irclog2html for #asterisk on 20051216

00:00.17justinuit'll make your MoH sound like shit tho :)
00:00.20TheCopsjustinu, i have a 6mbps/800kbps ADSL links, and I want to be able to get a PRI via voip
00:00.21Ariel_if you need to save bw but if you have the bw then ulaw is better sounding
00:00.55TheCopsAriel_, people say that 729 is very similar to ulaw
00:01.03TheCopstrue or not ?
00:01.09Ariel_not in all cases
00:01.14TheCops:/
00:01.38TheCopsjustinu, do you think I'll be able with 729 ?
00:01.45Ariel_if you really listen to tones there is a big difference in the sound
00:02.04*** join/#asterisk gnosys (n=ksford@ip68-9-201-250.ri.ri.cox.net)
00:02.08TheCopsAriel_, nice to hear, this is for incomming call hehe
00:02.40FuriousGeorgeanyone ever experience fluctuations of the volume on voicemail recordings
00:03.47*** join/#asterisk johngalt (i=John@64.30.193.55)
00:03.49rob0gnosys: you have to look in your /lib/modules/`uname -r` to see if the module is there
00:03.49FuriousGeorgerecordings=the messages left, not the greetings
00:04.57johngaltanyone care to comment on what happened and how openpbx got started - just curious and trying to get input from both sides.
00:05.04gnosysthanks, rob0. It is there.  It inserted properly.  The problem seems to be that there is no wcfxS module present.  And when I look in the sources, I see none there either.  Only a wcfxsUSB (my emphasis)...
00:05.32Ariel_argh windows sucks. I have to reboot this damm laptop.....
00:06.11rob0dmesg | tail # after you try to load it
00:06.31justinuyou can probably get 20+ channels out of 800kbps on G729
00:12.27gnosysya, i thought o' that too:  Zapata Telephony Interface Registered on major 196 is all that is there.  The error I'm getting when trying to load wctdm is this:FATAL: Module wcfxs not found.
00:13.00gnosysand indeed, there is no wcfxs module present in /lib/modules/`uname -r`/misc
00:13.25gnosysnor is there any wcfsx filename at all in the sources except for wcfxsusb
00:26.35hugo-v6whats the difference ebtween extensions.conf and extensions.ael?
00:27.56Gamercjmhmm for playback() does asterisk support .wav?
00:28.36TheCopsjustinu, more then 20?
00:28.50robl^Gamercjm: yes, depending on the format of the wav file
00:29.03justinug729 is ~8kbps/sec
00:29.20TheCopsjustinu, nice
00:30.02Gamercjmwhat formats are accepted? I downloaded a converter and converted a wav to a gsm wav, but didnt work
00:33.16*** join/#asterisk anthm (n=anthm@h4608c598.area4.spcsdns.net)
00:33.16*** mode/#asterisk [+o anthm] by ChanServ
00:36.29gnosysrob0?
00:40.25*** join/#asterisk Anjo_Malvado (n=eduardo@200.96.86.29)
00:43.00tainted-anyone use voicepulse for DIDs?
00:43.34justinutried to
00:43.44justinuquality is ass
00:44.08*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
00:49.44tainted-by ass do u mean good or bad
00:49.58justinubad
00:50.01tainted-cause there's good and bad ass
00:50.08tainted-did u use iax or sip
00:50.16justinutried both, iax seems to work a little better
00:50.27tainted-what problems where u having
00:50.32tainted-audio problems?
00:50.35justinunoisy calls
00:50.35*** join/#asterisk nagle (n=nagle@206-248-152-13.dsl.teksavvy.com)
00:50.36tainted-how long ago?
00:50.38justinuyeah
00:50.46justinuweek or two
00:50.47tainted-hmm.. that's actually what i'm getting too
00:51.01justinuthen I tried asterlink, and it's perfect
00:53.08tainted-what are their rates for incoming
00:53.36justinui don't remember, but they're toll free only
00:53.45justinutry joining #asterlink
00:56.21*** join/#asterisk coppice (n=chatzill@82.194.17.210.dyn.pacific.net.hk)
00:58.24*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
00:58.55*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
00:59.12alephcomHello Everyone
00:59.46dokhenchhowdy alephcom
01:01.33*** join/#asterisk javar (n=javar@69.79.133.185)
01:02.14alephcomNo big flamewars going on?? :-(
01:02.18alephcom:-)
01:02.33*** part/#asterisk mog_work (n=mogorman@gateway.digium.com)
01:03.42justinubeen very quiet today
01:04.09*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
01:04.20Dr-Linuxanybody can help for FOP
01:04.42Dr-Linuxi installed everything but haveing an error while run ./op_server.pl  file
01:04.43Dr-Linux[root@I2C-PBX i2c]# ./op_server.pl
01:04.43Dr-LinuxCould not write configuration data /use/local/apache2/htdocs/i2c/variables.txt.
01:04.43Dr-LinuxCheck your file permissions
01:04.43Dr-Linux[root@I2C-PBX i2c]#
01:04.56Dr-Linuxeven, the file doesn't exist :S
01:06.16alephcomjustinu: That's not all bad.
01:06.29justinunah, not at all
01:07.53*** join/#asterisk ManxPower (n=ewieling@host-216-226-240-72.wdb.ses-americom.net)
01:09.18*** join/#asterisk ShadowMaster1 (n=askme@host89-133.rancor.birch.net)
01:12.03ShadowMaster1I am seeking help with an error I am getting with my voip provider regarding a bad codec.  Is it bad form to paste the 1 line error message here or shoud I paste it on the code past web site?
01:12.25drraypaste it here
01:12.51ShadowMaster1I have a SIP trace on this as well..  WARNING[17166]: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: video 9058 RTP/AVP 31 34
01:13.02ShadowMaster1I am not tryingto do any type of video..
01:15.40ShadowMaster1It didn't matter if it was pre 1.2 or on version 1.2..  It also gives the same error on my new install of Asterisk@Home..  AXVoice the provider says they have others who work fine, but will not troubleshoot.
01:16.06drraywhich client?
01:16.31ShadowMaster1This occurs when trying to place a call into asterisk..
01:16.40drraywhich type of client
01:16.42file[laptop]from your provider?
01:16.49drraywhat kind of phone
01:16.50drray?
01:16.58ShadowMaster1yes..  From the AXVoice DID..  it never gets to my dial plan..
01:17.17file[laptop]they're sending you a video offer in the SDP?
01:17.19file[laptop]fun.
01:18.01ShadowMaster1That is what I was seeing as well.  Is there any way I can, from my side, reject that offer without rejecting the whole call?
01:19.00ShadowMaster1The offer for video that is..
01:19.01Dr-Linuxjustinu: i read in google  Qwell is helping a guy who has same problem that i have with FOP
01:19.39*** join/#asterisk toddf (n=toddf@ns0.fries.net)
01:22.35hugo-v6Dr-Linux: did u check the perms?
01:22.39*** join/#asterisk Mavantix (n=maverick@69-168-33-232.chvlva.adelphia.net)
01:23.22Mavantixwhooohoo, I got my phones and just compiled asterisk!! The fun begins! :)
01:26.01Mavantix..channel dead tonight?
01:26.40ptiggerdinenope.
01:26.50ptiggerdineall working on asterisk :)
01:26.54Mavantixheh
01:26.54Anjo_Malvadohehe
01:27.30Mavantixwell, it's running, but, I don't know what Im doing with it now... I guess I need to add a sip registration for a phone and try and get that working?
01:28.11jsaundersGood thinkin' Billy.
01:28.27MavantixI'm RTFM'ing ;)
01:28.34*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
01:30.15Dr-Linuxanyone is using FOP?
01:30.30*** part/#asterisk Anjo_Malvado (n=eduardo@200.96.86.29)
01:31.01javarwhat does FOP?
01:31.14jsaundershttp://xmlgraphics.apache.org/fop/
01:31.24Dr-Linuxjavar: Flash Operator Panel for asterisk
01:31.32jsaundersHeheh, guess I'm wrong.  :)
01:31.37javarthx
01:31.39Dr-LinuxCould not write configuration data /use/local/apache2/htdocs/i2c/variables.txt.
01:31.46*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
01:31.56Dr-Linuxi'm facing this problem while try to start it
01:32.51ShadowMaster1does the directory path exist?
01:33.07Dr-LinuxShadowMaster1:
01:33.19Dr-Linuxbut "variables.txt" file doesn't exist
01:34.10ShadowMaster1if it's trying to write it out, I would have thought it didn't need to exist.  Can you create one and see if it will update it?  Perhaps an empty file?  (I'm guessing as I'm new thoo this)
01:34.39Dr-Linuxhhm..
01:34.45Dr-LinuxShadowMaster1: let me try
01:35.30ShadowMaster1Dr:  It's worth the effort is it's that simple a fix..  IMHO..  Of course, I don't rate confidence high on that as a solution.
01:36.09Dr-Linuxhhm..
01:36.25Dr-LinuxShadowMaster1: same problem even i gave it full permissions
01:36.26Dr-Linux[root@I2C-PBX i2c]# ./op_server.pl
01:36.27Dr-LinuxCould not write configuration data /use/local/apache2/htdocs/i2c/variables.txt.
01:36.27Dr-LinuxCheck your file permissions
01:38.58Dr-Linux[Fri Dec 16 04:27:14 2005] [error] [client 72.129.86.208] File does not exist: /usr/local/apache2/htdocs/i2c/variables.txt
01:39.09Dr-Linuxthis is apache log :S
01:39.46*** join/#asterisk ldnblk (n=Just@212.183.128.185)
01:47.43robl^hrmmm..  has anyone come up with a way of using app_devstate to monitor parked calls?
01:48.43xachensweet
01:48.48xachenfound a good NSES emulator :)
01:48.52xachennow to download Mario Allstarts
01:48.56xachenAllstars*
01:49.28*** join/#asterisk azfhasterisk (n=rickb@VDSL-130-13-192-106.PHNX.QWEST.NET)
01:50.42justinuget yoshi's island too
01:52.39xachenok
01:52.39xachenthis is sad
01:52.40xachenMario is classic
01:52.45xachenand there is very few places that have it
01:52.48rob0asterlink $2/month and 2 cents/minute? Is that a typo?
01:53.12xachenWhy would that be a typo?
01:53.30rob0$2/mo seems low to me
01:53.34xachenerm
01:53.37xachenits pay as you go :)
01:53.54xachenyou pay 2c/min regardless if you call your neighbor or across the country
01:54.00file[laptop]it used to be $0/mth, but we started getting charged for having numbers
01:54.01rob0yes, I get that.
01:54.02xachenYoshis Island/Safardi?
01:54.12file[laptop]which is very very... so very expensive
01:54.12xachenCookie :P
01:54.15justinui dunno what safardi is
01:54.34xachenYogi Bear!
01:54.38xachenclassics :)
01:55.24xachensince all my voip stuff is on my laptop in a repair depot
01:55.29xachenI need to amuse myself with something
01:56.24ptiggerdineclassics are great
01:56.37justinui need to get an SNES emulator on my psp
01:56.59alephcomxachen: If that's the case, I'll lend you a box and you can spend the weekend programming for me. :-)  Free of course.
01:57.28xachenI work this weekend
01:57.28xachen:(
01:57.35xachenif it wass lastwekend I'd be more than happy haha
01:58.01alephcom:-)
01:59.58xachenits hard to code using SSH ond ialup
02:00.07file[laptop]that's death
02:00.14file[laptop]but, I'm spending the weekend on cellular...
02:00.17xachenwhich is why I was mad ewhen I completely goofed my CMOS
02:00.20file[laptop]so, I will feel the pain
02:00.36xachencellular is worse than dialup :(
02:01.11file[laptop]my provider's is about on par with dialup with latency
02:01.14file[laptop]except faster on the downstream
02:01.25xachenwell its not hte lag that kills on dialup
02:01.34xachenit actually works good 95% of the time amazingly with Telus
02:01.39xachenits just that your phone line is always tied up
02:02.07file[laptop]I intend on being on IRC while driving down... well, I'm not driving - that's just suicide
02:02.16trixterhttp://www.betanews.com/article/FCC_Wants_VoIP_Users_to_Pay_Tax_Too/1134671376   The FCC will likely force Internet telephone, or VoIP, providers to contribute to the Universal Service Fund -- when will the insanity end?
02:02.41alephcomxachen:  I sent a message to the guys the run the network up here and told them to swing a tower down by you guys.  :-)  Don't hold your breath waiting.
02:02.42xachenyeah
02:03.04xachenwe don't need spying anymore :p
02:03.11xachenthe guy here is a complete cracknut
02:03.31xachentrixster: Yes, the FC C wants complete control. They just hate VoIP because they can't regulate it as much.
02:06.48*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
02:07.58xachenlook at this: http://christmas.jason-m.com
02:08.02*** join/#asterisk JunK-Y (n=junky@67.71.108.114)
02:08.03xachenI want to do something like that know!
02:09.10*** join/#asterisk lexluther (n=s@62.56.236.237)
02:11.20*** join/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net)
02:13.05*** part/#asterisk Lurr (n=pr0ph3t@m615e36d0.tmodns.net)
02:13.27*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
02:21.34*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:21.42[TK]D-Fenderand... WHEEEEEEE!!!
02:25.08Mavantix[TK]D-Fender: got my phones!! :)
02:26.20justinuwhat kind?
02:26.57Mavantixpolycom 601s and 501s
02:27.01[TK]D-FenderYeah.. I forget which ones I advised :)
02:27.02justinugreat
02:27.04[TK]D-Fenderhehe
02:27.05[TK]D-Fendercool.
02:27.12justinui've got a 501 and 601 sitting on my desk
02:27.13Mavantixworking on setting up ftp provissioning
02:27.21justinuuse http
02:27.28Mavantixk
02:27.47[TK]D-FenderDon't use HTTP... means you need to run another service with its own set of hassles and is more snooped..
02:27.49Mavantixshould I use the files here: http://www.krisk.org/asterisk/pcom/
02:28.01robl^Polycom seems nice.. :)  Cisco has ceased being my favorite..  Snom is good.  :)
02:28.01MstlyHrmlsFTP means you can get logfile uploads without fiddling with getting HTTP PUT set up on your server
02:28.04[TK]D-FenderFTP is avaiable by default in practically all distros.
02:28.04justinui dunno, i'm a fan of http
02:28.17justinuit wasn't that hard to set up PUT,
02:28.21justinuuse HTTPS
02:28.27[TK]D-Fenderjustinu : Yeah it works, but FTP is just so much easier and less things to go wrong.
02:28.39justinui dunno, haven't had any problems using https
02:28.41MstlyHrmlsPUT is just One More Thing
02:28.51justinuthere's always just one more thing
02:28.55justinuit's technology
02:29.04justinui don't run ftp servers
02:29.07MstlyHrmlsplus, FTP gives you log appending by default
02:29.21MavantixI installed vsftp, but it's not on by default it seems
02:29.33MstlyHrmls*shrug* fair enough. however, if you're running neither, I'd say install FTP and use that :-)
02:30.15MavantixI'm fine with FTP, anyone want to let me know what config files I better be using? ...and do I need to upgrade the firmware in the phones?
02:30.19justinui like my provisioning to be availble over the net, so i use http
02:30.28justinui don't like running ftp over the net for whatever reason
02:30.43MstlyHrmlsjustinu: fair point
02:30.59Nuggetif it's not encrypted IT'S CRAP.
02:31.14MstlyHrmls\m/FTPS\m/
02:31.15*** join/#asterisk monachoi (n=bonvoyag@cpe-24-174-162-34.satx.res.rr.com)
02:31.17justinuyeah, which is wy new polycom boot loader does https
02:31.20justinuwhy
02:31.23MstlyHrmlsbest of both worlds! :-)
02:35.00[TK]D-FenderNugget : That last line in the tone of Billy Connelly no less ;)
02:36.21Nuggetthat's how I was saying it in my head.  :)
02:36.44*** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
02:42.13gnosysHi folks... just upgraded from 1.0.5 to 1.2.1 and ran into a problem loading module wcfxo with message: ZT_CHANCONFIG failed on channel 2: No such device or address (6).... but the module loads and dmesg output seems reasonable: wcfxo: DAA mode is 'FCC'; Found a Wildcard FXO: Wildcard X101P.  This is a 2.4 kernel.  Anybody see any boo boos here?
02:46.41[TK]D-Fendergnosys : did you download and recompile the new Zaptel as well?
02:47.10*** join/#asterisk junbug (n=harry@c-66-176-211-109.hsd1.fl.comcast.net)
02:47.53gnosysyup...  libpri too.  It actually seems to work (I'm fiddling with old modules like chan_modem.so now, which seems to be causing asterisk to fail to load, but the module stuff and ztcfg seemed to have worked.  Not sure what to make of the error message.
02:48.26rob02 zap channels and each is a different driver?
02:48.51rob0if so I think the error is harmless
02:50.01rob0ztcfg runs when each driver loads. When the first driver loads it gives an error because the second isn't loaded yet.
02:50.47gnosysDo I need chan_modem.so in 1.2.1?  I'm not even sure why I had it loading in the older version, but it's in my modules.conf file in /etc/asterisk just before res_musiconhold.so.  I did get a warning after asterisk make install about old modules, but I figured it wouldn't hurt to leave them there if they weren't loaded, but this one is getting loaded in modules.conf
02:53.55kshumard_homegnosys, noload it... you don't need or even want it
02:54.30kshumard_homeI think it'll be removed in 1.4
02:54.40*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
02:58.00*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
02:58.25*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
02:58.36FuriousGeorgedo i have to uninstall the old asterisk when i upgrade to 1.2.1 (from 1.0.9) or anything like that
02:59.16kshumard_homeyou should probably remove (or backup) /usr/lib/asterisk/modules, but that's about it
02:59.18kshumard_hometypically
02:59.46gnosysthanks guys.  seems to work ok now.  not sure what the error message was about.
03:00.05FuriousGeorgethanks kshumard
03:00.18kshumard_homenp
03:01.30GamercjmI have a DISA set-up how can i record all cames made from it?
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03:02.22*** mode/#asterisk [+o anthm] by ChanServ
03:03.50kshumard_homeGamercjm, you have an extension calling DISA, right? Just have a higher priority in that extension calling Monitor
03:03.56kshumard_homeor MixMonitor, prolly
03:04.09GamercjmWell its from a DID
03:04.28Gamercjmwould I do the same thing?
03:04.48kshumard_homesure
03:06.04QwellYou have got to be kidding me...
03:06.27Qwella cellphone company can't block a number from calling you...
03:07.26GamercjmI see that monitor asks for the filename to record as, So every call made will be in the same file unless i change the name everytime?
03:07.55kshumard_homewell you can use a channel variable like ${TIMESTAMP} or ${UNIQUEID} or something to have it dynamic
03:08.12kshumard_homecheck out /usr/src/asterisk/doc/README.variables to see what all the channel variables are
03:08.15Gamercjmooh i c
03:08.23Gamercjmyah ill do that, thanks
03:08.27kshumard_homenp
03:10.19gnosysI get so confused over the wcfxo versus wcfxs thing.  The wcfxs kernel module... has it been replaced by the wctdm module?  Is the wctdm module supposed to be for controlling PSTN lines or analog telephones?  I know the bit about wcfxo devices need wcfxs signalling, but it gets me so darned confused that I can never keep track of which is which.  If I want to load a module to drive a PSTN line.  Is that wcfxo or wctdm in 1.2.1?
03:11.06[TK]D-FenderWCFXO = X100p analog card.  WCFXS = TDM400 card
03:11.08kshumard_homeIn 1.2.1, wctdm is the driver for the TDM400P, whether it has FXO or FXS modules on it
03:11.30[TK]D-FenderHas been like this for years.
03:11.34CoaxDwhich makes no frickin sense
03:11.44CoaxDbut ok
03:11.56[TK]D-FenderCoaxD : My guess is they didn't think up FXO for the TDM card at first...
03:12.10[TK]D-Fendersupport modules)
03:12.12CoaxDtk: It has NOT been this way for years.
