irclog2html for #asterisk on 20061027

00:01.29*** join/#asterisk waz- (n=tjs@cpe-75-180-173-103.indy.res.rr.com)
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00:13.54earthsound/msg kram hey, mark, are you around?
00:14.12earthsoundsorry bout that
00:14.24*** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net)
00:14.31JTsomeone needs to resit their irc licence ;)
00:14.35QbYSo..  How many SIP phones (users) are too many of Asterisk?
00:14.50QbYs/of/for
00:15.06JThow long is a piece of string
00:15.40waz-3.85 meters
00:16.09earthsounddouble the length from one side to the middle
00:18.03waz-that's a trick question isn't it!?
00:21.52Drukendelmar: you shouldn't have a problem accepting multipul DID's over 5060...
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00:32.34map7Does anyone here view the Flash Operator Panel on a linux machine?
00:33.29map7I've got a problem where all the phones are flashing green/red all the time on the flash panel.
00:34.20Dovidmap7: it means that it isnt connected to asterisk
00:34.41map7I can view it on windows machines ok.
00:35.29Dovidu can see the calls ?
00:36.19map7on windows using firefox yes I can, anything else (like FreeBSD with firefox/konqueror or IE on windows) I cannot
00:36.32map7a lot of my machines are BSD/linux.
00:37.14Dovidhnn
00:37.17Dovidhmm*
00:37.18map7it loads all the pictures of the phones and text, but the little light which indicates if that phone is on call, keeps flashing green/red
00:37.22Dovidsounds wierd
00:37.40Dovidi only used it in a full windows enviroment
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00:38.07map7does it work under IE for you?
00:38.15Dovidyes
00:38.36map7In IE I only get a black line accross the top of the screen, worse than BSD/Linux browsers
00:38.39*** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net)
00:38.57map7I'm not really worried about IE though
00:40.11Dovidthats all i ever set it up in :(
00:44.50Cinen\
00:45.12DovidIE worked for me on windows
00:46.20map7Just tested IE on another computer and it works
00:46.38map7so it's just viewing it under linux/bsd which is a problem.
00:46.44Dovidso its a linux issue - dont know what it could be
00:46.52Dovidtry thier forum - i think they have one
00:47.08JTit could be the fact that flash majorly sucks
00:47.15Dovidlol
00:47.27*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:47.27map7JT, your right
00:47.33map7I personally hate flash
00:47.42JTmap7: i think there's a dhtml/ajax version of FOP
00:47.45Dovidwhat do u prefer ?
00:47.47JTbut i believe it's more limited
00:47.56JTotherwise look into fixing flash up
00:48.18JTDovid: non-prioprietry junk on the web
00:48.21JTflash is just junk
00:48.32map7David, a fast site with AJAX frontend and Java Servlet Pages at the back
00:48.33*** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net)
00:48.36map7or PHP
00:48.48map7I don't have time for all this animation crap!
00:48.54JTheh
00:49.09JTphp is a scripting system, different to presentation format
00:49.14JTbut i know what you mean
00:49.22JTmap7: try the dhtml version
00:49.29JTi have no idea if it's any good
00:49.34map7I will now I know it exists
00:49.37JTi've used some public online demo of it
00:57.11*** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com)
01:03.02hadsFOP is ming, you could read the source if you wanted to.
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01:21.45_DAWhello mates
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01:28.09DoceSup
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01:37.28apturaCunningPike stick around wont you?
01:37.30aptura:)
01:37.47CunningPikelol - wifi in the hotel blows
01:37.51apturahahah
01:38.06bkw__CunningPike: wayport is great
01:38.06apturahow far is the tranciver?
01:38.10CunningPikeAnd then, just as I get connected, my 24-hour time block expires
01:38.13CunningPikeDunno
01:38.18CunningPikeIt's crap everywhere
01:39.08apturaIts to bad the laws the FCC placed against wifi transmitters could have been changed to allow more power before the law was enacted.
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01:45.18insanity5I know this is off topic... but what is the best solution to keep a single download (tcp flow) from spiking pings up to 300 ms and killing your asterisk server?  T-1 link.  Only 5 users in the office.  Cisco Gear on our side.  Can this be done without the cooperation of the ISP providing QOS/better queueing?
01:48.36benjkaptura, more power on the transmitters would only make things worse
01:49.08orlockblah
01:49.10orlockwireless
01:49.12orlockwireless am the suck
01:49.18apturaohh because of the harmonics and such.
01:49.30benjkinterference would increase
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01:49.56apturaare all wifi channels on the same freq or slightly off.
01:50.21benjkif they'd be on the same frequency they wouldn't be different channels would they?!
01:50.27JTbecause it'd raise the noise floor, most importantly
01:50.34apturaare you talking about CSMA traffic?
01:50.48hoobastoobai have installed and configured the SanDSP for fax. I call in and receive the fax tone, but it does not actually produce a fax. has anyone else used this who may be able to point me in the right direction?
01:51.14benjkhoobastooba, try OpenPBX for fax
01:51.34hoobastoobayou mean instead of SanDSP
01:52.09benjkits SpanDSP and the developer uses OpenPBX for all his testing and integration work, not Asterisk
01:52.32justinu|laptopaptura: you're thinking of timeslots, not channels
01:53.07hoobastoobawell, i know that this works with asterisk
01:53.16hoobastoobaI would prefer to make it work with asterisk
01:53.29benjkthen you're out of luck
01:53.42hoobastoobawhat do you mean out of luck?
01:54.24benjkthe Asterisk version of SpanDSP is no longer maintained
01:55.01bkw__JT: we up stairs
01:55.46JTbkw__: ?
01:55.57bkw__second level
01:56.00bkw__couches
01:56.16bleblebleif none of your feature codes were working and everything was setup in extensions_additional.conf what would you look at next?
01:56.17JTare you talking to me?
01:56.33JTi think you have me confused
01:56.52bkw__yes
01:57.13JTi doubt i'm in the same country as you, even
01:57.30bkw__you're not John Todd?
01:57.40orlockheh
01:58.02bkw__benjk: their is NO version of spandsp for Asterisk
01:58.05bkw__or OpenPBX
01:58.07bkw__its a freakin lib
01:58.43JTno, i'm not
01:59.52*** join/#asterisk togni (n=chatzill@h19-ipv4-80-68-182.mynet.it)
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02:00.14togniHello!
02:00.50benjkbkw, the last version of SpanDSP that was reliably integrated with Asterisk was 0.2
02:00.58benjkand no further work is being done
02:01.47benjkof course you can always do the work yourself
02:02.00benjkbut then you would effectively create a fork of Asterisk
02:02.47benjkso why not use a fork for which other people have already done that work?
02:03.44togniAnyway on latest Asterisk, the only stable version I could integrate in various machines is spandsp 0.0.2pre18
02:04.05togniLatest version hit a lot of crashes in less than 48hours.
02:05.30*** part/#asterisk arcanine (n=arcanine@203.82.44.179)
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02:09.21benjktogni, SpanDSP 0.3 is NOT meant for Asterisk
02:09.30benjkit will not work unless you make changes in Asterisk
02:11.39togniYes, but even something > pre18 give throubles.. I'm not speaking of various sync problems related to various TDM cards and / or drivers, but in real software crashes. When I finnally upgraded to 1.2 two weeks ago from my aging, but very stable 1.0* setup, I faced various interesting troubles, one was that one.
02:13.13benjkwell, it works very well with OpenPBX
02:14.31togniBenjk, OpenPBX have a enhanced SIP channel or it's more or less the same of Asterisk?
02:18.42togniBy a simply diff -u from OpenPBX 1953-RC1 and Asterisk 1.4Beta3 branches I can't see anything really different.
02:19.44benjkOpenPBX still uses chan_sip
02:20.20benjkalthough its received some bugfixes Asterisk's chan_sip didn't and also STUN support and other minor changes
02:20.52benjkeventually this is going to be replaced though, probably with a Sofia based sip channel driver
02:21.11togniI have a question: as no known (by me) sip implementation have VAD, and now Asterisk 1.4 come with a Jitter buffer implementation.. how can a jitter buffer *decrease* in lenght without VAD / silence-cancellation?
02:21.12benjkfor the most important differences see wiki.openpbx.org
02:21.52*** join/#asterisk JohnJacob (n=dhorner@pool-71-127-121-21.aubnin.fios.verizon.net)
02:23.31togniI have another questions about IAX tunking TimeStamp bugs related to comunications where a SIP phone originates a call from an Asterisk box coming to a IAX trunk.. when the SIP callee takes the call on hold then resume it after a while, timestamps become "back in time" (something the IAX protocol prohibits).
02:23.44togniThis bug exists from a while.. but nobody cares.
02:23.46*** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com)
02:24.04togniDo you know if in OpenPBX there's some fix for that?
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02:24.50tmccraryDoes anyone know why Asterisk 1.2.13 keeps saying all my Polycom phones are Unauthorized, yet they work fine inbound and outbound? It's very odd as: A) They work, B) The credentials are correct
02:25.04benjkI don't know, but if you run opbx and ask in #openpbx, you stand a good chance somebody will take it serious
02:26.09togniThank you.
02:26.21*** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net)
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02:44.02tmccrarygood lord Audiocodes gear is terrible
02:45.11BigBadHoss_Workoff-topic, but has anybody here ever used opentaps crm software?
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02:59.15CunningPikeIs freenode fscked?
02:59.32CunningPikeI'm on the Italy node, from Dallas TX
03:00.27tmccraryhmm, seems okay here
03:02.39tmccraryI have a Goto command in my dial plan... but it only evalutes the first entry it goes to and hangs up
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03:04.53tmccraryI have a Goto command in my dial plan... but it only evalutes the first entry it goes to and hangs up
03:04.56tmccrarywhoops
03:06.39*** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net)
03:07.40KuJaXI just installed a wildcard X100P card, edited zaptel, edited zapata and edited extensions.conf and nothing.  When I type ztcfg -vvv it shows it successfully without any errors, and in asterisk CLI I type zap show channels and it shows status OK
03:08.28tmccraryYou mean you don't receive calls when it should?
03:09.10ManxPowerKuJaX: you do not have an X100P.  You have a clone card.  Digium has not sold the X100P in like 2 years
03:09.26ManxPowerJust to be clear.
03:09.41tmccraryKujax what does your dial plan look like?
03:09.44tmccrarypastebin it
03:09.47KuJaXCorrect MaxxPower
03:09.54KuJaX[from-pstn]
03:09.55KuJaXexten => s,1,Answer()
03:09.55KuJaXexten => s,2,Echo()
03:10.05KuJaXHere is from zapata.conf
03:10.08ManxPowertmccrary: I have NEVER seen a "unauthorized" message.
03:10.13ManxPowerUnspecified, yes.
03:10.30KuJaX[channels]
03:10.30KuJaXbusydetect=yes
03:10.30KuJaXbusycount=6
03:10.31KuJaXlanguage=en
03:10.31KuJaXcontext=from-pstn   ; Incoming calls go to [from-pstn] in extensions.conf
03:10.31KuJaXsignalling=fxs_ks   ;  Use FXS signalling for an FXO Channel
03:10.33KuJaXrxwink=300 ; Atlas seems to use long (250ms) winks
03:10.41KuJaX;usedistinctiveringdetection=yes
03:10.41KuJaXusecallerid=yes
03:10.41KuJaXhidecallerid=no
03:10.41KuJaXuseincomingcalleridonzaptransfer=yes
03:10.41tmccraryPastebin it :)
03:10.43KuJaXcallwaiting=yes
03:10.45ManxPowerKuJaX: regardless of your extensions.conf you should see "starting simple switch" on the CLI when a call comes in.
03:10.46JT~pb
03:10.55jbotpb is, like, a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
03:10.55tmccraryYou could get banned :)
03:10.55ManxPower~pastebin
03:10.57jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
03:11.04KuJaXSorry, I will do that. sorry about pasting.
03:11.13ManxPowerKuJaX: are you in the USA?
03:11.14KuJaXMaNxPower - I am not getting "starting simple switch" on the CLI
03:11.40KuJaXwhen I type "zap show channel" it shows pseudo - from-pstn context language en
03:11.45KuJaXYes I am in the USA
03:12.00ManxPowerKuJaX: then remove busydetect and busycount
03:12.29ManxPowerKuJaX: If you plug a regular analog phone into the 2nd port on the card, do you get dialtone?
03:12.44KuJaXLet me try that, I haven't tried plugging an analog phone into the other port.  One sec.
03:12.45*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
03:13.29ManxPowerKuJaX: put your /etc/asterisk/zapata.conf and your /etc/zaptel.conf on pastebin.ca
03:14.42benjkthere is no such thing as clone cards, they are all Ambient MD3200 softmodems
03:15.11benjkand the chips are no longer manufactured
03:15.24benjkAmbient doesn't even exist as a company anymore
03:15.39ManxPowerbenjk: Then why did wcfxo.c have to be updated to see the cards?
03:15.41benjkthe modems sold now are made from refurbed and reject chips
03:16.08benjkbecause it is a PCI feature to use a resistor on the board to set the PCI bus ID
03:16.23benjkyou change that resistor and you get a different PCI ID
03:16.38ManxPowerbenjk: as far as I'm cnocerned if a card has a different PCI ID, and is not from Digium then it is a clone card.
03:17.10ManxPowerheck, if it has the same PCI ID and Digium got no revenue from the card, then it is a clone card.
