh :-)
15:29.33dippo_:)
15:29.58dippo_is there a good compromise between them?
15:30.05silentfuryok, for the "Address" field in the Polycom phones under Line1 or Line2
15:30.10silentfurywhat do we put in there?
15:30.13Mimmusdippo_: peraphs g729?
15:30.14jbalcombdippo: just use gigabit ethernet and VLANs. ;)
15:30.16silentfuryi've been thrown into this without any help :(
15:30.23ms345anyone know how to get jbot to list the saved lore that starts with ~strip ?
15:30.30tzangermog_work: oh there it is
15:30.35tzangersorry I couldn't find it earlier :-)
15:30.41Broomthe autoconfig is putting this line on my zaptel.conf file, channel 1, WCT1, unhandled for now, for all channels
15:30.44Broomany ideas?
15:30.46ms345~stripdigits
15:30.49dippo_well this is for an IAX trunk
15:30.53dippo_over a cable modem
15:31.02jbalcombdippo: the g729 is great
15:31.05dippo_1Mbps up, so it's got SOME room, but more than 3 calls with ulaw would be tight
15:31.20dippo_is g729 the one that has some sort of weird restriction on it though
15:31.31Mimmusdippo_: exactly
15:31.38*** join/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:32.00*** part/#asterisk Lurr (n=pr0ph3t@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:32.00jbalcombdippo: you have to buy a license but its well worth it IMHO
15:32.02Mimmusdippo_: but ulaw is not 64 kbps? You have 1 Mbps. There is room for many calls
15:32.39silentfurythe Polycom phones claim they can't contact the boot server
15:32.42silentfurywhy is this?
15:32.48jbalcombdippo: 1Mbps of what? T1, DSL, Cable?
15:32.54Kattycan the ftp server ping the phone, silentfury?
15:33.00dippo_1Mbps of cable..
15:33.01[TK]D-Fendersilentfury : Are you provisioning your phones?
15:33.05silentfuryi am
15:33.06dippo_ulaw appears to be using significantly more than 64Kbps
15:33.09jbalcombsilentfury: have you contacted Polycom and asked them?
15:33.15dippo_more like 180-200Kbps according to pktstat
15:33.18[TK]D-Fendersilentfury : Then the server or credentials are wrong
15:33.21silentfuryPolycom won't talk to us since we bought it through a reseller
15:33.24iCEBrkr[TK]D-Fender: haha, yea they turned the PRI's up today!
15:33.26iCEBrkrfuckers
15:33.34Kattyjbalcomb: yeah i'm sure /that/ will help.
15:33.36silentfurythe Phones all get IP's from the exchange
15:33.46iCEBrkrjbalcomb: JIMBO?
15:33.48[TK]D-FenderiCEBrkr : So A104d up 100%?  EC & all?
15:33.50jbalcombiCEBrkr?
15:33.57jbalcombhaha.. whats up fella?
15:34.00iCEBrkrjbalcomb: GET THE HELL OUT
15:34.02iCEBrkrLOL
15:34.05iCEBrkrjbalcomb: Small world?
15:34.07silentfuryi'll try pinging the phones from the exchange
15:34.08iCEBrkrjbalcomb: what's up bro?
15:34.15jbalcombincreasingly small i'm afraid
15:34.28jbalcombicebrker: you running Asterisk down there?
15:34.31KattyiCEBrkr: oh, you.
15:34.34iCEBrkr[TK]D-Fender: it's all up and running, I just have to start making some test calls.
15:34.42iCEBrkrKatty: zip it, woman. :P
15:34.50KattyiCEBrkr: we have obligatory arguging to do now.
15:34.52iCEBrkrjbalcomb: for shizzy
15:34.53Katty...
15:34.58KattyiCEBrkr: oh do shut up.
15:35.00iCEBrkrKatty: Yes, I understand.
15:35.07iCEBrkrKatty: get do it..
15:35.07*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-225.claranet.co.uk)
15:35.09*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
15:35.09iCEBrkrerr to
15:35.12jbalcombicebrkr: excellent. contract me, im an 'expert'
15:35.17silentfuryok, I can't ping the phones from the exchange.
15:35.21silentfuryany ideas?
15:35.22KattyiCEBrkr: are you coming to etel?
15:35.26iCEBrkrjbalcomb: LOL
15:35.29[TK]D-Fendersilentfury : Tried plugging them in? ;)
15:35.36iCEBrkrKatty: WTF is etel and where is it?
15:35.37DaminiCEBrkr: Get back to work!
15:35.37silentfurythey are all plugged in
15:35.41Hmmhesaysbhwaaahahahahah "sip pass through nat and double nat easily"
15:35.43silentfuryhere's the thing
15:35.44iCEBrkrDamin: Boss gave me the day off LOL
15:35.44Kattysilentfury: can you login with the user/pass from a regular machine?
15:35.52DaminiCEBrkr: If you want a room at Etel, let me know..
15:35.54iCEBrkrDamin: Well, I'm taking tomorrow off.. I'm already in the office now..
15:35.57[TK]D-Fendersilentfury : well if you can't even ping them something is seriously wrong and nothing to do with VoIP
15:35.59Kattysilentfury: instead of a phone.
15:36.07Kattysilentfury: oh, you said can't.
15:36.13Kattysilentfury: are they on the same subnet thingy?
15:36.18*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi61.dialup.mindspring.com)
15:36.23iCEBrkretel? When? Where? How much?
15:36.26silentfuryno, the IP phones are on adifferent subnet
15:36.43silentfurythe router i have is 192.168.1.1
15:36.45jbalcombicebrkr: email me@work jbalcomb@imtco.com lets exchange detailed notes.
15:36.48KattyiCEBrkr: hold on.
15:36.50silentfurythe exchange is .55
15:37.03silentfurythe DHCP server i've setup should provision from 10.10.10.1 to .254
15:37.04koperniqsany ideas how i get useragent info for all peers in one list?
15:37.04KattyiCEBrkr: http://conferences.oreillynet.com/etel2006/
15:37.05*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfi61.dialup.mindspring.com)
15:37.13silentfuryand the phonse are picking up .254 and .253 ip addresses
15:37.16iCEBrkrjbalcomb: Are you flirting with me?
15:37.42[TK]D-Fendersilentfury : How are they being provisioned?
15:37.46Beirdooh boy, ASL goin on in the channel?
15:37.48jbalcomb*sniff* I miss you icebrkr.
15:37.49PrivalCall parking... Ok, found it... T versus t in the Dial command...
15:37.51silentfuryit should be automatically
15:37.57KattyBeirdo: you have such a pretty ASL.
15:38.03Beirdohehe
15:38.05KattyBeirdo: with it's lovely a and l, and curvacious little s.
15:38.08Beirdomy fiancee thinks so :)
15:38.10iCEBrkrSF!
15:38.13iCEBrkrUgh
15:38.14*** join/#asterisk fugitivo (n=ajf@201.255.177.172)
15:38.15Beirdomuhahaha
15:38.16iCEBrkrJAN?
15:38.21[TK]D-FenderAmerican Sign Language?
15:38.23Beirdogood morning, Katty :)
15:38.24KattyiCEBrkr: my room is already booked.
15:38.25iCEBrkrI'd have to order tickes today
15:38.28KattyBeirdo: morning (=
15:38.31Katty[TK]D-Fender: yes.
15:38.37silentfuryyou think it might work if i change the IP addresses to the same subnet?
15:38.38Katty[TK]D-Fender: i can speak to the deaf.
15:38.39iCEBrkrAND! I'd have to figure out how to schedule my PTO
15:38.42DaminKatty: I thought just you and I were sharing your room?
15:38.44Katty[TK]D-Fender: ...or sign.
15:38.44Beirdousing the powers of old Metallica to wake myself up this morning
15:38.53KattyDamin: no, i'm sharing with junky.
15:38.58DaminiCEBrkr: PTO? Tell him that this is critical for his business!
15:39.00ms345~stripleftdigit
15:40.01ms345~stripleftdigit
15:40.58DaminiCEBrkr: He needs to pay you for it! :)
15:40.58iCEBrkrLOL
15:40.58Daminkpfleming: Hey there..
15:40.59iCEBrkrDamin: I'm lucky this place pays for the coffee in the break room.
15:40.59dippo_g726 sounds pretty good and appears to be using around 100Kbps
15:40.59[TK]D-Fenderms345 : ${whatever:1}
15:40.59KattyDamin: think you can still recognize me? ;)
15:41.25DaminKatty: Probably not.
15:41.25Kattyexcellent.
15:41.25iCEBrkrhaha
15:41.53iCEBrkrThis can't be right
15:41.57iCEBrkrI found tickets for < $200
15:42.02Kattyit's probably one way
15:42.06iCEBrkrNope
15:42.15iDunnoone way ticket to nowhere?
15:42.15Kattymine's around 400
15:42.21Kattybecause i refuse to hop flights.
15:42.43Beirdoeek
15:42.51Broomany ideas on why A@H would not use the Digium Wildcard TE110P T1/E1 card?
15:42.52iCEBrkrOh, I got crazy layovers in here.
15:42.57Katty;)
15:43.01iCEBrkrBroom: cuz A@H sucks?
15:43.05*** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com)
15:43.08KattyiCEBrkr: hush.
15:43.12jbalcombicebrkr: you buy me one of those tickets too while you're at it eh?
15:43.13Ferrarigood morning all
15:43.15iCEBrkrBroom: It'll work, just have to make it work :P
15:43.18KattyiCEBrkr: we do not need any $stuff sucks comments.
15:43.27jbalcombicebrkr: ps, make sure mines for CLE
15:43.31iCEBrkrjbalcomb: Dude, I can hardly pay atttention, how am I gonna pay for you??
15:43.35Ferrariencountering a strange issue and wondered if anyone else had the problem.
15:43.46jbalcombicebrkr: credit cards
15:43.46Ferrari2 polycom IP500s call each other
15:43.50iCEBrkrjbalcomb: What's that?
15:43.52Ferrariasterisk show the calls talking
15:43.53cj-rmmog_work: Cheers man, I got it working based on that debug output!  My mysql database only accepted connections from localhost, but it was trying to connect from localhost.localdomain
15:43.59DaminiCEBrkr: You should go. Jan 24th, the first day of the conference, is my birthday!
15:44.01iCEBrkrjbalcomb: you think anyone would be crazy enough to give me a CC?
15:44.02Ferrariif one of the phones looses network connection
15:44.14iCEBrkrDamin: I'm thinking about it. It's only 3 days.. I can spare that, I'm thinking
15:44.18Ferrariasterisk continues to show the channels talking
15:44.21DaminiCEBrkr: And we could party like it's my birthday! :)
15:44.29iCEBrkrLOL
15:44.37jbalcombicebrkr: you do have a point there but i assume is someone would hire you then surely the cc can happen too. ;)
15:45.08gammacoderBroom: I've got A@H working with a TE110P
15:45.12DaminiCEBrkr: Let me know.. It is something you shouldn't miss! :)
15:45.20DaminiCEBrkr: Cleveland will REPRESENT!
15:45.25jbalcombferrari: CNG traffic maybe? how long does it take to recognize the drop?
15:45.32iCEBrkrDamin: 216 in the hizzous!
15:45.39Ferrariit never closes down the channel
15:45.47Ferrarii left it like that overnight
15:45.54Ferrariand this morning it still showed up
15:46.02jbalcombicebrkr is NO LONGER 216. excommunicated.
15:46.12FerrariVersion 1.0.10
15:46.26Daminjbalcomb: icebrkr can NEVER be excommunicated from 216..
15:46.28iCEBrkr727 in the hizzous!!
15:46.45iCEBrkrjbalcomb: I owned 216, LOL
15:46.49Daminjbalcomb: He still has a house in 216! ;)
15:46.50jbalcombits been done. I check with Mr. Mayor and signed off. sorry.
15:46.52FerrariI have qualify turned on
15:46.56iCEBrkrjbalcomb: Because of Damin, of course..
15:47.00DaminAlright.. I have to get to work.. :)
15:47.03Ferrariand asterisk sees that the one extension is no longer available
15:47.08iCEBrkrMe too. I gotta find tickets :P
15:47.19Ferrarithen when the phone reconnects asterisk show that the phone in now available
15:47.27*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:47.27*** mode/#asterisk [+o anthm] by ChanServ
15:47.31*** join/#asterisk obiwanmikenolte (n=obiwanmi@63.150.226.34)
15:47.39Ferrarihowever show channles continues to show that bothe sip channels are talking when they are not
15:47.43jbalcombicebrkr: perhaps you own the space at the feet of the one who owned 216.
15:48.11iCEBrkrlol
15:48.14*** join/#asterisk nguyep (n=chatzill@64.34.203.231)
15:48.29iCEBrkrI got a few 216 DIDs, and my cellphone is still 216..
15:48.31iCEBrkr:)
15:48.36iCEBrkrI'm hang'n on to my roots, yo
15:48.42jbalcombhaha indeed. anyway, i gotta get back to work and i hate when channels are flooded with irrelevant traffic.
15:48.48iCEBrkrlol
15:48.58iCEBrkrI'm gonna try to find some tickets for the etel thing.
15:49.18jbalcombicebrkr: you email me and ill see you here later to discuss internal echo, low traffic stats, drops on external calls, poor CDR stats, etc.
15:49.19*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
15:49.40iCEBrkrjbalcomb: man you got some issues.
15:49.55*** part/#asterisk nguyep (n=chatzill@64.34.203.231)
15:50.03*** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com)
15:50.20jbalcombicebrkr: yeah, man implements Asterisk and leaves 4 weeks later, half-assed admin takes over, four weeks later im hired.
15:51.08iCEBrkrHow do you get yourself in these situations
15:51.15Kattyvery carefully.
15:51.26Kattybut it's not /your/ job to run him into the mud.
15:51.34KattyiCEBrkr: saavy?
15:52.00KattyiCEBrkr: (you better talk back or this arguement won't go anywhere)
15:52.00jbalcombindeed. i think its cause i know how to do so much but have trouble keeping jobs. :/
15:52.24jbalcombhaha.. 'saavy' i love that word and 'gumption' too.
15:52.52iCEBrkrKatty: I've known jbalcomb for a VERY long time. I'll run him in the mud if I want :P
15:53.01KattyiCEBrkr: then can i help too?
15:53.12[TK]D-Fenderhe's Gump, he's Gump, he's Gump, is he inbred?
15:53.14iCEBrkrKatty: Pfft, no. He's my friend.. not yours.
15:53.26Kattywell then maybe fender will let me run him into the mud.
15:53.29*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
15:53.35iCEBrkrKatty: Yea, but fender likes it
15:53.44[TK]D-FenderKatty: Sorry, not my fight :)
15:53.45Kattywell that's the whole point.
15:53.55*** join/#asterisk _Sam-- (i=sam@phone2.kneedraggers.com)
15:53.56Kattyrunning into the mud is fun. it's skwishy.
15:54.04quadrataheh
15:54.15_Sam--if a caller is in the call queue and you are playing MOH...how do you interupt that and play another message?
15:54.25Katty[TK]D-Fender: you don't want to fight?
15:54.27_Sam--like while they are in the queue "did you know you can also check your order status at ....."
15:54.28Katty[TK]D-Fender: oh come on!
15:54.39tmccraryIf I register two phones on an asterisk, how do i get asterisk to ring both phones
15:54.41jbalcombIRC = Insane Residents of Crazytown?
15:54.51[TK]D-FenderKatty: I never said that.  Just not THAT fight :)
15:54.57Katty[TK]D-Fender: oh, right.
15:55.01Ferraridial(Device1&device2)
15:55.07ModcutsWhat would you recommend for setting up 4 incoming lines for a business asterisk box, Using ISDN or all using sip?
15:55.11iDunnoKatty: there are far better things to do in the mud ;)
15:55.31KattyiDunno: none of which i'd do.
15:56.01KattyHmmhesays: what's the name of that band with the kitty kitty song?
15:56.04iDunnoKatty: not make mud castles?
15:56.08KattyHmmhesays: my brain is not functioning
15:56.09HmmhesaysHot action cop
15:56.12Kattythanks.
15:56.17Hmmhesaysnp
15:56.20quadrataiDunno, I assumed that's what you were talking about
15:56.22[TK]D-FenderModcuts : ISDN.  I don't like basing a business' functioning on too many layers of liability.  Outside VoIP is one too many.
15:56.51[TK]D-FenderVoIP is at its best in a controlled environment like a LAN.
15:57.04*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:57.18*** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
15:57.52[TK]D-FenderiDunno : A facial perhaps?
15:57.58*** join/#asterisk linville (n=linville@azure.tuxdriver.com)
15:58.01quadrataisn't there a song about making mud castles?
15:58.17[TK]D-Fenderquadrata : The Hendrix....
15:58.33quadrataall along the MudTower?
15:58.51Hmmhesaysa kickass song by hot action cop is what she was thinking about
15:59.19[TK]D-Fenderquadrata : "Even castles made of sand flow to the sea eventually?"  Something like that in a song who's title escapes me.
15:59.38dippo_well I figured out why my zap channel is not answering calls: the phone line is dead
15:59.45dippo_turns out you need a live phoneline. who knew?!
16:00.28quadrata[TK]D-Fender, the song I had in mind was a bit more... cheesy
16:01.34KattyHmmhesays: you have that posted anywhere?
16:01.54Hmmhesaysthat song?
16:01.57KattyHmmhesays: yes.
16:02.22Hmmhesaysyeah I probably have it somewhere
16:02.57*** join/#asterisk harryk (n=me@harry.org.ua)
16:03.13*** join/#asterisk shido6 (n=bleh@i216-58-29-215.cybersurf.com)
16:03.14KattyHmmhesays: :>
16:05.09*** join/#asterisk gugaiz (n=gugaiz@host197.200.61.156.ifxnw.com.ar)
16:05.53gugaizhi, i need to know when the call start and when the call end, in realtime
16:05.59gugaizwith asterisk
16:06.13gugaizis that possible?
16:06.57fourcheezegugaiz: everything is possible
16:07.31*** join/#asterisk brc_ (n=spack_@pdpc/supporter/basic/brc)
16:07.34*** join/#asterisk SAM007 (i=akhq@82.215.66.105)
16:09.20tmccrary<PROTECTED>
16:09.44tzangertmccrary: read the asterisk handbook
16:09.47tzangerthis is explained quite clearly
16:09.50[TK]D-Fendertmccrary : Dial(SIP/phone1&SIP/phone2)
16:10.28SAM007ewrwerwer
16:10.32SAM007uouiouioui
16:11.11RoyKapowiehjrfapowsidfjcapowiehjrfapowsidfjcapowiehjrfapowsidfjc
16:11.26sivanacan sip1 register on box A, have voicemail reside on box B, and still get WMI?
16:11.34*** join/#asterisk green_earz (n=peter@213-232-83-67.dsl.prodigynet.co.uk)
16:11.56*** join/#asterisk BladeRunner05 (n=feelme@81.174.56.54)
16:12.03BladeRunner05hi all
16:12.20BladeRunner05does * support grandstream GPX-2000 ?
16:12.43gugaizfourcheeze: ok, I know but what is the way
16:12.44MimmusBladeRunner05: of course
16:13.07tmccraryi want the phones to have the same extension
16:13.09fourcheezegugaiz: when you say you want to know, what do you want to happen?
16:13.36BladeRunner05mimus: have u tried ?
16:13.53[TK]D-Fendertmccrary: exten => 123,1, Dial(SIP/phone1&SIP/phone2).  There.  person dials 123 adn rings both.
16:13.55fourcheezeBladeRunner05: it's well documented
16:14.04fourcheezeand it supports the flashy lights on it too
16:14.10gugaizfourcheeze: for example, execute a program
16:14.15*** join/#asterisk nguyep (n=chatzill@64.34.203.231)
16:14.17tmccrarybut if both phones are SIP/1011, that works?
16:14.32ruud_orgno
16:14.34sivanawhat about wav
16:14.50fourcheezegugaiz: you would just execute that inthe dialplan before you dial
16:15.02MimmusBladeRunner05: no but it is a frequently used SIP phone
16:15.05malverian[work]Is there a channel flag to tell me if a call is an assisted transfer?
16:15.09*** join/#asterisk PMantis_C (n=sswitzer@66.251.89.34)
16:15.44[TK]D-Fendertmccrary : SIP/1011 can't be in 2 places at the same time with *.
16:15.46fourcheezefourcheeze: you might need to write an AGI script to make the call and do the necessary DB work, or you might want to make a custom module
16:15.48malverian[work]I'm working on a module for asterisk, and I know i can check if it is a BLINDTRANSFER, but I can't differentiate between assisted transfer and a normal dial.
16:15.50iCEBrkrummm.
16:15.53fourcheezeoops gugaiz ^^
16:15.57iCEBrkr<PROTECTED>
16:16.00iCEBrkrIs that typical?
16:16.04tmccraryah, that sucks, thanks for the help Fender
16:16.09iCEBrkrLike just randomly restarting?
16:16.14*** join/#asterisk spunz (n=spunz@h081217096096.dyn.cm.kabsi.at)
16:16.16sivanaiCEBrkr: ya
16:16.21Mimmusif I dial an extension by a .call file, how can I check for result (BUSY, max temmpts, NOANSWER, etc)
16:16.30iCEBrkrok
16:16.31*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
16:16.33[TK]D-Fendertmccrary : Why would be trying to make 2 phones act as one?
16:16.43sivanaiCEBrkr: I get that periodically as well
16:16.56*** join/#asterisk HamYai (i=HamYai@125.24.3.126)
16:17.03sivanaiCEBrkr: you'll notice that it's only the channels that are not in use
16:17.10gugaizfourcheeze: ok, but I need to know only if the other side answer the call
16:17.16tmccraryBecause I have one person who moves from his apartment to office and I want both phones to work exactly the same. So if someone calls him, it doesn't matter where he's at.
16:17.40[TK]D-Fendertmccrary : it can still ring both at the same time, just as differnt ext's.
16:17.51HamYaihow is everyone detect the start of a call thru POTS?
16:17.57RoyKzoa: ping
16:17.57[TK]D-Fendertmccrary : Doesn't have to be the same as log as they both ring when they are supposed to.
