irclog2html for #asterisk on 20050415

00:00.17Tuplinkis theyr a free one
00:00.22Tuplinkmaybe not alison?
00:00.23tzangerharryvv: sure
00:00.28bkw_har ha rhar
00:00.33Hogiewhen you are flying and you just happen to have an ATC that's female, and being an asshole, when you are leaving her airspace, you say See You Next Time
00:00.41harryvvokay and in most cases that is what the channel bank is used for then.
00:00.43tzangerTuplink: use your own voice or convince a woman with a sexy voice to do it for you
00:00.46MikeJ[Jayden]a free person to record prompts for you, sure, a soft phone, and you!
00:01.25tzangerpersoanlly I find alison's recordings have FAR too much bass
00:01.25MikeJ[Jayden]but beware, you sound like shit
00:01.25MikeJ[Jayden]:)
00:01.45Hogieany idea why my tdm03b wants to dial out of a channel that is unplugged, when my x100 didn't do that?
00:01.50facek_anyone use linksys PAP2 adapter?
00:01.52TuplinkMikeJ i agree
00:02.04shido6facek_, whats up?
00:02.24tzangerHogie: I thought we discussed this already
00:02.27facek_shido6 I have problem with quality and some times with log
00:02.35tzangerthe wctdm driver does not detect line voltage, the wcfxo driver does
00:02.37Hogietzanger: if you replied earlier, i didn't see it
00:02.40harryvvBTW, Transport canada commerical affliate navair I think it was called had there ATC controler all equipment moved 1/2 away from Vancouver into our city. no one ever knew thay had a atc sitting behind cosco store until it was in the news paper.
00:02.51facek_shido6 and maybe you know how to set time (for digit wait) .. because i dont want press # at end of number
00:02.54harryvv1/2 hour away
00:02.57Hogieso I need to have wcfxo loaded, and not tdm?
00:03.12tzangerHogie: no
00:03.21tzangerthe driver for the fxo modules on the tdm400p does not do what you want
00:03.26tzangerthe driver for the fxo card does
00:03.32tzangerso fix the wctdm driver
00:03.51MichaelCatDoes anyone want to help me try to fix caller ID inbound to my X100P clone
00:03.58harryvvtzanger, do you recomend the rhino channel bank
00:04.01bkw_clones don't work right with callerid
00:04.02bkw_NEXT!!!
00:04.03facek_shido6 ?
00:04.06bkw_thats one thing you'll see
00:04.08HogieI thought it had already been done...  There was a bug made in Sep about it...
00:04.15tzangerharryvv: never used it
00:04.18harryvvk
00:04.19tzangerI used the adit600
00:04.29tzangerand the access bank i/ii for FXS ONLY
00:04.35harryvvI see
00:05.08MichaelCatbkw_, so how can I be sure when I buy one on ebay to get a real one?
00:05.09Hogiehttp://bugs.digium.com/bug_view_page.php?bug_id=0002359
00:05.17Hogiethat's what's happening to me...
00:06.00MikeJ[Jayden]MichaelCat, digium does not make them anymore
00:06.45MichaelCat<MikeJ[Jayden]>, I know but neet to get something working
00:07.00Hogieand as I see on there, mark said he fixed it in cvs back then, so it should be in stable now, right?
00:07.30MikeJ[Jayden]MichaelCat, you may be able to sniff one out, but the tdm cards will do what you need
00:07.35MichaelCatI do have a Dialogic D/4PCI but have no idea how to get it to work with Asterisk if it is even possible
00:08.24MichaelCatAll I need is one inbound FXO which supports Caller ID in the US
00:09.45MikeJ[Jayden]MichaelCat, http://www.voip-info.org/wiki-Asterisk+Hardware, there is a note on those cards, you will have to see if it is compatible or not
00:11.11MichaelCat<MikeJ[Jayden]>, actually the card I listed is different than the one on that web site, unless I am missing something
00:12.11MikeJ[Jayden]the question is, is it full duplex or not... and you will need to deal with digium for drivers
00:13.11MichaelCatIt is cuposed to be full duplex but I do not know were to find the drivers
00:15.44MikeJ[Jayden]did you read that page
00:17.20MichaelCatSo it is not possible to get caller ID working on the X100P clone (Ambient chip on it)?
00:19.27*** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net)
00:23.11MichaelCat<MikeJ[Jayden]>, thanks, I will just call Digiom and order the TDM11B bundle
00:23.15*** join/#asterisk netofsickcoder (~netofsick@200.121.129.178)
00:25.15shmaltzanybody here knows how much a Televantage or Avaya, or Toshiba system suporting 300 users in 3 different locaions, each having quad span T1 capabilities, would run (ball park figure) with the phones and all the equipment?
00:28.34shido6in the asterisk world
00:28.35shido6?
00:28.44MikeJ[Jayden]300 bucks a port?
00:28.56MikeJ[Jayden]including phones???
00:28.58MikeJ[Jayden]that
00:29.02*** join/#asterisk jf_ (~feulghulc@modemcable077.187-80-70.mc.videotron.ca)
00:29.04MikeJ[Jayden]'s a giess
00:29.09MikeJ[Jayden]guess...
00:29.11shmaltzshido6, no in avaya and toshiba
00:29.30MikeJ[Jayden]excuse me, I have to go teach myself to type...bbiab
00:29.32shmaltzMikeJ, including the phones?
00:29.39*** join/#asterisk Mentat (~mentat@pcp01260498pcs.nhaven01.ct.comcast.net)
00:29.41MikeJ[Jayden]wag
00:29.49MikeJ[Jayden]400.
00:29.58shmaltzI'm asking with the phones, I think its much more than 300
00:30.01MikeJ[Jayden]500
00:30.10MikeJ[Jayden]used or new?
00:30.20shmaltznew
00:30.44Qwellshido6: got a second for an odd question?  Maybe you can help me figure out why/where its happening
00:30.55*** join/#asterisk jf_ (~feulghulc@modemcable077.187-80-70.mc.videotron.ca)
00:30.57shido6k
00:31.00shmaltzmore than 500
00:31.03shmaltzI think
00:31.18jf_anyone have an idea why each time i reboot, i have to recompile zaptel to use it
00:31.29tzangerjf_: build the modules correctly
00:31.29jf_i use kernel 2.6
00:31.33jf_i did
00:31.35tzangersounds like you have a fucked up distro
00:31.59Qwellfrom my nufone account, if I call a certain 800 number, with my CIDNum set as my 800 DID...I get fast busy.  If I change the "area code", it works fine.  I'm also able to call to the direct lines of people, npanxx1234, 800 CID or not
00:32.12tzangerQwell: that is not nufone's problem
00:32.23jf_tzanger: or maybe i need to emerge another package
00:32.23tzangerthe 800 # you are calling does not have 800 as part of their acceptable NPAs
00:32.29tzangerugh gentoo
00:32.35jf_ya
00:32.55Qwelltzanger: Didn't say it was a nufone problem.  I was certain it wasn't actually.
00:33.04Qwelltzanger: any idea if thats fixable on their end?
00:33.11Qwellthey being the remote party
00:33.16tzangerjf_: find out specifically what seems to be missing (I'm gonna hazard a guess that the modules aren't present in an initrd) and fix it
00:33.19tzangerQwell: yeah
00:33.22MikeJ[Jayden]shmaltz, ok, you guess.. I was using used numbers, I havn't gotten new in quite a while, toshiba phones run around 350-400
00:33.34jf_initrd
00:33.36jf_ok
00:33.40jf_let's see
00:33.42tzangerQwell: don't set your CID to an 800# if it's calling an 80)#
00:33.46tzangerthat is what I did to fix it
00:33.48shmaltzMike per port with the system, or the phones itself?
00:33.55Qwelltzanger: I don't have any other DIDs.  heh
00:33.57jf_u mean u want me to do a rc-update
00:34.06tzangerQwell: you don't have a non-800# to set it to?  Then set it to empty
00:34.17tzangerjf_: I don't run gentoo, it was just a guess
00:34.30QwellI'm just gonna fake it.  939-555-0113
00:34.34tzangerQwell: that'sll work
00:34.35jf_gentoo is no good u mean
00:34.36Qwellbonus points if anyone knows what that number is
00:34.46tzangerjf_: I personally dislike that distro
00:34.52tzangerwww.funroll-loops.org
00:35.08jf_which one should i use then
00:35.16tzangerjf_: whatever you want to
00:35.28tzangerjf_: if you're comfortable with gentoo, then figure out what it doesn't like and fix it
00:35.32Qwelltzanger: Is that generally something at the provider or PBX level?
00:35.46tzangerQwell: it's what the company with the 800# wanted
00:35.52jf_k
00:36.16Qwelltzanger: I tried asking our telecom lady about it, but shes been ignoring me for the last 2 days.  heh
00:36.26tzangeryeah
00:36.28tzangershe won't fix it
00:36.29tzangershe can't
00:36.48Qwellso, something the provider has to "fix"?
00:36.53tzangerthe person with the 800# either did not realize or cared not to allow the toll-free NPAs to access their WAITS line
00:36.57tzangerQwell: NO!
00:37.02tzangerdammit listen to me
00:37.10Qwelltrying...not quite following
00:37.16tzangerwhen I buy an 800# I can decide who gets to and who doesn't get to call
00:37.35tzangersomeone either specifically said "no toll-free NPAs" or they didn't realize they should include them
00:37.45tzangeryou can't fix it and your telco can't fix it
00:37.51Qwellhmm
00:38.02tzangeryou can work around it by setting your outgoing CID to a NON-800# number when calling 800#s
00:38.38Qwellworkarounds due to lack of vision suck, heh
00:38.42file[laptop]it's the way it is.
00:38.48tzangerQwell: welcome to the real world, baby
00:38.49tzanger:-)
00:38.53Qwelltzanger: indeed...
00:40.04QwellSo, to avoid something like that, one should specifically state that they DO want toll-free NPAs to be able to call, when ordering service?
00:41.35tzangersomething like that, yea
00:44.31Tuplinkhow do i test my MOH?
00:44.58tzanger...
00:45.08tzangerTuplink: use your cell, call your house, answer and put yourself on hold
00:45.10tzangerjesus
00:45.14Tuplinkwell...  Started music on hold, class 'default', on SIP/20001-c100
00:45.20Tuplinkbut i heer nothing
00:46.09Tuplinkdefault => mp3:/var/lib/asterisk/mohmp3
00:46.23*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
00:46.54Tuplinkshouldnt i heed somthing?
00:47.34Tuplinkhear*
00:47.57QwellDo you have the right version of mpg123?
00:48.47Tuplinkits instaled but done see a version
00:49.09Qwelltype `mpg123`
00:49.12TuplinkFreeBSD dosnt come with it so i must have ported it
00:49.37Tuplink.59r
00:49.43Qwellthats a first
00:49.48Sedoroxahah
00:49.55Tuplinkwhat ver i need
00:50.07Sedoroxthat one...
00:50.12Tuplinkhehe...
00:50.17Tuplinkwell it dont work
00:50.19Tuplink;)
00:50.26Sedoroxapparently so...
00:50.31Sedoroxwhat problem are you having?
00:50.37Tuplinkno MOH
00:50.42Qwellno sound
00:50.43Tuplinkthere we go..
00:50.49Qwellhuge difference
00:50.49Sedorox?
00:50.53Tuplinkit took for ever to come on... like 5 min
00:51.12SedoroxI know the defaults are kinda quiet to start...
00:51.12Tuplink*CLI> Warning, flexibel rate not heavily tested!
00:51.12TuplinkApr 14 20:50:51 NOTICE[3956]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!
00:51.13QwellDid you put a song with 5 minutes of silence in there?
00:51.14Sedoroxyou could always change the level from quiet to loud...
00:51.27Tuplinkdefault
00:51.48Tuplinkit takes forever for MOH to start i think
00:52.00Tuplinkcause i took me off of hold and put me back and still
00:52.03SedoroxI know when I first did it.. it was just quiet
00:53.14Tuplinkcan i put any random MP3 in there? or need a specific birthrate?
00:53.37Qwellbirthrate...thats also a first
00:53.38SedoroxI did... worked for me.. but I don't think its recommended...
00:56.11*** join/#asterisk zilas (~1@adsl-065-015-074-044.sip.asm.bellsouth.net)
00:56.47Tuplinki think i got it figured out
00:56.54Tuplinkps -aux >  root     4328 91.7  1.1  3628 2868  p0  R+    8:50PM   0:57.48 mpg123 -q -s --mono -r 8000 -b 2048 -f 40
00:57.05Tuplink91% CPU usage
00:57.55Sedoroxhehe.. ours was doing that.. dunno if it still is...
00:58.07zilasdoes anybody have any good example to setup asterisk callback feature after voicemail is left please? The one at wiki doesnt work good..
00:58.08Tuplinkhow do i fix it?
00:58.23Sedorox<PROTECTED>
00:58.25Sedoroxactually no....
00:58.48Tuplinkwhat CPU you have?
00:59.10Sedoroxthat is from a Celery 333
00:59.11*** join/#asterisk techie (gus@asterisk.horizonte.us)
00:59.33Sedoroxwe also run a dual Athlon MP machine...
01:00.01Sedoroxand I forget what the last machine is
01:03.25Tuplinki killed the process and then the music came up
01:03.27Tuplink;)
01:03.43Sedoroxhmm
01:03.57Sedoroxmpg123 is a PITA anyway..
01:04.51TuplinkPITA?
01:04.55Sedoroxpain in the ass
01:05.11Tuplinkbut as soon as i put me back on hold it fucks me
01:06.13mgthtuplink: Is it good at fucking?
01:06.26Qwellhmm, I know I can do ${EXTEN:1} to remove the last char, but is it possible to remove the first char(s)?
01:07.40Sedorox${exten:1} is the first charactderf
01:07.44Sedorox(sp)
01:07.46Tuplinkdosnt that remove the 1st
01:07.54Qwellerm, right...other way around
01:08.03Sedoroxyes...
01:08.04SedoroxUmmm
01:08.07Tuplinkmaybe ${1:EXTEN}
01:08.11Sedorox${EXTEN:-1}
01:08.12SedoroxI think...
01:08.17Qwellhmm
01:08.20Sedoroxwill remove the last digit...
01:08.26Tuplinkkool
01:08.29Sedoroxits in the wiki.. just came across it the other day
01:08.37Tuplinkso... how do you remove the 1st and last?
01:08.48Qwelland, if I need to remove the first, and the last, I'd need something hackish like...  ${{EXTEN:-1}:1}?
01:08.51Sedoroxummm
01:08.56MajestiKDoes anyone have the Sipura Provisioning document that they could send my way?
01:08.57Sedorox${EXTEN:1:1}
01:09.00Tuplinkkool
01:09.03QwellSedorox: oh, sweet
01:09.04Sedoroxhold on.. let me find the wiki
01:09.41Tuplinkanyone have FWD?
01:09.42JunK-Y${EXTEN:-1} is suppose to be valid.
01:09.46QwellTuplink: yeah
01:09.53SedoroxTuplink: yes
01:09.54QwellTuplink: I'm sure a bunch of people do
01:10.10*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
01:10.15shmaltzQwell this should work:
01:10.17shmaltz${EXTEN:1:${LEN(${EXTEN})}-2}
01:10.26tzangershmaltz: jesus
01:10.29Sedoroxthere is a easier way
01:10.30JunK-YhuH?
01:10.33tzangerstep away from the computer
01:10.36tzangerand put DOWN the crack pipe
01:10.38Sedoroxjust have to find the damn page...
01:10.39shmaltzwhy :)
01:10.45JunK-Ytzanger: mouhahaa
01:10.47harryvvbtw, is it possible to tie a asterisk box into the skype network? I did not relize how big and how fast thay are growing.
01:10.48tzangerwe will be by shortly to confiscate your asterisk install
01:10.57JunK-Y${EXTEN:-1}
01:10.59tzangerharryvv: short answer: no
01:11.03shmaltzwhy ???????????????????????????????
01:11.30JunK-Yshmaltz: u passing by tokyo to go in europa.
01:11.47shmaltzwhy? JunK-Y
01:12.01harryvvmmm thats a bummer. So far thay have what 23 million users thats really large.
01:12.01Sedoroxhttp://www.voip-info.org/wiki-Asterisk+variables
01:12.05SedoroxLook under Substrings
01:12.09*** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net)
01:12.10Sedoroxand replace the number with EXTEN
01:12.30Sedorox<PROTECTED>
01:12.31MiccI do believe asterisk kicks ass. I just got asterisk setup with my broadvoice line.
01:12.40harryvvso what is skype a peer to peer network no centralized servers?
01:13.07harryvvmicc well yes it does then there are other issues :)
01:14.10*** join/#asterisk jbAU (~johnblade@61.8.110.41)
01:15.45JunK-Yharryvv: there's a lot of info related to skype
01:16.20harryvvyea downloading the rpm
01:16.30*** join/#asterisk asteriskn00b (asteriskn0@adsl-68-91-7-226.dsl.tulsok.swbell.net)
01:17.52JunK-Yoff topic: i wonder if i should get a Canon A75, whatcha think?
01:19.02shmaltzOK, well my original didn't work but this works:
01:19.04shmaltzexten => 12345678,1,Noop(${EXTEN:1:${LEN(${EXTEN:2})}})
01:19.05shmaltzexten => 12345678,2,Hangup
01:19.07shmaltzit returns 234567
01:19.32JunK-Ywhat ya want to return exactly?
01:20.44shmaltzor this:
01:20.45*** join/#asterisk iq (~iq@70-59-161-91.omah.qwest.net)
01:20.46shmaltzexten => 12345678,1,Noop(${EXTEN:1:$[${LEN(${EXTEN})} - 2]})
01:20.47shmaltzexten => 12345678,2,Hangup
01:20.52shmaltzthat it should strip the first and last digit
01:21.08Tuplinkdose exten => _2.,1,Dial(${EXTEN}) look rite to dial an extention from a VIR
01:22.00shmaltzTuplink, nope
01:22.24JunK-Yshmaltz: u just want to take off the 1st and the last digits of ur exten right?
01:22.26Tuplinkno...?
01:22.28shmaltzyou could do exten => _2.,1,Dial(Local/${EXTEN}@contextname
01:22.39shmaltzyep JunK-Y
01:22.48shmaltztzanger, I think I'm ok
01:22.56shmaltzwhat wrong with what I did?
01:23.14Tuplinki have all or my terminal ext under [localext]
01:23.24shmaltzTuplink, goto has much better results
01:23.41Tuplinkhum....
01:23.50JunK-Ywait, phone
01:24.01Tuplinki jsut want hte user to be able to enter it at any time
01:24.32shmaltzif it's all in the same context, then you don't need anything
01:24.37Tuplinkim new to this just started 3 days ago
01:24.45shmaltzor you could do an include if it's in a different context
01:25.00shmaltzTuplink, np, we were all new at one point
01:25.09Tuplinki put an include => localext in the [mainmenu]
01:25.16shmaltzyep
01:25.27Sedoroxshmaltz: maybe you.. but I had it loaded into my brain....
01:25.29Tuplinkand that did the trick
01:25.30Sedorox:-p
01:25.41shmaltzbut if you did any AbsoluteTimeout in your IVR then you will have some trouble with dissconnected phone calls
01:25.58Tuplinknope
01:26.09shmaltzSedorox, what did you have in ur brain?
01:26.47Sedoroxits the matrix man... one button and asterisk is loaded into my brain
01:27.02*** join/#asterisk hypa7ia (~leigh@modemcable176.166-203-24.mc.videotron.ca)
01:27.18shmaltzwell, I don't need asterisk to tell me that if you take off the first and last digit of 12345678 its 234567
01:27.22shmaltzheh
01:27.25*** join/#asterisk ethzer0 (~ethzer0@d141-238-51.home.cgocable.net)
01:27.32ethzer0hoi hoi
01:28.19shmaltztzanger, you tried that?
01:28.32shmaltzyou have another solution?
01:30.27JunK-Y${EXTEN:-1:1} should be an interesting option, no?
01:30.33JunK-Yor 1:-1
01:30.44shmaltzlets see
01:31.00shmaltzwhy? (I didn't test it yet)
01:31.07JunK-Yisnt working, but a patch should be make
01:31.23shmaltzthe first one tells * strip the first digit, negated it tells it strip the last
01:31.40JunK-Yyea, whatcha think?
01:31.43shmaltzthen the second number (after the :)
01:32.01shmaltztells * for the length of 1
01:32.25shmaltzthis should return just the first digit
01:32.35shmaltzor maybe just the second to last
01:32.44shmaltzam I wrong?
01:34.26*** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net)
01:34.34Mazda-MX5Hi,all~
01:35.35JunK-Yi said isnt make yet, that should be an option (a patch) to make, for like :1:-1 takes off the 1st and last digits
01:36.01shmaltzoh, that I agree with
01:36.12JunK-Yjust for negative value
01:36.25JunK-Yfor positive, that should take it as a length
01:36.37JunK-Yso :1:3 will stay the same as is it now.
01:37.08JunK-Ythat makes sense?
01:38.16shmaltznope
01:38.21shmaltzb/c right now
01:38.55shmaltzexten => 1234,1,Noop(${EXTEN:-1:2})
01:38.57shmaltzreturns 34
01:39.20shmaltzso ${EXTEN:-1:1} should return
01:39.22shmaltz4
01:39.26JunK-Yim taking about the 2nd option
01:39.40JunK-Y1:-1
01:39.46shmaltzbut this maybe
01:40.02JunK-Yits start:length
01:40.02shmaltzI was thinking -1:-1
01:40.12JunK-Yno, im talking about 1:-1
01:40.24shmaltzyep you are right
01:40.33JunK-Y1:-2 would take off the 1st digit and the 2 last one.
01:40.37shmaltz1:-1 should be the one
01:40.38Qwellthere...8xx toll-free DID, CIDNum hack
01:41.13JunK-Yu know where all that kind of parse in done in which file exactly?
01:42.22*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr)
01:42.42shmaltzdon't have a clue
01:42.55shmaltzsince I never (almost never) touch the source
01:43.47file[mac]okay, who stole res_gemini? IT WAS YOU JUNKY WASN'T IT?!?
01:43.55JunK-Yi think i found it
01:44.08JunK-Yfile[laptop]: what the hell is res_gemini?
01:44.15Mazda-MX5Hi, I have question , in my sip.conf can I write "bindaddr=192.168.0.0" to only allow 192.168.*.* phone connect ???
01:44.17file[mac]my billing system
01:44.37file[mac]I can't find my specific build for it
01:44.44JunK-Ynever heard about it.
01:44.47timecoper
01:44.55file[mac]AHA
01:44.56file[mac]found it
01:44.56timecoph323 requires some old crusty pwlib + oh323
01:45.04timecopis it gonna break if I use more recent shit?
01:45.11timecopI cant even find 1.8.1 pwlib
01:45.14file[mac]timecop: probably
01:45.17timecopsf only has 1.8.0 and 1.8.3
01:45.25JunK-Yshmaltz: file[laptop]: scroll up and tell me whatcha think about it.
01:45.37timecophttp://sourceforge.net/project/showfiles.php?group_id=80674&package_id=89974
01:45.39timecop:(
01:45.44file[mac]your idea? I think it's nifty enough
01:45.58shmaltzJunK-Y you mean the :1:-1 ?
01:46.17file[mac]I don't exactly take digits off from the end though, but meh - it may have it's application somewhere
01:46.18JunK-Yshmaltz: yep.
01:46.37shmaltzI thinks it should be done
01:46.39shmaltzmeaning if the second number is negative then it just means how many numbers to cut off from the end
01:46.39file[mac]and yes I realize the ability to do what I just said is already there
01:47.05JunK-Yshmaltz: if u want to take a look on it, it's in pbx_retrieve_variable in pbx.c
01:47.06timecopugh
01:47.11timecopwehre teh fuck do I get these old oh232 and pwlib
01:47.16timecopim just gonna compile 1.8.3
01:47.17timecopand if it fails
01:47.18*** join/#asterisk tessier (~treed@203.210.216.187)
01:47.21timecopblame on jerjer
01:47.22shmaltzwhich means that
01:47.24shmaltz:2:-4 on 123456789 will return 345
01:47.28Miccwhat are the issues with multiple broadvoice lines and asterisk?
01:47.29JunK-Yfile[laptop]: to go at cluecon, u pass by montreal?
01:47.42file[mac]JunK-Y: yes
01:47.49JunK-Yfile[laptop]: trudeau?
01:47.55file[mac]lemme  check
01:48.06JunK-Yprobably, mirabel is almost dead.
01:48.13*** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net)
01:48.25JunK-Yshmaltz: correct.
01:49.18JunK-Ywhen are ya passing by montreal? maybe we'll can go together from montreal to chicago.
01:49.43file[mac]it just says YUL
01:50.08JunK-YYUL?
01:50.14file[mac]airport code
01:50.23blitzrageblah
01:50.47JunK-YYUL is trudeau
01:50.53file[mac]there you go
01:50.54JunK-Ywith aircanada right?
01:51.00file[mac]Tuesday August 2nd, 12:49PM get in
01:51.02file[mac]yes - Air Canada
01:51.08file[mac]depart 1:40PM
01:51.22file[mac]flight 519
01:51.24JunK-Yso i should take my depart at 1:40pm too?
01:51.27JunK-Ylet me check
01:51.32file[mac]well we'll see, I have yet to buy this ticket :p
01:51.55file[mac]as it's still a ways away and bkw hasn't given me all 'da info
01:52.27JunK-Yits chicago ORD i think.
01:52.34shmaltzJunK-Y, ok one will have to add code that if the second one is negative it should do a backward strip instead of offset
01:52.42file[mac]yes
01:52.45file[mac]that's O'Hare
01:52.52shmaltzI mean offset from end insead of length
01:53.08JunK-Yshmaltz: i'll try to patch it
01:53.33JunK-YAC519
01:53.45file[mac]yup.
01:53.50JunK-Yat 214$ right?
01:53.51file[mac]and then for the return flight, AC518
01:53.52Mazda-MX5Hi, I have question , in my sip.conf can I write "bindaddr=192.168.0.0" to only allow 192.168.*.* phone connect ???
01:54.03file[mac]I use expedia myself
01:54.07JunK-Yfile: which day?
01:54.13file[mac]Tuesday, the 2nd
01:54.17shmaltzThanks but I'm ok with either:
01:54.19shmaltz${EXTEN:1:${LEN(${EXTEN:2})}}
01:54.21shmaltzor:
01:54.22JunK-Ythe return!
01:54.23shmaltz${EXTEN:1:$[${LEN(${EXTEN})} - 2]}
01:54.26JunK-Ynot the go.
01:54.32file[mac]ah 6th
01:54.45shmaltz;)
01:54.50file[mac]maybe the aircanada site is cheaper, let's try my
01:54.52file[mac]flight...
01:55.05JunK-Y6th at 13:25 right?
01:55.32JunK-Ygo is at 214$, return is at 225$.
01:55.48file[mac]not too bad
01:55.59*** join/#asterisk bjohnson (~bjohnson@66.11.165.158)
01:56.01JunK-Yfor a total of 563.01 $
01:56.12JunK-Ytell me if u find something cheaper.
01:56.13file[mac]mine is $606.75 grand total
01:56.22JunK-Ybut i'll call aircanada next week
01:56.32JunK-Yto know if they can offer me something cheaper.
01:56.41file[mac]but I don't know if I could stand being on the same flight as you *G*
01:57.04JunK-Yshut up, i'll kick ya out of the plane :)
01:57.09file[mac]haha
01:57.12QwellWhats in Chicago?
01:57.22file[mac]developer conference
01:57.23JunK-YQwell: my grand-mother lives there.
01:57.23JunK-Y:)
01:57.47Qwellare non devs invited?  heh
01:57.48*** join/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au)
01:58.11blitzragewell.. its a dev conf... not much point in going if you're not a dev :)
01:58.18blitzrageactually, I think they are having beginner tracks
01:58.24blitzrageso I guess there could be a point in going :)
01:58.25file[mac]yeah, we'll talk dev to eachother and you'll be scared!
01:58.29PBXtechwhat do SLIP's show up as in CLI?
01:58.37Qwellfile[mac]: sweet
01:59.03blitzragedamn you!
01:59.08*** join/#asterisk FryGuy- (~FryGuy@24.10.47.136)
01:59.13file[mac]blitzrage: muahahaha
01:59.20blitzragehrmmmm... I just programmed something, and then realized I don't know what its doing :)
01:59.33JunK-Yfile[laptop] : u reserved ur hotel?
01:59.34file[mac]oh right, I was going to work on this
01:59.36file[mac]silly me
01:59.37fugitivoblitzrage: skynet!
02:00.07file[mac]JunK-Y: I'm just paying the amount to anthm and bkw... includes some junk
02:00.13file[mac]and I get in for free since I'm a speaker, so yay
02:00.24JunK-Ywhere ya gonna lives?
02:00.31JunK-Yat anthm's house?
02:00.32Qwellfile[laptop]: get me a speaker pass.  I'll come up with something good. :p
02:00.41file[mac]hotel, he has some deal setup
02:00.43blitzrageJunK-Y: the 1:-1 doesn't work?
02:00.45Qwelllike...I'll read chan_sip as though it were poety...or something.  heh
02:00.45timecopso anyone running h323 channel with non-recommended pwlib/h323 bersions?
02:00.51Qwellpoetry*
02:00.54file[mac]Qwell: sing a song, do a dance!
02:00.56blitzrageJunK-Y: I thought the whole x:x stuff got fixed...
02:01.00file[mac]make chan_sip totally RFC compliant
02:01.01file[mac]HAHAHAHAHA
02:01.01JunK-Yblitzrage: yues
02:01.11file[mac]sorry, I like to amuse myself sometimes
02:01.19blitzragefile[mac]: I don't think the SIP RFC is SIP RFC complient
02:01.35file[mac]blitzrage: I like that statement, it's very very nice
02:01.40blitzrage:D
02:02.09file[mac]I tell you baby all my dreams come true
02:02.11blitzrageshouldn't Astricon be posted in the topic? it's before ClueCOn
02:02.13file[mac]when I'm laying next to you
02:02.23Qwellanother astricon already?
02:02.29file[mac]the Europe one.
02:02.30blitzrageQwell: already? :)
02:02.37file[mac]which I will not be attending, so there's no reason to go *G*
02:02.42Qwellwasn't there one like...2 weeks ago? :p
02:02.50blitzragefile[mac]: drumkilla is going, so tons of reason to go :)
02:02.51JunK-Yblitzrage: no, :1-1 is like the :1
02:02.52file[mac]oh oh oh I need to talk to oej
02:02.57shmaltzQwell, you got the it done?
02:03.03Qwellshmaltz: the it?
02:03.09shmaltzthe cutting the first and the last digit
02:03.21Qwellyeah, finished my toll-free DID CIDNum hack
02:03.30shmaltzhow?
02:03.34Qwelllemme pastebin
02:03.54malbechI search a softswitch for a good price but it's very diificult to find one ... no ???
02:04.57JunK-Yblitzrage: astricon europe yea
02:05.05blitzrageJunK-Y: you going?
02:05.07timecopi heard h323 compile will take half a day
02:05.08Qwellshmaltz: http://pastebin.ca/9579
02:05.08JunK-Yive no idea when the next ATL.
02:05.14QwellI could have probably done it cleaner, but...meh
02:05.18file[mac]it's Thursday
02:05.19JunK-Yblitzrage: only if the boss of boss wants to pay
02:05.19file[mac]new stargate
02:05.20file[mac]I'M GONE
02:05.21JunK-Ywhich i doubt.
02:05.23blitzrageJunK-Y: ATL is around October probably
02:05.33blitzrageSG1 sucks
02:05.33JunK-Yyes, around that.
02:05.44file[laptop]blitzrage: shutup you
02:05.47blitzragehehehe
02:06.01JunK-Yblitzrage: i'll need ppl to share hotel room
02:06.12JunK-Ywant to share a room?
02:06.21QwellJunK-Y: Stay with your grammy :p
02:06.25JunK-Yand dont take that for sexual advance!
02:06.28*** join/#asterisk TheEmperor (~mattn@203.114.48.47)
02:06.28JunK-Y:P
02:06.36JunK-YQwell: hhehe
02:06.38file[laptop]blitzrage is straight! OMG
02:06.42blitzrageOMG!
02:06.58file[laptop]blitzrage: really though, you are *hot*
02:07.00blitzrageJunK-Y: hahaha, what are we talking about?  Hotel for which conf? :)
02:07.03shmaltzQwell, I thought you wanted to strip the last and first digit of a variable
02:07.07file[laptop]anyway
02:07.11file[laptop]stargate!
02:07.17JunK-Yif we can share a huge room, that would be cheaper
02:07.19JunK-Ycluecon
02:07.21Qwellshmaltz: well, no, not technically first and last.  More like first, and last 4
02:07.21blitzragefile[laptop]: well, thats what everyone says anyways... doesn't seem to help me pick up chicks though :)
02:07.25JunK-Yfile[laptop]: star ya!
02:07.27bjohnsonQwell: why wouldn't you just do pattern matches instead of all those gotoifs?
02:07.28PBXtechwhat do SLIP's show up as in CLI?
02:07.29file[laptop]blitzrage: bah
02:07.42JunK-YSLIP show up?
02:07.45blitzrageJunK-Y: agreed. Sure, if I can figure out a way to get there, I'm in.
02:07.47PBXtechdont they
02:07.49Qwellbjohnson: Because 808 is a valid areacode
02:07.59bjohnsonI don't understand
02:08.06Qwellno, you are...I'm not
02:08.06shmaltzQwelland, if I need to remove the first, and the last, I'd need something hackish like... ${{EXTEN:-1}:1}?
02:08.07Qwellexplain?
02:08.10PBXtechdont SLIP error show up in the CLI?
02:08.11JunK-Yblitzrage: talk to that to others, i want a room.
02:08.26Qwellshmaltz: first and last was easier to understand then first and last 4
02:08.29blitzrageJunK-Y: I don't even care how big of a room. We had a room at Astricon with 2 double beds, and 4 of us in a room. I slept on my own cot though.
02:08.31JunK-YPBXtech: ive no idea of what u think.
02:08.45PBXtecht1 slips
02:08.46bjohnsonQwell: _1800NXXXXXX
02:08.49JunK-Ythen, reserve me a place.
02:08.53blitzrageJunK-Y: yah for sure. We'll figure something out. I'm sure it'll be easy to get a bunch of people to share, no one wants to spend lots :)
02:09.03file[laptop]I'll share
02:09.04Qwellbecause I'm lazy, and I would still need a goto at the end of that match
02:09.11blitzragefile[laptop]: I thought you were gone :)
02:09.13file[laptop]any way I can cut costs I'm down for
02:09.18file[laptop]I have a laptop, I might as well use it :p
02:09.22blitzragelol, fair enough
02:09.24Qwellor, I'd have to repeat the entire extension
02:09.26bjohnsonQwell: why?
02:09.28blitzrageJunK-Y: there you go... 3 people already
02:09.35shmaltzso here is how:
02:09.36JunK-Yblitzrage: why not going in a bar, find girls and go at their apt? :)
02:09.37shmaltz${EXTEN:1:$[${LEN(${EXTEN})} - 5]}
02:09.39shmaltzthis will do
02:09.40Qwellbjohnson: I need to get back into the "normal" flow anyhow
02:09.48blitzrageJunK-Y: its not that easy outside of Quebec :)
02:09.50bjohnsonsuperdial baby
02:09.50Qwellshmaltz: So will ${EXTEN:1:3}
02:09.56shmaltzI know
02:10.01blitzrage...well, sometimes it is... but not often :)
02:10.03JunK-Yblitzrage: damn, then move cluecon to montreal!
02:10.06JunK-Yhehehe
02:10.08blitzrageJunK-Y: no shit
02:10.10shmaltzbut what if you don't know the length?