03:12.24CoaxDer
03:12.34kshumard_homeI wanna say it's been like that since 1.0.9 but not 1.0.7
03:12.41[TK]D-FenderCoaxD : At least 2... since I got in and had mine :)
03:12.42CoaxDi'm thinking i'm wrong; one used wcfxs.c to handle both wcfxs/wcfxo
03:12.53CoaxDon tdm400
03:13.09[TK]D-FenderWCFXO = x100p :p
03:13.20CoaxDya
03:13.34CoaxDit dont make sense.  x100p.c anyone
03:13.47kshumard_homeso ln -s it. : )
03:13.54kshumard_homewell
03:13.58CoaxDkshumard: I dont think one can insmod a symlink :P
03:14.01kshumard_homealias in /etc/modprobe.conf anyway
03:14.05kshumard_homeright
03:14.06CoaxDheh
03:14.21kshumard_homebut that's what we do with wct2xxp for the dual span t1/e1 card
03:14.27kshumard_homeit's just an alias of wct4xxp
03:15.19dokhenchany way to get a 3.x version bootrom out of a polycom, and back down to 2.6.1? <crossing fingers>
03:15.51hugo-v6hmz. sometimes compiling mISDN is a pain in the ass. im going to bed for now.
03:16.15gnosysOk, in one server that's been operating for quite awhile now (just upgraded from 1.0.5) I have a TDM400P and an X101P.  In it, I have zaptel.conf with the first channel as fxsks=1 so I load the kernel modules in this order with 1.2.1: zaptel, then wcfxo, then wctdm.  Right?  That seems to work just fine.  But in another server (trying to get it set up for the first time), I have a TDM04B (all PSTN lines).  In this case, I would only load the
03:16.45gnosysThe first server's TDM400P is all telephone lines.  No PSTN lines.
03:17.26[TK]D-Fenderdokhench : Why downgrade?
03:17.46MstlyHrmlsdokhench: nope, none whatsoever
03:18.39GamercjmHow would i get the number dialed after DISA, I cant seem to find a variable for it
03:18.49*** join/#asterisk tengulre11 (n=tengulre@61.185.224.66)
03:18.53dokhench[TK]D-Fender, I have some that are 3.1 and others 2.6.. if I put the 2.6 on the same net they are going to pull down the 3.1 bootrom. don't want them to be locked into the 3 bootrom.
03:19.33MstlyHrmlsdokhench: *shrug* don't put a bootrom on your server then
03:20.09kshumard_homeGamercjm, you could use the Read() app in the dialplan ahead of the outbound dial to store it in a variable yourself
03:20.27[TK]D-Fenderdokhench : I have that situatio in my lan and jsut set a different boot server setup for those ones.
03:20.48dokhenchMstlyHrmls, 2.6 doens't seem to have a problem with it, but the 3.1 ones will give me "error loading bootrom" then loop.
03:20.49[TK]D-FenderSince I use a non-standard FTP name its easy to seperate.
03:20.54Gamercjmbut wouldnt that wait for the numbers to be dialed first.. or does it continue?
03:21.16dokhench[TK]D-Fender, ah, good deal. so you are just doing different group based on mac addr in dhcpd?
03:21.26MstlyHrmlsdokhench: yeah, that's why you take the bootrom.ld file out of your ftproot :-)
03:22.05MstlyHrmlsdokhench: the official word is that bootroms don't need to be upgraded unless there is a compelling reason
03:22.56dokhenchMstlyHrmls, ya, but it sucks to up 6 phones to see whats new, only to find out later that they won't go back downlevel. =)
03:23.01kshumard_homeGamercjm, show application read
03:23.23MstlyHrmlsdokhench: heh, whups :-)
03:24.07[TK]D-Fenderdokhench : not even that tricky.  I changed the FTP account IN THE BOOTROM to a different account on the same server ip :)
03:25.17[TK]D-Fendercheap trick that took 10 second.  esp since you only have to do it once.
03:25.59MstlyHrmlsone of the cool features of 3.x is you can give the phone a URL via DHCP, and it will use it to provision from
03:26.11Gamercjmkshumard_home: not sure what you mean, have disa() then read()?
03:26.36MstlyHrmlsso, if you want to change the account a Polycom is booting from, you can just add a custom entry to your DHCP server
03:26.48kshumard_homeGamercjm, yeah
03:26.56Gamercjmk ill try that
03:28.58Qwelloh lord
03:29.04QwellSprint charges you to check your voicemail
03:29.13Qwellpcs that is
03:30.04{zombie}that's pretty common around here - charge for the diversion to voicemail, charge to retreive your voicemail - and THEN they make money on you returning the call to whomever left the message
03:30.08{zombie}triple whammy
03:31.03QwellThat's just rediculous.  I could use asterisk to call me with my own cidnum, and check it from there...for free...
03:31.37QwellOR, call myself from my wifes phone
03:31.53FuriousGeorgethat was easy and painless
03:32.39QwellSo...what is a provider that isn't complete shit?
03:33.41dokhenchqwell: i like tmobile.
03:34.18dokhenchqwell: gsm and 19.95 for gprs. not nearly as fast a evdo, but it does the job.
03:34.22Sedoroxnow sprint... :p
03:34.23Sedoroxbut yea...
03:34.23QwellSedorox: nextel == sprint == complete shit
03:34.32Sedoroxnextel doesn't charge for voicemail tho..
03:34.47Qwellwell, it isn't that they "charge"...it justs uses minutes
03:35.10SedoroxI don't believe mine uses minutes... I can look...
03:35.17Sedoroxmight...
03:35.22Qwellbut, when you see an extra $20 on your bill...caused by voicemail...
03:35.24Qwellit's a little troubling
03:35.28SedoroxI have 300 out.. unlimited in.. so...
03:35.31Sedoroxwell yea...
03:35.34SkramXhey all
03:35.49{zombie}Qwell: I divert my mobile to a telular gsm gateway connected to my asterisk box
03:35.59{zombie}and the SIM in the telular is on my plan
03:36.08{zombie}and I get free calls between them
03:36.23Qwellby free, you mean $20/mo?  heh
03:36.27*** join/#asterisk Gamercjm (n=gamercjm@71.254.174.15)
03:36.35{zombie}no, free calls between them. :)
03:36.46Qwellbut, the second account doesn't cost?
03:36.47dokhench{zombie}: does the gsm gateway just act as a phone? I mean, do you just pop a sim in it, and you can SMS, cell call, etc?
03:36.56{zombie}dokhench: yup
03:37.04{zombie}pity the telular is such a piece of shit
03:37.08Gamercjmhmm read() after didnt work
03:37.18{zombie}can't quite make it talk nicely to the sipura it's plugged into
03:37.21dokhench{zombie}: is this in the states?
03:37.24{zombie}the ericsson is much better
03:37.30{zombie}no, this is in australia
03:37.45{zombie}Qwell: the second account costs $8/month and includes $8 worth of calls
03:38.00{zombie}which is shared amongst the 3 phones, so gets used easily
03:38.35{zombie}dokhench: you can get mobile phone gateways to suit the US cellphone networks too, I'm sure
03:38.51Sedoroxyea.. they have gsm -> fxs gateways
03:38.55Sedoroxsome on ebay are cheap...
03:39.24{zombie}yup, I got the telular from eBay
03:39.29SedoroxI'd do that.. but I need a special one since I'm iden
03:39.38{zombie}but as I said, it sucks
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03:39.52{zombie}I have an ericsson f250m on loan atm which is much nicer
03:39.53Sedoroxlol
03:40.00{zombie}but what I really want is one of the voiceblue gateways or similar
03:40.02{zombie}SIP -> GSM
03:40.12SedoroxI know...
03:40.14SedoroxI wanna try one of them
03:40.23Sedoroxuse a old gsm phone I have...
03:40.41kshumard_homeGamercjm, yeah, I see now that won't work. Sorry!
03:41.03Gamercjmits ok
03:41.20{zombie}there's also gateways around that take primary rate ISDN (E1 or T1) into GSM
03:41.22kshumard_homeGamercjm, I'm not sure how you would best get the number that was dialed
03:41.26{zombie}30 channels at a time
03:41.34Sedoroxhmm
03:41.46SedoroxI wish they made a cheaper sip -> gsm gateway
03:41.54Gamercjmyah I was trying to have the Monitor files organized and for finding out who's called what
03:42.06SedoroxI'd go and pick up another v60g...
03:42.10SedoroxI loved that little phone
03:42.20Gamercjmbut i guess ill just go by the timestamps and looking over the phone record if i have to do something like that
03:42.39kshumard_homeGamercjm, what does the CDR say when you use DISA? Does it show the outbound call?
03:42.45kshumard_homeI've never actually used DISA.... : )
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03:43.24GamercjmOh ive never looked at the CDR, looked at the voip records
03:43.26Gamercjmlet me see
03:44.05kshumard_homelooks like it should be there
03:45.43Gamercjmhmm
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03:46.41Gamercjmmy time seems to be off, or cdr arent recording
03:46.54Gamercjmand doesnt show my recent calls
03:47.40Gamercjmi have to go, bbl
03:48.01[TK]D-FenderSedorox : Voipsupply has a new GSM -> SIP for around 300$ now
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03:48.25Sedorox[TK]D-Fender: yea.. but I would have more of a want for sip -> gsm...
03:50.07[TK]D-FenderSedorox : may go both ways.. not sure.
03:50.44Sedoroxhmmm
03:50.48Sedoroxdoubtful for that cheap...
03:50.53Sedoroxbut I'll check it out
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03:51.36FuriousGeorgeanyone know why the volume on voicemail messages left fluctuates to nearly inaudible levels?
03:52.23FuriousGeorgeand do you think it will work right when i upgrade to 1.2
03:52.25FuriousGeorge.1
03:54.10Sedorox[TK]D-Fender: yea.. they are only to use cell service on a normal phone or into a pbx
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03:55.31gigabitfxanyone have asttapi working?
03:55.32kshumard_homeFuriousGeorge, that's a weird issue that's been around forever
03:55.42kshumard_homeFuriousGeorge, they added a gain adjustment to app_voicemail to counter it
03:55.52kshumard_homeshow application voicemail, the g(#) option
03:56.08kshumard_homeI haven't tried it but I read the commit logs so I've heard all about it. : )
03:56.36FuriousGeorgekshumard_home: thanks again!
03:56.36kshumard_homeI think that option isn't in the 1.0 stable branch.... pretty sure you have to upgrade to 1.2 branch to use it
03:56.46FuriousGeorgei just did anyway
03:56.56gigabitfxim getting a channel.c: Unable to request channel sip/2001
03:57.39kshumard_homegigabitfx, what else does it say?
03:58.39gigabitfxwell, its loggin in fine, but this unknown channel hook is resulting in an unknown output.. holdon ill paste the one line
03:59.01gigabitfx(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2005-12-15 22:56:25','','','s','default', '**Unkown**','','','',0,0,'FAILED',3,'','1134705385.110')
03:59.48kshumard_homehmm, what exactly is "Unkown"?  : )
03:59.57Sedoroxcid maybe?
04:00.08gigabitfxi think the unknown is cuased by not being able to request the sip channel..
04:00.23kshumard_homeyeah... amusing to me that there's that typo in channel.c though. : )
04:00.37Sedoroxchannel is **Unknown**
04:01.39Flautowhat do you guys use for give voip calls the most trafic?
04:01.49Flautoqueing? or somethign
04:01.50gigabitfxi just tried with cid, same error
04:01.52Flautowhat is it
04:03.13gigabitfxwhats the typo in the channel.c?  should it not have sip/ in it?
04:04.11gigabitfxwhen i do it without sip/ then the errors go away, but no call is placed
04:04.37kshumard_homewell the typo I pointed out is cosmetic, "unknown" is spelled as "unkown"
04:04.47Qwellkshumard: it's like that a bunch of places
04:05.04gigabitfxlol, your right :P
04:05.58Sedoroxwas checking on channels.c outsourced again? :p
04:07.29gigabitfxim really looking for a tapi driver, if theres another one you guys recommend let me know.  ive tried sip tapi and get a little further with that
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04:22.51UberbotHi all.
04:24.01UberbotCan someone help me with this line in extensions.conf?   exten => s,2, gotoif( $[${RESULT} = 0] ? 10:3)
04:24.19UberbotI jumps to #3 no matter what the value of RESULT is.
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04:27.56[TK]D-FenderUberbot : remove the spaces around the "="
04:28.35UberbotI'll give that a try.
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04:39.38UberbotThat didn't fix it.
04:40.05Uberbotexten => s,2, gotoif($[${RESULT}=0] ? 10:3)
04:40.06UberbotAlways jumps to #3.
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04:43.16QwellDon't you need spaces?
04:43.22Qwellor was that requirement removed?
04:43.57SkramXyo all
04:44.51UberbotWhere do I need the spaces?
04:45.00QwellYou used to need spaces around the =
04:45.07Qwellbut, that may have been "fixed"
04:45.19QwellHave you tried to NoOp ${RESULT} to see what it is?
04:45.27UberbotYes.
04:45.41Qwellhow are you setting it?
04:45.47UberbotI've set it to 0 and 1 for testing.  The result is the same, no pun intended.
04:45.55UberbotI've got an agi script that calls a sql database.
04:46.05QwellIt doesn't have quotes around it something silly?
04:46.15UberbotNo.  It's an int.
04:46.23[TK]D-FenderUberbot : Pastebin the whole section of your extensions.conf...
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04:48.13Uberbothttp://pastebin.com/466261
04:50.21Corydon76-homeUberbot: you can have spaces in the expression, but nowhere else
04:50.46[TK]D-FenderAlso it'd be nice to see where RESULT gets set..... (assuming that doesn't solve it
04:51.05Corydon76-homeNo space before or after the question mark
04:51.41Uberbotexten => s,2, gotoif($[${RESULT}=0]?10:3)
04:51.49UberbotThe agi script sets RESULT.
04:52.00Corydon76-homeexactly
04:53.25Corydon76-homeand technically, there shouldn't be a space before the gotoif, either
04:53.47Corydon76-homeWe may have fixed that, however
04:54.45Corydon76-homeYou may also want to check out bug # 5055... it eliminates the need to call an AGI for a simple SQL query
04:54.53[TK]D-FenderAnd we'd need to see if there's a bug in your agi setting RESULT.
04:56.39UberbotIt sets RESULT, correctly.  I've noop'ed it in the past.  I'm still looking at my log.  I'll look at that bug, though.
04:56.41UberbotThanx.
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04:57.32AgiNamuDoes anyone know of any NON-iaxclient based phones that we can control programmatically?
04:57.59AgiNamuWe're getting one way audio with IAX2... I thought that was impossible, but iaxclient does it ...
05:01.01Corydon76-homeNothing is impossible with the right firewall rule
05:01.42Corydon76-homeYou aren't sending through a tunnel, perhaps, are you?
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05:07.09AgiNamunope. clear thru.
05:07.16AgiNamuno nat, we're on a corporate network.
05:07.21AgiNamuthe packets are getting to the iaxclient
05:07.38AgiNamuiaxclient is simply not passing them to the audio device. every iaxclient based phone does this.
05:07.49UberbotFixed it.  Thank you all.
05:07.50AgiNamuif it doesnt say "call 0 progress" before we answer
05:08.11AgiNamuthen we cannot hear audio on the iaxclient. iaxclient still transmits just fine.
05:08.15Corydon76-homeUberbot: what was the problem?
05:08.25UberbotSpacing, it would seem.
05:08.31UberbotI
05:08.46UberbotI'll use that one as a template and fix a few other features.... <grin>
05:08.51dokhenchCorydon76-home, you dropping that te110p off in the morning? :)
05:08.58Corydon76-homedokhench: yep
05:09.13dokhench:D
05:09.15AgiNamuso im desperate to find any non iaxclient phone. FireFly (now cubix) works like a charm
05:09.45Corydon76-homeAgiNamu: how about an IAXy?
05:10.05AgiNamuwe want to control it from our PC software
05:10.15AgiNamuI have some IAX hardphones i could use
05:10.19AgiNamubut i want soft :(
05:10.48Corydon76-homeMight it be something simple, like some other program has locked the audio device?
05:10.53Qwelluse a cisco 7960, and control them with xml executes :p
05:11.11AgiNamuI wish... it also doesnt handle the hangup message either
05:11.22AgiNamuits like it somehow gets involved in sending audio, but doesnt figure out that its also receiving.
05:11.35AgiNamuWe've spent several days on it so far
05:11.41AgiNamuethereal shows the streem looks normal
05:11.54Corydon76-homePerhaps a PC-based firewall?
05:12.08AgiNamunope, it is intermittent
05:12.28AgiNamuill place a call, works fine. hangup try again, doesnt work. a few tries later, it works. It has to do with Iaxcleint thinking there's progress
05:12.33AgiNamuwell, thats what it seems like from the output
05:12.41AgiNamu"call 0 progress" or some message like that.
05:12.51AgiNamuif i get that, im ok, and there will be audio. if i dont... im hosed.
05:12.53Corydon76-homeYou're going to have to try the iaxclient mailing list
05:13.11AgiNamuI'm willing to pay
05:13.28Corydon76-homeWell, first find someone who actually knows that codebase
05:13.33AgiNamuyea :)
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05:24.15tessierI can't believe I put a Cisco ATA-186 on ebay and it's only going for $10.
05:24.21tessierThis might have been a mistake.
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05:40.34ptiggerdineanyone tried compile chan_misdn form asterisk-1.2?
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05:44.35ptiggerdinei get this error msg #error including kernel header in userspace; use the glibc headers instead!
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05:46.06AyanoAnyone here had a polycom hang during the boot and just go to a blank screen?
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05:53.36Gamercjmhmm is nufone.net down? not able to login and cant recieve calls
06:00.15ptiggerdinewhat's will mISDN support in asterisk???
06:00.22ptiggerdineit's all badness
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06:04.29ptiggerdinekernel headers in user space it bad...
06:04.41ncconquerdo anyone know how to setup asterisk for login to call form another network
06:04.54ncconqueri'm using ADSL and setting NAT
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06:14.24litagedoes sip split signalling and voice into 2 separate streams?
06:14.38Qwelllitage: yes
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06:24.29tonyshi, I may be a frequent question, but anybody knows if voipbuster is working with asterisk now?
06:25.33trixterdoes anyone have any problems when using rfc2833 and *not* ilbc that asterisk 1.2 complains that its an invalid ilbc frame and doesnt process dtmf?
06:27.58Ayanoanyone use polycoms here?
06:31.22FuriousGeorgeso if i set nat=route, is that more or less like making the user "roamable"
06:32.11FuriousGeorgedoes * just kinda "figure it out"
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06:42.20bsdfreakheh
06:42.55dokhenchAyano: yes
06:43.54AyanoI have ip300s, and they wont boot up
06:44.40UberbotAny answer about if boipbuster worked with Asterisk?
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07:13.38junbugany 800/8xx providers that use iax?
07:13.48junbugoutbound that is
07:14.46Qwelljunbug: nufone and asterlink.  I don't know if asterlink will be keeping iax though...ask them in #asterlink
07:16.11junbugusing voipjet for outbound,  and iam getting charged for toll free calls ...
07:16.21Qwellnice
07:20.34QwellMr. file !
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07:20.44QwellI'll bet he's sleeping...slacker
07:22.48trixterjunbug any 800/8xx providers that use iax?   ---  www.goiax.com
07:23.25trixterUberbot Any answer about if boipbuster worked with Asterisk?  -- yes it does but you have to sign up using their windows client
07:24.39FuriousGeorgehey qwell, you use asterlink?
07:24.42QwellFuriousGeorge: yeah
07:24.59Qwelltrying to figure out how to call Oz, heh
07:25.08FuriousGeorgeerrr, i just signed up, where are the directions :)
07:25.16Qwellthey email them to you
07:25.22FuriousGeorgeah
07:25.33Qwellugh...having trouble hitting arishost
07:25.44QwellAre you able to login to the site?
07:26.14FuriousGeorgearishost?
07:26.28FuriousGeorgeperhaps the email will come now that i have purchaced minutes
07:26.45QwellFuriousGeorge: dunno
07:26.49Qwellcogent.arishost.com
07:27.02FuriousGeorgehey Qwell, lemme axe you something now that i got your attention
07:27.14Qwell?
