irclog2html for #asterisk on 20041008

00:00.00salimfadhleyIf they are using iax2 without a secret that means anybody could pretend to be me... nice.
00:01.40SplasPoodhttp://www.protestwarrior.com/nimages/the_sign.jpg
00:01.41SplasPoodheh
00:04.38redder86is there a way to query if an extension is in-use?  setgroup/checkgroup doesn't perfectly fit the bill for me because I often dial multiple extensions like this Dial(extension1&extension2), and setgroup is useless here in distinguishing which extension answered the call.
00:06.45visik7anyone use res_config?
00:07.39bkw_salimfadhley you didn't fix the problem
00:07.43bkw_visik7 yes why?
00:08.02bkw_redder86 doesn't matter it will still work
00:08.24bkw_you want to know who answered you're SOL in that case
00:08.46bkw_thats why blah&blah2&blah3 suck on dial
00:09.02visik7bkw_ I can't figure how it works wiki page is not so clarifying
00:09.57*** join/#asterisk habakuk (~chatzilla@adsl-64-168-23-239.dsl.snfc21.pacbell.net)
00:11.01twistedhahaha
00:11.05twisted"we're gonna flex the tight end"
00:13.13*** join/#asterisk r1 (~erwan@www.thiscow.com)
00:13.29*** join/#asterisk doughecka_ (~dheckaman@doughecka.user)
00:13.30bkw_visik7 res_config in cvs-head is diffrent
00:13.31doughecka_Running ztcfg:  ZT_CHANCONFIG failed on channel 1: No such device or address (6)
00:13.35bkw_no device
00:13.37bkw_reinstall drivers
00:13.49bkw_or fix zaptel.conf
00:14.03doughecka_only happens after kudzu runs
00:14.07doughecka_and I turned it off
00:14.11doughecka_but it still happened
00:14.43doughecka_same error
00:14.56ManxPowerdoughecka: I'll bet kudzu thinks an ixj is in the system.
00:15.23ManxPowerremove the card, run kudzu to remove the configs, put the card back in, tell kudzu to not install the card
00:16.08twistedheh
00:16.10twistedbetter yet
00:16.13twistedremove kudzu
00:16.36*** join/#asterisk gabe (~gabe@c-064-186-245-122.sd2.redwire.net)
00:16.41PoWeRKiLLAny idea why I just restart my sever and asterisk is running back fine but when I call to moh or voicemail it's not working anymore ?
00:16.58redder86bkw_: is there any option other than Dial(blah&blah2&blah3) ?
00:17.27doughecka_twisted: heh
00:17.35JunK-Yredder86: what ya wanna do exactly ?
00:17.45bkw_redder86 not really
00:18.06redder86JunK-Y: I want to only have call-waiting work to the extensions for specified callers
00:18.15PoWeRKiLLThe only thing i've done is remove kernel support of RTC and compile zaptelrtc
00:18.25PoWeRKiLLdo you think this is the pb ?
00:18.56bkw_ya think thats the issue.. gee I wonder
00:19.15redder86JunK-Y: I don't want to have my ongoing calls disturbed by other people unless they are in my "priority calls" list.  But I *do* want to have those priority callers get through via call-waiting.
00:19.17doughecka_still giving me the error
00:19.56ManxPowerdoughecka: lsmod shows the ixj module loaded?
00:20.06doughecka_ManxPower: its rebooting now
00:20.10redder86JunK-Y: so I can't disable call-waiting on the phones.  I need to have a way to see if the extension is in-use before Dialing it
00:20.13DaminNo.
00:20.24tzangerdoughecka_: can you help me with firefly?
00:20.29doughecka_possibly
00:20.34ManxPowerredder86: See "setGroup" in the wiki
00:20.36doughecka_its mindboglinly simple
00:20.41tzangerdoughecka_: I thought so too
00:20.43doughecka_mindboggingly
00:20.46tzangerI can dial
00:20.49doughecka_ooh
00:20.51doughecka_recieving calls
00:20.54JunK-Yredder86: and its to dont disturb them right ?
00:20.56tzangerit gives me a funky ringback though (sounds european)
00:20.56doughecka_makesure callerid is passed
00:20.58tzangerand it connects
00:21.01tzangerbut no audio is passed
00:21.03doughecka_oh
00:21.04tzangerjust silence in both directions
00:21.05doughecka_no audio?
00:21.11doughecka_make sure your codecs match
00:21.12ManxPowerdoughecka: So that fixed it?
00:21.13redder86ManxPower: setGroup doesn't work exactly as I would need it when I dial extensions with multiple entries: Dial(x&y&z)
00:21.20tzangerdoughecka_: they should match
00:21.20tzangerheh
00:21.21redder86JunK-Y: yes
00:21.30ManxPowerredder86: Use chan Local.  also see the Wiki
00:21.33doughecka_ManxPower: no, lsmod?
00:21.44ManxPowerlsmod lists the modules loaded in Linux
00:21.44doughecka_# cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
00:21.47doughecka_eep
00:21.51doughecka_# cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds
00:21.55ManxPowerdoughecka: But it it's working.....
00:21.55doughecka_blah, stupid copy and paste
00:22.00JunK-Yredder86: that could be interesting yes, ive DND button on my phones, if ya find a way, please let me know how, im curious.
00:22.03PoWeRKiLLanyone using zaptelrtc ?
00:22.14doughecka_ManxPower: asterisk does not start up
00:22.21doughecka_ztcfg -vv gives that error
00:22.41doughecka_ok
00:22.47doughecka_it shows zaptel loaded
00:22.49*** join/#asterisk Phantom-X (someone@h133n2fls24o1068.bredband.comhem.se)
00:22.51doughecka_used by wcusb
00:22.55*** join/#asterisk VoiceLynx (~rda@user-0cdv656.cable.mindspring.com)
00:22.55Phantom-Xhello there
00:22.58doughecka_wcusb is unused and loaded
00:24.01doughecka_wcfxo says no such device
00:24.05doughecka_modprobe wcfxo *
00:24.23tzangerdoughecka_:
00:24.24tzanger<PROTECTED>
00:24.24tzanger<PROTECTED>
00:24.24tzanger<PROTECTED>
00:24.24tzanger<PROTECTED>
00:24.26tzanger<PROTECTED>
00:24.27JunK-Y~seen lennyt
00:24.32jbotlennyt <~lenny@rockbox-gw.voiping.com> was last seen on IRC in channel #asterisk, 5d 6h 36m 16s ago, saying: 'since the pap2-na is not avail ... anyone figure how to unlock the pap2?'.
00:24.32tzanger<PROTECTED>
00:24.32tzangerlooks like it would work to me
00:24.34tzangerbut no audio
00:24.53doughecka_tell it to disallow=all allow=ilbc
00:25.01doughecka_and turn ilbc on in firefly
00:25.24PoWeRKiLL~seen kapejod
00:25.25jbotkapejod <~kapejod@pD9E83D65.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 97d 8h 4m 29s ago, saying: 'later on you will be able to use any elcheapo hfc-pci card'.
00:25.34Phantom-Xim more or less unfamliliar with asterisk , I joined here just to get some info , I cot a cisco ata 186 , right now its wunning behind a NAT:ing firewall , now I wonder if I could use asterisk to act as proxy/answering machine etc  I mean when im not at home I wnant somethging to answer the dial or redirect it
00:26.38ChulJinyes.
00:26.43*** join/#asterisk crdobbs (~crdobbs@175.247.232.64.transedge.com)
00:26.48Phantom-Xis it hard to achieve it ?
00:26.55doughecka_ManxPower: still having the problem
00:27.14doughecka_removed the hardware entry in /etc/sysconfig/hwconf
00:27.17doughecka_and rebooted
00:27.22doughecka_still gives the same error
00:27.25tzangerew ilbc
00:27.25doughecka_cant find the card..
00:27.28doughecka_tzanger: ew?
00:27.31ManxPowerdoughecka: What about the lsmod showing?
00:27.31doughecka_then try ulaw
00:27.36ChulJin~ManxPower's 'Useful Asterisk Docs'
00:27.43doughecka_~manxpower
00:27.45jbotGo to http://www.digium.com/index.php?menu=documentation and look at the "Unofficial Links" section also see http://www.voip-info.org/wiki-Asterisk also see http://www.fnords.org/~eric/asterisk/
00:27.46ChulJinhmmm
00:27.48ManxPowerUseful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/
00:27.57doughecka_ManxPower: heh, zaptel and wcusb are loaded
00:28.09doughecka_wcfxo is NOT loaded
00:28.09ManxPowerdoughecka: ixj isn't loaded?
00:28.16doughecka_ixj?
00:28.16ChulJindough: hehe I knew jbot would have to have memorised those by now.
00:28.17doughecka_hmm, no
00:28.19ManxPowerdoughecka: Is this a new card or a new system?
00:28.24doughecka_both
00:28.26ManxPowerand if you modprobe wcfxo ?
00:28.29doughecka_just installed OS
00:28.36doughecka_that says it cant find the device
00:28.39doughecka_this is a TDM400
00:28.44doughecka_with 4 FXO
00:28.48ManxPowerYou need wcfxs
00:28.59tzangerdoughecka_: interesting -- it still  negotiates GSM even though iax.conf has disallow=all/allow=ilbc and firefly only has ilbc checked
00:29.05tzangerreloaded * and exited/reloaded firefly
00:29.07doughecka_tzanger: reload? :)
00:29.07ManxPowerdoughecka: the zaptel README lists the module -> card info
00:29.12Phantom-Xthanks for the urls
00:29.12doughecka_shoot :)
00:29.26doughecka_ManxPower: oh, I try loading the right module
00:29.37ManxPowerdoughecka: It usually makes things work better 8-)
00:29.40doughecka_heh
00:29.50ManxPowercitats: Are you around?