02:10.20Qwellbjohnson: back to the lazy part. :p
02:10.26Qwellshmaltz: then I have no business doing a hack like that, heh
02:10.28shmaltzlike in sip channels
02:10.44shmaltz${CHANNEL} holds the current channel
02:11.17shmaltzwhich always consists of SIP/sipaccountinsip.conf-16bithexnumber
02:11.41bjohnsonQwell: http://pastebin.ca/9580
02:11.46shmaltzand I want to know how long the sipaccount part is
02:11.53shmaltzso this is what I do
02:12.18*** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com)
02:12.36docelmowhadup?
02:13.14Qwellhmm
02:13.29Qwellif I do a Goto(abc), and have exten => abc,1,blah
02:13.33QwellDoes ${EXTEN} become abc?
02:13.38bjohnsonand then I do lond distance in a separate context
02:13.47tzangerQwell: why not try it?
02:13.55tzangerabc,1,noop(${EXTEN})
02:13.59docelmoQwell, yes it should..
02:14.01Qwelltzanger: lazy mostly
02:14.08shmaltz{LEN(${CHANNEL:4:${LEN({CHANNEL:9})})}
02:14.08bjohnsonand include them both in the main context so that I can control pattern match order (match toll free before it gets to the long distance pattern match)
02:14.09shmaltzor:
02:14.11shmaltz{LEN(${CHANNEL:4:$[${LEN({CHANNEL})} - 9])}
02:14.12JunK-YQwell: then we are too :)
02:14.15docelmoBut I dont know about alpha extensions
02:14.29JunK-Ytzanger: whatcha think about the 1:-1 ?
02:14.37tzangerdocelmo: This_is_a_valid_extension_name
02:14.41bjohnsonI don't use the same least cost routing for toll free and long distance
02:14.50docelmotz, then yes Q it will work
02:15.30bjohnsonQwell: yes
02:15.37Qwelldamn
02:15.45*** join/#asterisk Luhiwu (~marsosa@200.63.89.245)
02:15.49QwellI guess I could just setvar it before my goto
02:16.01JunK-Yyes u can.
02:16.10Luhiwuhi all
02:16.20PBXtechgetting IRQ misses  damn USB
02:16.22Luhiwuanyone is using cdr_addon_mysql?
02:16.26Qwellbjohnson: yeah yeah :p
02:16.31shmaltzgtg guys
02:16.38shmaltzgn
02:16.46Qwellbjohnson: I guess I'll go find a good one
02:17.20Qwellthere are a bunch though, aren't there?
02:17.34bjohnsonthere's only one superdial baby
02:17.49Qwelloh
02:18.26bjohnsonyou could edit it .. but it does what people want to do about 99% of the time
02:18.52docelmoLuh yes
02:19.27Luhiwudocelmo: are you using accountcode set by sip.conf? it doesn't get inserted in my database
02:19.53docelmoyes its broken Im using the setaccount something in the dialplan
02:20.24asteriskn00banyone have a opinion on the Aastra 480i ip phone?
02:20.30Luhiwuok, thanks for the information, i didn't know it was broken
02:20.55docelmoI dont know for sure if its broke.. But thats the only way I can get it to work for me
02:21.12*** join/#asterisk JerJer (~JerJer@DSL-224.210-rt-bras.che.centurytel.net)
02:22.40blitzragehey, whats an easy way of searching for a term in a bunch of text files in Linux, but outputing the line number and the filename when it matches the term
02:23.02Qwellgrep can do that, can't it?
02:23.25Qwell-H (which should be on by default) and -n
02:24.13*** join/#asterisk brimstone (me@146.229.188.198)
02:24.43JunK-Yi love grep -R (recursive)
02:25.07JunK-Yit doesnt print the line number, but u can do a regex in that file.
02:25.08blitzrageQwell: thanks, that works
02:25.15Qwellblitzrage: man grep :P
02:25.18brimstoneis there an app that will pause xmms if it's playing then resume it after the call ends if it was playing?
02:25.29blitzrageQwell: yep, thats what I'm doing now :)
02:25.31JunK-Ythx for info Qwell too.
02:31.04blitzrageok, thats it for me, time to go relax in the living room with a pipe, then head to bed and do this all over again tomorrow.
02:38.50Sedoroxhmm.. I thought you said pie...
02:38.52shepherdit's grep -r
02:40.57libpcpi would like to ask if anyone has already implemented an asterisk box running with 300 concurrent calls pstn or ip to ip ?
02:42.06|Vulture|300... thats some #
02:42.16|Vulture|load balance to multiple boxes...
02:42.32remmoip to ip should not be a problem
02:42.36|Vulture|pass internal to eachother and keep them apart from eachother
02:42.48remmoespecially if asterisk is NOT in the media path
02:46.03*** join/#asterisk urkle (~urkle@12-203-212-230.client.insightBB.com)
02:47.18urkleI'm trying to setup asterisk to connect to sipphone.com.  I have outgoing calls working perfectly fine. but incoming calls get dumped to sipphone.com's voice mail, and packet sniffing on my end I see that asterisk is sending a 407 "proxy auth required" packet back to sipphone.com
02:48.04ManxPower~docs
02:48.05jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
02:48.05bugbotdocs is assigned nothing and reported nothing.
02:48.06ManxPower~mailinglist
02:48.07jboti guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
02:48.07bugbotmailinglist is assigned nothing and reported nothing.
02:48.12*** join/#asterisk Rick_Hunter (~rhunter@01-098.008.popsite.net)
02:48.14*** part/#asterisk brimstone (me@146.229.188.198)
02:49.25urkleManxPower: I have looked through the docs for weeks.. and found at least 10 different docs, mailing list, forum posts on the subject.. and the examples provided that "claim" to work do not on my server.
02:49.30*** join/#asterisk kks (~kks@203.115.208.140)
02:51.57TheEmperorhi guys, can anyone tell me what Asterisk died with code 1.Aborting. means?
02:52.05JerJerseg fault
02:52.19denonTheEmperor: tail /var/log/asterisk/messages
02:52.29TheEmperorhow to rectify?
02:52.31denonTheEmperor: tail /var/log/asterisk/messages
02:52.43TheEmperordenon:ok, will check
02:53.03*** join/#asterisk Moc_ (~Moc@modemcable165.109-70-69.mc.videotron.ca)
03:00.28Yellow_Fuzzyhi
03:00.44Yellow_Fuzzyanyone got experience with Oztell?
03:01.07TheEmperordenon: got these error messages
03:01.20TheEmperordenon: Apr 15 11:02:03 ERROR[2930]: Unable to register channel '1-4'
03:01.20TheEmperorApr 15 11:02:03 WARNING[2930]: chan_zap.so: load_module failed, returning -1
03:01.20TheEmperorApr 15 11:02:03 WARNING[2930]: Loading module chan_zap.so failed!
03:02.21*** join/#asterisk jeffik (jefik@69.158.30.24)
03:03.42jeffikI need to add ability to access voice mail from outside asteisk using * during mailbox greeting
03:06.35timecopwatching openh323 compile is SO exciting
03:06.38JunK-Y~seen paulc
03:06.52jbotpaulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 7d 8h 13m 12s ago, saying: 'Vancouver BC, it just hit C$1.00/liter'.
03:06.52bugbotseen paulc is assigned nothing and reported nothing.
03:06.52timecop~seen timecop
03:06.53jbottimecop is currently on #asterisk (15d 20h 21s).  Has said a total of 185 messages.  Is idling for 1s
03:06.53bugbotseen timecop is assigned nothing and reported nothing.
03:06.53timecopbullshit :(
03:06.57JunK-Ywhat bugbot does here?
03:08.00drumkillahe's hanging out with us
03:08.13*** join/#asterisk doughecka_ (~Doug@doughecka.user)
03:08.16JunK-Y~junky
03:08.17bugbotjunky is assigned M3026, M3085, M3732, M3674, M3725, M3679 and reported M2922, M2593, M2951, M3878, M2643, M2627, M2776, M4023, M3949, M2869, M2635, M2868, M2923, M3212, M3257, M3679, M3145, M4032, M3661, M2968 et al.
03:08.22JunK-Ynice.
03:08.42drumkilla~drumkilla
03:08.43jbotwell, drumkilla is the Asterisk v1.0-stable maintainer.  ph33r him.
03:08.43bugbotdrumkilla is assigned M2338, M3154, M3758, M3857, M3320, M3012, M2140, M2790, M2983, M3979, M3989, M1595, M3733, M2968, M3977, M2755, M3150, M2662, M3188, M2669 et al. and reported M2814, M4000, M3746, M3046, M3842, M3254, M3124, M3858, M3838, M3864, M3280, M3130, M3083, M3749, M3997, M3990, M3876, M3934, M3989.
03:08.51drumkillaha
03:08.53timecop~timecop
03:08.54bugbottimecop is assigned nothing and reported M1178, M1426, M25, M1475.
03:08.59doughecka_~doughecka
03:09.00jbotdonations are accepted tad@heckaman.com
03:09.00bugbotdoughecka is assigned nothing and reported nothing.
03:09.03timecopLOL SOMEONE FIX MY BUGS
03:09.06doughecka_lol
03:09.07timecopPLZ
03:09.09Corydon76-home~corydon76
03:09.11bugbotcorydon76 is assigned nothing and reported nothing.
03:09.20drumkilladenied
03:09.27doughecka_does anyone know the default password on a cisco phone?
03:09.33drumkillaM4000
03:09.35bugbotM4000 is a feature bug that is new (unassigned): [patch] WaitExten option for Music on Hold. It was filed by drumkilla and was last updated on 04-13-05. http://bugs.digium.com/bug_view_page.php?bug_id=4000
03:09.37doughecka_to unlock it so I can set the tftp server?
03:09.52Corydon76-homeM2278
03:09.52Hogiedoughecka: why not set the tftp server in dhcp?
03:09.52bugbotM2278 is a feature bug that is closed (markster): [patch] Allow functions to be set. It was filed by Corydon76 and was last updated on 03-30-05. http://bugs.digium.com/bug_view_page.php?bug_id=2278
03:09.55JunK-Ywhen new bugs gonna be reported, it will be past here too (as in #asterisk-bugs)
03:09.56JunK-Y?
03:10.10doughecka_Hogie: dhcp server doesnt allow it
03:10.18drumkillanot sure ...
03:10.19doughecka_and I cant figure out how to configure linux dhcp to do it
03:10.26Hogiethat's easy
03:10.31timecopM1475
03:10.32bugbotM1475 is a minor bug that is closed (unassigned): SIP registration fails with Bad Request 400 until "reload" is executed. It was filed by timecop and was last updated on 05-05-04. http://bugs.digium.com/bug_view_page.php?bug_id=1475
03:10.36Hogieoption tftp-server-name "ipaddress";
03:10.37rikstaHogie: you can, i'm sure
03:10.40timecopM1426
03:10.41bugbotM1426 is a tweak bug that is closed (bkw918): [patch] Incoming SIP calls from SIP provider get wack channel names. It was filed by timecop and was last updated on 09-25-04. http://bugs.digium.com/bug_view_page.php?bug_id=1426
03:10.43Hogiethat's on rhel4
03:10.44timecopah this one
03:10.50timecopits stll broken
03:10.55timecopand nobody cares to fix it :(
03:11.00timecopi even had a patch, but it got ignored
03:11.06TheEmperorWARNING[2974]: chan_zap.c:771 zt_open: Unable to specify channel 2: No such device
03:11.07Hogiedoughecka: try cisco
03:11.09doughecka_in /etc/blah.conf?
03:11.11DaminM3660
03:11.11bugbotM3660 is a feature bug that is closed (markster): [PATCH] don't do codec matching until we know who the caller is. It was filed by KP "beeps" Fleming and was last updated on 04-13-05. http://bugs.digium.com/bug_view_page.php?bug_id=3660
03:11.12TheEmperordoes this mean my fxo port is busted?
03:11.12drumkillawas it determined that it was a bug with your provider?
03:11.26DaminM3630
03:11.27bugbotM3630 is a tweak bug that is closed (markster): [patch] allows 0 second retry intervals. It was filed by cmaj and was last updated on 02-20-05. http://bugs.digium.com/bug_view_page.php?bug_id=3630
03:11.28timecopTheEmperor: more like you probably didnt configure it
03:11.32timecopdrumkilla: what, me?
03:11.46drumkillayes
03:11.54timecopdrumkilla: its not a bug wiht a provider.
03:11.57Hogiedoughecka: i'd paste my dhcpd.conf, but the rhel4 box that has it is not reachable because its offnet atm...  my other dhcpds are windows, lol
03:12.09timecopincoming calls from FWD -> asterisk, the SIP/??? name is SIP/yourfwd#-random
03:12.14TheEmperortimecop: but all configured in zaptel.conf and zapata.conf...
03:12.15drumkillatimecop: ok, I see the bug now
03:12.18timecopwhich is highly pointless if one wants to know hte remote side
03:12.22timecopTheEmperor: did you run ztcfg?
03:12.23timecopor wahtever?
03:12.34TheEmperortimecop:yes
03:12.38timecopthen you might be screwed.
03:12.41drumkillatimecop: you can reopen it ... *shrugs*
03:12.48timecopdrumkilla: but it wont get fixed
03:12.52TheEmperorChannel 01: FXS Kewlstart (Default) (Slaves: 01)
03:12.52TheEmperorChannel 02: FXS Kewlstart (Default) (Slaves: 02)
03:12.52TheEmperorChannel 03: FXS Kewlstart (Default) (Slaves: 03)
03:12.52TheEmperorChannel 04: FXS Kewlstart (Default) (Slaves: 04)
03:12.55timecopbecause people probably think its dumb.
03:13.10TheEmperortimecop:would channel 2 be busted?
03:13.18HogieEmperor did it ever work?
03:13.32timecopyou shoul probably read
03:13.33TheEmperorHogie: first installation now
03:13.48urkleok.. outgoing calls work fine to sipphone.com and incoming calls can dial in and they can here me but I can not hear them.. and it looks (according to fiewall logs) that all the RTP traffic is going to port 5004 instead of the RTP range of 10000-10500..
03:14.23urklewhich is the RTP port of the internal sip adapter I'm using..
03:14.38*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
03:14.45TheEmperorany ideas?
03:15.50*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net)
03:15.52timecopheh, stop wasting time wiht sip behind nat.
03:15.56timecopjsut get a global IP
03:16.15HogieTheEmperor: tried swapping module 2 with another module?
03:16.43TheEmperorHogie: the card I have is a 4 port fxo card...
03:16.57Hogieyeah.. try swapping module 2 and 3 around
03:17.02TheEmperoroh..
03:17.04Hogieand see if it follows to 3
03:17.07TheEmperoryou mean take the card out and swap?
03:17.11TheEmperortake the module out from the card
03:17.14Hogieyes
03:17.21TheEmperor? ok, i can try that
03:17.22Hogiemight give you an idea if the module is bad
03:17.47Hogiethat's what I would do anyway
03:17.53TheEmperorok, i can give that a try
03:18.17TheEmperorHogie:cause i mean modprobe, ztcfg and all is working well
03:18.51Hogiethat happened with my 3 fxo
03:18.57TheEmperorhmm
03:19.02Hogieuntil I moved fxo from port 3 to 1
03:19.06Hogieand then it was fine
03:20.01malbechI search a softswitch for a good price but it's very diificult to find one ... any idea ?
03:20.19remmoasterisk $0 + pc
03:20.22JerJerasterisk  -- GPL
03:22.49pigpenhey..how would I set the outgoing caller id for lets say..fax numbers in * ?  in the sip.conf?
03:23.00HogieJerJer: is it possible to split my 1 account on nufone into 2, and have them pull from the same money poll, or should I just add a 3rd box on high bandwidth to direct 800's to the right offices?
03:23.57*** join/#asterisk file[laptop] (~file@mctn1-6079.nb.aliant.net)
03:24.27timecopPWLib version is 1.8.3... BAD
03:24.27timecopPlease read README for further details!
03:24.27timecopMakenshi: *** [checkversion] Error 1
03:24.29timecopheh
03:24.57timecoplorf.
03:24.57file[laptop]you're mad! totally bad
03:24.57file[laptop]er mad
03:24.57file[laptop]and bad too
03:25.00drumkillathen don't you dare report bugs :p
03:25.13timecopi cant evne find required versions
03:25.34timecopthey are NOT on sf.net's oh323 page.
03:26.37timecopha it compiled.
03:30.02*** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net)
03:31.56Hogiebleh blah
03:32.16WGFreewillAnybody every connect * via H.323 to Nortel carrier gear?
03:32.29WGFreewill(succession)
03:32.40timecopim about to connect it via H323 to some chinese voip provider
03:32.43timecopno idea what they use though.
03:32.58WGFreewillI have been passing 50k minutes a month
03:33.00*** join/#asterisk pigpen (~mark@fw.seamans.cc)
03:33.05*** join/#asterisk jdg (~jdg@CA03F857.adsl.mana.pf)
03:33.10WGFreewillusing JerJer chan_h323
03:33.20WGFreewillto Cisco
03:34.17WGFreewillvery reliable with this one small patch from PCadach
03:34.23WGFreewillfrom the bugtracker
03:34.49PCadachWhich one you mean? WGFreewill?
03:34.49WGFreewillGW - Gw
03:34.51timecopgood to know, too bad h323 is a pig
03:35.03WGFreewilldeadlock H323
03:35.24WGFreewill10 tries and still no kill
03:35.38PCadachThere still have an issues, especially when H.245 isn't embedded into signalling (H.225).
03:35.43WGFreewill#3643 and #3848
03:35.58WGFreewillI got it to crash again
03:36.02WGFreewillduring provisioning
03:36.06WGFreewillwith Nortel CS2k
03:36.14WGFreewillyou talk to a nortel box as GK
03:36.22WGFreewilland there are a farm of Gateway controllers
03:36.24WGFreewillthat pass the RTP
03:36.41WGFreewillhttp://products.nortel.com/go/product_content.jsp?parId=0&segId=0&catId=-9274&prod_id=37501&locale=en-US
03:37.04WGFreewillthis is some heavy duty voip gear the AS5850s were like flies
03:37.19PCadachI have crashes on callgen323 when H.245 goes through additional connection. It's not Asterisk-related but OpenH323...
03:37.49WGFreewillthats the noH245Tunneling = no
03:37.53WGFreewillright
03:38.14*** join/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca)
03:38.19*** join/#asterisk Moc[NX] (~mochouina@64.235.196.24)
03:38.34WGFreewillNortel Problems so far
03:38.37WGFreewillno DTMF
03:38.43WGFreewilland g.711 ulaw fails
03:38.49WGFreewillg.729 works with no DTMF
03:39.06Moc[NX]Hail
03:39.11WGFreewillI am just not finding where it dies in the debugs
03:39.21WGFreewillI took packet sniff
03:39.31WGFreewillasterisk sends a release complete
03:39.39WGFreewillafter the facilitycapability
03:39.52DaLionmoc can u pm me ?
03:40.33WGFreewillNoafter terminalCapabilitySet
03:40.41WGFreewill* ReleaseComplete s
03:40.45*** join/#asterisk Rick_Hunter (~rhunter@01-098.008.popsite.net)
03:41.06*** join/#asterisk file[laptop] (~file@mctn1-6079.nb.aliant.net)
03:41.08blitzragefile[laptop]: !
03:41.20file[laptop]what's happening?!?
03:41.46Hogiegah, flight school sucks sometimes
03:41.52PCadachAlso, I'd not tested outgoing calls. When I makes next: callgen323 -> (H323) -> Asterisk-> (H323) -> callgen323 I have Asterisk's crash after about 5-10 successful calls. Probably there is the same issue with outgoing calls when H.245 isn't tunneled.
03:43.30*** part/#asterisk urkle (~urkle@12-203-212-230.client.insightBB.com)
03:44.15blitzragefile[laptop]: not too much. Just heading to bed
03:44.26file[laptop]bah bed
03:47.52pigpenIs there any way to override the outgoing callerid info if a user has a did assigned to them?
03:47.59*** part/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca)
03:48.14file[laptop]pigpen: yes, it's called dialplan logic - learn it
03:48.34file[laptop]or you can set the default callerid in their entry in the respective configuration file usually
03:48.35pigpengee....thanks...you are so helpful.
03:48.36file[laptop]such as sip.conf
03:48.50pgpkeyspigpen: it'll take some time. I've been working on systems for over a decade and asterisk gives me the shits :)
03:48.57pgpkeysso expect some frustration :)
03:49.14file[laptop]but yes, learn about dialplan logic because you can do tons of stuff in it
03:49.14pigpenoh...I don't mind...but I have just been reading for 3 hours...
03:49.34*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
03:49.34pigpenyeah...I have been working "feaverishly" on it...
03:49.46pgpkeyspigpen: hehe I spent the night reading the manual. asterisk does a LOT. it's not a fire and forget like rogerwilco or teamspeak or ventrilo
03:49.51remmoanyone using those wuchuan ip phones? from aredfox.com
03:50.05pigpenbut in the sip.conf I see only examples of:   callerid=John Doe <1234>
03:50.09pgpkeysnot me, I'm still using softsip
03:50.29pigpenso If I want to override the outgoing phone number just replace the 1234 part...not the actual name...
03:50.45file[laptop]exactly...
03:50.45*** join/#asterisk iq (~IQ@70-59-161-91.omah.qwest.net)
03:50.53pigpengreat...I was thinking that...
03:51.02pigpenbut couldn't test at the moment.
03:52.05*** join/#asterisk Rick_Hunter (~rhunter@04-156.008.popsite.net)
03:53.14*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
03:54.21iq~int 13
03:54.22bugbotint 13 is assigned nothing and reported nothing.
03:54.28*** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc)
03:54.34iq~iq
03:54.35jbotrumour has it, iq is that apts IQ is lower than 1
03:54.35bugbotiq is assigned nothing and reported nothing.
03:55.13iqjbot got a girlfriend
03:55.44|Vulture|hmm is wiki down?
03:56.27PTG1234file[mac]: you alive?
03:56.31*** join/#asterisk Prowler1 (Prowler@d142-59-42-93.abhsia.telus.net)
03:56.34file[laptop]PTG1234: yes
03:56.54PTG1234didn't see your email :)
03:56.56PTG1234did you send it?
03:57.03file[laptop]yes
03:57.08file[laptop]maybe my SMTP server is freaking out
03:57.28file[laptop]lemme hop on gmail
03:57.33PTG1234what would the from name show?  I may have deleted it as spam, or the subject
03:57.42file[laptop]Joshua Colp
03:58.24PTG1234k one sec let me search deleted
03:58.24PTG1234hey you have a pda right
03:58.24file[laptop]I've got a Pocket PC GSM Phone
03:58.37PTG1234found you
03:58.44PTG1234moved it to inbox, so for now on you won't go in spam folder :P)
03:58.49PTG1234you use imap mail with it?
03:59.03file[laptop]it hasn't arrived yet :)
03:59.09file[laptop]but I hope so...
03:59.57PTG1234a bunch of that stuff i would be willing to send you but for stuff like the ups esp you wan tto pay the shipping and boxing charges? :)
04:00.06PTG1234oh well i can't get it to pull mail from folders other then inbox
04:00.09PTG1234thats why i was asking :)
04:00.26file[laptop]send back a list with what you can send :)
04:00.28QwellPTG1234: gonna be around in like an hour?
04:00.39PTG1234yah man i meant to talk to you hit me up then qwell
04:00.52Qwellalright, cool
04:01.20PTG1234well like i probably have alot of keyboards, mice upses, bunch of scsi stuff, etc
04:01.36PTG1234and i have 3 boxes of firbre channel drives, perfect for your storage array
04:02.06file[laptop]ooh
04:02.21PTG1234probably 24 fibre channel drives
04:02.22PTG1234etc
04:02.24file[laptop]like, ooooooooh
04:02.46PTG1234but the freight is on your shoulders if you want me to send that shit :)
04:02.46Micchow do I record gsm files?
04:02.56file[laptop]PTG1234: yes yes of course
04:03.00MiccIs there an audio program for linux that will record gsm files?
04:03.05PTG1234record a wav, use sox to convert,.. search wiki
04:03.10PTG1234or just call your voicemail and leave a message
04:03.14file[laptop]PTG1234: just e-mail back with a yes/no beside each thing, estimated weight, and I'll gather the cost and see what I can do
04:03.37PTG1234file[laptop]: its gonna require another trip up to my dads to figure out the weight and etc.. so it will be a few days
04:03.46PTG1234he stores all my stuff in his garage
04:03.52file[laptop]PTG1234: not a problem
04:03.55PTG1234anyone know anyone who would want a 7206VXR cheap?
04:04.29Sedoroxwhats that?
04:04.29file[laptop]I just want an e-mail back so I know exactly what you think you've got
04:04.32pgpkeysdon';t even know what that is.
04:04.35PTG1234cisco router
04:04.44PTG1234has an OC3 interface in it
04:04.46Sedoroxhow much?
04:04.53file[laptop]PTG1234: am I like, cleaning you out?
04:04.55pgpkeysSedorox: book
04:04.59pgpkeyserr book
04:05.01pgpkeysdamn that K
04:05.04PTG1234file: i can only hope so :)
04:05.08Sedoroxahaha
04:05.29PTG1234Sedorox: i don't know just want someone to offer me for it.. its a serious piece of machinery can handle an OC12
04:05.37file[laptop]PTG1234: what was the size of those drives btw
04:05.42Sedorox$12
04:05.43Sedorox:-p
04:05.44Sedoroxj/k
04:06.00PTG1234hah
04:06.12PTG1234file[mac]: no idea probably 16gigs i am betting.. maybe 32gigs
04:06.20file[laptop]PTG1234: cool
04:06.23PTG1234hey did someone see the original voice broadcasting email this guy replied to today?
04:06.41PTG1234on -biz
04:07.03file[laptop]I try to ignore the lists
04:11.18denonPTG1234: You have a tracking number for me? <G>
04:11.40file[laptop]my internet is dying, noooooooo
04:11.49Sedoroxeveryone's is
04:12.10pgpkeysNOOOO! not my internet! please! i haven't surfed to the end of the internet yet!
04:12.18PTG1234denon: its going out tommorow, i'll have it for you in morning :)
04:12.28denonah ok
04:12.44PTG1234sorry man :)
04:12.50Sedoroxanyone remember those DSL commericals where the guy gets a popup.. "you've reached the end of the internet"
04:12.50PTG1234just not set up to easilys hip, i am not an ebay junky
04:12.51Sedorox?
04:13.11pgpkeysSedorox: yeah
04:13.12file[laptop]and with all he's shipping me... ha ha ha
04:13.18Sedoroxhehe
04:13.18pgpkeysthat's what i was playing off of :)
04:13.24Sedorox:-p
04:19.41PTG1234haha
04:19.55PTG1234its ok file laptop may be answering alot of stupid * questions for me come the future :)
04:20.01file[laptop]yeah
04:20.04PTG1234file: you know anything about shitty app_queue? :)
04:20.14file[laptop]PTG1234: unfortunately yes
04:20.56PTG1234oh now your gonna earn your keep :)
04:21.43file[laptop]I also know that chan_sip slowly steals your sanity
04:22.24PTG1234yah thats what i am working on now :) complete rewrite.. but was hoping to avoid that with app_queue
04:22.34PTG1234i need to make it more virtualpbx friendly.. and iface with my new user stuff
04:24.44sivana~sivana
04:24.45jboti heard sivana is a putz
04:24.45bugbotsivana is assigned nothing and reported M2515.
04:24.56sivanaM2515
04:24.56bugbotM2515 is a tweak bug that is closed (markster): [patch] cleaned up cdr_mysql.c. It was filed by sivana and was last updated on 01-10-05. http://bugs.digium.com/bug_view_page.php?bug_id=2515
04:25.09sivanaM2515 open
04:25.09bugbotM2515 is a tweak bug that is closed (markster): [patch] cleaned up cdr_mysql.c. It was filed by sivana and was last updated on 01-10-05. http://bugs.digium.com/bug_view_page.php?bug_id=2515
04:25.13sivanaheh
04:25.37sivana~bugbot
04:25.38jbotwell, bugbot is a bot that gives bug statuses.  You can /msg bugbot help for info or visit him on #asterisk-bugs.
04:25.42bugbotbugbot is assigned nothing and reported nothing.
04:32.34*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
04:36.56timecopheh mysql
04:37.10*** join/#asterisk walnuck (~James@modemcable106.82-200-24.mc.videotron.ca)
04:37.11walnuckhi
04:37.12timecopwith a high performance opensores database like mysql, you're better off logging to flat files.
04:37.25*** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co)
04:37.41dec'opensores'?
04:38.58walnuckdec, /s/wh
04:39.14walnuck*lol
04:39.26walnuckgoddamit wtf are the channels dead?
04:39.31decheh
04:40.00walnuckdec, know any simple routting table tinker?
04:40.31decsorry, what do you mean by that? do I know how to setup a basic route table?
04:40.49walnuckdec, i have two nics on the card, just trying to add my laptop..
04:41.10decokay
04:41.28decso you have two machines, one has two nics and one is a laptop?
04:41.33walnuckdec, the desktop has eth0 and eth1 already hwaddress mac set, nicely seen, all that's left is my router table.
04:42.09walnuckdec, the laptop has eth0 and has been successful checked with dhcp, currently i'm on the desktop
04:42.55walnuckdec, i have no firewall rules to simplify the setup, can I show my table here?
04:43.03*** join/#asterisk U-238 (~U-238@CPE-138-217-33-205.vic.bigpond.net.au)
04:43.20decin here? i don't know what the channel rules are... you can PM me with it if you wish.
04:43.39decwhat's the problem you're having though? the desktop and laptop can't communicate ?
04:44.02walnuck169.254.0.0     *               255.255.0.0     U     0      0        0 eth0
04:44.02walnuckdefault         192.168.1.1     0.0.0.0         UG    0      0        0 eth0
04:51.17*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:59.23file[laptop]goodnight all
04:59.31*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr)
04:59.42*** part/#asterisk U-238 (~U-238@CPE-138-217-33-205.vic.bigpond.net.au)
05:01.40brc_hey hey
05:01.43brc_file[laptop],
05:02.19QwellPTG1234: still around?
05:02.36PTG1234yah
05:02.37PTG1234i am :)
05:02.42Qwellaim?
05:02.44PTG1234just message me i don't look in the channel much
05:02.45PTG1234sure
05:03.55*** join/#asterisk dec (~tom@203.87.91.78)
05:04.22deci'm back walnuck
05:05.23walnuckdec, if I want internet for my laptop and I were a guru, do I need anything else other than a proper router table and not any special progie to install?
05:06.31Qwellwalnuck: Should look at the advanced networking howto on tldp.org
05:07.03PTG1234well where are you? :)
05:07.08QwellI IMd you. :p
05:07.20*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
05:07.36FuriousGeorgewhats that bash command to list interrupts
05:07.45FuriousGeorgeim trying to find out what pci bay has a free irq
05:07.49drumkillacat /proc/interrupts
05:09.00FuriousGeorgegrunkilla:  thanks, i see that 4 and 6 are free, how can i find out which bay uses which interrupt, that would be in the bios, no
05:10.10*** join/#asterisk michael_t (~michael_t@c-24-20-234-51.hsd1.or.comcast.net)
05:11.22niZonanyone use the cisco 7905/12 with *?
05:15.55*** part/#asterisk walnuck (~James@modemcable106.82-200-24.mc.videotron.ca)
05:17.15niZondead in here..
05:17.59remmono thats just your brain ;)
05:18.56*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
05:20.43*** join/#asterisk danalien (~danalien@danalien.user)
05:20.58michael_thas anyone else seen this error in the asterisk console while playing a .wav? ast_waitstream_full: Wait failed (No such file or directory)
05:21.49Miccok, so I took a wav file and ran it through sox but when I use it in asterisk it plays at a really slow speed.
05:22.05MiccCall (425)278-0757 to see what it sounds like
05:22.54*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
05:23.20drumkillamichael_t: don't specify the extension of the filename
05:23.29drumkillaif it's foo.wav ... then Playback(foo)
05:23.29michael_ti didn't
05:24.24michael_ti'm using /path/filename (w/o extension)
05:26.05Miccdoes asterisk play wav files too?
05:26.18MiccSo I don't have to convert to gsm?
05:27.46Miccok I get unexpected frequency now
05:28.33MiccSo what format should the wav file be in?
05:29.06remmoanyone here using OSP?
05:35.53*** join/#asterisk jbAU (~johnblade@c210-49-42-214.rochd2.qld.optusnet.com.au)
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05:44.52remmosigh
05:53.29*** join/#asterisk dansoftware (~dansoftwa@tdata.ru)
05:54.40dansoftwareHi guys
05:54.46*** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net)
05:57.14*** join/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca)
05:57.34DaLionanyone know where script that dumps mysql to flat file configs ? or anyone used it ?
05:57.36*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
06:03.40DaLionwiki down ;(
06:04.07QwellDaLion: mysqldump?
06:04.21*** join/#asterisk ellvis (~ellvis@195.98.29.34)
06:04.24ellvishi people
06:10.22remmolllalla
06:10.57ellvissounds like moh :)
06:12.46JerJerdo you have ASCAP and BMI rights to play that MOH file?
06:15.24remmolol
06:15.29|Vulture|is there a way to change the pager email in voicemail.conf?
06:15.43ellvis:)
06:16.09remmoi'm trying to get OSP up and i have unresolved symbols
06:16.54JerJer|Vulture|:  um vi
06:16.57JerJerinsert mode
06:17.01JerJerscroll to appropriate line
06:17.05JerJerchange email
06:17.14JerJeresc shift zz
06:17.43remmoi prefer vim
06:17.49|Vulture|JerJer: but its in voicemail.conf? is it the same as emailbody?
06:17.54|Vulture|I like pico :P
06:18.14remmoewww i went pico - joe - vim - joe
06:18.27*** join/#asterisk brettnem (~mive29@user-0ccsr10.cable.mindspring.com)
06:18.29|Vulture|Ive just been using pico for so long
06:18.38brettnemEK pico?!
06:18.44brettnemgood morning all
06:18.46*** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu)
06:19.05*** join/#asterisk qkit (~qkit@60.48.11.74)
06:19.16yxayou shouldnt need to type 4 letters to get to an editor :)
06:19.16brettnem~seen bkw_
06:19.18jbotbkw_ <~brian@bkw.developer.and.friend.of.asterisk> was last seen on IRC in channel #asterisk, 6h 15m 12s ago, saying: 'thats one thing you'll see'.
06:19.26bugbotseen bkw_ is assigned nothing and reported nothing.
06:19.35brettnemyxa: heh, like "emacs" ?
06:20.04JerJerword
06:20.27brettnemha
06:20.28yxaheh
06:20.49qkithey guys? i wonder what can be difference in using h323,sip or iax protocol? i have read the readme in the source but still can really understand it......what can be the difference of it in using in asterisk????
06:20.58qkitcan=cant
06:21.24brettnemsupport and easy of use
06:21.29brettnemnat traversability
06:21.38brettnemmaturity
06:22.12The_ApeHave anyone tried the presense support in eyebeam with asterisk? i cant get it to work, everyone is constantly online(even when they are not).
06:22.14qkitbut will it have any problem with other system? as it used its own protocol and not the standard protocol
06:22.35brettnemof course it will have problems talking to devices that don't support IAX
06:22.39qkitwhat can be the disadvantace of the sip or h323 protocol
06:22.40brettnemdon't use it then
06:23.05brettnemh323 support really sucks on asterisk (I think that's due to problems in the actual stack which isn't part of asterisk at all)
06:23.25brettnemSIP dosen't support real line side signalling.. like flashhooks
06:23.35remmohey h323 is not that bad
06:23.46brettnemheh.. I knew I'd offend someone! :)
06:24.04brettnemhave you seen zoa's performance charts for h323??
06:24.10remmonot i, just had to bust my chops to get it working with my provider
06:24.13remmono
06:24.20brettnempretty nasty stuff!
