IRC log for #asterisk on 20090725

00:04.36*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
00:10.28Xetrov`is there a good free softphone app for windows mobile?
00:12.39*** join/#asterisk viq (n=viq@unaffiliated/viq)
00:14.36OrbixxSeems I have to compile a kernel as my provider uses a custom kernel on their servers.
00:16.29Snoogan[TK]D-Fender: what settings do i need for freepbx to register the ekiga sip trunk?
00:20.24*** join/#asterisk ariel_ (n=ariel_@c-71-196-99-26.hsd1.fl.comcast.net)
00:22.13*** join/#asterisk propellerhead (n=yogurt2u@host66.190-31-153.telecom.net.ar)
00:31.16*** join/#asterisk elite (n=elite@c-71-197-242-157.hsd1.wa.comcast.net)
00:31.46Guest441Is it possible to call a number and get the charger number and hangup so it never rings?
00:31.47*** join/#asterisk Alfio (n=Amunoz@adsl-54-26.tricom.net)
00:35.19ariel_the charger number? what is that?
00:35.45Guest441the charge number
00:38.11ariel_I still don't understand what you mean by the charge number.
00:38.46Guest441The carge number in the SS7 ISUP IAM
00:38.49Guest441charge*
00:39.35ariel_not unless you have access to the ss7 switch
00:41.38Guest441Is it possible to forward an incoming call to asterisk and then set the charge #?
00:42.53ariel_what you are trying to do is spoofing and actually it's doable but I can't and will not go into this
00:44.04Guest441Really wahts the point of idling in a channel and not going full out with your information, stop being a pussy
00:44.24Guest441Everyones security is already fked, WEP and WPA is cracked like butter anyways
00:46.10Guest441Can someone who is not a pussy please inform me
00:46.39Guest441http://www.binrev.com/forums/index.php/topic/14010-ss7-isup-number-delivery-fields/
00:47.34ariel_Like being called that, but I can tell you that I will not due to many issue's but mainly my own in being on the level and having respect for others as well.
00:48.01Guest441are you a christian dude?
00:48.29OrbixxNobody is obliged to help you.
00:48.41Guest441Is it in your hands as a christian to save the world from spoofing?
00:48.47ariel_don't have to be to be on the level
00:59.50*** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
01:09.05leifmadsenZup yo
01:09.36leifmadsenGuest441: you're odd
01:10.26Guest441is that an insult these days?
01:10.33leifmadsenno, I'm just saying you're odd
01:10.47leifmadsenyou've got a unique approach to asking questions is all
01:11.39Guest441leifmadsen: well thank you, the odd people make the world go around
01:12.24AlfioGuest441 like in my country say you are a macagrano
01:12.54*** join/#asterisk Iamnacho (n=Iamnacho@wsip-98-191-129-196.ks.ks.cox.net)
01:13.01Guest441does your country have clean water?
01:13.50Alfioyes and lambetallos like you
01:14.31Guest441i will glady be your king
01:14.49Guest441now go get your sister so i can fuck her
01:14.59Alfiothx
01:15.07[TK]D-FenderGuest441: Cool it.
01:15.38Guest441See you guys should follow fenders example
01:15.50Guest441hes dealt with my kind before
01:16.04*** mode/#asterisk [+b *!*@*.hsd1.wa.comcast.net] by leifmadsen
01:16.04*** kick/#asterisk [Guest441!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (leifmadsen)
01:16.12Alfio:)
01:16.25leifmadsenwho's the king?
01:16.32[TK]D-Fenderleifmadsen: Kill-joy :)
01:16.35Alfiohehehehhehehehhee
01:16.39leifmadsenhe'll be back in a minute telling me I'm ghey or something
01:16.55leifmadsen[TK]D-Fender: ya, I almost left he be just for fun, but i'm leaving soon
01:17.24[TK]D-Fenderleifmadsen: I'd have dealt with him...
01:17.26*** mode/#asterisk [-b *!*@*.hsd1.wa.comcast.net] by leifmadsen
01:17.34leifmadsentime out is over ;)
01:18.12leifmadsensweet, just found out I've had a record year for consulting
01:18.15leifmadsenrecession, what?!
01:18.26Alfiohehehheehhe
01:18.28jayteeyour patience is admirable, i'd have reduced him to a bowl of snot several minutes before you kicked him
01:18.41leifmadsenjaytee: he was kind of entertaining me :)
01:18.47Alfiothe recession its so good for asterisk
01:18.54leifmadsenit does seem to be
01:19.12leifmadsen"what? we can't spend $1m on Cisco or Avaya? I guess we'll look at this Asterisk thing"
01:19.15Alfiomany companys are lookint to cut their budgets
01:19.24jayteeyeah, it was fun to listen to. I was tempted to engage him but I usually have an effect on people like that that escalates into all out war :-)
01:20.03[TK]D-Fenderjaytee: I'd have just taken him apart one little piece at a time and then casually discarded the carcass...
01:20.12leifmadsenwith acid?
01:20.37[TK]D-Fenderleifmadsen: Mr. Pointy :p
01:21.03leifmadsenheh
01:21.17[TK]D-Fendershould be 2nd kyu now...
01:21.55leifmadsenweak
01:22.45[TK]D-Fenderleifmadsen: Sensei has only shown up once this month and I've actually been teaching the newbs with almost all the other senior students on vacation, etc.
01:22.59Alfioone question why itsnt an spanish channel like this?
01:23.36jayteeso was I wrong in getting the impression that he was pumping people for info on SS7 headers in order to commit toll fraud?
01:25.43*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
01:31.14*** join/#asterisk Faiz (n=otakucon@c-98-221-51-177.hsd1.nj.comcast.net)
01:33.10drmessanoShe's like a rainbow
01:53.16*** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au)
02:04.10*** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
02:34.54Faizis there a particular reason why the asterisk (1.6) CLI doesn't reload modules properly?
02:35.18Faizit hangs after i issue 'module unload chan_dahdi.co'
02:35.34Faizsince it does not allow me to run dahdi commands
02:37.21Snoogancan someone offer assistance and help me to diagnose why I can't register sip trunks on my Asterisk box?
02:42.30[TK]D-FenderSnoogan: What was the result of the last test I told you to do?
02:44.25*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
02:44.28Faizhey Fender, would it be alright if I private message you about a few issues?
02:44.34Snoogani couldn't work out the settings for freepbx for the free sip account.  I have signed up for a gotalk account, and put in settings according to what ppl have listed on whirlpool.
02:44.49Snooganthe go talk trunk won't register either
02:50.50[TK]D-FenderFaiAsk in channel;
02:51.23[TK]D-FenderSnoogan: Doesn't maater that it fails, it matter HOW it fails
02:53.19Snooganshould i do a sip set debug?