03:17.22benjkthen you are not using the word clone in its proper English meaning
03:17.49ManxPowerbenjk: Correct.  In proper english usage, clone mean "identical"
03:17.58benjkyou may also call it a credit card if you like and you certainly have the right to call things as you like
03:18.07benjkbut that doesn't mean that it is correct English
03:18.20tmccrarywhy does the goto command suck so bad
03:18.38benjkclone means that it is copied from the original
03:19.05benjkthereby implying that DIgium's Ambient MD3200 modems were originals and any others are copies
03:19.11benjkand this is not the case
03:19.28ManxPowerhttp://m-w.com/dictionary/clone
03:19.30benjkthey are all bulk China electronics, rebadged
03:19.51ManxPowerdoesn't really matter, if it is not supported or sold by by Digium then it is not a Digium X100P.
03:19.51benjkthe original was the Ambient MD3200 from Ambient while Ambient still existed
03:20.09benjkhe didn't say it was Digium
03:20.32ManxPowerbenjk: Who used the X100P part number before Digium?
03:20.33benjkin any event those modems are crap
03:21.05ManxPowerbenjk: the 3 or so Ambient chipset modems I got years ago for $10 each worked fine.
03:21.18benjkyes, the moniker is Digium's moniker, but that doesn't make those Chinese bulk electronics clones
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03:21.38KuJaXOKAY, sorry about that.  Here is my ZAPATA
03:21.39KuJaXhttp://pastebin.ca/223829
03:21.44benjkyears ago, the chips were still manufactured
03:21.48xaiManxPower: in an sip <-> ast <--> iax provider situation, what tools can I use to measure jitter and dropped packets,? is there an easy tool that tells you where the problems orginate?
03:21.59ManxPowerxai: I don't know.
03:22.05ManxPowerKuJaX: did you get dialtone on the 2nd port?
03:22.08benjkbut today, they are made from reject and refurbed chips
03:22.16KuJaXHere is zaptel.conf-  http://pastebin.ca/223834
03:22.18*** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
03:22.26KuJaXManxPower - Yes I did (when I plugged analog phone into 2nd port on card)
03:22.27benjkconsequently the cards you can get your hands on today are all crap
03:22.53ManxPowerKuJaX: you have a typoe "musiconhold=default channel => 1"
03:23.03benjkthe only thing you should be doing with those cards is throw it in a garbage bin
03:23.07ManxPowerKuJaX: let me fix the file for you
03:23.08KuJaXI put that there from a tutorial, it wasn't there before (and still didn't work)
03:23.13KuJaXThank you so much!
03:23.36benjkget a cheap FXO gateway instead
03:23.48benjkSPA 3000 is about 65 USD or so now
03:24.19KuJaXSPA 3000 you can use as a FXO gateway for Asterisk? (meaning, you can use the SPA 3000 to receive and do outgoing calls via Asterisk and not use it as a phone)?
03:24.33KuJaXI bought this card for testing and fun anyway, not production.
03:24.36tmccraryUnder any circumstances DO NOT GET AN AUDIOCODES
03:24.46benjkKujaX, yes you can
03:25.08benjkeven for testing I would recommed to throw them in the bin
03:25.16ManxPowerKuJaX: http://pastebin.ca/223840
03:25.46tmccraryReal digium cards are nice too
03:25.50ManxPowerKuJaX: yes, but the FXO on the SPA-300 is a bitch to get working with Asterisk
03:25.59benjkhuh?
03:26.22JTtmccrary: TDM400P is too expensive for most
03:26.38benjkit takes me 10-15 mins to get an SPA3K going as FXO gw with Asterisk
03:26.50tmccraryyeah, its not cheap, but so far, its been the most reliable type of analog adapter I've used
03:26.59JTheh
03:27.01KuJaXManxPower - alright uploaded an did a "reload" from CLI
03:27.17JTerr i think you need to restart
03:27.33KuJaXwill do a shutdown -r now
03:27.39JTrestart asterisk, if you changed the zaptel channel config
03:27.40benjkalso, the SPA has a power-off-passthrough feature
03:27.40JTnooo
03:27.43JTnot the computer
03:27.45JTjust asterisk
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03:27.59JTa reload is not the same as a restart of asterisk
03:28.03*** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net)
03:28.18KuJaXrestart gracefully from CLI
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03:28.47JTyes that or stop it and start it, they should both work
03:29.15KuJaXgot a "starting simple switch on 'zap/1-1'
03:29.34ManxPowerKuJaX: there you go
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03:29.51KuJaXhow do I hangup on it now?  Now that line is doing nothing (can't get a dial tone via a normal phone)  lol
03:29.59KuJaXbecause the dialplan never had hangup, only echo as last command
03:30.06KuJaXOkay it finally timed out
03:32.41KuJaXwhen I dialed from my cell to the land line, it connected well (did a playback that I just setup) and then dialed a softphone extension.  The softphone started ringing, but said "asterisk" instead of caller ID number
03:33.58ManxPowerKuJaX: callerid=asreceived  I forgot that
03:36.09KuJaXI just added that line in, did a restart gracefully, and still came in as "asterisk"
03:37.57ManxPowerKuJaX: You added it BEFORE the channel => 1 ine?
03:40.36KuJaXcorrect
03:40.44KuJaXright after usecallerid=yes
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03:46.53KuJaXhrrmm after starting simple switch on 'Zap-1-1' it says "notice[4206]: callid.c:322 callid_feed: Caller*ID failed checksum
03:48.45JTKuJaX: # jumps throught the dialplan by default
03:49.19KuJaXlooking online i am seeing possible zaptel CVS complication?
03:54.33kronicI've got a queue setup like this: http://pastebin.ca/223875
03:54.53kroniccalls don't seem to ring the other members, which are all available
04:00.49KuJaXManxPower - Thank you so much for your help.  I will play around with this weekend!
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04:14.42kronicanyone?
04:21.07*** join/#asterisk ShadowHntr (n=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)
04:21.16BigBadHossanybody ever use asterisk on an ultrasparc?
04:22.04benjkwith Solaris?
04:22.20BigBadHosslinux preferred
04:22.36benjkI tried to get a Solaris/Asterisk install going once
04:22.41*** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net)
04:22.42benjkgave up after 3 weeks
04:22.45BigBadHosshaha
04:22.57BigBadHosswhy not try linux
04:23.10benjkJohn Todd told me he was running OpenBSD on those UltraSparcs
04:23.17BigBadHossis it as easy to compile and setup for sparcs as for x86?
04:23.23benjkapparently that works
04:23.42justinu|laptophey benjk, you see that erlang video?
04:23.43BigBadHosswhy not linux
04:23.49benjkyes I did
04:23.57BigBadHosstheyress lots of linux support for sparc
04:24.16benjkwhy do I need a Sparc then?
04:24.29benjkif I want to run Linux I can easily get an x86 box
04:24.29justinu|laptopkinda cool, cutting edge technology when that was filmed
04:24.36justinu|laptopwish I knew about erlang when I was writing excel apps
04:24.58benjkheh
04:25.16*** join/#asterisk zhllg (n=zhangle@static-ip-178-123-134-202.rev.dyxnet.com)
04:25.21BigBadHossi tried to get jabber with erlang
04:25.28justinu|laptopejabberd
04:25.33BigBadHosswent to java instead
04:25.35justinu|laptopgentoo makes that a bit easier
04:25.49BigBadHosswildfire is very nice imho
04:25.59BigBadHossthey even have asterisk integration
04:26.10justinu|laptopi wonder how erlang would handle RTP
04:26.57BigBadHosshmm
04:31.21*** join/#asterisk alerios (n=alerios@190.24.99.75)
04:33.21tmccraryhow do you disable cdr csv? I have postgres going and I don't want the csv anymore
04:34.22Corydon76-homeunload cdr_csv.so
04:34.45tmccraryIs there a way to have that done any time asterisk is started?
04:35.58kronicI've got a queue setup like this: http://pastebin.ca/223875, I'm trying to get circular call distribution working (as you can see), though it only rings the first member listed
04:36.18Corydon76-hometmccrary: modules.conf
04:36.28tmccrarythx
04:37.27*** join/#asterisk bobby1234 (i=erokcxq@ems01.your-freedom.de)
04:46.48*** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com)
04:50.44*** join/#asterisk linlin (n=linlin@71.194.70.13)
04:50.45BigBadHosshey ManxPower
04:50.49BigBadHossu in AL?
04:50.58*** join/#asterisk aao_pwner (n=aao_irss@c-24-21-91-140.hsd1.mn.comcast.net)
04:51.26*** join/#asterisk lorinc (n=ang@caracas-4721.adsl.interware.hu)
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04:56.44benjkis there a LittleGoodHoss, too?
04:57.58BigBadHossnope
04:58.15BigBadHossanybody ever installed opentaps?
04:58.24benjkthat's what I suspected
04:58.33BigBadHossdo they have a chan here
04:58.45benjkhow about HugeEvilHoss?
04:58.58*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:59.02benjkGiantEvilHoss even
04:59.03BigBadHossno, im in the middle
05:11.19kronicshould a call on queue, if not answered be transferred to the next membe
05:11.22kronic*member
05:12.54predderwhere is voicemail data stored? I need to delete all stored voicemail messages
05:14.32kronic/var/spool/asterisk/voicemail/<contexT>
05:14.40predderthanks
05:19.15*** join/#asterisk argos73 (n=argos73@cpe-24-93-180-159.neo.res.rr.com)
05:21.50argos73question - assume a PRI between asterisk and a legacy PBX - 21/23 channels in use.  I issue a multiple destination Dial(Zap...) with 5 destinations to the PBX - 2 can find available channels, three can not.  any idea what happens?
05:23.32bobby1234hello
05:41.41*** part/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com)
05:49.44*** join/#asterisk oej (n=oej@apollo.webway.se)
05:53.58*** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net)
06:02.48*** join/#asterisk xpediant (n=admin@204.8.178.2)
06:15.01*** join/#asterisk cian (n=cian@cian.ws)
06:23.58*** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
06:25.18dzhhi guys! Any ideas why "iax2 show netstat" shows Lost packets when I run IAX-ZAP call ? server is just for test and 0% loaded and has just 2 calls
06:25.30xpediantI'm upgrading a cisco 7940 phone from firmware version 6.3 to 7.4.
06:25.30xpediantThe phone goes through the following:
06:25.30xpediantvlan
06:25.30xpediantip
06:25.30xpediantupgrading firmware
06:25.30xpediantresetting
06:25.32xpediant(phone restarts)
06:25.34xpediant(cycle repeats)
06:25.36xpediantThe 7.4 firmware does not show in status, firmware is still 6.3
06:25.38xpediantAll of the firmware files are copied into tftpboot. Is there a particular log file that might give me some clues to figuring this out.
06:32.18*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
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07:24.15*** join/#asterisk thdei (i=root@212.147.65.151)
07:24.43thdeiHi everybody
07:24.55thdeiI'm looking for a cluster solution with asterisk
07:25.16thdeiI tried DNS SRV, vovida load balancing, Alteon Switch
07:25.29thdeibut nothing very usefull
07:25.44*** join/#asterisk Jubei_ (n=Stormtro@147.27.47.12)
07:25.57*** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net)
07:25.58Jubei_has anybody ever compiles openssl and pwlib on a 2.6+ box?
07:26.01Jubei_compiled*
07:26.03thdeiI can make a cluster with SER with a radius DB but there is no failover issues
07:27.55thdeido you have ideas for me ?
07:32.20*** join/#asterisk [hC] (n=hardcore@12.127.180.58)
07:32.36[hC]anyone noticed issues with distorted audio in idefisk?
07:32.53*** join/#asterisk psk (n=psk@golia.caltanet.it)
07:35.16JT[hC]: i haven't had any problems myself
07:35.43[hC]im gonna try jackeniax.. weird
07:35.50[hC]like a strange timing distortion almost.
07:36.01JTis there any open source iax softphones out there?
07:36.10JTi didn't find any when i looked
07:36.52Jubei_JT, i'm pretty sure there are
07:37.00JTfor windows?
07:37.06Jubei_ye
07:37.38JTthey seem to mostly be based in the iaxclient open source libs, but the softphones all seem to be released under non-free licencing
07:39.12Jubei_check out www.voip-info.org , they have a section where they list tonz of em, i'm pretty sure i've seen a free iax one
07:40.19JTyeah i went right through it, unless one has been added to the listing in the last few weeks...
07:42.18Jubei_ah then i might be wrong. dunno sorry.
07:42.30*** join/#asterisk inspired (n=mikael@85.221.7.59)
07:42.52[hC]fwiw JackenIAX for MacOSX worked well
07:43.00[hC]better than idefisk
07:43.58*** join/#asterisk tlow (n=tlow@tor.blaqhat.com)
07:44.21JTthat'd require Mac OSX ;)
07:46.00[hC]im in luck :)
07:47.19JT[hC]: were you using idefisk under OSX?
07:47.23[hC]yup
07:47.36[hC]seems 1.35 has an audio bug in osx.. i guess.
07:47.39JTah ok, maybe that's why you experienced problems
07:47.41JThmm
07:47.51JTDIAX seems part open source
07:48.04JTstill haven't found a full OSS windows client
07:48.54*** join/#asterisk vaq (n=lars@0x57306388.rdnxx5.adsl-dhcp.tele.dk)
07:48.56vaqHello
07:49.07vaqare there any encryption solutions for Asterisk yet? TLS/SSL ?