06:24.24remmourl?
06:24.43brettnemwish I knew.. maybe on the astricon site.. hey, actually he has some astertest site I think..
06:24.58*** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net)
06:25.09brettnemcheck out www.astertest.com
06:25.14*** join/#asterisk znoG (gs@200.115.216.109)
06:29.08|Vulture|anyone know of the top of their head the reboot keys for the polycom ip500.. its like voicemail + mute, then the 2 volume keys?
06:29.38brettnem|Vulture|: try vol down, vol up, hold and... um.. messages?
06:29.45|Vulture|yea thats it
06:29.46|Vulture|thanx
06:30.01|Vulture|I am writing a letter to someone to do it and I don't have one in front of me
06:30.08remmo`hmmm
06:30.43brettnemhmm?
06:33.01remmohmmm how to open powerpoint in fbsd
06:33.06remmoopenoffice
06:33.54brettnemheh
06:34.11brettnemit's a windows world we live in
06:34.58brettnemanyone in here used SEMS?
06:35.21brettnemlets bring on the competition
06:35.32remmoWINDOWS SUX
06:35.58brettnemyeah, well
06:36.20tessierremmo: You are preaching to the choir, my friend
06:38.25brettnemso no one has used SEMS?
06:38.53brettnemhow is it that a room with 289 members is so quiet?
06:38.55remmotessier: well hey what can i say
06:39.05JerJerfriday night
06:39.13remmothey are sleeping
06:39.55JerJerAdd warning for _. match (bug #4032)   <--- THANK YOU
06:40.05brettnemisn't it friday morning?
06:40.13brettnemoh, nice!
06:41.00PTG1234anyone in here do any pocketpc development?
06:41.11brettnemcross compiling asterisk?
06:41.18PTG1234no for something else :)
06:41.32remmoc#
06:41.58PTG1234that a language they use alot for pocketpc?
06:42.16*** join/#asterisk gres (~serg@81.222.48.242)
06:42.28ellvispocketpc? ehm, palm rulez, not pocketpc :)
06:42.35remmoyup
06:42.41PTG1234any idea what compiler?
06:42.52remmomsdn.microsoft.com
06:43.07remmothey have a free ide called visual studio express beta
06:43.11PTG1234m$ compiler
06:43.16remmofree
06:43.32remmobut its missing dotNet stuff
06:43.35PTG1234i probably have visual studio but its a pain
06:43.42brettnemoh.. make it free to stamp out all the little guys making a business selling compilers
06:43.49remmoi have been using it and its not that bad
06:44.00brettnemsoon they'll be giving away clothes and shoes for free and put Walmart out of business
06:44.01PTG1234whats does C# add to the c language?
06:44.09remmonot much
06:44.14ellvisPTG1234: java mess:)
06:44.20remmomore like c++ , java, php
06:44.28PTG1234ugh :)
06:44.36brettnemjava.. barf
06:44.37remmoits actually really good when you get your head around it
06:44.38PTG1234are you sure that compiler will work on pocketpc as well?
06:45.13remmopretty sure. if its ppc 2003 should be right
06:45.34remmoi have only been doing console stuff with it, but does forms and all
06:45.48brettnemok.. I'm going to give sems a shot for voicemail.. wahoo
06:45.52remmoand the XML stuff, well lets just say hats off to m$
06:45.57PTG1234b/c i thought it was usually an addon
06:46.30remmob/c?
06:46.31PTG1234i need like the simpliest program for pocketpc, but i am not sure what would be the easiest route to do it
06:46.37PTG1234how is java's support on pocketpc
06:46.45PTG1234then it could work on palm as well :)
06:46.59ellvisjava suxx :)
06:47.04remmoewww java. good luck on ppc
06:47.11brettnemellvis: agreed!!!
06:47.13remmoc# for simple
06:47.26brettnemquickbasic, all the way
06:47.31remmonever thought i would be a m$ advocate
06:49.17drumkillamust be bored :p
06:49.21elricdoes ${DIALSTATUS} exist in CVS head only or does stable release have it as well?
06:49.38brettnemit should be in stable.. it's been around for a while
06:49.46drumkillayeah, it's in stable
06:49.50elricalright, thanks
06:50.13drumkillashow application dial   ;)
06:53.21elricwill Dial(${EXTEN}|60|gM(detect)^DIALSTATUS)  work?
06:54.06elricit doesnt seem to pass DIALSTATUS to the macro
06:54.54*** join/#asterisk TheEmperor (~user@203.121.47.165)
06:56.15elrics-${DIALSTATUS} shows up as s- on the CLI
06:56.47elricoh
06:56.57elrici forgot to read upon completion
06:56.59elric:|
06:58.50*** join/#asterisk luke-jr_ (~luke-jr@207.192.221.172)
07:12.32*** join/#asterisk pif (ldm@zenon.apartia.fr)
07:14.11*** join/#asterisk WorkTooMuch (~work@82.148.188.1)
07:14.39*** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com)
07:14.40RoyKanyone that knows what it'll take to support CallingPres on SIP?
07:15.48*** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za)
07:16.00clive-mwhois atacom
07:17.58decgrr
07:18.02decsilly SIP
07:18.04decnot working :(
07:18.09*** join/#asterisk pbxjunkie (~nkatzakis@videocomputer.gr)
07:18.15pbxjunkiehowdy hey:D
07:19.12*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
07:20.10PTG1234does gmail support imap?
07:20.45elricah the wiki is down
07:21.04elricits a good thing in a way, now I can take a break
07:21.57*** join/#asterisk Blackvel (~blackvel@dsl-213-023-033-191.arcor-ip.net)
07:23.10brettnemroyk.. callingpres=rpid
07:24.06RoyKbrettnem: er. what is rpid? I thought chan_sip didn't support callingpres.
07:24.36RoyKbrettnem: or is this in HEAD?
07:24.47brettnemoh, aybe asterisk dosen't but sip does
07:24.54*** join/#asterisk Martohtar (Martohtar@82.196.218.80)
07:25.04RoyKMartohtar: morgen
07:25.30*** join/#asterisk gres (~serg@81.222.48.242)
07:29.25QwellPTG1234: just pop3 I think
07:30.46facek_i am looking for two sound files.. beep (for good) and beep (for wrong)
07:30.49facek_do anybody have?
07:30.57wildgooseMy music on hold is blaring loud and distorted.  Setup says to use quietmp3.  Any thoughts on why?
07:31.40pbxjunkiewildgoose: there is an issue of compatibility between versions of mpg123
07:31.55pbxjunkieto my knowledge the LATEST version of mpg123 causes problems with asterisk
07:32.13*** join/#asterisk emitrax (~emitrax@ingnatdyn33.unime.it)
07:32.24Qwell0.59r is the latest stable, which is the only version thats supported
07:32.34pbxjunkieyea , that's the one
07:32.50*** join/#asterisk iamcool (Omega11@69-165-65-219.sbtnvt.adelphia.net)
07:33.11wildgooseok thanks
07:33.14*** part/#asterisk iamcool (Omega11@69-165-65-219.sbtnvt.adelphia.net)
07:33.23wildgoosedoesn't that have some security issues thouhg?
07:33.42*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
07:34.19|Vulture|0.59r is what is grabbed with make mpg123
07:34.29|Vulture|"make mpg123" is the safest way
07:34.59wildgooseis mpg321 supported instead?
07:35.09RoyKno....
07:35.18foobosonly mpg123 r59 is supported, no other version or product
07:35.24RoyKbut there's native mp3 support in -addons
07:35.57|Vulture|anyone know if there is a patch to be able to view the status of a Zap card through the manager interface... thinking about incorperating it in to Nagios
07:36.00The_Apeon my system the symlink to mpg321 wasn't overwriten when i did make mpg123. Might be an idea to remove that manually first.
07:36.07|Vulture|unless there is already a solution
07:36.28|Vulture|so if say a T1 drops... you will be able to get a notice
07:36.51*** part/#asterisk mozrat (~mozrat@80.68.89.215)
07:37.05RoyK|Vulture|: I guess you can start off with my nagios plugin and do a pro show span 1 or something
07:37.44|Vulture|RoyK: oh thats your plugin? the IAX one? it works great!
07:37.53|Vulture|yea I was thinking about that...
07:38.18RoyKit does iax and manager
07:38.22|Vulture|wasn't it JunkY who coded the show span utility?
07:38.31RoyKthen use the manager interface to check the pris
07:38.51|Vulture|I haven't tried the manager part just hitting IAX to check up status of *
07:39.28RoyKhttp://karlsbakk.net/asterisk/
07:39.34RoyKget the plugin from there
07:40.10RoyKhit. is the wiki down again?
07:40.33|Vulture|yea
07:40.38|Vulture|went down about 2 hrs ago
07:41.21RoyKshit
07:41.38RoyKsomeone ought to replace that
07:41.45RoyKor move it, I mean
07:41.58drumkillaI wish we had one that used MediaWiki
07:42.02drumkillait looks so much better :)
07:42.06*** join/#asterisk heison (~heison@p85.n-lapop06.stsn.com)
07:42.33RoyKwell. tikiwiki does  the job
07:42.40|Vulture|when its up :P
07:42.52RoyKonly you need to use google to search for something. the tiki search sucks big  tim
07:42.53RoyKe
07:42.55|Vulture|its usually up... just seems like when it goes down.... it dies
07:43.11|Vulture|RoyK: agreed... searching for a .conf never brings it up
07:43.25|Vulture|Ive just been using google now and using cache from the wiki since its down
07:44.13drumkillayou people and your attachment to that damn wiki :p
07:44.33RoyKit's a nice docs db
07:45.05drumkillayeah, its unfortunately the most complete thing out there, heh
07:47.50wildgooseseems that the latest version of mpg123 WILL work, but not the -mmx version.  Change it to the 486 or generic version in the symlink and it's playing fine here
07:48.07wildgooseMy guess is that the rescale flag is broken in the latest mmx version
07:49.16*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
07:54.09fenlanderHello, Is there a problem with the asterisk mailing lists? I've not had any mail from them in the last 17 hours
07:55.41drumkillai'm getting them
07:56.51*** join/#asterisk nitram (nitram@superblob.com)
07:57.14fenlanderHmm. I seem to be getting mail from other lists. What have I broken this time.
07:59.00emitraxI can't get my 7940 registered with asterisk. On the status messages of the phone I get: E640 REG msg unsupported
07:59.23emitraxI m useing SIP 7.0 as firmware
07:59.55newlI thought SIP was only up to 2.0. :)
08:00.24fenlanderThat explains the mail problem. Google has started marking them all as spam.
08:01.11*** join/#asterisk Delvar (~irc@83.146.53.34)
08:04.27PTG1234i am missing emails from mailing list
08:04.28PTG1234i have no idea
08:04.53elrichas anyone compiled the iax library on FreeBSD
08:05.24fenlanderGmail is marking all my mail from asterisk-users, dev and biz as spam
08:06.31drumkillano cvs?!
08:09.16*** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com)
08:09.29PTG1234who needs cvs when you got stable :)
08:09.53drumkillayou can track stable on the cvs mailing list, too :)
08:10.18PTG1234heh
08:10.35PTG1234i am now wiring my chan_sip for stable.... i got sick of cvs bugs
08:10.52PTG1234you know is an ipod mini the way to go
08:10.58PTG1234anyone have any mp3 player opinions
08:10.59PTG1234? :)
08:11.05drumkillai got an ipod photo
08:11.12PTG1234i had an ipod
08:11.16PTG1234it broke
08:11.18PTG1234those minis look slick
08:11.49*** join/#asterisk jwitte (~jwitte@port-212-202-101-206.static.qsc.de)
08:11.53drumkillabut, but ... ONLY 4 GB!
08:12.14PTG12346gb :)
08:12.22PTG1234do i really need more then that :)
08:12.43PTG123418 hours of battery life
08:12.45PTG1234thats pretty strong
08:14.27*** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it)
08:15.53*** join/#asterisk rvhi (~rv@66.175.65.89)
08:15.58rvhihi
08:16.25rvhianyone has paging working by sending sip call with "auto answer"?
08:16.51rvhii am going to post a bounty on this.
08:16.58rvhi$200
08:17.10DaLionyeah winamp rocks
08:17.25DaLionthis #mp3playersforsale now ?
08:17.28PTG1234yah i will just throw my laptop in my pocket
08:17.28DaLion;)
08:18.05*** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-65337.adsl.ttnet.net.tr)
08:18.21pinorvhi: what client are you using?
08:18.25*** part/#asterisk glLoadIdentity (~tuyan@dsl81-214-65337.adsl.ttnet.net.tr)
08:18.27rvhipolycom
08:18.43DaLionanyone tried the testsuite ?
08:18.46DaLionfor sip
08:18.47*** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-65337.adsl.ttnet.net.tr)
08:18.48DaLionand iax
08:19.01DaLionbtw got my mysql cluster up
08:19.07DaLionwas pretty fun
08:19.07pinorvhi: i see that there's already something on the wiki, but i can't load any page from it right now :(
08:19.19DaLionyeah wiki is down/slow/down tonight
08:20.13rvhii saw some basic code in wiki
08:20.41rvhii want the ability to do zone page
08:21.05rvhiweb front to add/remove zones, and add/remove users from a zone
08:21.43rvhiso it records a message, then does paging
08:25.09pinoi think you need the first configuration files for the polycom, first.
08:25.18pinoe.g. http://www.kriscompanies.com/modules.php?name=Downloads&d_op=viewdownload&cid=1
08:25.33pinos/first/right/
08:25.38*** join/#asterisk jonathh (~asd@217.46.145.65)
08:26.00jonathhmorning gents
08:26.06rvhipolycom can auto answer, it is not a problem
08:26.13rvhii got it figured out
08:26.24rvhithe * part is my issue
08:27.03pinoso the polycom already gets you on the speaker?
08:27.31rvhiy
08:27.51pinoyou just need to connect the same channel to multiple phones?
08:28.33rvhii don't know how to do it in *
08:28.58jonathhCan someone tell me.. if the wilcard x100p is being discontinued. what is it being replaced with?
08:29.53drumkillait has been discontinued for a while
08:29.57drumkillathe TDM400P is its replacement
08:30.38jonathhthe price difference from what i can see is HUGE
08:30.59jonathhall i need is a PCI FXO (i think) pc to phone line.. to play about with
08:31.26jonathhthe X100p i have seen for $6~
08:31.44drumkillabut you're not getting a Digium card
08:32.02jonathhif it works with asterisk? does it matter at this stage?
08:32.07pinorvhi: i'd try to pick a MeetMe room and connect each extension in the zone in monitor mode ('m' option).
08:32.35jonathhonce i have a working model and am making millons i'll plouge some back... but i cant afford to just yet.
08:32.51|Vulture|jonas: if you just playing with it, nothing more than testing.. its fine, but if you plan on using it for anything production wise.. don't count on it
08:33.15jonathhi am just playing with it currently
08:33.15|Vulture|urg jonathh
08:33.18jonathhon my home line..
08:33.26|Vulture|yea then by all means...
08:33.57|Vulture|just its funny to see people trying to throw 6 X100P clones into a box
08:34.07jonathhthe TDM10B: which has 1 FXS port.. like the Zaptel x100p does(did)...
08:34.14jonathhbut the price is huge..
08:34.17jonathhwhat happens then?
08:34.39|Vulture|jonathh: with 6 clones... well I don't think more than 3 work
08:34.47jonathhoh right..
08:34.49jonathhwhy is that?
08:34.53jonathhaddressing?
08:34.54|Vulture|not a clue
08:34.59jonathhoh :)
08:35.00|Vulture|possibly
08:35.05|Vulture|never tried it myself
08:35.09drumkillajonathh: the x100p is FXO
08:35.20|Vulture|only have had 2 x100p digiums in a box at once
08:35.24jonathhthat is where getting more FXO modules for the TDM card owrks
08:35.25|Vulture|that was just testing
08:35.34jonathhyeah sorry drumkilla
08:35.46jonathhi mean the port that goes from the PC to the phone line.....
08:35.52|Vulture|I am about to have a buncha FXO modules for sale.. moving most of my offices to TE110P cards
08:35.53drumkillano prob, just don't want you confused
08:35.55jonathhdid it work?
08:36.11jonathhmodules for the TDM?
08:36.31|Vulture|yea
08:36.54jonathhif your floggin'um on ebay.. make sure you notify the room!
08:36.55jonathh:)
08:37.01|Vulture|21 modules to be exact
08:37.06|Vulture|will do
08:37.06jonathhwhere in the world are you?
08:37.17|Vulture|Orlando, FL
08:37.23jonathhahh
08:37.30jonathhBrighton, UK :)
08:37.50|Vulture|ah... yes our neighbors across the pond ;)
08:37.56jonathh:) indeedy
08:38.01jonathhwhat i need to do
08:38.17jonathhis talk my employers into letting me disconnct our PBX
08:38.31jonathhso i can play about with an interface to a digital line
08:38.37jonathh(which i know nothing about)
08:38.43|Vulture|where * just blows away people is in new installs
08:38.45jonathh.. i dont think they will let me :)_
08:39.00jonathhi am amazed by the potential it has
08:39.02jonathhwhat do you do?
08:39.06|Vulture|you have a E1 going in there?
08:39.06*** join/#asterisk pranav (pranav@221.128.181.21)
08:39.21drumkillabe careful, you will become addicted to asterisk
08:39.25drumkillait will consume your life
08:39.28|Vulture|I use Polycom IP500s and Dell servers
08:39.30jonathhim not sure of the terminology.. ISDN something or other
08:39.36drumkillaand you will stay up all night working on it
08:39.37jonathh15lines in..
08:39.39jonathh5 out
08:39.44|Vulture|drumkilla: I just redesigned my dialplan for the 3 time since I started today
08:39.45jonathhdoning that already!
08:39.53pranavhello everuone
08:39.55|Vulture|drumkilla: but thats the 3rd time in a year... not too bad
08:39.57jonathhfasinating stuff
08:39.58drumkillaI went on a coding frenzy this week
08:40.22|Vulture|its amazing you learn a little then look back at your code like... dear lord what was I thinking?
08:40.26jonathhjust trying to decide what actual hardware to buy.. for the first way of playing with actual hardware
08:40.28pranavtell me the programing that we use in the extensions.conf is written in which language
08:40.45drumkillaits not a language
08:40.46DaLionin asterisk
08:40.47DaLion;)
08:40.52DaLionthey call it CLI
08:40.52|Vulture|jonathh: cheap PC and a x100p clone...
08:40.55DaLion;) j/k
08:40.58drumkillathe AEL ... asterisk extensions language
08:41.02drumkilla:)
08:41.03DaLionlol
08:41.07pranavok
08:41.17pinoADDL, asterisk dialplan definition language
08:41.17DaLioni love the wtf is this language when it crashes  for nothing
08:41.23jonathhyeah.. so far i got a shit box pc.. x100p clone... 1 cheapy sip handset.. 1 ata (probably the grandsteam)
08:41.27pinomaybe we can come up with something even better ... ;)
08:41.29|Vulture|jonathh: you mine as well get a nice IP phone though
08:41.30pranavso is there any prior requirement sbefor learning this language
08:41.47DaLionbah .. even on dual xeons 2 gigers u get problems
08:41.47drumkillano, it is fairly simple
08:41.49jonathhso far as in that is what we are gonna buy
08:41.58pranavok thanks
08:41.59DaLiona simple %%%#$@!# shit in diaplan can cause crashes
08:42.08smiley-is there any way to update fields in SQL (real time stuff) from extensions.conf ?   like realtime update  in the CLI?
08:42.20|Vulture|jonathh: okay Id recommend looking at IP500 or Cisco 7940 phones when you go to do a demo
08:42.23DaLionyes
08:42.23DaLionodbc something
08:42.25pinoDaLion: and if you can't get it, there's app_segfault! :D
08:42.28rikstaDaLion: you really shouldn't shit in your dial plan
08:42.33jonathhyeah the cisco look nicccce.
08:42.36|Vulture|I pitch the IP500s cause they are so cheap for what you get
08:42.39drumkillasmiley-: realtime doesn't have a way to do updates like that
08:42.48jonathhlemme make some notes :)
08:42.50DaLionhehe
08:42.54rikstajonathh: they are, i know :)
08:43.05|Vulture|jonathh: Cisco has the name, but Polycom IP500 is where it is at! just ask the channel
08:43.14jonathhi'll take a look
08:43.32DaLionx: realtime update sipfriends name bobsphone port 4343
08:43.33jonathhthey are abit pricey for me just now tho.. £200 compared to £70 for a shitty grandstream
08:43.36|Vulture|me saying all this and I have a 7960 in front of me... cause I haven't sprung for a IP600 yet
08:43.40DaLionthat would work
08:43.53DaLionsmiley- ? x: realtime update sipfriends name bobsphone port 4343
08:43.55jonathhso who make the IP500?
08:43.58smiley-drumkilla: ok..   since it's possible from the CLI and voicemailmain..   oh well..  I guess I have to do a small .c-application then..
08:43.59rikstai have a 7940, but i think it looks nicer than the IP500
08:44.00|Vulture|jonathh: true but they are very nice phones... Polycom
08:44.05rikstajonathh: polycom
08:44.06jonathhok
08:44.09jonathhlemme google i
08:44.11smiley-DaLion: from extensions.conf ?
08:44.13*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
08:44.23DaLionno in CLI
08:44.29DaLionasterick -vvvvvvvgrc
08:44.33|Vulture|the new firmware for the IP500s suck... anyone else try it?
08:44.35smiley-DaLion: yeah.. I know that ;)
08:44.39DaLionCLI> realtime update sipfriends name bobsphone port 4343
08:44.47smiley-DaLion: but I wanted to do that from extensions.conf
08:44.55|Vulture|I think its like 1.5 bootrom and 1.4.1 SIP...
08:44.56drumkillathat could be converted into an app easily
08:45.03DaLionoh
08:45.03DaLionyeah
08:45.11DaLionodbc set or something check the F. docs
08:45.12DaLion;)
08:45.28DaLiontry odbcget
08:45.32DaLionor something
08:45.41smiley-hehe..  ok ;)
08:45.46|Vulture|I just went through my DP today and make it much easier on CDR... so now its easy to track extensions
08:46.07jonathhforgive me if this is a obviously discovered answer. .but it has just struck me. can asterisk store the voicemail in a DB?
08:46.24|Vulture|files or users?
08:46.44jonathhthe actual voicemail files.
08:46.50|Vulture|users yes, files... not sure but don't think so
08:46.54drumkillain cvs head, yes
08:46.54jonathhi am aware you can store extention stuff in a DB
08:47.07jonathhoh version is the head at?
08:47.13|Vulture|*mutters* everything is in head
08:47.14jonathh^what version
08:47.31drumkillacvs head and the 1.0 branch are surely very different
08:47.31jonathhbut presumably the head isn't all the stable.....
08:47.41elricdoes ${DIALSTATUS} get set after the call is completed?
08:48.01|Vulture|yea I run v1-0
08:48.07DaLionsmiley-
08:48.07DaLionODBCput(family/key=value):
08:48.19DaLionStores the given value in the Asterisk database. Always returns 0.
08:48.20jonathhin what ways the 1.0 branch and the hdea different?
08:48.27*** join/#asterisk basta (~kqj@62-101-126-233.fastres.net)
08:48.38smiley-DaLion: ah..  that might be a way
08:49.11smiley-except for that I don't use ODBC for the real time stuff ;)
08:49.26bastaanyone using cisco 7912/7960 ? I've a problem with music on hold ...
08:50.01*** part/#asterisk pranav (pranav@221.128.181.21)
08:50.03|Vulture|7960 here
08:50.06jonathhwhat version is the head upto?
08:50.11DaLionexten => 111,1,ODBCput(sipfriends/password=1234):
08:50.12DaLionhehe
08:50.17DaLionnot sure anout that one
08:50.21drumkillacvs head isn't versioned
08:50.28jonathhoh right
08:50.36drumkillaat this point, anyway
08:50.41jonathhok
08:51.02drumkillathough it is commonly referred to as the branch for the future 1.2 ...
08:51.04jonathhis there a world of difference between 1.0.7 and the head?
08:51.10bastavulture, what version o * ? i think moh stopped working after an  upgrade to 1.0.6
08:51.11drumkillaor maybe the 1.1 dev branch
08:51.17drumkillabut most people just call it cvs head
08:51.38drumkillabasta: upgrade to cvs head
08:51.44drumkillabasta: no
08:51.48drumkillabasta: i meant 1.0.7
08:52.06*** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de)
08:52.30bastak, already compiled it, I'll try it (now i can't restart the servers)
08:52.39pinois anyone using *+spandsp to send faxes with a decent interface?
08:52.43jonathhi know this isn't really relivant.. but what distro are peeps using asterisk with?
08:52.48drumkillathere was a moh problem in 1.0.6, 1.0.7 should fix it for you
08:52.57drumkillajonathh: whatever you want
08:53.25bastahope it will, everyone hanging up calls, thinking the line is down !
08:53.28DaLionhttp://www.planetwayne.com/forums/viewtopic.php?t=211&sid=f308ab865deff3c83638ab9b15c40e2b
08:53.33DaLionthis guys an idiot
08:53.37jonathhwondered if there were any comments to be made about what distro to use if it is client side.. and only running asterosk
08:53.45DaLiongarage doror openein based on asterisk ?
08:53.47jonathhcurrent i have a red-hat, gentoo, and feudora
08:53.50DaLionhmm...
08:53.58DaLiongreat way to get robbed
08:54.00|Vulture|okay bedtime
08:54.01|Vulture|night guys
08:54.04jonathhit needs to be lean
08:54.06jonathhnight d00d
08:55.03*** join/#asterisk cinzas (~ashes@83.240.144.145)
08:55.07cinzasg'morning
08:55.16jonathhmorning
08:55.36cinzasI need help ... hehe
08:55.41jonathhdont we all :)
08:55.43cinzasast_readaudio_callback: Failed to write frame
08:55.50cinzasI'm getting this error
08:55.52jonathhi get that..
08:55.57drumkillawhat channel driver
08:55.58jonathhdunno what it means :)
08:56.08cinzasdrumkilla: sip
08:56.27cinzasand strange thins are happening ...
08:56.39jonathhwhen i get it.. the sound goes all shoppy
08:57.10cinzasI'm running * with 3 callcenters and voicemail
08:57.35cinzasAverage of 500-600 minutes by day
08:57.47cinzasWith a trunk SIP to Cisco Call Manager
08:58.06cinzasYesterday i got strange errors.
08:58.37cinzasSometime when a calls goes to a queue, it starts ringing a member, when he pickups the call is dropped and starts ringing in other queue member
08:59.05cinzasANd if he pickup, the call jump to ohter member. The ppl here are getting crazy with calls jumping phone by phone
09:02.05cinzasanyone ?
09:02.34*** join/#asterisk Newbie___ (me@60.48.55.141)
09:02.56jonathhsorry dude.
09:03.37cinzasmail time ;)
09:04.07jonathhi rekon your gonna be busy this weekend though :)
09:04.32cinzasbrbrbrbr
09:04.37cinzasPlease dont
09:04.39cinzaslol
09:05.03Newbie___hi, my X101P is successfully recognized by *, but i can dial out, * gives me " Unable to create channel of type 'Zap'"
09:05.09Newbie___please help
09:05.19Newbie___i mean can't dial out
09:07.13elricis there a way to check if a Dial()'ed call has been answered?
09:07.43jonathhask the person your calling? :)
09:08.03elricwell i wish extensions.conf could do that and then execute the macro
09:08.10elric:)
09:08.22drumkillaelric: there is an option to Dial to do that
09:08.39elricah ok drumkilla, i just need to read more on it then
09:08.42drumkillamight be only cvs head, though ...
09:08.48drumkillashow application dial ... look for it in there
09:09.45elricbecause Dial(${EXTEN}|60|M(macro)) executes as soon as the call connects. anyway i will read up on it
09:09.50elricand get cvs head
09:10.02drumkillaoh, just kidding
09:10.05drumkillathat's what I was talking about ...
09:10.46*** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl)
09:12.07elrichrm coz M() doesnt wait till the call is actaully answered
09:12.27elricalthought there is an option r
09:12.43elricringing but that stops all other callprogress info
09:18.41*** join/#asterisk Betu| (~betul@62.244.193.101)
09:19.52facek_i have this extensions
09:19.52facek_exten => _0XXXXXXXXX,1,Dial(SIP/202&SIP/203)
09:19.53facek_exten => _0XXXXXXXXX,2,Hangup
09:20.08facek_and why when SIP/202 answer the call.. the caller still have beep beep
09:20.09facek_?
09:21.00*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
09:28.22emitraxdoes anyone have cisco 7940 here? Im having in trouble with the registration process
09:28.39emitraxIm using 7.0 as firmware version
09:30.07*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
09:33.06rikstaemitrax: yeah
09:33.42rikstaemitrax: pastebin your sip.conf part        and the cisco .cnf
09:36.18emitraxI dont have a cisco.cnf
09:36.21tessierWith RealTime why aren't the contexts stored in the db also? Why do we have to still have [context] lines and then a switch statement in sip.conf and extensions.cnf/
09:36.23tessier?
09:36.51tessierAnd since each sip device has its own context in sip.conf do I still need a [sipphone] and a switch for each phone in sip.conf?
09:38.54facek_Why asterisk couldn't bridhe incoming call on ZAP. when SIP peer answered?
09:39.42TheEmperorif i've got 4 fxo ports, how do i make it so that in the extensions.conf file asterisk will use the free zap channel to dial out?
09:39.57TheEmperordial zap/2/3/4/5?
09:40.08drumkilladial zap/g1
09:40.13drumkillawhere all the channels are in group=1
09:40.27TheEmperorconsidering it's a 4 port fxo card?
09:40.43drumkillayeah, that should work
09:40.54TheEmperorg1 means any available channel?
09:41.07drumkillayeah, any channel in group 1
09:41.13*** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net)
09:41.25TheEmperordo i need to define that in zapata and zaptel?
09:43.08drumkillajust zapata
09:44.36TheEmperorthanks drumkilla
09:45.30TheEmperorfor signalling=fxo_ls for outgoing calls? and then channel=>1-4 is that correct?
09:45.41*** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
09:45.42drumkillasignalling=fxs_ls
09:45.59TheEmperorah..
09:46.01drumkillafxo uses fxs signalling and vise-versa
09:46.27TheEmperorsignalling=fxs_ls; group=1; channel=>1-4
09:46.30TheEmperoris that right?
09:46.34drumkillalooks good
09:47.03TheEmperorsweet. thank you :)
09:47.09drumkillanoooo problem
09:47.14drumkillayou can thank my insomnia
09:47.54TheEmperorshould i put group=1 on top instead? then signalling=fxs_ls ; channel =>1-4?
09:48.13drumkillaas long as its before channel
09:48.52*** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr)
09:48.55TheEmperorso in that order just now is ok?
09:49.18pbxjunkieanybody got any experience with zaptel and asterisk?:)
09:49.24drumkillayup
09:49.29cypromispbxjunkie: nobody
09:49.31pbxjunkie:D
09:49.55pbxjunkiemy boss is SO going to fire me if I don't get it working soon :D
09:50.37tessierAnyone know how context includes are handled when using realtime?
09:50.44drumkillapbxjunkie: support@digium.com
09:50.48tessierDo you still have to put the include => line in the extensions.conf file?
09:51.02Newbie___FXO uses fxs_ks or fks_ls signalling ?
09:51.17drumkillayou probably want fxs_ks
09:51.18tessierDepends on your fxo line. I suggest trying _ks
09:51.20cypromisbtw, /w 25
09:51.22cypromissorry
09:51.50*** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com)
09:51.53Newbie___ok, i am using _ks, but earlier TheEmperor said fxs_ls so was a bit confusing
09:51.58RoyKhmm
09:52.06RoyKpoor cypromis
09:52.14rikstacypromis: you have to wait to make coffee?
09:52.18rikstasounds like a shit job :P
09:52.28RoyKanyone that've used sip with call waiting, three-way calling etc?
09:52.30elricaccording to app_dial.c the M() actually should execute if status is set to ANSWER, does this mean if Zap/3 is the outgoing line and the caller is using Zap/1 it regards the bridging of Zap/3 and Zap/1 as ANSWER?
09:53.17*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
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10:10.35*** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr)
10:12.06pbxjunkiehmm... when asterisk first starts, I get (on the cli) Asterisk Ready.
10:12.06pbxjunkie*CLI>   == Primary D-Channel on span 1 up
10:13.42pbxjunkiethen when I first try to make a call I get : http://pastebin.ca/9592
10:14.08pbxjunkiebut ONLY the first time, every time after that, I get "unable to create channel zap"
10:18.27jonathhdoes anyone in here have any passion either for or against slackware? as a nice minimal.. distro for using asterisk with?
10:19.06rikstai used it when it was slackware 6, i bet it's changed since then.....back then the package management was terrible
10:20.03jonathhi dont use packages really... i mean i do with gentoo.. but i am a no thrills distro.. minimal so there is less to go wrong
10:20.16olivier_<pbxjunkie> try to better understand pb : CLI>pri debug span 1
10:20.24rikstagentoo shud be fine for you then
10:20.34pbxjunkiewhen I used channel 2 it worked
10:20.44pbxjunkieI only get this on channel 1
10:20.55pbxjunkiethe parameter after the first slash, is group or channel? Zap/x <---
10:21.08olivier_u can use both
10:21.35olivier_but if u use group for a call, and after an channel number, it can be busy
10:22.40pbxjunkieoh so.. since I don't care which channel to use I should use groups.. i see
10:23.18olivier_yep :)
10:26.03RoyKanyone here using a predictive dialler?
10:27.47pbxjunkiepredictive dialer? what's that?
10:29.08RoyKused for annoying telemarketers
10:29.10RoyKor by
10:30.33pbxjunkieand what does it do exactly?
10:30.55RoyKhttp://www.voip-info.org/wiki-Predictive+dialer
10:31.54pbxjunkieevil
10:32.02RoyK:)
10:37.08*** join/#asterisk jwitte (~jwitte@port-212-202-101-206.static.qsc.de)
10:37.14*** join/#asterisk gres (~serg@81.222.48.242)
10:43.58RoyKif used from agi, what is checkgroup supposed to return?
10:44.42*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
10:46.39elrici am coding a predictive dialer as we speak
10:46.47elricor well trying to
10:47.26RoyKelric: nice. I can possibly help
10:47.31RoyKboss wants one
10:48.43elricRoyK, cool i am doing one in perl right now but first I want to sort answering machine detection out
10:49.03elricthen do nifty shit like make a self learning GA predictive dilaer
10:49.14RoyKGA?
10:49.35elricgenetic algorithms
10:49.41RoyKheh
10:49.48*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
10:49.50elricevolution
10:49.54elric:)
10:49.55RoyKsounds reasonable
10:49.55RoyKI know what it is...
10:50.18elriccool use postgres as the backend, i dont like mysql
10:50.43RoyKalthough it might be getter to just use a fuzzy function or even nn, or what do you think?
10:50.59RoyKs/nn/ANN/
10:51.07elricyeah those are good options.
10:51.25elricbut first need to get answering machine detection sorted
10:51.44RoyKI'd start fuzzy. that should be good enough for a start, and a lot simpler than the two others
10:52.25RoyKhave you done some of the PD yet?
10:52.48elricnah just learning * yet
10:53.00elricwell i am getting alright at it
10:53.16elricwe are starting in May.
10:56.32pbxjunkiei find it incredible that people actually get paid to work towards making spamming easier
10:56.40elriclol
10:56.51pbxjunkieeither via phone.. or the web ..building address-collecting searchbots
10:56.51elricerr wrong window
10:56.54pbxjunkie:D
10:56.54elric:|
10:58.10RoyKpbxjunkie: boss tells me "we need a predictive dialer for our TM team"
10:58.12RoyKso I just do it
10:58.25pbxjunkieRoyK: fair enough
10:59.18elricpbxjunkie, its not only spamming, say our client is a company that needs to call people about pending bills/ overdue accounts
11:01.09elricits annoying but since its a job and it pays the bills
11:01.33*** join/#asterisk tessier (~treed@203.210.216.187)
11:02.04elricRoyK, may i message you?