02:54.22[TK]D-FenderSnoogan: Do you think looking at the problem might help you see what it is?
02:54.41[TK]D-FenderSnoogan: Its a really novel idea I'll admit...
02:54.54Snooganyeah, of course, but i don't understand the output ... that's why i'm asking for assistance
02:55.49Snooganyou've been helpful so far, and i appreciate that.  But i need guidance.
02:56.19[TK]D-FenderSnoogan: Yes, and I looked at it for you last time...
02:56.31[TK]D-FenderSnoogan: Do you really have to ask if I would again?
02:56.38*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
02:56.38*** mode/#asterisk [+o leifmadsen] by ChanServ
02:58.06*** join/#asterisk ingenius (n=alektro@host95.190-229-175.telecom.net.ar)
02:58.34Snoogani wasn't sure, that's why i asked.  http://pastebin.ca/1506545 shows the output
03:00.18[TK]D-FenderSnoogan: Ok, packets simply aren't making it back.  Either your router is bad and screwing up, or your internet connection is being filtered.
03:01.21Snooganthe mynetfone trunk registered yesterday
03:01.46Snooganand i've registered before on this connection with other boxes.  So i rule out the ISP connection
03:01.58Snooganso i'll try a different modem
03:02.01[TK]D-FenderSnoogan: well you confirmed your IP, I suppose I'm not going to see your router settings to trust that what needs to be forwarded is... we're jsut about out of options here
03:02.32Snoogani can send you a screen shot of my port forwarding
03:05.35*** join/#asterisk tflgen2 (n=clay@fl-67-233-27-60.dhcp.embarqhsd.net)
03:05.51[TK]D-FenderSnoogan: imagebin.ca
03:07.24Snooganhttp://imagebin.ca/view/1oW4Aq4.html
03:09.41[TK]D-FenderSnoogan: Ranges look right.  Does your router know anything about SIP aside from those settings you made?  AKA is it "SIP aware"?
03:10.07Snoogani'll check through the settings now
03:10.40*** join/#asterisk tflgen2 (n=clay@fl-67-233-27-60.dhcp.embarqhsd.net)
03:11.48*** join/#asterisk darkmadda (n=none@c-76-27-95-83.hsd1.ut.comcast.net)
03:13.12*** join/#asterisk Katty (n=Katty@mail.copi-rite.com)
03:13.22Kattymew?
03:13.36jayteemew
03:13.50Kattyjmirc has odd colors
03:13.57KattyHerroes
03:14.27jayteerowr
03:14.49Kattywhatcha doin
03:14.53[TK]D-FenderKatty: Mew.
03:15.08KattyHi fender
03:15.18jayteewatchin tv, playin Mafia Wars and watching [TK]D-Fender beat sense into newbs
03:15.37Kattyfun times
03:15.43[TK]D-Fenderjaytee: This one is getting it easy...
03:16.01jaytee[TK]D-Fender, yeah it's a case of [TK]D-Fender Lite
03:16.09KattyI'm in bed, avoiding sleep
03:16.26[TK]D-Fenderjaytee: Diet-D-Fender : just like the real thing, only half the ClueBat
03:16.30[TK]D-Fender....(tm)
03:16.41Kattyhehe
03:16.49jayteehahaha
03:17.01jayteeKatty, why are you avoiding sleeping?
03:17.08KattyFun size!
03:17.29Kattyno particular reason
03:17.58Kattyjust tinkering with the blackberry
03:18.12Snoogan[TK]D-Fenderk: can't find anything relating to sip
03:18.30Snoogani will try a firmware upgrade and get back to you shortly
03:18.44KattyI'm going to invent a sip fax
03:18.44*** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de)
03:19.20Kattyand sell it to kyocera
03:19.36jayteeawesome!
03:20.06Kattyyes. It will be grand
03:20.49KattyI will make billions, retire, and adopt jaytee
03:20.54jayteehad a 3 minute network outage today at work. all network traffic.
03:21.10eppigyhi Katty
03:21.12Kattyfrom what
03:21.19eppigyDONDE
03:21.24*** part/#asterisk Snoogan (n=asdf@115.69.179.5)
03:21.29Kattyherro
03:22.06jayteeerror logs on the Cisco Catalyst 4507 that is our main switch and includes our backbone registered about 3 minutes worth of HOSTFLAPPING errors on various gigabit fiber ports.
03:22.22jayteethe network engineer still hasn't figured it out
03:22.30Kattyeeeesha
03:22.37eppigythat is a terrible position to be in
03:22.41eppigyWE NEED A ROOT CAUSE
03:22.48tflgen2ouch
03:23.11KattyHow about a rootbeer float instead
03:23.16eppigy:D
03:23.24jayteewell, shouldn't get HOSTFLAPPING with spanning tree so odds are the Catalyst had some kind of "brainfart" and healed itself
03:23.55tflgen2had some issues with a 2811 earlier this month
03:23.58KattyI'd call cisco )=
03:24.00tflgen2voip related
03:24.33eppigyhttp://www.cisco.com/en/US/products/hw/switches/ps4324/products_tech_note09186a008063c36f.shtml
03:24.34eppigylol
03:25.09jayteethe errors were on multiple VLANS and ports and none of the connect points are close enough to each other that someone could have plugged in a cable looping any of them.
03:26.00tflgen2jaytee: hmm, sure that someone didn't misconfigure forwarding on a box that had access to multiple vlans?
03:26.40jayteetflgen2, we're a 5 person shop. Of the 3 of us that might make a net change all of us were at lunch
03:27.10eppigyi like the rootbeer float idea
03:27.29jayteeeppigy, this message is the one we were getting http://www.cisco.com/en/US/products/hw/switches/ps4324/products_tech_note09186a008063c36f.shtml#cg1
03:27.47jayteebut across multiple ports and from multiple addresses.
03:28.11eppigyi would question the otters
03:28.36jayteehahhahaa
03:29.03jayteemy first on the list of "Usual Suspects" is the lemurs and then Keyser Sose
03:29.48eppigytrue
03:36.32darkmaddaso i'm new to asterisk so i could use some help. Here is what i want to do. I'd like to make a dial plan/outgoing route. When a call is placed and matches a pattern i'd like it to place the call on hold. execute a script(bash/perl/php doesn't matter), then it changes an incoming route so that it connects to the line placed on hold instead of ringing it's normal extension.
03:36.53darkmaddaafter that the outgoing route should revert back to it's initial state.
03:37.37darkmaddaplaying a message to the call on hold would also be nice.
03:38.33darkmaddai have no clue how to do any of that. :-)
03:51.09*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
04:16.53Faizis there a particular reason why the asterisk (1.6) CLI doesn't reload modules properly?