00:30.21doughecka_or, hopefully will be
00:30.23ManxPowerdoughecka: You like pain, don't you?
00:30.26doughecka_well
00:30.27tzangerdoughecka_: hmm reSTARTed asterisk and it still negotiates GSM
00:30.27doughecka_it WAS working
00:30.29doughecka_great
00:30.31doughecka_till I rebooted
00:30.42doughecka_tzanger: sux to be you :)
00:30.50doughecka_thirdparty version correct?
00:31.21tzangerthirdparty what?
00:31.39doughecka_you downloaded the thirdparty version of firefly correct?
00:31.42tzangeryes
00:31.55tzanger1.9.5 build 3935
00:32.11*** join/#asterisk Kumbang (~kumbang@167.205.22.54)
00:32.14doughecka_huh
00:32.17tzangerin the about page it shows a bunch of 3rdparty links too
00:32.22PoWeRKiLLanyone using zaptelrtc ?
00:32.23ManxPowerdoughecka: If you continue to have problems you can /msg me.  You've helped enough people for me to spend time to help you.  Next time don't install a server the day before it goes into production
00:32.24crdobbsI am an * noob.  Is there a codec that doesnot bucher FAX transmissions?
00:32.30tzangeruses libiax2... uses vivida licence... uses speex...
00:32.38ManxPowercrdobbs: ULAW and ALAW are the only ones
00:33.03crdobbsdont thoes suck up a lot of bandwidth
00:33.04PoWeRKiLLcrdobbs: and you have to get a perfect bandwith :)
00:33.08Phantom-Xdamn asterisk seems really cool =)
00:33.24tzangerdoughecka_: I have route my internal and external calls using this netowrk checked, is that correct?
00:33.44tzangerif I uncheck all codecs I get an error when I try to call saying it can't negotiate codec
00:33.52doughecka_yes
00:34.00tzangerNOTICE[131080]: chan_iax2.c:5664 socket_read: Rejected connect attempt from 192.168.1.105, requested/capability 0x0/0x0 incompatible  with our capability 0x2.
00:34.04tzangerand that in the * log which is correct
00:34.07doughecka_[2006]
00:34.11doughecka_type=friend
00:34.15doughecka_secret=blah
00:34.18doughecka_host=dynamic
00:34.22doughecka_notransfer=yes
00:34.23crdobbswhat codec has the best quality vs bandwidth ratio
00:34.25*** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net)
00:34.26doughecka_thats what I am using
00:34.31doughecka_for my iax.conf for my firefly client
00:34.32doughecka_and it works
00:34.35tzangercrdobbs: I like gsm for free and g729 for pay
00:34.39tzangerdoughecka_: yeah that's what I have too
00:34.41redder86JunK-Y: I'm thinking about parsing the 'asterisk -r -x "show channels"' output in my AGI script that I already use for caller*id-based call routing
00:35.00Docelm0Anyone in here done any AGI PHP scripting?
00:35.02redder86JunK-Y: so if the extension is in-use, then it will know
00:35.02crdobbswhat aboud ADPSM, has anyone used that?
00:35.20crdobbsexcuse me ADPCM
00:35.24bkw_evil
00:35.25bkw_use g726
00:35.30bkw_ADPCM sounds like ass
00:35.51crdobbsthat good, ha :)
00:35.52tzangerbkw_: no, lpc10 sounds like ass
00:36.07tzangerdoughecka_: I have verified that it is only letting gsm through which is right
00:36.14bkw_speex 2,3 sounds EXCELLENT
00:36.26tzanger(I tried with the other checked one at a time and it failed to negotiate whcih is correct)
00:36.30tzangerbut yeah
00:36.32tzangerno audio
00:36.34tzangergrrr
00:36.41tzangerxlite works fine but I hate xlite
00:36.50crdobbsthankyou for the info.
00:36.58doughecka_tzanger: huh... :(
00:36.59tzangerhaven't played with speex
00:37.06tzangerwhat's the 2,3 behind that, bkw_ ?
00:38.53ariel_Has any one worked with a lucent Vina-intergrator Channel Bank here?
00:39.32file[laptop]Paypal is quite nice on 'da phone...
00:39.50file[laptop]"Can you check to see if my replacement card is being mailed out?" "Can you remove these preauth charges?"
00:40.06tzangerdoughecka_: I get no audio at all (no ringback) if I tell firefly to use SIP
00:40.13tzangeroh wait
00:40.26Docelm0Can anyone help me?   Im looking for a little help with the whole PHP AGI script deal..
00:40.29tzangerI get no ringback but I get audio once hte call is established
00:40.30doughecka_hmm
00:40.40tzangerIAX2: ringback but no call audio
00:40.47tzangerSIP: no ringback but call audio (and it's crappy quality)
00:40.54tzangerno other changes
00:42.58doughecka_HAHA
00:42.59doughecka_wierd
00:43.14habakukdocelm0 what are you trying to do?
00:43.36doughecka_tzanger: I would check my own config in firefly but I am not booted into windows
00:44.11*** join/#asterisk RageMax (~max@dhcp-064-247-097-106.eg4.ohiou.edu)
00:44.28PoWeRKiLLremoving rtc from my kernel completely broke up playbacking audio file from * is that normal ?
00:44.43RageMaxis that linux ip phone out yet that supports IAX out of the box?
00:44.47Docelm0Get the variables etc..  Just screw around with the interface to see what I can do with it.
00:44.57tzangerdoughecka_: no it's not that
00:45.01Docelm0Like how do I pass var's to php and pass them back etc..
00:45.01tzangerI'm sure of it
00:45.05ManxPowerPoWeRKiLL: for cvs -head anythng is possible, for cvs -stable that should not happen
00:45.17tzangerI did a make update I saw some interesting changes over the last few days
00:45.47PoWeRKiLLManxPower I'm currently using -head
00:45.48doughecka_hmm
00:46.00ManxPowerPoWeRKiLL: Do you need the -head features?
00:46.23habakukPoWeRKiLL, huh I had the same problem recently. why Do you think it is caused by rtc?
00:46.24PoWeRKiLLI always use -head so I don't know what the stable don't have :(
00:47.02ManxPowerPoWeRKiLL: Don't use -head then.
00:47.15ManxPowerAs of 1.0 -stable is what *I* think people should use for production
00:47.44*** join/#asterisk Inv_arp (junya@adsl-10-164-152.mia.bellsouth.net)
00:47.50PoWeRKiLLhabakuk cause I wanted meetme in my * box, but I can't use ztdummy I have usb-ohci, so I remove rtc from my kernel
00:48.47habakukPoWeRKiLL, right I have the same issue. so you are using zaprtc? and seeing this? when was the last time you updated your code?
00:48.47PoWeRKiLLand compile zaptelrtc I load it but I don't have any audio on MOH Voicemail or other playback
00:49.20PoWeRKiLLhabakuk
00:49.21PoWeRKiLLCVS-HEAD-09/20/04-23:17:39
00:49.26PoWeRKiLLand you ?
00:49.42habakukPoWeRKiLL, right I had the same problem, but when I disabled conferencing it all started working again.. didn't look at it too seriously though
00:49.54habakukmine is from august sometime
00:50.30PoWeRKiLLwhat do you mean you disable conferencing you are just not dialing to meetme app or you remove the apps ?
00:51.40habakukPoWeRKiLL, hmm.. I think all of the above I noticed that it didn't show up until I called the conference. So I just disabled the app, so I didn't call by accident
00:52.10tzangerdamn I get shit audio quality out of firefly, period
00:53.05PoWeRKiLLinteresting I'm currently download 1-0 from cvs  to see if the problem get resolve
00:53.27PoWeRKiLLif not I will try to remove the app and if not I will get back RTC in my kernel
00:54.22tzangerahh there better quality
00:54.25tzangerbut still weirdness heh
00:54.26ariel_I have a problem with a channel bank. I don't get a ring. I can take calls if I know there calling, I can make calls without problems ARgh.
00:54.41tzangerariel_: what channel bank
00:54.44ariel_Anyone know if signally should be loop start or ground.
00:54.53ariel_Lucent Vina
00:55.53ariel_Why can't they all just work out of the box like the Adtrans.
00:58.00ManxPowerariel_: I think loopstart is the default for most things
00:58.04PoWeRKiLLso habakuk currently you have or you don't have rtc support in your kernel ?
00:58.28habakukPoWeRKiLL, no rtc support at the moment
00:59.26doughecka_tzanger: works?
00:59.30PoWeRKiLLat all ?
01:02.44tzangerdoughecka_: better audio quality but still same problem as before
01:02.56doughecka_huh.
01:03.14habakukPoWeRKiLL, zaprtc is loaded
01:03.24doughecka_if it works on a cold boot
01:03.29doughecka_I will eat my hat
01:04.06doughecka_dang it
01:04.06ariel_ManxPower, that is what set it to. But it does not ring.. Argh. I need to get a shoot gun and shoot it.
01:04.07doughecka_it does work
01:04.16Jason357what is the smallest linux distro astrisk esily installs on?  Does it install well on DamnSmallLinux.org?
01:04.18doughecka_ariel_: shot gun? :)
01:04.39tzangerJason357: I have it running off slack91 in under 500M and that wasn't even trying to shrink it
01:04.48ariel_that way they get something like an Adtran or a CAC which I know work.