11:02.20RoyKyep
11:02.26elriccool
11:04.34*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
11:16.24*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
11:18.59*** join/#asterisk FLeiXiuS (~Nick@pcp0011094024pcs.essex01.md.comcast.net)
11:19.44FLeiXiuSIs it possible to use asterisk without a provider?  I just want to run this locally throughout my house then have a certain extention ring a specific IP phone.
11:20.33pbxjunkieFLeiXiuS: everything is possible
11:21.07FLeiXiuSpbxjunkie: :-P, would this feature already be built into Asterisk?
11:23.44*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
11:23.44foobosFLeiXiuS, providers are optional. no need to connect it to the outside world at all
11:24.07foobosyou can even run asterisk with two soundcards and have two extensions that way
11:25.03FLeiXiuSSound Cards?  I thought it would be using Ethenet ?
11:25.15FLeiXiuSHence NIC.
11:25.18pbxjunkieFLeiXiuS: yes. ethernet. it's already "built in"
11:25.59FLeiXiuShmm i'll have to keep doing my research thanks
11:27.59pbxjunkieFLeiXiuS: http://www.voip-info.org
11:46.02jonathhanyone know of good sip handset reseller in the UK
11:58.29*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
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12:08.45*** join/#asterisk Newbie___ (me@60.48.55.141)
12:08.48Newbie___hi all
12:09.11Newbie___exten => _95.,1,Dial(Zap/g5/${EXTEN:2},30)
12:09.21Newbie___can anyone please tell me if the above is fine
12:10.17RoyKfine for what?
12:10.18RoyK:)
12:10.29Newbie___RoyK: to make an outgoing call
12:10.38RoyKsure
12:10.41RoyKwhy not...
12:10.50RoyKonly 30 secs might be a little low timeout
12:10.53Newbie___i keep getting  "Unable to create channel of type 'Zap'"
12:11.02RoyKpastebin full debug output
12:11.06RoyK~pastebin
12:11.53*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
12:11.53*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th || Read bug guidelines before posting bugs or face deletion.
12:12.05Newbie___RoyK: http://pastebin.ca/9598
12:13.42WeezeyI have an SPA-3000 and the Line side is working great, but it's got a studdered dial tone.  Aside from having a message, what does that mean?
12:14.05Weezey(there's no mailbox= for it)
12:14.48*** join/#asterisk Mentat (~mentat@pcp01260498pcs.nhaven01.ct.comcast.net)
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12:15.54RoyKNewbie___: what hardware do you have_
12:15.55RoyK?
12:16.19Newbie___TE410P and just added X101P, X101P is suppose to be in g5
12:16.35RoyKpastebin zapata.conf and zaptel.conf
12:16.41Newbie___RoyK: ok
12:16.42RoyKhm
12:16.43RoyKreally
12:16.55RoyKx101p?
12:16.55elricthis is not working
12:16.56elric:(
12:16.56RoyKwhat is that?
12:17.03RoyKsingle pri card?
12:17.08*** join/#asterisk The_Duke (~the_duke@80.92.64.103)
12:17.10Newbie___is a clone FXO
12:17.13RoyKok
12:17.14The_Dukehi
12:17.17RoyKthen I have no idea
12:17.21RoyKI don't do analog stuff
12:17.36The_Dukecan someone help me connect a customer's cisco callmanager with my asterisk?
12:17.41elricdoes anyone know how to make macro execution wait till the call is actually answered?
12:18.23Newbie___RoyK: http://pastebin.ca/9599
12:18.39Newbie___errr
12:18.59RoyKsorry. can't say anything about analog stuff
12:19.19Newbie___RoyK: ok, tks
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12:40.43*** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
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12:44.17SuPrSluGNewbie___:exten => _95.,1,Dial(Zap/g5/${EXTEN:2},30). Why do you have it on g5? if its an x100p clone a Zap/1 is normal. And do normal pattern. matching. _9NXXXXXX etc...
12:45.34SuPrSluGNewbie___:Do you have 5 x100p clones?
12:46.33Newbie___i have 1 X101P and 1 TE410P
12:46.59PinholeWhat hardware do I need if I want to use * as an answering machine on my normal phone line?
12:47.45Newbie___span1-4 = g1-4. group5 = X101P
12:48.16ManxPower~doc
12:48.17jbotit has been said that doc is The command is "~docs", moron!
12:48.17bugbotdoc is assigned nothing and reported M3667, M2605.
12:48.19ManxPower~mailinglist
12:48.20jboti guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
12:48.20bugbotmailinglist is assigned nothing and reported nothing.
12:48.26Newbie___Pinhole: i think u need FXO to answer call
12:48.49ManxPowerPinhole, Asterisk does NOT make a good answering machine
12:49.01Pinholewhy not?
12:50.01_Briangood question...why not?  Isnt a answering machine just a basic version of voicemail?
12:50.13ManxPowerPinhole, It was never designed as an answering machine.  for example, imagine this: A call comes in, you don't pick it up in time and it goes to voicemail.  There is no way to monitor the message that is being left, if you pick up a phone connected to the line asterisk will nto stop recording the message.
12:50.27ManxPowernto == not
12:50.40PinholeMy current answering machine has the same trouble.  I have to unplug it.
12:50.52_BrianManxPower: you can purchase answer machine service from RBOC's and you cant screen calls either...
12:51.04_BrianManxPower: at least not with Verizon that is :)
12:51.09ManxPowerPinhole, you can buy an answering machine that works for like $20
12:51.27ManxPower_Brian, No, you are purchasing voicemail service, not answering machine service.
12:51.47PinholeI want to make my answering machine do "cool stuff". (read cool geeky toy)
12:51.55_BrianManxPower: all depends on your expectations :).....for me..I MUST screen my calls at home..... ... :)
12:52.14ManxPower_Brian, then you need a real answering machine.
12:52.16PinholeI don't screen calls, I usually answer on the first ring.
12:52.40ManxPowerPinhole, you also have to do some dialplan tricks to get asterisk not to pick up the line when the phone stops ringing
12:52.43SuPrSluGNewbie___:try dialling w/ just Zap/5
12:52.56Newbie___SuPrSluG: ok
12:52.58_BrianManxPower: that is why I got one :) ....hell, i dont have the cash to setup * as a answering machine at home....besides my wife would kill me or I would kill her while trying to explain how to get messages
12:53.45ManxPower_Brian, Aparently wives are technical morons.  Odd.  I wonder if it's being female or getting married that keeps them from learning anything tech.
12:54.04ManxPowerI suspect it's getting married.  Anyone that gets married has to be pretty stupid anyway.
12:54.15_BrianManxPower: ouch....
12:54.29Newbie___SuPrSluG: with zap/5, * pick up zap/5 from span1
12:54.51ManxPower_Brian, research the origins of marriage some time.  It was mostly about property rights.
12:55.26ManxPowerDid she come with a good dowry?
12:55.53ManxPowerIs she producing lots of kids to work as labor on your farm?
12:55.55Pinholedual ownership == marriage.
12:55.57_BrianManxPower: if dowry is defined as debt..yup :)
12:56.14_BrianManxPower: she does tend to the sheep, and milk the cows....so yes
12:56.29ManxPowerAnyway, you never hear "I want to use Asterisk, but it has to be easy enough for my husband to use."
12:56.33_BrianManxPower: cant get her on the tractor though...
12:56.46_BrianManxPower: rofl!
12:58.09_Briandoes anyone know if * has any type of audio detection or call progress detection?  I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF.  The problem I am having is that * will continue to the next step even before the remote party answers.  If I utilize a Dial string, then i use another channel......
12:58.11ManxPowerIt's not a popular view, but I think the whole gay marriage issue is silly.  ALL marriage should be abolished from a govt stantpoint.  Marriage should be personal/religious thing.  The govt should be concerned with contract law and that's how a "marriage" should be set up.
12:59.02ManxPowerMarriage should be replaced by a modified verison of the "LLP" or "LLC" that is common for businesses in the USA. 8-)
12:59.32_BrianOK..let me rephrase..."I want to use Asterisk, but it has to be easy enough for my kid to use."
12:59.34_Brian:)
13:00.00_BrianManxPower: now i know you are gonna say something about children... :)
13:00.33jakepdevBrian - there's a 3rd party module I was looking at for CPD
13:00.58bjohnsonManxPower: I thought you were into the GPL version of marriage .. it's all shared baby
13:01.00_Brianjakepdev: cool......got any urls?  i will check it out..
13:01.32SuPrSluGNewbie___:did u paste your zapata and zaptel files?
13:01.35ManxPowerGads!  Don't get me started about kids!
13:01.52Newbie___SuPrSluG: http://pastebin.ca/9599
13:02.26ManxPowerbjohnson, Yes, but we don't call it that
13:02.36_BrianManxPower: hell..we didnt mean to get you started about marriage..and look what happened
13:02.58_BrianManxPower: :)
13:03.07jakepdevbrian - http://www.voip-info.org/wiki-NVLineDetect
13:03.39_Brianjakepdev: thank you sir..
13:03.43jakepdevnp
13:03.56jakepdevif you try it, let me know if it works...
13:04.33_Brianyup..i will be looking at it shortly..got a meeting and then i will start testing..
13:04.49jakepdevtnx - /msg me please
13:05.06jefreyit seems like when A calls B, A transfer B - C, in CDR, it shows 2 record, 1 is A to B , another is B - C (shouldn't it be A - C) ?
13:07.19Delvarjefrey: no, CDR is correct
13:09.26*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
13:09.58*** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz)
13:10.28cervajshi, plz what is new method for playing mp3 (not mpg123)?
13:10.30SuPrSluGNewbie___:in zaptel fxsks=125. i'm not sure if that relates directly to the channel or # of cards.try w/ =1
13:10.44cervajsi have someone who can code support for ogg vorbis
13:10.52jefreyDelvar: hmm.. is a flaw if the transfer is PSTN - PSTN
13:10.55cervajsbut he doesnt know asterisk
13:10.55Newbie___SuPrSluG: ok
13:11.09jefreyDelvar: how am I going to bill B if B is a not a user?!
13:11.10cervajsfor MoH
13:11.48Newbie___SuPrSluG: i did that before i use 125, X101P crashed with Zap1 in span 1
13:12.02*** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com)
13:12.10Delvarjefrey: thats how it works in normal PSTN calls, if you get a PSTN call into your houm, then transfer that call out, YOU get chareged for teh transfer not the caller, thats how it works....
13:12.31Newbie___guys from mailing list adviced that even i only use 2 span, span 3 and 4 must be reserved
13:13.03jefreyDelvar: hmm.. you sure bout that? how can a receiver be charged?
13:13.18Delvarjefrey: they cant!
13:13.21Newbie___damn, is there a way to tell my wife to get off my back when i am working
13:13.22jefreyDelvar: B receives a Call from A, he was transfered to C and he to be charge?
13:14.43Delvarjefrey: A calls B, B talks to A, B tansfers A to C, C talks to A. in this, A is charged for call to B, B is charged for call to C.
13:15.06Delvarjefrey: thats exactly how it should work
13:15.25jefreyhmm
13:15.38jefreythat's assuming if B is one of sip user
13:15.40Delvarpm ->
13:16.35*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
13:16.42newlhowever, in his original example, A is transferring B to C which implies a second call from A to C (conferenceing all three, A drops out, B and C remain).
13:17.08newlA party placed two calls.
13:17.16PinholeManxPower, if I was to go against your advice and try it anyway, what would be the minimum (cheapest) hardware I would need to pull off an answering machine?
13:19.39*** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca)
13:21.48jonathhnewbie.. but i'd say pc.. asterisk... fxo card.
13:21.56*** join/#asterisk vinsci (~vinsci@dsl-sjkgw2jb1.dial.inet.fi)
13:22.13jonathhi think it is an fxo.. the one that geos from the pc to the phone line
13:22.52Newbie___jonathh: yes, is a FXO
13:23.24jonathhthanks
13:23.53*** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk)
13:24.08PinholeIt looks like FXO cards fit inside another card.  Can you just by a FXO card?
13:24.10Newbie___jonathh: why do u thank me for ?
13:24.21jonathhclarifying :)
13:24.29Newbie___ohm ok
13:24.36jonathhi think the x100p is a clone card
13:24.40jonathhthat does only that
13:24.40Newbie___yes it is
13:24.48jonathhso for 1 line.. at home.. tis fine
13:24.59*** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com)
13:25.00ManxPowerPinhole, Digium sells the TDM400P with FXO module.  You can still get cheap clone X100P cards, but they are no longer being made, so you won't be able to get them at some point in the future.
13:25.36cervajsdo you someone using MoH+mp3?
13:25.56Newbie___take ManxPower advice , go for digium TDM
13:26.30ManxPower"Microsoft Releases Public Beta of Data Protection"  It's not April 1, is it?
13:26.45jonathhlol
13:29.10darkskiezI saw an x100p clone for sale for £9.99
13:29.39darkskiezmight as well buy a TDM400 card with an FXO and FXS module.
13:29.40Gand_DJyeah, they are standard dialup modem cards, with MD3200 chipset
13:29.53Gand_DJwouldn't mind getting couple
13:30.07JerJerCaveat Emptor
13:30.12darkskieztelappliant seem to be only have FXS modules for the TDM tho
13:30.32darkskiezi hear the x100p cards are echotastic
13:30.37JerJermost X100P clones have horrible echo problems and some cannot deal with caller*id (on-hook audio)
13:30.58jonathhis that why the clones are sooo much cheaper
13:31.10*** join/#asterisk iq (~iq@207-224-100-229.omah.qwest.net)
13:31.11jonathhi mean sub £10 compared to ~£90 for a TDM and the correct module
13:31.27Newbie___anyone from www.iax.cc
13:31.57JerJerjonathh:  you get what you pay for
13:31.58darkskiezhttp://www.myphonecall.co.uk/voip/telephonycards/oem/default.aspx
13:32.03darkskieztheyve got em for a tenner
13:32.28jonathhsee you really cant complain about a tenner
13:32.55*** join/#asterisk BBRodriguez (~alex@pD956341D.dip.t-dialin.net)
13:32.58darkskiezjonathh, can i borrow a tenner off you at the next slug meet?
13:33.19BBRodriguezCan anybody confirm www.voip-info.org is down ?
13:33.33darkskiezNoooooooooo
13:33.39jonathhyou sure can
13:33.57darkskiezConnected...Waiting for.....
13:33.58jonathhhmmm dont seem to be work
13:34.04darkskiezworks for me
13:34.14jonathhit will be down with all the peeps in here checking!"
13:34.44darkskiezjonathh, I thought this was #scotlug, ooops. nevermind :)
13:34.50Newbie___is slow
13:35.02jonathhlol
13:35.09jonathhno skint flints here!
13:35.20jonathhyeah confirm... it worked finally!
13:35.41Newbie___what time is it now in US mountain standard time or something
13:36.03jonathhfork knows
13:36.49newl9:36AM EST so take two hours from that for mountain.
13:36.52ManxPower7:30am MDT
13:36.54jonathhhow many UK peeps we got in here then?
13:37.06Newbie___ok, thank you all
13:37.52darkskiezmaximum length of 1000base-t on cat5e anyone? google tells me different things.
13:38.15ManxPowerdarkskiez, search Cisco's web site.
13:38.22jonathhnot far enough?
13:38.45Gand_DJI think 100 meters?
13:39.01ManxPowerMost *Base-T is 100 meters
13:39.02Newbie___www.voip-info.org site finally finish loading
13:39.39jonathhsomeone posted a voip-info article on slashdot or something?
13:40.00darkskiezapparently its 10meters for stranded
13:42.16*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
13:43.40kajtzu1000base-t will run on cat5 or cat5e upto 100 meters.
13:44.39*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
13:44.53*** join/#asterisk durex (~ironman@weber.anpa.org.br)
13:46.10*** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
13:47.24darkskiezkajtzu, source?
13:47.36kajtzudarkskiez: cisco
13:47.40dmccollumHello peeps
13:48.47tzangerugh fucking dialplan bugs
13:51.28*** join/#asterisk CoolCat_ (~god@200.170.109.217)
13:51.38CoolCat_morning
13:52.14*** join/#asterisk netofsickcoder (~netofsick@200.121.129.178)
13:52.41Gand_DJne1 have fwd setup in *@home/
13:52.53Gand_DJI can dial out, but when I try to call my box, I get busy signal.
13:53.08*** join/#asterisk moy (~kvirc@201.135.105.124)
13:53.14Gand_DJusing a different softphone not linked to * that is
13:53.32EgonisWhen I try the sample (ext1000) it works, however, pressing '2' or '#' does nothing... I tried setting dtmfmodes, but nothing changes... any ideas?
13:53.34*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
13:53.49dmccollumIs it common to have static when using x100p cards? If so is there a setting that would help eliminate the static?
13:56.31foobosdmccollum, that is most likely due to badly insulated motherboard. try changing the PCI-slot
13:56.55foobosand if you have soundcard in the same server/computer try removing it
13:57.04foobosalso changing the PSU might help
13:57.15jonathhyou oculd try changing the whole computer
13:57.19jonathhand if that doesn't work
13:57.21CoolCat_does anyone has any idea how * gateway skype (i dont really used * yet, only installed *@home, but didnt got any clue how conect skype as * doesnt manage the soundcard)
13:57.23jonathha differenct card ;)
13:57.44*** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz)
13:57.44foobosthere's no skype connectivity as of yet
13:58.17carloshhello, anyone from voipjet support here ?? Thanks.
13:58.18CoolCat_foobos =o/  i thought i heard something like it was feasable! =o(
13:58.38CoolCat_calos o que voipjet?
13:58.42fooboscoolcat_, all on design phase at the moment i think
13:59.18carloshCoolCat: trying to get hold of them.. someone from voipjet..
13:59.21CoolCat_foobos =o) probably it will use the soundcard or the skype api, thought!
14:00.40CoolCat_foobos see if my way of think is right...
14:00.49CoolCat_i can open a FWD account
14:00.58fooboscoolcat_, still a long way to working solution
14:01.04carloshcan someone recommend better providers than voipjet ? I'm outside USA, and want to terminate calls in Southamerica, need cheap and good..
14:01.08bjohnsonand limited interest
14:01.10dmccollumfoobos: Thanks I'll give it a try. There's no soundcard in the server. Just an Intel dual 10/100 card and two x100p's.
14:01.15*** join/#asterisk Nivex (kjotte@user-0c8hq5r.cable.mindspring.com)
14:01.17CoolCat_and setup asterisk to redirect to my pbx, correct?
14:01.39bjohnsoncarlosh: teliax, nufone, livevoip, etc, etc are all possibilities.
14:01.40CoolCat_carlos to pstn?
14:02.40carloshCoolCat: yes to pstn in Southamerican countries..
14:02.50carloshiax2, ilbc
14:02.58*** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca)
14:03.33CoolCat_carlosh probably try some provider from the country, thought!
14:03.48CoolCat_bjohnson fwd is good?
14:03.50jf_can someone tell me that xlite does not transfert anything (audio, ringtone) but iax does, why ?
14:04.13jonathhport forwarding issue?
14:04.23jf_jonathh u think so
14:04.26bjohnsonCoolCat_: for what?
14:04.43jonathhi dont knw... but i know that sip is needy ont he ports.. iax isn't so much so
14:04.46CoolCat_jonathh im in this issue atm!
14:04.53jf_k
14:04.58jonathhok..
14:05.04bjohnsonCoolCat_: it's free and does certain .. it is even pretty reliable considering the price
14:05.08CoolCat_bjohnson for sip-sip conections (i signed it, as it was for free) =o)
14:06.11bjohnsonjf_: afaik, xlite is sip and requires extra screwing around to get it to work through NAT firewalls
14:06.29jonathhthe conclusion i have come to is dont bother :)
14:06.40bjohnsonCoolCat_: sip-sip connections you can do directly and don't need an intermediary like FWD
14:06.47jonathhuse sip for intra netwrok.. and iax connectiosn to other asterisk boxes for internet connections
14:06.53bjohnsonjonathh: (unless you have to)
14:07.01jonathhtrue.
14:07.27jf_bjohnson: it was working yesterday night, im quite sure
14:07.30CoolCat_well, i would need a sip server dont i?
14:07.54bjohnsonCoolCat_: asterisk talks sip
14:08.39bjohnsonCoolCat_: but even without *, a SIP device can be configured to allow you to directly call another sip device
14:08.44CoolCat_bjohnson well, i am still learning...i didnt configured one byte under asterisk
14:09.26CoolCat_bjohnson i would like to have a sip number...so ppl can dial me, and i want to redirect it to my regular pbx!
14:09.39bannerman*yawN*
14:09.40CoolCat_bjohnson i also studing the zoomtel v3
14:09.40bannermanmornin
14:09.41bjohnsonconsider FWD as kind of like a registry service and free voip voicemail
14:09.57bjohnsonCoolCat_: no idea what a zoomtel is
14:10.19CoolCat_bjohnson fwd is need for me to get the sip #, isnt it?
14:11.08EgonisWhen I try the sample (ext1000) it works, however, pressing '2' or '#' does nothing... I tried setting dtmfmodes, but nothing changes... i'm really lost
14:11.29EgonisI also tried ext8500 for voicemail, but entering the voicemail box number does nothing either
14:11.50jonathhegonis.. i got this with sipgate i think
14:11.54jonathhthey blamed BT
14:12.37Egonisjonathh: I tried dtmfmode=inband just now.. let's see what happens
14:13.04CoolCat_bjohnson http://www.zoom.com/products/voip_products.html v3!
14:13.15jonathhi tried all sorts of combinations in the [general] and for the sip setting of the connecting device.. also nat= effects it..
14:14.10CoolCat_bjohnson but i dont know exactly how it will integrate with asterisk, and if it is need, thought!
14:14.15Egonisjonathh: worked
14:14.25Egonisjonathh: what is the default passwd for mailbox 1234?
14:14.26jonathhso where did you set it?
14:14.33jonathh4242
14:14.34Egonisjonathh: sip.conf
14:14.34jonathhi think
14:14.38Egonisjonathh: ty!
14:14.41jonathhyeah but in general?
14:14.45jonathhor for the sip device?
14:15.12Egonisjonathh: the sip device itself
14:15.22Egonisjonathh: how do I find the mailbox passwd? 4242 was wrong
14:15.29jonathhhmm might stop mucking about with exim for a sec.. and tr
14:15.29jonathhy
14:15.33jonathhcheck voicmail.conf
14:15.37jonathhfor that extension
14:15.43jonathhit is the first CSV
14:16.31jonathhegonis
14:16.34*** join/#asterisk Moc[NX] (~mochouina@64.235.196.24)
14:16.35jonathhso you are using inband?
14:16.38jonathhand it works?
14:16.43jonathhwhat provider are you using?
14:16.45Egonisjonathh: works great
14:16.49Egonisjonathh: Primus
14:16.57jonathhhmm lets see what they offer
14:17.16jonathhprimus.com?
14:17.25Egonisjonathh: funny, although the passwd is set to 4242, it says login incorrect
14:17.27Egonisjonathh: primus.ca
14:17.39Egonisjonathh: I'm switching to allstream tho
14:17.57jonathhwell i use sipgate..
14:18.02jonathhit gives you any uk area number
14:18.07jonathhbut seems the tones dont work
14:19.24*** join/#asterisk cj-rm (~cjrm@81-178-22-214.dsl.pipex.com)
14:19.26cj-rmhey ppl
14:19.40cj-rmHow're things on #asterisk today?
14:19.48jonathhhmm no tones
14:19.52jonathhcant get into voicemail
14:19.53*** part/#asterisk langals (~icechat5@196.7.14.183)
14:19.55Egonisjonathh: That's odd... I wonder why... have you tried dtmfmode=info?
14:20.01*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
14:20.02jonathhbloody sipgate!
14:20.08jonathhyeah.. but i'll try it again
14:20.24Egonisjonathh: modes (from what I know) are inband, info, rfc2833
14:20.32fenlanderjonathh: sipgate does not work in incoming DTMF
14:20.34*** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226)
14:20.51_THEEND_someone uses spandsp for fax?
14:21.03jonathhso how can you operate menu's etc? via sipgate..
14:21.05jonathhor cant oyu?
14:21.11*** join/#asterisk Nuttah (~andrew@amber2.interdart.co.uk)
14:21.21fenlanderjonathh: no, at least not last time I looked into it. They are aware of the problem.
14:21.23Nuttahafternoon guys :)
14:21.30jonathhany alternatives?
14:22.00fenlanderjonathh: there are several other ITSPS in the UK that will give you a local number, but none that I know of are free
14:22.17jonathhyeah that is what sipgate is good for :-/
14:22.31jonathhi dont mind shelling out once i have proved the tech.. but not while i am still playing
14:22.46Nuttahdoes anyone here have experience with DECT units connected to sipura adapters? and have ever experience random tone sounds being generated mid call
14:23.44Nuttahaye, geographical numbers are never free from what ive found as well/
14:24.08jonathhsipgate is..
14:24.19jonathhmaybe i should quit moaning ;)
14:24.46Nuttahsipgate is also about twice as expensive as my current call rate for national calls
14:25.08jonathhwhat do you uise?
14:25.08Nuttahif i recall correctly that is
14:25.23Nuttahatm i'm testing out with magrathea
14:25.25fenlanderFor outgoing you can't really beat call18899 :-)
14:25.41Nuttahlinky fen?
14:26.10fenlanderwww.call1899.co.uk/coip.php
14:26.13_THEEND_someone uses spandsp for fax?
14:26.13Nuttahta
14:26.20fenlanderoops voip.php
14:26.26EgonisI just added a voicemail box to voicemail.conf -- what else do I need to do to get it to answer w/ voicemail?
14:26.27jonathhit no working :)
14:26.43jonathhlol
14:28.00Nuttahinteresting.. they charge for accounts fen?
14:28.15fenlanderno - just for calls iirc
14:28.21NuttahEgonis: you will need a voicemail line in extensions
14:28.24fenlanderoutgoing only
14:28.35Blissexfenlander: jonathh: in the UK Gradwell.com is free for the 1st 3 months, then  £4/month, and they give local numbers almost anywhere.
14:29.04Nuttahhmm I wonder how they can do that
14:29.16cj-rmhow do I get asterisk itself to establish a call between two of its extensions?
14:29.23cj-rmThe extensions are SIP phones
14:29.40jonathhhttp://www.gradwell.com/
14:29.41jonathh?
14:29.41EgonisNuttah: ty!
14:29.43Nuttahnotranfer=yes.. or is that iax..
14:30.20Blissexhttp://VoIP.Gradwell.com/ too.
14:30.28cj-rmIs there not some kinda spool that I can dump a file into to establish a call??  Or is there a pipe or socket I can connect to??
14:30.48BlissexGradwell resell Magrathea more or less, and they are fairly reasonable. They support IAX too, not just SIP.
14:31.08Blissexcj-rm: are your questions serious?
14:31.37Nuttahcj: adding a canreinvite=no will make asterisk manage transfers i believe
14:31.44Nuttahcj: into sip.conf
14:31.58cj-rmBlissex: sure are :)  Are they n00besque?
14:32.12EgonisNuttah: I added voicemail for ext611, worked.. but now 8500 doesn't work
14:32.15Blissexcj-rm: well, beyond your wildest dreams :-)
14:32.35Blissexcj-rm: all that Asterisk does is to switch calls between its extensions :-)
14:32.53Nuttahegonis: usepastebin and show us you conf
14:33.11Nuttahbliss: pedantic mofo :P
14:33.31cj-rmBlissex: Yeah I know :)  But I want a piece of software to tell Asterisk to call BOTH the extensions!
14:33.35Blissexcj-rm: and as to «pipe» or «socket» it uses usually VoIP, between IP phone.
14:33.48Blissexcj-rm: ahhh, thats completely different.
14:33.59Blissexcj-rm: describe then what you really want to happen.
14:34.03Nuttahseems we have a few UK guys here.. anyone use sipura and dect together?
14:34.06cj-rmBlissex: I know, its completely different, but thats what I want :)
14:34.17fenlanderNuttah: yes - never had a problem
14:34.24durexAsterisks... I just instaled Asterisk on a FBSD Box. All my desktops are with x-lite installed, and now I wanna that them talk with each other. Where do I start?
14:34.26Blissexcj-rm: asking intelligible questions is a somewhat important question.
14:34.44Blissexdurex: you need to do two things... ooops. three
14:34.49Nuttahhmm never get tone beeps in the call Fen?
14:34.51EgonisNuttah: nm, figured it out
14:34.53durexBlissex what?
14:35.02Blissexdurex: the FIRST thing is to read an introduction to Asterisk :-0
14:35.14fenlanderNuttah: no - but I have heard other people mentioning that problem
14:35.20Nuttah!docs
14:35.20durexBlissex I did it... :-)
14:35.28Nuttahman whats the command again
14:35.29cj-rmBlissex: I want a program (I'm writing) to decide that Bob (Extension: 2000) and Sue: (Extension: 2001) need to talk.  I want my program to then tell Asterisk to ring bob and sue's phones, so that they can chat to each other
14:35.37Blissexdurex: the other two are to define your phones as entries in 'sip.conf' or similar, and then define them as related extensions in 'extensions.conf'.
14:35.56Blissexcj-rm: that's a rather odd requirement indeed.
14:36.02cj-rmBlissex: Both my extensions are SIP phones
14:36.14jonathhim curious to know why you want auto initiated calls?
14:36.16Blissexcj-rm: I suspect that the best way would be a special purpose extension/plugin.
14:36.41durexBlissex and how do I do it?
14:36.43durex:P
14:36.46Blissexcj-rm: however you might be able to hack something together as a meeting room hack.
14:36.47cj-rmjonathh: Think scheduled calls :)
14:36.53Nuttahfenlander: out of curiousity.. what dect phones are you using?
14:37.02jonathhi like the idea..
14:37.03Nuttahpersonally think its a dect problem
14:37.06cj-rmBlissex: meeting room hack?
14:37.15jonathhbut maybe.. an email saying... 'hey dont forget to call thingy wahtsit'
14:37.22Blissexcj-rm: well, you create a meeting room and they connect via the meeting room.
14:37.24jonathhthen maybe 'click here to initiate the call'
14:37.33fenlanderNuttah: BT Synergy range
14:37.45Nuttahright using the BT ddiverse range myself
14:37.53Blissexcj-rm: but the idea is that in general Asterisk is sort of passive, and switches calls only on request.
14:38.00Nuttahcause i'm a cheapskate :)
14:38.21fenlanderNuttah: are they dtmf tones that you hear?
14:38.27jonathhbut astersik COULD call party 1.. and on party 1 answering.. call party 2?
14:38.35Blissexcj-rm: however, it could be something like call extension A, put it on hold, and transfer it to B.
14:38.47cj-rmjonathh: Thats exactly what I want :)
14:38.54NuttahFenlander: yes seemingly at random as well
14:39.11cj-rmBlissex: I guess thats exactly what I want
14:39.13Blissexcj-rm: but I doubt there is any logic that does that already. In part because it is hard to tell phone A to call B.
14:39.33fenlandercj-rm: use a call file?
14:39.44cj-rmfenlander: a call file?
14:39.47jonathhi can see some wisedom to this
14:40.16cj-rmjonathh: me too :)  Now how do I get Asterisk to do it?
14:40.18fenlandercj-rm: http://www.voip-info.org/wiki-Asterisk+auto-dial+out
14:40.33jonathhdunno.. me new to all this :_)
14:40.39jonathhbut if asterisk can initiate calls
14:40.55jonathhi see address books on the desktop.. adn double clicking on a number.. calling that perosn
14:41.05*** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34)
14:41.13*** join/#asterisk CoolCat_ (~god@200.170.109.217)
14:41.20fenlanderNuttah: which dtmf mode are you using?
14:41.45cj-rmjonathh: yup, its a cool idea, heh? :) click-to-dial its a big thing, thats under-understood :)
14:42.07cj-rmthere are even sip specs defining best practices for it
14:42.18jonathhi was talking to a mate about this last night .. and i decided it w asn't possible
14:42.21jonathhbut i take that back
14:42.27Nuttahfenlander: RFC2833
14:42.29jonathhwhere?
14:43.32fenlanderNuttah: I use inband to the sipura - maybe there is a bug in the rfc2833 code?
14:44.13Nuttahpossibly.. i'll test that out.
14:44.20Gand_DJDoes this info look like something you'd put into the incoming section of * or outgoing?
14:44.24jonathhany idea where sample.call is?
14:44.41Gand_DJcontext=inbound
14:44.44Gand_DJdtmfmode=inband
14:44.50Gand_DJtype=friend
14:44.56jonathhincoming
14:45.33Gand_DJok. I just signed up with voipforcanada and they sent me that info (and couple more lines) and told me to add it to my sip.conf for * to work for making outgoing calls
14:45.39cj-rmRFC 3725 - best practices for 3pcc
14:45.44fenlanderjonathh: asterisk/sample.call
14:46.01jonathhhmm i cant see it!
14:46.09Blissexfenlander: just checking... you do remember that only the 711 codecs can do inband DTMF, and that both RFC2833 and INFO DTMF can be broken in various phones...
14:46.40Blissexjonathh: there are several sample calls files in that Wiki page...
14:46.44fenlanderBlissex: yes, I use 711 everywhere at the moment
14:46.56Nuttahhmmm prefer GSM over my main pipe
14:47.18fenlanderBlissex: something chaged between 1.0.5 and 1.0.6 that changes the negotiation of dtmf mode.
14:47.45Nuttahwhich reminds me... must upgrade *.. on 1.0.4 atm :P
14:47.54*** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br)
14:48.09cj-rmBlissex, fenlander, jonathh - Thanks for your help... I'm gonna go check this stuff out :)
14:48.35jonathhhey
14:48.38jonathhcj-rm
14:48.39jonathhdude
14:48.52jonathhif you create that file.. in the outgoing spool file
14:48.55Gand_DJHrm, for inbound, should I have type= friend or user?
14:48.57jonathhit will prob do what your adter
14:49.01jonathhafter
14:49.08jonathhneed to do some tests first :)
14:49.28Nuttahfenlander: regarding 1899, seems voip service is only available for existing customers.
14:49.51fenlanderNuttah: yes, but you can become an existing customer by signing up :-) I did
14:50.01NuttahGand_DJ: user = inbound, peer = outbound, friend = both
14:50.13vaewynno friend = evil
14:50.16vaewyn:}
14:50.18cj-rmjonathh: I know I've been looking into it... Best of luck man.
14:50.19Nuttah:P
14:50.30cj-rmttfn
14:50.48jonathhl8rz
14:54.03PinholeI just had a really great idea:  You could use voice recognition to play the telephone game.  You say something and * interprets it, says it to another * box and when it gets back to you, you'll be amazed at what comes back!
14:56.02Gand_DJWhat is usually better.. making an outbound [] entry using peer, and then making an inbound [] entry using user.. making 1 [] entry using friend... or making an outbound & inbound [] entry using friend in both?
14:56.30Nuttahwell thats your call tbh
14:56.37Nuttahthere are risks using friend
14:57.14Nuttahif you have definite directions for your connections. then always best to use the user/peer approach
14:57.27*** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca)
14:57.45*** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2)
14:58.03jf_why when i dial an sip channel i got  auto-congestion everyone is busy/congested at this time
14:58.20Gand_DJI'm trying to configure asterisk@home for outbound & inbound. fwd works for outbound, along with simpletel... but fwd inbound gives me busy signal when I call it.