04:16.55Faizit hangs after i issue 'module unload chan_dahdi.co'
04:16.56Faizsince it does not allow me to run dahdi commands
04:28.46*** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au)
04:36.34*** join/#asterisk voxter (n=voxter@190.241.15.56)
05:32.04*** join/#asterisk Snoogan (n=asdf@115.69.179.5)
05:32.54Snoogan[TK]D-Fender: my problem is in my modem.  Tried a different dsl modem and it registered immediately and both calls in and out are working
05:40.42kn0xSnoogan: what does it have some kind of builtin router?
05:46.53FaizI'm sorry for sounding like a noob, but I still can't seem to understand why I can't load DAHDI modules from my asterisk CLI (v. 1.6) ?
05:47.21Faizis there a particular version of DAHDI linux+tools I need to install for ast.1.6.11 ?
05:54.57*** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au)
05:57.11drmessano1.6.11 is out????
05:57.19Faizyes
05:57.20drmessanoZOMG TORRENT PLZ!!!
05:57.49Faizwget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz
05:58.27drmessanoThats 1.6.1.1 not 1.6.11
05:58.32drmessano:(
05:58.50Faizmy apologies for the typo
05:59.02drmessano:``(
05:59.04Faizbut perchance, do you have any idea regarding my situation?
05:59.05[TK]D-FenderSnoogan: Glad you found it
05:59.39Faizi have installed asterisk after making sure that dahdi was started
06:00.01Faizconfigured the extensions, chan_dahdi and system.config files and double-checked thoroughly
06:00.01drmessanohmmm
06:00.53drmessanoShow us some CLI output
06:00.56drmessano~pb
06:00.57infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
06:00.58Faizsure
06:03.33Faizalso, when I try to issue command: "module unload chan_dahdi.so", i do not get an output, and the CLI hangs
06:04.13*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
06:04.34leifmadseny0
06:05.09leifmadsenthere will never be a 1.6.11
06:05.14leifmadsen1.6.11.0 maybe
06:06.36Faizagain, my apologies as it was a typo
06:06.40Faizmy pastebin: http://pastebin.com/m1c5fc08f
06:07.29Faizyou can see at line 117 that when i issue chan_dahdi.co to stop, all succeeding commands are ignored as the CLI hangs
06:07.53Faizi CTRL+Z'd out of the CLI, and included the DAHDI configs
06:10.00leifmadsendid you compile asterisk after installing dahdi
06:10.02leifmadsen?
06:10.15leifmadsenorder should be:   libpri, dahdi, asterisk
06:10.27leifmadsen(order of compilation and installation
06:10.35Faizyes, libpri first, dahdi, restarted and ran as: /etc/init.d/dahdi start
06:10.42Faizcompiled and installed asterisk
06:10.44*** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye)
06:10.50Faizsaw the chan_dahdi.so file being parsed
06:11.10leifmadsenwell, that *should* work fine
06:11.13leifmadsenwhat version of dahdi?
06:11.23leifmadsenwhat OS?
06:11.36Faizversion 2.2.0.2, tools: 2.2.0
06:11.39FaizCentOS v 5.3
06:11.50leifmadsennever had a problem with that configuration
06:12.00leifmadsenalthough 1.6.1.1 is getting pretty old
06:12.05leifmadsenyou should try 1.6.1 branch
06:12.15Faizhm
06:12.17leifmadsensvn co http://svn.asterisk.org/svn/asterisk/branches/1.6.1
06:12.21Faizthank you
06:12.31leifmadsen1.6.1.1 is a security release after 1.6.1.0
06:12.45Faizah
06:12.48leifmadsenwhich was like..... June 5th
06:12.49Faizi'm updating as we speak
06:12.56leifmadseni.e. 1.6.1.1 is 1.6.1.0
06:13.06Faizah, i see
06:13.10leifmadsenand 1.6.1.0 is even older, and we all know what .0 releases are
06:13.13leifmadsen(useless)
06:13.17Faizheh
06:13.24Faizshould i restart the machine?
06:13.26Faizor just asterisk
06:13.27leifmadsenI say that, and I'm even the one who makes the releases
06:13.31leifmadsenjust asterisk
06:13.38leifmadsenno point in restarting the machine
06:13.39Faizvery cool
06:13.49leifmadsencheck the ChangeLog :)
06:14.07leifmadsenanyways, you shouldn't have those locking issues
06:14.15leifmadsenit looks like you're getting deadlocks
06:14.38leifmadsenadd the DONT_OPTIMIZE and MALLOC_DEBUG flags under the Compiler Flags option of menuselect
06:14.47leifmadsenthen you can do 'core show locks' to see if you're getting deadlocks
06:14.54leifmadsen(which shouldn't happen)
06:15.03Faizalright
06:15.08leifmadsensuggest you try 1.6.1 branch with DAHDI verision you're running
06:16.18Faizi'll re-run the menuselect now
06:16.26Faizbut dahdi is still not being recognized by the asterisk CLI
06:16.57leifmadsenhuh?
06:17.05leifmadsenis chan_dahdi.so being loaded?
06:17.29Faiznope
06:17.33Faiznor unloaded
06:17.34leifmadsenthen it isn't being compiled
06:17.39leifmadsen./configure
06:17.52leifmadsenit will pick up dahdi if it was installed
06:17.56Faizshould i pastebin my ./configure output?
06:17.59leifmadsenno
06:18.04carrarheh
06:18.07leifmadsenit'll be obvious if it found it
06:18.43leifmadseninstall dahdi, run ./configure in asterisk source dir, install asterisk -- you will see chan_dahdi in the Channels section of menuselect if it found it
06:18.45[TK]D-Fenderleifmadsen: Just because its obvious... doesn't mean he's not BLIND :p
06:19.03leifmadsenotherwise, run "asterisk -cvvvg" to see the modules load, then look for "WARNING" and "ERROR"
06:19.20leifmadsenyou may have a couple WARNING, but never should have an ERROR
06:19.51leifmadsen[TK]D-Fender: your a first class dick...
06:19.54leifmadsenyou're*
06:20.50[TK]D-Fender*cough* that's what she said...
06:20.58[TK]D-Fenderzing!
06:21.19Faizwhen i run the CLI and issue: module show
06:21.24Faizchan_dahdi shows up on the list
06:21.52Faiz(i ran with asterisk -rvvvvg and it did not show me a list of modules loaded
06:22.25carrarwould 3rd class be better then 1st class in this case?