07:27.26FuriousGeorgeif i set nat=roam will * just "figure it out"
07:27.30Qwelldunno
07:27.45FuriousGeorgei mean "route"
07:27.55FuriousGeorgenat=route
07:28.22FuriousGeorgeis that like a "yes and no" or "depends"
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07:43.28nswintHello is it possible to for instance with a TDM400P to check each channel to make sure that it is not in use before dialing... if so can it route to another channel if one is in use?
07:43.51Qwellnswint: chanisavail, or use groups in zap
07:44.52nswintso for like priority 1.. check to see if channel is available then 2 dial
07:45.13Qwellsomething like that.  groups would be better though, imo
07:46.08nswintexplain groups.. not keen on that
07:46.31QwellIt automatically picks an unused channel
07:50.05nswintok.. Ill look into it.. I'm  going to be ambitious.. my sister and brother in law are moving in this summer. I need to add an extra line.. and make use of my skypebox with using skype credits.. and my callndock when the others aren't available
07:50.25nswintcallndock for my cell
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07:51.31TelamonIs there any way to change what is prepended to the name of a macro?  IE, instead of macro-<name> use macro_<name>
07:51.45*** part/#asterisk azfhasterisk (n=rickb@VDSL-130-13-192-106.PHNX.QWEST.NET)
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07:56.38QwellTelamon: change the source
07:58.53TelamonQwell: Yeah, that's one option I suppose.  I was hoping there might be a simpler method, ie, macroprefix=macro_. :)
08:00.16RaYmAn-BxTelamon: for the majority of people it doesn't really matter, so presumably that's why no one ever bothered to add that :) (why do you want it changed anyways?)
08:01.01nswintQwell are you using asterisk in the home or in the corporate env?
08:01.08Qwellnswint: both
08:01.20RaYmAn-BxTelamon: do you have contexts that are called macro-* which aren't macros or something? :>
08:02.22TelamonRaYmAn-Bx: Our web frontend (which I didn't write) uses a database (Realtime extensions) and prefixes all contexts with the customer name.  IE, company_dial_macro  So I can't create a macro labelled macro-_dial_macro unless I hard code it into extensions.conf, which I'd prefer not to have to do.
08:03.06TelamonRaYmAn-Bx: I'd like to be able to set it up so I can define a macro prefix and a macro suffix in the config files, but I guess I'll have to edit the source code to get that feature.
08:03.19Telamonprefix="" suffix="-macro" :)
08:03.23RaYmAn-Bxah, makes sense, but yeah, most likely needs source editing
08:04.06RaYmAn-Bxwould probably be easier to get whoever made the web frontend to change it so macro's are called macro-company_dial_macro etc :P
08:04.27TelamonRaYmAn-Bx: Yeah, sending that E-Mail now. :)
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08:07.22tessierAnyone know how much it normally costs to send an international SMS message?
08:10.31tessierI am mainly thinking of sending SMS messages between the US and Mexico.
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08:23.22nswintQwell: I've checked out the groups and I have a question...  I'm checking out automated.it  and they have a dialplan for group=1 pickupgroup=1-4 then below it has mailboxes specified and channels.. why is that
08:23.54Qwellbecause zap channels can have voicemail boxes
08:25.09nswintso basically it's assigning a phone/mailbox to a group and a channel at the same time
08:25.19*** join/#asterisk Newbie___ (n=me@211.24.146.11)
08:25.39*** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee)
08:27.11nswintwould you have to assign for each group which pickupgroup it can go to?
08:27.58Qwellgroups are generally used for outbound calls
08:28.19Qwellbed time
08:28.29nswintany places in particular I can read up on groups
08:28.41Qwell~wikis
08:28.43jbotsomebody said wikis was http://www.voip-info.org
08:28.53Newbie___hi guys, is there a way to make DISA dial faster?
08:28.55nswintgrazzie
08:29.45Newbie___when i call into * box with DISA, it takes approx 14 sec for DISA to accept the digits and then dial
08:30.17QwellNewbie___: press pound, or set the digit timeout lower
08:30.37Qwellor...response timeout?  dunno
08:30.48Newbie___Qwell: i did press #, * just ignore, wait for 10-14 sec and then start dialling
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08:30.49QwellI think the latter
08:31.05Newbie___hrm. digit time out
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08:34.06tessierHmm...I can do email to SMS usually by doing number@phoneprovider.com but how to do SMS to email?
08:38.07Newbie___hmmm, DigitTimeout and Response time out does not work either
08:39.57wasimtessier: kannel
08:41.17tessierwasim: Verrrry interesting. Thanks!
08:45.01rzaphoneprovider?
08:45.11rzaso i can do mynumber@nokia.com?
08:45.17rza:P
08:45.56tessiersomething like that probably
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08:46.20rzaisnt it more like @serviceprovider.com
08:46.48tessierYour service provider usually provides you the phone too. They don't manufacture it though.
08:47.00tessierStop being pedantic. :P
08:47.18rzano it doesnt
08:47.32rzain my country service provider doesnt provide phone
08:47.39rzain many countries its that way
08:47.47tessierAll the world is the US. If your country isn't yet, it will be soon.
08:48.02tessierI suggest you invest in a bomb shelter because I'm gonna vote for Jeb Bush next time around.
08:48.04rzait wont actually
08:48.08rzaits forbidden by law here
08:48.27rzato sell phone with service
08:48.49tessierWe'll rewrite your constitution, don't worry.
08:49.13tessierYou'll be sucking down BigMacs, drinking CocaCola, and getting service packaged with your cell phones in due time.
08:49.30rzai live in the homeland of nokia
08:49.47tessierFinland? It'll be easy then.
08:50.01rzahehe
08:50.14tessierI'm sure Qualcomm is making a generous campaign donation to ensure the Nokia HQ is first on the target list.
08:50.22razuno - kia
08:50.23razu:)
08:53.04junbughmm are there external fxo's->sip products
08:53.19tessierjunbug: yep
08:53.29tessierjunbug: I recommend them over digium fxo cards
08:53.41tessierI used a Mediatrix but it was a bit of a pain to configure with their silly SNMP app
08:53.54tessierBut once I got it configured it has worked for a year and a half without anyone touching it
08:55.17junbugtessier: so thats the only company right now?
08:55.37tessieroh, no. There are several.
08:55.39tessierCisco is the most popular.
08:55.52tessierFor about the same price as a digium card or mediatrix box you can get a cisco with an fxo port or two
08:56.02tessierIn fact that might be my recommended solution.
08:56.14junbugahh googling now
08:56.16tessierFXO-Cisco-SIP-Asterisk/SER/whatever
08:56.40tessierI'm nowhere near the guru status of some people here but over the last couple of years I've gotten a shitload of experience.
08:57.04tessierThere's a saying: Good judgement comes from experience. Experience comes from bad judgement.
08:57.11Ikarustessier: Cisco is a good choice, I agree
08:57.14tessierAnd like I said, I've gotten a lot of experience over the last couple of years.
08:57.37IkarusIf you don't want to muck around with FXO's in PC's, bad drivers, etc it is prolly the best
08:57.41tessierYep.
08:57.59tessierJust too many variables when you put FXO in a PC.
08:58.12junbugtessier: how much was the 4 port mediatrix?
08:58.19tessierjunbug: Around $500 or so.
08:58.43Ikarustessier: shame that Cisco sucks for SIP <-> BRI connectivity
08:58.59tessierIkarus: BRI? Like ISDN or something?
08:59.35Ikarustessier: yes
08:59.52tessierI've never run into that. But here in the US hardly anyone uses ISDN anyhow.
09:00.06Ikarustessier: here all small buisnesses use BRI and larger ones PRI
09:00.12Ikarusit is so awfully cheap
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09:00.17tessierWhere is here?
09:00.34tessierBRI is very rare and PRI is very expensive here.
09:00.34IkarusThe Netherlands, but this goes for Europe in general
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09:00.49tessierYeah. I've heard similar from others.
09:00.54tessierHow much is a PRI there?
09:00.58sbingneractually, I think it goes for everyplace but the US and canada
09:01.00hd420I just downloaded/installed the asterisk on my mac... I can't seem to find out where it is located though
09:01.03hd420PRI?
09:01.28sbingnerbut ISDN here isn't too bad either if you can get somebody to understand that you REALLY do want a VOICE ISDN line
09:01.36Ikarustessier: a full E1 (30 lines) is 200 euro
09:01.36tessierprimary rate interface. Funny thing about telco acronyms is that you can be told what the acronym stands for and still have no clue what it is.
09:01.43sbingnerthey all get really confised and try to send you to the wrong place
09:01.45Ikarusa month
09:01.57tessierIkarus: That is about 1/2 or 1/3 what it costs here.
09:02.52Ikarustessier: you can get it even cheaper, a 30 line E1 connection which is turned into VoDSL and VoATM is only 80 euro/month
09:02.59wasimpri are 2/3 the cost of pots here
09:03.05hd420I just downloaded/installed the asterisk on my mac... I can't seem to find out where it is located though... anyone care to point me to the root of the asterisk tree?
09:03.33IkarusYou can't even get more then about 15 POTS lines here to a single site, or well, you can't afford it
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09:06.35hd420this is shitty and stupid
09:07.12IkarusPOTS lines btw are 18 euro a line
09:07.15Ikarusyou do the math :)
09:07.33tessierThey are around half that here. Bizarre.
09:07.47hd420Ikarus: uhh... you can get VoIP in Eire for €10/month
09:08.25Ikarushd420: VoDSL and VoATM are much more reliable then VoIP
09:08.28tessierIs an E-1 for Internet similarly inexpensive?
09:08.44hd420of course, your broadband connection will run you €40/month
09:08.48drrayat 18 pots you are better off getting an E1
09:08.54hd420probably higher
09:09.01Ikarustessier: it is alot more expensive, but most people choose SDSL, which costs for 2.3Mb/s symmetric 220 euro
09:09.24tessierWow
09:09.30hd420Ikarus: in my area, in the US, there is no DSL available, only cable
09:09.38IkarusSDSL is the bastard grandchild of E1 anyway
09:09.45hd420so no one runs OS X here
09:09.47hd420?
09:09.51tessierhd420: No
09:09.56tessierhd420: Mostly Linux
09:10.18hd420tessier: I could run Linux for asterisk, but my laptop is a Mac
09:10.29tessierSo get a Linux box
09:10.37tessierWhy would you want to run asterisk on a laptop?
09:10.50hd420tessier: so it would roam with me?
09:10.53junbugtessier: dlink has one also  DVG-3004S
09:11.02hd420like skype does
09:11.15hd420or the gizmo project
09:11.33hd420the latter not being available on linux (yet)
09:11.43{zombie}hd420: I think you misunderstand what asterisk is
09:11.47{zombie}it's not a softphone
09:11.52{zombie}it's a VoIP server
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09:12.14hd420{zombie}: i suspect I do
09:12.23*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
09:12.28*** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
09:12.31hd420where's the daemon then?
09:12.32{zombie}asterisk is the VoIP server that you plug IP phones or softphone into (amongst other things you can do with it)
09:12.35DandreHello all,
09:12.45hd420{zombie}: what's the daemon called?
09:12.52{zombie}which daemon?
09:13.17hd420any one? i just want to find out where asterisk decided to install itself on this system
09:13.20DandreIs there any possibility, on a blind transfer, to take back the call on busy signal?
09:13.37wasimhd420: asterisk
09:13.59Dandrehd420: locate asterisk
09:14.08Dandreor which asterisk
09:14.13TheCopsor whereis
09:14.31TheCopsit search directly on your $PATH var
09:15.11hd420locate only works with the up to date databases, and which/whereis work only if your path has been refreshed, the proper way to find stuff is to use find(1)
09:15.21TheCopshd420, when installing asteris, you will have /usr/include/asterisk /usr/lib/asterisk /var/lib/asterisk /etc/asterisk and /var/spool/asterisk
09:16.56hd420I think I've found it
09:17.00hd420thanks for your help
09:24.06DandreIs there anyone you should help me with my blind transfer question?
09:24.18DandreIs there any possibility, on a blind transfer, to take back the call on busy signal?
09:28.02{zombie}Dandre: http://www.voip-info.org/wiki/view/BLINDTRANSFER
09:28.09*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
09:28.44{zombie}you could use that to build your dialplan so that at priority +101 (busy) it goes back to the extension in that variable
09:30.08Dandrenice tip! is this variable available for 1.0.x or only for 1.2?
09:31.56pifhi, how do I convert from passwd to shadow ?
09:32.00pifsorry
09:32.22wasimpwck
09:32.52TheCopsthis is pwconf, no ?
09:32.54jahanihow to secure asterisk ?
09:33.03wasimjahani: pull the network cable
09:33.13jahanilol
09:33.15TheCopswasim lol
09:34.08hd420is Asterisk akin to SkypeIN?
09:34.32hd420wasim speaks true, that would secure your connection
09:39.11razuit's true ... most secure case is without power and ethernet
09:40.35drraymost secure is to dig a hole, put the server in the ground and put a marine batalion on top of it.
09:45.08bsdfreakheh.
09:45.43bsdfreakactually secure is to take your server, compact it with a trash compactor, then run it over with a steam roller, then incinerate it in to dust
09:45.49Ikarusdrray: most secure is to remove the server from existence
09:45.49bsdfreak;p
09:45.55bsdfreakyup
09:46.00Ikarusbsdfreak: that is not a NSA approved methode
09:46.08bsdfreakikarus: lol.
09:46.20bsdfreakikarus: yes yes, i know they have their own way.
09:46.42Ikarusbsdfreak: I once helped out in a hospital where they had to destroy sensitive data (film, paper and CD's)
09:46.49IkarusThat was alot of fun
09:46.52bsdfreak=]
09:46.59IkarusIncineration, acid baths, etc
09:47.22Ikarus(the paper was acid treated, strips the ink right off and makes the fibers fall apart)
09:48.21bsdfreaknice
09:48.40bsdfreakmy gf's work which destroys top secret docs regularly uses a shredder that turns the paper in to dust
09:48.43bsdfreakit's pretty impressive
09:49.38IkarusIt is great stuff
09:49.44Ikarusbut destroying modern HDD's is a pain
09:50.58bsdfreakyeah heh
09:51.00IkarusBleah, zoning out at work
09:57.34*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
09:58.00nfi|ermeshi all
09:58.02nfi|ermesi have an hfc isdn card
09:58.04nfi|ermesand i use zaphfc
09:58.17nfi|ermeshow can i set my msn in the card ?
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10:15.23bsdfreakheh
10:17.05IkarusUrgh, ordered what I hope will be HFC-S cards
10:17.22IkarusElse the $boss will grill me (if she where to actually find out)
10:17.24tessierI would like to have a quality paper shredder
10:17.31tessierI only get cheap ones and they end up busting
10:17.39tessierBut good ones are quite expensive.
10:17.43Ikarustessier: don't shred CD-roms
10:17.46tessierSeems like shredding paper should be easy.
10:18.00tessierIf you can fit it into the slot the machine should be able to shred it.
10:18.03tessierI don't shred cd-roms.
10:20.00IkarusI wish buying single port HFC-S cards was a bit easier
10:21.16wasimnuke cd-roms
10:21.30*** join/#asterisk davidw (n=davidw@apache/committer/davidw)
10:21.45davidwhi - want to confirm something: hi-res timing is provided by ztdummy module?
10:21.53wasimdavidw: yes
10:22.05wasimdavidw: if you can define 1ms as high res
10:22.14*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
10:22.49davidwwell... I mean if you don't have a zaptel device, you should install ztdummy ?
10:23.02*** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
10:23.06IkarusIf you need timing for something, yes
10:23.06davidwor is it not worth bothering with?
10:27.08wasimis there a difference between microsoft gsm 06.10 and the asterisk 06.10 codecs?
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10:46.14astcryzUnable to find key '71398553' in family 'SIP/Registry'
10:46.19astcryzwhat does this mean?
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10:55.22jiro5281hi everyone..would like to ask help how to pass variables between context?
10:55.29jiro5281any advice?
10:55.40*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:55.46psirachello ! i've an Granstream ATA HT286, it seems to be good to call and receive call but i can't get ringing phone on incoming call
10:56.23psiracsomebody know what's the on hook Voltage for France (Europe) ?
11:03.50*** join/#asterisk zobia (n=laura_sh@218.6.242.212)
11:04.00zobiaHello every one.
11:06.50zobiaI have a question. i make a call through the asterisk to a IVR system. but if i press # key. it turn out to come back the asterisk box's "transfer-pbx"
11:10.50darkskiezso is there a chan_jingle yet ?
11:11.28coppicechan_jiggle? :-\
11:12.19darkskiezcoppice: http://code.google.com/apis/talk/index.html
11:13.10darkskiezzobia: edit features.conf and change transfer to something else, i set mine to ##
11:22.08fenlanderdarkskiez: I was about to ask that :-)
11:22.33fenlanderdarkskiez: very topical name for the time of year
11:23.01darkskiezindeed
11:25.29*** join/#asterisk zotz (n=zotz@24.231.47.168)
11:30.21fenlanderI guess it's time to get over my inherent distrust of C++ and hack a chan_jingle together using libjingle..
11:31.19fenlandermaybe this is one for woomera
11:37.49Ikarusfenlander: why on earth use Jingle at all
11:49.56*** join/#asterisk zoa (n=kkk@pirus.securax.be)
11:50.49darkskiezikarus: to take calls from googletalkers ?
11:51.12Ikarusdarkskiez: do we really want to ?
11:51.26*** join/#asterisk bangawanga (n=ahecker@ppp-82-135-3-44.mnet-online.de)
11:51.35bangawangahello
11:52.02bangawangacan anyone give me an example for chanspy app?
11:52.14darkskiezIkarus: you dont have to want to, but other people do and will want to
11:52.39IkarusWhich is why I had to board up the Skype ports here....
11:52.42Ikarusabusive protocols
11:53.30darkskiezI get asked at least once a week by colleages at work why skype sounds better than asterisk calls.
11:53.55Ikarus16khz vs 8khz, big bug in Asterisk really
11:54.17fenlanderdarkskiez, Ikarus: I'm often asked if there is a way to integrate skype into an office pbx
11:54.28darkskiezsame here
11:55.11Ikarusfenlander: the protocol is just waaaaaay too bad for your network to allow on your lan
11:55.15*** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
11:55.19lmehi guys !
11:55.27IkarusI actually have a few slides to demonstrate how it works and why it is bad
11:55.42darkskiezIkarus: are they published on the electronical interporn ?
11:55.45fenlanderIkarus: yeah, we don't like how it works, but it does and it is *very* popular
11:56.00*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
11:56.04Ikarusdarkskiez: not currently
11:56.17darkskiezI had to allow access to it after the CEO wanted to talk to another CEO with it
11:56.28Ikarusfenlander: then we need to figure out how to be compaitible without becoming a relay
11:56.50Ikarusdarkskiez: I am responsible for the network, if I allow something that endangers it, I am the guy who gets the problems
11:57.17IkarusBut I would really like * supporting 16khz audio
11:57.23fenlanderIkarus: that is unlikely to be acceptable to skype - which is why I support googletalk as an alternative
11:57.43fenlanderthere is some 16khz work going on
11:57.55Ikarusfenlander: only passthrough from what I heard
11:58.10fenlanderi guess the problem is the codecs - and licences...
11:58.26fenlanderiirc google/skype licence from voiceage
11:58.27zoafenlander: nopez
11:58.30zoathats not the problem
11:58.36zoaspeex does wideband for example
11:58.50fenlanderzoa: true - but does skype or googletalk do speex?
11:58.58zoathey use ilbc
11:59.21Ikarusfenlander: I don't care about being compaitible, I just wan the same quality in IP to IP calls
11:59.59fenlanderIkarus: you also need the phones to support it - are there any good IP phones that do 16khz?
12:00.22zoafenlander: idefisk will have it in a month probably
12:00.27Ikarusfenlander: well, my non-good BudgeTone supports it
12:00.45fenlanderIkarus: which codec?
12:01.34fenlanderzoa: that will be great - I guess it will be just speex and not wideband ilbc?