14:58.40Gand_DJusing context=from-pstn
14:58.46Gand_DJfor inbound []
14:58.57jf_unable to create channel type sip
15:00.41Nuttahi'm probably not going to look at this gand_dj.. currently getting harrassed by customers, but use pastebin if you want to give ppl access to you current conf
15:01.10*** join/#asterisk netofsickcoder (~netofsick@202.154.225.74)
15:01.59*** join/#asterisk heison (~heison@p85.n-lapop06.stsn.com)
15:02.01jf_why when i dial an sip channel i got  auto-congestion everyone is busy/congested at this time
15:02.16heisonis CVS head currently broken?
15:02.45*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
15:03.19heisoni get "ast_destroy is deprecated, use ast_config_destroy instead!" and a core dump
15:05.05CoaxDheison: Hah. Someone'll find it in 5 minutes.  Dont worry.
15:05.08*** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no)
15:05.11Gand_DJhttp://pastebin.ca/9608
15:05.32Gand_DJshows extensions.conf and extensions_additional.conf
15:05.36heisonit has been like that since yesterday :( and i wasn't sure if it was me...
15:05.49CoaxDheison: Hmmm.  Wonder if it could be in a module
15:05.49jonathhhey peeps.. i am playinh with dial-outs.. can i get it to route the outgoing call over iax?
15:06.04Gand_DJif your provider supports iax
15:06.11CoaxDheison: did you rm -rf /usr/lib/asterisk/modules before you did a 'make install'?
15:06.12jf_why when i dial an sip channel i got  auto-congestion everyone is busy/congested at this time
15:06.19jonathhwell it would go over the intenet to my mates ia box :)
15:06.44heisoncoaxD: haven't try that yet... hang on
15:07.01Gand_DJI think as long as you setup the outbound properly, and they setup inbound properly for iax.. I think it would work
15:07.04Gand_DJI'm kinda new at this
15:07.05jonathhprlbem is.. i want it to start the first call.. on a local sip.. then connect it to another sip over iax..
15:07.29*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
15:07.29*** mode/#asterisk [+o bkw_] by ChanServ
15:07.57CoaxDheison: The thing is, when you install new versions of asterisk, if the particular module code doesnt get updated, there's a chance that an old module wont get overwritten. (Tho I think that nowadays, the makefile circumvents that from happening, so thats probably NOT the issue)
15:07.59Gand_DJwell I think IAX from softphone -> asterisk works.. then * -> * can be over iax... then * -> softphone can be sip again
15:08.19jonathhhow do i specify that toh in the call thingy?
15:08.29CoaxDheison: Moreover, with a change like that, module code *would* get updated
15:08.51Gand_DJnot sure. kinda new to using *@home
15:09.43heisoncoaxD: that worked... but i had to move the codec_g*.so by hand
15:09.51CoaxDheison: AHHHH!!! :)
15:10.01CoaxDheison: WHO'S YOUR DADDY?!#$
15:10.01heisonlucky i didn't rm -f... phew
15:10.19jf_why when i dial an sip channel i got  auto-congestion everyone is busy/congested at this time
15:10.23CoaxDheison: Indeed!  Especially if you had custom stuff in there!
15:10.40heisonhmm... my DADDY? bkw_? kram?
15:10.40moydoes anybody has any idea why asterisk is stoping when i use musiconhold? i start asterisk with "asterisk -vvvvvvvvvvvvvc" and the end message before stoping is:
15:10.44moyFound new ID3 Header
15:10.44moyBeginning asterisk shutdown....
15:10.44moyExecuting last minute cleanups
15:10.44moy<PROTECTED>
15:10.44moyAsterisk cleanly ending (2).
15:10.55CoaxDheison: Yeah, those two qualify to be your daddy
15:11.01CoaxDheison: Maybe i qualify to be a cousin or something.
15:11.11heisonlol
15:11.13CoaxDheison: (thrice removed or something.)
15:11.24heisonthanks for your help...
15:11.29CoaxDheison: Welcome!
15:11.48CoaxDhate. api. changes.
15:12.53nestAranyone using monitor with queues? I'm using the monitor-join command
15:13.05nestArand it leaves me with 364 byte wav files
15:15.17*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
15:16.09moywhat does monitor does? i dont know that app
15:16.19*** join/#asterisk netofsickcoder (~netofsick@202.154.225.74)
15:16.21darkskiezrecord the call
15:17.05EgonisIn messages: Unable to open pseudo channel for timing ... Sound may be choppy (this is to name a few)
15:17.31CoaxDEgonis: insmod zaptel.o ; insmod ztdummy.o
15:18.10EgonisCoaxD: Non-existent
15:18.16EgonisCoaxD: Do I need to install zaptel?
15:18.37CoaxDEgonis: Um, yes
15:18.42*** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com)
15:18.44EgonisCoaxD: Actually, Zaptel is already installed
15:18.45jf_can someone help me about that : why when i dial an sip channel i got  auto-congestion everyone is busy/congested at this time
15:18.51EgonisCoaxD: I'm using kernel 2.6
15:19.13DrWho17jf_: no extension match?
15:19.14CoaxDegonis: You need to install zaptel else any timing functions wont work, and you wont be able to create meetme rooms, or such
15:19.31jf_it should.
15:19.54DrWho17well, watch the call come in from the console, this should tell you what is happening
15:20.06jf_exten => 103,1,Dial(sip/JF,20)
15:20.08EgonisCoaxD: adding those mods to my kernel config now
15:20.13jf_it should be o
15:20.27jf_i can see what is happening
15:20.36Gand_DJne1 know which countries require G729 to be licensed?
15:20.47CoaxDEgonis: If you can use 'zaprtc.o', use that instead of 'ztdummy.o'.  More accurate timing source
15:21.08CoaxDGand: Asterisk requires G729 to be licensed to enable it
15:22.08EgonisCoaxD: I modprobed zaptel and ztdummy, but still get the same error message about pseudo channel
15:22.18CoaxDegonis: does /dev/zap exist?
15:22.27EgonisCoaxD: let's see.. one sec
15:22.35EgonisCoaxD: nope.. :)
15:22.43CoaxDegonis: You never actually ran 'make install'
15:22.48CoaxDegonis: Or at least, not with that kernel
15:22.55EgonisCoaxD: I emerged it in gentoo
15:22.57CoaxD('make install' on the zaptel sources)
15:23.05CoaxDEgonis: Will you frickin gentoo users quit doing that? :)
15:23.15EgonisCoaxD: LOL!
15:23.23EgonisCoaxD: re-emerging zaptel for kicks
15:24.11CoaxDEgonis: WOO
15:24.17nestArmy internal search and replace read that as the more vulgar version..
15:24.22nestArmy mind rules
15:24.46Nuttahfenlander: you have use g729 for your 1899 calls?
15:24.54*** join/#asterisk cinzas (~ashes@83.240.144.145)
15:24.57cinzasHi !
15:25.00Nuttahbecause some reason its decided to use it
15:25.54*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com)
15:25.55jf_what mean RFC3389 support incomplete
15:26.11jf_turn off client if possible
15:26.55EgonisCoaxD: How do I restart udev?
15:26.58jonathhanyone in here familar with .call files?
15:27.36fenlanderNuttah: I use 711 over IAX2
15:27.52*** join/#asterisk lancey (Shady@support.net1.cc)
15:27.54lanceyhi guys
15:28.05lanceyanyone knows how to set the inter-digit timeout of LinkSys PAP2 ?
15:28.06*** join/#asterisk kairo (~kairo@200.251.61.124)
15:28.22kairohi. The asterisk is one gatekeeper too?
15:28.34lanceykairo yes it is
15:28.38*** part/#asterisk Egonis (~chultay@69.194.211.129)
15:28.40fenlanderjonathh: what is your problem?
15:28.42lanceyfar more than a gatekeeper
15:28.52lanceynoone here dealing with LinkSys PAP2?
15:29.12kairoAnd I I can use the asterisk as gatekeeper only?
15:29.19*** join/#asterisk netofsickcoder (~netofsick@stjhts23d054.nbnet.nb.ca)
15:29.29lanceykairo yes you can
15:29.49ManxPower~docs
15:29.50jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:29.50bugbotdocs is assigned nothing and reported nothing.
15:29.57ManxPower~mailinglist
15:29.58jbotfrom memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:29.58bugbotmailinglist is assigned nothing and reported nothing.
15:30.06*** join/#asterisk Egonis (~chultay@69.194.211.129)
15:30.55cinzasAnyone here using SIP phones widely ?
15:31.34jf_what mean RFC3389 support incomplete
15:31.35jf_turn off client if possible
15:31.43ManxPowerI've deployed about 15 SIP phones and am in the process of deploying 60 more.
15:31.48ManxPowerjf_, So do that.
15:32.16jf_server side or client
15:32.16cinzasManxPower: wich phone do you use ?
15:32.32kairolancey: thanks a lot.
15:32.37lanceyjf_ client
15:32.39lanceyobviousli
15:32.43lancey*obviously
15:32.44jf_k
15:33.29ManxPowercinzas, Polycom IP 300 and 500
15:33.36cinzasHmmm
15:33.39cinzasthanks ;)
15:34.26jf_which should be turn off rtp message on xlite
15:37.09lanceyanyone here with a LinkSys PAP2, last time :>
15:38.41ManxPowerjf_, no, in X-Lite "Transmit Silence = YES" turns off RFC3389
15:41.03*** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com)
15:41.21|Vulture|is it possible for voicemail.conf to accept 2 pager #s or do I need to write an AGI?
15:41.50ManxPower|Vulture|, Dude, /etc/aliases
15:42.13Gand_DJHere's a crazy question.. can you setup * to have 2 IVR systems, 1 set of prompts if call comes in from 1 trunk, and another set of prompts if call comes from trunk 2?
15:42.30EgonisCoaxD: zaptel is now working fine and is in /dev... I now have the following messages: unable to get our ip address ... Skinny disabled -- Unable to open /dev/dsp ... no such file or device (as my server has no sound board)
15:42.31jakepdevyes
15:42.48DrWho17Gand_DJ: sure, just send them to different contexts
15:42.55Gand_DJI want to setup zap for family IVR, and voip trunk for business ivr
15:43.06Gand_DJk
15:43.18ManxPowerGand_DJ, CONTEXTS!!!!!
15:44.03DrWho17hrm, is asterisk cdr csv format based on anything, or just custom to asterisk
15:44.08ManxPowerEgonis, Do you want to use chan_skinny?  Unable to open /dev/dsp is because you are trying to use chan_oss or chan_alsa (or it's autoloading)
15:44.30EgonisManxPower: Do I want chan_skinny? how do I disable the chan_oss/alsa? modules.conf?
15:44.49CoaxDOh fscking COOL.  There's a new article about my Variegated Streptocarpus yahoo group in the new GHA newsletter.  I'm dancing. woo.
15:45.13jf_manxpower : got it
15:45.49ManxPowerEgonis, You disable Asteirsk modules in /etc/asterisk/modules.conf  Skinny is a Cisco protocol for Cisco phones
15:46.34EgonisManxPower: no, no need for skinny.. how do I disable?
15:47.32EgonisManxPower: when I try to access ext 8500, using box # 1234 w/ pass 5678 or 4242, access is denied
15:49.01Nuttahhmm this is a long shot, as mobiles dont have carrier dependant number prefix's, is there any way to route outbound calls via various voip providers depending on the carrier?
15:49.04ManxPowerEgonis, noload =>chan_skinny.so  noload => chan_alsa.so noload => chan_oss.so
15:49.14*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
15:49.24ManxPowerEgonis, /etc/asterisk/voicemail.conf  issue a reload on the cli after you make changes
15:50.07ManxPowerNuttah, That's only the case in the USA/Canada.  In the rest of the world there mobile users use specific prefixes.
15:50.22EgonisManxPower: I left voicemail as is... should I change the pwd?
15:50.25ManxPowerIn the USA it does NOT cost more to call a mobile than it costs to call a landline.
15:50.38Nuttahnot USA sorry.. UK
15:50.48ManxPowerEgonis, I have no comment on that.  Asterisk sample config files are there to show you EVERY option available.  they are not designed to work.
15:50.50Nuttahcosts a fecking packet here
15:51.15Nuttahand MAnx ya wrong wit the number prefixs
15:51.24EgonisManxPower: lol.. okay, how do I delete the voicemail waiting for ext1234 manually?
15:51.25ManxPowerNuttah, that's because the CALLER pays when using mobile.  In the USA the person being called pays for the call (for mobile)
15:51.26*** join/#asterisk ronn (ronn@217.46.199.162)
15:51.32ManxPower~docs
15:51.33jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:51.35ManxPower~mailinglist
15:51.36jbotmailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:51.41ManxPower~RTFG
15:51.42jbotAnother variant of rtf*, 'g' refers to Google or newsGroups
15:51.43bugbotdocs is assigned nothing and reported nothing.
15:51.43bugbotmailinglist is assigned nothing and reported nothing.
15:51.43bugbotRTFG is assigned nothing and reported nothing.
15:52.05tzangerManxPower: no that's STFW
15:52.17Nuttahin the UK you you can have aa number on one carrier, and move the number over to another carrier
15:52.20ronnhi guys
15:52.31ronni just installed asterisk on rh el3
15:52.34tzangerjbot, STFW is Search The Fucking Web.  See http://justfuckinggoogleit.com/
15:52.35jbottzanger: please, watch your language.
15:52.42tzangerjbot, STFW is Search The F*cking Web.  See http://justfuckinggoogleit.com/
15:52.43jbottzanger: please, watch your language.
15:52.47tzangerjbot, STFW is Search The F*cking Web.  See http://justf*ckinggoogleit.com/
15:52.48jbot...but stfw is already something else...
15:52.48Wonkaha-ha
15:52.53tzanger~stfw
15:52.54jbotstfw is, like, Search the F|_|cking Web: http://www.google.com/
15:52.54bugbotstfw is assigned nothing and reported nothing.
15:52.54Wonkajbot: stfw?
15:52.55jbotmethinks stfw is Search the F|_|cking Wiki: http://voip-info.org/tiki-index.php
15:53.03tzangerjbot no, STFW is Search The F*cking Web.  See http://justf*ckinggoogleit.com/
15:53.04jbotokay, tzanger
15:53.08ManxPowerBTW, we can say FUCK here.
15:53.17shido6oh my ...
15:53.22jakepdevdouble oh my
15:53.22Wonkafork it!
15:53.23mishehumy virgin eyes!
15:53.30mishehuoooooowowowowowoww!!! ManxPower swore!
15:53.35Wonkastop rape, say yes...
15:53.40ManxPowerIt's not considered terribly polite, but we can say it.
15:53.42jakepdevi thought this forum was G rated
15:53.48NuttahManxPower: UK numbers are movable from one carrier to another
15:53.55ManxPowerNuttah, That must be new.
15:54.04Nuttahso I presume the innitial answer is no :P
15:54.15NuttahManxPower: not really.
15:54.19*** join/#asterisk brc-tux (~brc-tux@pD9E9A2F2.dip0.t-ipconnect.de)
15:54.19newlNumber portability isn't anything new. :)
15:54.26*** part/#asterisk brc-tux (~brc-tux@pD9E9A2F2.dip0.t-ipconnect.de)
15:54.32DrWho17Nuttah: portable here too
15:55.09jakepdev~STFW
15:55.11jbotsomebody said stfw was Search The F*cking Web.  See http://justf*ckinggoogleit.com/
15:55.11bugbotSTFW is assigned nothing and reported nothing.
15:55.28ManxPowerbugbot needs to be sprayed with poison
15:55.30jakepdevcase sensitive as well as profanity sensitive
15:55.40*** join/#asterisk bazzz (~baz@atlnga1-ar3-4-3-007-122.atlnga1.dsl-verizon.net)
15:55.59*** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net)
15:56.34jakepdevoh - my bad - it worked before
15:56.48Juxthello peeps
15:57.00jakepdevhello
15:57.19jakepdev~nickometer jakepdev
15:57.19bugbotnickometer jakepdev is assigned nothing and reported nothing.
15:57.19*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
15:57.34Zeeekgentlefolk...
15:59.20Gand_DJDoes this look right... I hope this is the right info to post.
15:59.21Gand_DJhttp://pastebin.ca/9608
15:59.25jakepdev~kill bugbot
15:59.27jbotACTION shoots a excited quark gun at bugbot
15:59.27bugbotkill bugbot is assigned nothing and reported nothing.
15:59.37Gand_DJincoming fwd calls get busy signal
16:00.39jakepdevGandDJ - IAX or SIP?
16:00.44EgonisAny suggestions for testing festival w/ asterisk?
16:00.47Gand_DJiax
16:00.53Gand_DJI can call out fine
16:00.57*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
16:01.01jakepdevcan you show your iax2 debug?
16:01.25Gand_DJwhat's the command again from root?
16:01.30*** join/#asterisk cp5 (~samy@chcgil2-ar7-4-3-040-086.chcgil2.dsl-verizon.net)
16:01.41jakepdevit's in the CLI
16:01.49jakepdeviax2 show debug - i think
16:01.56*** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net)
16:02.00BuckRogersgood morning
16:02.04ManxPoweriax2 debug
16:02.16Zeeekand iax2 no debug
16:02.20Gand_DJ[root@asterisk1 root]# iax2 debug
16:02.20Gand_DJ-bash: iax2: command not found
16:02.24ManxPower"help" is a good command to know, BTW.  It's spelled "help"
16:02.33jakepdevnow Gand - in the CLI
16:02.34jonathhnot in bash
16:02.36jonathhin the CLI
16:02.38ManxPowerGand_DJ, you putz.  Asterisk commands are done in the Asterisk CLI
16:02.49jakepdevasterisk -r
16:02.55Nuttahchrist manx... got a bug up ya ass as well? :)
16:03.00Gand_DJsorry..lol... been ages since I played with linux
16:03.00jonathhor asterisk -cvvvvvvvv
16:03.21Gand_DJok.. debugging enabled
16:03.27Gand_DJgoing to try to call my * box
16:03.48Gand_DJnothing shows up.
16:03.52Gand_DJseems to not even hit the box
16:03.58ZeeekManxPower - have you ever heard of a problem like this? A few days  ago most servers start becoming UNREACHABLE for one cycle, i.e., a few seconds then REACHABLE. It isn't the provider since several at once, so it's a network issue probably. When it happens, all flow stops to the connection. (ADSL)
16:04.04Gand_DJverbosity is at least 3
16:04.11BuckRogersQuestion, What is the largest calling capasity card that is supported by enterprise asterisk, running on sun solaris OS, looking into buying hardware probly need something that supports fiber a plus
16:04.12ZeeekThis happens once every 30 minutes or so
16:04.31Gand_DJI have port 4569 forwarded from router to *
16:04.31jakepdevGand_DJ - do a show peers
16:04.34Gand_DJok
16:04.34Zeeekreplaced router/modem, network= card
16:04.50Zeeektried every possible network test
16:05.04jakepdevis FWD listed?
16:05.11Gand_DJfwd/520214       65.39.205.121   (S)  255.255.255.255  4569      Unmonitored
16:05.28Gand_DJI'm using *@home
16:05.29algorithmnBuckRogers:  i've heard about that sbc card, but not fiber.. ;-(
16:05.35Gand_DJwant to see my setup through the amp interface?
16:05.51BuckRogershas anyone else had any large scale deployement experince here?
16:05.55jakepdevhmm..  looks like it registered
16:06.02BuckRogersAlgorithmn, thanks
16:06.13Gand_DJyeah. I can make outgoing fine
16:06.21Gand_DJwierd
16:06.30BuckRogersMr spencer are you watching the room?
16:06.37jakepdevoutgoing calls are made within your dialplan
16:06.56jakepdevyour incoming depends also on the config of iax.conf
16:06.59algorithmnBuckRogers:  lol.  there isn't much burocracy to circumvent if that one works...
16:07.13*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
16:07.15ManxPowerI need this http://story.news.yahoo.com/news?tmpl=story&cid=573&e=20&u=/nm/science_clock_dc
16:07.51jakepdevthat's great if it works
16:07.57ManxPowerZeeek, Yes, I've seen it.  It seemed to be to be an ISP issue.
16:08.24BuckRogersso has anyone ran asterisk on a sun solaris os
16:08.32BuckRogerswith fiber ethernet?
16:08.34ZeeekManxPower is there something like mtr for commandline?
16:08.56ZeeekI have them checking it but it's really hard to catch (every half hour for 10 sec!)
16:09.10ZeeekI got the two upstreams to see it
16:09.26Zeeekthey are now contacting the telco that owns the wires
16:09.41Zeeekanyway, it makes voip USELESS!
16:11.54Zeeekanyone know a network utility like mtr or pingplotter ?
16:11.55DrWho17BuckRogers: what's a large scale deployment?
16:12.05Zeeekbig people need a large scale
16:12.40BuckRogerstrafic at super "pop" on a national network
16:13.09nestArsuper size my deployment!
16:13.10BuckRogersDrwho17 needs to beable to handle super pop capistity
16:13.42DrWho17well, that is different per "super pop"
16:13.49newlindeed
16:13.50DrWho17how about calls/hour
16:14.04ManxPowerZeeek, run a ping and see if you drop packets during the UNREACHABLE"
16:14.24Zeeekdone it, yes
16:14.34Zeeekplus at the provider she saw packet loss
16:14.43Zeeekwhich is why she reported to upstream
16:14.44BuckRogersDrWho17 i think at 20,000 continous calls
16:14.47ManxPowerZeeek, It COULD be some other issue.  NAT timeout?  Crappy Router (my old Cisco Cablemodem/NAT Router could not handle many small packets, it dropped them on the floor every few mins during an IP call)
16:14.55Zeeekbut man, it's hard to troubleshoot a once in a while condition
16:15.17BuckRogersthe really high capisty where national network interconnections
16:15.19ManxPowerZeeek, Try trobuleshooting a problem that only happens every few WEEKS
16:15.21DrWho17BuckRogers: ouch, good luck with that
16:15.21ZeeekManxPower a) this is new and I've been running for a year. b) I swapped for different router
16:15.22fooboszeek, mtr is commandline.
16:15.49ManxPowerZeeek, Prolly ISP issue then
16:15.49Zeeekfoobos it won't make on my box with no win
16:15.52BuckRogersyeah i need ot contact a asterisk driver programer i think
16:16.05Zeeekwhat's the iptraffic prog again?
16:16.10Zeeekiptraf ?
16:16.14fooboszeeek, 0.26 version compiles with curses/ncurses atleast
16:16.16ManxPowerZeeek, SOME DSL modems allow you to see the line conditions.  Poke around the mfgr's web site.
16:16.24DrWho17well, you'll need 100's of asterisk boxes clustered together actually, you are better off buying a switch I think
16:16.35newlZeeek: if you suspect the fault is on a PRI or DSL, have the telco place QM on the service(s).
16:16.58BuckRogerswhy not have asterisk running on a sun micro machine with 24 64bit processors
16:16.59Zeeeknewl that's what is about to happen,but man this is gonna take forever to find
16:17.00DrWho174-port T1 cards for asterisk are $1500, each handles 96 calls
16:17.23ManxPowerI had a problem where my signal from the CO was too weak.  Showed up as packet loss
16:17.23BuckRogersyeah we already have one of those
16:17.25ZeeekThe telco HAD a monopoly, they may not be that cooperative with the ISP
16:17.26DrWho17BuckRogers: well you could do that, but Intel is much more economical
16:17.30DrWho17and powerful
16:17.46newlZeeek: nah, 7-10 days for the average QM will ususally show BER if there's problems.
16:17.50BuckRogersi think we may need to use a cisco cat switch to get it to ether net sip
16:17.56Zeeekin the meantime, I have to tell everyone, ok if you stop hearing me, wait thru 10 seconds of silence and I'll be back
16:18.16Zeeekfucking annoying to say the least
16:18.24BuckRogersDrwho what duel zeons or quads i dont think it would be that power fulll
16:18.26Zeeekand unusable for bizness to say the rest
16:19.00BuckRogershow much do you think markspencer charges for a day of his time?
16:19.16newlhow long is a piece of string? :)
16:19.30DrWho17how is going to help you? You need to have a better definition of your problem
16:19.35BuckRogersand how much does he like the night life, got to boogie
16:19.39darkskiezBuckRogers, How much you got? probably $1 more.
16:19.57BuckRogersi dont think he is an unreasable man
16:20.07BuckRogerswhy do you inply that he is darkskiez
16:20.28darkskiezbecause he gave his code to the hippies.
16:20.40BuckRogersright.....
16:20.41*** join/#asterisk jmacz (~jmacz@63.245.86.225)
16:21.20Moc[NX]BuckRogers: he is busy.  You better reach a consultant instead of mark itself.
16:22.09BuckRogersMoc[NX]: do you have any suggestions, i think if he knew what we are working on he would make time
16:23.06newlWas that 20k concurrent calls?
16:23.14BuckRogersyes
16:23.19Zeeeknewl what is BER
16:23.27BuckRogerscould be more in larger meto areas
16:23.28newlZeeek: Bit Error Rate
16:23.30Zeeekok
16:23.34newl~ber
16:23.35bugbotber is assigned nothing and reported nothing.
16:23.38Zeeekheh
16:23.47Moc[NX]BuckRogers: I dont see the need to talk to mark about that
16:23.48newlbugbot needs to die
16:24.12Zeeekthis is one of those "no one else is having this problem..." situations
16:24.17newljbot: BER is Bit Error Rate
16:24.18jbotokay, newl
16:24.20BuckRogersMoc[NX
16:24.27Moc[NX]He be in Toronto next week giving a conference at Von
16:24.31Moc[NX]or just email him
16:24.33BuckRogersi adgree but i do not know who else to talk
16:24.55Moc[NX]BuckRogers: , just call digium, if it need to get to mark, they will do that
16:24.58BuckRogersthat what will probly happen or ill call him
16:25.15Zeeekmaking mtr complains he can't find resolver library
16:25.56Moc[NX]Even if I had to do a 100k phone setup, I dont see the need to contact mark except telling him that I got this really cool sucess story ;)
16:26.00*** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca)
16:26.52jf_is there any to configure xlite that if im local to the * use subnet ip address, if im remote (internet) using the dns of *
16:27.03Moc[NX]salut jf
16:27.06BuckRogersit for the use of abstract hardware and signaling at that large cap level
16:27.09jf_Moc: salut
16:27.25newl20k concurrent users..that's definately a shirtload.  Would be easier to buy a switch that'd handle that number of users no sweat like AXE or if all you're doing is routing, a S12. hehe
16:27.30BuckRogersperferable sunsolaris
16:27.46BuckRogersits more than routing though
16:28.09Moc[NX]if I were you, I would stick with Linux
16:28.16BuckRogersits a whole knew level of voip trafficing
16:28.26Moc[NX]you will get all kind of weird problems if you try to use anything else
16:28.32newlWell, S12 is capable of hanging subscribers off of as well..it's just not primarily designed for that purpose.
16:28.39BuckRogersyeah but you can run linux on some sun hardware
16:29.02fooboseven if linux runs on sun hardware, its not necessarily a good idea
16:29.03Moc[NX]who care about sun hardware ?
16:29.17BuckRogershigh processing power
16:29.31BuckRogerslarge memory capasity
16:29.39algorithmnyou cannot use 120gb of system ram nor 180mb of proc cache using linux with sun hardware...
16:29.43Moc[NX]BuckRogers: Server price are soo low these day... just buy multiple of them
16:30.09foobosi'd just go with cheap opterons and link them together
16:30.10BuckRogersMoc that is an alternative
16:30.16foobosit would be pity if that one expensive box go down
16:30.17BuckRogersIBM blade servers
16:30.21BuckRogersect.
16:30.30Moc[NX]yep, and if 1 fail, only 1 fail
16:30.55*** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com)
16:31.04Moc[NX]if you need to do maintenance, you can ofload 1 blade, and do it on that one, and switch it back in production
16:31.05foobosonly thing is of course heat output, powerusage and rack space constraints
16:31.59*** part/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com)
16:32.01BuckRogersyeah it has some advantages, but that still doesnt address my problem of asterisk not being able to handle fiber cards?
16:32.09BuckRogersoc signalling
16:32.22denonBuckRogers: no OC cards, but it does handle a DS3 card now
16:32.25fooboswhy does asterisk has to handle that directly
16:32.52denonI cant imagine why you'd ever want to handle more than a single DS3 per asterisk switch anyway
16:33.09Moc[NX]same here
16:33.12denonunless you wanted to run an entire carrier solution on a single switch
16:33.14BuckRogersthere is reasons that i can not disclose
16:33.34denonyou're going to have to disclose them if you want us to help you with a solution :)
16:33.53DrWho17denon: routing calls via PRI
16:33.54BuckRogersi hear ya
16:34.30Moc[NX]1 DS3 per 1U server is a good ratio I think
16:34.31foobosbuckrogers, well you could buy an expensive Juniper router and run asterisk inside it (its basically freebsd)
16:34.53CoaxDMoc[NX]: Um
16:35.11denonI dont think asterisk is really designed to be a high-density termination server...
16:35.14DrWho17Moc: 2 DS3 per 2U then
16:35.14CoaxDMoc[NX]: Thats a hell of a lot to ask out of asterisk on a 1U server
16:35.37BuckRogerswho said anything about termination
16:35.41DrWho17denon: yea, probably not, I'm still investigating
16:35.56*** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
16:36.02CoaxDHell, there are people reporting good results w/ a quad T1 card in a dual Xeon
16:36.36Moc[NX]CoaxD: I got a Dual Xeon 2.8 1gig ram, PowerEdge 1850 (1U) and it could have being bosted
16:36.37DrWho17no transcoding
16:36.52Moc[NX]so we are talking of about 27k channel per Rack
16:36.53DrWho17just hairpinning calls around
16:37.07BuckRogershave you ran transcoding, or pri to sip signalling
16:37.18CoaxDMoc[NX]: Hmmm.  Wasn't aware that a xeon proc could fit in a 1U
16:37.21moyhi everybody, how can i increment the volume in wich the Playrecord() files are reproduced?
16:37.25bkw_this _. vs _X. is pissing me off
16:37.33CoaxDMoc[NX]: (Much less *2* Xeons.)
16:37.38Moc[NX]CoaxD: yes they do hehe
16:37.51CoaxDMoc[nx]: They must be making xeons considerably smaller than P3 Xeon days
16:37.56Moc[NX]check the PowerEdge 1850
16:37.58moybkw, whi?
16:37.58newlbkw_: go back to working on SMS then. :D
16:38.03CoaxDMoc: They were *huge*
16:38.10bkw_newl, waiting on modem to arrive
16:38.10bkw_:P
16:38.15Moc[NX]CoaxD: it not using cardrige anymore
16:38.28CoaxDMoc[NX]:  Socket based now?
16:38.33Moc[NX]yep
16:38.38CoaxDMoc[nx]: Much better. yes.
16:38.45BuckRogersall i need this asterisk sun machine to do is pass traffic
16:38.50CoaxDMoc[nx]: Those big ass cartridge monsters were for the birds. *lol*
16:38.58BuckRogersat high capsity=
16:39.08CoaxDBuckRogers: Its not just "passing traffic"
16:39.32BuckRogersWhat makes you say that?
16:40.05newlIf handling traffic is all that you want, considered SER?
16:40.45BuckRogersstill need to run a mysql database and api'modes also
16:40.55DrWho17newl: he's going to put something to handle SIP in a "super pop"?
16:40.55*** join/#asterisk [Outcast] (~bill@c-24-218-94-11.hsd1.ma.comcast.net)
16:41.16DrWho17newl: the calls will come in via PRI handoff, or SS7 Trunks
16:41.31BuckRogersDrWho17, exactly
16:41.40DrWho17asterisk is not a solution for this
16:41.46newlDrWho17: well, PoP implies termination.  He didn't say anything about termination.
16:42.13jmaczhi everyone, I have a problem with the atxfer function. I edited the features.conf en reload the "res_features.so" at the CLI but it doesn't work. Any idea?
16:42.17DrWho17newl: right, he wants to take calls in and send voip calls to their voip destination and terminate modem calls probably
16:42.27BuckRogersSuper pops have interconnections from one service provider to another
16:42.36DrWho17where asterisk makes the switching decision, but asterisk won't be good at that
16:42.43BuckRogersDrWho17, no
16:42.47DrWho17especially not for a deployment such as 20000 calls
16:43.13BuckRogersneeds to foward traffic or not foward traffic
16:43.24jmacznormal trasfer works ok but I can't get atxfer to work on 1.0.3
16:43.32[Outcast]DrWho17: you can do it with about 10 SGI servers
16:44.02DrWho17Outcast: well, they are connected to media gateways
16:44.11[Outcast]yep
16:44.25DrWho17which handle a lot of the heavy lifting
16:44.26_Briandoes anyone by any chance have a copy of the app: NVLineDetect ?
16:44.45[Outcast]you can still let asterisk handle the transcoding on those machines
16:45.09BuckRogersit needs to come in and out in the same signalling format..
16:45.19DrWho17well, I'd rather just get a switch that does it all you know
16:45.29BuckRogersthere not smart enough
16:45.40DrWho17if you are talking 20,000 simultaneous calls, you should have enough $$$ to buy one
16:46.14BuckRogerswe developed the programming to run in the envoriment no one else has it to sell so we developed it our selves
16:46.17DrWho17probably less then 2000 96-port digium cards
16:46.35[Outcast]convergent network has a level 5 switch that will do it I think
16:46.50[Outcast]big $$$ though
16:46.53DrWho17yea, or santerra
16:46.57DrWho17or metaswitch
16:47.04DrWho17or sentito
16:47.10DrWho17whatever, plenty of them
16:47.15[Outcast]convergent is easy to setup though
16:47.47newl$150k will get you an AXE switch capable of 32k subs outta the crate. :)
16:47.49CoaxDThere is not enough processing power within a 1U box to handle 2,000 calls, let alone 20,000 calls
16:47.54*** join/#asterisk goldenear (~goldenear@d149.dhcp212-198-168.noos.fr)
16:47.57DrWho17newlP oh, not bad
16:48.19DrWho17if it does SIP/ATM Switching/MGCP/h.248 I'll take it
16:48.24CoaxDhell, there's not enough processing power in a powerhouse quad xeon 4U boxes to handle 2,000 calls
16:48.28BuckRogersnah we figure we would need rack size equipment
16:48.30newlDrWho17: Alcatel isn't all that bad. ;)
16:48.31fearnorthere's telica 1U switch now ;)
16:48.40DrWho17yea, lucent bought them
16:48.47tclarkwhy does vm play the voice mails msg in reverse date sequence in the OLD msg folder ..
16:48.49fearnorcoax: it depends what exactly you are doing. single SER box can handle 2000 calls no sweat
16:48.55tclarkerr doesnt
16:49.13fearnorand it also matters if you are just doing mgc, in which case, you need to worry about call setup per second, not concurrent calls
16:49.16CoaxDfearnor: Hmm.  really? 2000 calls, 1 box?
16:49.20fearnorconcurrent calls take almost no resources
16:49.24fearnorcall setup/teardown does
16:49.39CoaxDfearnor: Hmm.  Well, a sip proxy doesn't really have to do any transcoding, i wouldnt think.
16:49.45fearnorthat's right ;)
16:49.46CoaxDfearnor: Nor does it actually have to process the data coming into it
16:49.51CoaxDfearnor: It just has to relay it
16:49.52fearnorthat's exactly right.
16:50.04CoaxDfearnor: Which basically makes it a data relayer.  Dumb mode ON.
16:50.10fearnorthis is why the world has a distinction between MG and MGC.
16:50.11CoaxDfearnor: 2000 calls @ 64kbps, tho.. Oww
16:50.17*** join/#asterisk Moc____ (~mochouina@64.235.210.66)
16:50.20fearnorit doesn't even relay the data.