06:22.29leifmadsenyou need to stop asterisk, and run it in the foreground
06:22.37[TK]D-Fenderfalz: "r" will not show you on start because you're connecting to a runnign process
06:23.28Faizah, misread, sorry
06:25.24Faiz"[Jul 25 02:24:06] ERROR[11647]: codec_dahdi.c:626 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory"
06:25.33leifmadsenno biggie
06:25.48Faizas you said, few warnings,
06:26.27Faizbut it isnt responding now
06:29.18drmessanoYou broke it
06:29.24drmessanoBad cop, no donut
06:30.08carrarBad cop, you're on video
06:30.42Faizsigh.. well I can't get a proper readout, since it hangs and I can't exit out
06:31.12carrarSCREWED
06:31.55drmessanotaskmgr > select asterisk.exe > end task
06:32.03carrarhahah
06:33.41drmessanoI spent more time getting ready to go geocaching than I do actually caching
06:33.43drmessanoThis is nuts
06:34.04carraryeah
06:34.27carrarWhere is the thrill in the hunt when you know right where to go
06:34.43drmessanoIf only that were the case
06:35.54carrarOnce there, look for the funny looking rock, 6 paces to the left and you will find a box with a eraser in it
06:36.00drmessano20ft accuracy of your GPS combined with 20ft accuracy of the person setting the cache + muggles + sneaky bastards
06:36.23drmessanoEraser?  Damn.. thats more than I normally get.. normally its all microcaches
06:36.31carrarheh
06:36.41carrarI once got a MicroSoft optical mouse from one
06:36.58carrargranted I do live in the capitcal of MicroSoft
06:37.05carrarcapital
06:37.24drmessano45 mins looking for a film can shoved in a fake log under a dead dog
06:37.30drmessanoTHAT is caching
06:37.35carrarhahah
06:37.45drmessanoand the hints
06:37.51Faizjust to make sure, install DAHDI as so:  make clean, ./configure, make, make install
06:37.53Faizcorrect?
06:38.08carrarand dahdi tools
06:38.41Faizand then proceed to asterisk as such:  make clean, ./configure, make menuselect, make install
06:38.41Faizyes, tools as well
06:38.41drmessano"Old mother hubbard, went to her cupboard, and a tree wouldnt leave her alone"... Which means look under the dead dog for the fake log
06:38.41drmessanoZOMG how didnt I see that?
06:38.55carrarno fake dog under a dead log?
06:39.14drmessanocarrar: That was the second one in the multi-cache
06:39.58drmessanocarrar: The third one was a 125lb pit bull with a pink ribbon on its neck holding a wet tree limb looking dog toy and the cache was hidden in his food dish
06:40.42drmessanoThat cache was called "Enuff with the Dogs, call 911"
06:41.19Faizis there a way i can tell that dahdi has been installed correctly, before i proceed to compiling asterisk?
06:41.38drmessanoQuestion..
06:41.51drmessanoHave you upgraded your kernel recently and not yet rebooted?
06:42.01Faizno
06:42.13Faizis my /usr/src/kernel folder supposed to have 1 or 2 directories?
06:42.56drmessanoDo you have two for two different kernels?
06:42.58carrarspeaking of kernels I need to upgrade my 2.6.30 box
06:43.23carrar.3
06:43.39Faizi'm not sure, i havent touched the OS settings
06:43.49drmessanoFuckin A
06:43.50Faizbesides configuring the network adapter
06:44.04drmessanoWhat are the DIRECTORY NAMES?
06:44.10carrarwait you really are running in windows?
06:44.12drmessanoDo they represent KERNELS?
06:44.34Faizissuing ls under /usr/src/kernels the directory names are:
06:44.46Faiz2.6.18-128.2.1.el5-i686  2.6.18-128.el5-i686
06:45.01drmessanoDo a uname -r
06:45.17Faizi get: 2.6.18-128.2.1.el5
06:45.26Faizis there an issue with symbolic links?
06:45.41drmessanoOnly if you make it an issue
06:46.08Faizi've been stressing about this for some time now
06:46.15Faizi just want  to hear a dialtone on my phone :(
06:46.43Faiznonetheless, i'll recompile dahdi kernel + tools, restart, start the service, compile asterisk
06:50.17carrarYou don't need * to hear a dialtone!!
06:51.39drmessanoNo, just a Yaesu FT-2400m and a lid with the repeater cheat sheet
06:51.49drmessanoDialtone ALL DAY
06:53.12drmessanoWas chatzilla not updated for 3.5?
06:54.34Faizdrmessano, would you like to see my ./configure output?
06:54.48drmessanosure
06:57.45Faizhttp://pastebin.com/m671aa4a1
06:58.57drmessanoOk
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06:59.06drmessanoLooks good to me
06:59.20Faizalright, proceeding to compile
06:59.30Faizdahdi service is running
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07:15.18Faizheh, holy hell
07:15.47Faizi think it was the USB_Radio module that was causing asterisk to hang
07:16.14Faizran asterisk -cvvv and got a clean scan; dahdi show channels has a proper output
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07:28.33Faizwow, let the fun begin
07:28.37Faizthanks to all that helped me
07:29.03Faizi'll be sure to document this as a guide for estranged folks in the future
07:29.08Faizgood night
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07:53.05pifhow does one set a minimum ping time for sip peers again?
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08:07.15pifhow do I test a function at the console? for example SIPPEER()
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09:02.13VooDooNOFXHey guys. I'm going through the astbook, and i've configured the base sip.conf. In the CLI, I get a register, then an immediate unregister for my xlite client in *1.4.26
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09:49.18ramindiaVooDooNOFX:  check over cli what happends when it unregisters.. enable debug
09:49.58VooDooNOFXi've got -vvvr enabled
09:50.14VooDooNOFXAll I get is "Registered SIP '1000' at 71.104.35.56 port 52184"
09:50.29VooDooNOFXthen on the next line "Unregistered SIP '1000'" immediately
09:50.50ramindiais the xlite behind nad ?
09:50.52ramindiaNAT ?
09:51.01VooDooNOFXyes
09:51.12ramindiadid u enable nat=yes
09:51.30VooDooNOFXhost-dynamic
09:52.46VooDooNOFXyeah, it was the NAT issue.
09:52.53VooDooNOFXty :D
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09:55.19VooDooNOFXOther than in the astbook, is there a single site to get phone configs for asterisk?
09:56.20ramindiavoip-info.org
09:56.55ramindiaget 2 books of *..both are well enough if u playing with *
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09:58.28VooDooNOFXyeah, the * book is in the mail, but it won't be here till Aug 4th or something
09:58.33VooDooNOFXstupid Borders bookstores
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10:49.14VooDooNOFXI've got some Aastra 55i's from an old packet8 system. Has anyone sucessfully reflashed these back to astra defaults so I can remove the Packet8 branding logos and such?
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11:00.14VooDooNOFXIs there a seperate loader firmware I need to load on this thing? Seems packet8's loader code fetches the newest firmware on every reboot from their servers.