12:01.39cypromisskype uses a gips codec
12:01.45cypromisbut its not the normal ilbc stuff
12:02.05bangawangacan anybody tell me how to use the chanspy app??
12:03.11Ikarusfenlander: G.722 atleast
12:03.30*** join/#asterisk Hmmmm (n=Hmmmm@221.135.51.19)
12:04.22lmebangawanga: have to configure an extension to use it. and to specify at least the beginning of a chan
12:04.42lmebangawanga: actually I use it this way : exten => _89,2,ChanSpy(ZAP)
12:04.53fenlanderIkarus: cool
12:04.58IkarusProlly some other codecs aswell, didn't do that much research as I don't need Asterisk in passthrough mode, so I can't use it anyway
12:05.09lmebangawanga: when i call the 89, it's propose me to swap between all the actives zap's chans
12:08.50RoyKzoa: ping
12:09.24IkarusiLBC is also supported it seems
12:16.05IkarusSomeone should make an Asterisk fork purely to implement wideband support
12:16.16IkarusSo that it doesn't get interference from "normal" development
12:17.04darkskiez*cough* openpbx :)
12:17.42lme-> []
12:17.57Ikarusdarkskiez: unlikely
12:19.03Ikarusdarkskiez: I need the basic functionality of Asterisk (mainly BRI support), which isn't availible in OpenPBX, which means I might be able to develop on openpbx, but I would never truely use it for other purposes then test calls
12:22.23fenlanderhmm, okay, so wideband ilbc is actually not ilbc but iSAC, which is proprietary - wideband ilbc doesn't exist
12:22.32Ikarusfenlander: ah, k
12:22.43Ikarusah well, G.722 seems to be widely availible
12:22.48Ikarusno idea about licensing
12:24.12fenlanderspeex.org lists the G.722 licence as '?'
12:24.54Ikaruslet's see
12:27.34IkarusI wonder if it is actually patentable in Europe the algorithm (some are, some aren't depends on how the patent was requested)
12:27.59*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
12:28.14IkarusI could care less about the USA
12:28.27Ikaruserm, that statement is a bit flawed
12:28.36Ikarusanyway, the USA is not the market I have to deal with
12:28.43bangawangalme: i created something like this: exten => 1004,1,Answer() exten => 1004,2,Chanspy(SIP)
12:28.53bangawangabut i cant hear anything on 1004
12:29.00fenlanderhmm
12:29.24RoyKzoa: ping!
12:29.25Ikarusif anyone owns the IP on it, it is AT&T
12:29.29RoyK~ping zoa
12:29.31jbotpong zoa
12:29.36RoyK~lart zoa
12:29.45Ikarusand possibly voiceage, but that is only on a subset
12:30.09fenlanderG.722.1 has a 'Royalty-bearing licnece' as it uses Siren7 which is patented by Polycom
12:30.28RoyKfenlander: wtf is siern7?
12:30.33fenlanderhmm. confused
12:30.48RoyKfenlander: just an algorithm, or would you think the patent's valid in EU as well?
12:31.05Ikarusfenlander: if it is purely an algorithm license it is not valid in the EU
12:31.20fenlanderI don't know - but I assume G.722.1 != G.722
12:31.29RoyKprolly
12:31.54IkarusG.722 is the wideband codec, .1 is the lowerbandwidth version
12:32.00Ikarus.2 is even lower
12:32.01RoyKIkarus: yeah. but like g.729 and g.723.1 it might hold 'methods' that are unique/new that indeed are patentable in EU as well
12:32.26bangawangalme: did you get me?
12:33.14IkarusRoyK: according to my info G.729 can't be validly patented in Europe, atleast not for software implementations
12:33.23*** join/#asterisk bmg505 (n=leon@dsl-146-38-142.telkomadsl.co.za)
12:34.51IkarusOfcourse for most people not giving a fuck about the USA is not an option so such things are rarely researched unless it is a really important algorithm (encryption and compression have happend in the past)
12:34.58RoyKIkarus: for what I can see, it can indeed. i've been talking to people with quite some law knowledge in .no and it looks like it's quite valid
12:35.12RoyKbut bbl
12:35.40IkarusNorway != EU
12:42.53bangawangalme: do you hanve an explamation why i do not hear any sound while spying??
12:43.24cj-rmI'm running * on a redhat 9 box with a 2.4.20-9 kernel.  If I want to install a digium tdm400 will the zaptel drivers require me to remcompile the kernel??
12:44.39*** join/#asterisk apardo (n=apardo@251.Red-83-46-189.dynamicIP.rima-tde.net)
12:47.29TonyMcj-rm: you shouldn't have to recompile the kernel, but you will need the kernel-source rpm installed
12:55.44*** join/#asterisk bangawanga (n=ahecker@ppp-62-245-163-47.mnet-online.de)
12:56.04bangawangahello all!
12:56.20*** join/#asterisk mcnobody (n=laaksola@laaksola.net)
12:56.44bangawangai am back with my question about chanspy and no sound, any ideas?
12:57.28wasimbangawanga: make sure you're listening to bridged channels and the ilke
13:00.34bangawangaok - beginners error, i always tried to listen between 2 interal phones - external works!
13:05.54*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
13:06.24asteriskmonkeyanyone know much about nortel systems and old school pbx's?
13:07.24rzanortel meridian 1?
13:07.28asteriskmonkeyyes
13:07.38asteriskmonkeywanted to ask if there usually just regular pots lines going in
13:07.46rzano idea
13:07.52asteriskmonkeywas going to retrofit a system with iaxys=>oldschool pbx
13:07.52rzai know something about ericsson md110
13:08.02rzaand something about nortel meridian 1
13:08.07rzabut from programming side
13:08.24asteriskmonkeyah ok
13:08.32asteriskmonkeyany of the digium guys on yet?
13:10.15asteriskmonkeyspooky quite in here
13:10.20zoahey ho:P
13:10.23zoa(im not digium)
13:10.56asteriskmonkeymorning
13:11.11znoGanyone know why the exten => h,1,...    don't work in macros?
13:11.47znoGi'm trying to do "something" when a call ends
13:12.24kaldemarput the h extension in the context you started the number analysis in.
13:12.25zoait depends on which ends ends the call
13:12.51znoGkaldemar: that's the problem, I want to set the hangup sequence only if a certain number is dialed
13:13.00*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:13.08znoGso, I have: exten => specificnumber,1,Macro(something,${EXTEN})
13:13.14znoGand inside macro-something I set what to do when they hangup
13:13.23znoGbut, it is ignored by asterisk
13:14.14kaldemarwell, you could set the called number to a variable and use a gotoif in the h extension to check if you want to perform the sequence. :P
13:15.26*** join/#asterisk bmg505 (n=leon@dsl-146-36-223.telkomadsl.co.za)
13:15.35znoGhaha, didn't think of that one.. ugly but it'd work
13:17.02kaldemari'd like to see a universal hangup option in the dial command. now you can execute a macro only if the remote side hangs up.
13:19.45*** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
13:22.34*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:22.51[TK]D-FenderOMFG over 20cm of snow since 3am ;(
13:23.41Sebbbbbjust rain here in germany ;)
13:24.01*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
13:24.01*** join/#asterisk outofjungle (n=outofjun@59.144.9.205)
13:24.06PakiPenguinevening guys
13:24.49asteriskmonkeymorning here :) evening to you though :)
13:24.59PakiPenguin:) yeah
13:25.16asteriskmonkeydude toronto is so clean of snow right now .. they said there was to be alot yet all the streets are sqeeky clean
13:25.31znoGkaldemar: that trick worked, not so ugly afterall :)
13:25.57*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:26.17asteriskmonkeyoh i got a good question... do you have to upgrade your cdr stuff when you are upgrading from 1.0.9 to 1.2.1 or can you just upgrade your zaptel/asterisk and be good :D
13:26.22newlSpeaking of Toronto, did the Leafs win their most recent game?
13:26.51kaldemarznoG: working tricks always look prettier that they are. ;)
13:27.06znoGkaldemar: true, true..
13:28.13newlhah unlucky you
13:28.17asteriskmonkey;(
13:28.28PakiPenguin:)
13:28.30newl401 is good in the summer. :)
13:28.42mutilatordoh he's not here
13:28.54asteriskmonkeyoh yes how enjoyable to go 2k/hr in rush hr with no air conditioning... how i miss that :)
13:29.29newlhaha last time I was on it, 130 was the cruising speed. :)
13:29.29*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
13:29.38asteriskmonkeymaybe the 401 wont be as bad now they have those hov lanes
13:29.46asteriskmonkeydamn you where lucks
13:30.15puzzledmorning
13:30.24[TK]D-Fender401 sucks at the BEST of times....
13:30.40asteriskmonkey:) there we go someone else local who knows the pain :)
13:30.56[TK]D-Fendersemi local :  Just have family in Mississauga :)
13:31.02[TK]D-Fenderandsurroundings
13:31.13PakiPenguincan anyone tell me , if this has been incorporated into 1.0.10 ? http://bugs.digium.com/view.php?id=4343
13:31.19[TK]D-FenderI'm in Montreal where we've got
13:31.28[TK]D-FenderUP TO HERE
13:31.40newlwhite stuff whee
13:31.44[TK]D-Fender&^%@# whit ^&%$
13:32.03[TK]D-FenderAnd imagine I'm only 10 minutes late to work!
13:32.07[TK]D-FenderI need a raise!
13:32.49*** join/#asterisk synthetiq (n=roger@64.201.13.50)
13:32.50newldon't we all? :)
13:33.02mutilatorwho would give a person late to work a raise..
13:33.20*** join/#asterisk pengyong (n=lala@218.93.147.154)
13:33.53lexlutherif the reason is compelling enough, like saving the world from another bin-laden attack
13:34.50fugitivosave the world from bush
13:35.02lexlutherthat too
13:35.04PakiPenguinhmm
13:35.46newlbloody bushes!
13:36.08mutilatorpfft
13:36.10mutilatorbe to work first
13:36.13mutilatorthen go save the world
13:38.24PakiPenguinis there any software i can use to simulate load / calls to asterisk?
13:39.30*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
13:40.05IkarusGrmbl, someone is glueing here, I am getting rather high
13:40.35*** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
13:40.47Dandrehello all,
13:43.06*** join/#asterisk azrishahril (n=azrishah@60.48.74.185)
13:44.51DandreI am trying to use on file to define my extension and then use this file in other config file. The first I am trying is queueu.conf:
13:44.52Dandrehttp://pastebin.ca/33886
13:45.02*** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee)
13:45.13DandreIs there someone you should help me?
13:45.57wasimPakiPenguin: /var/spool/asterisk/outgoing
13:45.58azrishahrilany one know how to trouble shoot Dec 16 19:25:52 WARNING[10514]: Ring requested on channel 0/19 already in use on span 2.  Hanging up owner.
13:48.04iCEBrkrWe've reactivated your
13:48.04iCEBrkr<PROTECTED>
13:48.04iCEBrkr<PROTECTED>
13:48.06[TK]D-FenderDandre : You can only do variables in extensions.conf <-
13:48.06iCEBrkroops
13:49.02[TK]D-FenderiCEBrkr : How'd day 1 turn our for your a104d?
13:49.06*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
13:49.07PakiPenguinhey wasim , adaab :p
13:49.23Ariel_morning everyone
13:49.27*** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226)
13:49.28wasimPakiPenguin: just use a perl script thingy to dump files in there
13:49.40Dandre[TK]D-Fender: so what # include may be used for? I don't really see
13:50.07*** join/#asterisk squall (n=squall@omaha.lnet.fr)
13:50.15iCEBrkr[TK]D-Fender: It's not.  I gotta figure out the signalling on our T1s
13:50.17PakiPenguinalright
13:50.27docelm0whadup?!!?
13:50.33iCEBrkrThey're not PRI.. and I can't find anything that makes sense as far as CAS
13:50.35iCEBrkrdocelm0: yoyoyo
13:51.09[TK]D-FenderiCEBrkr : No-one offering theservice in your area?
13:51.17docelm0Hay ice..  Where monday?
13:51.32iCEBrkrdocelm0: 600 1st Ave N.
13:51.52docelm0Got a zip?
13:51.58iCEBrkrdocelm0: 33702
13:52.11docelm0kewl..  time?
13:52.13docelm0:)
13:52.29*** join/#asterisk bmg505 (n=leon@dsl-146-57-202.telkomadsl.co.za)
13:52.34iCEBrkrTake 275 allll the way to 375, Get off at 5th Ave and turn right.  Or something
13:52.49docelm0thats fine..  But I still need a time..
13:52.56iCEBrkrdocelm0: I think we start about 7
13:52.58Kattymew.
13:53.09docelm0k..  I will come directly from work then
13:53.20docelm0I am usually outa here by 5:30ish anyhow
13:53.54docelm0Anyone here got any experience with Mera?
13:54.49iCEBrkrI dated a girl name Mira
13:54.53iCEBrkrOhhh. Mera.. oops
13:54.55bsdfreakhmm
13:54.59docelm0Im talking about Mera MVTS
13:55.03bsdfreakis xpro softphone for pocket pc any good?
13:55.07docelm0I have been tasked to set it up
13:55.14bsdfreakanyone tried it w/evdo?
13:55.31Kattymew?
13:55.37Kattyoh, right you people don't speak kat.
13:55.38Kattyhi!
13:55.39[TK]D-Fenderrawr?
13:55.53Katty[TK]D-Fender: you nearly almost speak kat.
13:55.53docelm0mew?
13:55.54iCEBrkr6hrs until PARTY time
13:56.03docelm0huh?
13:56.08iCEBrkrdocelm0: lol.. 'mew' coming from a guy your size is funny
13:56.15iCEBrkrToday is the company xmas party
13:56.44tzangerhahahaha
13:56.44docelm0ahh..  if we have one Im sure I will not be around to attend..   I am working today Monday and tuesday..  Then I Am off till the 3rd
13:56.53[TK]D-FenderI'm more of the jungle king with a giant paw over the bounty!
13:57.00[TK]D-Fender"My carcass!"
13:59.39azrishahrilanyone with chan_h323 experience ?
14:00.20docelm0iCEBrkr, you havent really seen me..  although many here that went to astricon have
14:00.54zoahe is the goaty
14:01.12zoanera is a mailserver right ?
14:01.14docelm0Not for long..  Planning to cut it off at the end of they year
14:01.16zoaah no thats merak
14:01.24zoaput it on ebay
14:01.27docelm0Mera is SBC
14:01.42PakiPenguinazrishahril, i do
14:03.32wasimPakiPenguin: we've got ss7box working with PK variant ITU for call setup and teardown ...
14:04.26PakiPenguinwoah neat? for which telco? worldcall?
14:04.50PakiPenguinwasim, i'd definately like to site down and talk / learn some stuff some day , you closed the office in isb?
14:04.54rzahmm
14:05.01wasimbrr ...
14:05.19wasimPakiPenguin: yeah, testing it at WorldCALL
14:05.27PakiPenguinneat!
14:05.31wasimPakiPenguin: they were kind enough to allow access
14:05.44PakiPenguinawesome !
14:06.11*** join/#asterisk docelm0 (n=docelmo@66.237.242.41.ptr.us.xo.net)
14:06.15Kattytoday is going to be horror.
14:06.44fugitivowhy?
14:06.44mutilatorvery horror show
14:08.05*** join/#asterisk apardo (n=apardo@251.Red-83-46-189.dynamicIP.rima-tde.net)
14:08.22mutilatorit's friday, i have my 40hr week in, i want to leave, but i'm sticking around til paychecks are cut
14:08.32mutilatorso like..
14:08.44mutilatoroptimally i'll only be here 4-5 more hrs
14:08.57mutilatormeans i have to deal with things for 4-5 more hours
14:09.09mutilatorlike.. *shivers* people
14:09.19Kattyfugitivo: because i'm installing a print traffic monitor (reports spooler inflimation to database) over at a client's office
14:09.36Kattyfugitivo: and from what i hear, their window's network isn't even a domain.
14:09.46Kattyfugitivo: and what's more......they're in like 5 different workgroups
14:09.49*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:09.51Kattyfugitivo: not on purpose, mind you.....
14:10.10fugitivoKatty: i understand, windows, horror...
14:10.13Kattyfugitivo: but because the outsource their IT stuff....and they never bothered with actually having a nice network.
14:10.17Kattyfugitivo: as do i.
14:10.24Kattyfugitivo: being a windowsy tech.
14:10.34Kattyfugitivo: this means no batch script installs :<
14:10.38Kattyfugitivo: it's going to be HORROR
14:10.41fugitivoyeah
14:10.45fugitivogood luck
14:10.52Kattythanks.
14:15.55azrishahrilPakiPenguin : you used which h323 version ?
14:16.30PakiPenguinazrishahril, i use the one from inaccessnetworks
14:17.48azrishahrilPakiPenguin : what OS version were you using ?
14:18.12PakiPenguinfc 4
14:18.17PakiPenguincentos
14:18.48Kattycentos?
14:18.57asteriskmonkeycentos rules
14:19.03Kattyi've never heard of centos before.
14:19.16lexlutherits cool
14:19.20azrishahrilPakiPenguin : i'm using chan_h323 ... seeking information on how to build stable asterisk h323
14:19.21Kattywhy?
14:19.26asteriskmonkeyits redhats commercial version gone freeware :D
14:19.30Kattyis it apt-getty?
14:20.25azrishahrilPakiPenguin : you connect to which h323-gateway vendor ?
14:20.31PakiPenguinhehe :)
14:20.33PakiPenguinits nice
14:20.38azrishahrilcisco ? mera ? veraz ?
14:20.51PakiPenguinazrishahril, cisco + mera + quintum
14:21.12PakiPenguininaccess's h323 has never let me down
14:21.19azrishahrilPakiPenguin : wow ! excellent ...
14:21.22PakiPenguinKatty, centos == awesome!
14:21.32azrishahrilPakiPenguin : any tips ?
14:21.34KattyPakiPenguin: but why?
14:21.39KattyPakiPenguin: TELL ME WHY
14:21.46azrishahrilPakiPenguin : are using asterisk@home ?
14:21.49PakiPenguinKatty, okay okay
14:21.54PakiPenguinazrishahril, nope
14:22.03asteriskmonkeyasterisk@home bad
14:22.05azrishahrilPakiPenguin : build from scratch ...
14:22.11asteriskmonkeywell its good for newbs
14:22.14PakiPenguinyeah :)
14:22.19KattyPakiPenguin: inflimation please.
14:22.37PakiPenguina sec , am on 2x phones
14:23.33*** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus)
14:23.49azrishahrilPakiPenguin : care to share knowledge ? you used chan_oh323 cvs version or stable
14:23.55asteriskmonkeyFreeBSD rules for * server but zaptel=not :(
14:24.22*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:24.52PakiPenguinazrishahril, give me a minute , am at work
14:25.00DandreI would like to provide a queue and in that queue I want the caller to dial in an account number end by # for instance 1234# and then provide this information in the CIDName to the agent that will get the call. How should I do that?
14:25.18*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
14:25.37[TK]D-FenderChange the callerID before dumping them into the queue.
14:26.15[TK]D-FenderSo " exten => 1,1,Read(accountnumber)"
14:26.37MRH2anybody using meetme (without option q) and would like to confirm something for me?
14:26.58[TK]D-Fenderexten => 1,2,Set(CALLERID(name)=${accountnumber}:${CALLERID(name)})
14:27.06PakiPenguinKatty, centos == up2date+yum+rhel+security+fast+redhatlike == very very good for someone who has worked on redhat series
14:27.14[TK]D-Fenderexten => 1,3,Queue(waitaroundawhilebuddy)
14:27.39PakiPenguinazrishahril, just goto http://www.inaccessnetworks.com/projects/asterisk-oh323 , read the README and build.. its pretty easy
14:27.59[TK]D-FenderCentOS is just RHEL with "RH" removed from all places possible
14:28.16PakiPenguinafaik , 0.6.7 is for 1.1.X , and 0.7.3 is for 1.2
14:28.16*** join/#asterisk mesfet (n=mesfet@host130-204.pool82188.interbusiness.it)
14:28.18PakiPenguinyeah
14:28.27gniretar_workugh, all this distros are sorry and crappy
14:28.32gniretar_workexcept Debian
14:28.42fugitivolfs
14:28.43gniretar_workDebian FTW, all others should just GTFO
14:28.51tzangerheh
14:28.55azrishahrili used 0.6.7 before ... seem unstable to me ...