01:07.33PoWeRKiLLright now no
01:11.36PoWeRKiLLManxPower I try -stable and still same problem
01:12.55ManxPowerPoWeRKiLL: What does "show version" show?
01:13.09Connor_~seen kram
01:13.13jbotkram is currently on #asterisk.  Has said a total of 4 messages.  Is idling for 3h 42m 7s
01:13.20PoWeRKiLLManxPower: Asterisk CVS-v1-0-10/08/04-02:54:28 built by root@sip on a i686 running Linux
01:15.50PoWeRKiLLI think I will recompile back the kernel with RTC
01:15.54*** join/#asterisk Poemius (~poemius@adsl-70-48-192-81.adsl.iam.net.ma)
01:16.27Poemiushi everyone
01:17.48*** join/#asterisk AsteriskJacob (~trillian@md-frederick-bw1-68-170-244-136.chvlva.adelphia.net)
01:18.22habakukPoWeRKiLL, hmm are you running a 2.6 / 2.4 kernel?
01:18.42PoWeRKiLLhabakuk 2.4
01:19.04AsteriskJacobhey peeps! I like to use the Cvsup script from the asterisk.org download page... Does anyone no how to modify it to get the 1.0 branch?
01:19.21habakukPoWeRKiLL, I heard that it's supposed to work very well with 2.6.. but I haven't tried it yet
01:19.21AsteriskJacobI thought *default release=cvs tag=1.0 would work...
01:19.27AsteriskJacobbut it doesn't seem to
01:20.00habakukPoWeRKiLL, zaprtc that is
01:21.07*** join/#asterisk MicroChip32 (~mc@DSL-49-70.che.centurytel.net)
01:21.08Poemiusanyone successfully use distinctive ring on incoming sip calls?
01:21.24Poemius(to route call to a different extension)
01:21.57outtoluncAsteriskJacob: why not mirror the whole thing?
01:21.59PoWeRKiLLhabakuk may be but I don't think that I will upgrade a major kernel version now on a production server :)
01:24.06MicroChip32is it possible to make sip work, reliably, behind nat. here's the situation: i use wifi at assorted hotspots, which are generally natted. I would like to be able to use my sip device (pap2) at these locations.
01:27.14PatrickDKmicrochip, only if they support stun, and you support stun, will that work
01:28.17PoWeRKiLLmy kernel is still compiling hope it will work i put the native RTC as module so I can load it and unload it easyly
01:28.20bkw_AsteriskJacob its tag=v1-0
01:29.05MicroChip32PatrickDK: 'they' refers to the hotspot ??
01:29.15*** join/#asterisk jeb-c4 (~jeb-c4@pcp03913576pcs.nash01.tn.comcast.net)
01:29.20AsteriskJacobjust noticed that... thanks
01:29.27PatrickDKyep
01:30.34MicroChip32and me supporting stun refers to my sip device (which does) or the voip server (which i dunno, its asterisk)
01:30.40jeb-c4just curious as to what CVS revision to track for stable? (It appears that v1-0_stable is dead)
01:30.59PatrickDKna, I think just the nat box and your box have to support stun
01:31.37MicroChip32"your box" = server (asterisk) right ?
01:31.53PatrickDKyour box = whatever your using on the hotspot
01:32.12bkw_Vonage is planning to use MP3 playback technology for voice mail soon, according to company CEO
01:32.16MicroChip32ah ok, i know my sip device does. who knows about the hotspot. the one i know is a linksys, no idea about the others
01:32.24bkw_OK I WANT MP3 playback of voicemail by tommorow
01:32.55icebalmcan't do it, patent encumbered, we can do ogg vorbis tho
01:32.56PatrickDKmp3 playback?
01:32.58PoWeRKiLLbkw_ Vonage use * for their voicemail right ?
01:33.09PatrickDKwhy would that be a customer benifit?
01:33.14hmodesaww comon bkw, give us our moment ;p
01:33.17hmodesit took long enough
01:33.22PatrickDKI only see that as a benifit to the company
01:33.45outtoluncyou should already be able to do mp3->ice->* <G> (ducking)
01:33.48hmodes(hi, by the way)
01:34.12amerVonage use * for their voicemail right ? is this true?
01:35.02bkw_hmodes we will have it by morning
01:35.05bkw_so na na ne boo boo
01:35.17bkw_accually anthm and I are very pissed off at the voicemail in asterisk
01:35.23hmodesdamn!
01:35.25bkw_expect some new shit to come from the anthm factory
01:35.41hmodesamer: http://investorrelations.utstar.com/ReleaseDetail.cfm?ReleaseID=137677
01:36.46Phantom-Xdo you guys know why emerge asterisk would want to install apache ( I have strong aversiona gainst webservers ) ?
01:38.22amerhmodes: so this means they are not using * for voicemail
01:38.25bkw_emerge -pv asterisk
01:38.42bkw_Phantom-X do not EMERGE asterisk
01:38.45bkw_you do it froms rc
01:38.50bkw_the portage is so out of whack its not even funny
01:39.05bkw_its 0.9.0
01:39.08bkw_its OLD OLD OLD
01:39.20amercan I apply patch for the latest cvs build to a month old build?
01:39.21Phantom-Xhmm
01:39.24bkw_Phantom-X do you not know how to use USE flags?
01:39.29Phantom-Xthe put up something newwer
01:39.43PoWeRKiLLhabakuk it's working :) The problem was the RTC no meet me but everything works back
01:39.47Phantom-Xbkw_, yes and still even if I have -apache and -apache2
01:39.54icebalmbkw_: 1.0 is in portage
01:39.57Phantom-Xit tries to emerge that mongoloid apcahe
01:40.17Phantom-XI really hate to have a webserver on my host
01:40.20bkw_emerge -pv
01:40.29bkw_emerge -pv asterisk
01:40.31Phantom-Xyou trie emerge -upv
01:40.35bkw_no u
01:40.36bkw_u bad
01:40.50icebalm-u is not bad
01:41.02bkw_it sure can be
01:41.09Phantom-Xand saw im supposed to be able to do USE="-apache2" but it still persist in instaling apache
01:41.09icebalmonly if you're a bloody retard
01:41.15bkw_haha no
01:41.21icebalmbkw_: how can it be bad
01:41.25bkw_export USE="-apache2"
01:41.42Phantom-X[ebuild  N    ] net-misc/asterisk-1.0.0  +alsa -apache2 -doc +gtk -mmx -mysql -nopri -nozaptel 10,993 kB
01:41.44bkw_icebalm I haven't run into the problem with u but someone else in here informed me
01:41.52bkw_Phantom-X its not going to install apache
01:41.52Phantom-Xim not haing to export it at all
01:42.12bkw_Phantom-X you export your USE flags
01:42.16Phantom-Xyou should just do : USE="-apache2" emerge asterisk
01:42.21icebalmbkw_: it's not bad, there's no senario where it can harm anything, unless you're not using portage properly
01:42.28bkw_maybe it was U
01:42.43Phantom-Xwell
01:42.45PoWeRKiLLManxPower habakuk I found the problem you have to compile kernel RTC as module not as NO
01:42.45Phantom-XI checked :
01:42.57bkw_Phantom-X never install asterisk from portage
01:42.59Phantom-XUSE="-apache2" emerge -upv asterisk
01:43.02bkw_never install mpg123 from portage
01:43.04Phantom-Xstill wanted to install apache
01:43.08bkw_you're doing it wrong
01:43.14Phantom-Xno im not
01:43.14bkw_export USE="-apahce2"
01:43.16bkw_then emerge
01:43.23Phantom-Xthats the way you are suppose to do with emerge
01:43.43bkw_you can export it to the USE var in your env too you ninny
01:43.46bkw_don't listen to me then
01:43.51bkw_I do this all the time
01:43.55icebalmPhantom-X: asterisk doesn't depend on apache, it depends on mpg123 and newt, the rest are configurable via USE
01:44.03Phantom-Xstill trying to install apache
01:44.09Phantom-XI did your variant also
01:44.14*** part/#asterisk AsteriskJacob (~trillian@md-frederick-bw1-68-170-244-136.chvlva.adelphia.net)
01:44.23Phantom-Xstill emerge wants to install apache
01:44.26bkw_if it has a -apache2 when you do emerge -pv
01:44.30bkw_then its not going to do apache
01:44.59icebalmPhantom-X: when you do emerge -vp asterisk what packages does it try to install?
01:45.35icebalmthe -apache2 use flag is moot, the asterisk ebuild doesn't even look at apache
01:45.51bkw_<Phantom-X> [ebuild  N    ] net-misc/asterisk-1.0.0  +alsa -apache2 -doc +gtk -mmx -mysql -nopri -nozaptel 10,993 kB
01:45.51Phantom-Xicebalm, I tries to instal some packages
01:45.56Phantom-Xand one of them is apache
01:46.06bkw_then something else that you need is depending on apache
01:46.09bkw_along the way
01:46.13icebalmthat does it install right after apache?
01:46.15bkw_but that shouldn't
01:46.21icebalmwhat does it install right after apache?
01:46.35Phantom-Xbkw_, well the list tells non is but asterisk
01:46.45bkw_the -apache2 doesn't mean shit
01:46.48Phantom-Xbut I put -apache2 so its not supposed to instal
01:46.56bkw_if you see -apache2 then you're good to go
01:47.04bkw_show me the full output of emerge -pv asterisk
01:47.08icebalmyou'll also want -apache
01:47.16bkw_put it on pastebin.ca
01:47.27bkw_I personally do not use portage for anythign asterisk related
01:47.40bkw_mpg123 from portage is fucked
01:48.15icebalmuh, howso?