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11:10.04VooDooNOFXOh, I think I found it. Perhaps someone can tell me how to make the aastra.xml Config file for this phone's auto-provisioning
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11:48.46tokozedghi, i have two asterisk connected with iax2, and on one side i make a outgoing call to through another *, and for outgoing application i use txfax, so it begins to send fax, and on another side i have rxfax at incoming extension
11:49.13tokozedgbut i cant receive fax? codec is g711
11:49.36tokozedgsr, i cant receive fax, and whats is the problem?
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13:15.07marl_scothi folks, anyone know if there is a way to show what commands are being sent through the manager interface? i am trying to debug a program that is connecting to the manager interface, but need to know what is being sent and recived by the manager interface something like : manager set debug ?
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13:34.59mwallingmarl_scot: i couldnt find anything, and ended up to watching packet captures
13:35.22marl_scotok, thanks, will go down that rout, just thought id ask first
13:35.55mwallingtheres a good chance i'm wrong
13:36.05marl_scotthink i might see if i can do something that will act as a passthrough proxy type thing, cus its a problem ive had in the past
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14:43.52artemmakhutovHello, is it possible to use wideband MOH with asterisk 1.6.2-beta3 ?
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14:51.43OrbixxAny idea why Record() is not responding to #?
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15:03.33OrbixxThis is so wierd.
15:04.37Orbixxhttp://pastebin.com/d451e52e6
15:04.44OrbixxAnything look wierd with that?
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15:38.48drmessanoWideband MOH?
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15:48.40Orbixxdrmessano: wav and alaw
15:49.19Orbixx[TK]D-Fender: Know of any issues that would cause Record() to ignore # to terminate the recording? (other than the option to ignore it)
15:49.33artemmakhutov16 KHz MOH with G722
15:49.56[TK]D-FenderOrbixx: Yeah, * not getting DTMF at all because its mode doesn't match
15:50.31Orbixx[TK]D-Fender: To get to the bit that does Record(), I have to traverse an IVR.
15:50.40OrbixxWhich would suggest DTMF is working, would it not?
15:51.07[TK]D-FenderOrbixx: I suggest your show me something useful
15:51.27Orbixx[TK]D-Fender: http://sprunge.us/IiQW
15:51.49OrbixxThe problem is at ivr,4
15:52.46[TK]D-FenderThat looks like my timeout code...
15:55.13OrbixxPossibly, nabbed it off voip-info.
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15:56.59[TK]D-FenderShow a failed call with SIP debug as appropriate
15:58.29drmessanoOrbixx: [10:44] <artemmakhutov> Hello, is it possible to use wideband MOH with asterisk 1.6.2-beta3 ?  <-- I doubt thats what he's asking about.. which is why I wanted a clairification
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16:00.56Orbixx[TK]D-Fender: Not sure how to do that. Sorry.
16:01.19Orbixxdrmessano: Ah my bad, didn't see that.
16:02.44Orbixxah
16:02.48Orbixxfound it [TK]D-Fender
16:03.52[TK]D-FenderOrbixx: and the problem?
16:04.28Jumpieanybody ever heard of a real time invoice summary off a call? i'm trying to see if i can do this (i know itll take work, just wanna be sure its possible), i wanna make it when a call comes in or is made, it checks incoming/destination number against a customer database, if so, in addition to the regular CDR addition, i want it to be able to generate a summary of the call, time/date/ to/from, length, in a pdf and emailed
16:05.24Orbixx[TK]D-Fender: Don't see it.
16:06.29drmessanoJumpie: Of course you can
16:06.44Jumpiecool, i figured...it'd do a similar check like it does against blacklist database
16:06.51Jumpiejust create another one and add/delete accordingly
16:07.01Jumpieone of the first things thast processed in a call if ir ecall
16:07.06OrbixxIf it makes any difference, when I call with a SIP phone to the IVR and then select 4 for record - when I hit hash, it hangs up, when I dial from PSTN, it does nothing.
16:07.22Jumpiehit hash? maybe thast your problem, you're too stoned
16:07.33drmessanoThe only relevant asterisk bit is a SQL connection to a DB to parse data, and writing the CDR info.. thats very basic DB stuff.. all else you mention has 0% to do with Asterisk
16:07.34Jumpieoh....pound sign ;)
16:07.38Orbixxyeah
16:07.41OrbixxI'm a Brit :>
16:07.45Jumpierofl im messin
16:07.50Orbixxi know
16:08.04Jumpiedrmessano well i thought maybe the conversion to pdf/mail was also part of asterisk engine
16:08.13drmessanoNo
16:08.21Jumpieok
16:08.47drmessanoAsterisk wont write the app for you.. it will handle the call, parse your db and write your CDR.. the rest is up to your magic app
16:10.27artemmakhutovJumpie, maybe you should try out a2billing, its a pretty nice billing solution for asterisk or check out this page: http://www.voip-info.org/wiki/view/Asterisk+billing
16:10.31drmessanoAs far as mail goes, Asterisk has a very limited/basic/but useful function to generate some ascii text that it pushes to a spool folder your completely configured external SMTP MTA will send off for you
16:10.59drmessanoBut thats a voicemail function and as close as asterisk gets to caring about mail
16:11.14falzpwd
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18:52.42intel352hey all, I've gotten Asterisk 1.4 latest and FreePBX 2.5 latest installed on my Centos 5.3 server
18:52.52intel352but i'm unable to connect with a sip device
18:53.09intel352nmap shows the ports open (5060 specifically)
18:53.17Dovidwhat does the asterisk cli show ?
18:53.22Dovidtry sip debug
18:53.34intel352k, checking
18:54.13intel352now that's handy
18:55.38intel352okay, Dovid, I'm seeing where it's connecting to proxy01.sipphone.com (Gizmo5 for forwarding)
18:55.47intel352but it doesn't show any connection from me (using X-Lite)
18:56.26intel352ah, just saw one
18:56.26intel3521s
18:56.26ddickensoncan anyone help me understand what this error is trying to tell me?  http://pastebin.com/d5ca04f6a
18:57.42OrbixxIf I have a group of people in [this] context using extensions 200-299 who have voicemail boxes 300-399, what would be the best way to construct a dial plan to let them access their voicemail boxes?
18:58.07ariel_ddickenson, you have something configured incorrectly
18:58.15intel352Dovid, here's the sip debug output regarding my conn: http://pastebin.org/4195
18:59.12ddickensonariel_: I figured this much, but this was a running production system that I needed to update dahdi and asterisk code and no changes have been made, also I have an identical "backup" server that updated with no problem.  Any idea how to figure out what it is that is configured incorrectly?
19:00.43ariel_ddickenson do dahdi_cfg -vvv to see what it's error's are from the cli
19:01.20ddickensonI was about to say I'd already done that but I guess I haven't had a cli running while I did it... hang on
19:01.22ariel_intel352, 401 not authorized pass word or type of security would be the first to look at.