14:28.56tzangerGTFO?
14:28.59PakiPenguinheh
14:29.02azrishahrilasterisk keep crashing
14:29.05tzangerSlackware > *, sorry guys
14:29.13PakiPenguinazrishahril, never gave a problem to me
14:29.19PakiPenguinwhats your call load
14:29.21gniretar_workno
14:29.25gniretar_workSlack sucks
14:29.36azrishahrilthat why i seek other experience to build stable / production system
14:29.36fugitivook, distro war
14:29.41fugitivothen db war
14:29.48gniretar_workMysql > *!!!!!
14:29.48azrishahrilPakiPenguin : do you yahoo messenger ?
14:29.57gniretar_workJabber > *!!!!
14:30.07fugitivowe could finish the day with a nice os war
14:30.15gniretar_workmeh
14:30.18gniretar_workLinux would win
14:30.21tzangergniretar_work: you're a debian weenie, I don't expect you'd understand.  :-)
14:30.34fugitivodistros sucks
14:30.41[TK]D-Fendertzanger : Yes.. Slackware = good :)
14:30.52gniretar_worktzanger: weenie eh?  I'm a weenie because i dont use sissly graphical tools and i edit comfig files like God intended?
14:30.57*** part/#asterisk lexluther (n=s@62.56.236.237)
14:30.59gniretar_workoh yea
14:30.59LostFroglol.. distros suck?
14:31.02gniretar_work.deb > .rpm
14:31.08tzangergniretar_work: slackware uses graphical tools?
14:31.08fugitivoLostFrog: yeah
14:31.13fugitivolinux from scratch for everyone
14:31.17LostFroglol
14:31.24mesfetHi guys. I've a problem with asterisk and zaptel: I'm not able to call any number which start with 1-9... only number which start with 0 can be called.   The command is Dial(Zap/g1/321231231,20) but it does not work. Log return "chan_zap.c: Call specified, but not found?"  Any idea?
14:31.25tzangerLFS is good if you have a real use for it... but that's not very often :-)
14:31.26[TK]D-Fenderfugitivo : Yeah thats a great way to go too.
14:31.32gniretar_worktzanger: OK fair enough but .deb are still > then .rpm
14:31.35LostFrogScrew that.. do it blind with no handbook/archive.
14:31.46fugitivotzanger: servers, no crap on it
14:32.02gniretar_workewwwwww
14:32.06gniretar_workAptitude ftl
14:32.07tzangergniretar_work: yeah the one thing I *do* like about debian is how militant they are about placing everything in the same locations...  I have to admit that the consistency is very welcome.
14:32.09fugitivogniretar_work: packages suck
14:32.18tzangerHowever GNU/everything drives me fucking batty
14:32.28gniretar_workyea, it annoys be too
14:32.32tzangerand yes, RPMs do suck. DEBs suck less, but .tgz is the One True Way
14:32.35gniretar_workpackages do suck
14:32.36[TK]D-FenderPackages are good for things to don't need to be compiled in with specific functionality.
14:32.38gniretar_workbut they are necessary
14:32.44fugitivopackages are good for lazy people
14:32.49gniretar_workfor keeping servers easily up to date
14:33.12fugitivoi keep my servers easily up to date
14:33.17[TK]D-Fendertzanger : Actually I'm not so keen on TGZ.  I'd like dependency checking personally which only seems to leave DEB
14:33.26gniretar_worktzanger: If i had to recompile everything on all my servers to upgrade to a new vesion then i would never get anyything done.  Packages are a necessary evil!
14:33.30fugitivo[TK]D-Fender: gentoo is better for that
14:33.40gniretar_workDeb dependancy resolution > *
14:33.54tzangerI hate dep tracking... totally hate it.
14:33.54[TK]D-Fenderfugitivo : Yeah... but I don't want to spend my life recompiling everything either :)
14:34.03gniretar_workGentoo is not a scaleable server opearing system for enterprise
14:34.05tzangergniretar_work: I never have to recompile everything
14:34.10gniretar_workbut it does kiick ass
14:34.10fugitivogniretar_work: why not?
14:34.34tzangerdep tracking is useless because if you need something not in the tree you now have TWO dep trackers... and that's assinine because they will always be at odds
14:34.38CyberPonygniretar_work, why not?
14:34.42tzangermy dep tracker is ldd  :-)
14:34.43gniretar_workit just isnt, it takes to long to do things and isnt barebones enough (then again, SuSe and RHEL fall intot hat too)
14:35.18fugitivogniretar_work: gentoo is great for servers, easy to manage
14:35.18gniretar_workbut like i said
14:35.25gniretar_workamazing for home
14:35.32gniretar_workhow it is any easier to manage then Debian?
14:35.45RaYmAn-Bxfugitivo: it's just not great when you need to install things...(unless you compile for it elsewhere)
14:35.46fugitivogniretar_work: emerge sync && emerge world
14:35.56fugitivogniretar_work: done, your server is up to date
14:36.02gniretar_workya but since it compiles everything there is a higher chance of bombage
14:36.16tzangertypically speaking I'd like to not have a compiler on my production servers
14:36.30gniretar_worki like having apt-get update and apt-get upgrade on cron jobs. Then if it needs dist-upgrade it emails me and i do it manually
14:36.32fugitivotzanger: me neither
14:36.38brad_msswheh, I sure as hell wouldn't add  emerge sync && emerge -uD world into a cron task ... talk about taking a server down
14:36.45fugitivotzanger: but i don't like to use packages on my production systems :)
14:36.48gniretar_worki'm with tzanger
14:36.59tzangerwell I use checkinstall to make packages and push them ot hte servers
14:37.03gniretar_workfugitivo: how many servers do you have?
14:38.25fugitivogniretar_work: right now, i have 1 openbsd, 5 gentoo, 4 lfs, not much right now
14:40.31KattyPakiPenguin: i don't run red hat
14:40.36KattyPakiPenguin: i run debian
14:40.45gniretar_workgood
14:42.21PakiPenguingood for you :)
14:42.59PakiPenguini run 2 x centos , 4 x fc4 , 2 x os x  , 1x freebsd
14:43.14gniretar_workwhy the freeBSD?
14:43.19gniretar_worknot that i have any problem with it
14:43.21docelm0debian.. cucca
14:43.24gniretar_worki'm just curious
14:44.31PakiPenguingniretar_work, for mera ..
14:45.02docelm0You run mera Paki?
14:45.13fugitivomera?
14:46.03PakiPenguinyes
14:46.20PakiPenguinfugitivo, a voip switch
14:46.38gniretar_workwhat does ti do that Asterisk doesnt?
14:46.41docelm0What do you think of it?   I am in the process of setting it up today
14:47.10docelm0Handle a hell of alot more calls
14:47.10fugitivowhy not ser?
14:47.27docelm0Cause it does both H323 and SIP seamlessly
14:47.45PakiPenguingniretar_work, i dont make decisions :(
14:47.50PakiPenguinyeah that too
14:47.56gniretar_workhaha, i know the feeling
14:47.59PakiPenguindocelm0, pretty neat!
14:48.02docelm0and easier to understand how SER doesnt have good documentation
14:48.24PakiPenguingniretar_work, i run ser and asterisk too  , and voipswitch too ( on windows )
14:48.27PakiPenguinlol
14:48.57docelm0ya
14:50.08bsdfreakheh
14:50.39*** join/#asterisk hadi57 (n=al_moghr@83.136.8.206)
14:51.29docelm0I wish someone would write a book on SER..   There are alot of things I would like to do with it but cant find any good information
14:51.36Dandre[TK]D-Fender: thanks for your tip. Is there an equivalent to Set for 1.0.x?
14:52.14docelm0SetVar
14:52.28PakiPenguinyeah
14:55.21[TK]D-FenderDandre : And you won't have access to the CALLERID() function likely...
14:55.28*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:55.28*** mode/#asterisk [+o anthm] by ChanServ
14:55.29*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
14:55.54DandreYes. May I use CALLERIDNAME variable instead?
14:56.29PakiPenguinanyone has used asterisk 1.2.X with quintum? I am having issues like hanged channels and stuff
14:57.52[TK]D-FenderDandre : SetCIDname(${accountnumber})
14:58.08Dandrethis: exten => 30,5,SetVar(CALLERIDNAME=${accountnumber}:${CALLERIDNAME}) doesn't seem to work
14:58.09[TK]D-FenderDandre : WIKI time for you!
14:58.11Dandrethanks
14:58.37[TK]D-Fendercallerid stuff was command based before
15:00.57[TK]D-FenderNew firefox themes & extensions, yippy.
15:01.11[TK]D-FenderStill like to see a SIP phone for it though....
15:05.34Ariel_any mediatrix experts or anyone with some writing doc's on how to get the 1204 working correctly with asterisk around???? help please...
15:05.46*** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr)
15:05.50bsdfreakis there any way to have asterisk not relay dtmf after it's dialed an extension?
15:11.59fugitivothe attack of the bots!
15:12.24cj-rmhey guys, I've got a hitatchi cable wireless ip5000 sip phone.  Odly the DTMF tones don't appear to work with asterisk where as our zultys zip 2 works fine... Anyone any ideas???
15:12.46PakiPenguinyup
15:12.59synthetiqfor default music on hold, should there be mp3 files in /var/lib/astersik/mohmp3  ?
15:13.13fugitivoyes
15:13.29synthetiqi wonder how they randomly disappeared
15:13.38wasimtermites
15:13.46fugitivohackers
15:14.53[TK]D-Fendercj-rm : Verify what standars for DTMF the phone supports and make sure it matches in sip.conf
15:15.09fugitivocj-rm: how much is that phone?
15:15.17Dandre[TK]D-Fender: It works fine, thank you for your help
15:16.04iCEBrkrHitatchi Cable Wireless IP5000?
15:16.06iCEBrkrhuh?
15:16.29iCEBrkrDoes not compute
15:16.36fugitivothe name?
15:17.02iCEBrkrYeah..
15:17.06iCEBrkrCable Wireless?
15:17.07cj-rmfugitive: it was somewhere in the region of 200-250 quid.
15:17.20*** join/#asterisk xtr (n=01928375@S01060012174cc0e1.vf.shawcable.net)
15:17.22fugitivoiCEBrkr: yeah, ironic
15:17.33cj-rmiCEBrkr: yup cable wireless... bizzare huh?
15:17.36iCEBrkrIs it cable or is it wireless?
15:17.44cj-rm802.11
15:17.47fugitivowireless
15:18.22cj-rmthe sip settings on the phone have the DTMFType set to 0
15:18.33[TK]D-FenderDandre : ywc
15:18.45Dandre:-)
15:18.50iCEBrkrUm, that was a rhetorical question
15:18.53cj-rmwhich is rtp...
15:18.54cj-rmhmmm
15:19.07wasimhu is on 3rd
15:19.16iCEBrkrwho's on first?
15:19.45LostFrogiCEBrkr: yes
15:20.48[TK]D-FenderI don't know who is on 3rd :)
15:24.33LostFrogNo.. "I don't know is on 3rd."
15:25.08[TK]D-FenderSHUP YUO!
15:26.16*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
15:26.20Ariel_hello
15:26.27Hmmhesayshey Ariel_ been awhile
15:26.30fugitivohe
15:26.31Ariel_I guess they disconnected some of us.
15:26.32fugitivoy
15:26.42Ariel_Hmmhesays, yes it's been a little while. I was moving
15:26.57HmmhesaysOh, from where to where?
15:27.09Ariel_Hmmhesays, do you have any writen settings or config files for the mediatrix 1204 setup to an asterisk box?
15:27.30HmmhesaysAriel_ no auto configs, but I can help you out
15:27.40Ariel_Hmmhesays, I moved from Miami(Kendall) area down south about 20 miles to Homestead Florida.
15:27.54fugitivoanyone using polycom IP 301?
15:27.54jeffikAriel_: Hi
15:28.04Ariel_jeffik, hello hope your doing well
15:28.13Ariel_fugitivo, I have a few polycoms out
15:28.23fugitivoAriel_: what do you think about the 301?
15:28.26jeffikAriel_: sure after 6 inches of snow yesterday
15:28.43jeffikmakes our network run faster :-)
15:28.46Hmmhesaysdo you need the mibs for the 1204?
15:28.53Ariel_Hmmhesays, ok any help with these unit's would be good. I am trying to write up a page on the wiki. There are many people asking for help with them and they seem to be a bear to configure
15:29.04Hmmhesaysnaw
15:29.18[TK]D-Fenderfugitivo : Polycom = great business phones.  Whats your question?
15:29.22Hmmhesaysthey are actualy very easy to configure once you've done one
15:29.39Ariel_fugitivo, there ok for cheap phones. But I would spend the difference between the 301 and the 501
15:29.56Ariel_Hmmhesays, sometimes they are sometimes there not
15:29.59fugitivo[TK]D-Fender: what's the difference between the 301 and 501 at functionality level?
15:30.07Ariel_<PROTECTED>
15:30.27Hmmhesaysyou can pm me Ariel, i'll be around
15:30.28fugitivohere the 301 is $180 and the 501 is $350
15:30.30Ariel_fugitivo, night and day.. actually screens are the main one
15:30.33fugitivothere's a big difference
15:30.38*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com)
15:30.44Ariel_here the 501 is around 172 to 199
15:30.45[TK]D-Fenderfugitivo : 501 = pixel LCD screen, SPEAKERPHONE, 3 line keys (not 2),
15:30.48*** part/#asterisk squall (n=squall@omaha.lnet.fr)
15:30.55[TK]D-FenderMuch better "general" office phone.
15:30.57Ariel_and the 301 is around 120 dollars
15:31.16fugitivohere the difference is bigger
15:31.37[TK]D-Fender301 is a good "courtesy" phone and for schmuck workers.  Not having speakerphone sucks though.
15:31.39Ariel_Hmmhesays, will do in a bit going to need to finish working on another problem first. thanks
15:32.05fugitivo[TK]D-Fender: yeah, that sucks
15:32.07[TK]D-Fenderfugitivo : Ok I guess the real question is : How many, and what are the people going to do with them?
15:32.48[TK]D-FenderI find the 301 the lowest end IP phone I suggest to anyone.  Next in line is the SPA-941, the the rest of Polycom's lineup
15:33.01fugitivo[TK]D-Fender: nothing important, i think the 301 will be ok for "normal" workers, and 501 for gerencial people
15:33.06zoa[TK]D-Fender: did you try the st302 ?
15:33.16[TK]D-Fenderzoa : link please.
15:33.44zoahttp://www.voipsolutions.be/public/product_info.php/cPath/41/products_id/206
15:33.47zoathey are pretty good
15:33.54zoabetter than the gxp2000's
15:33.56LostFroggerencial?
15:34.24[TK]D-Fenderzoa : ICK <-
15:34.29fugitivoLostFrog: did i invent the word? :)
15:34.41LostFrogIt appears so.
15:34.43fugitivolol
15:34.56[TK]D-FenderIAX2 for starters.  Buy something interoperable, bad LCD (only upside is the backlight) and buttons... *shudder*
15:35.18LostFrogA phone with no buttons.. lol
15:35.21docelm0zoa, I happen to like the GXP2000 thank you
15:35.31docelm0I have about 60 of them..
15:35.37fugitivois that the PA168?
15:36.07[TK]D-FenderGXP looks sort nice from a distance, but the recessed handset and cheap feel really detract from it.  I'd give it more credibility if they sharpened up the package.
15:36.15[TK]D-FenderWhich is why I love the SPA-941.
15:36.28docelm0I wish the firmware had more options..
15:36.36docelm0Would be nice to add a different graphic and such
15:36.39LostFrogfugitivo: would you care to define it for us?
15:36.53fugitivoLostFrog: what?
15:37.14LostFrogfugitivo: gerencial.
15:37.30fugitivoLostFrog: heh
15:37.44Ariel_I like the SPA-941 better for low end then the Polycom 301. But the Polycom 501 and 601 are in my view the best phones out there for the money.
15:37.59fugitivoLostFrog: managers
15:38.10LostFrogok.. thanks.
15:38.14Ariel_spangilsh
15:38.31[TK]D-FenderAriel_ : I agree.  The hard part is comparing the SPA-941 and IP 501.  They are dangerously close in price which might swing my to favor Polycom which has a better future.
15:39.06Ariel_well about 20 to 30 dollars difference depending on where you get them
15:39.30Ariel_But the spa are getting better. And now the polycom's are supporting asterisk so it's getting closer
15:39.33[TK]D-FenderPoE, 2 ethernet ports, and once you go all-polycom things feel more "solid."  But I still love my SPA-941 at home :)  Its a hard toss-up.
15:39.56[TK]D-FenderAriel_ : All in all great for the commodity SIP market :)
15:39.57fugitivohow much is the spa-941?
15:40.04[TK]D-Fender150$+/-
15:40.40fugitivoi think i have the worse ip phone out there
15:40.42fugitivoatcom 320
15:40.47develanybody using an cisco 1700 with fxo?  i'm seeing it "do stupid" transmitting the rtp back to asterisk (it thinks internally that it's tx, but the packets never get to the interface)
15:40.59[TK]D-FenderAriel_ : I never really USED it, but tested it once and it felt like an "adapted" POTS phone.  Not bad, PoE, but for a screen that size give me PIXELS dammit!
15:41.01Ariel_aastra is a good phone just expensive and hello on the tftp setup.
15:41.29Ariel_hello/hell
15:41.43Ariel_if the tftp is not around phone will loose it's settings
15:42.49[TK]D-FenderYeah, I heard about that.
15:42.54[TK]D-Fender&^%%^& stupid.
15:43.09[TK]D-FenderYou want dependency like that get MGCP :D
15:44.15Ariel_argh mgcp
15:44.33[TK]D-FenderOnly good thing about it is the dial-string RFC :)
15:46.45[TK]D-FenderUmmm is something b0rked on Mantis?
15:46.53nextimeanyone using mcc?
15:50.30astcryzOkay, i have something intresting here. I have a client, and when i call in to it, and the other end answers, it just hangs up, anyone known to that one?
15:50.59fugitivothe other end doesn't want to talk with you?
15:51.03astcryzNo, hehe :-)
15:51.11fugitivo:)
15:51.16astcryzthat so weird.
15:51.24astcryzi cannot see anything wrong in debug either
15:51.40fugitivospecify what you are using
15:51.46fugitivoprotocol, phones, etc
15:51.55astcryzSIP, AVM Fritzbox
15:51.59astcryzi call in from pstn
15:52.03fugitivowhat's that?
15:52.06fugitivoavm fritzbox
15:52.09astcryzATA
15:52.27zoathis 302 is really good, it doesnt look as good but has the best feature set i've seen so far
15:52.48fugitivoastcryz: can you call voicemail from the ata for example?
15:53.01astcryzoutbound calls from the ATA works perfectly
15:53.06zoahttp://www.asteriskguru.com/tutorials/st_302_ip_phone_hardphone.html -> look at some of the info screens
15:53.10fugitivowhat codec are you using?
15:53.19astcryzulaw
15:53.37fugitivoand you don't see anything on the CLI?
15:53.56astcryznothing wrong, now
15:53.57astcryzno
15:54.23astcryzonly about 5300 messages with sip seeding peer from astdb: 'clients number' at ....
15:54.27astcryzhehe :)
15:54.40fugitivozoa: it's the PA168 chip, stay away from it
15:57.37Ariel_astcryz, make sure you have canreinvite=no and both setups are using the same codec
15:57.58wasimwhats wrong with the pa168?
15:58.16coppicenothing really
15:58.42fugitivowasim: i don't know, i have a phone with that chip, and it sucks
15:59.28wasimfeature set, noise, or crappy quality?