16:18.07[TK]D-Fendertmccrary : other than that, whats the difference?
16:18.32tmccraryit's not as clean
16:18.50[TK]D-Fendertmccrary : how so?  its a few extra chars on a dial line.
16:18.55tmccraryI have to make special rules just for that user, to make them use the same voice mail, etc
16:19.11sivanatmccrary: macros
16:19.20[TK]D-Fendertmccrary:  copy & paste.  2 minute job TOPS.
16:19.33tmccraryI know it's not difficult to do, it's just not a clean solution
16:19.35RoyKwhy does a stun server require two IPs?
16:19.45tmccraryI'll have an extra extension that doesn't really do anything.
16:19.46RoyKi mean,  what does it do with them?
16:19.56tmccraryI'll have to looking into adding that functionality to asterisk
16:19.59*** join/#asterisk IOscanner (n=IOscanne@c-24-0-183-125.hsd1.tx.comcast.net)
16:20.08[TK]D-Fendertmccrary: definately not worth complaining about.  * doesn't support shared line appearances. Oh well.
16:20.20gugaizfourcheeze: I'm trying integrate radius server with asterisk, but I can't do it
16:20.32[TK]D-Fendertmccrary : sue it doesn... it rings at the same time!  They could be on seperat calls.  there's added value.
16:20.37tmccraryI'm not complaining, you're asking me why I wanted that functionality
16:20.45Mimmushow can I check for result (BUSY, max temmpts,
16:20.47Mimmus+NOANSWER, etc)
16:20.50tmccraryIt's better/cleaner the other way. This way is a hack.
16:20.51[TK]D-Fendertmccrary : Sure, shared appearances would be nice...
16:20.53Mimmusof a call file?
16:21.05[TK]D-Fendertmccrary : but entirely non-necessary
16:21.21sivana[TK]D-Fender: is it even possible?
16:21.38iCEBrkr[TK]D-Fender: This is pretty sweet. I didn't even have to do anything when they made these things go live.. The card just came up and asterisk worked.. Which is cool cuz if it didn't work, I wasn't sure how else to use wancfg to configure this card.. LOL
16:21.51[TK]D-Fendersivana : not yet, thats the point
16:22.08[TK]D-FenderiCEBrkr : Confirmed EC on it?
16:22.11sivanaI mean with the SIP protocol
16:22.13iCEBrkrNot yet.
16:22.18[TK]D-Fendersivana : yes
16:22.23iCEBrkrI've been busy doing some other catch up work
16:22.27tmccraryIt should be, SIP is just like a messaging protocol
16:22.29sivanaI see... just not supported in * is all
16:22.30tmccraryIt's not magic or anything
16:22.40[TK]D-Fendersivana : Linksys, Polycom, and others support it for a while now.
16:22.47[TK]D-Fendersivana : correct
16:22.53sivanaI see.. I have Polycom phones
16:23.14sivanaand that the number one complaint I've had with businesses
16:23.17[TK]D-Fendersivana : Same here.  I'm scoping out doing a mass-upgrade to SIP 1.6.3 soon.
16:23.21IOscannerWhen I dial into the PBX from remote and check voice mail the messages are very soft I can hardly hear them.  I can hear the mailbox options just fine.  Is there a way to boost the audio level of recorded messages.  Or could this be a bridged audio problem.  I can hear them when in the office.
16:23.25sivanabeing able to see if another phone is on the line
16:23.27shido6you can do that with voip providers, too
16:23.33shido6and dial multiple PSTN numbers
16:23.34[TK]D-Fendersivana : You CAN do that already...
16:23.42shido6whoever answers first will get the call
16:23.46sivana[TK]D-Fender: how?
16:23.51shido6or 1 after the next
16:23.55shido6or a ring group
16:24.09[TK]D-Fendersivana : read the WIKI for "polycom presence".  I do it a lot here.
16:24.13sivanaok
16:24.21sivanadoes the Linksys 841 do it?
16:24.21[TK]D-FenderQuick fix for it.
16:24.44[TK]D-Fendersivana : No, the SPA's only support "shared appearances", not straight presence.
16:24.48[TK]D-Fender(IIRC)
16:25.01sivanahrm
16:25.03[TK]D-FenderI use an SPA-941 at home as well
16:25.40fourcheezegugaiz: what do you want asterisk to do that's connected with what radius does?
16:27.12*** join/#asterisk thieums (n=darkmind@bea75-1-82-234-122-35.fbx.proxad.net)
16:27.25shido6ahh da 9 fourty-one
16:28.02*** join/#asterisk Cyon (n=cyon@216.179.31.166)
16:28.06CyonHey
16:28.18Cyon_Vile:  Around?
16:29.19Kattybkw_: looks like i'm flying into oakland.
16:29.23sivana[TK]D-Fender: so with the 501 where you have 3 appearances, 1 is yours and the other 2 could be colleagues
16:30.40[TK]D-Fendersivana : Yup.  I'm running all 60x here so I split them 3/3.
16:30.56dippo_i am having troubles with calls being dropped.. I see messages in the logs like
16:30.59dippo_"Bridge stops bridging channels"
16:31.00[TK]D-FenderAndthen there's the attendant module.
16:31.06dippo_and then it hangs up
16:31.08Cyon_Vile:  Oh well, maybe you'll see this when you get back.  It appears the T.38 patch had some bugs introduced when the newest copy was released, but unfortuantely as you know my experience is too limited to try and resolve it myself.
16:31.19sivana[TK]D-Fender: I'll have to test this further, I guess you need the hint priority
16:31.24sivanathanks for the info though
16:31.52dippo_er, that's after the IAX trunk missed a frame
16:31.56[TK]D-Fendersivana: Naturally.  a very quick setup.  2 parms to change in sip.cfg, hints to add, then tag your buddies int eh contact list.
16:32.03Cyon_Vile:  So, I tossed up a copy of ser, pointed my sipura at it, and had it point to the cisco...it's not working any more reliably, but I expect it's config related, so since you already had this implemented, I was going to ask you about it.
16:32.38CyonOf course anyone else here who is running ser to provide for reliable faxing, or has another way to provide reliable faxing...that would be great.  ;-)
16:33.28*** join/#asterisk thomastim (n=anonymou@ntserver01.thomastonschools.org)
16:33.50gugaizfourcheeze: excuse me the delay
16:34.05wunderkinteehee
16:34.31gugaizfourcheeze : if asterisk is connected to the radius server, i can get start/stop packages from radius server
16:34.31*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
16:34.31shido6now Im hungry j
16:34.38bonhow come
16:34.42boni don't have iax2 support
16:34.42bon?
16:34.46bonin my asterisk
16:34.47bonand how to change it? :P
16:34.51*** join/#asterisk gryzor (n=gryzor@fydelkass.inl.fr)
16:35.10gryzorgreetings
16:35.18gugaizfourcheeze: and the destination number, call id, auth user..
16:36.22thomastimwhy would you want to ban me?
16:36.24thomastimi'm a nice guy
16:36.38fourcheezegugaiz: I think you could use asterisk realtime to log to postgres
16:36.47wunderkinits the disturbing hostname
16:36.58thomastimlol it's running linux
16:36.59fourcheezegugaiz: I'm not sure exactly when the logging there happens
16:37.02fourcheezesomeone will know
16:37.22wunderkinthats even worse
16:37.39fourcheezegugaiz: even if it only logs at the end of the call, you can make use of that
16:37.42jsharpMost likely happens at the end of the call.
16:37.55fourcheezegugaiz: by putting your own record in your own db at the start
16:38.09thomastimwunderkin: what does your company run?
16:38.22fourcheezegugaiz: it all depnds how accurate you need it to be
16:39.07wunderkini mean its worse that a linux box is named ntserver
16:39.11fourcheezegugaiz: http://software.sunsaturn.com/ might help you once it's got some more development
16:39.25fourcheezeif syle is around he could tell you
16:39.48gugaizfourcheeze: I need in realtime, when the other side answer the call I need to know all information about the call
16:40.10fourcheezegugaiz: then you need to find a module that can do that
16:40.18fourcheezeor write one
16:40.38*** join/#asterisk interp1 (i=interp@loves.voltshells.com)
16:40.40jsharpapp_radius
16:40.43gugaizfourcheeze: And I need the user register, before can make a call
16:41.14fourcheezegugaiz: that all depends on your dialplan
16:41.19gugaizjsharp: app_radius exists?
16:41.21fourcheezei.e. your extensions.conf
16:41.37*** join/#asterisk Samoied (n=Samoied@200.247.141.111)
16:41.38jsharpappradius.minitelecom.org
16:41.54gugaizForbidden
16:42.03jsharpBah.  bastard.
16:42.09gugaizsay the bowser..
16:42.20bonhm
16:42.22*** join/#asterisk wwolfe (n=wwolfe@68-250-139-209.ded.ameritech.net)
16:42.37bonany clue about this? http://pastebin.com/501007
16:42.53gryzorif i know nothing about asterisk/voip/pbx, but know very well about linux/system/networking, will i have a hard time understanding/specifying/deploying asterisk? or is this question already bloated? :)
16:43.14bongryzor: well, not hard time, but won't go like 1-2-3
16:43.14bon:
16:43.18jsharphttp://www.dynx.net/ASTERISK/misc-progs/appradius/
16:43.20bonor at least is for me now so
16:43.35jsharpWhether that will build with latest asterisk, I don't know.
16:43.43gryzorbon: very approxcimate? how long will it take to understand things?
16:43.49*** join/#asterisk opus_ (n=opus@dahphish.org)
16:43.51opus_hello
16:43.54bongryzor: months :)
16:43.59opus_is there a way I can set a DID busy on my PRI?
16:43.59gryzorhehe
16:44.01bongryzor: to be honest, having it up and running
16:44.06bongryzor: is the easiest part :)
16:44.19bongryzor: but to really understand what's going on, it'll take ages
16:44.20gugaizfourcheeze: yes but I can't have the tools
16:44.24RoyKopus_: exten => somenumber,1,Busy
16:44.24gryzorok
16:44.25RoyK:P
16:44.40interp1How might I play music via MP3Player while I try to connect the call to a sip extension?
16:44.46bonhm, but anyway i am still having problem loading chan_iax2 :(
16:45.04*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
16:45.34jsharpbon:  Is that a fresh build/checkout of sources or did you upgrade some source code?
16:46.07gugaizjsharp: thanks, I will try this
16:46.11bonjsharp: no clue :)
16:46.14*** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net)
16:46.16bonbut it's not fresh 1.2.1 afaik
16:46.20gryzorbon: thanks, i think i see what you mean :)
16:46.28bonbtu as far as i'm concerned, i didn't upgrade since i installed
16:46.33jsharpTry it with a fresh 1.2.1.  Looks like you may have some code version mismatch.
16:46.37bonwould it be sufficient just to recompile from source?
16:46.48interp1or how can I play music while trying to connect to the sip extension?
16:47.05bonjsharp: but that would be problematic
16:47.13bonjsharp: i need support for res_mysql.so
16:47.14bonetc.
16:47.15bon:/
16:47.27*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
16:47.30jpabloHi
16:47.42jpabloanyone can recommend a good gsm<->asterisk gw?
16:47.51jsharpOh.
16:47.56bonhm
16:48.05bonas i am checking the full_log
16:48.16gambolputtyWhen is a congestion state usually encountered when making a call?
16:48.37*** join/#asterisk slak- (i=slak@rewted.biz)
16:48.56slak-hey, my new-vm tone no longer works
16:49.05bonjsharp: as i am checking the full it worked in the beginning
16:49.05bon:)
16:49.06slak-you know when you pickup the phone and the tone skips
16:49.10bonthis seems to be 1.2.0-beta1
16:49.13slak-did something change in 1.2?
16:49.21*** join/#asterisk SwK[Work] (n=SwK@64.89.118.139)
16:49.29*** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl)
16:49.32TagorHi
16:49.53TagorIs someone here interested in configuring an Asterisks server for some money?
16:50.33slak-Tagor: ok
16:50.39slak-how much
16:50.52*** part/#asterisk opus_ (n=opus@dahphish.org)
16:51.21*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
16:51.31TagorGive me a quote. I have a few phone numbers and a few software clients that I want to be configured. Also I need a menu on one of the numbers
16:52.01slak-100$/hr
16:52.08[TK]D-FenderTagor : PM
16:52.18TagorOne second need to identify to PM ;)
16:52.25[TK]D-FenderPrivate Message
16:52.28Tagorslak- >> How long do you think that will take?
16:52.36bon[TK]D-Fender: where? :)
16:52.38slak-[TK]D-Fender: i have a family of 10 and 8 starving children
16:52.42bonlol
16:52.56[TK]D-Fenderslak- : So you're eating all the food huh? :)
16:53.00slak-lol
16:53.14slak-Tagor: what kind of hardware do you haver
16:53.15slak-Tagor: what kind of hardware do you have
16:53.28slak-just software phones?
16:54.21slak-[TK]D-Fender: after upgrading to 1.2.1 new voicemail tone (the skipping tone) dissapeared
16:54.26slak-any clue?
16:56.17Tagorslak- >> Just software phones
16:56.33slak-who is your sip provider
16:56.47slak-whats platform will asterisk be running on
16:56.48gammacodercan you use a Digium TDM2400 series card to accept Centrex lines from phone company for use in Asterisk?
16:56.56Tagor12connect (a dutch provider), slak-
16:57.16slak-Tagor: linux?
16:57.50TagorYes, slak-
16:57.58TagorAsterisk is already installed :)
16:58.03*** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de)
16:58.14TagorYou get SSH access
16:58.38[TK]D-Fenderback
16:58.53brad_msswgammacoder: centrex is isdn, right ?? TDM2400 doesn't support ISDN
16:59.06iDunnoffs. the *one* thing that was bloody working today...
16:59.10iDunnohas died in a heap
16:59.14slak-Tagor: i could do it, but man that sounds so really simple just do it yourself
16:59.14*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
16:59.24slak-ill take maybe an hour for me to do
16:59.24*** join/#asterisk Strom_C (n=strom@198.172.114.2)
16:59.27iDunnothey've killed our ISDN line, the damned hippys.
16:59.44asteriskmonkeydoes anyone have any simple php agi scripts i could use for examples?
16:59.54Tagorslak-> I have to go now, I will contact you in 45 minutes, ok?
17:00.00*** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net)
17:00.01[TK]D-Fendersure
17:00.03Strom_Chello
17:00.24slak-ok
17:00.27asteriskmonkey[TK]D-Fender: you have php agi examples :) ?
17:00.36gammacoderbrad: not really sure http://en.wikipedia.org/wiki/Centrex
17:00.39slak-[TK]D-Fender: new vm tone skip disapeared,,any ideas?
17:00.42slak-what to check
17:01.19[TK]D-Fenderasteriskmonkey : Haven't done AGI yet...
17:01.33[TK]D-Fenderslak- : Dissappeared from where?
17:01.35asteriskmonkeydarn it... ANY ONE HERE DONE AGI WITH PHP YET?
17:01.52slak-tk: when i get a new vm, my dialtone doesnt skip anymore
17:02.00[TK]D-FenderWhat kind of phone
17:02.01slak-which would let me know i have a vm
17:02.05mog_workyes asteriskmonkey
17:02.06iDunnoasteriskmonkey: PHP IS EVIL, DON'T FUCKING SHOUT, HOW ABOUT ASKING ON THE FUCKING MAILING LIST OR USING GOOGLE.
17:02.14slak-tk: just regular phone attached to ata, the ata config didnt change
17:02.19mog_workbut if you type in all CAPS people whill not want to help you
17:02.21slak-asterisk got ugraded to 1.2.1
17:02.25silentfuryhrm
17:02.29[TK]D-Fenderslak- : Did you just upgrade to 1.2.1?
17:02.30asteriskmonkeyGoogle dosnt procude and good results, i will ask on the mailing list i guess
17:02.35silentfurynow i'm getting can't load macaddress.cfg on my 2nd polycom phone :(
17:02.41mog_workwhats your question
17:02.44slak-tk: not just now, but thats when it stopped working
17:02.47IOscannerWhen I dial into the PBX from remote and check voice mail the messages are very soft I can hardly hear them.  I can hear the voicemail prompt just fine.  How can I fix this?
17:03.10iDunnoalso: don't expect immediate responses, people have things like *work* and *lives*, just because they're in channel doesn't mean they're damned well reading it ;)
17:03.19[TK]D-Fenderslak- : make sure you are using "mailbox=abc@[context]" in your SIP.conf
17:03.31slak-okay..anything else
17:03.58tuxinator_linuxMlittle naps don't hurt
17:04.06mutwhats a good home wifi router? something that gets some decent range, lookin for $30-$50
17:04.10malverian[work]Anyone here worked on asterisk modules before?
17:05.29^HowleriDunno: what are *lives*?
17:05.47iDunno^Howler: buggered if I know, but I hear some people have them.
17:06.03malverian[work]Nevermind.. figured it out.
17:06.12tuxinator_linuxMI have my coding and my wife, a little time for anything else
17:06.16*** part/#asterisk secure75 (n=mic@ppp-62-245-162-105.mnet-online.de)
17:06.28*** join/#asterisk strib (n=ads@wompom.dur.ac.uk)
17:06.29^HowleriDunno: oh, alright. I've heard such rumors as well, thought you might be able to shed some light on it.
17:07.01brad_msswgammacoder: looks like they use centrex as a fairly generic term as a hosted pbx system of sorts by the telco
17:07.10stribCan anyone here recommend a nice earphone/microphone pair I can plug into my PC?
17:07.28stribOr at least tell me what they're called.
17:07.31anthmlogitech usb
17:07.32brad_msswgammacoder: if it's delivered over PSTN, then I see no reason why the TDM2400 with a bunch of FXOs wouldn't work
17:07.43[TK]D-Fenderstrib : Take your pick from Polycom, Logitech, Labtec.
17:07.44stribanthm: does it work with Linux?
17:07.54brad_msswgammacoder: though it looks like some may be delivered via isdn from other stuff I saw, so be careful
17:07.54anthmshould
17:07.59stribWhat's the generic name for these things?
17:08.12anthmusb headset
17:08.15stribThanks
17:08.41anthmnp
17:08.48gammacoderbrad_mssw: yep, I'm tring to get a clarification from the phone company - thx
17:09.13stribiDunno: how do you hear the ringer?
17:09.16Strom_Cgammacoder, if you're going to get 24 incoming lines, do yourself a favor and just get a PRI
17:09.21anthmyou will find tremendous improvement when using voip on a dedicated soundcard aka usb headset
17:09.26iDunnostrib: I don't :)
17:09.32stribiDunno: hehe
17:10.09Strom_Cgammacoder, centrex is a name telcos use for PBX-like services served off the main central office switch
17:10.12gammacoderStrom_C: yep - I've got PRIs for all my other clients, but this one has 25 months left on a contract for 11 Centrex lines
17:10.15Strom_Cyour lines are put in a centrex group
17:10.43Strom_C(sorry if im telling you things you know already; i came into this mid-conversation)
17:11.07iDunnoahh. erm.
17:11.18iDunnooops. seems that someone failed to pay the phone bill.
17:11.22[TK]D-Fenderlong-term telco contracts = BAD
17:11.35strib[TK]D-Fender: not if you're the telco ;)
17:11.58gammacoderStrom_C: thx - just striggling with this existing contract - hopefully the phone company will let them switch to pri
17:13.24*** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com)
17:15.14Strom_Cgammacoder, if you have a contract, I'm sure they would be willing to "renew" the contract early if it would save them money
17:15.28*** join/#asterisk crich1999 (n=crich@port-212-202-0-102.dynamic.qsc.de)
17:15.57gammacoderdamn contracts
17:16.54Strom_Chmm, I can't figure this out
17:17.26Strom_Cmy client's box has this problem where it will sometimes start filling its logs up to the tune of a gigabyte every three minutes with messages like this:
17:17.51Strom_CJan  9 11:59:04 DEBUG[5856] chan_zap.c: Exception on 57, channel 39
17:17.52Strom_CJan  9 11:59:04 WARNING[5856] chan_zap.c: We're Zap/39-1, not Zap/1-1
17:19.06*** join/#asterisk zoa2 (n=kkk@pirus.securax.be)
17:19.53tuxinator_linuxMzoa2, there is two zoa's?
17:20.01zoa2ping timeout
17:20.05Qwellscary ;]
17:20.05*** join/#asterisk roulduke_ (i=ziyrupus@p508D263F.dip0.t-ipconnect.de)
17:21.00*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
17:21.16*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
17:21.32shmaltzhas anybody noticed something wrong with Sipura and Polcycoms using reinvites in 1.2.1?
17:22.22*** join/#asterisk sysdebug_ (n=sysdebug@200.163.193.247)
17:23.02*** join/#asterisk chapeaurouge (n=chap@85.201.81.201)
17:23.47Strom_Calso, from what my client tells me, that problem is also accompanied by a memory leak
17:24.42*** part/#asterisk tmccrary (n=tmccrary@68.78.185.254)
17:25.11*** join/#asterisk denon (i=denon@synapse.subneural.net)
17:25.11*** mode/#asterisk [+o denon] by ChanServ
17:26.45*** join/#asterisk harryk (n=me@harry.org.ua)
17:27.53[TK]D-Fendershmaltz : reinvites = bad
17:28.16shmaltz[TK]D-Fender, can you explain?
17:28.23justinureinvites are cool
17:29.23*** part/#asterisk darkskiez (n=darkskie@194.247.78.146)
17:29.50*** join/#asterisk apardo (n=apardo@87.218.44.73)
17:30.07justinushmaltz: what kind of problem?
17:30.21zoa2strom_c, what is the problem ?
17:30.46Strom_Cmy client's box has this problem where it will sometimes start filling its logs up to the tune of a gigabyte every three minutes with messages like this:
17:30.46*** join/#asterisk Math` (n=Math_@toronto-HSE-ppp4123121.sympatico.ca)
17:30.51shmaltzjustinu, when xfering from polycom to sipura there is only one way audio (there is no nat)
17:31.01Strom_CJan  9 11:59:04 DEBUG[5856] chan_zap.c: Exception on 57, channel 39
17:31.03Strom_CJan  9 11:59:04 WARNING[5856] chan_zap.c: We're Zap/39-1, not Zap/1-1
17:31.03Math`is there any way to disable zaptel's hangup detection? (for fxo lines)
17:31.19justinushmaltz: can you turn on sip debug and pastebin the sip messages?