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19:03.34intel352thx ariel, I'm checking. turns out sipdroid on my g1 was still trying to connect (which apparently has problems with authentication), while X-Lite on my computer isn't generating any response
19:04.36ddickensonariel: yeah, no output to CLI, and no errors in the regular output from the command...
19:05.13ddickensonjust this: http://pastebin.com/d7a01ac6a
19:06.22ariel_then restart asterisk
19:06.38ddickensontried it..
19:06.44ddickensonjust now
19:08.44ddickensonside note: intel352:  How did you get sipdroid working?  I can get it to connect, but I have some sort of codec error or something and I think I have everything installed
19:09.46intel352ddickenson: not working, it's not auth'ing or something. same with x-lite. once i can find out the issue in my server conf, i'll be able to troubleshoot sipdroid :-)
19:11.31ddickensonintel352: gotcha, I had to add in a couple of extra things in my sip.conf like nat=yes and a few others that I didn't need for other phones/softphones on the same system to make it register, now I can make it ring but can't answer and if I call out I get "no compatible codecs, not accepting this offer!"
19:11.49intel352ddickenson: interesting, i'll check that out too ;-)
19:11.57intel352what codecs do you have setup?
19:12.24ddickensonpretty much everything but alaw, g729, and ilibc I think
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19:12.38ddickensoni think I got those right
19:13.16ddickensonon the business critical subject that has me working on a saturday... ariel, any other ides?
19:15.03tflgen2what is a minimum ram req for asterisk (1-2 active calls)? also is it a .deb already?
19:21.15[TK]D-Fendertflgen2: * can run on a dumb Linksys router.
19:21.50[TK]D-Fendertflgen2: And Debian has * in the repos
19:23.53tflgen2[TK]D-Fender: really? openwrt can support it?
19:26.45ddickenson[TK]D-Fender: Do you have any ideas on where I can go to at least figure out what that error means that I was talking to you about yesterday?  I've been searching and can't come up with anything.  Here's a refresher in case you don't remember http://pastebin.com/d1b1e7835
19:28.11[TK]D-Fenderddickenson: no idea
19:28.40ddickensondang..
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19:57.50ddickensonhow about this, can anyone tell me ALL the files chan_dahdi.so parses when it loads?
20:01.10ManxPowerddickenson: I believe it is chan_dahdi.conf and any files #include'd from that file.  If you're using some sort of "Asterisk GUI" then I doubt anyone knows other than people that use that GUI.  (We don't use GUIs here, those people are on a different channel)
20:03.37ddickensonManxPower: no gui involved.  I'm getting an error after recompiling new asterisk and dahdi codes.  this was a working system (pure asterisk system) and I have some config error that has come up on one of the two servers I updated and am trying to narrow down where it is
20:04.11ManxPowerWhat is the actual error message?
20:04.40ddickensonhang on a sec.  lemme get on some irc client other than my phone.  I just got back to office
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20:06.58ddickensonManxPower: http://pastebin.com/m52603664
20:07.39ManxPowerlooks to me like you might now have the DAHDI kernel modules loaded.
20:07.48ManxPowers/now/not/
20:07.49ddickensonas you can see it's getting through chan_dahdi.conf and then to users.conf and I'm pretty sure it should get to /etc/dahdi/system.conf somewhere but that file is identical to my working server
20:08.08ManxPowerno, /etc/system/dahdi.conf is read by the KERNEL drivers, not Asterisk
20:08.18ddickensonah
20:08.57ManxPowerdoes lsmod show any dahdi drivers loaded?
20:09.07ddickensonI didn't know that file existed... I've only written the /etc/dahdi/system.conf
20:09.30ddickensonyeah, it does
20:09.44ddickensonlooks pretty normal I think
20:09.53ManxPowerFirst you get the DAHDI kernel drivers loaded and configured correctly.  Then you tell asterisk (chan_dahdi.conf) about the dahdi config.
20:10.19ManxPowerwhat does (I think this is the correct filename) dahdi_cfg -vvv give you?  (might be dahdi_config -vvv"
20:10.27ddickensonwhere does that happen?  In the compile process?
20:10.32ManxPowerno.
20:10.36ManxPowerwhere does what happen?
20:10.40*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
20:11.38ManxPoweryup, "dahdi_cfg -vvv"
20:11.49ddickensonoutput of dahdi_cfg -vv looks normal http://pastebin.com/m223a0e0c
20:12.06*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:12.54ManxPowernow what is the chan_dahdi.conf contain?
20:14.19ddickensonhttp://pastebin.com/m1e9f99b
20:15.11ManxPoweryou sort of need a switchtype if you have PRI
20:16.17ddickensonI thought so too, those were commented out by digium support when I was first bringing the system online several months ago... Has been working good though..
20:16.39ddickensonIt's a nortel at CO, I've seen the stupid thing
20:18.53ManxPowertry it and see
20:19.00ddickensonjust uncomment?
20:20.23ddickensonactually, doesn't matter cuz I still have that other problem that won't let dahdi load (just tried anyway)
20:21.33ManxPowerare you running Asterisk as root
20:21.39ddickensonyeah
20:22.14ManxPowerwhat card do you have?
20:22.52ddickensonte410p
20:23.58ManxPowerwhat specific kernel drivers are loaded?
20:24.22ddickensonnoob questions, how do I check that?
20:25.03grandpapadotlsmod
20:25.15ddickensonI lsmod | grep dahdi but I don't know what I'm looking at
20:25.29ManxPowerno, pastebing the whole output, don't filter it
20:26.19ddickensonhttp://pastebin.com/pastebin.php?erase=m1e9f99b
20:26.34ddickensoncrap... http://pastebin.com/d4d529328
20:27.18*** join/#asterisk minsa (n=minsa@71.202.98.193)
20:27.45ManxPowerI don't see anything obviously wrong
20:28.17ManxPoweryou recompiled asterisk after you upgraded/installed DAHDI?
20:28.30ddickensondo you see something that should correspond to my card, or is it possible that I somehow didn't install the one for my card
20:28.41ManxPower(3:27:44 PM) ManxPower: I don't see anything obviously wrong
20:29.05ddickensonyeah, I actually recompiled new libpri, dahdi, then asterisk.  when that didn't work I tried recompiling old code in that order again and no go there either
20:29.26ManxPowerwhat is the output of "grep -i "pri" /var/lib/asterisk/modules/chan_dahdo.so"?