16:50.27DrWho17newl: yea, can't find those on Ebay though
16:50.29fearnorit just CONTROLS THE CALLS
16:50.45newlDrWho17: hahaha no, you probably wouldn't. :)
16:50.46CoaxDfearnor: (if ser is taken out of the media path right after start of the sip conversation, yea)
16:51.04fearnorits all about differentiating media path from call control path
16:51.12fearnorktnxbye
16:52.30CoaxDfearnor: Heh :)
16:52.47[Outcast]msg if you know how to get in touch with anthem outside of mirc
16:53.54*** join/#asterisk eidolon (~eido@seawall.homeport.org)
16:54.59eidolonhihifolks.  i have a quick question.  i have a PBX that I scrounged from eBay about 10 years ago, and it's really starting to fall apart.  We have 12+ extensions feeding 3 POTS lines into the house.  I'd love to toss the whole thing and replace it with Asterisk, but i'm not sure how costly it would be.  I don't want to park PC's at every extension - have VOIP phones gotten cheap enough to get a dozen or so for, oh, < $500?
16:55.58lanceyLinkSys PAP2 costs $63
16:56.03lanceyand you get 2 POTS lines with it
16:56.04lancey:)
16:56.16eidolonHmm.
16:56.17reallost1How do you detect phone numbers that are out of service?  The T1 line never answers and doesn't seem to return a code.
16:56.18lanceyand it really works :)
16:56.45[Outcast]hey is possible to unlock the sip adapter on the linksys
16:57.01eidolonohhh.  that's pretty neat.
16:57.18eidolonso i could use the PAP2 as a gateway to POTS phones, and it'll talk to the Asterisk server?
16:57.19gambolputtygrandstream bt102 about $75
16:57.23eidolon(2 phones at a time)
16:57.30gambolputtybt101 about $65
16:58.07Qwelleidolon: You need the PAP2-NA, make absolutely sure it has the NA
16:58.15Qwellanything else will not work with astrerisk
16:58.17Qwellasterisk...
16:58.24[Outcast]has anyone been having problem with res_perl in the cvs head?
16:58.40eidolonokay.
16:58.52*** join/#asterisk jets (~brian@guardian.pmt.org)
16:58.59eidolonthat's good, i could get a couple pap2's and use existing analog pots phones, plus get some BT101's for the power-users (like me :)
16:58.59Qwelleidolon: those ones are generally locked to a specific provider, and its difficult, if not impossible, to unlock them
16:59.19eidoloni'll need an FXO card for the server to hook into the POTS lines from the service, yes?
16:59.20jetsIs there any commercial vendors supporting asterisk text to speach?
16:59.24jetsspeech even
16:59.24Qwelleidolon: correct
16:59.27eidolonokay.
16:59.29eidolongood :)
16:59.37Qwelleidolon: something like a tdm400p, or maybe a SPA3000
16:59.43[Outcast]voice geine
16:59.48QwellI hear the SPA's are good
17:00.00eidoloni was looking at the voicetronix openline4
17:00.07[Outcast]the spa's rock
17:00.10QwellI'm all for the tdm though
17:00.17lanceybye guys
17:01.08Qwelleidolon: if you were to use a TDM for the pots lines, it would also give you a trusty timing device for things like MeetMe()
17:01.10eidolonokay, last question.  i have a dual-PIII-700 rackmount server that would make an excellent host for asterisk (1gig RAM) - is that enough horsepower to handle no more than 4 sessions at a time?
17:01.16Hmmhesaysgot a guy telling me he's got a loop start pri
17:01.26Qwelleidolon: That would probably handle a little more then 4 at a time
17:01.30eidolonheh
17:01.37eidolonokay, good 8)
17:01.58eidolonoh good, and asterisk is even in debian Sarge.  life is good.
17:01.59Moc[Work]www.voncanada.com
17:02.04*** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net)
17:02.06|Vulture|eidolon: that would easilly handle it
17:02.16Qwelleidolon: people have had problems with the debian packages
17:02.21eidolonoh?
17:02.23Hmmhesaysyou should get 1.07 if you apt-get asterisk    and have testing as your default
17:02.26QwellYou'd probably be better off using the source, and compiling stuff yourself
17:02.39gambolputtyDoes Asterisk support a plus sign in front of an E164 number?
17:02.46eidolongetting compilation working on debian stuff is... tedious.  mostly because debian by default doesn't install all the supporting stuff needed for builds.
17:02.50eidolonbut maybe i'll try it.
17:02.52|Vulture|hmmm strange... spandsp doesn't seem to be saving inbound faxes
17:03.05Qwelleidolon: well, try the (testing) packages, and see how it works out
17:03.10eidolonokee.
17:03.35eidolonwhat VOIP client do most folks under linux use?
17:03.37Qwellcompiling is good, because you can switch to cvs head easily if you need to test anything
17:03.47QwellFor a softphone you mean?
17:03.53eidolonyeah.
17:04.02QwellI like iaxcomm
17:04.06eidolon(i'm VERY new at this, sorry for the n00b questions)
17:04.24eidoloni'd like to use the internal speaker/mic for a headset, and just call right through the PC.  not using an external handset.
17:06.15goldeneariaxcomm is nice indeed
17:06.49goldenearbut it doesn't support for call transfer or hold when native bridged :(
17:08.03*** join/#asterisk jmacz (~jmacz@63.245.86.225)
17:08.28*** join/#asterisk Mike (~mike@201.135.48.119)
17:08.59*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
17:08.59*** mode/#asterisk [+o bkw_] by ChanServ
17:09.09fearnorhrmph
17:09.14fearnorearly media is pissing me off
17:09.45*** join/#asterisk kmelillo (~Armitage@wghi.net)
17:10.30eidoloniaxcomm looks good.
17:10.31Hogiehttp://gallery.cyberjunky.net/Work_Pictures/P0009081  <-- Ever seen a Cisco box taped up like that?
17:10.35eidolonnot in debian packages, unfortunately :)
17:10.56eidolonhogie: that host fails.
17:11.48Hogieeidolon: works for me, and everyone else I give it too...
17:11.57fearnortape packaging++
17:12.08eidolon'gallery.cyberjunky.net could not be found.'
17:15.32goldeneareidolon, did you try the call tranfer function ?
17:16.03*** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com)
17:16.06eidoloni was just looking at the site and the sscreenshots.  i don't even ahve a server active yet.
17:16.14*** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net)
17:16.19goldenearok
17:17.43*** join/#asterisk techie (gus@asterisk.horizonte.us)
17:17.46goldenearBut I guess the call transfer issue of iaxcomm is an iax conception issue...
17:18.50eidolonin all honesty, i'd rather use a SIP phone.  but having a softphone can be nice too.  particularly if i can tunnel through to the asterisk server remotely. :)
17:20.06goldenearthat's the big bennefit of IAX
17:21.00goldenearbut the fact that the sig doesn't go thrue the server during a native bridging has some bad consequences ...
17:21.04eidolonwell, phase 1 will be getting a platform and asterisk running on it :)
17:21.20*** join/#asterisk asteriskn00b (asteriskn0@wsip-68-15-113-233.ok.ok.cox.net)
17:21.43asteriskn00banyone here using or had any success with the xten eyebeam softphone?
17:22.03eidolonhm.  guy on ebay selling 85 grandstream bt101 phones for $65 each.
17:22.08*** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com)
17:23.10nvrsworkcan you use a hostname for externip, ie. externip = my.dyndns.org
17:23.13tzangerManxPower: do you really think I'm that stupid?  (re: dialplan 'i' and DIALSTATUS bugs)
17:25.44*** join/#asterisk AngelGabriel (~angel@host81-133-190-29.in-addr.btopenworld.com)
17:26.31AngelGabrielThe whole concept of asterisk, has blown my mind, WELL DONE to the developers, and people that have taken thier time to make it work
17:27.17AngelGabrielI have just one query - is it possible to route calls out onto the internet? And if so, how do I pay for them?
17:27.34tzangerAngelGabriel: you have a voice over internet provider and send the calls to them, and they bill you for the minutes used
17:27.55*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
17:28.26Gand_DJare there any voip providers that offer iax for inbound? Some offer it for outbound, but all seem to be sip for inbound
17:28.42kmelillowould it be possible to host Asterisk on the net, and give subscribers extension numbers, and have them use Soft Phones to dial each others extensions?
17:29.24eidolonkmelillo: i don't see why not.
17:29.33eidolonthat's basically what skype does, isn't it?
17:29.42kmelilloI dont know.. heh
17:29.53eidolonwww.skype.com :)
17:30.02kmelilloI just installed Asterisk on 2 machines, and I am trying to get them to communicate together... via prefix dialing
17:31.00AngelGabrieltzanger, Thanks ... I'll now go google for the info I need! I'm in the UK, can anyone recommend a supplier? and can I have more than one supplier?
17:32.50tzangerAngelGabriel: you can have as many as you like.  :-)
17:33.18goldeneareidolon, skype is p2p based...
17:33.19*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
17:34.36goldenearkmelillo, do you plan to have many subscribers ?
17:34.37eidolontrue, but they use a central accounting / auth system, as well as a gateway i'd assume.
17:34.53kmelillogoldenear, 20 or so...
17:35.12goldenearok so should not be a problem
17:35.23kmelillowe do a radio show, and need a way to record, once we are all connected... can Asterisk do conference recording?
17:35.54bjohnsontzanger: I find it advisable to avoid asking other people if they think I'm stupid
17:36.12kmelillobjohnson, heh.. if you have to ask?!?
17:36.40bjohnsonAngelGabriel: lots of voip suppliers listed on the wiki
17:36.51tzangerbjohnson: :-)
17:36.55*** part/#asterisk Egonis (~chultay@69.194.211.129)
17:36.57bjohnsonAngelGabriel: maybe you should take a cold shower
17:37.05*** join/#asterisk Egonis (~chultay@69.194.211.129)
17:38.01bjohnsonkmelillo: I think it can record conference calls .. but you'd better google on that
17:38.32bjohnsonthe normal call conferencing app is called meetme .. but I understand there are one or two other public ones and a few private (commercial) ones
17:38.55drumkillathe standard meetme has a recording option
17:39.03bjohnsonbtw .. meetme is p2p based as well
17:40.20*** join/#asterisk Signuts (~signuts@209.172.11.54)
17:41.04SignutsThis is all speculation, but Anyone aware of gmail marking all messages from Asterisk-Users as spam? Are they trying to pull a fast one, Perhaps this is related to them starting a Asterisk-test on google groups.
17:43.16PinholeIt's because of all the ex-M$ employees working at google that don't know they've changed employers.
17:43.19jf_is it possible for * to call a vonage number
17:44.34Signutsgmail just recently (perhaps yesturday) starting marking every message for the Asterisk-Users mailing as spam on my gmail account.
17:44.45SignutsThis is rediculous.
17:45.03Pinholethat's why I use hotmail.
17:45.09*** join/#asterisk AntiPool (~bostjan@BSN-77-143-148.dsl.siol.net)
17:45.21Pinholethey don't have any ex-M$ employees over there.
17:45.38Signutsman
17:45.47AntiPooli'm reading asterisk docs and if u understand correctly i can use isdn card for a start ?
17:46.31goldenearif it's isdn4l or capi compatible, yes
17:46.44Egonishow do I listen to the hold music?
17:46.48AntiPoolwhat about on fbsd ?
17:46.55Egonisis there an extension I can dial?
17:47.13*** join/#asterisk jeffik (jefik@69.158.30.24)
17:47.18goldenearyou have to create an extension for that
17:47.27*** join/#asterisk Dutts (~dutts@81.168.70.41)
17:47.29Egonisgoldenear: Where would I find a howto on this?
17:47.29jf_is it possible for * to call a vonage number directly trough the internet
17:48.13goldenearjf_, why could not * do this ?
17:48.16sivana~tzanger
17:48.17jbottzanger is probably some kind of fcking idiot
17:48.17bugbottzanger is assigned nothing and reported nothing.
17:48.28Signutsjf_, yes, it's called SIP, you must have a vonage account.
17:49.13sivana~toyk
17:49.17sivana~royk
17:49.18jbotextra, extra, read all about it, royk is .no body
17:49.19bugbottoyk is assigned nothing and reported nothing.
17:49.19bugbotroyk is assigned nothing and reported nothing.
17:49.21jf_ok can u tell me the parameter to do that, or do u have a tut
17:50.10Duttshi guys how do I find out which version of * is running from the CLI? show version doesn't seem to give me a proper version number like 1.x.x
17:50.14sivanajbot: no, sivana is one of the brightest stars out there, ok?
17:50.16jbotokay, sivana
17:50.26goldenearEgonis, many howtos on voip-info.org
17:50.41tzangerjbot no, sivana is not one of the brightest stars out there
17:50.42jbottzanger: okay
17:50.45sivanalol
17:51.09tzangerdamn I was hoping you weren't paying attention
17:51.21sivana~sivana
17:51.22jbotrumour has it, sivana is not one of the brightest stars out there
17:51.22bugbotsivana is assigned nothing and reported M2515.
17:52.07Duttsmy show version says Asterisk CVS-HEAD-03/09/05-02:14:09
17:52.40RoyKbugbot: I'm indeed reporting stuff
17:53.00drumkilla~drumkilla
17:53.02jbotsomebody said drumkilla was the Asterisk v1.0-stable maintainer.  ph33r him.
17:53.02bugbotdrumkilla is assigned M2338, M3154, M3758, M3857, M3320, M3012, M2140, M2790, M2983, M3979, M3989, M1595, M3733, M2968, M3977, M2755, M3150, M2662, M3188, M2669 et al. and reported M2814, M4000, M3746, M3046, M3842, M3254, M3124, M3858, M3838, M3864, M3280, M3130, M3083, M3749, M3997, M3990, M3876, M3934, M3989.
17:53.14drumkillasoo mannnyy buuuggsss ...
17:53.22CoaxD~coax
17:53.23jboti heard coax is a wierdo who screws around with asterisk for fun
17:53.23bugbotcoax is assigned nothing and reported nothing.
17:53.47goldenearEgonis, try this : in extension.conf : exten => 1800,1,Answer exten => 1800,2,MusicOnHold
17:53.49CoaxD~botsnack
17:53.49jbotCoaxD: thanks
17:53.49bugbotbotsnack is assigned nothing and reported nothing.
17:53.55CoaxDjbot: welcome
17:53.58Duttscan anyone hear me? =)
17:54.11tzanger~~bugbot
17:54.12bugbot~bugbot is assigned nothing and reported nothing.
17:54.13jbot...but bugbot is already something else...
17:54.15AntiPoolanyone runing asterisk on bsd ?
17:54.19tzangerdamn
17:54.21Qwellumm...
17:54.23tzanger~jbot
17:54.24bugbotjbot is assigned nothing and reported nothing.
17:54.28jf_anyone can help me with sending call from asterisk to vonage number
17:54.29tzanger~~jbot
17:54.32Qwellhmm
17:54.33CoaxDantipool: I DONT USE ASTERISK IN BED!
17:54.41CoaxDantipool: er. I misread you. :) :) :)
17:54.45Qwelltzanger: we got the same idea then, heh
17:54.45AntiPoolhahaha
17:54.52*** join/#asterisk MeTaBSD (metabsd@BlackBox.black4est.org)
17:54.57MeTaBSDhi all
17:54.57tzanger~~jbot no, jbot is heading for a crash
17:54.59jbotI think you lost me on that one, tzanger
17:54.59bugbot~jbot no, jbot is heading for a crash is assigned nothing and reported nothing.
17:55.00jbotbugbot: okay
17:55.03tzangerhehehe
17:55.04Qwell~north
17:55.06jbothmm... north is up today.
17:55.08MeTaBSDi have compile problem with asterisk-addons :(
17:55.09*** part/#asterisk Dutts (~dutts@81.168.70.41)
17:55.13bugbotnorth is assigned nothing and reported M3457.
17:55.13*** join/#asterisk Dutts (~dutts@81.168.70.41)
17:55.14CoaxDantipool: Asterisk on BSD works fine.  Trouble is, zaptel for it aint that great, from what iv'e heard
17:55.15MeTaBSDapp_addon_sql_mysql.c:162:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given
17:55.21Duttshello guys can you hear me now?
17:55.26CoaxDantipool: Oh, i'm sure it does get better every day. But.
17:55.35Qwelltzanger: evil :p
17:55.49MeTaBSDi have mysql-server,clien,devel,shared-compat installed .
17:55.50QwellWhy did bugbot ignore me?  heh
17:55.53CoaxDantipool: My advice is just to run it on linux
17:56.08CoaxDantipool: (Contrary to what some BSDers believe, it won't actually kill you to do so.)
17:56.19AntiPoolCoaxD: linux - me no go,
17:56.27CoaxDANtiPool: Guess you're screwed then :)
17:56.28MeTaBSDhttp://www.pastebin.com/271685
17:56.33AntiPoolCoaxD: i only wonder if it can work with isdn card
17:56.35AntiPoolon bsd
17:56.41MeTaBSDcan you help me :)
17:56.42CoaxDAntiPool: what, BSD? Probably not.
17:56.55CoaxDAntiPool: The APIs for talking to ISDN stuff are completely different between the OSs, i would think
17:56.57Egonishow do I install more music on hold? b/c it appears that 'default' is the only one that works, when I set to 'loud' the extension is busy
17:57.10sivana~jobt
17:57.11bugbotjobt is assigned nothing and reported nothing.
17:57.12tzangerQwell: I need a way to get jbot to and bugbot to talk to each other
17:57.12sivana~jbot
17:57.14jbotsomebody said jbot was heading for a crash is assigned nothing and reported nothing.
17:57.14bugbotjbot is assigned nothing and reported nothing.
17:57.21Qwelltzanger: yeah...
17:57.24reallost1MetaBSD, did you check the zaptel-bsd mailing list?
17:57.30Qwell~test
17:57.31jbotTest Passed!
17:57.31bugbottest is assigned nothing and reported nothing.
17:57.32Qwell~test abc
17:57.34jbotTesting abc... ROM BASIC NOT FOUND$#$
17:57.34bugbottest abc is assigned nothing and reported nothing.
17:57.38Qwellwtf
17:57.44RoyKjbot: lart bugbot
17:57.45AntiPoolCoaxD: true, do you know anything about X100P cards ?
17:57.46QwellHow is that even there?
17:57.55MeTaBSDreallost1 im not on BSD im on Linux
17:57.58tzanger~~jbot no, jbot is jbot no, jbot is going recursive
17:57.59jbottzanger: I think you lost me on that one
17:58.03Qwell~M2501
17:58.09bugbot~jbot no, jbot is jbot no, jbot is going recursive is assigned nothing and reported nothing.
17:58.10jbotbugbot: what are you talking about?
17:58.10bugbotM2501 is assigned nothing and reported nothing.
17:58.16reallost1oh, antipool sorry.
17:58.19Qwell~~M2501
17:58.20bugbot~M2501 is assigned nothing and reported nothing.
17:58.21jbotokay, bugbot
17:58.25tzangerhahaha
17:58.34pgpkeysas long as the card is supported it should work. I have freebsd, asterisk installed (not yet configured) which is in ports, and as i said so long as your ISDN card is supported it shoudl work
17:58.36reallost1nm
17:58.38pgpkeyss/dl/ld/
17:58.40Qwellwait, is that a valid username?  heh
17:58.40tzangerjbot forget M2501
17:58.40jbottzanger: i forgot m2501
17:58.44Qwell~totallyfakeuser
17:58.46bugbottotallyfakeuser is assigned nothing and reported nothing.
17:58.49AntiPoolreallost1: why sorry ?
17:58.51Qwellguess so
17:58.56tzangerQwell: no it worked because you said ~~
17:59.08sivanabugbot should need a msg word, like ~bug M2515
17:59.15sivanas/msg/special
17:59.15Qwellsivana: yeah, likely
17:59.22sivanadont' ask how I screwed that one up
17:59.45Duttsif I downlaod the latest Asterisk v1.0.7 how do I upgrade my current install?
18:00.08sivana~M2515
18:00.11bugbotM2515 is assigned nothing and reported nothing.
18:00.11reallost1antipool, I confused you with MeTaBSD in a conversation.
18:00.25sivanaM2515
18:00.25bugbotM2515 is a tweak bug that is closed (markster): [patch] cleaned up cdr_mysql.c. It was filed by sivana and was last updated on 01-10-05. http://bugs.digium.com/bug_view_page.php?bug_id=2515
18:00.31reallost1AntiPool, I'm running asterisk on BSD on several systems.
18:00.34AngelGabrieldoes anyone in here use SIPGATE?
18:01.12AntiPoolreallost1: any expiriences with isdn on bsd ? i have only once isdn card, i'm poor and i want to try ip telephony :)
18:01.30tzangerI need an ISDN card that works in north america
18:01.34tzangerBRI of course
18:01.40*** join/#asterisk Dutts (~dutts@81.168.70.41)
18:01.50Duttshello guys
18:02.04AngelGabrielSIPGATE.CO.UK - they are a VoIP provider
18:02.20Qwelltzanger: mind a msg?  heh
18:02.39tzangerdepends on what it says
18:02.40reallost1AntiPool, which ISDN card?
18:02.44Qwelltzanger: its funny. :)
18:02.45tzangerI have been known to shoot the messenger before
18:03.35AntiPoolreallost1: truth is i forgot which one, and i lost dmesg, how can i list pci devices without scanpci ?
18:03.48pgpkeyspciconf
18:04.09pgpkeysthis is a *bsd box correct? (net,free,open)
18:04.50*** join/#asterisk Balu (~balu@foghorn.bartels-schoene.de)
18:05.30AntiPoolpgpkeys: correct
18:05.36pgpkeyspciconf -lv
18:05.42BaluHi everyone
18:05.55BaluJust a quick question...
18:05.57*** join/#asterisk chris78 (~dg1nsw@saturn2.franken.de)
18:06.06pgpkeysquestions are never quick
18:06.12Baluhm
18:06.19BaluYour answer is hopefully ;)
18:06.23Balulemme phrase it 8)
18:06.28*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
18:06.50*** join/#asterisk WGFreewill (~WGFreewil@24-75-221-174.miamfl.adelphia.net)
18:06.58*** part/#asterisk eidolon (~eido@seawall.homeport.org)
18:07.14BaluI am one of the guys who likes to start from scratch, so I thought to clean /etc/asterisk (containing the debian examples) and start with a clean sip.conf and extensions.conf
18:07.26Baluis that ok or will I miss something absolutely needed?
18:08.31Baluasterisk.conf probably :)
18:08.34pgpkeysif you have to ask that then you've not read the manual or any of the docs that come with it. claiming 'i like to start from scratch' is a little over the top
18:08.45AntiPoolreallost1: i have HFC-S PCI A Cologne Chip
18:08.52AntiPoolreallost1: ever heard of it ?
18:09.02Baluthe problem with the manuals and example config files is that they are cluttered with things I don't need
18:09.07Balu:)
18:09.09tzangerAntiPool: wow I bet that card stinks
18:09.18pgpkeysthen ignore the things you don't need
18:09.31AntiPooltzanger: why ? they have support for linux on page :)
18:09.33BaluI've red some docs and tutorials and now think I can start with my own configs
18:09.58tzangerAntiPool: nevermind
18:10.00chris78Balu: i started from scratch as well .. like noted in the asterisk-documentation-project .. its a good way i think
18:10.19AntiPooltzanger: but probably sux elephant's ass in real life :)
18:10.20Balupgpkeys: But then I will not know if my stuff is not working because it conflicts with some not needed example in some file I don't know yet
18:10.24pgpkeysyou still need to read the manual at the very least from end to end
18:10.29Baluof cours
18:10.30Balue
18:10.36Signutspgpkeys, end to end is easy
18:10.41*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
18:10.47pgpkeysSignuts: hehe
18:10.52chris78Balu: then step by step go through and taking look at the example-configs helped me understand
18:10.52pgpkeysok, start to finish ;)
18:10.56Signuts:)
18:11.15AntiPooltzanger: http://www.colognechip.com/isdn/controllers/hfc-pci-a.gif <- diagram of that card maybe you can tell if sucks ?
18:11.30tzangerAntiPool: I didn't say it sucked, I said it stunk
18:11.39pgpkeyschris78: excellent comment.
18:11.51Baluasterisk is fairly chatty on problems as I've seen so far, so it will help me too
18:12.17pgpkeysi've built more LFS style deployments than I care to count in the last 10 years. even I read the docs from top to bottom and look through the example configs and try to fathom what's being done.
18:12.29pgpkeysit's really the only way to do it if you're a 'start from scratch' guy
18:12.51BaluI was very active in LFS up to V4 - some parts are from me :)
18:13.01pgpkeysSignuts: or would that be bottom to top as well, depending on the language? ;)
18:13.15Balupgpkeys: if we are talking about the same LFS ;) (Linux From Scratch)
18:13.21pgpkeysi was also one of the LFS mirrors
18:13.25pgpkeysBalu: yes
18:13.26Balu:)
18:13.27pgpkeysvery same
18:13.38[Outcast]how does the amd chips handle asterisk?
18:13.48pgpkeysi started out as one of the original beta testers, before there was a 1.0 ;)
18:14.12*** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com)
18:14.29pgpkeys[Outcast]: well it would come down to the horsies you have. the cpu itself has no issues
18:14.52Gand_DJne1 here installed G729 on their * box. I just got a non-commercial license but can't find info for installing the ipp. :(
18:15.01Balu:)
18:15.02pgpkeysi have a duron 1.3GHz for a fbsd dev box with 1GB RAM. i've yet to see any app it can't handle or an app that can't handle the cpu
18:15.18[Outcast]pgpkeys: is there a proformance gain from using the amds at all?
18:15.29pgpkeysnot really except for games.
18:15.36[Outcast]k
18:15.43pgpkeyswhich is why i happened to have an AMD around in the first place :)
18:15.48Balupgpkeys: I've learned a lot from LFS even though I was a linux user for >7(?) years at that time :)
18:15.59pgpkeysmy game server box became my development box when i got tired of gaming
18:16.04WGFreewillGand_DJ: I have commercial g729
18:16.06WGFreewilldigium
18:16.08WGFreewillworks great
18:16.09PBXtechGand_DJ, non commerical?
18:16.36Gand_DJYeah, non commercial license from intel website
18:16.36PBXtechoh that hack, yea nevermind
18:16.48pgpkeysBalu: so did I. I started out with SLS Linux (Slackware's father) and followed along for several years. then LFS came along and even having worked in the industry for some time, I learned more working through the LFS than i had previously
18:17.00pgpkeysVERY good choice for those that really are into the technicals of distrib building
18:17.13Baluyep
18:17.15DrWho17Outcast: my Opteron boxes destroy my Xeon boxes for SQL database serving
18:17.23Sedorox:-p
18:17.32Balupgpkeys: I've created a Dreamcast-Linux with my LFS-knowledge
18:17.43Balupgpkeys: :)
18:17.49*** join/#asterisk Grooby (~Grooby@12.22.232.212)
18:18.03pgpkeyscool.
18:18.07Sedoroxthats alotta 2's
18:18.15Groobywierd
18:18.17pgpkeysi should email gerard and say hello
18:18.22Groobyhas anyone have problem with speex codec?
18:18.28pgpkeyshe and chris were great.
18:18.35Baluyep
18:18.49Balupgpkeys: I wonder if there still are the irc-servers
18:18.57pgpkeysprobably
18:19.38Baluyep irc.linuxfromscratch.org
18:19.41Baluno Gerard though
18:19.57pgpkeyshe never stayed connected
18:20.05Balutrue
18:20.07Balu:)
18:20.10pgpkeyshe's busy has hell. i wouldn't expect him to stay connected all the time
18:20.33Baludon't even know what he is doing now
18:20.39Baluhe switched and moved a lot
18:20.56pgpkeyswell his company gave him permission to write the book in the first place, gave him the down time to do it.
18:21.21pgpkeysso now that it's written and it's well into new renditions he's probably back working and offloaded the majority to individual maintainers
18:21.33BaluWasn't he going/working to do some LinuxLab work?
18:21.48Baluno
18:21.48pgpkeysi don't know for sure. i hven't touched a linux in oh neigh on 2 years now.
18:21.58pgpkeyseverything i've got is all *bsd these days
18:22.14pgpkeysso i don't know the goings on anymore like i used to with most of the linux projects
18:22.34Balu:)
18:22.40tzangerheh
18:22.42BaluNo BSD FS? :)
18:22.57pgpkeysfree, net, and open. only things i run
18:23.08tzangerlfs is pretty good for learning but if you're going to just copy and paste the text you ain't learning any more than watching hours of compiler output scroll by with gentoo
18:23.15tzangerI tried BSD
18:23.18tzangerI tried really hard to like it
18:23.20Balu:)
18:23.25pgpkeyswell occasionally i handle a solaris box or two on contract but not much beyond that
18:23.33pgpkeystzanger: exactly
18:23.40BaluThe learning in LFS is if you help people when they have problems
18:23.45tzangeryup
18:23.51tzangersame as asterisk, but really same as anything
18:23.53pgpkeysBalu: the learning is if you take the LFS apart
18:23.57pgpkeysto see what it's doing.
18:24.03tzangerI used LFS to build small firewalls
18:24.05pgpkeysthe passing ON of what you've learned is in the helping
18:24.30tzangerI had a firewall with ipsec, perl, iproute2 and an xmlrpc configuration server fit into a 16M CF
18:24.32ManxPowerGads!  _., "i", and "h" is generating almost as many messages as the GPL stuff!
18:24.34tzangeruncompressed was like 40
18:24.39pgpkeysthere's some learning in the helping too but nothing like when you take something apart
18:24.39tzangerManxPower: that is good though
18:24.48BaluI started using LFS because I got fed up with all distros
18:24.50tzangerit's like stuffing 10 pounds of shit in a 5 pound bag
18:24.52Balunot did what I liked
18:25.03tzangerBalu: I felt the same way about everything which is why I use slackware
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18:25.19BaluNow I am back to "grah - all that compiling, fixing errors, recompiling - I don't have that time!!1!"
18:25.31tzangerBalu: heh
18:25.40tzangerI am trying out suse 9.2 for a lug presentation next month
18:25.43pinotazanger: i'm going to try with the uClibc-based debian sooner or later
18:25.44tzangerit's an interesting experience
18:25.54tzangerpino: that's what I did with lfs... uclibc+busybox for most things
18:26.04pgpkeysthe only distro i touch these days is debian. I've even let my RHCE lapse out, but that's for political and internal differences with Red Hat's dealing with things. nothing they've done so far invalidates my RHCE knowledge so i can easily do what i need to do on the redhat boxes i controct on
18:26.08pgpkeyserr contract even
18:26.08BaluYep, nice for workplaces and my fathers box :)
18:26.21tzangerI hate hate hate HATE debian
18:26.36pgpkeysthen you hate one of the strongest if not THE strongest linux distribution going
18:26.49tzangerpgpkeys: debian is going nowhere fast, don't kid yourself
18:26.51BaluI am also a Debian dude, but they absolutely need to do something about their release cycles
18:26.53pgpkeysNO ONE does the level of testing and auditing on their distros that debian does
18:26.55tzangerredhat's got more pull than debian
18:26.58Groobythis is really wierd
18:26.59pgpkeystzanger: you can think that if you want
18:27.01tzangerand I hate redhat too :-)
18:27.03pinotzanger: no flame intended but... i love it :) so i'm curious to know why...
18:27.05Groobyi am getting 1 way audio with iax
18:27.10Groobyand just start happening recently
18:27.18Gand_DJis there a way to see what audio formats * supports for transcoding?
18:27.20pgpkeysanyways.. i'm not getting into a distro or old vs. new war
18:27.24Gand_DJor what's installed
18:27.25tzangerpino: no no people can use whatever distro they please, but I really really hate debian
18:27.26pgpkeysso.. how bout that asterisk!
18:27.49Baluasterisk?
18:27.55pinotzanger: exactly, i was asking why *you* hate debian! :)
18:28.02Baluah - the number of people in here let me think it is #debian :)
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18:28.21tzangerpino: I hate the politics behind debian, I hate how unless you stick to stable, it's unstable.  I hate how old stable is, but I understand why
18:28.32pgpkeystzanger: that's horse shit, but ok
18:29.02Balupino: I am using debian on all servers too, but I hate all the backports.org-stuff in my sources.list
18:29.03tzangerI hate how everyone says how stable it is yet they all run unstable and end up butchering the package tree, yet debian is still somehow immune to it
18:29.06ManxPowertzanger, It's pretty obvious that _. is confusing to new users and makes things happen that they don't want.
18:29.07pinotzanger: ok, release cycles mostly :)
18:29.09tzangerpgpkeys: refute it if you like :-)
18:29.18pgpkeysdebian's testing is far more stable than any other distro's stable release is. the versions are usually no more than 1 behind the current upstream version and that's to ensure that they have a known codebase to secure and audit from
18:29.25tzangerManxPower: _. is confusing, period
18:29.34tzangerpgpkeys: -stable is not 1 rev behind
18:29.39tzanger-testing is not -stable
18:29.44tzangerand -unstable is not -stable
18:29.58QwellTherefore, -unstable must be -testing
18:30.01tzangerhaha
18:30.04Balupgpkeys: no security support in testing and unstable
18:30.04ManxPowertzanger, You hate Debian?  I KNEW there was a reason I liked you. 8-)
18:30.09tzangerManxPower: :-)
18:30.11pgpkeysthat's stable, that's the official release. and the debian project's goal isn't to be latest and greatest. it's to build THE most stable distribution possible for administrators
18:30.18tzangerI mean yes, if everything you want is in -stable then it is *amazing*
18:30.24tzangerif you can put up with the politics
18:30.29pgpkeysBalu: unstable there is none
18:30.34pgpkeysbut testing most assuredly does
18:30.37tzangerpgpkeys: I am not arguing that -stable is not stable
18:30.49ManxPowertzanger, if "i" worked as people expect it would not be as much of an issue.
18:30.51tzangerpgpkeys: as I said, if everything you want is in the -stable tree, you have a very well tested and very stable distribution
18:30.57tzangerManxPower: I intend on making 'i' work as intended
18:31.04pgpkeysit's 1 day behind stable and that's because stable is an official release. they have to get it out for the official before they start on the unofficial
18:31.21ManxPowerI think "i" should be fixed to work as expected and then remove _. as a valid pattern.  Require at least 1 X or Z or N before the .
18:31.38tzangernah accept _. still but it does not match any of the oshiat extensions
18:31.56pgpkeystzanger: well considering that debian has almost 3 times the number of packages available than most of the other distros have available i'd say you'd have a VERY hard time not fiding something in stable
18:32.03tzangerpgpkeys: bullshit
18:32.07tzangerutter, complete bullshit
18:32.13pgpkeystzanger: that's NOT bullshit,. that
18:32.15pgpkeysis FACT
18:32.24Balupgpkeys: was it last year when all those ssh-problems came up?
18:32.27tzangerI have MANY utilities that do not appear in debina -stable or even -unstable
18:32.44pgpkeysBalu: that was ssh upstream which translated to ALL the distros, not to debian alone
18:32.49ManxPowerpgpkeys, Um, I want ease of management, reasonably up to date packages, package format supported by many vendors.  Mandrake does that for me.
18:32.57Gand_DJhow do you force linux to delete a directory that is not empty?
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18:33.03pgpkeysrm -rf
18:33.08DeeJayTwohi guys..
18:33.16EgonisGand_DJ: also, to see WHAT you are deleting... ls .*
18:33.23Balupgpkeys: I switched to a different ssh software (not sure what it's name was, need to look up), which had a bug too (found a few days later :-)
18:33.26DeeJayTwoHas anybody tried to use postgresql with odbc?
18:33.41pgpkeysManxPower: apt is the #1 package management system out there which is why redhat supported so heavily porting apt to handle RPMs
18:33.42Balupgpkeys: that was fixed in upstream, but not in the stable packages after two months
18:33.55pgpkeysas for ease of management debian/rules is as easy as hell
18:34.00Gand_DJthx
18:34.13Balupgpkeys: I contacted security-mailinglist and they told me there are problems backporting the stuff :-(
18:34.28Balupgpkeys: not sure if it was ever fixed...