17:31.22*** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com)
17:31.29justinushmaltz: i'm somewhat familiar with SIP
17:31.33shmaltzjustinu, ok, just a min
17:31.37*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@host.190.115.68.195.rev.coltfrance.com)
17:32.44*** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it)
17:32.48zoa2that comes with a memory leak ?
17:33.14Strom_CI think so
17:33.21Strom_CI don't know if the two are associated
17:33.24Strom_Cbut they seem to be
17:33.27justinustrom_c: that sounds somewhat serious
17:33.28fndudeHi all. When I try to transfer a call to park, I can press # and get the transfer message, but any number I type gets a "not valid extensions". I have added exten 700 to my dialplan and parking works, but only directly, not transfer style. Any hints?
17:34.07*** part/#asterisk strib (n=ads@wompom.dur.ac.uk)
17:34.10Strom_Cjustinu, uh, yeah ;)
17:34.19zoa2when does that happen ?
17:34.19zoa2what card is that ? te410p ?
17:35.15Strom_CTE406
17:35.34zoa2how often do you get those messages ?
17:35.41zoa2and do you get them only under heavy load ?
17:36.13zoa2the flooding is something i also saw before, its extremely fast
17:36.30Strom_Czoa2, I don't know when this happens yet
17:36.43Strom_Cthe logs had filled the disk and my client had to delete them to get the system back up
17:37.23shido6dont want to assume - but whats exten 700 say in your dialplan :)
17:38.02*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
17:38.05*** join/#asterisk seele_ (n=seele@200.124.172.72)
17:38.09*** join/#asterisk zoa (n=kkk@pirus.securax.be)
17:38.22zoagrr
17:38.26zoadamn internet connection
17:38.35seele_Hello... somebody could recommend me a good GUI for the end-user???
17:38.48seele_That doesnt suck as much as AMP
17:38.49DaminWTF good is relaxdtmf on sip connections?
17:39.08zoadtmf recognition on inband
17:39.12fndudeshido6: exten => 700,1,Park()
17:39.12fndudeexten => 700,2,Hangup()
17:39.35*** join/#asterisk santiago (n=santiago@208.195.215.97)
17:39.54fndudeshido6: like I said, dialing 700 from the menu will park the call fine. But the transfer does not work, even to other working extensions.
17:40.00*** join/#asterisk lodeon (n=not4u@h75n5c1o1023.bredband.skanova.com)
17:40.09seele_Hey anything else besides AMP that sucks?
17:40.22justinueverything sucks
17:40.48zoathat is set looser
17:40.48zoa? :)
17:40.48zoathat matches faster
17:40.48zoaso that when you talk to you wife, it thinks you send digits ?
17:40.48zoahey ho DAMIN!
17:40.56fndudeseele_: I just went through that myself, did you check out freshmeat?
17:41.13[TK]D-Fenderseele_ : Basically any GUI for *.  You lose control on stuff and debugging is a pain in the ass.
17:41.46seele_i'm looking for an easy solution for my end-users
17:41.48Daminzoa: Hola..
17:41.58Strom_Cseele_, give 'em touchtone phones
17:42.00Strom_Cthat's easy
17:42.09Strom_Cleave the switch administration to the switch administrator
17:42.13[TK]D-Fenderseele_ : If things go wrong, every GUI is bad.  For general quality I'd say the one I'm stuck with rates above the rest (ScopServ).
17:42.17Daminzoa: I'm trying to track down a DTMF issue using G.729 passthrough and RFC-2833 to SIP.
17:42.34Daminzoa: It works fine for just about everything except 1 specific number..
17:42.47Kattyblitzrage: looks like i'm not going to etel )=
17:42.47Daminzoa: But everything else works fine w/ that number..
17:42.49*** join/#asterisk alk (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
17:42.55Kattycheapscape company that i work for.
17:42.55seele_[TK]D-Fender, basically i want this so that no user has to be a master-geek in order to make a simple config of his PBX
17:43.03Daminblitzrage: Did you get your Flight booked?
17:44.07seele_Anyway, does anyone knows why my ring groups are always busy??? why i cant use them ?
17:44.08*** part/#asterisk santiago (n=santiago@208.195.215.97)
17:44.35seele_I put some extens to them and they sound busy....
17:45.13*** join/#asterisk kpettit (n=keith@69.15.174.114)
17:45.30zoai want to go to etel
17:45.33fndudeAnybody here use TelaSIP for asterisk termination?
17:45.39zoasomebody willing to pay for my flight  ? :)
17:45.53zoadamin: thats really strange
17:46.47zoadoes inband dtmf recognition work on g729 pass thru ?
17:46.50zoathat would be weird
17:46.51seele_I' trying to assign some extens to a ring group and they sound busy (of course they really arent). HELP1
17:47.03zoaas i suppose asterisk needs to decode the audio before it can match dtmf
17:47.28Ferrarianyone using polycom phones
17:47.45Broomhello all, for some reason utilizing the genzaptelconf command I get this message: channel 1, WCT1, unhandled for now :on all of my TE110P card channels, any idea?
17:47.49zoaso that would make it impossible to work if you dont have a g729 license
17:48.08Ferrarii have found a problem when 2 polycoms are talking to each other and one of them looses network connection asterisk does not see that it dropped off and still shows the channel as active
17:48.32zoaferrari, thats normal
17:48.40zoayou need rtptimeout to fix that
17:48.51Ferrarizoa thanks
17:49.02Ferrariany urls i can read on that rtptimeout
17:49.09Ferrarior is it an rfe
17:49.15zoahttp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout
17:49.23zoaim faster than google
17:49.23zoa:)
17:49.26Ferrariyou rock thanks
17:49.36zoareturned reply in 0,0001ms
17:49.43seele_zoa, any help with my issue??
17:49.49[TK]D-Fenderseele_ : I'm far from a master geek andam doing just fine.
17:49.52seele_zoa, I' trying to assign some extens to a ring group and they sound busy (of course they really arent).
17:49.58docelm0DAMIN!
17:50.02Ferrarimaybe you should go public and you could make BILLIONS also
17:50.07zoagoatie!
17:50.12docelm0ZOA!
17:50.16Ferrariwith that type of return speeds its a golden goose
17:50.16docelm0not anymore.. I shaved..
17:50.21seele_[TK]D-Fender, im farest than you then.
17:50.41zoahow are you making the ringgroup, i only did that with zaptel before, and only once
17:52.33Kattydocelm0: ...
17:52.39Kattydocelm0: why on earth did you do that?
17:52.39*** join/#asterisk Strom_C_ (n=strom@216-80-66-245.lem-bsr1.chi-lem.il.cable.rcn.com)
17:53.04Drukenhmm.....
17:53.38seele_zoa, can i make my rg with SIP and ZAp extens ??
17:53.43dippo_man teliax is not so good with the ol' customer service
17:53.57dippo_anyone know of another good IAX trunking service that can handle on the order of 6-8k minutes/mo?
17:54.47*** join/#asterisk calennert (n=calenner@adsl-068-017-103-165.sip.gsp.bellsouth.net)
17:54.49Drukento where?
17:55.15*** join/#asterisk Samoied (n=Samoied@200.247.141.111)
17:55.44iDunnoI hate phone companies.
17:55.46iDunnothey suck
17:55.49Drukenditto :)
17:56.04Drukendippo_: 6-8k mins to where?
17:56.20justinuwireless phone companies especially
17:56.42docelm0Katty, I didnt want to attract freaky chics now that I am single..  besides..  I was up for a raise..  So I thought I would begin to act the part.
17:56.58dippo_Druken: mostly domestic
17:57.02dippo_in the US
17:57.21Drukennot canada eh... hehehe
17:57.21dippo_teliax seems decent enough but their biggest plan is only 2500 min/mo
17:57.27dippo_and also they won't answer the phone when I call them
17:57.31dippo_which is a bit ironic
17:57.47Kattydocelm0: k
17:58.03Drukeni know a canadian company that can handle the volume, but not sure on the costs...
17:58.25zoaseele, explain me exactly what you want
17:58.55docelm06-8k a day, week, month?
17:58.55CyonHmmm, anyone here familer with sipura->ser->cisco for the purpose of faxing over t.38, or any reliable faxing method?
17:58.58docelm0I can take it.
17:59.23dippo_6-8k a month
17:59.39*** join/#asterisk secure75 (n=mic@ppp-82-135-87-186.mnet-online.de)
17:59.39zoacyon, does the sipura actually do t.38 ?
17:59.42docelm0Whats your target price?  Is it just canada?
17:59.47zoaor is it like the others, a fake message that it does ?
17:59.59green_earzhello all, I just subscribe to the "asterisk-users@lists.digium.com" but I have not yet received a email to confurm my registration. dose the registration need to be approved by the list moderator ?
18:00.07*** join/#asterisk FastJack (i=fastjack@p5091CD8E.dip.t-dialin.net)
18:00.15zoagreen_earz: no i dont think so
18:00.23dippo_it's not canada, we're in the US.
18:00.26blitzrageDamin: not yet, but I could do that now
18:00.38Ferrariyoa thanks that rtptimeout fixed my woes
18:00.44blitzrageI was thinking of doing it on Saturday night once I knew I was actually going to be in the country though
18:00.55dippo_we're paying $45/mo right now for 2500min/mo.. so pretty much anything under $135/mo would be an improvement
18:00.59zoayou are welcome, now give me your ferrari
18:01.00zoa:p
18:01.07green_earzzoa: thanks so it looks like it being very slow
18:01.25justinudippo: for whatever reason most low volume VoIP termination/origination seems to have poor QoS
18:01.28docelm0so 6-8K a month domestic?  Killer..   www.plainvoip.com  .92c per minute
18:01.35NDTThere is a problem downloading from sourceforge...well I never thought I would see the day hehe
18:02.07dippo_interesting.. thanks
18:02.22iCEBrkrAnything different I have to do to make Asterisk answer a PRI?
18:02.22tuxinator_linuxMNDT, what seems to be the problem?
18:02.23iCEBrkrLike different from VoIP calls?
18:02.24*** join/#asterisk ikey (i=ikey@220.226.19.92)
18:02.31docelm0iCEBrkr, yes..  Learn how ISDN works..  :P
18:02.34iCEBrkrLOL
18:02.37dippo_do they do IAX trunking?
18:02.38blitzrageKatty: boooo... thats too bad. I haven't met you yet
18:02.47blitzrageI"ve met a good number of people from IRC though
18:02.56dippo_aha, i see
18:02.58docelm0No IAX trunk isnt support cause its not efficient @ high amounts of calls..  SIP is better
18:03.03*** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com)
18:03.10NDTtuxinator_linuxM: must just be busy searches timing out...heh
18:03.12iCEBrkrdocelm0: "your call cannot be completed as dialed, please check the number and dial again"
18:03.44iCEBrkr...odd, I don't even see anything scroll on the CLI
18:03.50docelm0iCEBrkr, "If you think you have reached this recording in error,  please hangup and call someone who cares...."
18:03.56iCEBrkrhaha
18:03.58Kattyblitzrage: you could come to cluecon.
18:04.03*** join/#asterisk Splas (i=jwb@206.252.198.100)
18:04.06Kattyblitzrage: the company has instructed me to go to it instead.
18:04.10dippo_hm, really? maybe i should switch our teliax trunk to sip
18:04.20justinuwhen's cluecon?
18:04.27*** join/#asterisk lorinc (n=ang@caracas-3803.adsl.interware.hu)
18:04.28NDTtuxinator_linuxM: You know...normal stuff...like everyone else needs to be using it when I am not LOL
18:04.31Kattyjustinu: anthm doesn't know yet.
18:04.31tuxinator_linuxMNDT, it was slow registering a new project the 2 days ago
18:04.34blitzrageKatty: booo
18:04.39Kattyblitzrage: i sorry. :<
18:04.46blitzrageKatty: not much you can do I suppose :)
18:04.52docelm0dippo_, I dunno 99% of what I do is SIP..  IAX is supported only on Plainvoip.
18:04.53Kattyblitzrage: not really :<
18:05.03docelm0dippo_, our wholesale platform is sip only
18:05.12blitzrageI'm only going because I'm getting a bunch of stuff for free :)
18:05.15blitzragejust have to pay for the flight
18:05.18*** join/#asterisk Rowter (n=SilverDr@201.145.5.26)
18:05.24justinui think i'm going to go to cluecon this time
18:05.37Kattyblitzrage: yeah. that's all i'd have to pay for. lunch, hotel, and flight.
18:05.44Kattyblitzrage: and the company had a heartattack over 800 bucks.
18:06.03docelm0skip that..  Im going to Astricon 2006 in Dallas!
18:06.18Kattyi probably won't get to go to that either.
18:06.22justinui'll skip that one
18:06.34mog_workyay!
18:06.38mog_workerr no
18:06.53mog_workbut astricon 2006 so close to waco ill get to see my brother
18:07.07NDTI am going to Von
18:07.16*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-227-68.claranet.co.uk)
18:07.31KattyVONs way to expensive for us.
18:07.33blitzragedocelm0: cool! I'll be there too :)
18:07.45blitzrageVON is useless in my opinion
18:07.58NDTKatty: Me too...we just eal with a place that gets us in free heh
18:08.02NDTerr deal
18:08.10iDunnois it home time?
18:08.11blitzrageunless you have too much money
18:08.26Kattycluecon was nice though.
18:08.27blitzragehome time? I have no home time... I work at home, thus, I'm always working
18:08.28*** join/#asterisk oej (n=oej@apollo.webway.se)
18:08.32NDTblitzrage: No I am just going cause I dont have to pay
18:08.33Kattysmall enough to mingle a bit.
18:08.44blitzrageNDT: ahhh I see
18:08.44Cyonzoa:  Sorry for the delay; sipura 2002s support t.38 supposedly
18:08.48Cyonzoa:  None of the others do yet.
18:08.56NDTblitzrage: If I had to they could keep it LOL
18:09.05blitzrage:)
18:09.22blitzrageoh yah, now I remember what I was going to do... find a flight
18:09.25zoadoes somebody know the manufacturer of the st302 ?
18:09.36*** join/#asterisk tainted- (n=somewher@mail.k2usa.com)
18:09.41iCEBrkrI'm really thinking something isn't 100% configured for our PRI to accept inbound calls.
18:09.48iCEBrkrAsterisk doesn't spew anything
18:09.53tainted-what could cause audio to cut out for a few seconds?
18:09.57tainted-like 5 seconds
18:10.05blitzrageDamin: when do we get access to the hotel room?
18:10.30iCEBrkrblitzrage: I'm trying to find a flight too.
18:10.50zoadamned i want to come :(
18:10.55justinuice: any more details?
18:11.07iCEBrkrjustinu: on?
18:11.13*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)
18:11.18blitzrageiCEBrkr: cool!
18:11.25justinuice: your problem
18:11.39blitzrageEVERYONE GOING TO E-TEL: Make up a badge with your IRC name on it!
18:11.41justinuyou have a PRI up, but inbound calls are blocked?
18:11.54blitzragesometimes I only know people by IRC name (most of the time usually :))
18:12.05shmaltzjustinu, here:
18:12.06shmaltzhttp://pastebin.ca/36501
18:12.08iCEBrkrjustinu: Nope.  Everything appears to be good/up/green.  Inbound isn't really important, but I'd like to get it working.
18:12.19iCEBrkrjustinu: outbound seems to work, I've dumped a call file and my cellphone rang.
18:12.22docelm0iCEBrkr, do you want me to come set it up for you?
18:12.22*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
18:12.45*** join/#asterisk [chico] (n=chico@p5491111D.dip0.t-ipconnect.de)
18:12.55justinuice: you could turn on pri debug and see if you get a SETUP when you make an inbound call
18:12.56Kattyblitzrage: you mean they don't hand out pretty badges? :<
18:13.01Kattyblitzrage: i got a shiny one for cluecon.
18:13.11iCEBrkrdocelm0: Naaa. I'm just confused.. Cuz Asterisk (Typically with VoIP) will bitch about how there's no way to handle "123-123-1234" and Hangup()
18:13.29blitzrageKatty: I'm sure they do, but I doubt it'll have my IRC name on it :)
18:13.31iCEBrkrI got my contexts setup..
18:13.38Kattyblitzrage: well just give everyone a crayon.
18:13.43blitzragelol
18:13.48docelm0sigh.. women
18:13.51Kattyfree crayon with ever badge!
18:13.59Kattys/ever/every/
18:14.09Kattyjbot: thank you, dear.
18:14.09jbotKatty: de nada
18:14.35*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
18:14.39Kattydocelm0: yes. we're quite annoying. aren't we?
18:15.00iCEBrkrI get a disconnect message if I call from our office phones..
18:15.01iCEBrkrlol
18:15.08iCEBrkrSo it's not 100% configured yet
18:16.04Daminblitzrage: Monday, the 23rd..
18:16.22KattyDamin: are you going to etel?
18:16.35Kattyi need favor.
18:16.36DaminKatty: Yes.
18:16.40iCEBrkrDamin: I'm looking for flights..... FYI
18:16.45DaminKatty: A favor is going to cost you..
18:16.46docelm0DAMN RIGHT!
18:16.52iCEBrkrDamin: ^5
18:16.54KattyDamin: k, i'll ask someone else.
18:16.58iCEBrkrKatty: *points and laughs*
18:17.00docelm0Where is ETEL?
18:17.02iCEBrkrHA HA
18:17.06iCEBrkrdocelm0: SF
18:17.10docelm0screw that..
18:17.11KattyiCEBrkr: having fun?
18:17.15iCEBrkrKatty: yup!
18:17.18KattyiCEBrkr: k
18:17.31Kattymaybe bkw_ will do it for me.
18:17.32docelm0I have enough crap here w/o dealing with trying to go there.. Maybe in a couple months..
18:17.43Kattybkw_: in fact, i know you'd do it for me.
18:17.58iCEBrkrdocelm0: oh, you have a life? See, I don't :P
18:18.25docelm0Im getting divorced..  I dont believe that constutes a life..
18:18.37justinuack... that sucks
18:18.42Kattydocelm0: so /thats/ why you've been so cranky lately.
18:18.50rob0BTDT lost the T-shirt
18:18.53docelm0yes among other things..
18:18.54iCEBrkrdocelm0: :( Boooooo
18:19.02DaminKatty: what do you need done?
18:19.05Kattydocelm0: and here i thought i had pissed you off.
18:19.17Kattydocelm0: instead, it was another female....and all females have gone to hell for it.
18:19.18docelm0No not you directly..  Just your gender
18:19.23Kattyexactly.
18:19.26KattyDamin: nothing.
18:19.30KattyDamin: i'll ask someone else.
18:19.49DaminKatty: OK.. I was joking around before, but it is entirely your discretion..
18:19.54blitzrageDamin: leave the 27th?
18:19.56NDTKatty: looks like you been stereotyped ;)
18:19.58*** join/#asterisk brockj49464 (n=brockj49@22.105.dhcp.hope.edu)
18:20.03KattyNDT: isn't everyone?
18:20.06NDThehe
18:20.08Daminblitzrage: I plan on leaving that Friday, yea...
18:20.09KattyDamin: i don't think you'd actually be good for it anyway.
18:20.19KattyDamin: unless you know junky
18:20.22blitzrageDamin: I should have bought that flight the other night... its not up to $477
18:20.24Daminblitzrage: We might have to split the cost of the Hotel room on the 26th.
18:20.37DaminKatty: I know junky pretty well..
18:20.41iCEBrkrblitzrage: Where you flying out of?
18:20.45blitzrageDamin: no biggie...
18:20.49KattyDamin: probably not well /enough/ though
18:20.54iCEBrkrblitzrage: the flights I've found from Tampa are < $300
18:20.55docelm0We need to all chip in to the Katty Conference fund..   This way since she is poor she can fly to astricon, cluecon, and whatever else..
18:20.59DaminKatty: Well, I dont' SLEEP with him..
18:21.01blitzrageiCEBrkr: going to try to fly out of KC since it'll cost me over $700 to fly out of YYZ
18:21.15blitzrageiCEBrkr: through who?
18:21.19Kattydocelm0: whoo!
18:21.21blitzrageI'm checking continental right now
18:21.29rob0YYZ? Canada somewhere?
18:21.30KattyDamin: uhh, not quite what i was going for there.
18:21.39iCEBrkrblitzrage: Go to www.cheaptickets.com
18:21.40blitzrageYYZ == Tornto PEarson
18:21.47rob0KC = ??
18:21.47iCEBrkrblitzrage: I found AA and Delta there.
18:21.55blitzrageummm.. KIansas City?
18:22.14rob0Isn't that a fur piece from Toronto? :)
18:22.28blitzrageI'll be in KC next week
18:22.42KattyDamin: you just need to hug him for me.
18:22.44rob0oh oh I figured it was something I was missing :)
18:22.46blitzrageoh nice -- $422 from YYZ on cheap tickets
18:22.53KattyDamin: but it must be longer than 2 seconds.
18:22.56blitzragestill going to cost $529 :(
18:22.56docelm0Katty, you tried to get me to do that.. HELL NO!
18:23.15Kattydocelm0: you just have issues.
18:23.16docelm0Junk's cool and all..  But huggin another guy.. um, no
18:23.21DaminKatty: I'll molest him for you!
18:23.25Kattydocelm0: see, hug issues.
18:23.27*** join/#asterisk arbius (n=arbius@c-67-173-45-34.hsd1.il.comcast.net)
18:23.28KattyDamin: that's the spirit!
18:23.35docelm0Damin you sick bastard..
18:24.00Kattydocelm0: who was i trying to get you to hug? twisted?
18:24.43docelm0Josh, Junk and someone else.. dont remember
18:24.55Kattyhmm, don't recall who the other one could have been.
18:25.03DaminJosh still owes me an Voicemail...