20:31.23ddickensonhttp://pastebin.com/d9657dd6
20:32.32*** join/#asterisk grantm (n=grant@68.142.138.4)
20:33.03ManxPowerI don't have an asterisk install handy to find the modules directoty.  Maybe /usr/lib/asterisk/modules
20:33.55ddickensonBinary file /usr/lib/asterisk/modules/chan_dahdi.so matces
20:34.09ddickensons/matces/matches
20:34.25grandpapadotomg, we just pulled off our first automated asterisk update to 46 servers using a script that was less than 25 lines, nice ... (pats self on back)
20:34.44grandpapadotTotal per server downtime: < 15 seconds
20:34.50ddickensonand I can't even manage one server... ouch!
20:34.58ManxPowerddickenson: I have no more suggestions
20:35.18ddickensonManxPower: thanks for the help anyway
20:35.23grandpapadotwget; configure; make; asterisk -rx "stop now"; make install; /etc/init.d/asterisk start
20:35.51grandpapadotoh, and a tar
20:40.49*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
20:41.31StroggYar!
20:41.42Strogghacks on his extensions.conf
20:46.34OrbixxIs it possible to do the following...
20:46.51Orbixx${EXTEN} should be equal to "200" for example.
20:46.58OrbixxI want to dial 300 programmatically.
20:47.18OrbixxI need to do something similar to Dial(${EXTEN}+100)
20:47.27OrbixxBut that obviously doesn't work, else I wouldn't be asking.
20:52.58StroggCan I use wav files with my asterisk dialplan or do I have to convert to ulaw?
20:56.31*** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net)
20:56.54WindowsUserStrogg: they need to be mono and 8khz sample cycle
20:58.02*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
20:58.07Strogghow do I tell asterisk where my sound files are?
20:58.23Stroggmy custom sound files.. I had someone record a bunch for me
20:58.44[TK]D-FenderStrogg: absolut path when you play them, or relative to the folder in asterisk.conf
20:58.52[TK]D-Fenderastlib
20:58.57Stroggahh ok. thanks!
20:59.22ManxPowerOrbixx: If you want EXTEN to be "300" thhen use a Goto
20:59.28ManxPowerGoto(300,1)
20:59.56ManxPowerThat is just about the ONLY way to do that other than dialing 300 from a phone.
21:00.08ManxPowerEXTEN = the currently executing extension.
21:00.55ManxPowerwhy not exten => 200,1,Goto(300,1)?
21:02.25[TK]D-FenderOrbixx: how do you Dial() "300"?  300 is not a tech or device name
21:05.17ManxPower[TK]D-Fender: It's the new chan_300!
21:06.42carraryeah
21:06.45carrarget with it
21:07.08carrarI'm always on the phone with 300
21:09.34*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
21:09.36L|NUXhello
21:10.17carrarHARRO
21:11.09L|NUXi have simple but compilcated query :)
21:12.20L|NUXhttp://www.pastie.org/private/kabdocstsp82saf7xyxq
21:12.26L|NUXi have these rules
21:12.38L|NUXwhat i want to do is to access them with single DID
21:13.05L|NUXmeans if user call on number it will see if the number is not listed in db it will ask for PIN
21:13.12L|NUXand list their CallerID into db
21:13.18L|NUXand next time give them call back
21:14.40OrbixxAny idea why when I call internally, and enter pins and stuff via DTMF they go through fine, yet when I call via PSTN, only the first digit of the pin gets caught and the authentication fails...?
21:16.02ManxPowerOrbixx: "PSTN" = ???
21:16.13ManxPowerSIP, IAX, Analog Zap, PRI, CAS T-1?
21:16.15grandpapadotOrbixx: Are your calls coming in via sip or direct through a card in your asterisk server?
21:16.42Orbixxgrandpapadot: They're coming in via IAX, which eventually leads to a PSTN and a landline phone number.
21:16.58OrbixxIf you see what I mean.
21:17.35ManxPowerOrbixx: So calls come in via ??? -> IAX2 -> Asterisk -> ?? -> analog standalone phone
21:18.11*** join/#asterisk SlicerDicer (n=SlicerDi@24.138.244.145)
21:18.30OrbixxCaller (PSTN) => Switch => Provider with number allocations => IAX2 => My Asterisk Server => IVR => SIP client
21:18.40ManxPowerI guess I could list every possible problem/fix for every possible combination of "PSTN" technologies, but I don't think you have the time for that.
21:18.48OrbixxThat ^ does not let anything but the first DTMF digit through.
21:19.05OrbixxSIP client => My Asterisk Server => IVR => SIP client
21:19.06ManxPowerOrbixx: Caller = analog phone or cellular phone?
21:19.11OrbixxLets all DMTF buttons through.
21:19.15OrbixxManxPower: Either.
21:19.25ManxPowerOrbixx: So the IVR is not detecting your DTMF?
21:19.49OrbixxManxPower: I'm not so sure, because it detects the first digit correctly without flaw.
21:20.03OrbixxIt just refuses to take into consideration any further digits pushed.
21:20.12ManxPowerIf the problem was just with Cellular I'd suggest turning on long DTMF codes, but since it also does not work from analog, then the only thing I can think of is your carrier has something screwed up.
21:20.46OrbixxAnd if I said I had my analog/cellular phones working with multiple DTMF digits before?
21:20.50OrbixxConfig issue :>
21:20.51ManxPowerOrbixx: pastebin the CLI output of a failed call?
21:20.53OrbixxI just don't know what.
21:21.06OrbixxSure.
21:23.14ManxPowerMost of the time IAX2 DTMF issues are caused by the carrier using SIP from their DID provider to their Asterisk box, then IAX2 to your Asterisk box.
21:24.10OrbixxHmm.
21:24.20OrbixxI have a Background() and a WaitExten()
21:24.31OrbixxIf I try to dial whilst Background() is executing, I get 1 digit through.
21:24.35ManxPowerwaits for the pastebin
21:24.46OrbixxIf I wait until WaitExten() executes, nothing gets through and it times out.
21:25.01OrbixxManxPower: I'm just saying, a pastebin doesn't say everything with Asterisk.
21:25.46*** join/#asterisk swaj (n=scott@unaffiliated/swaj)
21:25.59ManxPowerOrbixx: let go of my ears
21:26.12Orbixxhttp://pastebin.com/d6c0d5a8b
21:26.58swajWe just implemented a new Asterisk 1.4.18 PBX here, and the AMI and web interfaces are enabled.  I'm try to Originate calls programatically through AMI.  I'd like to ring the local extension first, and then dial an outside number.  I cannot get it to work.  It will ring my extension, but then it will give a busy signal when I pick up.