15:59.55coppiceI have a vaio that sucks, containing a Pentium, a Hitachi disk, and a Hitachi screen so I guess they must all suck too
15:59.56fugitivoit's full of features
16:00.00fugitivobut crappy quality
16:00.27fugitivoi have 52ms to that phone with the sip firmware
16:00.34fugitivoif i use the iax2 firmware, i get 20ms
16:00.48fugitivothat's something wrong with the chip, not the phone
16:01.13coppicereally? it sounds more like the software
16:01.48*** part/#asterisk rue_mohr (n=not@h24-207-96-50.cst.dccnet.com)
16:02.26fugitivowell, yes
16:02.42fugitivobut i think it's the same software for all the phones using that chip
16:03.14[TK]D-Fender"One firmware to rule them all....."
16:03.22fugitivohehe
16:03.32filestupid people...
16:03.38fileannoy me.
16:03.55LostFrogThat's what people are for.
16:08.00*** join/#asterisk ldnblk (n=Just@82-35-232-48.cable.ubr01.camd.blueyonder.co.uk)
16:08.23azziegood morning gurus. Which SIP phones support "auto-dialing" ? Like, user picks up the phone and it automatically dials a pre-defined number
16:10.30docelm0Good question..  I use ZAP for that
16:10.39*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfil2.dialup.mindspring.com)
16:14.28brad_msswmost SIP phones and adapters support 'batphone' or 'hotline' features
16:14.38brad_msswfor instance: http://www.sipura.com/Documents/faq/Section_2.html#5
16:14.57brad_msswusually configured via the dialplan
16:15.03brad_msswof the SIP adapter itself
16:17.02fugitivothat's nice, didn't know it
16:19.11*** join/#asterisk Psy0rz (n=psy0rz@lounge.datux.nl)
16:19.17iCEBrkr3 hrs and counting
16:19.24iCEBrkrIt's Friday.. It's payday and party time
16:19.54nswintwoo hoo
16:20.20xachenbatphone :P
16:20.24Psy0rzno interface for PLCI=0x702, MSGNUM=0x13!
16:20.30Psy0rzwhat does that no interface message mean?
16:20.36Psy0rzi can call out with misdn
16:20.46Psy0rzbut the received calls arent pickedup by asterisk
16:21.52nswintI need to add an additional 2 line but want to do it cheaply.  I was thinking off adding another line with my current provider for $29 a month.. and then using my skypebox as another line.. and dockntalk with my cell as a last resort
16:21.54Psy0rzin verbose mode i only see this:
16:21.57Psy0rzVerbosity was 0 and is now 4
16:21.57Psy0rz<PROTECTED>
16:22.07nswintis that possible?
16:22.08Psy0rzwhat is supposed to happen after this initial message?
16:22.24movertherecan i put the Local channel to another context?
16:22.36mover+ hi
16:22.38mover:-)
16:22.55Psy0rzit just seems to totally ignore incomming calls
16:23.05Psy0rzshoudlnt it say something like 'blabla context is bla..etc'?
16:23.11gniretar_workhmmm, silly question people
16:23.20gniretar_worki have a macro
16:23.25gniretar_worker
16:23.26gniretar_worknevermind
16:23.29gniretar_workthink i got it
16:23.30gniretar_worklol
16:25.40*** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net)
16:26.35*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
16:29.37gniretar_workhey so i cant figure out how to set the outgoing CID
16:29.44gniretar_workits just not working at all
16:30.04gniretar_workthe remate phone is saying 'caller id unavailable'
16:32.57*** join/#asterisk santiago (n=santiago@208.195.215.154)
16:34.30waddyoutgoing to pstn?
16:34.57gniretar_workyes
16:35.50*** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it)
16:35.54waddyyou using SETCALLERID ?
16:36.12waddyanyone got the Snom Sidekick to work with asterisk?
16:36.33gniretar_worki'm using SETCIDNUM
16:37.19waddy,1,SetCallerID(XXXXXXX)
16:37.22waddyworks for me
16:39.16Psy0rzso can anyone paste me the verbose output of an incomming capi/misdn call?
16:39.21Dandananyone using gxp-2000?
16:39.25Psy0rzso that is can see whats supposed to happen
16:39.32*** join/#asterisk newl (n=newlook@203-59-61-62.dyn.iinet.net.au) [NETSPLIT VICTIM]
16:39.32*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM]
16:39.32*** join/#asterisk kshumard_home (n=ksh@pcp01931374pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
16:39.32*** join/#asterisk ptiggerdine (n=ptiggerd@c220-239-93-118.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM]
16:41.22waddyok
16:42.20waddyPsy0rz: http://pastebin.com/466732
16:42.24waddyis that what you need?
16:44.08jontowany clean way to dial a number and after a timeout dial that number and another number (without dropping the original ring?)
16:44.16Psy0rzyep
16:44.18Psy0rzthx waddy
16:44.26jontowie. dial(sip/100&after-timeout-of-20s(Local/NXXNXXXXXX))
16:44.33Psy0rzcan i see your config?
16:44.38waddyi reposted, it may be a bit better now
16:44.44Psy0rzi'm trying for hours
16:44.56waddywhat you got and what you trying to do?
16:45.00LostFrogjontow: read the wiki
16:45.16LostFrogThere is a specific example of that.
16:45.25Psy0rzi'm trying to receive incomming calls
16:45.42jontowmy problem is i don't want to see the original call drop and start again (because i know how to replicate that easily)
16:45.47jontowdon't want to create a wrongly missed call, as it were
16:45.52jontowbut i'll look ;)
16:46.25waddychange capi.conf context to point to your incoming in extensions.conf, find out how your MSN is sent in 3 or 4 digits
16:46.38Psy0rzi'm trying to receive incomming calls with chan_capi, but i only see the first line
16:46.52Psy0rzwell i'm using AMP and changed it to from-pstn
16:47.02Psy0rzbut it seems to ignore it completely
16:47.07*** join/#asterisk Assid (n=assid@203.115.64.62)
16:47.09Assidheya
16:47.10Psy0rzalso no warnings or furter messages
16:47.18Psy0rzonly the boring 'incomming call' message
16:47.19Assidi wonder what happened to irc.freenode.net it doesnt connect
16:47.27waddybot attack
16:47.37Assidhrmm.. sucks
16:47.38waddyincoming context - exten => _150,1,Wait(1)
16:47.39waddyexten => _150,2,Dial(SIP/200&SIP/201,20)
16:47.40jontowassid; irc.freenode.net is a round-robin DNS pointer.. try a few more times, or find a real host for a server
16:47.42fugitivoAssid: problems with bots
16:47.43jontow:)
16:47.59Assidjontow: has to be the bots.been trying since over 1/2 hrs
16:48.00jontowoh, real problems :(
16:48.05jontowdamn, sorry
16:48.10Assidhrmm.. okay i think 1.2.1 has an issue
16:48.13waddyThe msn sent is 150 from ISDN to asterisk
16:48.15AssidMOH refuses to play the next song
16:48.28fugitivohttp://freenode.net/news.shtml
16:48.44Psy0rzwaddy i want it to accept any msn
16:48.45PakiPenguinassid: chat.freenode.net for the time being
16:48.49LostFrogwaddy: why the patter matching? _150 matches the same as 150.
16:48.55LostFrogpettern
16:49.02fugitivoAssid: what kind of moh?
16:49.15*** join/#asterisk bkw_ (n=bkw_@adsl-69-154-145-194.dsl.tulsok.swbell.net)
16:49.18LostFrogDamn fingers.
16:49.27waddyLostFrog: yes i dunno
16:49.34Assidusing mp3 .. i mentioned a directory with a few mp3 files. but it always plays one song.. over and over..
16:49.45wasimAssid: annoying isn't it?
16:49.48fugitivoAssid: native moh?
16:50.00*** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net)
16:50.02Assidnah.. using mpg123
16:50.07*** join/#asterisk jahani2 (n=k@adsl-227-47-192-81.adsl.iam.net.ma)
16:50.10Assidps shows all the mp3's mentioned in the mpg123 .. but it never plays
16:50.11Psy0rzso if i use a non existing context, i wouldn't see any debug or verbose messages?
16:50.21waddyyou should
16:50.29QwellAssid: What version of mpg123?
16:50.34Assidi dont think * can play native mp3.. atleast it doesnt manage to for me
16:50.35LostFrogPsy0rz: it normally tries default.
16:50.37QwellIt *must* be 0.59r
16:50.39waddygrab ngrep and run that
16:50.49waddysee where you are failing
16:50.50AssidQwell: yes
16:51.01Qwelland yes, * can play natively
16:51.09fugitivoAssid: yes it does with the addon
16:51.12Qwellasterisk-addons, get format_mp3
16:51.14Qwellbbl
16:51.29fugitivothat
16:51.36Psy0rzi'm using msn=* and incommingmsn=*, so why wouldnt i see more messages?
16:51.44Psy0rzlike Exectuing...blabla
16:51.50fugitivoAssid: but everytime you put someone on hold, the mp3 will start from the beggining
16:51.53Assidokay lemme get addon
16:52.02Assidfrom begginng ?
16:52.06Assidcan it play random files?
16:52.16LostFrogAssid: does 'ps axwwww' show multiple mp3s behind the mpg123?
16:52.17waddycan you make outgoing isdn calls?
16:52.20Assidhow come it wont play from wherever
16:52.24waddywhat does capi info say?
16:52.31fugitivobegining
16:52.34Psy0rzoutgoing is no problem
16:52.38Assidi see 2 processes
16:52.51Psy0rzContr1: 2 B channels total, 2 B channels free.
16:52.51Psy0rzContr2: 2 B channels total, 2 B channels free.
16:52.53*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
16:53.17LostFrogShow one of your mpg123 processes.
16:53.31Assid<PROTECTED>
16:53.32fugitivoAssid: check /etc/asterisk/musiconhold.conf, [native-random]
16:53.35moveri there a nice tool to dump the sip packets on an ehternet?
16:53.55Psy0rzwhen i dail out i see tons of messages
16:53.56Assidfugitivo: i tried that.. and put whatever is in that contexct into default
16:54.09fugitivoAssid: but you're not using native
16:54.14Assidnah
16:54.16fugitivoAssid: first get the addon
16:54.20Assidlemme do that
16:54.22Psy0rzi use protocol=2 while loading the drivers, i'm in the netherlands/europe
16:54.34Seldon1975Arg! Im at my wits end - can someone tell me why Comedian Mail isnt sending email notifications? sendmail is running.  here is voicemail.conf http://pastebin.com/466736
16:55.05fugitivoSeldon1975: check sendmail logs
16:55.17fugitivothe answer is always there
16:56.17Assidi gotta also figure out a way to interface with mozilla's address book.. thats freaky
16:56.27Psy0rzi'll upgrade from chan_capi 0.6 to 0.6.1 first
16:56.29Psy0rzlots of fixes
16:57.01Assidwanna have a central directory for contact for telephones, etc
16:57.47wasimAssid: moziax
16:57.57Assidmoziax?
16:58.20Seldon1975fugitivo can you tell me whene to find the sendmail logs, typically?
16:58.24Assidyou kidding me.. there is something ready?
16:58.45fugitivoSeldon1975: /var/log/mail.log or /var/log/sendmail.log or something like that
16:58.55fugitivoSeldon1975: i use postfix, the log is /var/log/mail.log
16:59.05Seldon1975ok ta
16:59.36Assidwasim: i just want to get the address book outta mozilla, the phones are polycoms.. and will get the addressbook via ftp
17:00.05wasimeww
17:00.07AssidQwell: is that an issue with mpg123 to not play the rest of the files?
17:00.12Assideww ?
17:00.46fugitivoAssid: i never had that problem
17:01.09TonyMmover: check out ethereal - it understands many, many protocols
17:01.49Assidi only started getting it on my other box once i loaded 1.2.1, the old CVS release worked fine
17:02.46fugitivoAssid: i'm using native moh with 1.2.x, so i can't tell you if it's an issue with mpg123
17:03.12Assidyour using format_mp3 ?
17:03.29Assidokay i ran a make install on the addons.. do i need to edit the modules.conf ?
17:03.52fugitivono
17:03.55fugitivono need to do that
17:04.12fugitivojust setup musiconhold.conf with native
17:06.11Assidcool
17:06.12Assidworks
17:06.22Assidbut it always starts from the beginning
17:06.24Assidhrmm
17:06.41LostFrogThat's why I don't want native.
17:06.48docelm0Does anyone know if the database is broken in asterisk?   or am I just doing something wrong..  Cause it was working b4 I upgraded now its not.
17:07.32fugitivoAssid: that's what i told you, i like it because i have a message for the people, not music
17:07.54Assidhrmm
17:07.57fugitivobrain wash :)
17:08.11Assidhehe
17:08.15Assidsame message over and over
17:08.29[TK]D-Fenderdocelm0 : They depreciated DBGet and DBPut if thats what you mean
17:08.30Assidlemme guess: buy my product.. dont buy anything else.. you love us..
17:08.37fugitivoyes
17:08.47fugitivowith oriental music for relax
17:09.02SkramXUPS says my package is out for delivery
17:09.18SkramXlast time it said that, they mysteriously lost it and it took them another 2 days to get it to my residence
17:09.22SkramXheh
17:09.59docelm0What is it now?
17:10.25docelm0cause I was pulling DB variables for call forwarding
17:10.33LostFrogYou guys must be unlucky bastards.
17:10.43LostFrogI have never had a problem with a carrier.
17:11.41Assidfugitivo: any way to come to know which file its playing?
17:12.34SkramXLostFrog: Yeah..
17:12.59SkramXit's a 19" LCD from Tigerdirect.com, after rebates its $199-USD, which is a damn good price
17:13.13nextimeanyone using mcc for billing?
17:13.23Seldon1975fugitivo:  my mail log says: Dec 16 11:48:18 asterisk1 sendmail[13358]: jBGGg0fL013358: to=Vlayko Knezik <vlayko@radintl.com>, ct
17:13.24Seldon1975laddr=root (0/0), delay=00:06:18, xdelay=00:06:18, mailer=relay, pri=235183, relay=[127.0.0.1] [127.
17:13.24Seldon19750.0.1], dsn=4.0.0, stat=Deferred: Connection timed out with [127.0.0.1]
17:13.33fugitivoAssid: err.. check the moh command from the cli
17:15.42docelm0[TK]D-Fender, What did they change it to?
17:18.15[TK]D-Fenderjust use the DB() function.
17:19.03[TK]D-Fenderso like "exten => 1,1,Set(DB(family/key)=value)
17:19.14[TK]D-Fenderso like "exten => 1,1,Set(value=DB(family/key))
17:19.19fugitivoSeldon1975: well, you should check your sendmail config
17:19.31fugitivoSeldon1975: is sendmail working?
17:19.41[TK]D-FenderSkramX : What kind of monitor?
17:19.57SkramXX2gen something
17:19.59SkramXwant a link?
17:20.07docelm0how go I pull information from the db then?
17:20.08[TK]D-Fendersure
17:20.19SkramXhttps://www.tigerdirect.com/applications/SearchTools/item-details.asp?Sku=C911-1904
17:20.27SkramXits not the best ever, but for the price, a pretty good deal
17:20.37[TK]D-Fenderdocelm0 : my first sample is the style for DBPut, the second was for DBGet
17:20.43gniretar_worksetcallerid isnt working
17:20.44[TK]D-Fenderso like "exten => 1,1,Set(value=${DB(family/key)})
17:20.45gniretar_workstrange
17:20.47docelm0ohh ok
17:20.56[TK]D-Fenderthis is a corrected version.  I forgot the braces
17:21.27[TK]D-FenderDB()=, then =${DB()}
17:22.08docelm0exten => _XXX,2,set(DID=${DB(CFIM/${EXTEN})})
17:22.11docelm0is that right?
17:22.38[TK]D-Fenderexaclty right
17:22.47[TK]D-FenderCall foward immediate?
17:23.36Assidhow do i stop mysql being loaded with asterisk?
17:23.51docelm0yes
17:24.46[TK]D-Fenderdocelm0 : Careful with the ASTDB stuff... or you'll end up with a 20k dialplan!!! ;)
17:25.26[TK]D-FenderThough mine is very nice now :)
17:25.28fugitivoAssid: what distro?
17:25.52Assidgentoo and debian
17:26.03Assidactually i just added a noload cdr_addon_mysql
17:26.16Assidi think it did the job
17:26.30fugitivoAssid: you don't want mysql to be started?
17:26.38fugitivoor just the asterisk module?
17:27.13Assidasterisk module
17:27.17fugitivook
17:29.19docelm0I wish they would stop screwing with shit..  It breaks my dialplan!
17:30.20docelm0is there a howto on the DB function somewhere?
17:30.22*** join/#asterisk ao2 (n=u@2001:1418:117:0:0:0:0:1)
17:30.29ao2hallo
17:30.36[TK]D-Fenderlookup "functions" on the WIKI and you'll get the list
17:30.50docelm0ao2 where the hell are you connecting from?   You have an IPv6 IP
17:31.00[TK]D-FenderThere is a DBEXISTS() function you'll use instead of priority jumping
17:31.34ao2docelm0: you can easily guess from my IP
17:32.58synthetiqis there an iax2 stress tester outh there?
17:33.10docelm0I cant guess shit.
17:33.32Assidhaha.. anyone tried loading the id3ed on a debian box?
17:35.00zoawe are making one soon (iax2 stresstester)
17:35.02SkramX[TK]D-Fender: Nice deal..
17:35.38ao2I have a question about zap channels. I have a digium clone card (a modem) which uses the wcfxo zaptel module, and I want to ask if it is possible to make asterisk detect when I _lift the handset_ attached to the loop port of the modem.
17:35.39[TK]D-FenderSkramX : which?
17:36.01SkramXthe monitor i linked to ya
17:36.05Assidconfigure: error: installation or configuration problem: C++ compiler cannot create executables.
17:36.10Assidridiculous
17:36.32ao2docelm0, I'm connecting using a tunnel broker, just to let you know :)
17:36.39*** join/#asterisk KranZ (n=user@sme.bestline.net)
17:36.58*** join/#asterisk apardo (n=apardo@207.Red-83-50-107.dynamicIP.rima-tde.net)
17:37.00[TK]D-FenderSkramX : This is what I use at work and am looking to get for home. http://images.ncix.com/imagesthumbnails/16517_20051013_1.jpg
17:37.34SkramX[TK]D-Fender: nice, yet probably very expensive?
17:37.35KranZthat pic is huge
17:37.56[TK]D-Fenderumm
17:38.22[TK]D-Fenderhttp://images.ncix.com/images/16517_20051013_1.jpg
17:38.25[TK]D-Fenderthere
17:38.38[TK]D-Fender19" widescreen. +/- 300$ USD
17:38.50Assidbah.. debian.. getting on my nerves: sox: Sorry, no MP3 encoding support
17:38.53[TK]D-FenderAcer AL1916W.
17:39.25gniretar_workAssid:   lame is your friend
17:39.27[TK]D-Fender14440x900.  very nice res and incedible bang/buck for widescreen.  20" will coust you > 50% more
17:40.11*** join/#asterisk mutilator (n=animenod@65.111.201.79)
17:40.30SkramX[TK]D-Fender: Yea, I have an acer laptop, pretty good brand
17:41.43gniretar_workPowerBooks > *.laptops
17:41.57gniretar_workor laptops.* if your thinking OO like
17:42.24SkramXlol
17:44.05[TK]D-FenderI love most things Acer makes.  Good commodity company.
17:45.03zoaacer is bad bad bad
17:45.08zoaive never seen such crappy laptops before
17:45.09[TK]D-FenderLIES!
17:45.12zoaand i have had 3
17:45.14BakermdI agree - not an Acer fan
17:45.21zoaand still bought one a month ago
17:45.23zoaand its crap again
17:45.25zoabut it was cheap
17:45.25ao2excuse me, but i can't find any good answare on the net for my question. Is it possible to make asterisk detect when an handset is liften on the zap channel? It can be useful to make some things whith outgoing calls.
17:45.28[TK]D-FenderWell I swear by their monitors at least...
17:45.53Bakermdao2 - Not that I know of...