07:49.23JTIAX can be run with encryption
07:49.34JTit's about the only native encryption solution
07:49.49*** join/#asterisk Chris-H (n=chris@caitlin.archnetnz.com)
07:49.52JTfor everything else, it requires patching or encrypted UDP tunnels
07:50.01*** part/#asterisk Chris-H (n=chris@caitlin.archnetnz.com)
07:50.08vaqIAX ?
07:50.46JTInterAsterisk eXchange protocol
07:50.51JTan alternative to SIP
07:51.14vaqHmm, never heard of it...
07:51.33*** join/#asterisk Magicianx (n=chezvous@24.122.205.9)
07:51.46jeremy_g"\033[1;35mhaha"vi a.c
07:51.53JTyou musn't have used asterisk much then
07:51.54vaqOh, reading about it.
07:52.20vaqJT: no, i did just a simple asterisk setup with SIP, however i will try IAX now. Which is the best IAX client?
07:52.42jeremy_gis there a way to colorize ur log in asterisk, i dont want to logcolorizer
07:52.52jeremy_glike replace Dial with green colored Dial
07:53.05jeremy_gin /var/log/asterisk/messages
07:53.23JTvaq: i've had good success with Idefisk, but not sure if it supports encryption
07:53.41JThttp://www.voip-info.org/wiki/view/Asterisk+IAX+clients
07:56.10*** part/#asterisk tlow (n=tlow@tor.blaqhat.com)
08:01.13jeremy_gtail -n 50 /var/log/asterisk/log_notice |replace NOTICE `echo -e "\033[1;35mNOTICE"`
08:01.23jeremy_gthis line showed all file as pink
08:01.34jeremy_gsorry yellow
08:04.39*** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net)
08:05.08E-bolaAnybody else have tried having problem with register lines?
08:05.14E-bolasometimes mine stalls with request sends?
08:05.18E-bolaand i cant receive incomming calls
08:05.48*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)
08:11.47*** join/#asterisk qdk (n=qdk@193.164.155.35)
08:13.15dzhAnyone can explain output of IAX2 JB DEBUG ?
08:13.23*** join/#asterisk pigpen (n=mark@fw.seamans.cc)
08:14.44*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
08:17.18*** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net)
08:21.48vaq<PROTECTED>
08:21.58vaqcan't seem to find a IAX client that has encryption functions?
08:22.08Arnarseen Ambriento
08:22.12vaqbut does this mean anything if the server has encryption under the IAX account?
08:22.32Arnarhey ppl.. is there a bot here?
08:26.25jmlsall your bots are belong to us
08:26.39Arnar:)
08:28.18*** join/#asterisk xpediant (n=admin@204.8.178.2)
08:31.25vaq*CLI> Oct 27 10:30:34 NOTICE[27547]: chan_iax2.c:3924 register_verify: Peer 'larsiax' is not dynamic
08:31.28vaqwhat does this mean?
08:34.40hegemoOnttp://www.xml-dev.com/
08:34.58vaq?
08:41.43*** join/#asterisk qdk (n=qdk@193.164.155.35)
08:46.32Jubei_guys "chan_h323.so" is looking for another module to load, where should I put that module so that it's found?
08:47.47Jubei_[chan_h323.so]Oct 27 07:46:49 WARNING[19687]: loader.c:325 __load_resource: libpt_linux_x86_r.so.1.9.0: cannot open shared object file: No such file or directory
08:48.17Jubei_i have that libpt module, where must I copy it to so that asterisk finds it?
08:50.59kaldemarldd <module>
08:51.04Jubei_huh?
08:51.29kaldemarman ldd
08:51.48kaldemarit's command that gives you library dependencies of a module. with a path.
08:51.59Jubei_but isn't that for kernel modules etc?
08:52.43vaqexten => 10,1,VoiceMailMain(210@default)
08:52.48vaqwhen i call the number 10 i get:
08:53.03vaq<PROTECTED>
08:53.04kaldemarmy chan_h323.so is satisfied with libpt in /lib/
08:53.07vaqhow come?
08:54.20Jubei_kaldemar: ok i'll try putting it in there, thanks.
08:54.24vaq?
08:55.08Jubei_kaldemar: it worked :D thanks!
08:55.21Jubei_kaldemar: i wonder though why make install on asterisk didn't copy those dependencies there too
09:00.03vaqanyon?
09:04.08*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
09:04.40vaqexten => 10,1,VoiceMailMain(210@default)
09:04.42vaqwhen i call the number 10 i get:
09:04.44vaq-- Got SIP response 404 "Not found -- unknown service number" back from
09:04.48vaq(setting up a voicemail)
09:06.05shellsharkerr
09:06.10shellsharkVoiceMailMain()
09:06.11vaq?
09:06.19shellsharkshouldnt need any arguments there
09:07.00*** join/#asterisk rkr245 (n=ravi@cw.callsat-telecom.com)
09:07.04vaqhow should it bee then shellshark  ?
09:07.17*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
09:07.19shellshark09:06 <shellshark> VoiceMailMain()
09:07.38vaqhmm okay
09:08.19vaqsame error
09:08.35vaqbut see this:
09:08.37vaq<PROTECTED>
09:08.43kaldemarif you leave the parameters out, the application will prompt the caller for a mail box.
09:08.52vaqit's trying to call number 10 trough my VoIP Provider.
09:09.01vaqit should call it local
09:09.15shellsharkthen your VoIP provider's pattern matches first ;)
09:09.24shellsharkshow us your VoIP provider Dial statement
09:09.31hwthow do i chop off the first 4 digist of a number and the last one as well in the dialplan?
09:09.57hwti want the XXXX: ####XXXX#
09:10.06vaqmy extensions.conf shellshark  ?
09:10.45shellsharkvaq: yes
09:11.05*** join/#asterisk RoyK (n=roy@80.239.107.70)
09:11.35vaqhttp://pastebin.ca/224271
09:11.41vaqthere you go shellshark
09:11.45hwtanyone?
09:11.48*** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net)
09:13.01FlatFootmorning all
09:13.27FlatFootjust about to order a new dell for * any recommendations ie model etc
09:13.36vaqshellshark: ?
09:14.23jeremy_gwhere do i setting a higher debug level in asterisk
09:14.33jeremy_gi dont want to type set debug 24 every time at cli>
09:14.52jeremy_gasterisk -d ??
09:14.58shellsharkFlatFoot: HP ;)
09:15.07vaqjeremy_g: /etc/default/asterisk
09:15.15vaqshellshark: did you see the extensions.conf?
09:15.20shellsharkvaq: yeah man
09:15.41jeremy_gvaq:what do i do there man?? :>
09:15.52jeremy_gwhat variables to set
09:15.54shellsharkvaq: your musimi_outgoung is VERY vague
09:16.02shellsharkvaq: you in the US?
09:16.43vaqshellshark: no Denmark, what do you mean by vague?
09:16.44FlatFootshellshark: why HP ? prob is we have account at dell and are sposed to use them only
09:16.55jeremy_gcmon vaq tell me
09:17.13vaqjeremy_g: PARAMS="-g -vvvvc"
09:17.29*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)
09:17.33joelsolankiHello all.
09:17.39shellsharkFlatFoot: ah... i've just always had MUCH better luck with HP servers
09:17.54shellsharkvaq: "_X." should be a lot longer ;)
09:17.58joelsolankii m facing problem with installing zaptel on centos 4.4
09:18.06*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com)
09:18.09joelsolankilet me pastebin the error
09:18.11vaqshellshark: what should it be?
09:18.21shellsharkvaq: for example, in the US we would use something like "_NXXNXXXXXX"
09:18.40shellsharkvaq: longer than your local extensions, that's for sure ;)
09:19.09vaqshellshark: so i should write "_NXXNXXXXXX" instead of "_X."
09:19.47shellsharkvaq: i was telling you how we would do it in the US
09:20.00shellsharkvaq: i have no idea how the numbering system works over in denmark ;)
09:20.40*** join/#asterisk Ahrimanes (n=michael@81.7.159.2)
09:20.46vaqshellshark: hehe okay, it works.. with _X..... But what is wrong with my voicemail dial?
09:20.59Ahrimanesanyone using cisco phones?
09:21.45kaldemarhwt: ${EXTEN:0:${LEN(${EXTEN:1})}} would chop the last number. if you come up with a solution to remove the first 4 on the same line, please share. :)
09:22.43jeremy_gahh did it! watchdog started with -d ands lots of vvv
09:23.05vaqshellshark: Hm?
09:23.35shellsharkvaq: should work
09:23.43shellsharkvaq: you try with no arguments?
09:24.13vaqyes
09:24.22shellsharkwhat error did you get?
09:24.46vaq<PROTECTED>
09:24.49vaq<PROTECTED>
09:24.51vaq<PROTECTED>
09:24.54vaq<PROTECTED>
09:25.02hwtkaldemar: i will, thanks.
09:25.08Ahrimanesvaq: danish are we?
09:25.15vaqAhrimanes: yes
09:25.19Ahrimanesvaq ;)
09:25.22key2is G711 the best quality we could find ?
09:25.26key2since it's not compressed
09:25.46shellsharkvaq: err, i told you to change your musimi_outgoing dial statement!
09:26.15vaqshellshark: no you didnt.
09:26.22vaqshellshark: what should i change?
09:26.25Ahrimaneskey2: provided you have around 80k of andiwdth per channel, yes g711 would be the best quality
09:26.25JTkey2: yes
09:26.44Ahrimanesman my spelling is bad today
09:26.45shellshark09:17 <shellshark> vaq: "_X." should be a lot longer ;)
09:26.52key2Ahrimanes: why is it said that g722 is better quality
09:26.54shellsharkvaq: did you miss that?
09:27.00JTkey2: what?
09:27.07key2Ahrimanes: aparently it uses the same bw but it's compressed
09:27.16JTg.711 is the best quality codec for 8k voice
09:27.22Ahrimaneskey2: hm havent read about 722.. but also havent seen support for it in  *?
09:27.25JTeverything is converted back to it
09:27.33JTso it's impossible for anything else to be better
09:27.34AhrimanesJT: 722 is wideband?
09:27.38JTmaybe
09:27.43JTasterisk doesnt do widepand
09:27.46JTwideband
09:27.47*** join/#asterisk qdk (n=qdk@213.150.62.32)
09:27.54JTwe're talking about PCM8000
09:28.27JTalso, telco networks use PCM8000, so wideband won't help if you have to interconnect with one
09:28.54Ahrimaneskey2: g722 is 16khz.. so sound quality is better yes, but i doubt it that * or any current endpoints really support it
09:29.06vaqshellshark: but why should that help
09:29.23key2Ahrimanes: ah ok
09:29.26vaqshellshark: could you paste your setting instead of _X. again i cant scroll up (Screen)
09:29.28Ahrimanes_X. <- matches a lot....
09:29.40*** join/#asterisk xnon (i=xnon@200.8.30.50)
09:30.31vaqAhrimanes: are you also using _X.
09:31.04qdk_X. is just stupid in a real setup.
09:31.09Ahrimanesvaq: no, i tend to use longer patterns to avoid problems
09:31.14shellsharkvaq: please read how dialplans work and how patterns operate ;)
09:31.31vaqAhrimanes: which do you use?
09:31.43Ahrimanesvaq: depends on what i'm trying to accomplush
09:31.54shellsharkvaq: like i said, normally in the US we use something like _NXXNXXXXXX
09:32.08shellsharkvaq: of course you're confused, you havent read any docs ;)
09:32.09vaqshellshark: but HOW could this affect voicemail ?
09:32.23vaqshellshark: i did read docs on how to setup asterisk, IAX, and the voicemail.
09:32.39shellsharkvaq: you are including the outbound context before your context containing the voicemail extension
09:33.22vaqshellshark: yes
09:33.29Ahrimaneswhich means _X. steals the call
09:33.37shellsharkvaq: and since _X. will match ANYTHING, and it comes before your voicemail extension, it gets priority
09:33.37*** part/#asterisk joelsolanki (i=joelsola@202.160.161.94)
09:33.47vaqahhhh
09:33.53vaqokay, will try to use _NXXNXXXXXX
09:34.47vaq*CLI> Oct 27 11:34:09 NOTICE[28253]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 213.237.44.34, request '10@dialout' does not exist
09:35.15vaqshellshark: ?
09:35.16Ahrimanesvaq: _NXXNXXXXXX is a pattern for the US.. you need one for dk....
09:35.35vaqye
09:35.51vaqAhrimanes: could i copy one of yours since your in the DK?
09:36.43Ahrimanesvaq: not relly, all my * are using least cost routing from a database currently.. but read up on the dialplan, it's not all that hard to get
09:36.49shellsharkvaq: read the documentation and you could make your own ;)
09:37.36kaldemarhwt: ${EXTEN:4:${LEN(${EXTEN:5})}} <-- that will take the first 4 off too...
09:38.47vaqAhrimanes: https://musimi.dk/index.php/wiki/p/Vejledninger/KonfigurerAsterisk
09:38.55vaqAhrimanes: they use the same
09:39.37qdkvaq: dont believe everything you read.
09:39.54Ahrimanesvaq: sure, but they dont have any other extensions configured.. if all you want is for your asterisk to send everything to musimi, that's fine.. but if you have local extensions.. its not all that good
09:40.10shellsharkvaq: they are assuming you'll only use thier VoIP service, and not do anything else with your Asterisk box ;)
09:40.24qdkvaq: musimi is  lowbudget newbie system, but for people somewhat skilled.
09:40.46vaqtrue
09:40.56kaldemarvaq: include the musimi context as last, there's a simple solution for you.
09:41.21kaldemarvaq: if you want to send everything except your local extensions there.