01:48.20Phantom-Xhttp://pastebin.ca/1324
01:48.37Phantom-Xive never had any probs with portage
01:48.55Phantom-Xuntil now with asterisk , im certain that the ebouild is wrong built
01:49.01bkw_your portage is smokin crack
01:49.14bkw_icebalm anything but 0.59r is broken
01:49.20Phantom-Xbkw_, blame the person who made the ebould for astersik
01:49.20bkw_it gives strange results
01:49.23Phantom-XI could check
01:49.37bkw_my emerge sync from today is doing 0.9.0
01:49.40bkw_odd
01:49.46bkw_the 1.0 ebuild is fucked up then
01:49.54Phantom-Xbkw_, change sync server
01:49.55brc_suprise!
01:49.58brc_it's gentoo
01:50.10bkw_gentoo on a server ROCKS
01:50.10DaminDiscovery has a 1 hour show on SpaceShip One.
01:50.12brc_use debian
01:50.15DaminGentoo is lame.
01:50.22DaminGentoo should be banned.
01:50.23brc_in debian we had 1.0 THREE MONTHS AGO!
01:50.25brc_beat that!
01:50.30icebalmbkw_: it's not fucked up, it's masked, geez, you have a great tendancy to say things are fucked up or broken when they really arent
01:50.40brc_Damin, when is that on?
01:50.41brc_now?
01:50.42bkw_icebalm fuck you
01:50.51Daminicebalm: He is a drama queen. :)
01:50.52brc_calm down bkw_
01:50.55DaminYeah.. it's on now..
01:50.57bkw_brc_ fuck you
01:51.02DaminLast ten minutes..
01:51.05icebalmDamin: so it would seem
01:51.33DaminIt's called "Black Sky"
01:51.37bkw_icebalm no I just don't pay much attention when its not my problem
01:51.40bkw_and thats the truth
01:51.40DaminLots of awesome footage..
01:52.02icebalmbkw_: fair enough, but you shouldn't say things are "fucked up" when they are working as intended, it's FUD
01:52.14bkw_0.59s is fucked
01:52.16bkw_i know that for sure
01:52.34*** join/#asterisk denon (denon@synapse.subneural.net)
01:52.34*** mode/#asterisk [+o denon] by ChanServ
01:53.11brc_thanks Damin ...Scientific Atlanta'ing it
01:53.30brc_oh...it's on at 7pm here
01:53.38icebalmPhantom-X: the only thing I can think of is that the asterisk ebuild does make mention of a voicemail webapp, but it doesn't (or shouldnt) get installed if you dont have a webserver
01:54.06bkw_icebalm that was my first thought
01:54.18bkw_but I dont have the ebuild yet
01:54.19bkw_so I can't see
01:54.34brc_the voicemail 'webapp' is called 'webvmail'
01:54.44brc_if you have the asterisk source you can 'make webvmail' iirc
01:54.50Phantom-Xicebalm, right , and I have no webserver or ever will install one such since I hate them =)
01:55.31bkw_Phantom-X this is why I don't use ebuilds for asterisk
01:55.31bkw_they have been so messed up since day one
01:55.32file[laptop]or... day -1!
01:55.39DaminWhat is so hard about typing "make"
01:55.46Phantom-Xbkw_, im not listening to that
01:56.06bkw_Phantom-X good luck
01:56.15bkw_trust me compile it yourself
01:56.18Phantom-Xbkw_, I will make it =)
01:56.19bkw_if youw ant packages use Debian
01:56.35Phantom-Xdebian is crap , using your own words =)
01:56.39bkw_yep
01:56.40DaminIf you want problems, use Gentoo! :)
01:56.43bkw_I totally agree
01:56.47bkw_doh
01:56.48bkw_you ass
01:56.50Phantom-Xgentoo is fine
01:56.52bkw_No gentoo isn't a problem
01:56.58Phantom-Xbeen using if for many years
01:57.02bkw_now whats fun is when it fucks up your gcc-config during emerge
01:57.14bkw_now thats a fun one when it happens to ya for the first time
01:57.30Phantom-Xbkw_, are you rally making sense ?
01:57.51Phantom-Xbkw_, why use a computer in the first pace then ?
01:58.42bkw_um
01:58.47bkw_wtf did I do to you?
01:58.48mishehubah, hylafax is pissing me off.
01:58.53bkw_hylafax is easy
01:58.56file[laptop]9..9..6!
01:59.00DaminI prefer using an Abacus myself..
01:59.20*** join/#asterisk paulc (~paulc@S010600062586a0b4.vc.shawcable.net)
01:59.34file[laptop]paulc: 9..9..6!
02:00.06mishehubkw_: I constantly get "Waiting for modem to come free", my modem is on ttyS1, and faxgetty launches on ttyS1...
02:00.36mishehunothing ever changes, and I've tried multiple configurations.  the linksys pap2 has nothing to do with it.
02:01.06icebalmnah, what's fun is when you apt-get dist-upgrade and then you go to login and your keyboard isn't inputting anything because some idiot debian package manager compiled /bin/login without local console support
02:01.49*** join/#asterisk Mike (~mike@201.135.48.52)
02:02.25mishehuin fact, I got the pap2 to be able to test out hylafax.
02:02.29*** join/#asterisk intrin (~fasdfa@c66.188.173.60.stc.mn.charter.com)
02:02.47intrinpaulc!
02:02.48intrin:D
02:03.01paulcevening all..
02:03.04*** join/#asterisk PhilM (~a@r42h98.res.gatech.edu)
02:03.09intrinyou just get here too?
02:03.21paulcyeah, few mins ago
02:03.24intrincool
02:03.26intrinperfect timing
02:03.40intrinman, watched cabin fever
02:03.43intrinboy was that gay
02:04.02bkw_haha
02:04.11mishehubkw_: you able to help at all?
02:04.13JerJerPower over Wireless Ethernet
02:04.19bkw_mishehu does the modem work from minicom?
02:04.25mishehubkw_: *nod*
02:04.37mishehuit's a USR ISA modem
02:04.41*** join/#asterisk invi_ (~undisclos@dsl-cap-209-5-169-185-cgy.nucleus.com)
02:04.46bkw_evil
02:04.47file[laptop]JerJer: power over bagel over wifi over toast over dingbats
02:04.56bkw_mishehu not sure never used an ISA modem wif it
02:05.14mishehubkw_: I thought winmodems were evil, not per-say ISA hardware modems.  ;-)
02:05.16invi_is this the only way of doing Call-Waiting flash > http://lists.digium.com/pipermail/asterisk-users/2004-September/060960.html
02:05.30bkw_mishehu I used a 48 port Patton 2977 card
02:05.47mishehubkw_: appearantly not
02:05.48bkw_48 modems on that bad boy
02:05.49mishehuheh
02:06.00mishehulittle warfaxing going on there?
02:06.00mishehuhehe
02:06.05bkw_just a bit
02:06.08bkw_:)
02:06.12*** join/#asterisk Stealth_Man (~no@ool-18bc203f.dyn.optonline.net)
02:06.13bkw_FAXBLAST
02:06.26bkw_I got tired of waiting on app_rxfax/txfax
02:06.35bkw_and 9600bps is for the birds
02:06.39mishehuwell, i'd put a PCI USR hardware modem on there, but I have no free PCI slot
02:06.45bkw_I have one of those too
02:06.47bkw_that works great
02:06.52ManxPowerAlcohol and Calculus don't mix. Don't drink and derive.
02:07.00bkw_OLD
02:07.17mishehuand my X101P is sharing interrupts with the usb controller already
02:07.45mishehuoh and I don't have a serial cable for my external USR modem
02:07.46florzWhat's the state of app_*fax
02:07.48florzBTW?
02:07.50florzgnah
02:07.54bkw_florz CRAP 1.0
02:07.55mishehugnaw
02:09.01florzbkw_: Hmm ... "it might work under unknown circumstances"?
02:09.22ManxPowerflorz: .01 had problems sending to Canon fax machines.
02:09.53mishehuI wonder when the SHIT 2.0 version will be out.
02:10.01florzManxPower: That sounds more like "works most of the time but is not stable"!?
02:10.03bkw_dont know but I don't have tow orry with it
02:10.13bkw_app_txfax_and_segfault.c
02:10.43jgaviriaim installing zapata in debian with kernel 2.4.26 but i have unresolved symbols when i tried to load zapata module... somebody have an idea.. somebody could helpme?
02:10.50florzhow about receiving faxes?
02:11.07*** join/#asterisk Wi_Fi (~OUT@c-66-229-160-14.we.client2.attbi.com)
02:11.35jgaviriai would say zaptel
02:11.42ManxPower.02 seems to do better
02:11.47Wi_Fiheyaz
02:12.10intrin~google sip.conf.sample
02:12.13Stealth_Man~showtime
02:12.35Stealth_Man~seen showtime
02:12.36jbotshowtime <show@65.171.196.4> was last seen on IRC in channel #asterisk, 5d 1h 27m 43s ago, saying: 'dumb proxy and registrar'.
02:12.42florzManxPower: txfax -> Canon, you mean?
02:12.56ManxPowerflorz: Yes.
02:12.58bkw_BIG IS BAD
02:13.01Wi_Fi~google iax.conf.sample
02:13.09bkw_Does size matter?