17:45.56zoathe laptop adaptors always fail after 1 year, the casing bursts
17:46.03zoathey use shitty drivers
17:46.13zoathe mainboards break
17:46.26zoabut they are cheap
17:46.28zoareally cheap
17:46.31Bakermd1st Acer I had was a 486 100 (AMD) when it was new, and it was garbadge
17:47.09ao2thanks, Bakermd. I tought that since ztmonitor can "visualize" the ready tone when i lift the handset so asterisk also can detect this event.
17:47.11BakermdI cringed when I saw Ferrari chose Acer to be the manufacturer of their branded computer :-(
17:48.08BakermdI am not saying that it is not possible - I have simply not run across anyone trying to utilize that functionality... I imagine documentation is scarce
17:48.42Bakermd40% of Dell laptops are rebranded Acer's - The lower lines
17:49.19LostFrogDon't tell me that, Bakermd. :(
17:49.37LostFrogWell.. I switched to HP, anyways.
17:50.03fugitivobrands suck
17:50.05fugitivo:)
17:50.06ao2Bakermd, So should i ask to developers if I can achieve what I want?
17:50.10BakermdI love my HP laptops
17:50.25fugitivocustom laptops and servers are great
17:50.27Bakermda02 - Yes, I think that is really the best route.  See if they have any insight
17:51.02LostFrogbtw.. Chocolate Oranges rule.
17:51.36*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
17:53.25zoamy dell laptop is great
17:53.33zoacompaq also sucked
17:53.38zoadunno if they are better now
17:54.17BakermdHP/Compaq - all of their stuff now has both brands on it somewhere - Compaq PC, HP Serial number, etc.
17:54.20BakermdConfusing
17:54.24[TK]D-FenderSkramX : I'm loking to pick one of those monitors up for $334CAD (288$USD
17:54.40SkramXwhich?
17:54.48SkramXthe acer or the one you linked to? or the one i posted?
17:55.07[TK]D-Fenderthe 19" wis=descreen
17:55.12[TK]D-Fenderwide*
17:55.42[TK]D-FenderBetter specs : http://www.acer.com.sg/products/monitoral1916w/p_monitoral1916w.asp
17:55.42SkramXnice.
17:55.55SkramXi wish i had one, but this x2gen should suffice
17:56.17[TK]D-Fender19" really is the sweet sopt these days
17:56.24SkramXIndeed.
17:56.36SkramXThis will be my first LCD :)
17:56.49[TK]D-FenderI have been considering Dells FP-2005 20" 1680x1050 but its $600CAD
17:57.00zoaim considering the dell 30"
17:57.00BakermdBeware - the res is 1440x900 - that is the same as my 17" widescreen Dell Laptop... you may wish for higher res
17:57.11lofi-revI've tried two different softphones (kiax & iaxcomm) and with both I get choppy sound.  It seems to be X related, because if I switch desktops during the call it seems to fix the problem.  Any ideas for a fix/cause?
17:57.26SkramXi think this one i am getting is 1280x1024.., eg
17:57.26[TK]D-FenderBakermd : I used to have a 19" 1280x1024... I don't miss the eight at all.
17:57.32[TK]D-Fenderheight
17:57.47fugitivomy 17" is 1280x1024
17:57.48[TK]D-FenderI find the witdh far more appealing.
17:57.56SkramXthis one has a 12ms refresh time, thats pretty good, right?
17:58.01SkramX700:1 contrast..
17:58.02fugitivomy laptop 15" is 1400x1050
17:58.10SkramXfugitivo: nice.
17:58.19SkramXmine is 15" 768x1024
17:58.32[TK]D-Fenderfugitivo : Dell?  Sounds like their XGA size
17:58.36fugitivothe screen is really nice, but i'd like more bright...
17:58.43fugitivo[TK]D-Fender: no, it's a custom hypersonic, amd64
17:59.05*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:59.43fugitivothe lcd brand is AU, model B150PG01
17:59.59SkramXmy 15" laptop is a acer 3002lci, 1.ghz Sempron Mobile 2800+.. 512mb ram
18:00.23BakermdI love my Apple 23" Cinema display - 1900x1200
18:00.38fugitivoBakermd: hell, that's nice :)
18:00.48Bakermdthx
18:00.53zoatoo bad they are so expensive
18:00.59fugitivohow much is that monster?
18:01.05BakermdWere over $2000, now they are $1299
18:01.06SkramX2000++
18:01.06zoahttp://www.neowin.net/forum/index.php?showtopic=403932
18:01.13BakermdI had to pay $1799 when I got mine
18:01.22SkramXId spend my money on something else, but hey
18:01.25robl^I want a 72" plasma wall mount monitor :)
18:01.40*** join/#asterisk t0ke (n=t0ke@51.Red-83-46-136.dynamicIP.rima-tde.net)
18:01.40fugitivoi want one of those too
18:01.41*** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org)
18:01.52BakermdI use a Mitsubishi projector at home, 1280x1024 @ 108"
18:02.11Bakermdnative res is only 1024x768 though
18:02.14Seldon1975in voicemail.conf how do I set the full 'sender' address
18:02.19fugitivomitsubishi? isn't that a car's brand?
18:02.36coppice4 wheel drive projectors
18:02.48fugitivoSeldon1975: serveremail=
18:02.56BakermdCars, trucks, construction equipment, tv's, phones, microchips  - Multi-billion doller conglomerate
18:03.08Bakermder, dollar
18:03.30SkramXcoppice: ;p;
18:03.32SkramX*lol
18:03.35*** join/#asterisk Igbothom_III (n=HiltonT@203-206-114-129.dyn.iinet.net.au)
18:03.42fugitivothat's a giant
18:03.55t0kehi, I receive this message in my asterisk Dec 16 07:05:47 NOTICE[7193]: app_dial.c:955 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
18:03.55t0ke<PROTECTED>
18:04.00LostFrogAll the biggest diesel engines in the world are Mitsubishi.
18:04.06t0keanyone know how to fix it?
18:04.16fugitivot0ke: plug the line to the card?
18:04.20LostFrogMake sure chan_zap.so is loaded?
18:04.21Seldon1975fugitivo: I have serveremail set to 'asterisk' but mail still appears to come from root@(myserver).local
18:04.32coppicecircuit breakers :-) I was in Mitsubishi's circuit breaker factory a couple of weeks ago. they must make one heck of a lot, cos its a big factory
18:04.48fugitivoSeldon1975: check your sendmail.cf
18:05.07t0kefugitivo..line is plugged to card. Is there any command to show if span is active?
18:05.28t0kefugitivo..It is TE410P card
18:05.43fugitivowhat color are the lights?
18:05.50t0kegreen
18:06.19t0keis there any command to check if span have signal?
18:06.41fugitivotry zap show status
18:06.42tzangerzap show spans?
18:06.43tzangerer status
18:07.12t0ke*CLI> zap show status
18:07.13t0keNo such command 'zap show status' (type 'help' for help)
18:07.16fugitivothen
18:07.20fugitivozap module is not loaded
18:07.22tzangert0ke: what version of asterisk
18:07.23SkramX"help"
18:07.34t0ketzanger: asterisk business solutions
18:07.37tzangerahh
18:07.40tzangeryeah that's probably not in there
18:07.42fugitivot0ke: show modules like zap
18:07.53fugitivot0ke: you should see chan_zap.so
18:08.02LostFrogt0ke: load chan_zap.so
18:08.05*** join/#asterisk kpettit (n=keith@69.15.174.114)
18:08.09Seldon1975fugitivo: do I need to restart sendmail after editing that fiule?  how do I restart it?
18:08.33fugitivoSeldon1975: depends on your distribution, maybe /etc/init.d/sendmail restart? or /etc/rc.d/init.d/sendmail restart
18:08.33t0ke*CLI> show modules like chan_zap.so
18:08.33t0keModule                    Description                              Use Count
18:08.33t0kechan_zap.so               Zapata Telephony w/PRI                   0
18:08.35t0ke1 modules loaded
18:09.07t0kei think module is loaded
18:09.22fugitivozap <tab>
18:09.23fugitivonothing?
18:09.31fugitivowait
18:09.41t0kezap detroy or zap show
18:09.44t0keno more options
18:09.48fugitivoif you have bussiness version, why don't you ask at digium?
18:10.12*** join/#asterisk RoyK (n=roy@ti211310a080-12051.bb.online.no)
18:10.14RoyKzoa: ping
18:10.20*** join/#asterisk SERGEUS (n=s@195.112.98.13)
18:10.27zoaim here
18:10.36zoabut i need to go now
18:10.38zoaso hurry
18:10.40t0keok..I will ask then..I thinked I would launch here problem and look if anyone knew problem
18:10.41RoyKzoa: just a sec
18:11.04t0kefugitivo: tnks for your help
18:11.07fugitivot0ke: maybe you should get what you pay for :)
18:11.21SkramX:)
18:11.39[TK]D-FenderIn a good way at least
18:11.47fugitivowe could help you, but if you pay for the bussiness version...
18:12.13*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
18:13.06t0kefugitivo..ok, I tried it becaus if was a easy mistake by me in configuration..I havent to wait answer from digium
18:13.12SkramXthere are a couple digiym fulk in here anyways.
18:13.12t0kewell..thnk you again
18:13.16SkramX*folk
18:13.57fugitivot0ke: zap show channels
18:14.07*** join/#asterisk dash (n=washort@adsl-147-122-226.bhm.bellsouth.net)
18:14.59dashif I make a SIP call to asterisk and the dialplan ends up directing it to a SIP address, does asterisk always stay in the media path?
18:16.02[TK]D-Fenderdash : "canreinvite=no"
18:16.33[TK]D-FenderSet it to "yes" and the phones will reconnect with each other directly
18:16.53ms345Anyone know why a user doesn't get an email if they are forwarded a vmail through the web interface? The vmail notification emails with wav attachment  are sent if the vmail is left in a mailbox but not if the message is forwarded from one vmail box to another. Is this normal?
18:18.30dash[TK]D-Fender: hmm. yeah, that would make sense
18:18.38t0kefugitivo...1            iax_asterisk    es
18:18.43t0keuntil 124
18:18.49Seldon1975fugitivo i added "Dmmyserver.com" to sendmail.cf but messages still appear to come from root@asterisk1.local
18:19.06fugitivot0ke: iax_asterisk is the context?
18:19.19strain17is anyone here successful in using h.323?
18:19.46t0kefugitivo: yes
18:22.12dash[TK]D-Fender: that's gonna be tricky with the stuff i'm doing though :)
18:22.17ms345Seldon - what is serveremail set to in vm_general.inc ?
18:23.00*** join/#asterisk klictel (n=klictel@207.107.208.137)
18:23.07fugitivovm_general.inc? what is that?
18:23.09*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)
18:24.03Seldon1975ms345: ill take a look
18:24.07BlackthornHello. When I do "show sip peers" it shows the port numbers that the ata's are attacked to. All of the units are set to use 5060 and 5061 why do they show up under differnt numbers?
18:24.36ms345also look at your /etc/hosts file and put your preffered localhost name in there
18:24.42t0kefugitivo..u know if there is any irc channel specified for asterisk bussiness version?
18:25.07Blackthornattacked = attached
18:25.22[TK]D-FenderATTACK!!!!
18:25.34dasht0ke: probably you are supposed to pay digium for help with that :)
18:25.41ms345my vmail emails seem to come from the vm@ specified in vm_general.inc but the @domain is overwritten with what I have as localhost in /etc/hosts
18:25.53ms345i don't know why - but it works for me :-)
18:28.41strain17does anyone knows how to setup h.323?
18:31.34BlackthornOk, let me try a different question.  * on a routable ip ---> router --->  Wireless -->router2 ---> wireless --> linksys -->and users nat behind this. Including an ata and it shows up in the * box as the router2 ip?
18:32.16malverian[work]Has anyone used the Sangoma WINPIPE software with Zaptel 1.2?
18:32.38malverian[work]This utility to determine the taps needed for echo cancellation seems useful.
18:32.48malverian[work]*WANPIPE
18:33.53*** part/#asterisk dash (n=washort@adsl-147-122-226.bhm.bellsouth.net)
18:33.55[TK]D-Fendermalverian[work] : Funny... I don't have to tell Zaptel ANYTHING with mine :D
18:35.15iCEBrkrSignalling requested on channel 1 is SF (Tone) Signalling Immediate but line is in PRI Signalling signalling
18:35.20iCEBrkrYou're on crack.
18:35.24iCEBrkrIt IS NOT PRI
18:36.51[TK]D-FenderOr... maybe its split signalling :)
18:37.10LostFrogI prefer semaphore.
18:37.53iCEBrkrSPAN 1: CAS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1)
18:38.01iCEBrkrok, so how's that PRI?
18:39.22[TK]D-Fenderummm
18:41.02Blackthornok, i've got to go but also just realized my * version is several versions behind the time. I'll go update it later today.
18:42.55*** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net)
18:46.04*** join/#asterisk Mavantix (n=maverick@69-168-33-232.chvlva.adelphia.net)
18:46.50Mavantixanyone have any tips on getting voice to work on remote X-Lite clients? (What ports to open, etc?)
18:46.52*** join/#asterisk e3c (n=chatzill@sin-dhcp-4-096.si.umich.edu)
18:48.43SkramXhehe, im stalking taxis at http://labs.google.com/ridefinde
18:50.30wunderkinwow.. scary
18:51.11rob0Mavantix: SIP is 5060/udp by default (or whatever you have in your sip.conf)
18:52.26Mavantixrob0: that part works, but when an X-Lite client connects and dials another extention, no voice goes either way. I forwarded ports 10000-10100 to the * box, and configured rtp.conf to use just those ports, so it should be trying to send voice on those ports, right?
18:53.00rob0dunno
18:54.10[TK]D-FenderMavantix : You need to set "localnet" & "externip" in SIP.conf
18:54.23[TK]D-Fenderlook those up on the WIKI and you'll know what to do.
18:59.40strain17robo...can u tell me where to start on setting up H.323?
19:00.26rob0no strain17 I cannot, I said I do not use it.
19:00.52strain17hmm....someone told me...lol
19:01.00strain17* rob0 knows how/why NOT to use h.323 :)
19:02.08Seldon1975hey guys - is there a toll-free number accessable from canada that I can call and just speak down the lin to see if I echo?
19:02.18Seldon1975like a talking clock would do
19:02.48*** join/#asterisk roulduke (i=gvzeaqb3@p508D03DF.dip0.t-ipconnect.de)
19:06.05*** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net)
19:06.10jsaundersG'day
19:06.38SkramXHiya
19:07.10*** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca)
19:12.08docelm0Anyone know how I would evaluate in the dialplan for this:  DB_EXISTS(<family>/<key>)   to find if the key exists?
19:12.35*** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com)
19:12.54iCEBrkrdocelm0: You wanna know if there's data there of if the key exists?
19:13.13iCEBrkrCuz you could DBGET and assign it to a variable and then GotoIf() if there's a value assigned
19:13.30*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
19:15.12SupaplexI must be on a roll. 5th time restarting asterisk today.
19:15.43docelm0DB_EXISTS(<family>/<key>)
19:15.46docelm0crap
19:15.55docelm0exten => _XXX,2,GotoIf($[${DB_EXISTS(CFIM/${EXTEN})} = "1"]?103:3)
19:16.03docelm0is that correct IF the key exists?
19:22.28*** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net)
19:23.24*** part/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net)
19:29.48Supaplexffs
19:31.22docelm0I figured it out..
19:31.26docelm0pain in my fricken ass
19:31.43Supaplexlet's start a support group :p
19:33.33Supaplexis it possible to ring all phones after * (re)starts?
19:34.08iCEBrkrdocelm0: Looks about right.. :P
19:34.10Supaplexjust a short ring, so they know to relogin :-/
19:34.34docelm0I got it..
19:34.48*** join/#asterisk freezer (n=freezer_@ACB339EB.ipt.aol.com)
19:34.50freezerhi
19:35.45freezermh asterisk doesnt see any incoming sip calls and then calling the sip number a voice says "Number is temporaly not available"
19:35.58freezeri mapped all needed udp ports
19:38.38*** join/#asterisk areski (n=areski@152.Red-83-44-70.dynamicIP.rima-tde.net)
19:41.59*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
19:46.03FuriousGeorgeim emerging gnophone over here.  i just installed asterisk 1.2.1.  i notice gentoo wants to emerge net-libs/iax-0.2.2    i assume a version of the iax lib was installed by asterisk 1.2.1 too.  anyone know what verson that is
19:46.07FuriousGeorge?
19:46.45rob0gnophone has not been maintained AFAIK
19:47.11FuriousGeorgei see, i just noticed it on digiums web page and remembered it existed
19:47.52rob0I tried kphone yesterday, followed instructions on the wiki, but can't get it to register
19:48.05rob0(it registers but doesn't think it did)
19:48.13FuriousGeorgekphone huh, lemme try that one
19:48.56rob0I really need a softphone until I can get more FXS ports
19:49.06FuriousGeorgefor linux?
19:49.12rob0yes
19:49.18FuriousGeorgethere is an x-lite for linux
19:49.28FuriousGeorgeive nver found one on say the level of eyebeam
19:50.02ctooleyDoes anyone use the Asterisk::Manager perl module from gnuinter?
19:50.09FuriousGeorgespeaking of libs, did you hear about the google voip for xmpp libs they put on sourceforge recently?
19:50.36iCEBrkrPARTY TIME
19:50.44ctooleyIf so, what all things have been fixed in your version (because I've got several fixes for mine)
19:53.27rob0I got x-lite. Will try that next. I think kphone should work, I just did something wrong.
19:53.52FuriousGeorgerob0: i got mine logged in no problem, but tis fugly
19:54.38rob0mine registers according to the * console, but then it pops up a timeout error anyway.
19:54.49FuriousGeorgeyeah i think thats what i got
21:01.33*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
21:01.33*** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org
21:01.38[Latre]i have a problem compiling zaptel in mandriva 2006 kernel 2.6.12.12 this is the error:  http://pastebin.com/467055
21:01.40trixterthat is retail if you are doing any sort of volume you get discounts
21:01.59marcus2theres a cac 12fxo+12fxs channel bank on ebay from a reputable dealer for $500 right now
21:01.59trixterand that is new with warantee and all that
21:02.11marcus2and $500 for a te100
21:02.40tzangermarcus2: don't get it
21:02.40trixterits quite often hard to sell used equipment to customers
21:02.44*** join/#asterisk amir_ (n=amir@gentoo/developer/amir)
21:02.49tzangermarcus2: CAC AB1/2 do *NOT* do CPD on the FXO ports
21:02.51Seldon1975has anyone here tweaksed the jitterbuffer settings on their Polycom 301 or 501?
21:02.51junbugi prefer external fxo
21:02.52tzangerI have one
21:02.58tzangerthe sales literature is just plain wrong
21:02.59LostFrogI have a T1 card.
21:03.02marcus2trixter; well, i guess thats differnt
21:03.02tzangerAdit600 works *great* though
21:03.08tzangerand CAC AB1/2 work GREAT for FXS
21:03.18marcus2yeah i'm only uss my ab2 for fxs
21:03.19marcus2and it does work great
21:03.25trixterI had a te110p but no more
21:03.38trixterneveru sed it
21:03.53marcus2i still have one unused port of my te410, too
21:03.58trixteronly took it out of the package to see what it was when it came ...  was free afterall
21:04.21CyberPonyanyone have experience getting SIP connections to work across a VPN tunnel between 2 watchguard firewalls?
21:04.34trixterfor customers I get only what is needed and dont keep stock for the sacaug.org stuff I would rather have sip and analog devices
21:04.43LostFrogWell.. 'nough said.. eventually I will switch to either a channel bank or a TDM4200 for my FXO.
21:05.06freezerGot Unsupported Frame with Format:0
21:05.07marcus2how many FXO ports do you need?
21:05.08freezermh?
21:05.13trixterits kinda hard to justify spending $1000 to provide analog lines for demonstrations and lectures :(  so I rely on free gear from companies like thevoipconnection.com and digium :)
21:05.33LostFrogmarcus2: 6
21:05.39trixterwhy the tdm410 is a good deal, it gives us the ability to do lectures with something that is more reasonable to use
21:06.12marcus2ah
21:06.28trixterI can get a dual fxs port dlink ATA for $32.50 to my door
21:06.32marcus2i wish fedex would hurry up!