09:41.28qdkkaldemar: yes, if either first match or best match works as intended.
09:41.40vaqso context=incoming will make it work with _X. ?
09:42.13shellsharkugh
09:42.17qdkvaq: wrong, _X. is for your outgoing context.
09:42.59vaqyes
09:43.10qdkvaq: incoming will be whatever phone no. musimi gave you... and extens of your own imagination.
09:43.20vaqyes
09:43.22[hC]Anyone using faxdetect?
09:43.35vaqqdk: however, how do i find out what i should use instead of _X. ?
09:44.05qdkvaq: i would probably use somethnig like _XXXXXXXX.
09:44.36vaq*CLI> Oct 27 11:43:58 NOTICE[28387]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 213.237.44.34, request '10@dialout' does not exist
09:44.44vaqwith:
09:44.44vaq[outgoing]
09:44.45vaqexten => _XXXXXXXX.,1,Dial(Sip/musimi/${EXTEN},120)
09:44.45vaqexten => _XXXXXXXX.,2,Congestion
09:44.52qdkvaq: there is no reason to send anything less that 8 numbers to musimi, as they just route it out through their PSTN provider.
09:45.51vaqbut still doesn't work.
09:46.01qdkvaq: thats seems correct... perhaps use the 'n' priority after the 1 priority.
09:46.17vaqhuh
09:46.27qdkvaq: could be your SIP account or typo or a lot of things.
09:46.37vaqim using IAX
09:46.41kaldemarvaq: did you just try to access the voicemail when you got that NOTICE?
09:47.22vaqkaldemar: what do you mean by that?
09:47.54kaldemarvaq: errr.. what did you do to get that error message you posted last? dial 10?
09:48.04vaqdialed 10
09:48.08vaqnumber 10.
09:48.42kaldemarok, so asterisk was searching for 10 in context dialout. it has nothing to do with what you have in outgoing.
09:49.27vaqkaldemar: should i place the exten under dialout?
09:50.07kaldemaryes, or include the context in dialout that has your voicemail extensions.
09:50.44kaldemaryou should obviously do some studying on the dialplan structure.
09:50.58vaqOct 27 11:50:18 NOTICE[28501]: chan_iax2.c:2455 iax2_read: I should never be called!
09:51.25*** join/#asterisk Dragonmen (n=dragonme@212.200.115.53)
09:51.28Dragonmenhi
09:52.04*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)
09:52.17vaqkaldemar: hm?
09:52.19Dragonmeni have a problem with packet rate
09:52.29Dragonmenwhen using sip protocol
09:52.36Dragonmenit's over 50 packets/sec
09:52.43Dragonmenand we have wifi here
09:52.48Dragonmenso it's an issue
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09:52.57kink0hello
09:53.08key2Dragonmen: change ur codec
09:53.20chapeaurougehi.. got a pb with isdn-bri.. wether the led is red or green on my card (quadbri), pri show span 1 always show the status as down, even though provisionned and Active. All the rest looks absolutly normal, ztcfg -vv is fine, etc...
09:53.39vaq<PROTECTED>
09:53.41vaqhmm
09:53.52kink0I originate a call to a PSTN phone from my Asterisk, and then I need to capture the DTMF key pressed on the called side. Any sugestion ?
09:54.41Dragonmenkey, with every codec it's the sam problem
09:55.01Dragonmeni tried gsm also
09:55.09Dragonmeni need an feature
09:55.20Dragonmento limit the number of packets/sec
09:55.33Dragonmeni searched the net
09:55.52Dragonmenand didn't find any solution
09:56.15kink0Dragonmen: see for "traffic shaper" on freshmeat.net , or may be iptables will be enough for you
09:56.21vaqkaldemar: works now but it asks me to dial my local number, is that the number that the mailbox has been allocated for?
09:56.36Dragonmenkink0, that would limit the bandwidth
09:56.38Dragonmenbut
09:56.51kink0those are ok in the event you want to limit traffic on you Linux box, but is better if you have some Cisco or similar router in your network
09:56.53Dragonmenthe problem will be the same
09:57.04Dragonmeni have mikrotik
09:57.08kink0no just bps also packets
09:57.09Dragonmenbut that's not the point
09:57.26kaldemarvaq: yes, you removed the parameters from VoiceMailMain, right?
09:57.34Dragonmenthe client packet rate is limited to 50 packets/sec
09:57.40Dragonmenand asterisk force it to 80
09:57.49vaqkaldemar: yes
09:57.58Dragonmenso limiting on traffic shaper will not do the job
09:58.12Dragonmeni need to limit it on asterisk
09:58.31kink0Dragonmen: ahh ok, I see, then configure your codec, I did that sometime, but I don't remember how did it
09:58.43kink0I did for g729 only
10:00.16kink0about DTMF... I am able to do Read() when my asterisk gets the call, but no when my asterisk originate the call, because while Dial() is not possible to execute next priority
10:00.36vaqkaldemar: when i write: Oct 27 11:59:46 WARNING[28661]: app_voicemail.c:3389 vm_execmain: Unable to read password
10:04.10vaqOct 27 12:02:39 WARNING[28689]: app_voicemail.c:3389 vm_execmain: Unable to read password
10:04.13vaqUnable to create lock file: Permission denied
10:04.18vaqhmm, how do i fix his?
10:05.43vaqah the client has problems.
10:08.24*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:18.14xpedianthi...  I've been trying to get asterisk up and running for a couple of weeks.  I'm struggling through this and I wanted to know if there were any specific places that I might be able to learn a better base of asterisk other then manual, which I have been reading.  Suggestions are apreciated.
10:21.27Jubei_guys I'm trying to setup an addon that astbill requires, "res_config_mysql". I downloaded asterisk-addons-1.2.5 and did a make in the directory and then a make install. what do I need to do to make sure asterisk loads that certain module?
10:22.15EyeCue~book
10:22.17jbotsomebody said book was a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
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10:34.43pifxpediant : unless your time costs nothing, hire a consultant
10:35.16*** part/#asterisk Ahrimanes (n=michael@81.7.159.2)
10:35.17xpediantpif: knowledge is power
10:35.23qdkxpediant: what EyeCue said.
10:35.49xpediantjbot: I'll check that out, I apreciate it.
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10:36.16pifa consultant will _show_ you how things are done
10:36.37pifelse spend more weeks struggling....
10:37.20xpediantI can handle struggling I was just shooting for pointers, struggling has it's downside but it's o so sweet once you get it
10:37.54JTxpediant: know about voip-info.org?
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10:41.54xpediantJT: I've been reading a lot there, the biggest problem I have found is that my base knowledge is lacking, voip-info.org strikes me as more for people who have a decent understanding and while it hasn't   not been helpful there are a lot of concepts that I just don't know. I like to understand WHY things work the way they do and not just that they work
10:44.01JTi guess you need to keep reading Asterisk: TFOT then :)
10:46.03RoyKTFOT?
10:47.43JT~thebook
10:47.45jbotfrom memory, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
10:49.19RoyK~pebkac
10:49.21jbotSounds like the Problem Exists Between Keyboard And Chair
10:50.57*** join/#asterisk festr__ (n=festr@ns.regnet.cz)
10:51.37festr__hello, is it possible to dynamic change of codec at established IAX call?
10:51.57xpediantI'm actually browsing through the book right now,I hadn't seen anything about this yet shockingly enough, I think this is exactly what I was looking for
10:52.10RoyKfestr__: I seriously and utterly doubt so
10:53.19festr__RoyK: btw did you seen patches which i've send you as you requested?
10:53.19festr__s/did/have
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10:53.21Dragonmenhi
10:53.26Dragonmeni got power outage
10:53.36Dragonmeni was adking about packet rate for sip
10:53.40Dragonmen*asking
10:53.50Dragonmen50 packets/sec on wifi is too much
10:54.03Dragonmenis there a way to solve this ?
10:54.19RoyKfestr__: yes, but haven't tried them yet
10:54.49RoyKfestr__: one question: did the jitterbuffer/plc log as usualy, only not actually use the PLC?
10:54.59festr__RoyK: yes
10:56.02RoyKhm. perhaps I'd better try them, then :)
10:57.03festr__RoyK: i will try to explain it: when frame is lost jitter buffer send frame with zero datalen and log to file it will be interpolated. but it depends on used codec if PLC is really done
10:57.31festr__RoyK: but this is not logged if it is really done or not. if you use iLBC -> zaptel, plc is OK. this PLC issue is only when alaw -> zaptel
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11:02.59pjohi all, any recomendations for a Wi-Fi SIP phone? I'm currently looking at the WIP300 from Linksys and the DPH-540 from D-Link. Any other ideas or preferences of one over the other?
11:03.31festr__RoyK: the main trick is that zaptel is forced to use SLINEAR so when negotiating bridged channels alaw is translated to slin and PLC works only if transcoding (every transcoding is from source to SLIN and SLIN to destination)
11:03.38Dragonmenpjo, it looks like u can forget about it on 2.4 wifi
11:03.55pjoDragonmen: why?
11:04.08Dragonmentoo much packets
11:04.16Dragonmeni have the problem with it
11:04.34shellsharkDragonmen: works fine for me ;)
11:05.04Dragonmenfor me it doesn't
11:05.18shellsharki setup a 40 acre campus (huge car lot) with ~20 access points all in bridge mode with the same ESSID, even roaming and handoff is seamless
11:05.27pjoshellshark: which? the linksys or the d-link? Dragonmen: okay, any other options in terms of wireless SIP?
11:05.31festr__RoyK: i mentioned dynamic IAX codec change. i've modified 1.2 and working experimental dynamic codec changes. it is easy to integrate.
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11:05.58pjoDragonmen: (from the perspective of someone who has to build the network as well)
11:06.18shellsharkpjo: i dont use either of those, and because of an NDA I'm under, I can not disclose the certain manufacturer we went through to get the phones custom made ;)
11:06.19festr__RoyK: but i'm not sure if this feature is in trunk (it is not i've tested it and it does not work) but dont know if there are some new options
11:06.47pjoshellshark: understood. thanks.
11:07.28shellsharkpjo: but i would discard Dragonmen's statement, as it is very possible
11:07.30RoyKfestr__: but this looks like it's breaking usage of other codecs.....
11:07.49RoyKdeflaw = AST_FORMAT_SLINEAR around line 5020 in chan_zap.c
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11:08.12festr__RoyK: yes, its correct. you can pass alaw or slinear to zaptel
11:08.22festr__RoyK: its releated only to zaptel
11:08.43pjoshellshark: I do use wiFi (on my laptop and ipaq) to do SIP calls, I just need a wifi phone.
11:08.45festr__RoyK: it does not mean that you cannot use ilbc -> zaptel
11:09.48festr__RoyK: when you set deflaw AST_FORMAT_SLINEAR | AST_FORMAT_ALAW and RTP frames are coming also alaw there is no translation (zaptel can natively handle slinear alaw and ulaw) and thus no PLC
11:09.58shellsharkpjo: i can't comment on linksys nor dlink wifi phones, as I've not used them
11:10.08RoyKfestr__: but this overrides all those checks above. doesn't this break things?
11:10.17shellsharkpjo: the UTstarcom clamshells are decent in my experience
11:10.22festr__RoyK: tested in production
11:10.24shellsharkF3000 i think is the model
11:10.27shellsharkcheck those out
11:10.35RoyKhm. i'll check
11:10.35festr__RoyK: tens simult. calls no issues
11:10.44pjoshellshark: thanks. will look those up as well
11:11.03festr__RoyK: also look at translate.c
11:11.30RoyKwhere in it?
11:11.40festr__RoyK: there is some racecondition if severeal frames comes from RTP arrives at the same time (which jitter network does)
11:11.59RoyKyour patch doesn't touch translate.c
11:12.06festr__RoyK: aha
11:12.19festr__RoyK: so i send you only zaptel
11:12.46RoyKsend me the full patch again, please
11:12.53RoyKas an attachment, please
11:12.53JTRoyK: pebkac with relation to what?
11:13.48RoyKJT: none particluar
11:13.58JTi see
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11:16.39Dragonmen2.4GHz AP is limiting clients to 50 packets
11:16.41Dragonmenso
11:17.09Dragonmenif voip is run below 50 packets u will get small pauses while talking
11:17.43Dragonmenand that will distrupt talk
11:18.10shellsharknever had any problems
11:18.20shellsharkperhaps you have a crap AP :)
11:18.22festr__RoyK: check mail
11:19.36festr__RoyK: this modifies translator path (its not real fix for RTP racecond.). old behaviour: RTP->frame -> translate -> put to jitter
11:19.57festr__RoyK: this change cause: RTP->frame -> put to jitter -> translate
11:26.38RoyKfestr__: can you please email me the whole patch as an attachment? email creates line breaks and all sorts of shite?
11:27.57festr__RoyK: it IS attachment
11:28.15festr__RoyK: if you see it in message it does not mean it is not attachment
11:29.20festr__RoyK: also previous patch is attachment pls confirm
11:31.10RoyKit's inline text
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11:33.08festr__RoyK: interesting becuase my thunderbird shows it as attachment
11:34.11festr__RoyK: and i also put it as attachment
11:34.38festr__:)
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11:37.40RoyKfestr__: erm
11:37.41RoyK+       if (ast_set_read_format(peer, src) < 0) {
11:37.41RoyK<PROTECTED>
11:37.50RayJWPihas somebody ENUM and e164.org lookup running? asterisk 1.2.9
11:37.53RoyKthe ast_log should perhaps use the src, not dst?