02:13.14Stealth_Man~google asterisk forums
02:13.18bkw_and how far would you go to change the lenght or size of your goodies?
02:13.21ManxPowerUseful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/
02:13.46florzWell, actually receiving faxes would be more important for now, as it could save some paper =:-)
02:14.43Phantom-Xok I managed to make sense with the emerge =)
02:15.09Wi_Fii need help gettin iaxy in dialplan
02:15.15*** join/#asterisk Moc (~mochouina@modemcable021.49-80-70.mc.videotron.ca)
02:15.20bkw_Dial(IAX/iaxy)
02:15.22bkw_NEXT!!!
02:15.45*** join/#asterisk lattera (~lattera@dialup-4.228.195.82.Dial1.Denver1.Level3.net)
02:16.00Wi_Fimake it default ring from zap incoming
02:16.17latterais there a list of asterisk-compatible VoIP clients anywhere?
02:17.29Wi_Filattera try firefly
02:17.35Wi_Fivery good with iax
02:17.41tzangerWi_Fi: I can't get it to work
02:17.44latteraand, will VoIP work well with dialup?
02:17.55lattera(I'm quite the newb when it comes to VoIP)
02:18.02Wi_Fitzanger ???
02:18.25*** join/#asterisk xpasha (~pavel@217.30.252.68)
02:18.30tzangerWi_Fi: with identical settings, I get ringback with IAX but no audio once connected... and changeing from iax to sip I get no ringback but audio
02:18.50tzangerI use gsm codec (and can verify that if I change codecs, * complains, as I only allow gsm with iax and sip)
02:18.53mishehuhaha that's whacked
02:18.56tzangerxlite works just fine
02:19.19xpashaanybody alive who worked with E1/T1 cards?
02:19.22tzangermishehu: tell me about it :-)
02:19.24tzangerxpasha: yup
02:19.25tzangerT1
02:19.28*** join/#asterisk afrosheen (~afro@c-67-166-162-197.client.comcast.net)
02:19.29tzangerT100P and TE405P
02:20.06xpashatzanger so may be you know what is to be done in order to make E1 card being a clock source for remote device?
02:20.10*** join/#asterisk Kb1_Kanob (johnsmith@sec2d47.dial.uniserve.ca)
02:20.17tzangerxpasha: it won't
02:20.20tzangerwell it can
02:20.29xpashahow?
02:20.32tzangerbut it'll clock off the internal timer which I wouldn't trust with a 10 foot pole
02:20.46tzangerwell in any link you have a clock source
02:21.02xpashaI have cisco 5350 and it can't generate clock signal
02:21.02*** join/#asterisk voiper (~none@pcp09528872pcs.eatntn01.nj.comcast.net)
02:21.07tzangerif the card is not syncing to the other side, then it's running free
02:21.20tzangeryeah the 5350 won't clock you're right
02:21.33xpashaso how to force it to generate signal?
02:21.34tzangerso tell the digium card not to sync from the line
02:21.41tzangeryou don't you jus ttell it not to sync ot hte line
02:21.42xpashawhat option?
02:21.44*** join/#asterisk telme (~non@c-24-8-57-124.client.comcast.net)
02:21.47tzangerspan=
02:21.58tzangerread the zaptel.conf it's listed in there
02:21.59xpasha?
02:22.04tzangeryou just say 0 = don't use the line for clock sync
02:22.12Kb1_Kanobspan=1,0,0 to push timing, 1,1,0 to pull timing.
02:22.18tzangerum
02:22.18xpashahmmm
02:22.18tzangerno
02:22.22tzangerKb1_Kanob: that is not true
02:22.26xpashaso
02:22.30xpashalisten to me guys
02:22.42xpashaI have done  span=1,0,0  and  span=1,1,0
02:22.53xpashaand anyway
02:23.17xpashaIt wroked well with remote PBX I connected
02:23.18Kb1_Kanob0 for push and 1 for pull, according to digium support on t100p.
02:23.19tzangerthe card has an internal clock; you can have it try to sync to incoming line or not.  span=1,0,0 says not to try and sync, 1,1,0 says to try and use recovered clock as primary sync src
02:23.44voiperdoes any know about this message ?¿ "ast_rtp_read: Unknown RTP codec 72 received"
02:23.45xpashaI have PRI trunk from Alcatel S12 PBX
02:23.48tzangerthere really is no "pushing" of the clock signal, only an attempt to lock ot the recoverd clock
02:23.50xpashaand it generates clock
02:23.55tzangerxpasha: is this a 4-span card?
02:23.57xpashaand anyway
02:24.04xpashait worked well
02:24.09xpashawith my E100P
02:24.14Kb1_Kanobtzanger: I take it you're quite familiar w/internals - can you answer some bus questions?
02:24.18tzangerxpasha: everything generates clock, don't think of it that way
02:24.26tzangerKb1_Kanob: might be able to :-)
02:24.26xpashaif I set span=1,0,0 it worked
02:24.31xpashaif I set span=1,1,0 it worked too
02:24.35tzangerxpasha: stop
02:24.38xpashawho to explain it?
02:24.38latteraare there any test PBXs available?
02:24.38tzangerxpasha: listen to me
02:24.52tzangerxpasha: all T1/E1 devices self-clock.. ALL of them
02:24.56tzangerthe 5350 will
02:25.20tzangerxpasha: the problem is that if you have two sides both clocking to different sources (i.e. themselves) you will get frame slips
02:25.32florzlattera: Depending on what you want to test, I'd say: Just install Asterisk.
02:25.33Kb1_Kanobtzanger: relative clock drift.
02:25.34tzangerso you designate one side to lock on to the other
02:25.47*** join/#asterisk monzsca (~monzsca@monz.marmoset.net)
02:25.56tzangersince the 5350 demands that you lock on to the other side, you tell * not to try and lock on to the other side
02:26.03tzangerand therefore you use a span line of 1,0
02:26.04voiperdoes anyone know about this message "ast_rtp_read: Unknown RTP codec 72 received"
02:26.16tzanger(span 1, don't try to lock on to recovered clock)
02:26.30tzangervoiper: not me, I don't use RTP unless I can't help it
02:26.37xpashatzanger so how to explain that my E100P worked well anyway if I set 1,0,0  or 1,1,0 ??
02:26.47tzangerxpasha: because the clocks are pretty much in sync
02:26.58latteraflorz, I would, but I want to test to see how well it would (or would not) work on dialup
02:27.16tzangeryou won't notice desync'd hardware until frame slips happen and if the clocks are worth anything at all that won't happen very often
02:27.23tzangerbut when it odes happen you'll hear a buzz or chirp
02:27.26xpashaI meant worked with PBX that generates signali itself
02:27.32tzangerxpasha: yes
02:27.35Wi_Fibkw_ would that be IAX2/iaxy?
02:27.42tzangerxpasha: the one end is going
02:27.54xpashatzanger but if I set 1,0,0 it have to generate clock
02:28.00tzanger__________----------_________----------__________----------__________
02:28.02tzangerand the other side is going
02:28.08voiperthanks tzanger
02:28.15tzanger__________----------__________----------__________---------__________
02:28.17xpashaand I would get no working
02:28.19tzangeri.e. ALMOST the same
02:28.28tzangerbut you can see there it's starting to slip
02:28.37xpashahmmm
02:28.40tzangerwhen you get enough of a difference you'll slip a frame
02:28.47tzangerxpasha: THEY ALL GENERATE CLOCK
02:28.50xpasha:)
02:28.51tzangerplease understand that
02:28.57tzangerthey ALL generate their own clock signal
02:29.06xpashatzanger okok :)
02:29.15tzangerwhat you are saying with the span command is wiehterh to try and lock on their own internal clock ot the recovered clock from the line
02:29.31Kb1_Kanobtzanger: hence 'master'/'slave' terminology rather than 'push'/'pull'?
02:29.37tzangerthe 5350 you have no choice, you have to say whether the incoming line is clock primary or clock secondary (or teritary or quaternary)
02:29.42tzangerKb1_Kanob: kind of yeah
02:29.54xpashatzanger did you try span=1,0,0 with 5350?
02:29.58tzangerall you're doing is telling the driver whether to try and use the recovered clock as a sync source
02:30.02xpashagot it working?
02:30.13tzangerxpasha: I don't have a 5350, I have a 5248 and an ancient TNT
02:30.18tzangerer not TNT but MAX
02:30.21tzangerand it works just fine
02:30.38xpashathe same situation it can generate clock signal?
02:30.38tzangerI have a TE405P.  span 1 is a PRI from Bell Canada.  I say to lock on to that
02:30.44xpashait can't I meant
02:31.01tzangerspan 2 I have connected to the Ascend MAX.  I tell it NOT to sync there, and I tell the MAX ot sync to the line (which is span 2 on *)
02:31.13xpashahmmm
02:31.14xpashaok
02:31.15xpasha:)
02:31.20tzangerspan 3 and 4 go to an Adit600 and again, I tell * NOT to sync to them, but I tell the Adit600 to sync to those lines
02:31.29bkw_klasjdfowkdjfoiejf
02:31.39xpashaI will make cross over and try to get it working with Cisco 5350
02:32.01xpashathanks tzanger :)
02:32.04tzangerevery T1/E1/J1 device generates clock...  all you are doing is specifying whether the device should try and lock its clock to the incoming recovered clock
02:32.25Kb1_Kanobtzanger: so, I understand that the clock on the digium hardware is implemented as a high resoloution hardware timer _but_ if it's the 'master' end then it's succeptible to causing slips when the PCI bus gets contention. However,
02:32.26tzangerunless there are SERIOUS issues you will always sync up, but if one side is not trying ot match the other you will eventually get frame slips
02:32.34tzangerno
02:32.40Kb1_Kanobif it's hardware then it's not the clocking that's slipping, but a buffer underrun?