18:34.49pgpkeysBalu: yes, backporting an upstream issue that's filtered into over 12 architectures (which is more support for various arches than any otehr distro has) is not exactly easy
18:35.02pgpkeyss/otehr/other/
18:35.16tzangerpgpkeys: psi (jabber client) is in stable, but it's quite a bit more than "one rev behind".  openswan (ipsec gateway).  unstable only.  postgresql.  hardly one rev behind.  MANY revs behind.  xen - unstable only.
18:35.20tzangerneed more examples?
18:35.33Balupgpkeys: of course, but it gave me a bad feeling all over...
18:35.37Baluanyway
18:35.39pgpkeystzanger: you did exactly what i expected you to do, you ignore the keyword in my comment
18:35.40ManxPowerpgpkeys, Cite your source.
18:35.44pgpkeysi said USUALLY
18:35.47tzangerpgpkeys: what was the keyword in your comment?
18:35.51tzangerusually what
18:35.55pgpkeysManxPower: I was ON the project. i AM onje of the sources
18:35.56pgpkeys:)
18:36.10pgpkeysi said USUALLY one revision behind
18:36.12tzangerHow can you tell a tough lesbian bar? ...Even the pool table doesn't have balls.
18:36.16tzangerhahahaha
18:36.16ManxPowerpgpkeys, I mean, cite an OBJECTIVE source.
18:36.19Gand_DJI noticed that for manitoba canada, all voip providers have access to 204-480 area only.. where would I look to getting ability to resell DID for that area also
18:36.23pgpkeysusually is not equal to subjective
18:36.35pgpkeysManxPower: Umm redhat support what is it, 6 architectures?
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18:36.49pgpkeysmandrake supports the same amount as does suse
18:36.53pgpkeyssame with slackware
18:37.10pgpkeyslook at the stable release support of number of architectures
18:37.18Balupgpkeys: and the debian people are thinking to reduce all their supported architectures
18:37.21Balu:)
18:37.23sivanalol
18:37.27*** join/#asterisk corlis (~corlis@HSI-KBW-082-212-051-230.hsi.kabelbw.de)
18:37.30pgpkeysif that doesn't constitute subjective then i don't know what to tell you
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18:37.47corlisheyas
18:37.52pgpkeysBalu: on the more difficult ports like sparc which even redhat had issues with porting to.
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18:38.03tzangerpgpkeys: the difference is that on slackware, building from source does not fuck up your package manager
18:38.07BaluOk, stop bashing distros now
18:38.08Balu:)
18:38.15durexhello folks
18:38.19pgpkeyswhich people are naming as the competitor (who i also worked for) so I'll speak on that score as well :)
18:38.21tzangerpgpkeys: that's the EXACT reason I don't fuck around with these "dep tracking" distros... too much fucking work
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18:38.48durexit´s my first time I instaled Asterisk. I have two computers with X-lite installed in the same network thank asterisk, and I wanna make them talk with each other
18:38.51durexdoes anybody can help me?
18:39.15jf_is it possible to remove the girl saying in the voicemail please leave your message after, hangup or press the # key, by a simple beep
18:39.16pgpkeystzanger: hehe. well trying to mix source and pkgs is always hell on wheels
18:39.21corlisdurex: you been googling around and read that manymany examples?
18:39.28tzangermy main main main GNU/beef with GNU/debian is GNU/the GNU/politics
18:39.30BuckRogersdurex you need to set up a sip config and extenctions
18:39.33Balugtg dudes - I will come back if I totally ruined my ACFS (Asterisk Configuration From Scratch) :)
18:39.40DrWho17slackware has a package manager?
18:39.42tzangerpgpkeys: exactly.  with slackware it's trivial
18:39.46tzangerDrWho17: yup
18:39.52durexcorlis yes... and I got the following error in Xlite: Call failed: 408 Timeout
18:39.54MeTaBSDits ok my problem solve :)
18:40.01MeTaBSDbut i have other question :)
18:40.01pgpkeystzanger: it's only trivial because slackware by default doesn't exactly have package management
18:40.02MeTaBSDAsterisk RealTime Voicemail
18:40.12Balulater
18:40.13DrWho17hah, I ditched slackware 6-7 years ago, because they were so far behind
18:40.14MeTaBSDwhere i specify the Privilege user
18:40.16corlisdurex: have you checked that the firewall is open/ok?
18:40.28durexcorlis yes... it's ok
18:40.30tzangerpgpkeys: untrue.  slackware has great simple package management.  Slackware does not have DEPENDENCY TRACKING which is the problem with every other package manasgement system out there... they try too damned hard
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18:41.06DrWho17moved to that new Redhat thing, with glibc2 and rpm
18:41.10pgpkeystzanger: ok, so it's cool when slackawre let's you overwrite libperl.so by some perl related package without checking to see if it's already installed
18:41.11pgpkeyscool
18:41.27durexcorlis sorry... I think really could be a firewall problem... let me see...
18:41.28tzangerpgpkeys: dep tracking is 90% solved with ldd and a search of the manifest.  debian (and the others) try for 100% and they all fail because if something's not in the package tree and you need it, you're fucked... now you have the package manager the disto has and you have the one you've now set up in your head.  thanks but no thanks
18:41.39Gand_DJIs there a way to change a peer / registry entry from being unmonitored to monitored?
18:41.47tzangerpgpkeys: cite me a real world example, not some contrived bullshit one
18:42.04jf_is it possible to remove the girl saying in the voicemail please leave your message after, hangup or press the # key, by a simple beep
18:42.07DrWho17tzanger: mysqlclient, glibc version
18:42.10pgpkeysi AM citing real world. you just choose not to let the possibility enter into YOUR world
18:42.12pgpkeys:)
18:42.33tzangerpgpkeys: hardly.  I do a LOT of work with perl and never have I had libperl.so overwritten.  Your example is contrived
18:42.39tzangerDrWho17: huh?
18:42.44DrWho17yea perl, eek
18:42.47pgpkeysit's an example of a VERY real world possibility
18:42.52tzangerpgpkeys: no it's not
18:42.57pgpkeysok, if you say so.
18:42.59tzangerpgpkeys: name a perl module that creates libperl.so
18:43.01tzangergo ahead
18:43.06pgpkeysi didn't say creates it
18:43.28pgpkeysnow you're dodging the issue in an attempt to prove your disaffection
18:43.30pgpkeysi won't play that game
18:43.30tzangerpgpkeys: you actually give me great ammunition -- perl has its own package management: CPAN.  If Debian doesn't have a package for a specific perl module you need, now you're dealing with CPAN too
18:43.36DrWho17well, don't mix and match that is for sure
18:43.42Groobyhmmmm
18:43.54DrWho17don't compile and install via the distro package system
18:43.54tzangerpgpkeys: I'm not dodging anything, I am calling bullshit.  I have been working with slackware and perl for 7+ years...  I have NEVER had a perl module overwrite libperl.so.
18:43.55pgpkeys*sigh*
18:44.02MeTaBSD--> Asterisk RealTime Voicemail i configure all but where i specify the username and password for databases and table privilege
18:44.11tzangerpgpkeys: that's like me saying I can create a deb package in -unstable that does the same...  your point is meaningless
18:44.12Groobywhat could cause the suddent change from working setup to only 1 way sound?
18:44.20pgpkeystzanger: and I'ev been working on various distros for neigh on 10 years and this is a VERY real world situation.
18:44.20DrWho17MeTaBSD: in extconfig
18:44.25pgpkeysso think as you wish.
18:44.41tzangerpgpkeys: again... where in the blue fuck do you get a perl module that overwrites libperl.so?  Any distro
18:44.43pgpkeysi suggest we stop bashing at each other now because we are boviously in the opposition camp for each other's thoughts
18:44.45DrWho17extconfig.conf
18:44.53corlisAnyone here could help me with this, when i enable CAPI debug: "found capi with omsn = 123456", next is: "Segmentation fault"
18:44.59tzangerpgpkeys: I just want a real world example, not a "what if" that can happen on any distro
18:45.01pgpkeysi used the libperl.so as an example of a potential
18:45.06pgpkeyspeople KNOW what libperl.so is
18:45.36tzangerpgpkeys: but the same thing cna happen in debian
18:45.37DrWho17I had problems with mysqlclient and dbi yesterday
18:45.41pgpkeystzanger: you CAN'T overwrite libperl.so on debian via any known package unless you --force
18:45.48MeTaBSDDrWho17 voicemail => mysql(methode),asterisk(DB),voicemail_users(Table) but where is the username and password ?
18:46.03DrWho17updating mysql from source, broke the rpm installed perl DBD modules
18:46.10tzangerpgpkeys: and you can't do that on slackware without a bad package either...  I fail to see what makes Debian better in this particular example?
18:47.02jeffikall: using *@home, need to add a line to extensions.conf to allow access to long on to voice mail by pressing * during greeting
18:47.02DrWho17MeTaBSD: well, that's not how I access the voicemail users database from within asterisk
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18:47.19pgpkeystzanger: the last slackware i used (7 iirc) did not do adequate checks of versioning to ensure that proper care was taken to stop the overwrite. it was automatic assumption that it was ok TO overwrite from another package.
18:47.26pgpkeysand THAT is intolerable
18:47.31DrWho17I just store the mailbox information in a table, and access it just like a normal extensions
18:47.47tzangerpgpkeys: yes we are on opposing sides...  as I said, if -stable has exactly everything you want, it is *GREAT*.  It really and truly is... but at least for my installations, it doesn't...  and -testing and -unstable are no better than no package manager in my opinion because you either end up breaking the dep tree by stepping around it to get what you want or end up keeping two dep trackers around
18:48.12jeffikDrWho17: need to call in from outside
18:48.18pgpkeyswhy not make packages and submit if the ones you want are missing?
18:48.19tzangerpgpkeys: that's correct; slackware 10.1 does that too.  you say "Installpkg somepkg.tgz" and it will, and anything in it will overwrite anything that was on the system before
18:48.30tzangerpgpkeys: that has a VERY simple answer.
18:48.49pgpkeysyou don't have to maintain it, just submit it and ask the project to assign a maintainer
18:49.01pgpkeysif it has intrinsic value they'll do it
18:49.04DrWho17jeffik: add it to your context to allow the *
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18:49.29DrWho17exten => a,1,VoiceMailMain(${CALLEDTO}) ; If they press *, send the user into Voic$
18:49.29DrWho17exten => a,2,Hangup
18:49.44tzangerpgpkeys: 1) I must create a patch to GNU/ify the entire documentation tree.  2) I must include in that patch moving the document tree to where Debian prefers it.  3) I must write proper startup/shutdown scripts and 4) I must now MAINTAIN that abomination since a) the software author will never accept the debian-ified setup as "stock" and b) nobody else will take it over because it's such a holy shit pile of work to keep up :-)
18:49.44*** join/#asterisk durex (~ironman@weber.anpa.org.br)
18:49.58DrWho17just tack those two lines to the end of your voicemail context
18:50.07MeTaBSDDrWho17 i dont need privilege ?
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18:50.13jeffikok i'll try it
18:50.17tzangerpgpkeys: I had no idea that someone else would accept a paackage maintenence job from lil ole me :-)
18:50.21barmalif Voicemail was left then do something. Is there something that tells you the voicemail has been left?
18:50.28DrWho17MeTaBSD: you specify the priviledges in extconfig.conf
18:50.31corlisWhat should a normal extension look like for a call from a IAX-client to an isdn?
18:50.44pgpkeystzanger: i think we'd both agree that the only 100% way to manage something is to remove package management completely, move the management back to the admin and let him face the tedious task of version control and overwrites. packages were simply meant to lower the level of time-consumption the admin had to face.
18:50.44tzangermaintaining slackware packages is dead simple in comparison
18:51.17MeTaBSDDrWho17 how
18:51.17DrWho17pgpkeys: statically link everything !!!
18:51.35pgpkeystzanger: again, as i said create the package and submit. if it has value (as in your not the only one, or only a small subset of the userbase would actually USE teh damn thing) they will usually assign a maintainer
18:51.45DrWho17MeTaBSD: have you read the realtime instructions? They are very clear
18:51.56pgpkeysDrWho17: hehe ok, you just shot your memory usage up by probably 70%
18:51.57tzangerpgpkeys: correct -- slackware packages are easy to maintain and work with but you have to have a bit of clue to be able to use them effectively.  What every other distro (lfs and gentoo aside) do is lower the level of clue required...  it's a noble and laudable goal but it ties my hands as an admin
18:51.59barmalis there anything what returns anything if voicemail has been left?
18:51.59MeTaBSDCopy asterisk-addons/configs/res_mysql.conf.sample to /etc/asterisk/res_mysql.conf
18:52.00DrWho17extconfig.conf.sample is probably in your default asterisk install as well
18:52.04MeTaBSDi think i find
18:52.15pgpkeys(arbitrary number to show the extreme increase in memory you just caused)
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18:52.28nexIAX<PROTECTED>
18:52.29tzangerpgpkeys: as I siad I had no idea you could do that, that reduces my number of arguments against debian's package system a little.  :-)
18:52.37pgpkeyshehe
18:53.03DrWho17MeTaBSD: well, if you are using the mysql you specify database user/pass/host in res_config_mysql.conf
18:53.11pgpkeysthere are tons of maintainers in the queue looking for packages. if you did NOT find at least one looking for something I'd be truly suprised
18:53.22DrWho17but you specify how to get to your database in extconfig.conf
18:53.25jeffikDrWho17: I will try it thank yuo
18:53.31FLeiXiuSDoes asterisk come with a SIP server?
18:53.33tzangerManxPower: quit fucking with bkw, that's hilarious :-)
18:53.57*** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net)
18:53.57*** join/#asterisk TEKjacob (~chris@70-32-21-41.frdrmd.adelphia.net)
18:54.08TEKjacobHappy Friday!
18:54.08barmalgotoif (Voicemail has been left = true?3:2) how can I use it? does anybody know?
18:54.36MeTaBSDok
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18:54.42nexIAX<PROTECTED>
18:54.43pgpkeystzanger: anyways, it *has* been clueful and enjoyable speaking with you. need to make some coffee and step into an area *I* have no experience in, asterisk setup :)
18:54.46tzangerbarmal: you can't
18:54.52MeTaBSDi dont have the good version ... i download the asterisk-addons 1.07
18:54.55corlisTEKjacob: Happy Friday? with asterisk segfaulting here? *sniff* no
18:54.56TEKjacobOh wise ones... Do I need to use a T1 Cross over or straight cable to go from the dmarc to the Digium T1 card?
18:55.13DrWho17straight through all wires wired
18:55.15tzangerpgpkeys: :-)  I enjoy flaming as much as anyone but this wasn't a flame (at least I hope you didn't think it was)
18:55.43pgpkeysnaww, i've met far rougher. you have clue, whichis always good. i enjoy clue so long as it's clue and not clue+ego.
18:55.46TEKjacobgroovy thanks folks
18:56.03tzangerpgpkeys: yup... I have no ego when it comes to this stuff, it's just what I've found to work (or not work) over the years
18:56.07pgpkeysi can always debate with clue. his brother ego has to go though. which (i think) neither of us brought to the party
18:56.12tzangerI didn't write the slackware package manager so none of my ego's in it
18:56.20pgpkeyshehe
18:56.22tzangernope it was a good (heated) debate
18:56.39pgpkeyslittle bit of heat always makes the meat taste better :)
18:56.44barmaltzanger: thre is no way to do the function when the voicemail has been left and user hangs up * calls back to given number?
18:56.50*** part/#asterisk jmhunter (~jacob@wire3-215.razzolink.com)
18:56.59tzangerbarmal: of course, but it doesn't work with just asterisk
18:57.03tzangeryou need a cron job and a shell script
18:57.13DrWho17well
18:57.13tzangerbecause the voicemail will only go on properly if they hit # and 1 to accept the message
18:57.16tzangerwhich is bullshit
18:57.17pgpkeysok, coffee time. wake up fully enough to start laying down this config
18:57.17tzangerbeen there, done that
18:57.29tzangerit's far easier with a shell script to watch for new messages and copy a .call file over
18:58.36barmaldamm the problem is I am not good with shell script. Do you know if thre is any example would appreciate man
18:58.47tzangerbarmal: yeah I have mine
18:58.50tzangerit ain't pretty bu tit works
18:59.18barmalcan you share please...
18:59.29tzangeryueah just looking it up
18:59.31barmallitcomp@bellsouth.net
18:59.54*** join/#asterisk In-Side (~Lowgitek@es-217-129-30-41.netvisao.pt)
18:59.55In-Sidehi
18:59.59barmalhi
19:00.01cypromislo
19:00.37In-Sideanybody knows what that means? pr 15 20:00:05 WARNING[1527]: chan_sip.c:603 __sip_xmit: sip_xmit of 0x812b21c (len 414) to operator_ip returned -1: Bad file descriptor
19:04.01In-Sideanybody ?
19:04.18In-SideI have no clue what hell is that
19:04.20*** join/#asterisk mbranca_home (~matteo@host-84-222-7-10.cust-adsl.tiscali.it)
19:04.56tzangerbarmal: http://pastebin.ca/9612
19:05.00*** part/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
19:05.09tzangerthat's my messy callback script; it's called every 5 minutes from crontab
19:05.29*** join/#asterisk redG (~nik@67.107.241.3.ptr.us.xo.net)
19:05.49*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
19:05.56In-Sidetzanger: can you help me ?
19:06.57*** join/#asterisk redG ([U2FsdGVkX@67.51.185.15)
19:06.59redG<PROTECTED>
19:07.17tzangerbad file descriptor?  weird
19:08.02ManxPowerWOW!  Useful suggestions to _. and h
19:08.59tzangerManxPower: which one
19:09.11ManxPowerhttp://bugs.digium.com/bug_view_page.php?bug_id=0004036
19:09.24tzangeryeah
19:09.32In-Sidelet take a look
19:09.35In-Sidedamn message ...
19:09.35ManxPowertzanger, info about proper use of _. in extensions.conf.sample and add a URL to the warning message to the proper use of _.
19:09.39In-Sidevery unusefull
19:09.40tzangerManxPower: uh 4036 is about parking
19:10.22ManxPowertzanger, I pasted that before I saw your "whoich one"
19:10.27In-Sidethe url is for me ?
19:10.27tzangerManxPower: 4038 is the one
19:10.37ManxPowermy paste for 4036 is just desperation to get it fixed.
19:10.42ManxPowerI'll post a bounty if
19:10.44ManxPoweri have to.
19:10.59ManxPowerWe go live with an office that will use a lot of parking on Friday
19:11.09tzangerCorydon-w: wow that was fast
19:11.17tzangeris it easy to do that for SIP and IAX too?
19:12.18In-SideIU just got franzy it that
19:12.35*** join/#asterisk roamer323 (~sing@HSE-MTL-ppp64197.qc.sympatico.ca)
19:12.40In-Sidemy * don't get registered in my sip provider
19:13.13ManxPowerEGADS!  I have a negative karma!
19:13.24In-Sidehow can i set a realm on my resister
19:13.47In-SideI trying with realm=name.realm in providerconfiguration
19:14.19In-Sideand it still persist to use the host name after @ as realm in register in provider how can I set another one ?
19:15.23*** join/#asterisk jmacz (~jmacz@63.245.86.225)
19:16.12*** join/#asterisk darby_t (~tom@182-tor-6.acn.waw.pl)
19:16.24corlisany isdn-god here, that could help me with my lill problem calling out?
19:16.38*** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca)
19:17.15Corydon-wtzanger: dunno, haven't looked
19:17.34corlisor any c-guru/asterisk-guru here, that could help me to find out why my asterisk SEGFAULTS?
19:17.38*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
19:17.43Corydon-wThat's the advantage of having poked around in the code for a long time...
19:17.44WGFreewillcan anybody help with H.323 codec negotiation problem
19:17.47corliswoops, i hate caps :P
19:17.52WGFreewillNortel is beating me about asterisk
19:17.59*** join/#asterisk Hymie (hymie@L8R.NET)
19:18.02tzangerCorydon-w: I'm looking at it and I don't know if that's right
19:18.08tzangerwouldn't you use ast_canmatch_exten() ?
19:18.11Hymiehey guys, I seem to be getting a lot of problems with hard drive usage and clipping / noise in asterisk.
19:18.16HymieI don't think it's purely interrupts, but I am monitoring each call... so if I have 8 or 9 calls occuring at once, there's some thrashing.. add a cp operation, and it seems that asterisk freaks out at the inability to dump the buffer fast enough
19:18.20Hymieanyone else notice behaviour like this?
19:18.39In-Sidehey guys how can i set a realm in register ?
19:18.42Nivex<PROTECTED>
19:18.45Corydon-wtzanger: I'm looking for a single extension called "i".  Why would I use a different function?
19:18.52Corydon-wtzanger: btw, have you tried it?
19:18.59tzangerCorydon-w: no I haven't tried it yet, it seems too simple
19:19.01In-Sidemy sip provider doesn't accept the host name as realm
19:19.11tzangerCorydon-w: you are looking for an 'i' exten but what will match it later on?
19:19.16In-Sidehow in register i can set a new realm ?
19:19.24WGFreewillhttp://bugs.digium.com/bug_view_page.php?bug_id=0003980
19:19.33FLeiXiuSDoes Asterisk already include SIP?
19:19.34Corydon-wtzanger: once you start the dialplan, the dialplan takes care of the invalid extension
19:19.41tzangerCorydon-w: ahhhhhhhhhhh
19:19.42*** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE)
19:19.43corlisFLeiXiuS: yes
19:19.57In-SideFLeiXiuS: see sip.conf
19:20.02DaLion!seen bkw_
19:20.02FLeiXiuScorlis: It includes the SIP server also correct?
19:20.09In-SideFLeiXiuS: shure
19:20.11Corydon-wtzanger: that's the nice thing about how pbx.c's dialplan logic works
19:20.22corlisFLeiXiuS: yes, yes and yes
19:20.33FLeiXiuSlol cool :-p
19:20.42Corydon-wBecause, in the words of that Ronco commercial, we can "set it and forget it"
19:20.56In-Sidedoesanybody knows how to set the realm at register in sip.conf?
19:21.32eKo1realm=myrealm under [general] in sip.conf
19:21.35tzangerCorydon-w: ok, I did something simialr for chan_iax2.c
19:21.41eKo1I think
19:21.42tzangerexcept I return CANEXIST
19:21.48tzangertesting now
19:21.49In-SideeKo1: no that is only for clinets
19:21.50*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
19:21.55FLeiXiuSI'm looking to just use Asterisk locally, without connecting to a VoIP provider.  Would I need SER to do the routing?  Or does Asterisk already deal with this.
19:22.01In-Sideit doesn't uset it at register
19:22.14Corydon-wtzanger: I would test it myself, but I'm in the middle of a drastic change to manager.c
19:22.15In-SideFLeiXiuS: asterisk is all in one solution
19:22.20In-SideFLeiXiuS: is enough for you
19:22.32tzangerCorydon-w: :-)  I will test it tonight
19:23.22tzangerbah got 3 active calls
19:23.31ariel_FLeiXiuS, in Between 95% to 99% of the time you will never need to use SER with asterisk.
19:23.48tzangerer two now
19:24.00ariel_Good afternoon everyone
19:24.43FLeiXiuSariel_ and In-Side: Thank you.
19:24.47corlisWhy does asterisk crash, when using: "exten => _0049XXXXXXXXXX,1,Dial(CAPI/12345678:${EXTEN},90)"
19:25.30corlisWhen i try to call out with my IAX-client, asterisk does a segmentation fault
19:26.07ariel_corlis, seems like you have an issue with your drivers. Are you using CVS Head or stable?
19:26.20corliscvs
19:26.46ariel_corlis, make sure that your capi is the right one for cvs head and not stable.
19:27.09corlisusing chan_capi 3.5, as it's the only one out there?
19:27.52corlismodified it a little, tho, as it won't compile with cvs
19:28.20*** join/#asterisk drbrown (~chatzilla@65.121.240.182)
19:28.28pinocorlis: there's a patch set for running chan_capi on cvs-head, AFAIK
19:28.39fooboscorlis, depending on card, you can also run isdn with chan_misdn http://www.beronet.com/?PageID=3017
19:29.17corlispino: i applied that patch, but it still was referring to a non-existant channel_pvt.h
19:29.25DeeJayTwois there any stable asterisk version with "realtime" ?
19:29.44corlisfoobos: hrm. gonna check that one out, thanks
19:29.50pinocorlis: the one I have does not... yours is probably old
19:31.01pinohttp://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 does no longer use channel_pvt.h.
19:31.03ariel_DeeJayTwo, no
19:31.11Hymiehey guys, I seem to be getting a lot of problems with hard drive usage and clipping / noise in asterisk.   I don't think it's purely interrupts, but I am monitoring each call... so if I have 8 or 9 calls occuring at once, there's some thrashing.. add a cp operation, and it seems that asterisk freaks out at the inability to dump the buffer fast enough
19:31.17durexdoes anybody can help me with asterisk and x-lite ???
19:31.21*** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net)
19:31.38LoRezHymie: using scsi?
19:31.43HymieLoRez: ide
19:31.57corlispino: Hrm. then there went something wrong... gonna recheck that....
19:32.19ariel_durex ask a question lots of people here use both.
19:32.55pinoHymie: if I ran * on Linux with IDE disks, I'd try hdparm -u1 -3c -d1 on them.
19:33.06barmaltzanger: usage is exten => voicemail_callback.sh u101   ???
19:33.07pinos/3c/c3/g
19:33.36*** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
19:33.42tzangerjust 101
19:33.53barmalok
19:34.42DrWho17DeeJayTwo: no
19:36.40corlispino: weird, here it's still referring to channel_pvt.h....
19:38.22_Briananyone here ever play with AVST?
19:38.28FLeiXiuSCould I just use any ordinary phone adacpter from RJ11->RJ45(VoIP)?  I was looking at the vonage one, but it says subscirption required.
19:38.33pinocorlis: the applications do, but the channel does not
19:38.49DeeJayTwoDrWho17: So there's still no asterisk version able to be feeded dynamically from a SQL database?
19:39.04DrWho17not the stable branch
19:39.14DrWho17CVS has had it for quite some time
19:39.14DeeJayTwook...
19:39.37DeeJayTwocan we expect some "unstable" version to be technically stable? ;)
19:39.40corlispino: hrm? but then "make install" will fail....
19:40.01DrWho17DeeJayTwo: well, stable is a new thing to asterisk, the CVS is pretty stable
19:40.19pinoyou just need chan_capi.so to compile OK with the patch and without further editing. then you have a starting point to investigate the crash!
19:40.38DeeJayTwoHEAD ?
19:40.43DrWho17yes
19:41.05DeeJayTwoyou'd build an mid-telco on it?
19:41.06DrWho17I've always run out of CVS
19:41.08DeeJayTwo(1000 users)
19:41.14*** join/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com)
19:41.30DrWho17DeeJayTwo: well, like I said, it's only a new thing, that asterisk has had a "stable" branch
19:41.45DeeJayTwook
19:42.07DrWho17I've got a device without a reload in 6 months
19:42.14DrWho17running off CVS
19:42.35DeeJayTwowe built a management interface for "realtime"...
19:42.37DrWho17and that was because I upgraded it
19:42.42DaLionman i ahad a sun box i didnt reboot for 790 DAYS
19:42.47DeeJayTwowe're gonna get our server monday...and we're gonna start testing in 1 week..
19:42.59DrWho17Daliion: well, I'm talking about an asterisk reload
19:43.03corlispino: ok, and where do i have to put chan_capi.so?
19:43.07DrWho17of course the machines have been up fine
19:43.09DaLionit could take load avg of 1500 before goign to a crawl.. in the good old 1995's days.. where porn sitres where huge ;)
19:43.16DeeJayTwowith Adtran TA750 channel banks..
19:43.18DeeJayTwofor 250 users
19:43.20DaLionah
19:43.21DaLionheheh
19:43.23pinousually, in /usr/lib/asterisk/modules/
19:43.26DaLionthat nice then
19:43.36DrWho17DeeJayTwo: oh, we have some of those
19:43.41DaLioni think we reload asterisk each night at 5 am when 0 calls
19:43.47DaLionmade a small script for that
19:43.52DrWho17and some carrier access bits lying around
19:43.58DaLionsits andwaits til 0 calls then does whatevr
19:44.07eKo1I restart * every day at 1:00 AM.
19:44.13DaLion1am is busy here
19:44.14DrWho17really, hrm
19:44.19DaLionyeah
19:44.24*** join/#asterisk ChristianK (~Christian@p54A3FD7C.dip.t-dialin.net)
19:44.26PBXtechyou do that in a cron job?
19:44.37DrWho17why do you do that?
19:44.37DaLionno
19:44.45eKo1It will restart regardless of how busy it is here.
19:44.52*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)
19:44.53corlispino: still the same: executes the dial, then segfaults
19:45.16DaLioni do a while look in bash .. test for lock file (means it restarted) then asterisk -rx 'show channels'  | grep -a 'active'| gawk -F' ' '$1==0 {print "./reloadast.sh" }' |/bin/bash
19:45.17*** part/#asterisk Grooby (~Grooby@12.22.232.212)
19:45.24eKo1My problem is, by the end of the day, * is full of zombie channels and the only way to get rid of them is using a restart.
19:45.27DaLionthat reload is just what ever i want to do
19:45.29corlisgonna check out that misdn thing
19:45.59*** join/#asterisk ves1820 (~root@o-254-108.hosts.cablelink.de)
19:46.01DrWho17eKo1: really, what type of channels?
19:46.20DaLioneko u want it ?
19:46.26*** part/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com)
19:46.30eKo1SIP, which are related to my FXO gateways.
19:46.37eKo1Stupid analog crap.
19:46.40DaLionu can test with showing print" I WOULD RELOAD" instead ofdoing anyshit
19:46.51eKo1Otherwise, I would never reload or restart.
19:46.55DrWho17eKo1: so they are losing connection or something?
19:47.19eKo1Well, no but those zombie channels eat memory.
19:47.20DrWho17I've got my asterisk boxes sipping over OC3/OC12/ethernet only
19:47.24DrWho17stable ethernet links
19:47.30eKo1Lucky you.
19:47.42DrWho17and TDM inside, never noticed an issue
19:47.57DrWho17TDM -> SIP
19:48.05eKo1Are there any good FXO gateways out there?
19:48.14DrWho17pretty high volume per machine, I never reload unless making a config change
19:48.33*** join/#asterisk mbranca_home (~matteo@host-84-222-11-21.cust-adsl.tiscali.it)
19:49.05DrWho17If I had to reload/restart services like that, I'd look for something else
19:49.15eKo1Well, you should reload ONLY when you make config. changes.
19:49.15DaLion???
19:49.35eKo1Well, I don't have other choices.
19:49.43ves1820hi, has somewone got MWI working with CCM 4.2 ?
19:49.43DrWho17ok
19:49.47PBXtechbuy a HP server
19:49.58ves1820yo
19:50.00DaLionhey .. yeah .. but a config change could be a simple as changing folow me .. or new password for vm
19:50.01eKo1I'm at the mercy of these shitty analog gateways.
19:50.01ChristianKsrt: how are you? :D
19:50.54DrWho17Damin: yea, got most of those things in a MySQL table though
19:51.03DrWho17DaLion rather
19:51.10DaLionyeah
19:51.19DaLionso whats the prob again ?
19:51.32PBXtechi want to play with that ds3 card
19:51.45tzangerCorydon-w: unfortunately the fix isn't that simple in chan_iax2 :-(
19:51.51*** join/#asterisk netofsickcoder (~netofsick@cpe-24-170-74-115.stx.res.rr.com)
19:53.22eKo1I don't get this. I ping my gateway from one machine and I get <10 ms ping times. I ping it from another machine and the ping times >100 ms.
19:53.54*** join/#asterisk don_oles (~bill@pajaro.alfabank.kiev.ua)
19:54.28tzangerCorydon-w: oh wait, I think I found it
19:54.35don_oleshello coolhackers ;-)
19:55.36don_olessomeone can tell me why this stuff uses 97% of CPU on my 2 Ghz FreeBSD box?
19:55.49don_oles.. when doing nothing????
19:56.00*** join/#asterisk bobessutio (~bobessuti@c-67-180-96-152.hsd1.ca.comcast.net)
19:56.20Corydon-wdon_oles: because the FreeBSD port is a hack?
19:56.38bobessutiocan you have one did spread across multiple origination providers?
19:57.57barmaltzanger: thanks something works only it sends a sms message but i'll firure out the rest, Thanks
19:59.35sivanajbot: no, sivana is one of the brightest stars out there, ok? :)
19:59.36jbotokay, sivana
20:01.33corlishrm. i already hate my idea of using asterisk at my office, so i can receive calls wherever i have my laptop....
20:02.36corlisfirst day that it works, and already 3 calls... although it's 10pm here
20:02.59*** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca)
20:03.42jf_is there any to make * dial faster on a zap channel, for instance if i use my sip phone and want to call over a zap channel it take like 4 seconds before start rigging
20:05.14CoaxDjf: It should start ringing immediately on the SIP phone and dial silently and bridge the SIP and the ZAP channel together
20:05.25*** part/#asterisk oden (~oden@194-237-146-22.customer.telia.com)
20:06.22tzangerwhat the blue fuck
20:06.59tzangerCorydon-w: I've replaced every instance of ast_exists_extension() with one that checks for the original AND "i" and it still doesn't work
20:07.35Corydon-wtzanger: I haven't looked, so I don't know what to tell you
20:07.36Qwelltzanger: Whats the problem with i?  I tend to ignore the list
20:07.39jf_CoaxD:  right now i dont think it does
20:08.26jf_can i configure something
20:08.39kairoReading, ok the asterisk is one gatekeeper, but it is a sip gk? I use the gnugk, one gk h.323 and I need sip on the momment.
20:08.40CoaxDjf: Hmm. no, it just does what it does
20:08.49CoaxDjf: But if yours is doing something different than mine, there's obviously a difference
20:08.55tzangerQwell: read the list
20:09.10Qwelltzanger: Too much drama
20:09.21tzanger<PROTECTED>
20:09.24tzangerooh
20:09.24tzangergetting closer
20:09.43*** part/#asterisk bobessutio (~bobessuti@c-67-180-96-152.hsd1.ca.comcast.net)
20:10.07*** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net)
20:10.44TUplinkhow do i set an umtamate timeout?
20:11.07jf_CoaxD: does i ring right now how it take like 4 sec before
20:11.53tzanger~google IAX2 RFC
20:11.54bugbotgoogle IAX2 RFC is assigned nothing and reported nothing.
20:12.02tzangerno no no
20:12.06tzanger~google IAX2 reference document
20:12.07bugbotgoogle IAX2 reference document is assigned nothing and reported nothing.
20:12.14tzangeroh for fuck sakes
20:12.53*** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net)
20:13.39tzangerthere we go
20:13.39*** join/#asterisk durex (~ironman@weber.anpa.org.br)
20:13.42tzanger~google iax2 white paper
20:13.48tzangerha
20:13.50bugbotgoogle iax2 white paper is assigned nothing and reported nothing.
20:15.36Hymiehey guys, I seem to be getting a lot of problems with hard drive usage and clipping / noise in asterisk.   I don't think it's purely interrupts, but I am monitoring each call... so if I have 8 or 9 calls occuring at once, there's some thrashing.. add a cp operation, and it seems that asterisk freaks out at the inability to dump the buffer fast enough
20:16.10*** join/#asterisk Veryhot (~tho@adsl-69-109-159-239.dsl.sndg02.pacbell.net)
20:16.33Veryhothi all, anyone using netlogic.net?
20:17.38Veryhotdo they have good network for netlogic.net?