11:39.51*** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il)
11:40.00Dovidmorning everyone
11:40.14RayJWPihi Dovid
11:41.07RoyKfestr__: still no translate.c patch. should I have that?
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11:41.40festr__RoyK: i'm sorry, i dindt mean translate.c this change is in channel.c which i've sended you... :(
11:41.41RayJWPimorning Dovid ... it is 13:41 in Germany
11:41.57Dovidsame time for me
11:42.07Dovid(in Israel) but morning is when ever i wake up ;)
11:42.31RayJWPiok ;-)
11:42.44festr__RoyK: hmmm i'm idiot.... this change is already in full patch
11:43.02festr__RoyK: delete my last mail
11:43.24festr__RoyK: i've not touched translate.c. i've only confused you :)
11:44.37RoyKok. got it then
11:44.42festr__RoyK: if you could do tests of there changes, pls use "tc qdisc add dev eth0 root netem delay 0 100ms"
11:44.48festr__s/there/these
11:45.05festr__RoyK: you can hear big differences
11:45.45festr__RoyK: and also packet loss "tc qdisc change dev eth0 root netem delay loss 10%"
11:45.46DovidRoyK: We dont like trolls here ;)
11:46.19RayJWPiok cu ... :-(
11:46.28RoyKfestr__: what's that supposed to do?
11:46.42festr__RoyK: http://linux-net.osdl.org/index.php/Netem
11:46.47*** part/#asterisk RayJWPi (n=RayJWPi@pD9E83F78.dip0.t-ipconnect.de)
11:47.01festr__RoyK: this sch_netem.ko is in vanilla kernel
11:47.12festr__RoyK: it should be the part of most linux distributions
11:47.39festr__RoyK: ok then include netem scheduler
11:48.19festr__RoyK: just make menuconfig press "/" and search for netem, it should navigate you enable this
11:49.12festr__RoyK: it's good to mention, that "tc qdisc add dev eth0 root netem delay 0 100ms" will cause latency to outgoing packets only
11:49.47festr__RoyK: if you want to test incoming jitter you have to enable it on the other side
11:50.01festr__RoyK: or put router between test asterisks
11:50.13RoyKhow can I put a qdisc on a particular IP?
11:51.18festr__RoyK: check http://linux-net.osdl.org/index.php/Netem at the bottom
11:52.03RoyKthat tc syntax reminds me of configuring sendmail by hand :P
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11:52.57festr__heh
11:53.20festr__RoyK: you can also use iptables -t mangle -I PREROUTING ... -j CLASSIFY minor:major
11:53.29festr__RoyK: instead of using tc filter
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11:55.13festr__RoyK: btw do you know if digium g.729 does PLC?
11:55.17RoyKso 'tc qdisc add dev eth0 root netem delay 0 100ms' and 'tc qdisc change dev eth0 root netem delay loss 10%'
11:55.22RoyKno they don't
11:55.32festr__intels implementation does
11:55.36RoyKi've been emailing them about it
11:55.53RoyKbut I guess it needs to interact with the jb?
11:55.55RoyKno?
11:56.06festr__RoyK: original intels impl does not do plc
11:56.13festr__RoyK: i've to modified it to use it
11:56.16RoyKso 'tc qdisc add dev eth0 root netem delay 0 100ms' and 'tc qdisc change dev eth0 root netem delay loss 10%' <-- but then what'll the syntax be to do this to only one IP?
11:56.17RoyKah :)
11:56.18RoyKnice
11:56.21RoyKme have?
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11:56.25festr__RoyK: this PLC totaly rock
11:56.31festr__RoyK: better than ilbc
11:56.35festr__RoyK: a little bit
11:56.37RoyKnice
11:57.04festr__RoyK: on 25% loss you hear nice interpolated sound
11:57.10RoyKhehe
11:57.20RoyKbut can you help me with the tc above, please?
11:57.41festr__RoyK: i've send email to creator of intel g.729 but with no response :(
11:57.46festr__RoyK: tc.. sure wait
11:57.58RoyKI want this done only for one IP and only for the RTP traffic
11:58.06festr__so
11:58.18RoyKusing iptables and/or tc or whatever
11:58.27RoyKsch_netem is loaded
11:58.39festr__i give you example only for particular IP with tc filter:
11:58.50festr__tc qdisc add dev eth0 root handle 1: prio
11:59.06festr__tc qdisc add dev eth0 parent 1:3 handle 30: netem loss 10%
11:59.19festr__tc qdisc add dev eth0 parent 30:1 tbf rate 100Mbit buffer 1600 limit 3000
11:59.32festr__tc filter add dev eth0 protocol ip parent 1:0 prio 3 u32 match ip dst 65.172.181.4/32 flowid 10:3
11:59.36festr__thats all
11:59.53festr__after that you can modifie paramters to netem like this:
12:00.06festr__tc qdisc change dev eth0 parent 1:3 handle 30: netem loss 0%
12:00.07festr__or
12:00.17festr__tc qdisc add dev eth0 parent 1:3 handle 30: netem delay 0ms 100ms loss 10%
12:00.37festr__this last example will simulate jitter from 0 to 100ms and loss 10%
12:01.27festr__if you want only udp protocol change tc filter...proto udp
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12:02.15festr__s/qdisc add/qdisc change
12:02.40festr__once qdisc is added you have to : tc qdisc change instead of add
12:06.21RoyKthanks. i'll try this
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12:06.47festr__RoyK: test these patch pls
12:07.09RoyKI am.....
12:10.00festr__RoyK: test it without and with and compare results
12:10.39Murdock_Anyone experienced intermittent failure of commands in AGI scripts in 1.2.12.1?
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12:47.22Murdock_ob_implicit_flush(true);
12:47.29Murdock_that was it...dammit
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12:51.56festr__RoyK: btw it is possible to byr licence from vonage and use intel codec legally?
12:52.33festr__RoyK: off caurse bye intels libraries
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12:56.40lero_hi
12:56.45RoyKfestr__: don't think so. anyway can I have that intel code with your fixes?
12:56.51lero_how can i remove this warning messageS?
12:56.51lero_Oct 27 09:56:37 WARNING[8703]: chan_unicall.c:2644 handle_uc_event: Unicall/66 event Connected
12:58.00coppicefestr__ I don't know if its correct, but someone said voiceage have a licencing scheme now for people who want just a few channels of G.729
12:58.01RoyKlero_: s/asterisk//gi will remove them
12:59.38lero_but i'm in console of asterisk :~
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13:00.49festr__coppice: cool to know thanks
13:00.57festr__RoyK: ok
13:01.44festr__coppice: i'l contact them directly
13:02.07b11dmorning all
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13:02.53b11dbrb
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13:14.08vaqHello, i have two questions: 1 : Im using IAX with my Asterisk can the encryption be higher than 128bit? (encryption=aes128) - 2 : How can i doublecheck that my asterisk traffic really is encrypted? (See if it's working)
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13:24.55lero_http://pastebin.ca/224550
13:25.06lero_why it's saying non-zero and ZOMBIE?
13:26.53CunningPikelero_: afaik, all that 'non-zero' means is that the priority that the macro exited at was greater than zero - I could be way wrong there - at any rate, every macro we have ends 'non-zero'
13:27.22CunningPikelero_: I'm not sure that zombies are a concern either - we get the occasional one - it doesn't seem to cause problems
13:29.26lero_hmm
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13:39.18lero_CunningPike: hey, could you look this please? http://pastebin.ca/224571
13:41.47CunningPikelero_: 'Executing GotoIf("Local/3083@padrao-0e56,2", "1?bsy") in new stack' - what does that GotoIf look like in your dialplan?
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13:44.39RoyKfestr__: ping
13:44.54festr__RoyK: pong
13:45.04pingwin|workfrick
13:45.05pingwin|workfrack
13:45.30pingwin|worksyn
13:45.31RoyKfestr__: I keep getting some strange behaviour here. it seems I get quite variable latency
13:45.33pingwin|workack
13:45.37RoyK~lart pingwin|work
13:45.56festr__RoyK: could you describe it?
13:46.27RoyKI'll try. Just need more testing first.
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13:49.10lero_CunningPike: in extensions.conf?
13:49.21CunningPikelero_: Aye
13:49.56CyonUpgraded my servers over to 1.2.13 last night, suddenly I'm getting reports of audio cutting out for a few seconds here and there....is this known is any way?  (I didn't check bug reports yet, moving there now.)
13:50.11*** join/#asterisk seele_ (n=juliangu@64.76.191.12)
13:51.13lero_CunningPike: hm.. the conf here is extended.. what can i search for to facilitate ?
13:51.48CunningPikelero_: That line - I'd like to see the line of code that determines whether or not to go to your 'bsy' label
13:52.18seele_please help my log grows without control
13:52.42nortexseele_, look at the wiki for logger rotate
13:54.13lero_CunningPike: right, wait a minute =]
13:54.34seele_nortex, logger rotate is enabled
13:54.35CunningPikeAstricon hasn't started yet anyway
13:54.47caio1982how could I notify a phone device (text or beep/ring) without bridging it, before answering the call?
13:57.22seele_look please http://pastebin.ca/224594
13:57.34seele_how can I solve this ???
13:58.45lero_CunningPike: http://pastebin.com/814284
13:59.08lero_take a look, i just don't know what macro it runs, but i think it starts at line 30
14:00.16CunningPikeseele_: Looks like a codec mismatch
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14:01.11blophello :)
14:01.31CunningPikelero_: Use pastebin.ca - pastebin.com is fscked
14:01.50seele_CunningPike, is possible solve this??
14:02.04blopi'm looking for some kind of mini linux distribution which i could load with PXE to run only asterisk on it (from ram) :)
14:02.13CyonAre there any known iax2 issues with asterisk-1.2.13?  IAX2 is dying miserably for me...
14:02.48lero_CunningPike: http://pastebin.ca/224600, i think it starts at line 30
14:04.55CunningPikelero_: It looks like ${GROUPCOUNT} is not > 1, which is why you're getting busy - NoOp(${GROUPCOUNT}) to see what you get
14:07.51seele_how can I make a codec upgrade??
14:08.55pifiumorning everyone
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14:12.58CyonAnyone have any thoughts on:  chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock
14:15.01CunningPikeseele_: What does the codec section of your sip.conf look like?
14:16.45CyonCan't find any current issues reported on matis for iax...
14:17.00CyonI also had:  chan_iax2.c:7665 socket_read: Received mini frame before first full voice frame
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14:21.10seele_CunningPike, http://pastebin.ca/224624
14:22.30CunningPikeseele_: OK - couple more questions - 1) what version of asterisk are you running 2) what end-points are you using 3) what are you doing in your dialplan when you get these errors?
14:23.18lero_CunningPike: NoOp above the gotoif?
14:24.09CunningPikelero_: Yes - we need to inspect the value that you're getting immediately before you do your GotoIf test
14:24.52lero_ok.. it's the macro that starts at line 30 right?
14:25.34seele_CunningPike, ok Asterisk 1.2.9.1, LinkSys are my endpoints and is a call center whit many incomming calls
14:26.11CunningPikeseele_: But more precisely - where in your dialplan do you get the errors - can you pastebin the relevant section?
14:26.29CunningPikeseele_: Of extensions.conf
14:26.39seele_CunningPike, ok ...
14:27.07lero_CunningPike: ok.. it's the macro that starts at line 30 right?
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14:28.26CunningPikelero_: Correct - just above line 36
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14:30.44lero_CunningPike: and just type extensions reload?
14:30.54CunningPikelero_: Yes
14:30.59lero_right =]
14:31.08seele_CunningPike, I don`t know ...  perhaps this?  http://pastebin.ca/224644
14:31.10CunningPikelero_: And then watch the CLI to see what value you get
14:31.24seele_CunningPike, I`m lost .... sorry
14:31.47CunningPikeseele_: You're not running trixbox are you? :)
14:32.29seele_CunningPike, yes
14:32.53seele_CunningPike, Trixbox 1.2
14:33.20CunningPikeseele_: Ya - it's going to be next to impossible to debug it then - check the channel topic
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14:34.06seele_CunningPike, ok thanks
14:34.17CunningPikeseele_: Not a problem - good luck
14:35.07lero_CunningPike: http://pastebin.ca/224653
14:35.12lero_it's jumping the NoOp
14:35.36CunningPikepastebin your CLI output again
14:36.06CyonI know I'm being quite impolite, but if anyone would be able to provide any insight, it would be most appreciated...
14:36.29lero_CunningPike: http://pastebin.ca/224655
14:37.13CunningPikelero_: pastebin your extensions.conf again, too
14:37.15CyonI installed 1.2.13 on a single box as a test, everything ran perfectly for 24 hours; the moment I installed it on a second server, meaning I had IAX2 (with ilbc or ulaw) talking from a 1.2.13 box to a 1.2.13 box, iax started having issues under higher load.
14:37.21lero_CunningPike: ahh ok
14:37.34CyonIf I have one box as 1.2.7.1 and the second box as 1.2.13, IAX2 has no problems at all.
14:37.48CyonIt's only occuring when I have 1.2.13 talking to the same version with IAX2...
14:37.57lero_CunningPike: http://pastebin.ca/224658
14:38.39CunningPikelero_: Your NoOp() needs to be before the GotoIf
14:38.46lero_ahhh
14:38.48lero_;)
14:39.07CunningPike:D
14:40.25lero_CunningPike:     -- Executing NoOp("Local/3083@padrao-21c6,2", "2") in new stack
14:41.45CunningPikelero_: OK - so is it supposed to be 2? Your dialplan is doing what it's told.......