02:32.47tzangerthe digium timer has nothing to do with the PCI bus
02:32.54xpashaso what is the option to be a primary clock source, secondary etc?
02:33.05tzangerthe digium timer is on-card and it can either free-run or try and lock to a recovered clock
02:33.06xpashato be source for what?
02:33.10Kb1_Kanobtzanger: I'll let you finish w/xpasha first. sorry.
02:33.30tzangerxpasha: exactly what it says -- you tell the 5350 how to sync to the incoming T1s
02:33.49xpashaso what is primary and secondary then?
02:33.50tzangeryou can say controller 1's recovered clock is your primary clock source (or secondary/tertiary/quaternary) and so on
02:33.56*** join/#asterisk infinii (~wayland@host661461427f.dsl.res.tor.fcibroadband.com)
02:34.03tzangerprimary = if I have multiple incoming clocks, use this one first.  it's the best
02:34.13tzangersecondary = if my primary dies goes away, use this one
02:34.19tzangertertiary = if my secondary dies, use this one
02:34.25tzangerquaternary = if my tertiary dies, use this one
02:34.25intrin~google x-lite
02:34.34tzangerjust the same as teh span= lines in *...
02:34.38tzanger0 = do not sync to this span's clock
02:34.50tzanger1 = use this span's clock as the primary clock sync source
02:34.58tzanger2 = use this span's clock if the primary clock source is gone
02:35.08tzanger3 = use this span's clock if the secondary clock source is gone
02:35.14infiniigood evening
02:35.19tzanger4 = use this span's clock if the tertiary clock's source is gone
02:35.27tzangerpretty straightforward
02:35.35therealadept"You have entered a number that cannot be reached from your calling area"  :/
02:35.36tzangerconfusing at first, but pretty straightforward after :-)
02:35.36xpashaOk understand this :)
02:35.39xpashatnx :)
02:35.40tzangerso
02:35.55tzangersince the 5350 DEMANDS to sync ot something, yuou tell * NOT to sync to the 5350
02:35.56xpashatzanger what area are you from?
02:35.59infiniiAnyone using the digitnetworks x100p fxo card? seems really affordable
02:36.00tzangerxpasha: canada
02:36.06xpashawhat province I meant
02:36.11tzangerontario
02:36.14xpashaah ok
02:36.20tzangerabout an hour and a half from Toronto
02:36.27xpashaI have a relatives in Quebec
02:36.48xpashamay be visit canada soon :)
02:36.58tzangerxpasha: where are you?
02:37.05xpashaI'm from Russia
02:37.12tzangerright on
02:37.18tzangerone of the places I want to visit is Russia
02:37.19xpashabut work for Czech company
02:37.33xpashaand visit Prague frequently
02:37.35Wi_Fican somebody splain me on adding iaxy to dialplan
02:38.05tzangerPrague is another place
02:38.06xpashatzanger Russia is not so wild as most ppl in the world can imagine :)
02:38.11tzangerI dunno I like those kinds of places
02:38.12tzanger:-)
02:38.21tzangerno I don't want to visit it because it is wild
02:38.27xpashahehe $)
02:38.31tzangerI like the architecture, I like the "old world" I think
02:38.32ariel_Wi_Fi, what is your question?
02:38.42Kb1_Kanobtzanger: may I continue clocking discussion w/you?
02:38.48tzangercertainly
02:38.56xpashatzanger btw Russia is the same nationalities mix as USA and Canada
02:39.04Wi_Fiive added it but i cant connect to it from other phones
02:39.08xpashaI live in the place Tatarstan
02:39.20xpashahere are ppl tatars
02:39.20tzangerxpasha: right on
02:39.22tzanger:-)
02:39.25xpashathey are muslims
02:39.26therealadepthas anybody been having nufone problems recently?  some toll free calls haven't been going through and my callerid is always showing up as unavailable :/
02:39.29Kb1_KanobOk, so if clock on t100p is hardware then that explains why I get no BPVs on a crossover link, but I do get audio defects during long DMA transfers - because of buffer starvation on the card?
02:39.30tzangerI want to see Germany, Russia, the Ukraine
02:39.45therealadeptseems like its been happening for the last few days
02:39.49xpashamy father is ukrainian :)
02:39.52xpashabtw
02:39.53ariel_for real wild things go to Thialand...
02:39.53tzangerKb1_Kanob: you likely get audio defects because you're taxing the bus
02:40.01tzangerare you getting BPV's on DMA transfers?
02:40.04infiniiI'm going to bangcock in Dec
02:40.09tzangeror is that a red herring
02:40.10infiniibut first japan
02:40.16ariel_Wi_Fi, what error are you getting.
02:40.18Kb1_Kanobno BPVs or alarms detected.
02:40.31tzangerKb1_Kanob: ok
02:40.37Kb1_Kanobbut some occasional audio defects when master clock is t100p.
02:40.48Kb1_Kanobvoicemail is running on the host server.
02:40.52ariel_infinii, if you want when you get there go to pattia beach it's better.
02:41.08tzangeryour problem is likely bus exhaustion -- try reducing the PCI latency timer for all your PCI devices, and then giving the T100P a higher latency timer value
02:41.16Kb1_Kanobalso getting intermittant timing issues w/RBS signalling on the t1.
02:41.18tzangeri.e. give it the ability to preempt the other devices on the bus
02:41.19infiniiThe extensions.conf that comes with * is kinda full of commented stuff, did you guys simply start from an empty file and build it up to your liking? this std one is hard to read with so much in it
02:41.31Kb1_Kanobtzanger: setpci?
02:41.35Wi_Fiariel_
02:41.37infiniiariel_: pattaya is overriden with tourists and isnt as clean as it used to be
02:41.38tzangerunless you're getting frame slips clocking isn't the problem
02:41.39Wi_Fihttp://www.pastebin.com/108294
02:41.44tzangerKb1_Kanob: yes
02:42.04infiniiariel_: phuket in thailand has nicer beaches
02:42.09Kb1_Kanobthat makes more sense - your description this evening was extremely good. It belongs in zaptel.conf.
02:42.31tzangerKb1_Kanob: :-)
02:42.36tzangerI just checked my own system
02:42.43tzangerI have the latency timer set to 64 across the board
02:42.50tzangerI thought I had the TE405 set a little higher
02:42.50latteraso, asterisk doesn't run on FreeBSD?
02:43.03tzangerlattera: it does, I don't think the digium hardware drivers have been ported
02:43.15Kb1_KanobOn a related note - some services in * consume zaptel clock for timing (eg. signalling). Is it an absoloute timer read or relative? Ie. if there was a bus delay would the signal run on?
02:43.21florzlattera: *lag* That depends mostly on the codec you are using - which you easiest can test with asterisk's echo application, I think. The other "limiting factor" on dialup (56 k modem?) is latency, which is acceptable ...
02:43.43tzangerKb1_Kanob: http://www.reric.net/linux/pci_latency.html
02:43.53Kb1_KanobPut another way, could a 200ms wink stretch out to 250ms?
02:44.07florzlattera: (no, not whether it runs on FreeBSD, but voice quality =:-)
02:44.14Kb1_Kanobor an incoming 200ms be read as a 150ms?
02:44.18tzangerKb1_Kanob: well nothing "consumes" the timer -- it's just a free running timer that can be optionally locked to a recovered clock
02:44.30tzangerKb1_Kanob: and what the software uses the timer for is a 1000Hz interrupt
02:44.47*** join/#asterisk enmaca (~emartinez@201.129.118.69)
02:45.00Kb1_Kanobahhh.... and it has a tight timing routine tucked away somewhere...
02:45.16tzangerthat is why digium hardware is so picky about having its own IRQ -- if multiple devices have the same IRQ the system must check each driver to see which one the IRQ is really for, and some drivers are really shitty about not being consistent in the time they take to service the interrupt
02:45.24tzangerand jitter is the #1 killer of VOIP quality
02:45.35tzangerwhether that jitter is network or system doesn't matter
02:45.41Kb1_KanobRight, that also makes sense.
02:46.04Kb1_Kanobwould kernel 2.6 provide better service to zaptel hardware?
02:46.16tzangerso for MOH timing and whatnot it is expecting that 1ms interrupt to occur with precision
02:46.24tzangerKb1_Kanob: not sure, I don't use 2.6 for *
02:46.52Kb1_KanobSo it's just a trusted interrupt source, not a rolling counter in hardware.
02:46.59tzangernow SIP uses the incoming RTP stream to time (I call it self-timing) -- every time an RTP packet comes in the RTP clock advances...  that's why VAD kills it at this time
02:47.13tzangerKb1_Kanob: the zaptel hardware has this timer running and generating IRQs
02:47.41tzangerif the system doesn't respond to the IRQs in a timely fashion doesn't make any difference to the timer, it just keeps slapping the IRQ line
02:48.01Kb1_KanobHence setting the latency and priority would seem critical.
02:48.24tzangerI've seen that RTP (and IAX fromthe looks of it) timing is coming from an internal software timer now, or at least that is in the works -- the CVS logs showed some of that about a month back
02:48.52Kb1_Kanobthis also explains the trunking/jitterbuffer incompatibility.
02:48.55tzangerso I believe that VAD and packet loss recovery will soon become features in * :-)
02:49.07Kb1_Kanobrecovery = concealment?