18:25.14docelm0dunno..  You are female with raging horemones..
18:25.18blitzragecheaper if I fly out of KC
18:25.23Kattydocelm0: female, yes.d
18:25.27Kattydocelm0: raging hormones? hardly.
18:25.35docelm0How old are you?
18:25.37Kattydocelm0: i am a straightedge, i have absolutely no interest in males that way.
18:25.46Kattydocelm0: now you give me a female, that might be a different story ;)
18:26.03Kattyrob0: thx.
18:26.06rob0yw
18:26.07blitzragegive me one too
18:26.12*** join/#asterisk dustyservers (n=admin@d205-206-85-75.abhsia.telus.net)
18:26.14dustyservershi
18:26.19Kattyhi.
18:26.19docelm0um..  interesting..   But I will still drool..  :)
18:26.25dustyserverscan anyone tell me what I need to start up a pbx for my home
18:26.26dustyservers?
18:26.29tuxinator_linuxMYou guy should stop picking on Katty
18:26.30dustyserversI know I need linux
18:26.32Kattydustyservers: a box!
18:26.34rob0I have one female cat left now ;)
18:26.34Kattydustyservers: and linux.
18:26.38Kattydustyservers: and software (=
18:26.40blitzragelol!
18:26.44dustyserversyup that too
18:26.44*** join/#asterisk [sP] (i=jwb@206.252.198.100)
18:26.45dustyserversbut like what cards
18:26.48docelm0dustyservers, server, linux, asterisk, knowledge of how to do it.
18:26.49[TK]D-Fenderdustyservers : depends on what kind of lines/phones you have in mind.
18:26.50dustyserversto connect the phones
18:26.55Kattydustyservers: analog or dig it all?
18:27.02dustyserversI have knowledge of linux and all
18:27.09dustyserversI want digital phone for calling out
18:27.23dustyserversbut I want to hook my lan phone line into the pbx
18:27.24Kattybut what are outbound calls going over
18:27.27Kattyanalog or digital lines
18:27.44dustyserverslan lines so that would be analog
18:27.46dustyserverspstn
18:27.55[TK]D-Fenderdustyservers : What kind of lines?  Analog, ISDN, VoIP?
18:27.56PMantis_Cdustyservers: Ok, you need a card with FXO port(s)
18:28.24dustyserversoh ok that all I need is fxo?
18:28.26[TK]D-Fenderdustyservers : how many lines?
18:28.29dustyservers1
18:28.37dustyserversit for my home so I only have one line
18:28.37*** join/#asterisk SwK[Work] (n=SwK@64.89.118.139)
18:28.52dustyserversand I also want to beable to hook up the gnet phones I have
18:28.53[TK]D-Fenderdustyservers : How many internal ext's?  What kind?  (Aanalog phones, IP)
18:28.56dustyserverswhihc are digital phones
18:29.00dustyservers9
18:29.15*** join/#asterisk enemy`x (n=johnny@85.196.70.98)
18:29.20[TK]D-Fenderdustyservers : each as a seperate extensions (independant)?
18:29.26dustyserversyuppers
18:29.40Kattydustyservers: http://www.voipsupply.com/product_info.php?manufacturers_id=13&products_id=291&osCsid=60d054f77af55fd0ef4d998cf2e0d49d
18:29.41[TK]D-Fenderso 9 analog, 1 Gnet IP phone?
18:29.56Kattydustyservers: that's for a single outgoing analog line.
18:29.57dustyservers9 digital ip phone
18:30.05Kattydustyservers: you can also buy your ip phones there.
18:30.17*** part/#asterisk secure75 (n=mic@ppp-82-135-87-186.mnet-online.de)
18:30.28[TK]D-Fenderdustyservers : So 9 IP phones no analog at all?
18:30.31dustyserverswhat I need to do is hook up my pstn phone line into the box so I can also hook up the digital phones to the pbx and do calls
18:30.44Kattydustyservers: see that url, it's what you need.
18:31.00dustyserversok sweet so what is fxo and fxs what the difference?
18:31.03Kattydustyservers: you can get your ip phones there as well and plug them into various switches around the house.
18:31.10[TK]D-Fender~fxsfxo
18:31.12jbot[fxsfxo] An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
18:31.14Kattyfxo is for analog phone lines, fxs is for digital phone lines, like a t1
18:31.32tzangerKatty: uhm... no
18:31.36denonwha?
18:31.37dustyserversaww ok that make more sence now to me
18:31.38dustyservers:D
18:31.42denonthats the craziest thing Ive read this week
18:31.43Kattytzanger: no? that's what i've been told by everyone.
18:31.51[TK]D-Fenderdustyservers : FXO is for LINES, FXO is for PHONE.
18:31.55denonKatty: fxo is for telco lines, fxs is for handsets
18:31.57tzangerFXO is for analog phone lines, you're absolutely correct.  FXS, however, is for analog phones.
18:32.12tzangerFXO = you connect it to a telephone *O*ffice.
18:32.13tuxinator_linuxMhttp://www.digium.com/index.php?menu=fxsvfxo
18:32.18tzangerFXS = you connect it to a telephone *S*et
18:32.32dustyserversok so fxs is for ip phones
18:32.36tzangerno
18:32.37denondustyservers: no
18:32.39dustyserversand fxo for analog fphone?
18:32.41denondustyservers: it is for analog phones..
18:32.45Kattyapparently not
18:32.47tzangerFXS is for analog phones (the phones you have with a normal phone line)
18:32.48tuxinator_linuxMno speacial hardard for IP phones
18:32.54denondustyservers: FXO is to plug in a phone line, FXS is to plug in a regular old analog phone
18:32.56dustyserversaww
18:32.58dustyserversic
18:33.09dustyserversso fxo would be for like use with a gnet ip phone
18:33.16denonuh
18:33.19denonwtf?
18:33.21Kattyoh ah
18:33.25Kattyi'm all edjimicated now.
18:33.29[TK]D-FenderNo, FXO is just so that * can access your physical phone line <-
18:33.29tzangeran FXO port emulates a telephone.  An FXS port emulates the telephone company.
18:33.46dustyserversoh ok
18:33.57gammacoderip phones just talk IP to astetisk - no need for an FXS
18:33.57blitzrageIP has nothing to do with FXO/FXS.
18:33.57dustyserversso would I need fxo and fxs ports then?
18:33.57denonor you can jack a T1 into any of them, and watch em fry :)
18:34.03justinuwhich leads me to ask why are FXO more expensive?
18:34.14denonjustinu: more circutry is needed
18:34.17tzangerjustinu: Part 68 compliance is my guess, the BOM on an FXS card is actually more expensive
18:34.32justinui figured that more circuitry would be required in FXS, because you need to generate ringing voltage, etc.
18:34.37tzangerjustinu: precisely
18:34.38justinuBOM?
18:34.46[TK]D-FenderBill Of Materials
18:34.50justinuoh
18:34.54marcus2can a carrier access channel bank grok pri signalling?
18:35.02tzangermarcus2: generally not
18:35.15tzangermarcus2: access banks deal with CAS not CCS
18:35.18wunderkinzoa: *knock knock*
18:35.18gammacoderdustyservers: you would need 1 FXO to get Asterisk connected to your single POTS line, and 0 FXS (your IP phones don't need them)
18:35.20justinuso it's another govt tax?
18:35.21dustyserversso if I just go with a fxo how would I connect my ip phone to the pbx?
18:35.35tzangerjustinu: not really.  You *want* strict compliance
18:35.37[TK]D-Fenderdustyservers : With a network switch
18:35.59dustyserversbut my telephone line be plug into it
18:36.00dustyserversright
18:36.10dustyserversso how do I plug in the rest of the phone in the pbx
18:36.14[TK]D-Fenderdustyservers : An IP phone is just a network appliance.  Shove them all on the sme switch (or get them talking one way or another) and off you go
18:36.17dustyserverssorry am a new bie to asterisk
18:36.31tuxinator_linuxMdenon, T1 frying, have experience?
18:36.45tzangerwell the T1 won't fry
18:36.49[TK]D-Fenderdustyservers : You don't plug your phone line into the switch.  You need an FXO gateway like the SPA-3000 or a PCI card in your * server
18:36.58tzangeryou see, the telco will be providing 130VDC on the transmit pair, IIRC
18:37.06tzanger(it's been a while, I forget if it's on the transmit pair or the receive pair)
18:37.22dustyserversoh so there for I only need one card
18:37.32tzangerit's used to power the repeaters and customer-side DSU/CSU in legacy installations
18:37.34dustyserversone port for my pstn line and one for my swith
18:37.36marcus2why "generally" not?
18:37.38[TK]D-Fenderdustyservers : Correct
18:37.38zoa~seen zx81
18:37.42jbotzx81 <n=ZX81@81-208-60-204.ip.fastwebnet.it> was last seen on IRC in channel #asterisk, 6d 3h 5m ago, saying: '~ping'.
18:37.51marcus2are there circumstances where it can?
18:37.56dustyserversok so do I got with both port fxo then?
18:38.05tzangermarcus2: I'm sure you can find a super expensive channel bank that understands PRI signaling but it's definitely not standard
18:38.08tuxinator_linuxMtzanger, 130VDC, I learned something new today
18:38.12marcus2oh ok
18:38.14tzangertuxinator_linuxM: eh?
18:38.18marcus2i was wondering about the cac-ii specifically
18:38.42wunderkinzoa, hey dude im looking at bug 6196, i accidentally submitted a duplicate bug.. i cant program and i saw you submitted a patch.. but i think you are unlocking at the wrong place.. should it be unlocked there *and* if weight is defined? because i have weight defined and im still getting the unlocked already notices
18:38.44tzangermarcus2: CAC2: absolutely not :-)
18:39.26marcus2anyone using the sangoma pci/t1 cards?
18:39.28tzangermarcus2: I have a number of CAC1 and CAC2s, and some Adit600s as well
18:39.31tzangermarcus2: yes
18:39.32*** join/#asterisk Katty (n=angela@64.82.232.54)
18:39.34zoawunderkin: it works if we patch it there
18:39.34dustyserverssorry just want to make sure I got here so I need 2 fxo port then one for pstn and the outher for my router I hope am right
18:39.40zoawe tested it with several thousand calls
18:39.46zoatook us 16 hours start till finish
18:39.48wunderkinzoa: with weight enabled?
18:39.49marcus2is there any reason i shouldnt get a sangoma insteda of a digium?
18:39.51tuxinator_linuxMtzanger, I though the T1 would have a max of 48VDC or something, not 130
18:40.09zoawunderkin: i will mail your comment to the programmer for a reaction
18:40.17tzangertuxinator_linuxM: nope, 130VDC
18:40.28tzangerthat's the spec, anyway
18:40.55wunderkinzoa, ill take a look at it again, maybe i did it wrong, it didn't apply at all so i had to do it manually..
18:40.57marcus2i guess thats a no? :)
18:41.57drumkillawell, if you want to support the people that are improving Asterisk every day, then yes
18:42.21marcus2but thats it?
18:42.42tzangermarcus2: sangoma is a little better on unusual systems, but they both function identically
18:42.50marcus2i've purchased plenty of digium hardware, this particular company that wants to set up voip stuff doesn't have much money
18:43.00marcus2so the $125 diff between the sangoma and the digium is significant to them
18:43.01tzangermarcus2: uh, the sangoma kit is the same price as the digium
18:43.13marcus2um
18:43.24marcus2looking at atacommm, i see $775 for the sangoma and $891 for the digium
18:43.25marcus2(dual port)
18:43.39dippo_is the ring that a phone uses controlled by asterisk or by the phone itself?
18:43.54dippo_the phone, right?
18:44.17tzangerdippo_: the phone
18:44.18drumkillawell, that's kind of a loaded question :)
18:44.39tuxinator_linuxMdippo_, both, if I am not mistaken
18:44.51marcus2am i missingsomething about digium pricing?
18:44.53zoawith weight enabled, yes
18:45.00marcus2is there some place i can get the dualspan for $775?
18:45.24wunderkinzoa, no it doesn't work for me.. maybe this is another problem
18:45.30*** join/#asterisk bkw_ (n=bkw_@ppp-70-128-119-100.dsl.tulsok.swbell.net)
18:46.06zoai didnt see those messages, (with or without the patch)
18:46.08tuxinator_linuxMdippo_, "distinctive ring" style
18:46.19tuxinator_linuxMdippo_, http://www.voip-info.org/wiki-Asterisk+ZAP+channels
18:46.58drumkillamarcus2: call sales at digium.  they will help you out with getting the price you need.
18:47.03marcus2i see
18:47.23drumkillai can't guarantee anything
18:47.24wunderkinweird
18:47.40wunderkinmaybe you are using a different version, since it didn't apply cleanly for me
18:48.17*** join/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)
18:48.22zoahe patch is for 10 januari
18:48.54bkw_since when has sales@digium ever talked to the public about buying hardware?  Every time we wanted to deal with them we were shoveled off to a reseller
18:49.02marcus2yeah thats kinda what i figured
18:49.20marcus2i'm not going to go out of my way to get a better price from digium than i can get from sangoma in an online purchase
18:49.24marcus2just doesn't make sense
18:50.55denonoh no! it's mr giggles
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18:52.45gammacoderi had a meaningful conversation with sales@digium earlier today - the guy was very helpful
18:54.02asteriskmonkeydoes anyone have a php agi that i can see.. ive tried google and stuff. but im looking for something simple and basic to study, like an ivr in php or something with if statements etc..
18:54.03marcus2to buy a single card from him?
18:54.58*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
18:56.22*** join/#asterisk Defraz (n=t0tal@72.24.26.215)
18:56.42ravenpiSo: is there any (reasonably easy) way to have voicemail attachments encoded as MP3?  (Yeah, I'd prefer .gsm or whatever, but MP3 is the one format I know all my *nix and *doze boxen can play.)
18:56.47gammacodermarcus2: no, I didn't even want to buy anything - just needed info on a TDM2400
18:57.10[TK]D-Fenderravenpi : WAV49 <-----
18:57.24[TK]D-FenderWindows standard WAV format.
18:57.49ravenpiIs it compressed at all, or will I have huge attachments?
18:58.47[TK]D-Fenderravenpi : not so big IIRC.
18:59.19[TK]D-Fenderits still 8khz mono so a fraction of full WAV
18:59.38ravenpi[TK]D-Fender : OK.  Thanks -- I'll seje how it works.  Oh, you're right.  8KHz -- no biggie.  Not like it's 44.1 stereo...
19:00.15NDTWhat does digium sell quadspans direct from them for? They are like $1263 from netxusa
19:00.25NDTI mean how much
19:00.58znoGwhat are people using these days as an on-hold music player? chan_mp3?
19:02.28drumkillathere is no such thing as chan_mp3  :)
19:03.06drumkillaNDT: store.digium.com
19:03.57znoGchan_ohmp3?
19:04.18drumkillalol, i think you mean format_mp3
19:04.19drumkilla:)
19:04.56*** join/#asterisk apardo (n=apardo@62-15-238-17.inversas.jazztel.es)
19:10.13*** join/#asterisk JMcA (n=jmcadams@pixout.appriss.com)
19:10.19znoGthat's the one drumkilla :) is it supposed to be stable?
19:11.11drumkillaznoG: supposed to be.  it doesn't make sense to use it, though, really
19:11.30drumkillathere is no channel in asterisk where mp3 will be their native format
19:11.51drumkillait makes more sense to convert the files to ulaw, slin, or whatever is most commonly used on your box, using sox
19:12.01drumkillaso that the least amount of transcoding is necessary
19:12.20drumkillaformat_mp3 is there for convenience, i suppose, but surely not for efficiency
19:13.11*** part/#asterisk silentfury (i=anubis@CPE0013104cefd8-CM000f9f5011d8.cpe.net.cable.rogers.com)
19:15.21znoGyea, convenience it seems
19:15.39znoGdoes it use less CPU than having mpg321 processes around_
19:15.40znoG?
19:18.16bkw_mpg123 don't use any CPU in the first place
19:18.26bkw_lets see 100 channels.. 100 mp3 decoders.. much more inefficient than having a single shared mp3 decoder pipe
19:19.42*** join/#asterisk Katty (n=angela@64.82.232.54)
19:20.46iCEBrkrbkw_: umm, what are you trying to say?
19:22.21*** join/#asterisk rainkid (n=rainkid@gemini.os5.com)
19:22.56rainkidhere's a silly question.. is the proper way of stopping asterisk just killing its process? is there a more 'proper' way?
19:23.11Strom_C_log into the console and say "stop now"
19:23.43rainkidinteresting.. i didnt know asterisk had a console (first time install)
19:23.45JMcAor "stop gracefully"
19:23.58JMcAwhich I assume will let calls complete before shutting down
19:25.15rainkidim reading Getting Started with Asterisk.. no mention of a console
19:25.23fourcheezeasterisk -r
19:25.53rainkidthanks
19:27.46dustyserversdo anyone know how to connect a telus phone line too an asterisk box
19:28.11twisted[asteria]with a dowhangle
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19:31.18*** join/#asterisk thomastim (n=anonymou@ntserver01.thomastonschools.org)
19:34.13*** join/#asterisk eNEMY^x`q (i=lkqw@212.62.250.98)
19:35.18IOscannerWhen I dial into the PBX from remote and check voice mail the messages are very soft I can hardly hear them.  I can hear the voicemail prompt just fine.  How can I fix this?
19:36.42denonIOscanner: you can increase the vmail gain, Show Application Voicemail
19:37.41IOscannerAlso I have the same problem if I forward calls from my phone to my cellphone.
19:37.48*** join/#asterisk Katty_ (n=angela@64.82.232.54)
19:38.24IOscannerinbound calls and outbound calls from my Cisco 7960 are just fine.
19:38.46*** join/#asterisk santiago (n=santiago@208.195.215.97)
19:39.18gugaizhi, i need to know when the call start and when the call end, in realtime (radius)
19:39.29*** part/#asterisk santiago (n=santiago@208.195.215.97)
19:39.30gugaizand authorization
19:39.37IOscannerI don't see anything about vmail gain when doing show application voicemail
19:39.46*** join/#asterisk Ciber (n=Ciber@216-211-204-48.firstgate.net)
19:40.59asteriskmonkeyanyone had this problem? out of area calls from result in echo becuase rx and tx levels are different than local calls.. if so is there a way around it?
19:41.13rainkid\
19:42.03Ciberi'm getting "your call could not be completed as dialed" msg's when i dial numbers myself, but no errors when i hit the redial button on the phone?
19:42.20Strom_CCiber, from the telco?'
19:42.25Ciberyes
19:42.35Ciberi had it before and fixed it with some wait thing
19:42.39Ciberbut i forgot how lol
19:42.52Strom_Cwhat are you dialing
19:42.59quadrata*******
19:43.10Cibermy cellphone
19:43.30Strom_Cno
19:43.30iCEBrkrCiber: Dial from a land-line
19:43.40Strom_Ci meant what string are you actually dialing into the telephone sey
19:43.41iCEBrkrCiber: you'll get a different message :P
19:43.43Strom_Cer, set
19:44.01Strom_CiCEBrkr, he's getting the completion recording from the telco
19:44.05Ciberlike 16465094835
19:44.18Ciberlike i said i had the issue before
19:44.24Ciberand adding some wait thing fixed it
19:44.27Strom_Cand what is the asterisk box dialing onto the line?
19:44.30Ciberi just can't remember what that is lol
19:44.31iCEBrkrStrom_C: I dunno about that.. I got two different messages when I dial my huntgroup...
19:44.41IOscannerany other ideas how to increate low voice calls when checking dialing in to check messages.  calls are coming in via TDM 4 port fxo card
19:44.48IOscannerCiber: is this an FXO card?
19:44.53Ciberyes
19:44.58IOscanneropenvox?
19:45.07Ciberhuh?
19:45.14*** join/#asterisk simul8ted (i=user@adsl-070-155-044-220.sip.bct.bellsouth.net)
19:45.31IOscanneris the card an openvox card or Digium?
19:45.35Ciberdigium
19:45.40IOscannerI have seen this problem before
19:45.40rainkidcan anyone point me to documentation on how i can allow dialed in users to dial out?
19:45.51Strom_CCiber, what is the digium card dialing onto the line?
19:45.56gugaizsomebody has work with PortaOne Radius auth
19:45.57IOscanneryou need to add w to the exten string when you dial.
19:46.05simul8tedrainkid: check out "adis"
19:46.09Ciberahh that's it IO
19:46.19Ciberi did that before to fix it lol
19:46.21IOscannerjust before the number and it will resolve the problem.  the card is not picking up the line fast enough
19:46.26Ciberwhere do i put the w again? lol
19:46.38iCEBrkrDo you really need ADIS? Cuz I was able to script a dialplan to alllow inbound calls to dialback out via a set context.
19:46.41IOscannerwhat are you using?  AMP or just config files?
19:46.46Ciberamp
19:47.04IOscannergo to AMP interface zap interface and add w to the prefix for that trunk
19:47.07simul8tedicebrkr put your dialplan on pastebin ;)
19:47.13iCEBrkrhaha
19:47.21IOscannersave you from adding it to the conf file.
19:47.37*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
19:47.37iCEBrkrsimul8ted: It's a simple IVR
19:47.40*** join/#asterisk dustyservers (n=admin@d205-206-85-75.abhsia.telus.net)
19:47.42IOscanneralso if you are using SIP it would break it if you are change the conf file for amp
19:47.48gugaizcan I in asterisk exec a command when the other side pickup the phone?
19:47.54Ciberi already have a 9 there
19:47.55simul8tedim too busy writing my own ast manager to write ivr menus :)
19:48.01Ciberdo i just add the w in front?
19:48.02dustyserverscan some one tell me what Alarm Receiver do?
19:48.11simul8tedthats why i just wanna copy your work
19:48.25iCEBrkrsimul8ted: sure sure, everyone's busy doing something..
19:48.37*** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net)
19:48.45[av]banifender -> another point for gxp-2000: latest firmware has true intercom support, without having to use the autoanswer-extension hack like other phones
19:48.51IOscannerput ww infront of it
19:48.52iCEBrkrDISA is what you're looking for..