14:43.29lero_but why when i transfer i get that 3083 is busy if it isn't?
14:44.09CunningPikelero_: For some reason, your channel group count is 2 - you'll need to try and figure out why
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14:44.51lero_it need to be 1 ?
14:45.22CunningPikeYes - if you want to not go to busy - that is what your dialplan is doing
14:45.55lero_right
14:46.12lero_i'm just learning this dialplain and asterisk too ;)
14:47.01CunningPikelero_: OK - well, there's no error in your code - it's doing exactly what the dialplan is telling it to
14:47.29lero_right.. gonna do some tests... thanks for your help =]
14:48.38CunningPikelero_: np
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15:03.44[TK]D-FenderCunningPike: And when DOESN'T the dialplan do exactly as its supposed to? :)
15:04.43CyonAnyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes?
15:04.45CunningPike[TK]D-Fender: True - lol
15:04.58*** join/#asterisk blebleble (i=godie@caesar.godie.net)
15:07.18pifiucould there be any reason why i can have IAX working without a password, but when i put a password in, it craps out? the passwords  are fine! lol i checked them like 20 times!
15:09.50*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
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15:10.36[TK]D-Fenderpifiu: Go check the USER NAMES & IP's 20-30 times then :)
15:10.42pifiulol
15:10.43pifiuok
15:10.59pifiui am so flaky with all of this, lol im soo almost done
15:11.12pifiui basically replicated the working config that i had in the first location over to the other and changed teh names accordingly
15:11.20pifiueverything READS liek it should on paper, but it keeps failing
15:11.32pifiuand only when i comment out the password, so the usernames must be right?
15:12.08pifiulocation 1 can talk to the colo, and the colo can talk to 1, but location 2 can only receive calls, and not place any to 2
15:12.14pifiuto colo i mean
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15:19.42pifanyone got problems with asterisk & kernel 2.6.18 ?
15:19.54pifmy polycoms stop registering
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15:20.29joelsolankiHi all.
15:20.45lero_CunningPike: i tried to do something different
15:20.53joelsolankisucessful in installing and configuring rhino 4 fxo port card.
15:21.06joelsolankiincoming is working perfectly but i am not able to make outbound calls.
15:21.46lero_CunningPike: i dial from 3083 to a cellphone. them, from the 3083 i dial *2 to transfer and them 3062 to a other phone here, and all i saw in the CLI is "playing 'beeperr'"
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15:23.33lero_CunningPike: when i press *2 it started playing the playback music, but when i type 3062, it returned to the original call. maybe there's no dialplan to transfer external calls?
15:23.36[TK]D-Fenderpif : thats the most ridiculous assoication I'd heard in a long time.
15:23.53hoobastoobaI am still having problems where quite a few of my calls will go into queue and then be delivered to a queue member but the queue member cannot hear the caller. The caller can hear the queue member. Can anyone give me some assistance? I have 3 other asterisk servers set up nearly identical to this one and they have had absolutely no trouble at all.
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15:25.04CunningPikelero_: Check your features.conf
15:25.18hoobastoobai am using the sangoma t1 card and asterisk 1.2.12.1
15:25.23CunningPikelero_: And check your CLI output _carefully_
15:27.41[TK]D-FenderAnyone here able to provide testimonials to the qualiy/performace of the newer Otasic-powered Digium digital interface boards?
15:28.21hoobastoobaalso i am getting errors: Oct 27 09:27:54 WARNING[19328]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh-native/fpm-world-mix': No such file or directory
15:28.30hoobastoobabut the directory does exist and has full access
15:28.44tzanger[TK]D-Fender: I'm waiting to get mine (RMA'd their Oki one)
15:28.51tzanger[TK]D-Fender: I have *not* had luck with Sangoma's, which REALLY surprised me
15:29.02tzanger[TK]D-Fender: we're starting the process of RMAing that one too
15:29.09hoobastoobaif i do moh reload i get no errors
15:29.09[TK]D-Fendertzanger: Would suprise me as well... and what is "Oki" one?
15:29.26tzanger[TK]D-Fender: TE406 used an Oki ASIC for the echo can/dtmf detection
15:29.52tzanger[TK]D-Fender: fought this sangoma octasic echo can for over two months now
15:29.55[TK]D-Fendertzanger: Oh thats the maker of the old VPM chip?
15:30.11tzangertry new versions of drivers (which helped a LOT but did not eliminate the problem), tweaks, tried different MB... all the same
15:30.12coppiceIts just fabbed by Oki. its not really their chip
15:30.16tzangerI know
15:30.19[TK]D-Fendertzanger: Have you updated the firmware on the card and changed drivers a few times?
15:30.26tzangeroki fabs our VFD ASICs too
15:30.48tzanger[TK]D-Fender: yep, all that was done through Sangoma's tech support
15:30.56[TK]D-Fenderbugger.
15:30.59[TK]D-Fenderbbiab
15:32.59coppiceMark must like Oki. The ADPCM codec in * which is supposed to be DVI is actually the Oki codec. I guess nobody ever uses it, since they haven't noticed
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15:50.05FlatFootanyone got a country code list in db or csv format ?
15:50.22*** join/#asterisk pdtwork (n=ptinsley@209.12.249.243)
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15:50.46drcodehi all
15:51.04drcodex-lite can have speex ? I try the reg fix, but I don't see speex
15:51.11drcodeonly ilbc
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15:57.40pdtworkare there any known issues with the latest versions of 1.2 and random one way audio.  I have a few installs that have never experienced such a problem which were recently updated to 1.2.12.1 for some other fixes and this problem has shown up
15:58.03pdtworkthe setup is pstn -> asterisk -> polycom
15:58.43*** join/#asterisk CunningPike_ (n=CunningP@204.239.8.149)
15:59.19qdkpdtwork: perhaps try another version. 1.2.9 is herhaps the most stable version of the recent ones.
16:00.35pdtworkI had to move to 1.2.12 to fix another problem :(
16:01.13pdtworkguess I could backport the other fix
16:03.24pdtworkseems i get more complaints about it on analog than pri customers but that could be chance
16:05.43*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
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16:06.23*** join/#asterisk Jubei_ (n=Stormtro@147.27.47.12)
16:06.37Jubei_guys what is the most popular/widespread open source web administration interface for * ?
16:08.42Corydon-wThere isn't one
16:08.57Jubei_hmm.. why?:)
16:09.09Corydon-wThose in the know edit their configurations by hand
16:09.14qdkJubei_: because its stupid.
16:09.35Jubei_ok here's a little scenario for you.
16:09.44Corydon-wbecause it's incredibly difficult to abstract a computer language into graphical widgets
16:10.57Jubei_I've got a new user who wants to use my * server to make calls (i work at a university) and I need to make a sip account for him. I can go in to sip.conf and make it manualy but not everybody at the university's admin center knows unix/vi etc. How do they do it?
16:11.10Corydon-wThey learn vi
16:11.18*** join/#asterisk McLazarus (n=mcallist@pool-72-78-131-160.phlapa.east.verizon.net)
16:11.41Jubei_ok, say they do learn vi and someday they decide to go make the changes themself, what if they make a mistake and screw up my whole conf
16:11.54Jubei_or even worse delete sip.conf.. or.. whatever, funk things up
16:11.56*** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv)
16:12.11QwellYou fire them
16:12.16Jubei_loool :)
16:12.24Corydon-wJubei_: what happens when somebody screws up one of your Windows servers?
16:12.53Jubei_eer.. i'm not responsible for windows server so I wouldn't know :)
16:13.07Jubei_i think u get my point
16:13.13Corydon-wJubei_: I suggest that learn about change management
16:13.24qdkGUIs makes novice people think they know what they are doing, which isnt the case 99% of the time.
16:14.15pdtworkJubei_, there are some commercial ones that do work very well, that seem to have mastered the impossible of giving a frontend to textual config files
16:14.15qdkJubei_: SO GUIs replace the need for backup?
16:14.33pdtworkbut config guis aren't sexy so you don't see much effort in the open source world to build them
16:14.39*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)
16:15.05Corydon-wpdtwork: config guis are very sexy.  They're just difficult to do without restricting functionality
16:15.32FlatFootok who's ready for a daft question ?
16:15.35*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)
16:16.41pdtworkI personally wouldn't attempt to do command line configuration of the complexity of some of my installs.  Setting up presence, voicemail, follow-me, company directories for phones, etc for more than a couple of phones is silly
16:17.04FlatFootHOW ? copy the libpri on my current box to a new box ( reason is it's been adapted )
16:18.41pdtworklots of copy paste humans make mistakes nightmares waiting in the wings if you try that on grand scale
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16:22.43pif[TK]D-Fender : sip nat option and polycoms don't mix well
16:22.58pifnew option on 2.6.18
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16:29.49[TK]D-Fenderpif: Kernel directly interferes with packet creation?  DISABLE it.
16:29.51*** join/#asterisk saftsack (n=oliver@p54A7E660.dip.t-dialin.net)
16:30.07saftsackhi does asterisk accept calls on port 5062 on default?
16:30.13Qwellsaftsack: 5060
16:30.23saftsackwhat is with 5062?
16:30.35Qwellexplain
16:34.32saftsacki have a patton gateway
16:34.47saftsackif i add a new virtual gateway there the second is on port 5062
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16:39.35sizzlaHi All !!!
16:40.06sizzlaI need help to install G729 codec on my asterisk box
16:40.40carrardigium sells them
16:40.45carrarbuy how ever many you need
16:40.54carrarand follow their instructions
16:41.02carrarpretty straight forward
16:41.08*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
16:41.09sizzlaI have this error when I restart my server : loader.c: /usr/lib/asterisk/modules/codec_g729a.so: object file has no dynamic section
16:41.47carrardid you email them?
16:42.43CrashHDwaiting for the "email who"?
16:42.54*** join/#asterisk oej (n=oej@apollo.webway.se)
16:43.04sizzlayes but they don't answer me
16:43.26carrarI haven't seen that error
16:43.30carrarwhat version of *
16:43.36*** part/#asterisk Tili (n=tili@202.133.65.90)
16:43.52sizzla1.2.13
16:44.04carrarah bleeding version
16:45.18*** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com)
16:45.28sizzlaI have a sempron processor
16:46.00sizzlawhich version of the codec I have to use?
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16:46.09carrarsizzla, was this a clean install of the OS?
16:46.10JonR800lol.. bleeding edge? he/she is running stable.  That shouldn't be considered bleeding edge.
16:46.26carrarhahah
16:46.33ManxPowersempron fi!
16:47.45sizzlayes install is clean
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16:51.56ManxPowersizzla: what issue are you having?
16:53.41syzygyBSDsizzla: did you purchase a g729 license?
16:54.52syzygyBSDkeep it in the channel sizzla
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16:56.04sizzlaYes I have purchase a g729 license
16:57.36syzygyBSDok, is it the correct version for you version of asterisk?
16:58.37sizzlayes I use 1.2
16:58.54sizzlamy processor is a sempron
16:59.28syzygyBSDthat shouldn't matter, the codec file should be a binary from digium
16:59.50sizzlafor you which version of the codec I have to use?
17:00.20syzygyBSDcan you ask that a differnt way?
17:01.26saftsackhi, is it possible to have different realms for one asterisk server where users can authenticate themselves?
17:01.33ManxPowersaftsack: no.
17:01.56syzygyBSDManxPower: u sure?
17:02.16ManxPowersyzygyBSD: There has been extensive discussion about this on the mailing lists.
17:02.36syzygyBSDcan't you run two instances of asterisk...
17:03.08ManxPowersyzygyBSD: maybe.  assuming you don't need IAX2 trunking, MeetMe, or Zaptel
17:03.31syzygyBSDthey could talk to eachother...
17:03.54syzygyBSDjust a thought... but anyway, not trying to distract from the real question
17:05.09saftsackManxPower, if a sip telephone connects with two different accounts to the * server it is possible that the accounts communicate on different ports?
17:06.51ManxPowersaftsack: I don't know, but I strongly doubt it.
17:07.36ManxPowerAsterisk is not designed to run multiple instances at the same time.
17:07.36ManxPowersaftsack: Why would you want two phones with the same account name?
17:07.36saftsackand one instance can just act on one port?
17:08.05saftsackManxPower, i have a gateway with fxs and fxo port and the gateway isnt capable to connect to the same server on the same port at the same time
17:08.37ManxPowersaftsack: I had no trouble with my SPA-3000 getting both lines to talk to Asterisk
17:08.39saftsacki want to have 2 accounts because of the ability to differ from two contexts, outgoing and incoming
17:08.50saftsackManxPower, yes but i havent got a spa-3000
17:09.01saftsacki have a patton 4552
17:09.08saftsackthis is what the manufactor writes
17:09.15ManxPowereach port had a different SIP user id (the MAC Address -a and -b), the source port was 5060 for one and 5061 for the other.
17:09.39ManxPowerthey both talked to a DESTINATION port of 5060 on the Asterisk server of course.
17:09.44saftsackThe only way to use two services in a single gateway is that you MUST use realms, and they MUST be unique to each service.
17:09.52saftsackthis is the first option and the second ist
17:09.53saftsack-t
17:10.09ManxPowersaftsack: does each port support different SIP user IDs?