02:49.13tzangersorry yes concealment
02:49.29tzangeri.e. g729 and iLBC's PLC will become functional
02:49.42Kb1_KanobThis is fascinating - I have been seeking someone with understanding.
02:49.50tzangerright now they just never get called to do those functions because if packet doesn't come in, it doesn't get slapped to fake it
02:49.58tzangerKb1_Kanob: I'm at the bottom end of the understanding
02:50.08Kb1_Kanobbut you share. :-)
02:50.14tzangerkram or bkw or citats or coppice... those guys are really in the know
02:50.38Kb1_Kanobhowever, few seem to have the few minutes it takes at the moment.
02:50.42Wi_Fithey should be heading back home now
02:50.43Wi_Fihehehe
02:51.07Kb1_Kanobtzanger: how are you with understanding the zap echo can?
02:51.18tzangerhaha
02:51.23tzangerthat is black juju, my friend
02:51.29Wi_Fianyone in california?
02:51.32tzangerI don't think even the people who wrote it fully understand it
02:51.34tzanger:-)
02:51.55Kb1_KanobI've been tring to understand where the taps occur and such. Been having a horrible local echo from my hybrid.
02:52.14tzangerit's kind of like the first Terminator -- the guys who built the skynet processors were given funky technology and told not to ask where it came from, but it gave them ideas and took them in directions they hadn't thought of :-)
02:52.19Kb1_KanobGot an impedance check today from telco. Am getting 100ohm into a nominal 600ohm interface.
02:52.23Kb1_KanobNot nice.
02:52.29tzangerKb1_Kanob: is it balanced at least?
02:52.42tzangertake your ohmmeter and measure from tip to ground and ring to ground
02:53.03Kb1_KanobHmmm... I will check, however should be - have 11 trunks.
02:53.11Kb1_KanobI am less than 500m from serving CO.
02:53.12sudhir492Can anyone tell me why would registration fail in a SIP device, but the phone can still make calls through the server
02:53.13tzangeryou can "balance" it by adding resistance to the low leg
02:53.22tzangeryou're less than 500m and your impedance is fucked?
02:53.29Kb1_Kanobbottom of the scale.
02:53.33tzangerheh
02:53.47tzangerwith 11 lines you're almost worth going to a PRI or cT1
02:54.01Kb1_KanobIf I turn of the can the sidetine is horrible - almost as loud coming back as going out.
02:54.06tzangerjust tell me you're using a channel bank :-)
02:54.07Kb1_KanobPRI on order.
02:54.28Kb1_Kanoboh yeah - but it's got op amp hybrid and I can't twiddle to match down to 100ohm.
02:54.30tzangerand not 3 TDM404Ps
02:54.44Kb1_Kanoban old Newbridge Mainstreet 3624
02:54.50tzangerhmm not familliar with that one
02:54.53tzangerfunny thing is
02:54.55tzangerI am on PRI
02:54.57tzangerand I still get echo
02:54.58Kb1_Kanobmid 90s modular.
02:55.13tzangermy voice is bouncing off the far-end's hybrid and reflecting
02:55.15Kb1_Kanoboh? telco mismatch on their indermediate frame?
02:55.22tzangerbut only for some numbers
02:55.27Kb1_KanobRight, but it's fairly attenuated?
02:55.41Kb1_Kanob(if can is off?)
02:55.45tzangermost numbers are fine... but some aren't and it's not switch related
02:55.51tzangerno it's pretty loud actually :-)
02:55.58Kb1_KanobHmmm.... I had a report of one of those today.
02:56.00tzangersome it's barely noticeable but ohters it's distracting
02:56.12*** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
02:56.20tzangerI've got the echo can on and set to 128 taps, played around with that a lot it's just not working right
02:56.23BrixiusHello
02:56.27Kb1_KanobI'm assuming the can is synced to the near end sidetone and the far end echo was being ignored.
02:56.43tzangerprobably, I can't say for sure
02:57.00tzangerI was going to buy a tellabs hardware 64ms echo can but I'm going ot try a few other things first
02:57.11Kb1_KanobI can dial down to 32 taps w/o training and get 100% supression when I dial quiet termination test number, but in practice I need at least 128 w/ training=800
02:57.25file[laptop]ARE YOU ON THE INTERWEB???!!!
02:57.49Kb1_Kanobtaps are in 1 tap per sample, and one sample on t1 is 1/8000 of second, right?
02:58.33tzangerthat's a good question I'm not sure but that certainly makes sense
02:58.36Kb1_Kanobso what are the units of the 'training' factor?
02:58.42tzangerKb1_Kanob: taps :-)
02:58.55Kb1_KanobHmmm....
02:59.21tzangerbut yeah in a digital filter taps are usually discrete units of samples
02:59.29BrixiusI just got a 7960 to replace a barbietone, that's like night and day
02:59.33Wi_Figuys is it possible to have an iaxy be the default ring on incoming zap channel
02:59.40tzangerBrixius: yeah?
02:59.40Kb1_Kanobalso makes sense, but training 'mutes' the channel to get the initial delay factor - however it can only mute the near end outbound, so it'll really try to compensate for far end echo initially.?
03:00.13tzangerKb1_Kanob: it doesn't so much mute -- it mutes you off and sends an impulse down the line and measures the response to try and quickly train the echo canceller
03:00.17tzangeryou can kill that by setting echotraining=off
03:00.23tzanger(or some # of ms)
03:00.27tzangerI think it defaults ot 800
03:01.11Kb1_Kanobyes, 800 default. So it's really geared to supressing far end echo rather than excessive sidetone.
03:01.53infiniican someone show me their /etc/modprobe.d/zaptel file? I get errors on bootup
03:01.56*** part/#asterisk invi_ (~undisclos@dsl-cap-209-5-169-185-cgy.nucleus.com)
03:02.09Kb1_KanobSidetone would have a delay of 1 or 2ms.
03:02.23tzangerKb1_Kanob: that seems more of a txgain issue than echo
03:02.28tzangerbut I really despise playing with gains
03:02.49tzangersidetone = sounds like intermod in the old ham radio days?
03:02.54Kb1_Kanobyes - I've attenuated down at CB to help the can, but the sidetone is wicked loud.
03:03.14Kb1_KanobYes - perhaps we could squeeze 3 channels into 1 that way!
03:03.19tzangerhehehe
03:04.15Kb1_KanobAny idea what the algorithmic implications of 'AGRESSIVE' supression are?
03:04.30Kb1_KanobI understand it tends to mute higher frequencys.
03:04.47tzangernot sure there, youd' have to piss around in the source
03:05.02Kb1_KanobOk.
03:05.27Kb1_KanobThanks - I really appreciate your time discussing this.
03:07.32_rcsin sip.conf, what entry to you create so that when you refer to a fone from extensions.conf, you refer to it as a number and not an IP. eg 2000
03:07.36Phantom-Xhttp://bugs.gentoo.org/show_bug.cgi?id=66720
03:07.40_rcslike with SIP/2000
03:08.47*** join/#asterisk NormAst (NormAst@Toronto-HSE-ppp3677931.sympatico.ca)
03:09.31NormAstf
03:12.56Brixiusthere is really good documentation as well as samples in the default sip.conf file for that, or on the wiki it's at http://www.voip-info.org/wiki-Asterisk+config+sip.conf
03:13.51Kb1_Kanobtzanger: another question if I may - how much drive should a 0db signal (ie. a milliwatt test tone) represent on the zap channel? ztmonitor provides a notional representation of levels for conversation, but I'm unclear what to expect from a 0db source.
03:15.30tzangerthat's actually a REALLY good question
03:15.34tzangerI wish I had a good answer for you
03:15.42tzangerI've only ever eyeballed it
03:15.47tzangerer... earballed I guess is the term
03:15.52Kb1_KanobSame here.
03:16.29Kb1_KanobIn particular I noted that when dialed in to a milliwatt source provided by asterisk the zap gain adjust didn't seem to bring it down in ztmonitor but it did acoustically.
03:16.39tzangerhmm
03:16.46tzangermaybe the ztmonitor output is linear
03:17.11Kb1_KanobOr the gain is inside the card and ztmonitor taps the bitstream.
03:17.19*** join/#asterisk Mike (~mike@201.135.48.52)
03:17.19tzangeryes that'd do it too
03:17.21Wi_Fican IVR be interupted?
03:19.47pfnwhat do you mean by interrupted
03:20.37Wi_Fisuch as a phone that has PA when incoming calls ring and would like to listen to person calling then pick up in middle of leaving msg
03:21.16Kb1_KanobWi_Fi: are you describing 'caller screening' or 'recall from voicemail'?
03:21.26Wi_Ficaller screening
03:21.27Wi_Fiyes
03:21.30Wi_Fithats the ticket
03:21.44*** join/#asterisk lattera (~lattera@dialup-4.228.195.121.Dial1.Denver1.Level3.net)
03:21.55latteraanyone ever used asterisk on FreeBSD?
03:21.56tzangeranyway
03:21.57tzangernight all
03:22.01Wi_Fitryin to make this transition as smooth as possible
03:22.02florzAny clue whether it is possible to find out the MSNs of a EuroISDN line using only the line and a computer connected to it?
03:22.06Kb1_Kanobtzanger: gnight & thanks again.
03:22.12tzangernp glad to help where I can
03:22.18Wi_Finite tzanger
03:23.27*** join/#asterisk andy_b (~me@208-186-220-90.nrp4.brv.mn.frontiernet.net)
03:23.43Wi_FiKb1_Kanob is caller screening possible
03:23.55latteraanyone ever used asterisk on FreeBSD?