19:48.55znoGbkw_: so you're saying format_mp3 is more efficient than using mpg123?
19:49.01[av]banithough it seems a pretty obvious thing to me, wonder why other vendors dont implement it
19:49.05Ciberneed comma or anything?
19:49.06simul8tedthats the one hehe
19:49.08IOscanneryou shouldn't need 9 infront of that
19:49.17dustyserverscan some one tell me what Alarm Receiver do?
19:49.22IOscannerit would add 9 to the number you dial
19:49.30simul8tedIcebrkr, if you ever feel like sharing let me know, i'd like to have a look at how you did it
19:49.31Ciberi need the 9
19:49.37Ciberbusiness line needs it
19:49.42IOscannerah okay
19:49.48IOscanneryep then add ww9
19:49.52gugaizHow I do to exec a command when the other side pickup the phone?
19:49.59iCEBrkrsimul8ted: Sure, here it is..
19:50.01iCEBrkrsimul8ted: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA
19:50.02iCEBrkr:D
19:50.09gugaizCan I do that?
19:50.09Ciberthanks IO
19:50.11iCEBrkrThere's 3 examples.
19:50.13IOscannerno problem
19:50.14simul8teddont need voip-info link to wiki ;)
19:50.14Ciberlet me check if it works now
19:50.30iCEBrkrsimul8ted: There's 3 examples.  It's cake.. use it. :P
19:50.39dustyserversdo anyone know what the feature Alarm Receiver dose?
19:50.44IOscannerso any other ideas on how to increase low voice when calling in to asterisk from remote to check voicemail?
19:51.57dustyserversguest not
19:52.39IOscannernope
19:53.00Ciberit worked IO
19:53.03Ciberthanks :)
19:53.17IOscannersays it provides support for receving alarm reports from a buglar or fire alarm panel
19:53.26IOscannerCiber: cool
19:53.38IOscannerCiber: enjoy
19:53.43Ciberthanks
19:53.52Ciberi gotta head out, i'll see you guys later.
19:54.00iCEBrkrdustyservers: When in doubt..
19:54.07iCEBrkrdustyservers: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
19:54.40IOscannercool I didin't know asterisk supported that
19:55.22dustyserversoh ok thanks
19:55.31*** join/#asterisk [chico] (n=chico@p5491111D.dip0.t-ipconnect.de)
19:56.31znoGso the question remains... is it best to use the format_mp3 decoder... or mpg321/mpg123?
19:56.44Kattyhmm
19:57.20IOscannerI think better to stream it and keep cpu cycles down
19:57.48gugaizHow I do to exec a command when the other side pickup the phone?
19:57.49gugaizCan I do that?
19:58.34IOscannerI think you would have to have an AGI script handle the calling
19:58.37IOscannernot sure
19:58.44bkw_znoG, No
19:58.45bkw_znoGbkw_: so you're saying format_mp3 is more efficient than using mpg123?
19:58.47bkw_its not
19:58.54*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
19:58.55bkw_you have one decoder per instance..
19:59.05bkw_where with just mpg123 you have a single instance for all channels
19:59.18*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
19:59.21IOscannerI would write an AGI script to do what I need when someone answers the call.
19:59.25*** join/#asterisk Chiardon (n=yo@200.71.58.39)
19:59.32sivanabkw_: does * launch mpg123 thread for each use or just what it needs an channel go in and out of them?
19:59.36ChiardonHello
20:00.06bkw_sivana, it launches one sitting on a pipe and only uses CPU when you read from said pipe
20:00.33sivanaso to have a ulaw moh, can I do that with mp3 and mpg123?
20:00.45ChiardonSomeon with big knowledge and experiencce with TE400P card or Tor2 driver that could give us some help??
20:01.19*** join/#asterisk rue_work (n=not@h24-207-96-50.cst.dccnet.com)
20:01.41rue_workgreetings, I'm haivng a problem with blank voicemails, any general pointers?
20:01.56twisted[asteria]joy
20:02.06ChiardonHello BKW!!
20:02.08twisted[asteria]anyone know why cdr_odbc would fail to write cdr records to the dsn specified?
20:02.10znoGbkw_: oh, ok.. i thought it was the other way around.
20:02.21twisted[asteria]i know the dsn is valid, i use it in res_odbc also.
20:02.49Kattytwisted[asteria]: you can't have my chips.
20:03.05KattyiDunno: i'll eye you in a minute.
20:03.07twisted[asteria]Katty, but I just ate fish! they'd go great together!
20:03.10iDunnohot chips, with salt and vinegar.
20:03.11Chiardonbkw have you seen my last email??
20:03.15Kattytwisted[asteria]: fishy :<
20:03.16twisted[asteria]anywho - bkw, any ideas?
20:03.33Kattytwisted[asteria]: and these are american chips, not uker chips.
20:03.41Kattytwisted[asteria]: Lays Classic
20:03.41twisted[asteria]Katty, oh. n/m.
20:03.57Kattythey're quite greasy. and good.
20:03.58*** join/#asterisk Lee619 (n=Lee@netblock-66-245-232-162.dslextreme.com)
20:04.05twisted[asteria]i like salt + vinegar chips.
20:04.06Lee619hello
20:04.12Kattytwisted[asteria]: know what i like?
20:04.28twisted[asteria]Katty, that's a very open ended question :)
20:04.39Kattytwisted[asteria]: yes, but the answer is obvious.
20:04.55Lee619has anybody in here successfully used A@H and FWD?
20:05.15iCEBrkrI wish there was a A@H channel
20:05.24twisted[asteria]so make one
20:05.35iDunnoKatty: you like Lays Classic 'Chips'? ;)
20:05.40iCEBrkrtwisted[asteria]: I would if I cared to help support it...
20:05.45twisted[asteria]iCEBrkr, :P
20:05.47iCEBrkrtwisted[asteria]: but I don't care. :P
20:05.48znoGwhat exactly is A@H ?? another FWD?
20:05.54iDunno(which are blatently crisps)
20:05.59twisted[asteria]Katty, was I right?
20:06.09znoGasterisk@home... oh!
20:06.11znoG:)
20:06.11[TK]D-Fender~A&H
20:06.14tzangerno, twisted[asteria] you're not right
20:06.14znoGduh
20:06.15tzangerI could have told you that :-)
20:06.16[TK]D-Fender~A@H
20:06.23twisted[asteria]tzanger, and you're not Katty :P
20:06.24Kattytwisted[asteria]: you were right!
20:06.28twisted[asteria]Katty,  ;)
20:06.29tzangerno, I am definitely not :-)
20:06.37znoGright about what exactly?
20:07.08Kattytwisted[asteria]: i suggest not doing that.
20:07.14[TK]D-Fendertwisted[asteria] :  Just wait and it'll explode all by itself :)
20:07.27twisted[asteria][TK]D-Fender, that's the problem.
20:07.28tzangertwisted[asteria]: just use cdr_pgsql and be happy :-)
20:07.28twisted[asteria]NO
20:07.31twisted[asteria]cdr_pgsql has locking issues
20:07.35twisted[asteria]and leakage
20:07.42twisted[asteria]and I don't have the time to fondle the code to fix
20:08.18twisted[asteria]besides, with odbc, i can have multiple db types bound to a single engine.
20:08.43twisted[asteria]i'm just trying to figure out why it's not loggine.
20:08.48twisted[asteria]s/loggine/logging
20:09.18*** part/#asterisk PMantis_C (n=sswitzer@66.251.89.34)
20:09.28Lee619is there any way to "recover" an FWD number if you forgot what it was?
20:09.29docelm0Does anyone know a good app for testing packet loss for linux?
20:09.35iDunnomaybe no one's calling it? :)
20:09.41*** join/#asterisk A-jay (n=quirc@62.217.245.194)
20:09.56*** join/#asterisk Naturalblue (n=Kay@195.26.12.229)
20:09.57Lee619nah... just signed up a long time ago, but never used... now, i'd like to use it...
20:10.19twisted[asteria]docelm0, ping.
20:10.21Lee619rather then sign up for a new number- is there any way to figure out what my old number was?  (besides finding the fwd e-mail)?
20:10.29*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
20:11.09znoGLee619: if you don't remember what it is, chances are you haven't given it to anybody, so sign up for a new number!
20:11.31Lee619znoG: Thanks-- yah, that's what i'll do if i can't find the old one....
20:11.46Lee619just wondering if there was any way to "recover" my old number...
20:13.36*** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net)
20:13.37iCEBrkrLee619: they have a phone book, if you filed out your info.
20:13.53Lee619icebrkr: thanks-- yep, i tried the "white pages", if that's what you mean?
20:14.11docelm0twisted[asteria], funny..  I need something better than ping..
20:14.14iCEBrkryeah
20:14.41zoadocelm0: smokeping ?
20:14.43zoa:)
20:14.47zoaping -f
20:14.47rob0docelm0: nc(1)?
20:15.14Lee619thanks all
20:16.23DaminWoah..
20:16.32Daming729 just freaked out!
20:16.34*** join/#asterisk zotz (n=zotz@24.231.47.175)
20:17.41*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
20:18.12*** join/#asterisk graphyx (n=mike@67.50.46.118)
20:18.29graphyxIt is true that the 4 port Digium T1 cards start to have issues if they near saturation?
20:18.56tzangergraphyx: only if the hardware they're running on can't keep up
20:19.24*** join/#asterisk [sP] (i=jwb@206.252.198.100)
20:19.26graphyxWhat hardware would you suggest using if you wanted to have a 4 port T1 card?
20:19.34KattyDamin: well if you'd quit torturing it with a stick...
20:19.38graphyxwould dual xeon 3.4 ghz be sufficient?
20:19.50tzangergraphyx: what are you planning on doing with it
20:20.02graphyxasterisk.
20:20.06tzangerlots of transcode, tons of AGI and doing something assinine like running a DB on the same server?  :-)
20:20.42rue_workhmm I dont know how to start debugging my blank voicemail problem....
20:20.42RoyKwasim: ping
20:20.48RoyK~seen wasim
20:20.53jbotwasim is currently on #asterisk (10h 18m 50s). Has said a total of 2 messages. Is idling for 5h 17m 29s, last said: 'rob0: he didn't do too well though'.
20:21.09graphyxtzanger: Good point.
20:21.19rue_workare any of the fellows from the district of vancouver here?
20:21.50KattyA-jay: you'll get more support in here than talking to me.
20:21.59graphyxSo any report of the 4 port t1 card having issues were more than likely caused by the server load instead of the card?
20:22.07rob0I scared him away!
20:22.12tzangeryep
20:22.16*** part/#asterisk Samoied (n=Samoied@200.247.141.111)
20:22.17rue_work"file.c: File outage does not exist in any format
20:22.17rue_workJan 11 04:22:14 WARNING[9569] file.c: Unable to open outage (format unknown): No such file or directory
20:22.18rue_work"
20:22.19graphyxtzanger: Ok good to know.
20:22.21tzangermany people use the quadspan card, even two in a machine without issue
20:22.33rue_workwhats that one on about?
20:22.48rue_workhow can I trace it down?
20:23.50rue_work??
20:24.12wunderkinls -la /var/lib/asterisk/sounds/outage*
20:24.39*** part/#asterisk graphyx (n=mike@67.50.46.118)
20:24.40*** join/#asterisk shido6 (n=bleh@i216-58-29-215.cybersurf.com)
20:25.10rue_workits not there all right
20:25.22rue_workso is that part of the ivr then?
20:25.28wunderkinthat's where it should be
20:25.38rue_workwhat message is that?
20:25.42wunderkinyou may be able to supply a path
20:25.59rue_workapp_playback.c: ast_streamfile failed on Zap/6-1 for outage
20:26.02rue_workI dont like that either
20:26.13rue_workoh its the same
20:27.28tainted-anyone know what audio problems could be? my latency is fine
20:27.40rue_workthe file seems to be missing... it would be outage.gsm, right?
20:28.00tainted-the audio cuts out for 5 seconds at a time
20:28.06tainted-and comes back
20:28.07tainted-could it be RTP stream problems?
20:28.18tzangertainted-: only a tcpdump can tell
20:28.21rue_workapp_setcallerid.c: SetCallerID requires an argument!
20:28.22JMcAit certainly seems like it could be
20:28.29*** join/#asterisk trym (n=trym@194.63.254.6)
20:28.35rue_workthis would be out missing callerid data woulnd't it
20:28.44tainted-tzanger hey there.. what should i look for?
20:28.57rue_workis that a bad function call or is it a scripting error?
20:29.16tzangertainted-: capture it and look at it with ethereal and see if you are getting RTP during the blankouts
20:29.28tzangerand then look at the RTP contents if so and see if there is actually audio in there
20:29.31iCEBrkrIllegal function call, line #23... 1st down.
20:29.40rue_workhmm, is there somewhere I can get the file outage.gsm?
20:29.48tainted-tzanger by the way, do u have a 1.2.0 patch for those files (parkandannounce.c, etc)
20:30.09tzangerI can create them
20:30.22tainted-is it the same patch?
20:30.33tzangerthe parkandannounce patch for 1.2.0 is WAY smaller (I have a patch in mantis right now to make it part of the main * code)
20:30.40tzangerthe other stuff (MOH) I would have to test
20:30.41rue_worktzanger the audio file or the patch?
20:30.46tainted-k
20:30.54tzangerrue_work: ?
20:31.06rue_workwell, you pointed out I need outage.gsm
20:31.15tzangerno I didn't
20:31.16svenna_hi all, i ve got a strange(to me) problem: i d like to use SIP. and it works so far, what means, that i can call out - but when i call my number from the outside, i always get a busy sign... sip show registry tells me, that the state is registered. so, does anyone have an idea?
20:31.24*** join/#asterisk chapeaurouge_ (n=chapeaur@85.201.81.201)
20:31.28*** join/#asterisk ty_tex (i=uglyy@85.101.128.147)
20:31.32rue_workI dont know what its supposed to sound like and I assume its part ofhte asterisk files
20:31.34badboyzwheres a good place to start to post an * application so that people can download / test / etc ?
20:31.46tzangerbadboyz: on your website
20:32.15asteriskmonkeywhat does youur application do?
20:32.20badboyztzanger: but id like to drop some links around so people can mess with it -- any spots in specific that people post apps?
20:32.38asteriskmonkeybadbozy: mailing lists
20:32.49*** join/#asterisk razu_ (n=razu@87-98-87-252.hps.norby.ee)
20:32.55gammacoderCould someone help me will call waiting, from my logs:
20:32.58gammacoderJan 11 15:05:50 DEBUG[2481]: Unable to find key '8205' in family 'CW'
20:33.18gammacoderJan 11 15:05:50 VERBOSE[2481]:   dialparties.agi: Extension 8205 has call waiting disabled
20:33.23svenna_badboyz: http://www.voip-info.org is a wiki, doesnt that work?
20:33.44gammacoderhow do I enable call waiting on these SIP extensions?
20:34.13gammacoderusing Grandstream GXP-2000s which seem to support call waiting indications
20:34.44badboyzhttp://www.invalidrequest.com/features/ <-- screenshots of the app
20:36.29tzangerbadboyz: nope
20:36.32svenna_hey, badboyz: that looks nice :-)
20:37.00kink0if asterisk receives a SIP message with a TO like phone_number@ip , and where ip is from outside and not is the local ip or where asterisk is running, would run RTP ?
20:37.11badboyzsvenna_: ty :)
20:37.52kink0I got a problem because there no RTP between my asterisk and a remote gateway, and the only thing I see may be wrong is he is sending INVITE with TO:phone@HER_ip instead phone@MY_ip
20:39.17ChiardonWARNING[13881]: chan_zap.c:791 zt_get_index: Unable to get index, and nullok is not asserted
20:40.45*** join/#asterisk mjmtaiwan (n=michael@ip68-100-65-52.dc.dc.cox.net)
20:40.59*** join/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)
20:42.26*** join/#asterisk graphyx (n=mike@67.50.46.118)
20:42.56graphyxWould an asterisk system service cisco 7960 phones really need RCP port TCP 32768 open?
20:43.51*** join/#asterisk badboyz (i=bbz@adsl-70-128-78-21.dsl.stlsmo.swbell.net)
20:44.34blitzragegraphyx: TCP -- no. Asterisk doesn't use TCP
20:44.39blitzrageonly UDP
20:45.07Chiardon15:46:31 WARNING[14265]: chan_zap.c:791 zt_get_index: Unable to get index, and nullok is not asserted
20:45.14moralei used to work with someone who pronounced it 'udip' for udp
20:45.26blitzrageI don't take those people seriously ;)
20:45.45graphyxblitzrage: Ok.  Thanks.
20:46.04kink0curiosity, how is pronunced udp in english ? au - di -pi ?
20:46.11tzangereww dee pee
20:46.11Chiardonthis message is repeating frecuently . .How can I manage it?
20:46.13fourcheezeyou dee pee
20:46.35[av]banidoesnt iax use tcp?
20:46.36*** part/#asterisk graphyx (n=mike@67.50.46.118)
20:46.44tzangerjust like IAX is eye ay axe not EEKS
20:46.48tzanger[av]bani: no
20:46.52blitzragetzanger: its EEKS
20:46.56Chiardonor it isn't has big relevancce??
20:46.58tzangerblitzrage: fuck you it is :-)
20:47.00fourcheezetzanger: according to the asterisk book it's EEKS
20:47.03*** join/#asterisk Storm (n=StorM@stardust.noc.frontier.fr)
20:47.18blitzragetzanger: I don't ever want to hear you say SIP again... its ESS EYE PEE
20:47.18tzangeryeah that's because they're wrong ... and mark spencer's wrong...  heh
20:47.26blitzragetzanger: screw you :)
20:47.26tzangerblitzrage: hehe
20:47.27blitzrageI'm right
20:47.36blitzrageand Mark wrote the damn thing :)
20:47.43Stormhello, is this possible to return a channel a different error than busy/answer/congestion? I need return Q.931 specific code
20:47.45tzangerdoesn't matter.  no protocol should be named "eeks"
20:47.54tzangersince when does "IA" => ee
20:47.59fourcheezeanyone here using Yate and * together?
20:48.05tainted-someone add timeout to parkandannounce.c :D
20:49.36fourcheezeYate looks quite good for SIP
20:49.43bkw_blitzrage, ok what about PRI
20:49.47bkw_I wanna hear you call it pry
20:49.55mjmtaiwanI can't stand the pronuciation of VoIP as voyip... but I guess V O I P is a bit long.
20:50.13moralevo eye pee
20:50.17moralevow eye pee
20:50.18`SauronI just pronouince it voip
20:50.21fourcheezemjmtaiwan: just say "I Pee Telephony"
20:50.23JMcAsip, and voip are pronounced as words AFAIC...PRI is spelled out P-R-I
20:50.26blitzragebkw_: I do say PRI :)
20:50.29blitzragepry*
20:50.37bkw_pee are eye
20:50.45blitzrage:D
20:50.45fourcheezepee are us
20:50.50blitzrageI'm just being a smart ass
20:51.02tzangerI say pee are eye
20:51.10blitzrageeye aye ex is just too damn hard to say
20:51.15blitzrageeasier to say EEKS in a sentence
20:51.16mjmtaiwanyea. pri - But ISDN is I S D N
20:51.18blitzrageand I talk really fast
20:51.26*** join/#asterisk darkskiez (n=darkskie@bb-194-6-115-241.ukonline.co.uk)
20:51.36blitzragewho's going to say "is den"... seriously
20:51.57JMcAblitzrage: I don't particularly like EEKS, but I know too many unix-heads who end up saying A-I-X rather than I-A-X
20:52.09blitzrage:D
20:52.20blitzragethere you go -- a good point :)
20:52.46znoGhrm, I should have a mpg123 process running once I have musiconhold.conf configured, right_
20:52.47bkw_eye ess dee en
20:52.49znoG?
20:52.53JMcAso...just out of curiousity...how many folk here run multiple * servers?
20:53.04bkw_ok whats this http://www.voip-info.org/wiki/index.php?page=Asterisk%20GPL%20Compliance
20:53.15Chiardonwher i can find a list aboute Asterisk error at CLI and how to undesrtand every one??
20:53.16postelJMcA: is that a survey?
20:53.21bkw_asterisk violates the GPL?
20:53.28blitzrageJMcA: I do...
20:53.39rob0<== 2 * servers
20:53.40mjmtaiwanHow is GPL pronounced?
20:53.50bkw_GEE PEE ELL
20:53.50znoGbkw_: what are the requirements for mpg123 to work?
20:53.50JMcApostel: uhm...I'm just curious...'cause I'm looking at running 3 or 4 to get the dialing capabilities to/from various places
20:53.51simul8tedChiardon: www.voip-info.org
20:53.58postelGee Pee Elle
20:53.59justinugipl
20:54.00blitzrageoh come on... are people just asking questions to hear themselves types? :)
20:54.05mjmtaiwanyse
20:54.08justinugiple
20:54.23blitzrageI have like... 10+ boxen
20:54.26JMcAthat makes LGPL be elgiple?  blah, no thanks
20:54.40blitzrageyou have to pick the one that makes sense
20:54.46JMcAblitzrage: are those clustered for performance/scaleability, or 10 different sites?
20:55.01blitzrageI'm not going to say S-I-P, but I'll say I-S-D-N
20:55.12Chiardonsimul8ted thanks!!
20:55.25marcus2if i get a PRI from my telco with 6 voice channels, and internet service on 8 of the other channels, i'm assuming that i can terminate both on the same linux box
20:55.25blitzrageJMcA: 2 physical locations for redundency and load balancing
20:55.30marcus2is that correct?
20:55.52JMcAmarcus2: oh, I hate integrated access setups from telcos
20:56.11JMcAblitzrage: hrmm...ok...thanks
20:56.20rob0If I was going to sound out "IAX" it would be "yacks" or maybe "E-acks"
20:56.21*** join/#asterisk elvisthedj (n=Johnny@th20.montanavision.com)
20:56.36*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
20:56.38marcus2jmca; so how do you get to the IP channels on the linux box?