21:27.10swajI can get it to call other extensions, just not outside lines
21:27.35OrbixxManxPower: Here's the context the problem lies in: http://pastebin.com/d6741fbde
21:27.46OrbixxCLI output further up
21:28.11swajWhen calling with originate, I use context=default, exten=<numbertocall>, channel=Local/5002, priority=1, action=originate
21:28.43swajif I reverse things, I.E. channel=Zap/g2/<number>, exten=5002 -- it will work, but it calls out first
21:28.45*** join/#asterisk jmacz (n=mcorb@201.244.169.20)
21:29.01swajwhat I want it to do is ring my extension first and then let me listen while it calls out
21:29.02ManxPowerOrbixx: Now pastebin your [dialextension] context
21:29.10OrbixxI just did.
21:29.17Orbixxhttp://pastebin.com/d6741fbde
21:30.02swajany help would be greatly appreciated
21:30.36ManxPowerOrbixx: that is the entire context?
21:30.40OrbixxYes.
21:31.17ManxPowerswaj: chan_local requires Local/extension@CONTEXT
21:31.31swajManxPower: well I've tried using SIP too
21:32.04[TK]D-Fenderswaj: Channel: should point to your phone, and Context, Exten, Priority to an exten that will dial out
21:32.10ManxPowerOrbixx: your dialplan looks correct.  Iax has virtually no options for DTMF since IAX2 doesn't have DTMF issues.  Have you tried using SIP?
21:33.06OrbixxNo, I'll give that a shot.
21:33.14swaj[TK]D-Fender: so if I want to dial out from a local phone, I should do Action=Originate, Channel=Local/5002@DID_span_3, Priority=1, exten=8005551212 ?
21:33.56ManxPowerswaj: have it dial your phone, then handle everything else in the dialplan
21:34.26ManxPowerOrbixx: I suspect if you do an iax2 debug you'll see that the carrier is not sending all the digits.
21:35.44ManxPowerswaj: Define "dial out from the local phone"?  The only you can "dialout from a phone" is to pick up the phone and dial.
21:36.05swajManxPower: I guess I'm not understanding.  All I really want to do is call AMI or the HTTP interface (rawman) and have it ring my extension, then dial an external number
21:36.41ManxPowerswaj: I believe the term is "click-to-call"
21:37.12ManxPowerswaj: look into .call files rather than using AMI.  I don't know which way is better for you.
21:37.28swajManxPower: yes, basically I'm trying to write a one-touch-dial program where a user can click a button on their screen and it will dial their phone for them.
21:37.50ManxPowersounds like you want to use AMI to generate a call with 2 legs.  the first leg is your phone, the second leg is a PSTN number.
21:38.03OrbixxManxPower: iax2 debug output: http://sprunge.us/gHCC
21:38.33swajManxPower: yes basically.  Right now I can make it call other extensions... for example, if I set Channel to Local/5002 and Exten to 5001, it will ring my extension at 5002, and then ring the person at 5001
21:39.05swajManxPower: and I can make it work in reverse... setting exten to 5002 and channel to Zap/g2/8005551212 will call 8005551212 first, wait for them to answer, and then ring my extension
21:39.32ManxPowerOrbixx: looks to me like you may be getting a max of 2 DTMFs from your carrier
21:39.47ManxPowerswaj: how are you connecting to the PSTN?
21:39.49[TK]D-FenderSw* calls the CHANNEL first
21:39.53[TK]D-Fenderswaj: * calls the CHANNEL first
21:40.00[TK]D-Fenderswaj: THAT should point to your phone
21:40.34*** join/#asterisk serph (n=serph@70.50.133.67)
21:40.45swaj[TK]D-Fender: right!  so what I want to do is Channel = Local/5002, Exten = outside number, but when that happens, I get a busy signal after answering the phone, like it never completes the call
21:41.29ManxPowerOrbixx: Looks to me like you are not getting ANY DTMF from your carrier.
21:41.30swajManxPower: we have I think 2 PRI's coming from the phone company with a T1.  We're using SIP.
21:41.35[TK]D-Fenderswaj: Show me the full originate and your dialplan.
21:41.42ManxPowerswaj: as long as it's not analog
21:42.13OrbixxManxPower: That cannot be correct, as I have to choose an extension to get to the context I am having a problem with.
21:42.35Orbixx(and choosing said extension works with DTMF)
21:43.33[TK]D-FenderOrbixx: I never saw the failed call
21:43.57Orbixx[TK]D-Fender: Hmm?
21:45.15swaj[TK]D-Fender: the originate is this-> http://10.7.30.21/rawman?action=originate&channel=Local/5002&exten=6363573061&priority=1
21:45.16swaj[TK]D-Fender: unfortunately I don't know what you mean by dialplan
21:45.16OrbixxGood job, give him a local IP why don't you.
21:45.16ManxPowerOrbixx: Looks to me like the carrier is sending the DTMF begind and END frames but no actual dtmf frames.
21:45.37[TK]D-Fenderswaj: You are not specifying CONTEXTs anywhere!
21:45.43[TK]D-FenderSwDialplan = EXTENSIONS.CONF
21:45.54*** part/#asterisk ManxPower (n=manxpowe@69.73.94.162)
21:46.02swaj[TK]D-Fender: I use &context=default
21:47.12[TK]D-Fenderswaj: In that URI you specified nothing.
21:47.25[TK]D-Fenderswaj: pastebin a failed attempt an your dialplan.
21:47.26[TK]D-Fender~pb
21:47.27infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
21:47.32swaj[TK]D-Fender: http://pastie.org/558944
21:48.30[TK]D-Fenderswaj: [default] has no match for the # you are dialing <-
21:51.16swaj[TK]D-Fender: I think I'm starting to understand
21:52.04swaj[TK]D-Fender: I noticed my phone is set to use DialPlan1 -- is there a way to originate using that Dial Plan through AMI?
21:52.46[TK]D-Fenderswaj: your local channel should specify the context, as should your context field.
21:53.09[TK]D-Fender[17:31]<ManxPower>swaj: chan_local requires Local/extension@CONTEXT <---
21:53.37[TK]D-Fenderswaj: and ManxPower already told you how to properly format that line
21:56.47swajsigh
21:57.26swajhttp://10.7.30.21/rawman?action=originate&channel=Local/5002@numberplan-custom-1&exten=6363573061&priority=1&context=numberplan-custom-1
21:57.40swajI used the plan in extensions.conf
21:58.49swajwait I got it
21:58.49swajneed the 91 in front of the exten
22:03.46*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
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22:38.56Dovid~pb
22:38.57infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit
22:39.00Dovid~google
22:39.01infobothmm... google is http://lmgtfy.com/?q=google
22:39.56Dovid~trfm
22:40.02Dovid~rtfm
22:40.02infobothmm... rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM
22:55.53*** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com)
23:06.51WindowsUser~spam
23:06.52infobotit has been said that spam is a preferred environment.  SPAM; Shut up, You damn Vikings! SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM, or to destroy it, try spamassassin (spamd+spamc) and razor.
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