03:23.57*** join/#asterisk chap (chapster@woodstock.cleburne.com)
03:24.14andy_bhello all - is anyone interested in helping a newbie tonight?  (yeah yeah, I hear the groans...=))
03:24.24Wi_Figood
03:24.24Wi_Fihehehe
03:24.35Kb1_KanobWi_Fi: honestly, I don't know. It could probably be done with zapbarge() if they were coming in over a zap channel.
03:24.36Wi_Ficaller screening on asterisk
03:24.38Poemiuslattera : I tried using it 3 months ago, it did not compile, I hear they made progress since then, not sure
03:24.41chapChannel rules prohibit helping newbies.
03:24.47andy_bsweet
03:24.49andy_b=)
03:25.00latterais zaptel _required_?
03:25.09chap:)
03:25.10latterabecause zaptel does not compile
03:25.22florzchap: Can't you see that telling that a newbie helps him?
03:25.40Wi_Fithey are comming over zap channel
03:25.58Poemiuslattera: you may need ztdummy for some functions, for music on hold to work
03:26.02chapflorz: I see that now.. Damn my incompetence!
03:26.04andy_bI stumbled onto asterisk by accident actually - looks like a GREAT solution....I want to try it out - have it installed on a RedHat 9 box, but would like some guidance on initial setup...?
03:26.10florzchap: *gg*
03:26.32Kb1_KanobWi_Fi: yeah, try getting creative with zapbarge and the dialplan.
03:26.40Wi_Fiyikes
03:26.49Wi_Fino guru here
03:26.50Wi_Fihehe
03:26.54andy_b;)
03:26.54Kb1_Kanobmyself, I need 'recall transfered call' and/or 'recall from voicemail' in a PRI network.
03:28.12andy_bmy goal for testing the basic functionality of this puppy out is to hook it up in my office - one analog phone line, one server, one handset
03:28.24Connor_~seen kram
03:28.25jbotkram is currently on #asterisk.  Has said a total of 4 messages.  Is idling for 5h 57m 20s
03:28.41chapandy_b: read, read, read. It is pretty cool stuff, I have only been messing with it for a few months- so my newbiew advice is to read. Don't forget about the wikki.
03:28.49andy_bwikki?
03:29.06andy_bI have the handbook from digium - going through that baby right now
03:29.06Poemiusvoip-info.org
03:29.38Kb1_Kanobandy_b: googling against digium.com and voip-info.com will prevent you pulling out most of your hair during your implementation.
03:30.08andy_bnice
03:30.14andy_bthanks all...=)
03:30.32chapanytime. Since I am rarely here. :)
03:31.47Wi_FiKb1_Kanob zapbarge in zapata.conf?
03:32.34Kb1_Kanobhttp://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
03:35.32Wi_Fithnx bud
03:37.10Kb1_Kanobnp
03:38.20*** join/#asterisk drumkilla_ (~russelb@user-24-214-77-148.knology.net)
03:38.20*** mode/#asterisk [+o drumkilla_] by ChanServ
03:39.20JunK-Ymoooo
03:39.46mishehuThe Kow says Mu.
03:39.57Poemiusmooose?
03:40.03*** join/#asterisk carlosmm (carlosm@adsl-66-142-143-244.dsl.hstntx.swbell.net)
03:41.10carlosmmGood Morning.
03:41.38mishehuMoose Poemius?
03:43.53Poemiuslol
03:43.54PoemiusI've had breakfast already, thank you :)
03:45.02infiniiMy 'ship show peers' shows my 2 klite phones logged on. however I can't call each other, I get 404 not found. Any ideas?
03:45.13Kb1_Kanobdial() to multiple zap targets - if one gives back a PRI busy then the dial falls through to the next step. How can I make it continue to timeout first (to ring other targets)?
03:45.23PoemiusMoosehehu? :^)
03:45.52*** join/#asterisk carlosmm (carlosm@adsl-66-142-143-244.dsl.hstntx.swbell.net)
03:45.56carlosmmHola !!
03:46.06Connor_checkout bug 2601
03:46.39enmacacarlosmm: Hola !!
03:47.12carlosmmepale
03:47.35*** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net)
03:48.33carlosmmNE1 knows how to make asterisk use MySQL or any other DB to autenticate and manage extensions ?
03:48.49Poemiuscheck out the wiki
03:48.57Q-At-Homehows todays CVS
03:49.15Q-At-Home:)
03:49.25*** join/#asterisk manipura (~mike@dsl-ep-209-115-250-i114-cgy.nucleus.com)
03:49.52pfnq if you have to ask, use v1-0
03:50.05Q-At-Homeoh come on now, its what I ask every time I show up
03:50.15Q-At-Homeits like... how I greet everyone :)
03:50.50carlosmmNE1 ?
03:51.07pfncarlosmm:
03:51.09pfn~asterisk wiki
03:51.10jbotmethinks asterisk wiki is http://www.voip-info.org/wiki-Asterisk
03:51.15pfnread and be enlightened
03:51.23*** join/#asterisk plungerboy (meowmeow70@sdn-ap-003watacoP0336.dialsprint.net)
03:51.24carlosmm:D thanks i will
03:51.47pfnI hope the diamond I ordered comes out nice
03:51.53Q-At-Homeso what have I missed during my 3 week absense
03:52.26bkw_ack
03:52.52Q-At-Homehiya bkw
03:57.10Q-At-Homehrmm, does anyone know if theres a way to get the 2 port FXO to report when a line is unplugged ala red alarm?
03:57.25Q-At-Homeor what the new way of telling if the line is gone is?
03:57.27*** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com)
03:57.32blankmanHey guys.
03:58.08plungerboyyo
04:01.49Q-At-Homeeh
04:04.23blitzragewe have an older Bell payphone in our classroom.  I'm curious... could the tdm400p and the zaptel drivers have the ability to support the tones when you drop in the money?
04:04.29pfn2 port fxo?
04:04.38pfntdm400p won't go red
04:05.14blitzrageits plugged into a 2 port fxs
04:05.31blitzrageit works (I can ring the phone)... just curious about the coin mechanism tones
04:06.02blankmanSo I have a question ... what is the best media player on linux for the wav49 format? Is it the same as wav in the sense that all player should handle it? When I try to play it back with xmms it can't ... is there a plugin for it?
04:07.36*** join/#asterisk InetNomad (~lenny@rockbox-gw.voiping.com)
04:07.52Q-At-Homehrm
04:07.55ManxPower"play"  It comes with "sox"
04:08.05InetNomadHas anyone tried the parking and respective *8 parking pickup with chan_sip on 1.0.X?
04:08.12Brixiusanyone know if there is a way to reboot a 7960 from the keypad, or do you just have to power cycle it
04:08.14Q-At-HomeI've got a pair of tdm400p boards.. red was handy
04:08.28Q-At-Homehey ManxPower, long time no see :)
04:08.32InetNomadI keep getting "Oct  7 22:27:28 NOTICE[-170013776]: chan_sip.c:7284 handle_request: Nothing to pick up"
04:09.08Q-At-Homejust waiting for the wife to get home before I compile tonites cvs... just in case
04:10.24blitzrageBrixius: try * 6 settings
04:10.39blitzrageBrixius: or telnet into it and do a 'reload'
04:10.58Q-At-Homeisnt *8 for picking up a ring group?
04:11.11Q-At-Home(its been a while sinve I've been here so it may have changed)
04:11.38InetNomadOh yea, you might be right.
04:11.40InetNomadDoh!
04:11.57ManxPowerHow do you do a factory reset on the SPA-3000
04:12.04Brixiusblitzrage: thanks
04:12.07drumkilla_yep, it's for picking up a ring group
04:12.12InetNomadYou dial the Pickup extension to get the extension... doh!
04:12.14drumkilla_and by the way, that is configurable now to any extension
04:12.24InetNomadThanks guys
04:13.24infiniiwoohoo!! ext to ext working internally. even voicemail works and it's sending an email with WAV attachment.
04:13.31Brixiusnow if I can only find the error in my dialplan.xml I'll be happy(er)
04:18.42Juggiecan you guys take a look at this http://lists.digium.com/pipermail/asterisk-users/2004-October/066245.html and tell me what you think.... its really driving me insane.
04:24.53pfnso the first test would be to see what happens if you dial that number from a normal phone as opposed to your sip phone
04:25.25pfnyou also have that timeout message
04:25.38paulc~seen showtime
04:25.40jbotshowtime <show@65.171.196.4> was last seen on IRC in channel #asterisk, 5d 3h 40m 47s ago, saying: 'dumb proxy and registrar'.
04:27.10*** join/#asterisk kb1_kanob_ (johnsmith@sec2d48.dial.uniserve.ca)
04:27.19infiniihrm. voicemail works, but these attachments aren't meant for me to simply go into email and detach and listen to. I guess asterisk's voicemail system decodes it differently
04:27.56infiniianyways, I'm pleased to have internal crap working. I'll read more and explore later. g'nite
04:31.53*** join/#asterisk Wi_Fi (~OUT@c-66-229-160-14.we.client2.attbi.com)
04:35.11blankmanHey ManxPower, so you are saying to use sox to "play" or to convert it to a new file type and play that?
04:36.31Juggiepfn, were you talking to me? there... sip phone works and i can hear the message, from outside the pri i only get fast busy after a few seconds, no audio
04:36.48Juggiebut if i bridge a call to another sip client/server. it works