17:10.11saftsackcreate to sip gateways but they doesnt act on the same port
17:10.32*** join/#asterisk lsald (i=lsald@gw.percipia.com)
17:11.04saftsackyes one service can support more than one gateway but it isnt possible to assign them to a line or to a telephone directly. at least you have to create two services or two gateways
17:11.24ManxPowersaftsack: correct.  Seems pretty easy to me.
17:11.42ManxPowerline 1 uses username1 and line 2 users username2
17:12.03ManxPowersaftsack: I think you are confusing SOURCE port with DESTINATION PORT
17:13.24ManxPowerSPA-3000 FXS. username "fred", IP 172.16.5.2, port 5060 <-> IP 172.16.5.1, port 5060, Asterisk
17:13.29saftsackdont know but it doesnt work actually but i wrote a mail to the manufactor
17:13.43saftsackThe only way to use two services in a single gateway is that you MUST use realms, and they MUST be unique to each service
17:13.43ManxPowerSPA-3000 FXO. username "bob", IP 172.16.5.2, port 5061 <-> IP 172.16.5.1, port 5060, Asterisk
17:13.58saftsackthis isnt possible with one * server, or?
17:14.11ManxPowersaftsack: then you should return the device and get a different one.
17:14.35*** join/#asterisk rootfield (n=rootfiel@200.103.96.98)
17:14.37rootfieldhi all
17:14.37saftsacki bought it 2 months ago ^^
17:14.52rootfieldhow can I solve hangup problem ?
17:18.08*** join/#asterisk mv00 (n=mv00@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk)
17:18.12mv00hi
17:18.56mv00i need some help to configure a caller ID system on my Asterisk box.. i am looking to be able to do: <incoming DID> :Asterisk: <enduser> (end user see's set callerID)
17:19.10mv00i don't have a problem with paying.. just please someone help :) i have looked a lot about this..
17:20.49Strom_Cmv00: I can probably help you out
17:20.50carrarYou incoming calls do not have caller ID already?
17:21.46mv00no, they don't
17:21.50mv00and i need a custom caller ID..
17:21.59carrarwhat is your connection to the outside?
17:22.11carrarpri?
17:22.16mv00i use a sip provider..
17:22.30carrarThats odd that they are not sending you ANI
17:22.41*** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net)
17:22.49mv00i have no idea if they send ANI carrar - but
17:22.53mv00i am looking to set custom caller id..
17:22.59carrarah ok
17:23.48carrarSet(CALLERID(name)=Outside Line)
17:23.58mv00yeah, but i have no idea what to do
17:24.04mv00carrar, i think Strom_C is helping me, but thank you
17:24.07mv00and i'll let you know how it goes
17:24.09carrarok
17:26.03rootfieldanybody solved hangup problem ?
17:26.35*** join/#asterisk DjPepse (n=pepse@71-223-116-141.phnx.qwest.net)
17:26.36*** join/#asterisk ACiDV (i=ACiDV@bas3-sherbrooke40-1177840132.dsl.bell.ca)
17:26.41DjPepsemorning, gents.
17:27.30carrarhihi
17:27.40carrarYou need a i in your name ;)
17:28.01carrarDjPepsi
17:28.18DjPepse:O that would be copyright infringement!
17:28.20DjPepse:)
17:28.25ACiDVAnyone have a working setup with Pickup() application ?
17:28.36DjPepsebesides, this has an alternate connotation to it.
17:28.53syzygyBSDwhat do I need to get SMS text messages with asterisk?
17:29.27ACiDVI'm totally unable to made it working :(  exten => 222,1,Pickup(1000@default) ...  always say not originating channel
17:30.05DjPepseI'm having trouble finding a feature that I swear I've used.. A way to grab up a call from another extension. Or maybe I transfered it? It was nothing I had to specially configure, anyway.
17:30.28*** join/#asterisk Odie_Flocon (n=chatzill@S01060011953d7c4c.cg.shawcable.net)
17:30.35Odie_Floconhey all.
17:31.04Odie_FloconHelp
17:31.10Odie_Flocon:D
17:31.31ACiDVDjPepse... use Redirect Manager API, BRIDGEPEER variable and transfer call to a Meetme
17:34.12DjPepseacidv: yeah, i've seen a bunch of documentation with stuff like that
17:34.30DjPepsebut this is something i've done with my existing setup. i don't even know how i found it last time :/
17:36.59*** join/#asterisk ast_freak (n=jesse@h69-130-172-115.69-130.unk.tds.net)
17:37.34ast_freakAnyone used the Set: field of a .call file?  Having problems getting the variable into the dialplan.
17:38.02CyonAnyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes?
17:38.52*** join/#asterisk tdawgpharaoh (n=chatzill@62.135.94.232)
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17:42.41*** join/#asterisk mfroes (n=mfroes@200-162-218-81.corp.ajato.com.br)
17:43.48mfroescan someone help ??? i have a TE110 board but it gets too much echo ... tried to configure like they say on the net, but it wont work. any ideas?
17:46.54mfroesmostly on long distance calls
17:48.04qdkmfroes: have you tried different EC's in the zaptel driver?
17:48.43*** join/#asterisk lero_ (n=rootz@200.192.160.100)
17:48.52lero_CunningPike: Oct 27 14:47:33 WARNING[16640]: res_features.c:844 builtin_atxfer: Did not read data.
17:49.05mfroesqdk, ueap
17:49.06lero_what can be this
17:49.31qdkmfroes: say what?
17:50.24*** join/#asterisk drega (n=drega@80-47-247-119.lond-th.dynamic.dial.as9105.com)
17:51.18ast_freaknm, got it.
17:53.09CunningPikelero_: I have no idea :)
17:53.29lero_:~
17:56.27DjPepseHas anyone tried out the australian Uplink Skype to SIP app?
17:56.34DjPepseIt's pretty cool
17:56.50DjPepsewish I could find something like that for OsX or linux
17:58.52*** join/#asterisk rene1 (n=rene1@201.122.36.212)
17:59.20*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
17:59.40rene1can one get skype-comparable audio quality with speex in asterisk?
18:00.49*** join/#asterisk Gunde (n=spamyous@82.153.170.213)
18:01.25DjPepsegoogle tells me skype uses ilbc
18:01.42DjPepseand isac
18:01.46DjPepsei guess isac is better
18:02.03mfroesqdk, yes.. i have tried that already
18:02.50DjPepserenel: i was -just- saying how cool the uplink skype to sip app is before you joined
18:04.34rene1well i have read that speex is wideband and so does skype. i just wonder if any body has actually done speex calls or other wideband calls with asterisk
18:04.44rene1DjPepse: cool
18:05.05rene1speex can sample 32 / 16 / 8 khz
18:05.21rene1but the wiki says something about asterisk being hardwired to 8 bit
18:06.03qdkmfroes: how is the latency YOU control?
18:06.36qdkmfroes: EC is not the magic that fixes "br0ken" lines.
18:06.57DjPepsei wonder what codecs my e61 supports
18:08.05rene1s/bit/khz/
18:09.45mfroeswhat do you mean latency i control ?sorry my englhish is not that good
18:09.45sahafeezsomeone tell me whats wrong here. i am on the phone w/sip provider and they have no clue. been going around with them for a week now http://rafb.net/paste/results/OIG7ld47.html
18:10.12*** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu)
18:10.18b11dhello all
18:10.46b11dok..  is there any way I can disable my Polycom 501 from showing people in the directory on the "idle" screen as Speed Dial options?
18:10.49b11dI hate that..
18:11.34CunningPikeb11d: Blank out the speed dial number in the directory entry
18:12.28b11dgot it
18:12.30b11dthanks
18:12.35b11dthat was freaking me out
18:12.36b11d:)
18:12.38CunningPikeb11d: ytw
18:12.56b11di dont know what ytw means
18:13.05CunningPikeYou're totally welcome
18:13.10b11doh.. sweet :)
18:13.10Juggie!seen mog
18:13.11b11dthanks
18:13.30sahafeezanyone http://rafb.net/paste/results/OIG7ld47.html
18:13.52CunningPike~seen mog
18:14.10jbotmog <i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net> was last seen on IRC in channel #asterisk, 3d 1h 50m 13s ago, saying: 'thats what most of us do'.
18:15.28Juggieanyone in the CZ?
18:15.33b11di wish..
18:15.38b11di could have a name like fester or something
18:15.43b11dwhich would fucking rock
18:15.54*** join/#asterisk mfroes (n=mfroes@200-162-218-81.corp.ajato.com.br)
18:15.59CunningPikesahafeez: Are you using reinvites?
18:16.02*** join/#asterisk gaspiz (n=gaspiz@86.35.240.238)
18:16.05mfroessorry ... my pc had crashed
18:16.17b11dthats the worst
18:16.33mfroesqdk, if you cloud explain better what you meant i'd be glad to answer
18:17.09sahafeezCunningPike: not as far as I know. one sec..here is what it is now - provider made changes http://rafb.net/paste/results/3HQYHb48.html
18:17.49gaspizhi, the most wierd thing happend: when calling a sip address it goes the normal call flow: invite,100 trying,183 session progress,200,ack back nad then my asterisk sends another invite
18:17.52CunningPikesahafeez: That looks better
18:18.03sahafeezyes, still get a busy tho.
18:18.11gaspizI use asterisk 1.2.12: any ideas?
18:18.34CyonAnyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes?
18:19.12[TK]D-Fenderb11d: You would want to use line-keys for speed dials for either convenient #'s, or people whose status you'd want to track
18:19.17[TK]D-Fenderb11d: (Presence)
18:19.21CunningPikesahafeez: Different reason - is that last pastebin the entire sip debug?
18:19.29sahafeezyes.
18:19.46CunningPikeHmm - 'SIP/2.0 407 Proxy Authentication Required' - looks like you're not authenticating properly
18:20.07mfroesgaspiz, have you tried with the default conf of asterisk ?
18:20.57[hC]anyone know how long it taxes for faxdetect to detect a incoming fax?
18:21.18sahafeezCunningPike: http://rafb.net/paste/results/V8UmPQ55.html - just did it again and mark the start/end on myside so i am sure.
18:21.53[hC]I have a box that id like to run faxdetect on, before connecting via iax to my clients asterisk box, to be able to receive faxes for them on the same number they take voice calls on, but i suppose i need to answer the call first, and let it ring a certain number of times.
18:22.20qdkmfroes: you have some endpoints configured on your*, right?
18:23.06sahafeezCunningPike: ok, my provide cannot seem to tell me what I am doing wrong re: auth. are we talking the registration part in sip.conf or the other parts were the provider is defined?
18:23.14HarryRok, just learning to write asterisk modules (e.g. dialplan functions & such), anybody willing to spare a second pair of eyes on some C?
18:23.17qdkmfroes: what is the latency from them to your server and what is the latency from your server to your provider, if you know that?
18:23.24CunningPikesahafeez: Make sure you're registered properly with your ITSP - 'SIP/2.0 407 Proxy Authentication Required'
18:23.26gaspizmfroes: it's pretty mutch the basic setting
18:24.20sahafeezCunningPike: proxy2.bandtel.com:5060         2038700001         280 Registered
18:24.33sahafeezdamn.
18:25.44rene1i hate c cuz i cant read it
18:28.51*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)
18:29.19[TK]D-Fenderrene1: if u cn rd ts tn u cn pgm n c
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18:32.21florz[TK]D-Fender: thr s nn wh cn nt?
18:32.57hmmhesays~seen oej
18:33.13jbotoej is currently on #asterisk (1h 50m 19s), last said: 'hmmhesays: Need to go off line, it's late here. Keep me posted.'.
18:33.18hmmhesaysbah is jbot down?
18:33.20oejhere, now
18:33.32oejsoon there, then
18:33.36oej:-)
18:34.03rene1oej: do you have  90 secs?
18:34.16oej63,78 :-)
18:34.28oejfriday night, drinking red wine, eating cheese.
18:34.33oejNeed to focus
18:34.38hmmhesayshaha oej, those would be my thoughts exactly about the t.38
18:37.38hmmhesaysT38 pt UDPTL : Yes
18:37.46hmmhesaysfor all of my peers involved
18:39.34*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
18:39.34*** mode/#asterisk [+o russellb] by ChanServ
18:46.01CyonAnyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes?
18:48.03*** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net)
18:48.47GaVakWould this be the proper channel to ask a config file question?
18:49.08CyonGaVak:  Yeah.
18:50.08GaVakOk, i've set up a 1.2.13 server and have my SIP channels working, and I'm currently working on setting up call Queues.
18:50.14CyonSure
18:50.32GaVakIn my extensions.conf, I put in: exten => s/4258820921,n,Queue(support)
18:50.35GaVakand
18:50.41GaVakexten => s,n,Queue(support)
18:51.04GaVakWell, it seems to ignore these steps and hits the timeout value
18:51.10GaVakwhen i do a show queues
18:51.27GaVaki have: support      has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s
18:51.27GaVak<PROTECTED>
18:51.27GaVak<PROTECTED>
18:51.27GaVak<PROTECTED>
18:51.27GaVak<PROTECTED>
18:51.28GaVak<PROTECTED>
18:51.34*** join/#asterisk brookshire (n=greg@dsl253-055-082.dfw1.dsl.speakeasy.net)
18:51.38GaVaki'm wondering what i'm missing
18:52.15Cyonpastebin the config.
18:56.46*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
18:59.19saftsacki did bindport=5062 but it isnt possible for a client to connect on this port. do i have to change an other option?