20:56.57BeirdoI would use "eye-axe"
20:57.02Beirdopersonally
20:57.04blitzragerob0: which is pretty close to EEKS if you say it fast
20:57.04rue_workfile.c: File outage does not exist in any format
20:57.15elvisthedjany cisco phone gurus around?  I just need an example file for 7960-font.xml and 7960-tones.xml
20:57.15rue_workis this file supposed to come with asterisk?
20:57.18elvisthedjcan't find them anywhere
20:57.25jsharpmarcus2:  Is it a true PRI or is it just a channelized robbed bit T1?
20:57.30JMcAmarcus2: well...I avoid integrated access services like that like the plague, so I'm not really the person to ask, but you may be better off splitting the data channels off from the voice channels with an adtran or something
20:57.43tzangerBeirdo: yeah I use that and the letters individually
20:57.45marcus2ahh
20:57.55marcus2its a true PRI, as far as i can tell from the att/sbc quote
20:58.09*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
20:58.19tzangermarcus2: are they charging you for a D channel?
20:58.24rue_workso it shoudl have been put there when asterisk was installed?
20:58.31JMcAhow is the data being delivered, then?  is it ISDN dialup?  or are they "clear-channel" data on the data channels, and the rest are configured for PRI?
20:58.32rob0but I'm in the deep South USA, nobody says anything fast :)
20:58.46*** join/#asterisk Cyon (n=cyon@216.179.31.166)
20:59.04marcus2not sure, the line items are "PRI", "voice channels", "512kbit BIA"
20:59.17Cyon_Vile:  Happen to be around?
21:00.27rue_workwhere do the files in /var/lib/asterisk/sounds/ come from? is there a site that hosts them so i can do individual downloads?
21:00.48simul8tedrue: ftp.digium.org , search for asterisk-sounds*.tar.gz
21:01.12JMcAmarcus2: ok...what is sounds like they're suggesting is to take a T1 circuit and mux 6 B channels and a D channel onto it and then mux the rest of the channels with a pseudo clear-channel data...if that's the case, you're almost gonna have to have an adtran or something such as that (a drop-insert mux/demux) to pull those back apart
21:01.13rue_worktahnks!
21:01.23simul8tedrue: here's the link http://ftp.digium.com/pub/asterisk/asterisk-sounds-1.2.1.tar.gz
21:01.38marcus2the service includes a cisco 1721
21:01.54JMcArue_work: your distro (assuming Linux, here) may have them in a seperate package
21:01.59*** part/#asterisk eNEMY^x`q (i=lkqw@212.62.250.98)
21:02.30JMcAmarcus2: ah, ok...that's actually pretty reasonable of them
21:02.40elvisthedjnobody can help me out :(  I'm just trying to upgrade sccp firmware and it keeps asking for these two files.. empty ones won't work
21:03.00jsharpYou should be able to terminate everything on your * box, then.  If you want to.
21:03.12marcus2hm, "smart trunk interface", "6 voice B channels" is exactly what they're charging us for
21:03.22marcus2on the voice side, that is
21:03.41marcus2so if we use that 1721, how do we get to the voice channels, i wonder
21:03.47marcus2does the 1721 have voice interfaces in it?
21:04.01*** join/#asterisk razu__ (n=razu@87-98-87-252.hps.norby.ee)
21:04.07rue_worksimul8ted there is no outage.* in that archive
21:04.13jsharpI don't think so.
21:04.30marcus2hm
21:04.37JMcAmarcus2: yes, I believe the 1721 does support voice stuff
21:04.55marcus2this is definately real isdn
21:04.59marcus2reading into the contract
21:05.07marcus2"this provides 23 b channels and 1 d channel", etc.
21:05.21marcus2so presumably the internet service is just clear channel on 8 b chans
21:05.24JMcA23B+1D doesn't leave any room for data
21:05.33jsharpYeah.  What JMcA said.
21:05.58rue_worksimul8ted outage* dosn't exist in the 1.0.9 archive either
21:06.00jsharpYou're not going to get there from here with a 1721.
21:06.00marcus2well, only 6 of the b channels will be used for voice
21:06.12*** join/#asterisk rainkid (n=rainkid@gemini.os5.com)
21:06.34JMcAjsharp: unless they're doing some sort of drop and insert sorta think on the 1721...it does have 2 WIC slots
21:06.41marcus2ohhhhh
21:06.43rainkidhow do i create an exten to dial out using IAX2?
21:06.43rue_workif this is justhte word "outage" I can make myself one via voicemail and copy it into sounds...
21:06.44marcus2ok
21:06.51jsharpDrop and insert of PRI is ugly.
21:07.51JMcAjsharp: oh, agreed...but I've see a setup once where there was a PRI with only certain channels turned up where those channels (plus the D) are D-I'ed onto another T1 circuit....ugly it is, but this is SBC/AT&T we're talking about...they'll sell you anything if they can make money on it
21:07.59znoGguys, is there any way to pass variables to the externpass command in voicemail.conf ?
21:08.28jsharpYeah, but that wouldn't be 23B+D. That'd be somethingB+D plus the D&I channels.
21:08.41marcus2i dont understand
21:08.42rue_workI'm gonna assume that the file outage.gsm is completely missing from all the asterisk stuff everywhere, and that everyone gets this warning and that its not fatal, that or kb1canobe was supposed to record a custom one and didn't
21:08.59JMcAyup, but how else are you going to do data on 23B+D unless you do a multi-link ppp dial-up bit?
21:09.27jsharpYou're not, with 23B+D.
21:09.53JMcAk, so its gotta be some sort of drop and insert bit
21:10.03marcus2or its multilink ppp
21:10.11jsharpYou crazy EuroISDN kids.
21:10.32marcus2well we're calling att shortly
21:10.35JMcAmultilink ppp...as cool of a hack as that was...was just that, a hack...avoid it in this day and age for a real circuit like that
21:10.37marcus2we'll see how they say the data is delivered
21:10.37jsharpI'd call your...yeah, do that.
21:11.00JMcApersonally...I'd "Just Say 'No'" to integrated access...but that's just me
21:11.23marcus2so lets say that they give me 6B+D, and then the data over 8 other channels
21:11.23kink0rainkid from the CLI add extension and follow the info
21:12.06*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
21:12.18marcus2in zaptel.conf, i'd say something like bchan=1-6, dchan=24, and clear=7-15 ?
21:12.42svenna_good night
21:12.51jsharpYeah, and you'd set up an HDLC interface in Linux.
21:12.58marcus2ok
21:13.13JMcAyou can do that with a zaptel card?
21:13.14marcus2we have no real choice but integrated services at this site
21:13.22justinuyou can do it with a sangoma card, for sure
21:13.25JMcAmarcus2: you can't do two seperate circuits?
21:13.29marcus2nope
21:13.37jsharpDouble the local loop charges.
21:13.39jsharpNot pretty.
21:13.45marcus2(a) not enough pairs into the facility and (b) too much $$
21:13.45justinuthe wanpipe drivers allow for fractional T1s to be terminated by a single piece of equipment
21:13.53JMcAjsharp: but is it less pretty than integrated access crap?
21:14.05marcus2whats wrong with integrated access
21:14.20marcus2i mean, all we need is 6 pots lines and 512kbit of internet
21:14.24jsharpDepends.  The ugliness for integrated access happens once during setup.  The ugliness of double local loop keeps on giving.
21:14.28JMcAI've never met a person that's used an integrated access setup that hasn't come to later despise it
21:14.54JMcAmarcus2: since you're in #asterisk, can you just run a full T1 and do VoIP over it?
21:15.07JMcAor even a fractional, but non-integrated-access T1
21:15.09jsharpI've done plenty of integrated setups.  Never really had a problem.
21:15.15marcus2i've used voip enough to know that i'm not running a business's incoming lines over the public internet
21:15.21marcus2no f'in way
21:15.28ChiardonWhat's the signalig in the spans that belong to a 2 channels bank all of them fxo???
21:15.42jsharpFXO ports use FXS signalling.
21:15.45JMcA*shrug* ok...I, from experience, have learned do avoid integrated access like the plague...YMMV
21:16.06*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
21:16.10generalhanwhats going on guys ?
21:16.18*** join/#asterisk viLeR (i=1000@66.128.47.232)
21:16.20*** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
21:16.21marcus2this is just my father's small business
21:16.48marcus2at my "real" job, we have a real data T1, and a full PRI for voice
21:16.54marcus2and dsl and a cable modem
21:17.03JMcAand integrated access will probably be fine if the business is going to stay small...if it grows, though, you'll outgrow integrated access and then you'll *really* be in for the pain
21:17.08generalhancan anyone help me out with a 1.2.1 issue im having ?
21:17.29Chiardonjsharp . .yes but in relation with coding and framing??
21:17.36marcus2its a second plant for a manufacturing company
21:17.38generalhanim trying to do a make clean: make install for Zaptel and i keep getting this : You do not appear to have the sources for the 2.6.14-1.1653_FC4smp kernel instal
21:17.38generalhanled.
21:17.38generalhanmake: *** [linux26] Error 1
21:17.38jsharpIf you grow to that point, you can afford "real" connections.
21:17.48marcus2so theres not much real business going on at this facility
21:17.49JMcAwell...I'll admit that I'm something of a purist...my job buys voice service at the DS-3 level, so we can afford to make decisions on philosophy a bit more
21:18.00marcus2its unlikely that we'll need more than 10 voice channels in the next 2-3 years
21:18.05jsharpChairdon:  Whatever you set the channel bank to.  I'd recommend B8ZS coding / ESF framing.
21:18.14marcus2and yeah, if we do, we'll drop the data off of the integrated line and have room for a full 23 voice chans
21:18.31jsharpIts a problem you hope to have.
21:18.40marcus2to some degree
21:19.01marcus2if they need more voice lines, they want them to be at the business office, not at the manufacturing facility
21:19.01elvisthedjgod i hate cisco right now :(  somebody should drop a bomb in that place
21:19.07marcus2the idea is to use voip between the two facilities
21:19.18gammacoderI have a TimeWarnerTelecom VersiPak - Integrated sultuion 5B + 1D + Internet at a client's site that has worked flawlessly w/ Asterisk
21:19.22JMcAmarcus2: how far apart are the two facilities?
21:19.25Chiardonjsharp  . .but I'm having *Box downs and think that is related with a bad framing and codin signaling and perhaps must I aadd crc4!
21:19.34marcus2jmca; texas and new york
21:19.48marcus2gammacode; so you have the PRI from timewarner plugged directly into the asterisk box?
21:19.50jsharpIs it an E1 or T1 channel bank?
21:20.07JMcAanother possibility might be to get a private T1 loop between the two facilities (which really won't be *all* that expensive) then you can run data and VoIP on your private line (ie, not over the Internet), and life is good
21:20.11Chiardonwe are talkin about one E1 . .pri EuroIsdn!
21:20.15jsharpOh.  Heh.
21:20.17jsharpSorry.
21:20.29A-jayhello there
21:20.35marcus2jmca; a private t1 between texas and ny? :)
21:20.37jsharpI dunno much about EuroISDN.  You can try adding crc4.
21:20.46Chiardonjsharp . .but where this problem coul be comming?
21:20.47JMcAmarcus2: not as expensive as you might think
21:20.48jsharpIt'll either fix it or break it in an ugly way.
21:20.54marcus2they will still need local voice lines in texas
21:21.03gammacodermarcus2: incoming service is broken out by TWTC's equipment to a frac PRI (to a Diguim TE110P in Asterisk box) and an Ethernet handoff (to Cisco firewall)
21:21.03tzangerhmm apparently my son cracked his head open at the sitter's
21:21.07JMcAah, good point
21:21.11A-jayanyone out there talking from theyr mobile phone???
21:21.13jsharpEw.
21:21.27marcus2also, both of these facilities are in rather rural areas
21:21.27Chiardon2 T1 channel banks working with an E1 EuroIsdn
21:21.31tzangermom's running out to grab him and get him to a hospital to assess the damage :-)
21:21.48jsharpArooo?  T1 channel banks on E1 lines?  Uh.
21:21.48JMcAmarcus2: amazingly, with FCC T1 tariffs, that really doesn't matter
21:21.53justinutzanger: hope he's not too messed up
21:22.01Chiardonhummmmmmmmm!!
21:22.02tzangerhe'll be fine I'm sure
21:22.18jsharpKids are hard headed.
21:22.37jsharpI took a header into a rock fireplace and came out no worse for wear.
21:22.47justinumy sistercut her head good on a jungle gym when we were kids
21:22.52justinuscared the living hell outta me
21:22.57tzangerany kid of mine is hard-headed
21:23.05tzangerI cracked my head open when I was a few years older than him
21:23.12tzanger5 stitches later I was no worse for the wear
21:23.15JMcAI took out some teeth as a kid doing a face plant off a jungle-gym into a rock from about 8 feet up
21:23.16znoGsay I wanted to "tell" my externpass program in voicemail.conf which user changed pass by passing it a variable or something... how could this be done?
21:23.18tzangerjsharp: heh you stole my joke
21:24.24Chiardonjsharp  . .no a harhead problem . . .that was the only one alternative that we had wen we bougth . .from digium . . becaus they don't have E1 channels banks!!!
21:24.33IOscannerWell I still can't find a solution to the low audio when checking voicemail.  The prompts are perfect, but the messages are very soft.  This seems to be really bad when I call in to check the messages from remote.
21:24.49IOscanneranyone have an ideas what I can try?
21:25.02tzangerIOscanner: sounds like you have your rxgain set really low and recorded the audio weak
21:25.22IOscannerWhat options should I try?
21:25.24Chiardonjsharp . .durinf 3 yeras they are worked in a more or less stable condition
21:25.33*** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com)
21:25.33jsharpBut now they're broke?
21:25.37Seldon1975Hi; I get the NOTICE:Fax detected, but no fax extension. Even though I have exten=>fax,1,... in my dialplan
21:25.48Seldon1975is there something else I should check?
21:25.54AyanoIOscanner; send it to your e-mail and see if it is still weak
21:25.58Chiardonjsharp but we have a big intrussion and plenty of inestability now!!
21:26.02tzangerSeldon1975: you dont' have exten => fax,1 in the context the zap channel is dumping in to
21:26.16jsharpHmmm.
21:26.19*** join/#asterisk kiwnix (n=egarcia@82.158.157.192)
21:26.23AyanoSeldon1975; is it zap?
21:26.51Seldon1975yes
21:26.53Chiardonjsharp . .we los the confogs . . at the bigining and having lot os inestability
21:26.54jsharpI'm not sure what to tell you, then.
21:27.01jsharpOh.
21:27.05Zodiacalanyone know how important it is to have ToS switches, so called voice capable prioritizing switches? i just wanta setup a small lan (5-10 phones) with simple linksys switches. will that work ok?
21:27.06IOscannerk ioscanner@yahoo.com
21:27.11IOscannerayano: thanks
21:27.13tzangerhttp://www.kottke.org/06/01/letter-to-apple-support
21:27.15tzangerbkw_: that's for you :-)
21:27.31Seldon1975tzanger: but it is in that context
21:27.45tzangerdid you reload your extensions.conf?
21:27.53Seldon1975tzanger: yes
21:27.53AyanoSheldon1975; you might try nvfaxdetect.  That's what I used.  it worked great.
21:28.05Seldon1975well it seems to be detecting the fax ok
21:28.14jsharpYou can try the crc4 stuff and see if it fixes it.
21:28.16RoyKzoa: ping
21:28.19RoyK~seen zoa
21:28.34jbotzoa is currently on #asterisk (3h 50m 25s). Has said a total of 42 messages. Is idling for 1h 13m 47s, last said: 'ping -f'.
21:28.34Ayanooh
21:28.34tzangerSeldon1975: show dialplan name_of_context_the_channel_uses
21:28.35tzangeris the fax extensions showing up there
21:29.09Chiardonjsharp in this strange mix of Euro pri and T1 channel bans how the coding and signaling and framing should be managed
21:29.09Seldon1975tzanger: aah I think I see whats wrong
21:29.15Seldon1975tzanger: thanks
21:29.28Zodiacalany ideas?
21:30.10jsharpLeave the T1s as B8ZS/ESF.  And leave the E1s at whatever your provider says.  Oh, and set your system to clock off your E1.
21:30.15jsharpThat may be part of your problem.
21:30.20jsharpConflicting clocks.
21:31.02*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
21:31.27Chiardonjsharp ... yeah, I guessed that could be it,so I did what you propose
21:31.28Naturalbluehi there
21:31.35Chiardonbut no luck =/
21:31.40jsharphmm
21:31.42Naturalbluecan some shed so light on a problem im having
21:32.01jsharpSunspots.
21:32.12Ayanowish some shadows
21:32.15Ayanowith
21:32.15justinucosmic ray shower
21:33.53Ayanonaturalblue; we nee a problem to shed light on...  ...
21:34.47IOscannerAyano: did you send the example?
21:35.13*** join/#asterisk shido6 (n=bleh@i216-58-29-215.cybersurf.com)
21:35.32Chiardonhey , Anybody here has any experience with a T400 card?
21:35.38RoyKka-boom
21:35.53shido6yes
21:35.57Naturalbluesorry: when a call comes in and someone hangs up my asterisk box doesn't recognise the call has hung up and instead continues going til it sends to voicemail
21:36.01Ayanoyep
21:36.19jsharpLack of disconnect supervision.
21:36.22Naturalbluethis means no other calls can come in until it has finally hung up after about a minute
21:36.43Ayanojsharp; damn, your quick today.
21:36.47jsharpWhere are the calls coming from?  PSTN?
21:37.17Ayanochiardon; what problem are you having?
21:37.19Naturalbluealso if i answer a call and then hang up it does seem to let the other phone know which stays on for about 30sec-1 min
21:37.29Naturalblueyes calls are from pstn
21:37.42jsharpDo you have your channels set for fxs_ks in zaptel.conf?
21:38.17Naturalbluein zaptel.conf i have fxsks=1
21:38.32jsharpHm.  That'll do it.
21:38.43jsharpPerhaps your telco isn't sending disconnect supervision, then.
21:39.03Naturalblueit works fine if i hook it up to a normal phone
21:39.03ChiardonAyano --> Umm I have an E1 connected to my T400 card
21:39.42Chiardonbut every few hours I get a message that says "No D_Channels available, using channel 16 anyway"
21:39.50Naturalbluejsharp: anything else you can think of
21:39.51Chiardonand then the phones just die
21:40.12jsharpNot off the top of my head.
21:40.36AyanoNaturalblue, did you check the wiki.  I think I remember something like that on one of the systems I put together.
21:40.54Naturalbluecan you give me the wiki address, please
21:41.36Ayanohttp://www.voip-info.org/wiki/index.php?page=Asterisk
21:41.51elvisthedji find lots of guide for upgrading this phone (7940), but none mention this 7960-font file :(
21:42.40Ayanoelvisthedj; those are two different phones...
21:42.46*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
21:42.47elvisthedji know
21:43.03elvisthedjbut i have a 7940 and this is what it is requesting from the tftp server
21:43.05RoyKbrb. reboot into 10.4.4....
21:43.13*** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
21:43.23IOscannerHow do I change the rxgain?  what are the valid number to provide.  0.0  I changed to 0.5 or 1 and asterisk will not restart.
21:44.02Ayanoelvishedj; really?  Did you accidently upgrade the firmware a wrong version?
21:44.41*** join/#asterisk |Vulutre| (n=Vulutre@175.220.204.68.cfl.res.rr.com)
21:44.47elvisthedjAyano, I bought the phone off of ebay, so who knows.  It asks for the .tlv, then the SEP<MAC>.cnf then it asks for that font file
21:45.24elvisthedjAyano, if i send it an empty file, it just loops.  If the file doesn't exist, it then asks for 7960-tones.xml .. and then loops
21:45.44Ayanoelvisthedj; do you have copies of the correct firmware for the 7940?
21:45.45elvisthedjAyano, don't these phones use the same firmware?? 79xx?
21:46.16ChiardonAyano, --> Any ideas whatcould be happening?
21:46.29Ayanoelvisthedj; not sure, dont remember.  Its been a long time since I have had to do firmware stuff to them.
21:46.31elvisthedjAyano, I do supposedly have the correct files, however it never asks for them.  I never make it that far.
21:47.00AyanoChiardon; not sure.  I have no clue about e1s.  I'm sorry
21:47.20*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
21:47.45Chiardon:s
21:47.46elvisthedjAyano, Gotcha.. well, it must not be a common issue.  There is one mention of it in the mailing list archives with the response: Just check cisco's site for telephony locale.  (which i did.. and found nothing relevant)
21:47.47Chiardonanybody?
21:48.25RoyK~seen zoa
21:48.33jbotzoa is currently on #asterisk (4h 10m 24s). Has said a total of 42 messages. Is idling for 1h 33m 46s, last said: 'ping -f'.
21:48.36RoyK~nudge zoa
21:48.56Ayanoelvisthedj; I would tell you to set it back to factory, but you need to make SURE that you have the correct files to get it to where you want
21:49.18Ayanoelvisthedj; there is a certain order that you have to upgrade it in.
21:49.48elvisthedjChiardon, did you try reloading the zaptel module and running ztcfg after making the change? (I'm not sure if you have to do that.. been awhile since i've had a zap channel)
21:50.01elvisthedjAyano, Right, i've been reading A LOT about that :)
21:50.13Chiardonelvisthedj, --> Yes
21:50.55elvisthedjAyano, but, i can see what files it's requesting, so unless i've made an error in my SEP<mac> file, then i'm not sure.  Maybe if i leave the locale tags blank it won't ask for the font / tone files..  i don't know.  I hate to do something dumb that totally bricks it
21:51.07Ayanoelvisthedj; ebay is notorious for getting firmware screwed up phones to the world.  I would guess that something is hosed in the firmware, and you need to start it from scratch.
21:51.09elvisthedjChiardon, So, what does asterisk die with?
21:51.24*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
21:51.52*** join/#asterisk aNaSTaCia_geBeri (n=LiDeR@85.108.150.190)
21:51.55jsharpI have a copy of 7960-tones if you still need it.