21:06.41trixterdoing the math that is a better deal, it doesnt look as pretty but its a volunteer group :/
21:07.07LostFrogWell.. I ordered a card from X100P, it's a clone.. but I'll try it.
21:07.13LostFrogX100P.com
21:07.18trixterthat makes it less than half a te110p and channel bank even if used
21:07.34trixterlastfrog: how much did it cost?
21:07.44LostFrog$30+$8
21:07.48trixterouch
21:07.54trixterwhat country are you in?
21:08.05LostFrogUS
21:08.16LostFrogWhy ouch?
21:08.43trixter$17 to your door for a clone off ebay
21:09.00trixternkans or someone is realy the only seller, where I got my parents from and they work great
21:09.28trixter4000-5000 feedback and others in  here have bought from him too with the same results, what you expect for the price you expect shipped when you expect, etc
21:09.31*** join/#asterisk HolyGod (i=nobody@got.securebinary.com)
21:09.45trixterI dont often recommend ebay sellers but this one is fairly good
21:09.51trixterat least with this one product
21:10.30*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
21:10.50Sedoroxtrixter: I think thats who I got mine from
21:10.52Sedoroxboth actually
21:10.54[Latre]i bought 10 clones to nkans and never sendme anything..... i send 50 email asking what happwnd and nothing....
21:11.13Sedoroxweird... mine was quick shipped
21:11.24[Latre]i live in MExico
21:12.40marcus2sweet
21:12.46marcus2my polcom 601 and gxp2000 just showed up!
21:13.58justinubeauty and the beast
21:14.31marcus2heheh
21:14.55marcus2$85 vs $250
21:14.59marcus2and i need to buy 50 of them
21:15.17LostFrogSedorox: what does your card detect as?
21:15.27justinuthe gxp2000 is missing a lot of features
21:15.40marcus2yeah but
21:15.46marcus2is it missing any features that matter to my users
21:16.00justinuwell, it won't ring on the speaker if you have a headset plugged in
21:16.26*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
21:16.29marcus2and thats not a firmware-solvable issue?
21:16.36justinunot from what I hear
21:16.48justinunot certain tho
21:18.53zoamarcus2: buy the st 302
21:18.58zoaits great
21:19.00zoaroyko!
21:19.02zoaim back
21:20.38marcus2st302?
21:21.33*** join/#asterisk Gamercjm (n=gamercjm@pool-71-254-174-15.lsanca.fios.verizon.net)
21:21.38*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
21:21.47Sedoroxwho was asking me about my x100p clone?? (sorry.. comp locked up)
21:22.01Gamercjmhow would i make an incoming call go to my cell phone?
21:22.31LostFrogSedorox: I was.
21:22.50Sedoroxthought so.... Suse Linux detects it as a ISDN card....
21:22.57Sedoroxumm
21:23.04LostFrogwhat does wcfxo detect it as?
21:23.13SedoroxFound a Wildcard FXO: Generic Clone
21:23.19LostFrogahh.. ok.
21:23.28SedoroxRegistered tone zone 0 (United States / North America)
21:23.42Sedorox0000:00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
21:24.01zigmanhahah ISDN interface ;)
21:24.14LostFrogwell.. it is a modem.
21:24.24zigmanyeah.. ANALOG
21:24.25trixterjustinu the gxp2000 is missing a lot of features  -- like what?
21:24.30zigmanisDn is digitial
21:24.39trixterIsdn is integrated :P
21:24.42LostFrogIt says Modem *or* ISDN.
21:24.50zigmanyeah i know
21:24.53zigmanjust think its funny
21:24.59trixterit was a joke cause its really not integrated
21:25.01SedoroxSuse said it was a Tiger ISDN interface.. but I didn't go through the setup :p
21:25.02justinutrixter: like call forwarding
21:25.03trixterunfufilled promises
21:25.06*** join/#asterisk kolorado (n=kolorado@129.82.174.111)
21:25.09Sedorox(its in a Novell OES (suse 9 pro)
21:25.09trixterit cant forward calls?
21:25.12Sedorox)
21:25.20justinudon't think so
21:25.21trixteris that it?
21:25.28zigmanSedorox if you can get a rj45 plug into the card let me know ;)
21:25.34justinudialed, missed calls
21:25.38justinudirectory
21:25.38trixterthere is a work around if you create a forwarding system within asterisk but meh
21:25.50justinuyeah
21:25.52justinupita
21:25.57Sedoroxzigman: ahahaha
21:25.58justinuand no on-phone indication of the forwarding
21:25.59Sedoroxyea.. right.. :p
21:26.03marcus2heh. this 601 is *so* much nicer than the gxp2000
21:26.08Sedoroxhas two rj11's on it
21:26.09trixterit should do a call log at least..  that is bothersome
21:26.11justinumarcus2: see?
21:26.11Sedoroxnormal modem....
21:26.12koloradoHey does anyone know how the new VMCOUNT() app works.  Just upgraded to 1.2 and hasvoicemail broke :(
21:26.16Sedoroxwell.. winmodem
21:26.19trixteroh well at least my customer that is getting 15 of em doesnt expect those features
21:26.29justinutrixter: on phone interface is a joke, very difficult to nagivate
21:26.36marcus2and i havent even got them working with asterisk yet, heh
21:26.59justinui have a customer getting 20
21:27.05marcus2this thing came with old firmware
21:27.15marcus2and polycom wont let end users download firmware from their site. lame.
21:27.26justinuone nice thing is the gxp2000 is PoE
21:27.29justinufor 85 bucks!
21:27.39marcus2so is the 601, for a mere $160 premium ;)
21:27.43Sedoroxor just save up for a 601
21:27.59trixteryeah the customer wanted $50 phones but I couldnt recommend any for that price
21:28.08marcus2budgettone!
21:28.08trixterthis was the cheapest I could find that would be reasonable
21:28.12Sedoroxbudgetone
21:28.12robl^Snom is kewl!  and firmware easy to obtain from vendor website.  No extra fees or password needed.
21:28.17Sedoroxif you want JUST phone functions
21:28.18Sedoroxits fine
21:28.21trixterthere are a lot of problems with those from what I have read
21:28.32SedoroxI haven't had too many problems with mine
21:28.36twisted[asteria]mine flung itself off the desk a few times
21:28.37Sedoroxjust want more functionality outta it
21:28.37Sedorox:p
21:28.43justinusnom doesn't support connected partty ID
21:28.46trixtereverything from a 0.4mm size difference in the power adapter (so it occasionally falls out) to audio problems to ...
21:28.46justinuwhich is lame
21:29.13Sedoroxtrixter: for the BT100... I've used ilbc, gsm, and g729... all works nicely...
21:29.17Sedoroxnever had a problem with power
21:29.22zoathe bt100 is crap
21:29.23zoaCRAP!
21:29.27Sedoroxlol
21:29.36justinuilbc is crap too
21:29.37zoai lost a costumer due to it
21:29.39Nivexwho here has played with the spa-941?  opinions?
21:29.55*** part/#asterisk [Latre] (n=latre@148.233.19.133)
21:30.50robl^the BT101 isn't even useful as a paperweight.  One sneeze and it goes flying off the desk. ;)
21:30.59Sedoroxlol
21:31.01trixteroh speaking of ilbc I have a strange problem...  a dlink dvg1120s (sip) connecting to asterisk via any codec other than g.711 (a/ulaw) using rfc2833 dtmf results in asterisk not processing dtmf and reporting bad ilbc frames.  the dvg1120 doesnt do ilbc, I have disabled the module in asterisk, disallowed it in sip.conf, etc..  the only way it will work for dtmf is inband signalling and g.711..  any ideas?
21:31.05SedoroxI've never had these problems with it..
21:31.06Sedorox:p
21:31.44koloradoAnybody familiar with Aterisk 1.2, specifically vmcount()????
21:32.03trixterI have tried g.729, g.726, g.723, and g.711..  only g.711 works all the others when using rfc2833 report invalid ilbc - when using g.711 inband dtmf works
21:32.06justinutrixter: it's a problem with the dynamic payloads, i think
21:32.19Sedoroxanyone try those linksys/sipura/cisco sip phones yet?
21:32.28justinuphone is sending the wrong dynamic payload number and making asterisk think anthing other than g711 is ilbc
21:32.29trixtera friend has the same problem, he is using 1.0.7 I am 1.2.0 ...  so this problem isnt new aparently
21:32.31justinu?
21:32.34justinushrug
21:32.41trixtermaybe
21:32.46lofi-revIs it possible to dial 2 numbers, and when the first answers it is bridged to the second call?
21:32.48justinui found problems with eyebeam and speex
21:32.54Sedoroxthere is issues between 1.2.0 and 1.0
21:32.55trixterthat is a problem though..  I might hunt around for firmware upgrades on it and see if there is any solution that way
21:33.08SedoroxI think specificly trunking I believe
21:33.19justinueyebeam tries to use the ilbc payload type, and starts talking speex to asterisk which thinks it should be ilbc
21:33.22trixtercause it would be nice to use g.729 especially when I take it into the sac AUG and do the live install ...  wanted to bring a edge card and stuff..  but meh
21:33.23justinuasterisk freaks and drops the call
21:33.54justinutrixter: in your dlink interface can you tweak the dynamic payload values?
21:33.56justinusipura can do this
21:35.19trixterother than framesize no I dont believe so
21:35.25trixterthis device works a lot like it wasnt finished
21:35.35justinulol, just like the gxp2000 :)
21:36.06justinutrixter: pastbin some sip traces if you want to figure it out
21:36.14*** join/#asterisk amir_ (n=amir@gentoo/developer/amir)
21:36.27trixterI was just wondering if anyone else saw this type of problem before..  I will work on it later
21:36.38*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:36.42justinuaight
21:36.47trixterI have a few weeks until the next meet and then its not required I can do ulaw, just thought it might be nice to nothave to :P
21:36.53*** join/#asterisk bmg505 (n=leon@dsl-146-5-216.telkomadsl.co.za)
21:37.02nswintI'm experiencing somethign interesting.. asterisk is somehow converting the callerid from my addresbook into a name and number and recording it to the Master.csv.... how is that possible?
21:37.25GamercjmIs this correct for trying to call an a cell phone exten => s,2,Dial(IAX2/NuFone.net/*cellnumber*,15,rtm)
21:37.27file[laptop]it uses whatever is present for the callerid values on the channel
21:37.30file[laptop][TK]D-Fender: !!!
21:38.03[TK]D-Fenderfile[laptop]: !!!
21:39.15file[laptop][TK]D-Fender: driving... to Nova Scotia
21:39.50justinueh?
21:39.59[TK]D-Fender...yay!  Try driving HERE :D
21:40.08file[laptop]or not
21:40.09lofi-revhas someone already written a tool for retrieving callerid name data from an external DB?
21:40.16[TK]D-FenderIt ^&%^&% ugly out there.  We topped 40cm!
21:40.25file[laptop]112 km/hour!
21:40.28file[laptop]oh no, we're speeding
21:40.44[TK]D-Fenderlofi-rev : yes go check the voxilla formus, its in the first page on Asterisk
21:40.52lofi-rev[TK]D-Fender, thanks
21:40.53LostFrog15"
21:40.55LostFrog?
21:40.57LostFrogwow.
21:41.41[TK]D-Fenderyes, its shit-on-a-stick out there.....
21:41.51file[laptop][TK]D-Fender: you should go play in the snow!
21:41.55file[laptop]:P
21:41.58LostFrogIt even stopped snowing here.
21:42.01file[laptop]walk in a winter wonderland
21:42.39nswintlofi-rev: it's weird
21:43.07[TK]D-Fender*FAP* !
21:43.30nswintlofi-rev: numbers not in my addressbook show up fine.. don't know how asterisk is connecting to my address book.. or where is't getting my addressbook info
21:44.52lofi-revnswint, asterisk or your client?
21:44.54justinufrom the softphone
21:45.39nswintasterisk
21:45.50lofi-revnswint, what address book?
21:46.11nswintits on a seperate box
21:46.18[TK]D-FenderJust upgraded the firmware on my SPA-941 :)
21:46.47*** join/#asterisk Equinox (n=secret@pool-71-251-73-183.tampfl.fios.verizon.net)
21:47.00EquinoxAny idea why playing sound files(gsm) stops working when I insmod zaprtc & zaptel?
21:56.51*** part/#asterisk kolorado (n=kolorado@129.82.174.111)
22:04.27lofi-revIs it possible to return from a Dial command before the call is hungup?
22:08.18RenacorIs there any apps for asterisk that will extract extensions or create a directory/address book?
22:08.52trixtergrep
22:08.52trixterperl
22:08.54trixter:P
22:09.13Renacorlol
22:09.18RenacorYour great
22:09.30trixterafaik no there is no existing tool to parse extensions.conf and pull that out, and you would want to limit that so you dont create a directory of every route you have
22:10.10trixterI would rather import extensions into asterisk (there are ldap modules, enum, etc) rather than export from asterisk but ...
22:11.53Renacorthat would be cool
22:12.07trixterif you could integrate extensions, voicemail, etc all from ldap or some other central repository it would be a better enterprise solution (note I did say you could do this above before anyone jumps all over me saying you can)
22:12.27trixteralthough I dont know about voicemail from that, havent looked
22:12.47trixterbut that way you have one place to add someone for access to machines on the network, phone system, etc
22:13.01trixterbut now I must work
22:17.09Renacortrixter: Yeah thats exactly what I would need
22:17.13Renacorto make my life easier
22:23.32Sedoroxhmm
22:32.02*** join/#asterisk sterne (n=art@246-84.customer.cloud9.net)
22:32.10*** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org)
22:44.53*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
22:49.49*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
22:56.22HolyGodb00 TheCops
22:56.40TheCopshey
22:58.20TheCopsSomeone is familiar with G.729a codec from digium ? I just installed it, registered it and loading it. When I'm making a call from a SIP -> PSTN, it's written g729tolin_framein: Out of G.729 Decoder Licenses!, I have 2 licenses.
22:58.35*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
22:58.44mog_homedare you doing a monitor?
22:58.53mog_homeand do a show g729
22:59.13junbugany recommendations on external  4 or 2 port fxo -> sip gateways
23:00.17mog_homesipura
23:00.56junbugthey make external fxo-> sip?
23:02.22SedoroxAnyone currently using ldap with asterisk?
23:02.52Sedoroxjunbug: yes...
23:03.22*** part/#asterisk joelsolanki (i=joelsola@202.160.161.93)
23:04.46Sedoroxok.. actually seems like sipura only does a 1/1 combo.. not 2 fxo..
23:05.20*** join/#asterisk santiago (n=santiago@208.195.215.154)
23:06.01*** part/#asterisk santiago (n=santiago@208.195.215.154)
23:06.41junbugyea i would prefer external fxo's over pci's
23:07.20justinuaudiocodes
23:07.43justinulots of people make that kind of stuff
23:07.57*** join/#asterisk bweschke (n=bweschke@72.242.52.52)
23:08.25mog_homewhy not get a card?
23:08.36TheCopsintranet*CLI> show g729
23:08.36TheCops1/2 encoders/decoders of 2 licensed channels are currently in use
23:08.46TheCopsthis is mean, 3 license ?! 1/2
23:09.21LostFrogNo.. 2 licenses.
23:09.23Sedoroxlooks like one is decode/encode.. the one is just decode
23:10.00TheCopsLostFrog, I have already 2 license, and I have a out of license
23:11.27junbugmog_home: I here external setup's are just more stable
23:11.36junbughear that also
23:15.02mog_homeeh
23:15.20mog_homeit think the initial set up is harder
23:15.24*** join/#asterisk pablo4545 (n=pablo@181-77-246-201.adsl.terra.cl)
23:15.24mog_homebut after that
23:15.30mog_homeid rather have  a pci card
23:17.56*** join/#asterisk sese (n=sese@host81-159-76-154.range81-159.btcentralplus.com)
23:18.59TheCopsLostFrog, nice, found my problem, was doing a monitor before..
23:18.59LostFrogDamn ebay idiots, list channel banks, don't give part numbers, don't say what modules are installed..
23:19.34TheCopsweird, the call is established but I'm not hearing something
23:20.29mog_homethen its 0 modules lostfrog
23:20.35mog_homeor 1 or the other
23:21.11LostFrogBut, to try to sell one without info..
23:21.13*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
23:21.31mog_homewell they are hoping someone is a nubb
23:21.34mog_homeand doesnt check
23:21.35LostFrogLike this one, especially: http://cgi.ebay.com/Channel-Bank_W0QQitemZ5841724271QQcategoryZ11908QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
23:21.37mog_homethink its a steal
23:22.54mog_homeyeah looks shady
23:23.26LostFrogThese guys are idiots.
23:23.35LostFrogAt least put the model number on the listing.
23:23.52*** join/#asterisk Zach^^ (n=Zachary@65.121.244.130)
23:24.00Zach^^anyone setup skype on asterisk?
23:24.12mog_homeNO
23:24.14mog_homeand you cant
23:24.17mog_homesorries
23:24.23Zach^^you cant?
23:24.32mog_homeonly through skype to sip gateway
23:24.39Zach^^any like skype that you can?
23:24.55mog_homegizmo i guess
23:26.15marcus2huh, how is this grandstream figuring out its timezone?
23:26.40bweschkesip.esp
23:26.41bweschke:)
23:26.47marcus2ah
23:26.49marcus2i see
23:27.01mog_homelol
23:27.07mog_homentpdate maybe?
23:27.08bweschkedo you have a timezone offset field in your dhcp scope?
23:28.10marcus2i do
23:28.26*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
23:28.28marcus2the polycom figures its timezone out and displays the proper time... the grandstream is 3 hours ahead
23:29.02justinuthere's a place in the web interface to set it
23:29.07marcus2ah
23:29.17justinunot sure if it takes the parameters from dhcp or not
23:29.20justinunever thought of trying that
23:29.30marcus2i'll try the web interface
23:37.11TheCopseyebeam is supporting g729 ?
23:37.15bweschkeyep
23:37.47TheCopsthe same codec as asterisk ? (G729a I think) ?
23:38.04TheCopsWhen I'm calling somewhere using g729, I dont hear anything.
23:38.39justinueyebeam works
23:40.23TheCopsvery weird
23:40.27LostFrogTheCops: what version of *?
23:40.31TheCops1.2.0
23:40.45LostFrogThere was a bug with g729 and 1.2.0
23:40.51TheCopsduh
23:41.00TheCopsI just finished to update *grins*
23:41.01TheCopslol
23:41.27LostFrogOk.. just making sure you weren't dealing with that issue.
23:41.47TheCopsWhat's the issue, precisly ?
23:43.10LostFroghttp://bugs.digium.com/view.php?id=5780
23:43.30LostFrogok.. that may only be a DTMF issue.
23:44.03TheCopsAsterisk sends invalid SDP, causes RFC2833 not to work
23:44.04TheCopsnice
23:44.07TheCopsthat's the bug I have
23:45.08LostFrogedit chan_sip.c
23:45.12LostFrogand rebuild.
23:45.37*** join/#asterisk bkw__ (n=brian@adsl-69-154-145-194.dsl.tulsok.swbell.net)
23:45.45TheCopsI prefer to update the whole * version
23:45.57LostFrogThen, move to 1.2.1
23:46.06TheCopsThanks LostFrog
23:46.11TheCopsvery appreciated
23:48.58LostFrogNot a problem.
23:49.55LostFrogI like to be helpful once in a while. :p
23:50.35*** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com)
23:51.56_Simonhey gang. where can I go to get some information on 1.700 and 1.800 VOIP allocations? I'm looking to create some extensions but I would like to play friendly with other VOIP servers and not conflict
23:52.16_Simonor any information where I can place my extension allocations to be compliant
23:58.14trixterby 1700 do you mean NANPA 1700 numbers?
23:59.14_Simonumm not sure, I'm new :) I'm trying to figure out what extensions to give my users for our test/dev server but I don't want to conflict with existing servers
23:59.15*** join/#asterisk sese (n=sese@host81-159-76-154.range81-159.btcentralplus.com)
23:59.24_Simonso when we make it public etc we play friendly with everyone
23:59.39_SimonI was told this was important if I want my server to connect to others

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