18:59.43[hC]arg, newmantelecom's website is down, i need a copy of nvfaxdetect
19:04.59*** join/#asterisk alerios (n=alerios@190.24.99.75)
19:05.36*** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-f8f3fcecb708ecda)
19:07.12hmmhesaysbah
19:07.13hmmhesays<PROTECTED>
19:07.16hmmhesayswhat do I do about that
19:07.20hmmhesaysasterisk 1.4
19:10.20hmmhesaysbetter yet how do I fix it
19:11.21*** join/#asterisk zeppelin_ (n=fpcmdv@201.21.219.33)
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19:28.17hmmhesaysargh this is driving me nuts
19:31.31*** join/#asterisk Arnar (n=arnarb@landi.oddi.is)
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19:43.27WGFreewilli have a question about codecs on an IAX trunk
19:43.44WGFreewillshow channel XXXX show nativeformat readformat and writeformat
19:43.49WGFreewillwhat do they mean?
19:44.05WGFreewillwhats the actual utilized codec?
19:44.16WGFreewill(I have a case with 1024-64-64)
19:48.00*** join/#asterisk cian (n=cian@cian.ws)
19:51.07pifiuwhat are mini-frames?
19:51.18pifiuin the cli sometimes it says received mini frames before firs tfull voice frame
19:53.40*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)
19:56.13GaVakI'm running a test that goes out ZAP/6 and comes in ZAP/1 and the sound is really faint.
19:56.24GaVakShould I increase the rx/tx gains?
19:59.35*** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net)
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20:19.08tdawgpharaohany voipgateway.org users here? (I have number from guest-voip.ch) sip debug shows incoming call on my swiss number, sip show registry shows it is registered, but no calls comes in... any ideas?  Sorry I am kinda new to *
20:19.29*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
20:21.20rene1td...h: do you have an appropiate incoming context in your dialplan..
20:21.59tdawgpharaohwell, I wrote in sip.conf to go to swissin in dialplan, which should ring my extension directly
20:22.08rene1ah ok
20:22.24rene1maybe it is a codec issue
20:22.29*** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net)
20:22.31tdawgpharaohcould be
20:22.36rene1if codecs can not be negotiated then the call wont go tru
20:22.42tdawgpharaohNever though of that
20:22.45tdawgpharaohI'll try
20:22.53tdawgpharaohI'll let you know, thank you
20:22.57rene1sure
20:23.34rene1if you see a tall hot blond chick in zurich named chrstine say hi to her for me
20:23.45rene1emm nevermind
20:24.21tdawgpharaohsure will :)
20:24.34joerene1: there are many, including my ex gf that lives there :)
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20:25.06*** join/#asterisk steveaj (n=steve@62.55.147.53)
20:25.08tdawgpharaohyou might have been right, I did disallow=all, and allow=gsm and I got no dial error, but instead a "dead line" sounding tone
20:25.29*** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com)
20:25.51joesomeone remdind me a way at the cli to see which ports in a tdm400 fxo/fxs card are fxo and which are fxs?
20:26.31hmmhesaysshow channels?
20:26.32rene1:)
20:27.28joeumm no
20:28.19rene1zap show channels?
20:30.02rene1that wont work either
20:30.43rene1maybe ! cat /proc/zaptel/*
20:30.48rene1in the CLI?
20:33.43tdawgpharaohseems that wasn't it, I tried several codecs... the dead line sound was unrelated.
20:34.37ghenryinteresting http://www.asteriskvoipnews.com/asterisk_hardware/polycom_and_digium_partner_to_offer_integrated_sipbased_telephony_solution.html
20:34.50ghenrygood for pushing * solutions!
20:36.09ghenrydoes this prove poltcom phones are the best?
20:36.24joerene1: I think that's it thanks
20:36.42joerene1: I know there is a function or tool to do that from the * cli but I just can't remember
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20:42.39*** join/#asterisk smash- (i=smash@216.sub-70-193-104.myvzw.com)
20:43.42smash-Hey
20:43.51smash-anyone familure with old school telephone systems
20:44.13smash-i have a unique problem that i could use some minor help with
20:44.33smash-i have http://www.amdevcomm.com/voice-mail-products/voice-mail-components/rhetorex/rdsp_400.html
20:44.43smash-in a computer and the hard drive broke i have no idea what software runs it
20:45.11smash-i hear there is a more recent windows version for it and a older DOS version for it
20:46.48tdawgpharaohgot it now, thanks rene1
20:46.57*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
20:50.02smash-found it
20:50.06smash-amanda@work.group
20:50.13smash-fuck but where can i down that ancient shit
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21:02.02*** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net)
21:03.40GaVakI have a system set up using a Digium FXO card and using polycom 501 SIP phones. I just brought the system online today and the calls seem very quiet.
21:03.56GaVakDoes the RX/TX Gain in Zapata.conf effect FXO card volume?
21:03.56rene1the calls to the PSTN?
21:04.01rene1yes
21:04.04rene1they do
21:04.20rene1crankup the rx gain
21:04.25GaVakOk, thankee.
21:04.27rene1there is an app to do it interactively
21:04.37rene1like with a live call
21:04.50GaVakthe one that needs a test 1024hz number?
21:04.52GaVakI didn't have one.
21:05.17rene1cant remember Gavak sorry
21:05.28GaVaknpnp, thanks though, I'll play around with manually setting them first
21:05.45rene1ztmonitor
21:06.15rene1may have been compiled for you or you may need to manually compile it in zaptel source
21:08.10GaVakI see it compiled in the zapa source directory
21:08.24GaVakI'll monkey around with it and see what i get.
21:08.26GaVakthanks.
21:09.25rene1np
21:11.06tzafrir_homeGaVak, BTW: 'reload  chan_zap.so' will update those values, you don't need to restart asterisk
21:11.57tzafrir_homeanyway, where exactly is Digium's new GUI? Which SVN branch? Which repo?
21:12.55*** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br)
21:13.38GaVaktzafrir_home: Will that kill current zap buildouts?
21:13.59rene1tzafrir_home: in asterisk-gui
21:14.21rene1in trunk
21:16.55*** join/#asterisk CunningPike_ (n=CunningP@204.239.8.149)
21:18.41smash-hey
21:18.52GaVakAwseome, RXGAIN 20.0 did the trick.
21:18.52smash-what was the convention called that devoloped asterisk?
21:18.53GaVakThanks again.
21:19.06hadsWow, that's quite high.
21:19.18GaVak10 was still kinda quiet.
21:19.20CunningPike_smash-: ?
21:19.34GaVakcould there be negative impacts for a high gain?
21:20.24CunningPike_GaVak: One word - echo
21:20.31*** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk)
21:20.53CunningPike_GaVak: But the correct value is the one that works
21:20.59GaVakHmm, I'll tell support to keep an 'ear' out for it then.. if they get it, I'll trim it some.
21:21.27GaVakI had problems with the lines with the analog phones I had on them before, I think it puts out too much power from the ADTRAN multiplexer they are on.
21:23.59*** join/#asterisk simoncion (n=simoncio@DHCP-144-48.resnet.ua.edu)
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21:36.09ARRIBAHi all, how do i change the filename convention of recordings in /var/spool/asterisk/monitor e.g. <phonenumber>_<timestamp>.wav ; I am using Asterisk 1.2.12.1
21:37.18*** join/#asterisk ScurvyDawg (n=scurvyda@S0106000d883f28a0.gv.shawcable.net)
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21:53.50*** join/#asterisk megasquid (n=asdf@ip3d.campustech.net)
21:54.46megasquidanyone know if it would be possible to connect my comp to a voip line and use fax software to send faxes?
21:56.05tzafrir_homemegasquid, a fax software generally expects a real fax. A VoIP "line" is not exactly a modem. And faxes don't work very well over voip
21:56.33pifiuis ulaw better than gsm?
21:56.38tzafrir_homeBut then again, iaxmodem and hylafax is exactly that...
21:56.44tzafrir_homeulaw is better, yes
21:56.46hadspifiu: Define "better"
21:56.53*** join/#asterisk angryuser (i=uk@d07m-89-86-90-231.d4.club-internet.fr)
21:57.13hadsIf you are referring to sound quality then yes.
21:57.14tzafrir_homeless chances of losing data
21:57.28hadsIf you are refering to bandwidth use then no.
21:57.45tzafrir_homepifiu, ignore me. I thought that this was regarding the fax issue
21:58.01hadsIf you are somewhere other than the US or Japan then alaw would be much better :)
21:58.39megasquidtzafrir_home: so using an adapter connected to my fax modem and voip line wouldn't really work?
21:59.02hadsNot reliably if at all.
22:00.37pifiuwhats not reiable hads/
22:00.44pifiubetter sound quality
22:00.46tzafrir_homethrough voip? may or may not work
22:00.59hadspifiu: That was for megasquid
22:01.15megasquidhmm.. im basically just looking to be able to fax without long distance charges :)
22:01.19pifiuim wondering how many calls can i have on a 384Kbits upload DSL connection on ulaw vs gsm
22:01.23hadspifiu: If you are after sound quality then ulaw/alaw is good, yes.
22:01.40pifiui need 3 AT MOST i think
22:02.00hadsulaw you won't get more than 3 or so.
22:02.09hadsLess if there is other traffic using the link.
22:03.39pifiuand its pppoe so there's overhead
22:03.40pifiuhmm
22:06.07shellsharkpifiu: use g729 :)
22:06.19shellsharkpifiu: $30 one-time investment for three channel licenses
22:06.29tessierWhen I register with a SIP provider how do I tell it what context incoming calls from that provider should go into? I used to know but it's been ages since I have set this up.
22:06.41shellsharkpifiu: you'll use a tiny fraction of the bandwidth that you would use for any other codec
22:07.04pifiuwhere do i set what codec i want to use?
22:07.08pifiuin iax.conf?
22:07.25hadsIf you are using IAX.
22:09.35pifiuyeah
22:09.42pifiuunder each defined user and client
22:09.51pifiuwhats the context to use? codec=?
22:10.01hadsRead the sample configs.
22:14.17pifiulol i deleted them
22:14.22pifiuok dont worry ill figure it out i guess
22:14.36hadsThey are in the configs directory in your source.
22:17.51*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
22:18.04qseekhi all
22:18.30qseekhow do I configure a TDM400P to handle telco rollover correctly
22:18.57*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
22:19.03*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
22:19.54qseekhello
22:22.57qseekis anyone online
22:22.59*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
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22:38.53GaVakIs there any way to get the asterisk -r to code in color like the main console does if started with asterisk -c?
22:40.18*** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com)
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22:56.59un_jneed some help with 7960s is this a right place?
22:57.57Strom_Cit's worth a go :)
23:00.44un_jI cannot figure out how to different versions of firmawres (on the phone I got P00303020204) what should I put in XMLDefault.cnf.xml and OS79XX.TXT??
23:01.01linageewow, that's pretty nifty. voip over GMRS/FRS using asterisk. :-D
23:01.36maclihi, I have two xlite softphone setup, one on powerbook, one on ibook,  they rings to each other, but no audio , any clue?
23:03.37un_jmacli: looks like firewall/nat setttings tome
23:04.46maclithere is not nat/firewall, the server and the two computer are in the same network, none of them have firewall setup
23:04.57maclinot = no
23:05.12*** join/#asterisk R3PT||3 (i=reptile@82.79.232.132)
23:06.17macliI thought there might be no audio input on my powerbook/ibook, but they past the audio test when I install xlite, I can both use skype on the two computer
23:15.34*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
23:26.57*** join/#asterisk file2 (n=IrcNet@out.clearnet.com)
23:26.57*** mode/#asterisk [+o file2] by ChanServ
23:27.19file2hey
23:27.36file2i am a nub
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23:28.26file2nooooooo
23:29.25file2oh no im a nubb
23:29.44file2thgpol
23:29.46file2a.kl
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23:35.21Invertedis there a way to implement a call-limit for iax in a similar manner as sip?
23:37.20*** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es)
23:37.23kink0hello
23:38.01kink0I am trying to do Dial() to a PSTN phone, and then capture the pressed keys on that phone, any idea ?
23:38.39kink0I did this when my Asterisk receives a call , ussing Read(), but.. how to do when my Asterisk originates the call ?
23:42.12*** join/#asterisk coder_cotton (n=coder_co@12.206.134.251)
23:42.22coder_cottondoes anyone have any experience running a local wifi PBX?
23:42.29coder_cottonwould have the linksys POE phones attached to wireless-N Access points
23:44.35kink0where is the people ? at lunch ? sleeping ?
23:44.51kink0too much silence on the room :)
23:45.12GaVakHmm, it is 8pm EST.
23:45.15GaVakOn a Friday
23:47.01hmmhesaysyeah
23:52.31un_jwhere can I get old cisco 7960 firmwares?
23:52.37*** join/#asterisk R3PTII3 (n=reptile@ACA23BED.ipt.aol.com)
23:52.56R3PTII3is there anyone available to help me with something please
23:54.16Nivex~data
23:54.18jbotDon't Ask To Ask. Just ASK
23:56.05R3PTII3i have installed asterisk on my server and configure it ... i am using it to make calls trough nufone.net but when i am making the call a sample voice answear my call and is telling something like ... you have succesfully installed the asterisk pdx .. and the call didn`t go trough ... so can anuone help me to configure it?
23:57.06hmmhesaysyou have your dialplan set up wrong
23:57.13hmmhesaysand that is the default
23:58.17R3PTII3hmmhesays i can show you the link that shows me how to set it up .. maybe you can figure what is wrong and help me ... can you do that for me please?
23:58.57R3PTII3i will give you access to my server to be easy for you
23:59.53tzafrir_homeAnybody started playing with asterisk-gui ?

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