21:52.00elvisthedjAyano, I'd love to :)  If this stupid thing would let me.  I have the UAL ready to put on there, but it just wont ask for it.  Never requests OS79XX.txt which tells it what you want to load (supposedly from what i've read anyway)
21:52.01Chiardonelvisthedj, --> With a "No D-Channels available, using channel 16 anyway"
21:52.10elvisthedjjsharp, Thanks so much!  can you throw it on pastebin for me?
21:52.28Chiardonelvisthedj, --> Strangely enough, the card does NOT go red or amber, it stays green
21:52.42jsharphttp://pastebin.ca/36522
21:53.08elvisthedjChiardon, I apologize.. I mixed up the issue / nick .. thought you were asking about rxgain on a zap channel :(  sry bout that
21:54.06elvisthedjjsharp, i suppose if you had 7960-font, you'd have mentioned that, huh?
21:54.07elvisthedj:)
21:54.27jsharpLemme see if I can find it.
21:54.31*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
21:55.38Chiardonnp
21:56.54*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:00.16*** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
22:01.03troybanyone here have experience with getting cisco ip phones to dial SIP URLs?
22:02.15Kattyhttp://news.scotsman.com/scitech.cfm?id=16902006 <- i knew it! i knew it! someone stole my prototype!
22:04.00jsharpI don't have it anywhere, unfortunately.
22:05.03elvisthedjjsharp, Thanks for looking.  At least I got one of the two
22:05.20elvisthedjjsharp, i'm guessing you have a 7960?  I've got the 7940 and don't know why it wants these two files...
22:05.39jsharpAs far as I can tell, its part of the language package that gets downloaded to the phone from a Call Manager.
22:05.47jsharpNo,  I have a bunch of 7940s.
22:05.56jsharpAnd none of them want that file, in SIP or SCCP mode.
22:06.49*** join/#asterisk Chiardon (n=yo@200.71.58.39)
22:06.52elvisthedjWow.. sounds like i got problems
22:07.08iDunnoKatty: cooool. I hope you patented it first ;)
22:07.25Ayanoelvisthedj; it sounds like you have the wrong firmware.
22:08.01jsharpNo.  Just an oddball configuration.
22:08.28jsharpLike a phone that is expecting to download international fonts from the server.
22:08.29Motheranyone around that knows bluetooth stack parameters?
22:08.59elvisthedjjsharp, P00306000400 is the app load ID ..  that way old?
22:09.32jsharpIts not bleeding edge, but not the oldest one.
22:09.34*** join/#asterisk linlin (n=linlin@c-67-184-231-233.hsd1.il.comcast.net)
22:10.19jsharpAre you going to leave it in SCCP mode or change it to SIP?
22:10.24elvisthedjjsharp, I wish I could just connect to chan_sccp.. but I get some weirdo message (it basically connects and then asterisk closes the connection) .. chan_sccp dev told me to upgrade firmware.
22:10.36*** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-200-134.miamfl.adelphia.net)
22:11.00elvisthedjjsharp, I'll take either one that will connect.  i hosed my IAXY so now i have no phone .. (using console / headset .. yuk)
22:11.22*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
22:12.21justinuhow'd you hose your iaxy?
22:13.05*** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net)
22:13.14jsharpI'd try converting it to SIP, then.
22:13.23elvisthedjjustinu, Static I think..  Was talking on it.. (my place is really dry and static is awful.  i got zapped by a loaf of bread yesterday).  Anyway, got a big shock right to my ear.. phone plugged into iaxy made a modem type noise.  now iaxy no workie anymore
22:13.39jsharpbang.
22:13.42*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
22:13.46justinuack
22:13.51justinuthat sucks
22:14.41*** join/#asterisk Reverend (n=owned@68-169-204-147.agstme.adelphia.net)
22:14.44ReverendJan 11 17:13:35 WARNING[21436]: file.c:967 ast_writefile: Unable to open file /var/spool/asterisk/voicemail/default/200/unavail.WAV: No such file or directory
22:14.44ReverendJan 11 17:13:35 WARNING[21436]: file.c:978 ast_writefile: No such format 'wav49'
22:14.45Reverend-- x=0, open writing: /var/spool/asterisk/voicemail/default/200/unavail format: wav49, (nil)
22:14.47ReverendJan 11 17:13:35 WARNING[21436]: app.c:739 ast_play_and_record: Error creating writestream '/var/spool/asterisk/voicemail/default/200/unavail', format 'wav49'
22:15.10Reverendno such format?
22:15.39elvisthedjjustinu, Yeah.  it will register with *, but doesn't ring.. no dialtone.  IAXY-5 is Ringing  .. umm.. no, it isn't
22:16.45synthetiqanyone know why asterisk on freebsd wont open port 5060?
22:16.50justinuyeah, sounds like something on the analog side died
22:18.09jsharpelvisthedj: If you want them, I've got all the config & firmware files already setup to convert a 7940/60 to SIP.
22:18.58*** join/#asterisk A-jay (n=quirc@62.217.245.194)
22:20.07*** join/#asterisk bziherl (n=bziherl@cpe-212-18-59-51.dynamic.amis.net)
22:20.18*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
22:21.05*** join/#asterisk santiago (n=santiago@208.195.215.97)
22:21.44*** part/#asterisk santiago (n=santiago@208.195.215.97)
22:21.46Reverendso my * was working perfectly fine, and all of a sudden 3 days ago, the folder /var/spool/asterisk/voicemail/ disappears...
22:21.53Reverendnow no one can make recordings
22:22.06bziherlHello everybody. How are you doing? Is here anyone, who has any experiecens with the Alcatel 4400 and Asterisk interoperability?
22:22.34elvisthedjjsharp, When i send the phone a blank 7960-font.xml the phone displays Invalid Glyph
22:22.40iCEBrkrReverend: so make a new one
22:22.59Reverendi did
22:23.00*** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com)
22:23.08ReverendiCEBrkr still can't record
22:23.23Darwin35need help on 1.2.1 with fax spandsp 0.0.2.pre22
22:23.29Darwin35http://pastebin.ca/36525
22:23.29Reverendi created the voicemail directory, inside that created the 'default' folder
22:23.36Darwin35no
22:23.37Reverendthen inside that created the 200 folder
22:23.37iCEBrkrReverend: cuz the permissions are wrong
22:23.39Darwin35npo
22:23.40Darwin35no
22:23.50Darwin35voaicemail will create it
22:24.01Darwin35you just add the user to voicemail.conf
22:24.08Darwin35then restart asterisk
22:24.10iCEBrkrDarwin35: apparently now if nothing is getting recorded.
22:24.16iCEBrkrs/now/not
22:24.27Reverendthe users are all all defined in voicemail.conf
22:24.39Darwin35and when the user login for the first time to vm it creates the boxes
22:24.41Reverendthe directories that asterisk created are no longer there
22:24.56|Vulutre|Darwin35: what kinda help do you need?
22:24.59Darwin35what did you do to them
22:25.13Reverendi didn't do anything that i know of.
22:25.15Darwin35look at the pastebin error
22:25.15jsharpI'm guessing the 7960-font.xml is the XML file that gives it instructions on how to build the fonts.
22:25.23Darwin35http://pastebin.ca/36525
22:25.33Reverendthe only thing i've done recently is create a cron job that restarts asterisk every 6 hours
22:25.45iCEBrkrReverend: that's kinda gay
22:25.49Reverendbecause it kept locking up
22:25.58Darwin35I installed spandsp 0.0.2pre22
22:26.01iCEBrkrSo figure out the problem and fix it
22:26.11Reverendi tried...
22:26.14Darwin35but it asterisk fails to compile now
22:26.20*** part/#asterisk Utah_Dave (n=boucha@0-1pool139-119.nas28.salt-lake-city1.ut.us.da.qwest.net)
22:26.21|Vulutre|Darwin35: then you applied the patch manually or with the patch command?
22:26.30Darwin35manualy
22:26.38|Vulutre|Darwin35: this is under the make command of * right?
22:26.42Darwin35yes
22:26.52Darwin35on 1.2.1
22:26.54bziherlHello everybody. How are you doing? Is here anyone, who has any experiecens with the Alcatel 4400 and Asterisk interoperability?
22:26.55Darwin35not head
22:26.58|Vulutre|this looks like 0.0.3
22:27.01Reverendafter asterisk has been running for about 2 days, calls incomming from voicepulse don't trigger anything to happen with IAX debug on
22:27.05Darwin35nope
22:27.20|Vulutre|strange I didn't think t38 was in .2
22:27.28Darwin35it seems to be
22:27.46iCEBrkrt38 is a fairly new addition/update
22:27.52iCEBrkrThere was talk about it just a few weeks ago
22:27.53|Vulutre|Darwin35: I know it works I have it running, can you pastebin your portians of Makefile ?
22:28.01Darwin35the patch for t38 sucks
22:28.17jsharpDoes the T38 stuff actually work?
22:28.28|Vulutre|no clue I have stuck with the .0.0.2 tree
22:28.41*** join/#asterisk Lee619 (n=Lee@netblock-66-245-229-224.dslextreme.com)
22:28.45|Vulutre|but apparently the .3 tree t38 made its way into the .2 tree
22:29.03Lee619hello
22:29.32Darwin35hold a mi
22:29.34Darwin35n
22:29.42*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
22:29.45jsharpI'd love to flush the statically assigned fax gateways I have and remove them from my Quintum route manager.
22:29.50jsharpAnd just hang em off Asterisk.
22:30.10Lee619doe anybody have any experience getting * to work with FWD behind a firewall/NAT?
22:30.51Lee619i'm forwarding tcp/22 and udp/4569, 5060-5082, 10000-20000.  does that look right?  any comments?
22:30.52|Vulutre|took me awhile but I finally got fax to work through * FXS with a channel bank
22:31.38Lee619i can call from one extension to another (behind the firewall), but can't get out to FWD...
22:31.44Darwin35ok what do you want from my files
22:32.02Lee619yes, i turned on IAX on freeworlddialup.com settings....
22:32.07|Vulutre|Darwin35: just the portions that have "fax" references just paste like a line above and below
22:32.09bziherlDoes anyone know if Alcatel is capable of off-switch fransfering of the calls that come into the queue, and not strictly to the agents defined in the Alcatel 4400 PBX?
22:32.41Darwin35ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h $(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),)
22:32.41Darwin35APPS+=app_rxfax.so app_txfax.so
22:32.41Darwin35endif
22:32.49Lee619iax2 show registry shows a state of "Rejected"
22:33.00Lee619any suggestions?
22:33.12Darwin35app_rxfax.so : app_rxfax.o
22:33.13Darwin35<PROTECTED>
22:33.13Darwin35app_txfax.so : app_txfax.o
22:33.13Darwin35<PROTECTED>
22:33.23|Vulutre|yea looks good
22:33.25Darwin35thats the makefile in apps
22:33.34|Vulutre|Darwin35: you try the an older release?
22:33.58gambolputtyHas anyone done an insert from * using the mysql command?
22:34.14Darwin35what dbput
22:34.22Darwin35dbget
22:34.43gambolputtya mysql database, not the internal * database
22:34.49fndudeIs there a way to trace through a dialplan? I am trying to figure out why I have no valid extensions to transfer to: Unable to find extension '6' in context ''
22:34.58Lee619|Vulutre|: How did you get it to work?
22:35.15Darwin35I will try a older ver
22:36.38|Vulutre|Lee619: I have a TSU600 with FXS connected to the fax machine
22:36.57|Vulutre|thats connected to a Sangoma T1 card
22:37.00generalhanCan some one please help me out with an issue im having compiling the zaptel-1.2.1 ??? this is the error that i keep getting :: http://generalhan.pastebin.ca/36524
22:37.19|Vulutre|inbound faxes go to spandsp to .pdf documents from a PRI
22:37.20Lee619fax over FWD?
22:37.24|Vulutre|hell no
22:37.36|Vulutre|fax over phone lines
22:37.55|Vulutre|FoIP is still awhile out
22:37.55Lee619behind NAT/firewall?
22:38.11Lee619what problems did you have getting FWD to work behind NAT/firewall?
22:38.40|Vulutre|I don't use FWD but I can tell you, you may want to look into your sip.conf for setting an externalip= and a nat=yes
22:38.56elvisthedjjsharp, I gotta split.  thanks A LOT for the help
22:39.13RoyKeverything you never wanted to know about coffee http://commons.wikimedia.org/wiki/Coffee
22:39.19jsharpNo problem.
22:39.32Reverend0xC0FFEE
22:39.36Lee619sip? i was using iax2 to FWD... should i be using sip?
22:39.58Naturalbluejsharp: found the answer
22:40.33jsharpWhat was it?
22:40.43Naturalbluei added busydetect=yes and callprogress=yes into the zapata.conf and when i reboot it was detecting hangups
22:40.48|Vulutre|Lee619: no IAX2 works but you need to look into using IAX2 behind NAT
22:40.59jsharpYup, that'll do it.
22:41.14kink0is possible for Asterisk star a SIP dialog ok from one remote IP who requests to start RTP on another remote IP ?
22:41.25Naturalbluenot sure which one of these caused fixed
22:41.27Naturalblueit
22:42.13kink0I have the following scenary, where one remote gateway connects to my Asterisk and start SIP , then they have other IP where RTP must be used.
22:42.24*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:42.48bziherlHello everybody. How are you doing? Is here anyone, who has any experiencens with the Alcatel 4400 and Asterisk interoperability?
22:43.16kink0bziherl, GSM ?
22:43.19*** join/#asterisk SplasPood (i=jwb@206.252.198.100)
22:43.19Lee619Vulutrel: where do i set externip and localnet... iax.conf?
22:43.46*** join/#asterisk ToTo (n=ToTo@host243-91.pool8260.interbusiness.it)
22:44.31CyonHmmmm, if I'm running ser to try and get my sipura 2002 working with T.38 to my cisco...should I use ser, or openser?
22:44.39bziherlKink0, nope in fact it's a queue related problem. Do you know if is it possible to make Alcatel 4400 deliver calls from the queue directly to the Asterisk?
22:44.40justinudoesn't matter
22:45.14*** part/#asterisk mkrufky (n=mk@68.160.103.77)
22:45.52kink0bziherl, I don't know, probably you will need an AGI to manage queue
22:46.14*** join/#asterisk graphyx (n=mike@67.50.46.118)
22:46.28graphyxIs there a way in Asterisk CLI to get a list of SIP accounts that are valid?
22:46.37bziherlkink0, what's an AGI?
22:46.48iCEBrkrgraphyx: Why not look in sip.conf?
22:46.53graphyxI mean the SIP.conf file lists the client credentials.  Is there a command to list theones that asterisk has in it already?
22:46.54*** join/#asterisk bjohnson (n=bjohnson@i216-58-90-3.cybersurf.com)
22:47.07iCEBrkr'has in it'?
22:47.08iCEBrkrwtf?
22:47.09graphyxI have altered sip.conf, but the phone doesn't seem to react lik it is in ter.
22:47.25obiwanmikenoltesip show registry?
22:47.27graphyxreact like it is in there.
22:47.28iCEBrkrtab-completion is your friend.
22:47.29iCEBrkrmy friend
22:47.43graphyxok
22:47.44iCEBrkrlike type help at the CLI
22:47.49iCEBrkrand then you'll see some commands
22:47.53iCEBrkrit's pretty straight forward
22:47.59*** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz)
22:48.38patpatnzHi guys, I'm having trouble with having extensions starting with a * .. it just doesn't seem to match them at all, is there any setting I need?
22:49.12patpatnzI just have 'exten => _*77,1,...'
22:49.13iCEBrkr*55 works for me on my setup
22:49.17iCEBrkrwhy _?
22:50.00patpatnz_ matches from the start of the number
22:50.20iCEBrkrso why the _? again? :)
22:50.52iCEBrkrSo if it's not working.. try without the _
22:50.54iCEBrkrlol
22:51.00patpatnzI have
22:51.23iCEBrkr; NightOutNow
22:51.23iCEBrkrexten => *88,1,Goto(nightoutnow,s,1)
22:51.31iCEBrkrI dial *88 on my phone and bam, it works.
22:51.51KattyiDunno: :<
22:51.53*** join/#asterisk ZeMMaD (n=ZeMMaD@209.59.105.69)
22:51.55*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)
22:52.07KattyiDunno: twisted[asteria] was going to break me into the nasa place so i could build it.
22:52.22*** join/#asterisk BeHappy_ (n=willy@host54-203.pool877.interbusiness.it)
22:52.30SkramXhaha
22:52.31iDunnoKatty: ahh - it's all thier fault, then ;)
22:52.36patpatnziCEBrkr: is that from a SIP phone?
22:52.40KattyiDunno: exactly.
22:52.40iCEBrkryeah
22:52.43iCEBrkrfrom any phone
22:52.47Lee619when using IAX2 behind NAT, is it as simple as port forwarding UDP/4569 to my * machine, or is there another configuration step I am overlooking?
22:52.50patpatnzweird
22:52.51twisted[asteria]!@#%
22:52.52*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
22:53.02KattyiDunno: oh well, i'll just patent the pilot wave calls and banking transactions.
22:53.07BeHappy_god bless IAX, really.
22:53.12BeHappy_<- having nat troubles with SIP
22:53.14Kattytwisted[asteria]: hi.
22:53.21Darwin35[app_rxfax.so]Jan 11 15:52:13 WARNING[51351]: loader.c:325 __load_resource: /usr/local/lib/asterisk/modules/app_rxfax.so: Undefined symbol "fax_set_phase_d_handler"
22:53.21Darwin35Jan 11 15:52:13 WARNING[51351]: loader.c:554 load_modules: Loading module app_rxfax.so failed!
22:53.29twisted[asteria]Katty, hi
22:53.34iDunnoKatty: heh - what you need to do is patent it's comms system, because they're going to want to keep in touch ;)
22:53.44twisted[asteria]<PROTECTED>
22:53.45twisted[asteria]yay.
22:53.55Kattytwisted[asteria]: are you still going to break me into the nasa place?
22:54.03Darwin35this fricking sucks
22:54.11Darwin35its pissing me off
22:54.18twisted[asteria]Katty, heh. shh.
22:54.22Kattytwisted[asteria]: k
22:54.46Kattyoooh, shiny!
22:54.57lesouvageI have 2 swisvoice IP-105 phones and a sipura ata SIP- connected to my asterisk box. I can make outbound calls with all phones and call extensions like the echo test but I can't make a call from one phone to an other phone. Any idea what can cause this problem?
22:55.26iCEBrkrlesouvage: you didn't put those extensions in your dialplanb
22:55.48hardwiremanager isn't going to let me buy lightnigh protectors for some wireless equipment
22:55.50hardwiretoo expensive.
22:55.51Darwin35whats the command in csh to pipe output of errors and wrnings to a file
22:55.58hardwiremofo
22:56.29Lee619hardwire: lightning protectors are required to be up to code...  play the legal card
22:56.46patpatnziCEBrkr, do you use ael or normal dialplan?
22:56.50Lee619hardwire: we've lost 4 APs to lightning strikes... stuff happens...
22:56.52iCEBrkrpatpatnz: .conf
22:56.56sivanaif I've got * running as foreground app (-gcvvv), can I turn it into a process without restarting?
22:57.06iCEBrkrlol
22:57.35hardwireLee619: we have no exact lightning.. but we do have hella low fog.
22:57.50twisted[asteria]wow
22:57.53hardwireI have spent a while making isolated pole mount ap's now.
22:57.56twisted[asteria]no exact lightning.
22:58.00hardwirethis should work.
22:58.06hardwiretwisted[asteria]: oh poo on you
22:58.10Lee619hardwire: we're not in a lightning prone area... but it does rain every now and then.  :-)
22:58.15hardwireheh
22:58.32*** join/#asterisk Chiardon (n=yo@200.71.58.39)
22:58.33twisted[asteria]irony police.
22:58.37iCEBrkrsivana: that's a knee slapper there..
22:58.39Lee619hardwire: maybe i should buy a lottery ticket... you know what they say about lightning striking twice and all...
22:58.40twisted[asteria]hardwire is talking about wireless stuff.
22:59.00hardwiretwisted[asteria]: everything requires a cable sooner or later.
22:59.09Lee619wireless stuff is cool... but has nothing to do with Asterisk  ;)
22:59.10Kattyhardwire: lies.
22:59.13lesouvageIceBrkr: you mean a line like this: exten => 501,1,Macro(stdexten,SIP/501,,501)
22:59.15twisted[asteria]haha... that's what my ex-gf said
22:59.19Kattyhardwire: you don't, afterall.
22:59.19sivanaiCEBrkr: lol.. I was semi serious.. but :)
22:59.21hardwireKatty: maybe not a bug.
22:59.26hardwireor yes.. me.
22:59.28iCEBrkrsivana: Ain't happen'n buddy
22:59.31sivanaya
22:59.39sivanaI ended up restarting it.. hehe
22:59.42iCEBrkrlesouvage: If you have a phone 501
22:59.48hardwiremoo.. thats all I gotta say.
23:00.00Lee619so-- any suggestions on how to get IAX2 working behind NAT?
23:00.04hardwireLee619: used wbc100 cable?
23:00.06*** part/#asterisk graphyx (n=mike@67.50.46.118)
23:00.09twisted[asteria]i think it's time to go outofdoord
23:00.15Kattytwisted[asteria]: WHAT
23:00.17twisted[asteria]s/outofdoord/outofdoors
23:00.18hardwireLee619: open it up. set your outbound IP in iax.conf
23:00.19Kattytwisted[asteria]: are you nuts?
23:00.23Lee619hardwire: mostly LMR400...
23:00.24Kattytwisted[asteria]: it's dangerous out there!
23:00.34hardwireLee619: ok.. I need 6ghz pigtails.. this is being a bitch.
23:00.41lesouvageiCEBrkr: yes, I have the idea that everything is in place. I have done this before but nut with the swissvoice phones.
23:00.43twisted[asteria]yeah, but so what.. it's not like i'm going to get hit by a truck or an airplane
23:00.44ChiardonRoyK have made the jumpers movement . . .done some modifications (if the second span doesn't has a card  . .. totake out this jumper  . . nothing works but the initial position yes!!
23:00.48twisted[asteria]brb
23:00.48Chiardonsome idea?
23:01.40patpatnziCEBrkr, thanks anyway
23:01.44iDunnoheh.
23:03.45*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
23:04.45*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
23:06.22[TK]D-FenderDon't make me bass y0 ass!
23:07.01*** join/#asterisk zotz (n=zotz@24.231.47.175)
23:07.01iDunnoha!
23:07.53generalhanCan some one help me out with this compiling issue im having with zaptel-1.2.1 :: http://generalhan.pastebin.ca/36524 :: ive done everything in every forum that i have seen and i cant figure it out .... anyone have any suggestions or ideas ??
23:10.07[TK]D-Fenderthere, we've covered both coasts :D
23:10.18generalhanlol
23:10.26generalhanis that a slamon ?
23:10.29[TK]D-FenderAnd a decent chunk of in-between fresh-water!
23:10.29generalhansalmon rather
23:10.33[TK]D-Fenderindeed :D
23:10.40generalhanthought that sounded familiar
23:10.44RoyK[TK]D-Fender: we don't have those.....
23:11.15[TK]D-FenderI do.... right next to the butcher's :D
23:11.28RoyKgeneralhan: salmon is salmo salar
23:12.25*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
23:12.25generalhancan anyone explain to me how to get the kernel-source for 2.6.14-1.656_FC4smp ?? i think ive gotten them like 10 times but i must be doing something wrong cause i cant compile zaptel, and its killing me !
23:12.56iCEBrkrapt-get blahblahbl-devel
23:12.59iCEBrkryum install blahblahblab-devel
23:13.03RoyKgeneralhan: generally i say 'use kernel.org kernels and NOT distro fscked-up kernels'
23:13.14iCEBrkryeah, what RoyK said
23:13.20[TK]D-Fendergeneralhan : Make sure to get a lower version of DCC, I've heard funny things about FC's use of GCC4
23:13.26Lee619hmm... i put externip=ip in my iax.conf, but that did not help to get iax2 to work behind NAT...  any other suggetions?
23:13.37generalhannoo dont tell me that. i JUST updated GCC to 4
23:13.40[TK]D-FenderLee619 : Added the localnet clause as well?
23:13.48[TK]D-Fendergeneralhan : Sorry...
23:13.56generalhancrap, should i go back ?
23:14.10generalhanthis is a fresh install of the distro i can even reformat and only lose 15 minutes worth of work !
23:14.11[TK]D-FenderI guess that should be a "yes"
23:14.48[TK]D-FenderFedora Core : Putting the "bleeding" back into "bleeding edge"...
23:14.53generalhanlol
23:15.05Lee619fender: yes
23:15.14Lee619fender: thank for the suggestion
23:15.20Lee619s/thank/thanks
23:15.31generalhani know nothing about linux, well a very small amount, and people told me not to use FC and to go with CentOS or Whitebox ... but i didnt listen
23:15.36[TK]D-FenderIts things like that & fsck'd up bastardized kernels taht make me want to avoid all things RPM...
23:15.56Lee619when i try dialing 613, i get "all circuits are busy now, please try your call again later"... weird
23:16.08[TK]D-Fendergeneralhan : its mostly FC4 going a little "Gonzo"
23:16.27generalhani still have the FC3 disks !! should i load that back up ?! LOL
23:16.37drumkillamy development box is running FC4, and it pretty much went unusable a couple days ago after a 'yum update'
23:16.41drumkillausing the stock fedora repos
23:16.42[TK]D-Fendergeneralhan : What other distro's do you have handy?
23:17.08drumkilla:-p
23:17.09generalhannone, when i started using asterisk about 6 months ago was the first time i had ever touch Linux
23:17.09*** join/#asterisk ZeMMaD (n=ZeMMaD@209.59.105.69)
23:17.16drumkillaRoyK: already downloaded
23:17.16[TK]D-FenderI have a small mountain (I'm more a collector than a user)
23:17.19drumkillaabout to install it, I think ...
23:17.42[TK]D-Fendergeneralhan : how comfortable are you at the CLI?
23:17.44ctooleyThere any Asterisk/Network consultants in Austin, Texas looking for clients?
23:18.02Lee619can you confirm that the FWD host is iax2.fwdnet.net?
23:18.08generalhankind of comfortable, im picking it up slowly but surely
23:18.47[TK]D-Fendergeneralhan :  What else does your box do for you?
23:18.52ZeMMaDhow do i have my asterisk box answer my Zap on the first ring??
23:19.00generalhanthats it
23:19.02generalhanwell
23:19.34generalhani mean i have a few different things tied into asterisk ... tftp server runs on this box, as well as NTPD and DHCPD but its all related to running asterisk and the phones
23:20.02Lee619nevermind-- iax2.fwdnet.net and iax.fwdnet.net resolve to the same place....
23:20.13tzafrir_laptopZeMMaD, immediate=yes in zapata.conf?
23:20.14[TK]D-Fendergeneralhan : Try Slackware or Debian out.  They tend to stick to a more solid base and avoid the bloat (sorta)
23:20.18Lee619back to the drawing board....
23:20.43[TK]D-FenderI'm running everything you described on Slackware 10.2 at work/home.
23:20.43Lee619my fwd registration is rejected...
23:21.08Lee619but my voipjet registers fine...  what gives?
23:21.12generalhanwhen i went to college all they taught us about was $M and Active Directory, but when i heard about an "open source pbx" is was worth it to me to get my hands a little dirty and stumble around for answers and help ... sure as hell beats the $18,000 pbx Cisco was pitching me on ! lol
23:21.39drumkillayay open source
23:21.42generalhanlol
23:22.03Lee619three cheers for open source...
23:22.45generalhanive been running asterisk 1.09 for about 6months and it has worked great for me, but because of one major change with 1.2.1 i HAVE to get over there, we bought a new faster stronger server and now i cant get 1.2.1 to work. not much fun ! lol
23:23.08drumkillaheh, I'm about to release 1.2.2 :-p
23:23.30generalhanfortunately i convinced my boss to buy a new server so i still have 1.0.9 running on the old one till i get 1.2.1 working on the new one ! that helps a lot !
23:23.34[TK]D-Fender:)  thanks for the good news drumkilla
23:23.44Lee619generalhan: what problem are you having?
23:23.50drumkillageneralhan: good plan
23:24.01drumkilla[TK]D-Fender: yeah, I have been meaning to do it for days, but things keep pushing it back
23:24.05drumkillait's annoying
23:24.23Lee619Russell: what's new in 1.2.2?
23:24.24generalhanwhen i try to compile zaptel i get a "you dont have the sources for the kernel installed"
23:24.37[TK]D-Fenderdrumkilla : Isn't that always the way it goes?  Was the bug-marshal rally effective?
23:24.45*** join/#asterisk korihor (n=humberto@200.35.210.134)
23:25.07BeHappy_shit... anyone got troubles with nat and sip?.. i mean i have this configuration... SIP provider --  |NAT| --  * box -- sip softphones   the call seems to go ok, if i make a call from the inside softphones to the outside sip provider can hear the first few seconds of audio in both directions, but i get an SIP response 408 and the call is terminatted
23:25.15Lee619generalhan: what os/version are you using?
23:25.19drumkillaheh, a little bit i guess
23:25.30*** part/#asterisk korihor (n=humberto@200.35.210.134)
23:25.32drumkillaresources are stretched very thin ...
23:25.36generalhanFC4 kernel 2.6.14-1.656
23:25.50BeHappy_any ideas? (now i'm blindling changing nat related values in sip.conf... i'm near the desperation :)
23:26.01drumkillamost people have other full-time jobs - mine being a full-time student
23:26.03drumkillait's really hard ...
23:26.05[TK]D-FenderBeHappy_ : Fill in one of "externhost" or "externip", and "localnet" in sip.conf.
23:26.19[TK]D-FenderBeHappy_ : And set all of your phones to "canreinvite=no"
23:26.23*** join/#asterisk Soul (n=Soul@87-196-8-134.net.novis.pt)
23:26.38BeHappy_[TK]D-Fender, in order to use localnet i MUST have externhost/ip ?
23:26.41Lee619what is the difference between externhost and externip?
23:27.02Lee619i thought you need externip and localnet... (?)
23:27.14[TK]D-FenderBeHappy_ : You need BOTH.  For anything that does NOT fall under your localnet, it uses either "externip" or resolves "externhost"
23:27.52BeHappy_kay, i tought that anything that did not fall under "localnet" was treated as external
23:27.55blitzragedrumkilla: good call -- I should do that too
23:27.59BeHappy_thanks for the info, lets try
23:28.21Lee619got it-- so you would use externhost if you have a dynamic IP and want to use a hostname instead of an address?
23:28.45[TK]D-FenderLee619 : Yes, if you're using a Dynamic DNS service for instance.
23:29.06Lee619Fender: i understood that all of this applied to SIP-- is it also necessary for IAX2 behind NAT?
23:29.17Lee619fender: Thanks
23:29.25[TK]D-FenderLee619 : Because * does not have a mechanism for detecting your external IP intermittently it depends on an external DNS updater
23:29.34[TK]D-FenderLee619 : fairly certain.
23:29.38*** join/#asterisk ManxPower (i=ewieling@62.sub-70-197-11.myvzw.com)
23:29.53blitzrageso when the IP changes... there will be some delay between being able to recieve and place calls
23:29.55Lee619Fender: makes sense...
23:30.10Lee619Fender: I just don't understand if i need this in both iax.conf and sip.conf...
23:30.13blitzrageManxPower: zup zup!
23:30.15ManxPowerblitzrage, did my example last night help you?
23:30.28[TK]D-Fenderblitzrage : "Life sucks, but rarely swallows."
23:30.33RoyKLee619: are you using sip or iax2?
23:30.45blitzrageManxPower: for what I wanted, not really... but its damn cool and I'm going to probably use it at some point :)
23:30.47Stormhello, is this possible to return a channel a different error than busy/answer/congestion? I need return Q.931 specific code
23:30.52Lee619RoyK: trying to get IAX2 working with FWD...
23:30.52blitzrage./ignore [TK]D-Fender
23:31.00[TK]D-Fender:O
23:31.06hardwireanybody know good pizza in San Jose?
23:31.14blitzrageshut your mouth when you're talking to me
23:31.40bziherlhehehe, nope hardwire. Do you know any in Fort Lauderdale, FL?
23:31.46hardwirenot in the least.
23:31.50Lee619RoyK: my asterisk machine cannot register at iax2.fwdnet.net.  iax2 show registry shows a state of "rejected"
23:31.55gambolputtyIs call duration stored in a variable?
23:32.04*** join/#asterisk coolhp (n=crap@mtl149-99-190-66.dedicated.sprintdsl.ca)
23:32.10coolhpGood day all !
23:32.19RoyKLee619: then what do you need sip.conf for?
23:32.21RoyK:)
23:32.23bziherlWill anyone from here attend the VoIP Expo in Fort Lauderdale this month?
23:32.41blitzragewas supposed to, but instead going to E-Tel now
23:32.57coolhpI was wondering if anyone would happen to have versions 7.2.3 or 7.2.4 of the SCCP images for Cisco 7940 and 7960s ? I'd like to experiment with chan_sccp for a bit.
23:32.59bziherlwhere's E-Tel taking place?
23:33.02blitzrageSF
23:33.04blitzrageSF, CA
23:33.12bziherlOh, the Orelly's one, right?
23:33.15blitzrageyah
23:33.25Lee619RoyK:  I know, i know...  :-)   but i do use SIP to one extension (a voip phone) behind the firewall....
23:33.30bziherlI was supposed to go there, but am now going to FL, hehe
23:33.45blitzragegot asked to speak at that, but turned it down to go to Internet Telephony (was supposed to go for work), but now I don't need to go there, so I'm going to E-Tel for fun
23:33.46BeHappy_[TK]D-Fender, thanks! now it works
23:33.56RoyKOreally
23:33.57*** join/#asterisk EvilMetal (n=StorM@stardust.noc.frontier.fr)
23:34.12bziherlNice, blitzrage. Whats your profession in the VoIP world?
23:34.30ManxPowergambolputty, you like as is documented in README.variables?
23:34.48*** join/#asterisk enemy^x (n=null@85.196.70.98)
23:34.50BeHappy_but i have a strange problem.. if the outside telephone (the call is from the provider) ends the call, * does not recognise that the call is ended.. i have to end the call in the softphone too
23:34.57bziherlYeah, Oreally...
23:34.57*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
23:34.57blitzragebziherl: formally... telecommunication engineering manager now... but I'm a co-founder of Asterisk Docs project and co-author of the Asterisk book...
23:35.03*** join/#asterisk korihor (n=humberto@200.35.210.134)
23:35.14*** part/#asterisk korihor (n=humberto@200.35.210.134)
23:35.14[TK]D-FenderBeHappy_ : YWC
23:35.26BeHappy_hem... that was an acronym? :P
23:35.53RoyKnice... sip show peers with asterisk realtime and caching
23:35.54bziherlWow, excellent, blitzrage. Have to remember that :)
23:35.55RoyKLOTS of peers
23:35.58[TK]D-FenderBeHappy_ : Thats just not normal... have you forwarded the required ports to * as well?
23:36.06Naturalbluehow do i make my phones ring straight away, people ring and have to wait for 3-4 rings before my system starts my phones ringing
23:36.16[TK]D-FenderYou're WelCome.
23:36.22enemy^xcan anyone help me with hint`s? tryin to get xten to actually update the states in the contact list. I`ve put hints into the extensions.conf file, I see using show hints that there is no changes. I use peer-to-peer under presence on the xten phone. Is there something I`m missing here?
23:36.22BeHappy_oh :)
23:36.28ManxPowerblitzrage, Did you find a way to do what you want?
23:36.59bziherlBlitzrage, I am designing a few larger call centres in the Eastern Europe, Russia and Asia, and we would like to get a bit 'closer' to the whole Asterisk development process. Do you know anyone that I could talk to?
23:37.12Lee619glitzrage: love the book... thanks for your efforts.
23:37.51blitzrageLee619: great! glad you're enjoying it -- don't hesitate to leave a positive review on amazon.com :)
23:37.52BeHappy_[TK]D-Fender, i've forwared only the 5060
23:38.06*** join/#asterisk trym (n=trym@194.63.254.6)
23:38.08blitzrageManxPower: not really... but I was pretty much just dreaming, and not actually trying to implement anything for real
23:38.32ManxPowerblitzrage, I'm in a creative mood.  What did you want to be able to do?
23:38.44NaturalblueBlitzrage: whats the name of the book
23:39.34blitzrageManxPower: well... I wanted to be able to take a SQL statement, send that to a DB, then if it returned multiple rows, be able to access the information with an array in the dialplan -- but thats going to be a LOT of crazy coding :)
23:39.38blitzrage~thebook
23:39.41jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
23:39.41ManxPowercreative mood as in "I finally have my computer desk, monitor and keyboard back, have a chair and no longer have to work at the kitchen table"
23:39.42Lee619i'm fowarding UDP/4569, 5060-5082, 10000-20000  ... does that look right?
23:39.54blitzrageManxPower: nice!
23:39.58Lee619and TCP/22 for SSH....
23:40.01ManxPowerblitzrage, how is the information returned anyway?
23:40.17blitzrageManxPower: its not... you'd have to create the SQL function :D
23:40.25blitzrageManxPower: what I'm thinking isn't just something to do in a night....
23:40.30blitzrageunless you're rediculous :)
23:40.37blitzrageare you rediculous ManxPower? :D
23:41.04ManxPowerblitzrage, Not for programming, I'm not.
23:41.15blitzrageManxPower: yah... I wouldn't worry about it
23:41.41ManxPowerblitzrage, Ah.  I thought you were getting data back from the application in a way that was not easy to work with, like comma delimited or something like that.
23:41.56blitzrageyah, not really
23:42.03Lee619are there any other TCP ports i should be forwarding or are they all UDP?
23:42.14blitzrageall UDP
23:42.51blitzrageManxPower: tiz coo though. Got a LCR routine for me? :)
23:42.54ManxPowerblitzrage, I had an idea for doing named subscripts of an array.  So you could do something along the lines of SetVar(TEMP=${RESULT[LNAME]})
23:42.57Lee619blitzrage: hmm... any other ideas why my * (behind NAT) cannot register with FWD?
23:43.03[TK]D-FenderLee619 : Looks good
23:43.09lesouvageWhat does this message indicate: chan_sip.c:7936 handle_request: Registration from '<sip:501@192.168.1.200:5060>' failed for '192.168.1.108'
23:43.20ManxPowerblitzrage, Nope.  I don't bill for calls and my carrier is flat rate fo most destinations.
23:43.34blitzragelesouvage: the registration for that device failed (usually due to username/secret mismatch)
23:43.49blitzrageManxPower: yah, pretty much same
23:43.56[TK]D-FenderLee619 : You said it returned "rejected"  My guess is a bad user/pass
23:44.25blitzrageManxPower: thats cool -- I'm going to build one of those within a private DUNDi network I'm building. Got a really cool idea. DO you program in C at all? I have a pretty "simple" task that I'd like to get done... but no time :(
23:44.31*** join/#asterisk watchy (n=watchy@adsl-69-152-41-249.dsl.ltrkar.swbell.net)
23:44.40Lee619Fender: Good Thought.  I checked a few times-- also had no problems manually logging into Freeworlddialup.com....
23:44.42watchyon a cisco 7960 how do i make it automaticly dial
23:44.47watchyinstead of hitting the dial button
23:44.56ManxPowerblitzrage, I try to avoid programming unless it's REALLY REALLY needed for something.
23:45.02Lee619Fender: should i be using the FWD number or my FWD "Username"?
23:45.09lesouvageblitzrage: I checked that all a couple of times but I will do it again.
23:45.34Lee619Fender: Does secret need to be in quotes or anything special?
23:45.56watchyis it possible to dial without hitting the dial button?
23:46.01ManxPowerblitzrage, Apparently there's a rumor going around at one of my clients that I wrote Asterisk and work from my mountiantop cabin.
23:46.12tzangerManxPower: I KNEW IT!!
23:46.14watchyis it possible to dial without hitting the dial button?
23:46.18tzangerpay up,  blitzrage!!
23:46.20tzangerI won the pool
23:46.23Lee619lol
23:47.08[TK]D-FenderLee619 : NOT SURE REALLY.. i HAVEN'T TOUCHED MINE IN A LONG TIME..
23:47.35ManxPowerI'm waiting until the rumor evolves into me taming a pride of mountian lions before I dispute it.
23:48.22ManxPower"[TK]D-Fender: NOT SURE REALLY.. i HAVEN'T TOUCHED MINE IN A LONG TIME.."  That would make a great /topic.
23:48.32Lee619Fender: guess i could just try both and see what happens.  :-)
23:48.36[TK]D-FenderLee619 : Double check against this : http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
23:49.06*** join/#asterisk FastJack (i=fastjack@p5091CD8E.dip.t-dialin.net)
23:49.12[TK]D-FenderManxPower: .... go back to your cabin and get coding!
23:49.15Lee619Fender: Thanks
23:49.24tzangerManxPower: hahaha
23:49.29tzangertwisted[asteria]: come on/ topic that
23:49.46Ariel_ManxPower, so you have a montain cabin... hummmm nice.....get away....
23:49.50ManxPowerIt is kind of cool to coast most of the way down the mountian in my car.
23:50.15ManxPowerAriel_, tecnically is's an "RV Camper Trailer", but close enough.
23:50.43Ariel_RV hummm did not figure you for a trailer person...
23:50.51*** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com)
23:51.26*** join/#asterisk r0d3nt_m (i=r0d3nt@tinfoilhat.net)
23:51.45ManxPowerAriel_, This one is rented, still evaluating my long term options for housing at this location.  An actual cabin IS on the list of possibilities.
23:51.55*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
23:52.08Sedoroxyay for more join spam.... same person...
23:52.10rob0but going uphill the poor thing has to pay for the free ride down
23:52.24ManxPowerrob0, *nod*
23:52.43ManxPowerThere's only 2 roads up here, steep and REALLY STEEP.
23:52.53Lee619Manx: what part of the country are you in?
23:53.05*** join/#asterisk Powerkill (n=PoWeRKiL@84.205.154.241)
23:53.17Powerkillhi
23:53.21blitzrageManxPower: thats a nice rumour to have :)
23:53.36ManxPowerLee619, 50 miles NE of Birmingham AL, on "Chandler Mountian" (really should be called "Chandler Mesa")
23:53.50Powerkilli have this probleme http://bugs.digium.com/view.php?id=4096 but don't understand the solution ?
23:54.08ManxPowerUgh.  Looks like I'm installing postgrey tonight.
23:54.30*** join/#asterisk BasketCase (n=BasktCas@asylum.sanitarium.net)
23:54.56ManxPowerI wish my last damn customer of the day would call so I can finish up this trouble ticket and get i the hottub.
23:55.15RoyKwtf is this?
23:55.15RoyKJan 12 00:54:45 WARNING[2834]: frame.c:1239 ast_codec_get_samples: Unable to calculate samples for format unknown
23:55.38sivanaheh
23:55.55BasketCaseanyone ever setup a Handytone 488?  When I call its extension the phone rings but when I pick up no sound is transmitted.  No NAT is involved.  The FXS port isn't configured.
23:56.02ManxPowerMe and of the cats had an argument last night.  She wanted the futon moved back to she thinks it belongs and I wanted to go back to sleep.
23:56.11RoyKif i dial into zap and send dtmf, i get that shite
23:56.21RoyKand dtmf never gets through
23:56.26ManxPowerSo I put in ear plugs so I could not hear her meowing. 8-)
23:56.33*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
23:57.08sivanaRoyK: 1.2?
23:57.12RoyKyes
23:57.19sivanano idea :)
23:57.49ManxPowerRoyK, is the call answered at that point?
23:59.13ManxPowerAriel_, she's not my cat.
23:59.20Ariel_BasketCase, hummm I have never gotten the 488 to work correctly via it's pots port. But the line works fine
23:59.26ManxPowerI did pick her up and scream at her. 8-)
23:59.27Ariel_ManxPower, so

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