20:17.44tzangerCorydon-w: are you sure that check for 'i' will work?  I see * falling back to 's' if the exten doesn't exist, not 'i'
20:18.01jf_is there any to make * dial faster on a zap channel, for instance if i use my sip phone and want to call over a zap channel it take like 4 seconds before start rigging
20:18.16Corydon-wtzanger: but it DOESN'T fall back to s
20:18.46tzangerhmm
20:18.46Corydon-ws is only for if you don't have ANY extension when you start
20:18.46tzangerchan_iax2 is working very differently from zap
20:18.50tzangerCorydon-w: yeah, chan_iax2 is wrong
20:19.00tzangerit says "extension 5342432 doesn't exist, falling back to 's'"
20:19.18Corydon-wInteresting
20:21.46CoaxDThe taxes are DONE, MAN
20:22.09Corydon-wCoaxD: You don't have your refund, already?
20:22.15CoaxDCorydon: I had to pay in $3k
20:22.20Corydon-wOuch
20:22.25CoaxDCorydon: You think I'm gonna give 'em money BEFORE 4/15??
20:22.36Corydon-wI'm thinking extension...
20:22.45tzangerCorydon-w: pbx.c around line 2250
20:22.48CoaxDCorydon: Heh :) You can't get an extension for money
20:22.54CoaxDCorydon: You can get an extension for the RETURN
20:22.56tzangerI think it should check ast_strlen(c->exten) for nonzero
20:23.00tzangerand only drop to s if it is zero
20:23.08tzangeroterwise jump to 'i'
20:23.14Corydon-wYou mean ast_strlen_zero
20:23.47tzangeryes
20:23.48tzanger:-)
20:24.02Corydon-wGo for it...
20:24.17*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
20:24.30shmaltzwho overhere is an admin to the asterisk-users list?
20:24.30PBXtechim getting quite a few dropped IAX <->IAX calls over a lan  (no internet calls) any ideas off the top of your head?
20:24.33tzangerwhat the hell
20:24.34tzanger<PROTECTED>
20:24.48tzangercopy "s" to c->exten for a size of the orignal exten length?
20:24.55tzangerthat seems wrong
20:24.57*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
20:25.48cp5it says don't copy more than what can fit in c->exten
20:26.26*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
20:26.51*** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net)
20:28.14Bentleyhi all - can anyone recommend a utility for measuring packet loss on a network?
20:28.57rikstagood question
20:29.25TUplinkpacket loss.... packets dont get lost inles their is a colision
20:29.34TUplinkrite
20:29.50rikstanot just that
20:29.51PatrickDKtuplink, you wish
20:29.52TUplinkifconfig i belive will tell you Collisions
20:29.56TUplinkok...
20:30.03TUplinkhowels dose it die?
20:30.10rikstahowels? lol
20:30.14rikstahow else?
20:30.16PatrickDKmost routers drop packets
20:30.23PatrickDKif a link is over 80% full
20:30.31rikstaTUplink: what about when you receive more data than you can process?
20:30.36PatrickDKalso, interference
20:30.44PatrickDKbuffer overflows :)
20:30.56TUplinkyea... true
20:30.59PatrickDKriksta, that doesn't happen too much now a days though
20:31.16rikstatrue
20:31.25PatrickDKI have some 3c501 cards, that can't do back toback packet reception :)
20:31.37PatrickDKon 10base-2
20:31.38Bentleyi was about to try out iperf: http://dast.nlanr.net/Projects/Iperf/iperfdocs_1.7.0.html
20:31.45Bentleythought i'd check here 1st tho
20:31.59rikstai actualy didnt know of any tools for it Bentley
20:32.16rikstai dunno if ntop does that kinda stuff
20:32.45rikstaBentley: i think you want smokeping
20:34.00Bentleylooks interesting riksta
20:35.03*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
20:45.20*** join/#asterisk rvhi (~rv@66.175.65.89)
20:47.43shmaltzwho overhere is an admin to the asterisk-users list?
20:48.07Corydon-wThere are no mail admins
20:48.21Corydon-wNot any one person dedicated to the task, anyway
20:48.50denonhmm .. am I losing my mind here? I thought asterisk would playback .wav files by default as well ..
20:48.55denonits whining cause its not a gsm ..
20:50.00*** join/#asterisk toyos (~gabri@216.90.111.178)
20:50.15*** join/#asterisk h3x0r (Justino@ip70-180-167-6.lv.lv.cox.net)
20:50.22h3x0rwtf, ds3000p
20:50.27toyosHow do I do IAX2?
20:50.31h3x0rthis is a joke right
20:50.38Qwellh3x0r: no
20:50.39toyosnope
20:50.43shmaltzCorydon-w, someone unsubscribed me from the list
20:50.53shmaltzand when I try to resubscribe, it doesn't work
20:51.00h3x0ryoud need like 8 machines to codec translate anyway
20:51.01h3x0rheh
20:51.26h3x0rmaybe more
20:51.33Qwellh3x0r: 8 way opteron? :p
20:51.52h3x0ryeah right
20:52.13h3x0rit would cost more than a brand spanking new sonus, telica, apx 8000, anything
20:52.19h3x0rprobably
20:53.05Corydon-wshmaltz: have you considered that your ISP may have instituted an overly aggressive spam policy that it dumping all your list traffic?
20:53.21shmaltzCorydon-w, no
20:53.28shmaltz1. My ISP is Gmail
20:53.50shmaltz2. I got a message from the list telling me that I'm unsubscibed
20:53.51Corydon-wAnd you've checked all your spam folders?
20:54.00shmaltzof course
20:54.12Corydon-wWell, try emailing someone at Digium
20:54.19shmaltzoh, I just got my confirm for resubscribing
20:54.39Corydon-wCould also have been too many bounces which got you unsubscribed
20:55.14h3x0rand the new IAXy looks like a chinese radar detector
20:55.20shmaltzwhat do ya mean too many bounces?
20:55.53shmaltzlike NDRs?
20:56.53Veryhotanyone using Netlogic.net?
20:57.19*** join/#asterisk jhowardPA (~jhoward@12.25.177.120)
20:57.28jhowardPAHello people!
20:58.11jhowardPAriksta: I threw a copy of Xorcom Rapid on my box, rather than A@Home...  working great now.
20:58.26jhowardPAriksta: sorry I didn't get back to you - I was out sick yesterday.
20:58.36*** join/#asterisk greg_work (~greg@d221-73-240.commercial.cgocable.net)
20:58.40Veryhotjhowardpa: yeah I seem to like xorcom too
20:58.43greg_workwhat are the IAX2 and SIP ports?
20:59.10shmaltzgreg, SIP is 5060 by default
20:59.17jhowardPAAnyone know why my hold music would be silent, though I''ve got mpg123 (0.59r) playing properly?
20:59.18shmaltzcheck iax.conf, and sip.conf
20:59.19Veryhotjhowardpa: A@home should be good in couple more version
20:59.26greg_workand iax is 4569/udp ?
20:59.32jhowardPAVeryhot: I hope so  ;)
20:59.57shmaltzjhowardPA, any messages on the CLI?
21:00.14Veryhotjhoward: I have problem recording my menu on the A@home
21:00.17jhowardPAshmaltz: none - none at all.
21:00.17*** join/#asterisk bah (048830696@AC869EBC.ipt.aol.com)
21:00.39jhowardPAVeryhot: I gave up on it.  Made me sad.
21:00.41shmaltzjhowardPA, not evan telling you starting music or stopping music?
21:01.03Veryhotjhowardpa: then I have to do custom menu :)
21:01.06jhowardPAshmaltz: I believe it says it's starting on Asterisk startup..  double-checking...
21:01.25*** join/#asterisk lohelle (~post@213.161.252.253)
21:01.29shmaltzjhowardPA, when you put someone on hold, is it syaing anything?
21:01.54Veryhotanyone know of a opensource CDR program?
21:02.29jhowardPAshmaltz: no messages regarding music at all.
21:02.39Veryhotfor doing billing.
21:02.40jhowardPAshmaltz: nothing.
21:02.44shmaltzjhowardPA, even when putting on hold?
21:02.50jhowardPAshmaltz: yeah, not a thing.
21:02.55shmaltzthen your device is not putting on remote hold
21:03.00shmaltzwhat are you using?
21:03.12jhowardPAshmaltz: I think I don't have either oss or alsa initialized in modules.conf - is that the problem?
21:03.14shmaltzhow many vvvvvvvvvvvvvvv are you using with asterisk?
21:03.33shmaltzjhowardPA, how are you testing it?
21:03.43shmaltzno you don't need oss or alsa
21:03.54psycodaddid anybody ever run * on a vmware guest system ? I have linked a real system and a test host in virtual machine, an when i dial the *-demo on the remote via iax2 the sound is about 3 times toooo slow..
21:04.29jhowardPAshmaltz: I've got 2 Cisco 7940G's, and I've got extn 888 pointed at WaitMusicOnHold(30)
21:04.45lohelleDoes anyone have an example ser.cfg (SER proxy) to forward all all sip "traffic" to two (or more) asterisk servers?
21:04.48shmaltzand you are dialing 888?
21:05.26jhowardPAIf I do, it waits a bit, then says "something is terribly wrong" audibly, and hangs-up.  Placing a call on hold from phone-to-phone doesn't do anything (audible)
21:05.32rikstajhowardPA: great
21:05.42jhowardPAriksta: thanks  :)
21:05.45rikstajhowardPA: i'm just about to head out to the pub
21:05.49shmaltzand what is the CLI output
21:05.54shmaltz?
21:05.57jhowardPAriksta: I'm insanely jealous.
21:06.03jhowardPAshmaltz: none
21:06.07rikstajhowardPA: uhh? ok! hehe, laters
21:06.09shmaltzmake sure you have like 20 vvvvvvvvvvvvvvvvvvvvvvvvvv when starting the console
21:06.25jhowardPAshmaltz: "-vvvvvvvvvvvvvvvvvvvvvvvvvvv" ?
21:06.30jhowardPAfor verbose?
21:06.33shmaltzyep
21:06.37jhowardPAok, one sec...
21:07.09*** join/#asterisk habakuk (~chatzilla@24-119-164-129.cpe.cableone.net)
21:07.21jhowardPAexecuting moh on new stack...  started class 'default'
21:07.41shmaltzis that all it is doing?
21:07.47jhowardPAplaying tt-somethingwrong
21:07.52jhowardPAhangup
21:07.54shmaltznow check your musiconhold.conf
21:08.02shmaltzwhat does default say?
21:08.24jhowardPAIt timed-out before the somethingwrong.
21:08.28Veryhotcan someone recommend a opensource cdr program?
21:08.29shmaltzthen check the path that musiconhold.conf shows and confirm that you are hearing whats in the path
21:08.47jhowardPADefault says mp3:/usr/share/asterisk/mohmp3
21:08.53shmaltzVeryhot, doesn't really exist under GPL
21:08.58shmaltztry ASTPP
21:09.12Veryhotshmaltz: or something like under $500 :)
21:09.37shmaltzand ls in /usr/share/asterisk/mohmp3 returns what?
21:09.47jhowardPAIn /usr/share/asterisk/mohmp3, I've got the two mp3s that come with AMP (AMP is not installed): QuajiroPromo.mp3 and TristeAlegriaPromo.mp3
21:10.33jhowardPAthey were playing fine when I had Asterisk@Home installed on this machine, so I assume they're in the right format.
21:10.37jhowardPA(I copied them by hand, I didn't leave the drive intact between installs)
21:10.39shmaltzI have no clue what this mp3 are so do yourself a favor move those out of there, and put in your own that you know play something and test it
21:10.54habakukI'm trying an Originate from the manager interface. However I notice that CDR's generated are way out of whack. Has Anyone else noticed this problem?
21:11.11jhowardPAshmaltz: I'm playing one through the speaker right now, works fine.
21:11.14habakukI'm using CVS today btw
21:11.21shmaltzhabakuk, yeah
21:11.25jhowardPATried the second, works as well.
21:12.21habakukshmaltz: did you find out why that's happening?
21:12.25shmaltzdo pstree -G -a do you see mpg123 holding on to those files?
21:12.28jhowardPAHmmm...  Is there a config which binds the output of mpg123 to a channel on the PBX? :\
21:13.30jhowardPAshmaltz: files in pstree?  I see the mpg123 procs...  let me run lsof
21:13.39*** join/#asterisk bannerman (~bannerman@209.216.176.42)
21:13.54bannermanIs asterisk@home pretty cool?
21:14.07foobosbannerman, not really
21:14.13jhowardPAYeah, it's holding the first one open.
21:14.52bannermanfoobos: how so?
21:15.06jhowardPAshmaltz: where does the mpg123 output go?
21:15.18jhowardPAahh, stdout
21:15.21shmaltzwhere ever you tell it to
21:15.31shmaltzasterisk redirects it to itself
21:15.32foobosbannerman, too many noobies breakng the pre-programmed extensions.conf thingie
21:15.54bannermanfoobos: newbs break all sorts of things, whether they use asterisk@home or not
21:16.00bannermanfoobos: I know this, I am newb.
21:16.18jhowardPAshmaltz: what does asterisk do with it, once it's piped in?
21:16.26jhowardPAie, is it configged somewhere?
21:16.35shmaltzit will wake it up whenever MOH starts
21:16.41shmaltznah
21:16.56*** join/#asterisk PBX_Boy (L0ck@82-37-180-88.cable.ubr04.wals.blueyonder.co.uk)
21:17.52jhowardPAHmmm...  worked on A@H, so something's configged different...  is Asterisk calling mpg123 to wakeup by name?  Maybe it's off because I've got it in as mpg123-oss which is linked from mpg123...
21:18.41*** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com)
21:19.20jhowardPAHmmm...  strace mpg123 is showing very little going on.
21:19.26barmaltzanger: you there?
21:19.36sivanahe's gone
21:20.00barmalok....
21:20.39bannermanare there any other Bad Things in asterisk@home?
21:22.28shmaltzjhowardPA, try downloading mpg123 again and compile
21:23.00*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
21:23.00*** mode/#asterisk [+o bkw_] by ChanServ
21:23.09jhowardPAshmaltz: I did that, too.
21:23.16*** join/#asterisk file[laptop] (~file@mctn1-7126.nb.aliant.net)
21:23.20jhowardPAbefore I came here.
21:23.23jhowardPA0.59r
21:25.10*** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co)
21:26.29jhowardPAWHOA!
21:26.50*** join/#asterisk jmacz (~jmacz@63.245.86.225)
21:26.59jhowardPAManually running mpg123 with "--mono -r 8000 -b 2048 -f 8192" outputs "No supported rate found!"
21:27.10jhowardPAthere's a quick answer, I'd bet.
21:30.07bannermanquick, need handholding! I've got to go, and I would like to delay a download by an hour and a half (or make it run at 4 pm my time)
21:30.23bannermanhow do I schedule a command like that? is cron the way?
21:32.01eKo1Why are you asking such a question here?
21:32.19pgpkeysexactly
21:32.27pgpkeys#unix or #linux
21:32.47eKo1You're assuming he/she uses a *nux.
21:32.53shmaltzor maybe #downlad
21:32.59shmaltzI mean #download
21:33.11jhowardPAIf my mpg123 (0.59r) doesn't like the "--mono -r 8000 -b 2048 -f 8192" set of options, what's the matter?
21:33.31eKo1mpg123 sucks. Avoid using it if you can.
21:33.44shmaltzjhowardPA, get the defaults that come with asterisk
21:34.17eKo1You can always go the the source code dir. and type 'make mpg123'.
21:34.47*** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca)
21:35.02jf_any way to make * dial faster on zap channel
21:35.14eKo1Dial faster?
21:35.20jhowardPAI was wrong, it works if I pipe to stdout...  but it gives me a different error: "Warning, flexibel rate not heavily tested!"
21:35.25jhowardPAIs that important?
21:35.28Qwellno
21:35.34eKo1Ignore warnings.
21:36.02jhowardPAI would, if things were working  ;)
21:36.13jf_eK01:  yes, when i use my sip phone, i want to call someone on zap channel (fxo) it take like 4sec before i hear the ring
21:36.25jf_in sip phone
21:36.30jf_it's long 4 sec
21:36.30eKo1jf_: Press the send or pound button on your sip phone.
21:37.00jhowardPAI see, looks like that's due to the mp3 being VBR?
21:37.29eKo1Do not use variable-bit-rate mp3s.
21:37.35jf_ek01: u mean end the number by a #
21:37.42eKo1Yes.
21:38.01jf_let me try
21:38.08jhowardPAeKo1: Hmmm...  it's no a vbr mp3
21:38.18jhowardPAbrb
21:39.11jf_eko1: is it possible to auto do that
21:39.34eKo1What do you mean 'auto do'
21:39.51jf_so i do not have to press #à
21:39.53jf_#
21:40.14eKo1uh, what's wrong with pressing pound?
21:40.37jf_i want to * do that not me
21:41.19*** join/#asterisk hypa7ia (~leigh@67.71.86.109)
21:41.32eKo1I think there's a way to set the amount of time * will wait for digits when you type a number but I don't recall.
21:42.31eKo1But if you make it too short, then you will have trouble dialing.
21:42.44jf_oh ok
21:43.04Qwellor, don't use . as part of your extension matches
21:44.54bannermaneKo1: Because I need to schedule an Asterisk-related ISO download sometime after hours, and I was in here already. Sorry..
21:45.02*** join/#asterisk doughecka (~dheckaman@doughecka.user)
21:45.16Qwelluse `at`
21:45.19dougheckawhat would cause asterisk to not recognize dtmf tones from a cisco phone
21:45.36dougheckamine works fine, but this new setup I have its not wanting to see dtmf
21:45.45eKo1doughecka: check the dtmf mode.
21:45.59dougheckaits set to that rtf thing
21:46.02dougheckaorwhateveritis
21:46.11dougheckarfc
21:46.15*** join/#asterisk Tili (~Tili@202.133.65.241)
21:46.18dougheckabut it wasnt set before
21:46.37dougheckaand it said inband wont work with other codecs and info doesnt work with voicemail
21:46.41eKo1rfc2833
21:47.03eKo1Make sure _both_ * and the phone are using that mode.
21:47.27dougheckahow do I set it on the cisco
21:47.44eKo1Hell if I know. Read its manual.
21:47.59dougheckalol
21:48.09bannermanI have an occasional issue where I can hear the other side, but they can't hear me. Happens on both incoming and outgoing calls.
21:48.13dougheckaI am acting like a user today cause I feel like it
21:48.14doughecka:P
21:49.54bannermanMid-call.. the call works fine.. then incoming audio quits, for about 20 seconds.
21:50.57corlisHrm. great. misdn won't work with asterisk-cvs....
21:52.31*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
21:52.59dougheckahuh, odd
21:53.00FuriousGeorgedoes the source for the driver that i need to compile for the tdmp400 come with the zaptel package
21:53.02dougheckait doesnt see numbers
21:53.05dougheckabut it sees the pound sign
21:53.41QwellFuriousGeorge: yes
21:53.51*** join/#asterisk likwid-- (likwid@nc-205-240-44-137.dyn.sprint-hsd.net)
21:53.54FuriousGeorgemake <drivernaqme>?
21:54.00FuriousGeorge<drivername>*
21:54.04Qwellno
21:54.09*** join/#asterisk girabraz (~christian@200.121.129.178)
21:54.09Qwellits included
21:54.11FuriousGeorgemake config?
21:54.18Qwelljust make
21:54.54FuriousGeorgei get a module not found
21:55.00*** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63)
21:55.01FuriousGeorgeisnt it wctdm?
21:55.07FuriousGeorgefor the tdmp4000
21:55.10Qwelldid you make install?
21:55.26FuriousGeorgewhoops
21:55.32JohnnyCPortugal esta por ai alguem ?
21:55.44FuriousGeorgeeu fallo espanhol
21:55.55*** join/#asterisk gein (~gein@213.134.110.241)
21:56.11JohnnyCestou mesmo a procura de um tuga !
21:56.48CoaxDJohnnyC: no hablo el portuguesa o italiano pero.. sabes espanol? :)
21:57.01eKo1s/espanol/español
21:57.03FuriousGeorgeque e uma tuga?
21:57.18CoaxDeko1: o algo asi. jeje
21:57.20JohnnyCum portugues
21:57.38FuriousGeorgenos os chamamos "pork chops" ;)
21:57.52CoaxDFuriousGeorge: mmm.  carnes!
21:57.55eKo1eu no soi portugues
21:58.00CoaxDfuriousgeorge: yo queiro  yo quiero!
21:58.03FuriousGeorgelol
21:58.09FuriousGeorgechorizo na brassa
21:58.12FuriousGeorgeyummy
21:58.18CoaxDMmmmmmmm
21:58.20CoaxDsiii
21:58.36eKo1Chuleta?
21:58.38FuriousGeorgefrango assado, verdad, johnnyc.  o chraasco e muinto bon
21:58.59CoaxDhe estado estudiando espanol hace tres anos o algo..  necesito aprender mas pero entiendo, mas o menos
21:59.19eKo1CoaxD: You needs a spanish keyboard.
21:59.22FuriousGeorgemudase a newark NJ.  apprenderas espanol, portuguese, todo
21:59.52FuriousGeorgelo malo es que to olvidaras del ingles ;)
21:59.56CoaxDeko1: nahh
22:00.00CoaxDeko1: just a lappytop
22:00.04FuriousGeorgebut the food is soooo good
22:00.05DaLionheheh
22:00.06DaLionyeah
22:00.15CoaxDfuriousgeorge: jeje si!! pero poco a poco :)
22:00.24DaLioncreo que estas un poco loco cabron
22:00.40FuriousGeorgebest in new jersey, if you are near by, check it out.  i sh*t u not
22:00.43eKo1Cuantos anos tienes?
22:00.43CoaxDDaLion: hahahahaha
22:00.50FuriousGeorgedalion:  no manches, carnal
22:00.52CoaxDeko1: moi?
22:01.06eKo1jbot, translate es en Cuantos anos tienes?
22:01.13eKo1lol
22:01.16CoaxDhahahahaha
22:01.17FuriousGeorgelol
22:01.20Hmmhesays~e&m
22:01.21bugbote&m is assigned nothing and reported nothing.
22:01.29Hmmhesays~wink
22:01.31jbotACTION winks at hmmhesays
22:01.31bugbotwink is assigned nothing and reported nothing.
22:01.32Hmmhesaysheh
22:01.49Hmmhesaysi'm looking for some info on wink start
22:01.54CoaxDfuriousgeorge: tengo 28 anos ahora.. pero tengo mi cumpleanos en "may"
22:01.59Hmmhesaysso I don't sound like an idiot when this carrier calls back
22:02.38FuriousGeorgecoaxd:  ya eres un "viejete" en este mundo
22:02.48FuriousGeorge(el mundo de *)
22:03.13FuriousGeorgeps. may is  ez:  mayo
22:03.54CoaxDfuriousgeorge: mayo si
22:04.05CoaxDfuriousgeorge: como se dice "viejete" en ingles?
22:04.18FuriousGeorge"geezer"
22:04.24FuriousGeorgemore or less
22:04.29CoaxDfuriousgeorge: jeje siii.  de acuerdo :)
22:04.30|Vulture|yea
22:04.34|Vulture|geezer lol
22:04.37|Vulture|old fart
22:05.00*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
22:05.15eKo1pedo viejo
22:05.19FuriousGeorgei was brought up speaking gallego in a portuguese/spanish speaking neighborhoos, so i take up spoken language much better than *
22:05.32|Vulture|CoaxD: que dia en mayo?
22:05.38CoaxDVulture: 2nd
22:05.40QwellFuriousGeorge: Are you suggesting a speech config for asterisk?
22:05.41CoaxDVulture: tu?
22:05.50|Vulture|29th
22:05.53FuriousGeorgeqwell:  no
22:05.55Qwell:p
22:05.57CoaxDVulture: mmm.  not taurus!
22:05.58FuriousGeorgebut should i be?
22:06.09|Vulture|hehehe no I don't even know my sign
22:06.27CoaxDFuriousGeorge: The thing is, dude, i'm the exact opposite. I pick up config languages like mad.  but try to get me to learn a language..
22:06.31CoaxDFuriousGeorge:  I *suck* at it
22:06.49CoaxDFuriousGeorge: Only reason I learned so much spanish is because of this chick I used to work with; she didn't speak a word of spanish, but.. Man, was she ever hot.
22:06.52FuriousGeorgemais savoire des idomes est tres important
22:06.54FuriousGeorge!
22:07.23FuriousGeorgecoaxd:  thats great, what women will make us do
22:07.40CoaxDFuriousGeorge: I went out with her one night.  Got the spanish cook to come take her order and everything
22:09.11FuriousGeorgelol, chics dig spanish speakers, unless they are rednecks and you look like one too
22:09.17CoaxDFuriousGeorge: Hahahaha
22:09.23FuriousGeorgeiow, cant speak for middle americans
22:09.37CoaxDFuriousGeorge: Honestly, dude, I'm a total redneck. Always ahve been
22:09.50CoaxDFuriousGeorge: Note: High functioning redneck.  I don't have old beat up cars in my front yard, etc
22:09.51FuriousGeorgewould you date a borriqua?
22:09.53eKo1Chicks don't dig spanish speakers.
22:10.00eKo1Where did you get that from?
22:10.00CoaxDFuriousGeorge: wtf is that?
22:10.02FuriousGeorgelol, high functioning
22:10.12FuriousGeorgeborriqua is a porto rican chic
22:10.13CoaxDEko1: Chicks like guys who can speak more than 1 language. they think its interesting
22:10.18CoaxDFuriousGeorge: Oh, sure I would
22:10.29FuriousGeorgeek01:  then i guess chics dig me
22:10.30CoaxDFuriousGeorge: Not just because she was hot, tho. I never would do that. Gotta be somebody like me
22:10.30eKo1Fuck, I can speak three.
22:10.39CoaxDeKo1: Fluently?
22:10.40eKo1I don't see chicks falling all over me.
22:10.48CoaxDeKo1: You hang in the wrong crowds then
22:10.56FuriousGeorgeeKo1:  were you baeten with the ugly stick?
22:11.09eKo1That's besides the point.
22:11.22FuriousGeorgeright, because the fairer sex is never superficial
22:11.38marloweIs anyone using livevoip w/ sip using g.729 ?
22:12.00eKo1I'm kidding. I get with 3 new chics every week.
22:12.44FuriousGeorgeeKo1:  look out for rashes
22:12.50CoaxDeko1: Heh
22:13.05eKo1I use boots instead of condoms so don't worry.
22:13.06CoaxDeKo1: The crabs are easy to get rid of, too.  Just go to walmart. they'll hook you up
22:13.43FuriousGeorgethe HPV is harder though
22:13.44eKo1I wouldn't know. Never had crabs. Looks like you have though.
22:14.19FuriousGeorgewhich is green the fxo or fxs
22:14.30eKo1Green?
22:14.54FuriousGeorgei bought a tdmp400 and i got 2 fxos and 2 fxss, which is which
22:14.58FuriousGeorgetwo are green two are red
22:15.58FuriousGeorgeCoaxD:  so you would date a PRcan, where are u from
22:16.13eKo1I think red indicates an error condition.
22:16.36FuriousGeorgekeep in mind i spanish, so you wont offend me, i'm probably more gringo que tu, con mis pelos rubios
22:16.46eKo1You're from spain?
22:16.48FuriousGeorgeeKo1:  the actual daughter cards are made of green or red
22:17.06FuriousGeorgeno im from nj but everyone else in my fam is
22:17.16eKo1What? I have a bunch of FXO daughter cards and they don't have any color.
22:17.46eKo1Unless you mean the PCB.
22:17.56FuriousGeorgei do
22:18.00FuriousGeorgei forgot the name
22:18.02eKo1and if you're from NJ, then you're not spanish.
22:18.03CoaxDeKo1: Nah. never actually had crabs
22:18.20CoaxDeKo1: Had lice tho.  Just as bad. (Same genus, different species.)
22:18.26*** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net)
22:18.52FuriousGeorgeeK01:  we are getting into semantic arguments.  ask people what they are and they will say "I'm Irish" having never met anyone from ireland.  i actually have spanish citizenship, and as per you, im not spanish
22:19.07FuriousGeorgestandards keep getting tougher.  u must be against affirmative action
22:19.09eKo1I'm talking about nationality.
22:19.24FuriousGeorgei hope ur not talking culturally
22:19.24CoaxDFuriousGeorge: Have mercy on the guy. he's dating a puerto rican
22:19.29FuriousGeorgecuz u'd be way off base
22:19.48eKo1Not ethnicity (as there rarely are any 'ethnic' people around).
22:20.26FuriousGeorgei have spanish citizenship when i go through customs in spain, i go through the "nationals" line.  thanks for playing
22:20.41FuriousGeorgeeKo1:  u live somewhere with no ethnic people?
22:20.42eKo1So you are spanish then.
22:20.51FuriousGeorgedid we just make a circle?
22:20.57FuriousGeorgei have dual citizenship
22:21.03CoaxDeKo1: Do you think one must be spanish to have citizenship in spain?
22:21.15eKo1Well yes.
22:21.21CoaxDeKo1: (Do you have to be white to have citizenship in the USA?)
22:21.41FuriousGeorgeand conversely, asians born here must be asian nationals not americans
22:21.44eKo1No, you can be a dog for all I care.
22:21.48CoaxDNo. People here in the USA might well be Hmong, but they'll answer to "HEY, AMERICAN!"
22:22.00eKo1I like to keep things simple.
22:22.08CoaxDeKo1: Nothing is ever simple.
22:22.17FuriousGeorgelol, uh oh
22:22.33eKo1Well, I try.
22:22.45FuriousGeorgeso anyway, back to *
22:22.48FuriousGeorgewhich has the red pcb?
22:23.01eKo1I would imagine the FXS.
22:23.48FuriousGeorgefxs connects office phone to station (verifying, i always get those confues)
22:23.51FuriousGeorgeright?
22:23.59eKo1~fxs
22:24.00jbothmm... fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx
22:24.00bugbotfxs is assigned nothing and reported nothing.
22:24.02eKo1~fxo
22:24.06jbotforeign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo
22:24.12bugbotfxo is assigned nothing and reported nothing.
22:24.18FuriousGeorgeoh that clever jbot
22:24.32*** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
22:25.11eKo1Que hambre tengo, joder.
22:25.20CoaxDeKo1: :/
22:26.07eKo1jbot, I want a gyro.
22:26.28eKo1~gyro
22:26.30bugbotgyro is assigned nothing and reported nothing.
22:26.39eKo1Hmm...no gyros.
22:29.01FuriousGeorgeeKo1:  looks like the fxo is the red one
22:32.10BoRiSI know this probably doesn't exist but is their such thing as a virtual modem(sip that can talk back to *) adapter for windows?
22:33.29ElsharI suppose I don't see what you mean. If you wanted a modem modem, you could just buy one. If you wanted a sip phone for windows, there's a couple out there..
22:33.29*** part/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca)
22:33.36Elsharalong with an iax phone program too
22:34.05FuriousGeorgeyeah i think u r talking about www.xten.com
22:34.49ElsharYep, that's one
22:35.02Nuggetwell, it isn't a "virtual modem".
22:35.10BoRiSElshar: Its like that, I want to emulate a modem on my windows box... This emulation adapter connects back to asterisk via SIP. Then I can use winfax or some softfaxing programs that will use the virtual adapter to send faxes
22:35.19ElsharI see
22:35.20Elsharhmm
22:35.27*** join/#asterisk tsetane (~tsetane@212.4.33.58)
22:35.36Nuggetmodems don't talk SIP, real ones or virtual ones.
22:35.50ElsharI could've swore I've seen something like that though.
22:36.10Nuggetbut more importantly, trying to mix faxing and asterisk is nothing but disappointment and headache.
22:36.15BoRiSvirtual ones *can* as long as someone had written a driver/program to do that.
22:36.35Nuggeta "virtual modem" is a concept which exists only in your head.
22:36.48Nuggetyou're using that nonsense phrase to describe a device whose real name you don't know
22:36.49fearnornugget: uhhh
22:36.51BoRiSUmmm, no
22:36.51fearnoryou are so wrong.
22:36.58fearnorthere is 'virtual modem' thang.
22:36.58BoRiSYou are wrong Nugget
22:37.00Nuggetok.
22:37.03fearnormy TNT supports it just fine.
22:37.07fearnormodem emulation via TCP
22:37.13Nuggetand those virtual modems connect to asterisk for faxing?
22:37.25Nuggetinteresting.  ok, I'm wrong.
22:37.29fearnorthere's a rfc for that even. so you can control modem DCD and other things via tcp ;)
22:37.44BoRiSNugget, You were saying that a "virtual modem" concept was only in my head........Pffffff
22:37.48fearnornot to asstricks, but to my PC, it looks just a remote serial port.
22:37.58fearnorand i can use windows fax to send faxes and shiznit ;)
22:38.07Nuggetclever
22:38.24fearnorthis concept is about 10 years old
22:38.31fearnorgotte shiva!@#
22:38.37Nuggetwhat's it used for?
22:38.38BoRiShehe
22:38.47Nuggetmodem pooling or something?
22:38.52fearnornugget: yep pretty much
22:39.00fearnororiginal application was modem pooling
22:39.09fearnorbut you can use it for whatever
22:39.25fearnorbuy a 200$ max 6000 and rock on, 4 T1s worth of that ;)
22:41.48*** join/#asterisk guugmember (~guugmembe@200.6.223.209)
22:42.00eKo1virtual modem? You mean the ones on a T1 modem card?
22:42.33guugmemberhello, I am in a proyect where I have to put 48 remote extensions, besides IAXy is there other hardware solution more unexpensive and that talks IAX?
22:42.55eKo1no
22:42.56eKo1next
22:43.08fearnorguug: why iax.
22:43.14fearnorfuck iax.
22:43.23eKo1eh, no.
22:43.23guugmemberbecause sip is blocked in my ISP
22:43.24eKo1iax rocks
22:43.27Sedoroxfearnor:
22:43.33guugmemberfearnor, what do you recommend then?
22:43.39SedoroxGunnar: call them and bitch
22:43.39fearnorguug: get a non-ghetto ISP or just change ports, and SIP will probably work.
22:43.42tzangeriax > *
22:43.55fearnorif your ISP is blocking something, they'll block your iax too.
22:43.58eKo1iax >> ISP
22:44.26fearnormy main complaint about iax is that it makes no distinction between call control path and media path
22:44.39fearnorwell, that and the fact that only ghetto low-density devices support iax ;P)
22:44.39tzangerfearnor: that's its only shortcoming IMO but it's also a strength
22:44.44eKo1Who cares. It works, especially through NAT.
22:44.56fearnorso does SIP
22:44.58tzangerall you need to do to add it is to allow callbacks from dropped out calls
22:44.59fearnorktnxbye
22:45.07fearnor:)
22:45.25eKo1sip + nat = pain
22:45.32guugmemberso, besides of the protocol, anyting besides the IAXy?
22:45.40eKo1h.323
22:45.44tzangerI've heard from the yate crew that they have a sip stack that owrks very well through nat
22:45.48tzangerbut I've not tested it yet
22:45.51fearnorbingo
22:45.59fearnorits all in the stack, not teh protocol
22:46.18eKo1Well, there's nothing wrong with SIP conceptually.
22:46.31eKo1It's just that it sucks in NATed environments.
22:46.31tzangereKo1: I think it's a hideous protocol
22:46.35tzangertries to do everything for everyone
22:47.01fearnortzanger: once you realize what *could* be done with sip, you'll understand *why* it is so complex.
22:47.06eKo1What's wrong with that.
22:47.09FuriousGeorgei followed http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf and i get a line 142: Unable to open master device '/dev/zap/ctl
22:47.13tzangerfearnor: I don't want my VOIP stack to do everything
22:47.23tzangerfearnor: I want my VOIP stack to handle voice and maybe video, that's it
22:47.25FuriousGeorgeim trying to load the drivers for the tdm400 i just got
22:47.36HogieFuriousGeorge: 2.6 kernel?
22:47.38jhowardPAAnyone know how to debug mpg123 MoH problems?
22:47.45FuriousGeorgehogie: yessir
22:47.46jhowardPAI'm still not getting anywhere...
22:47.52HogieFuriousGeorge: udev?
22:47.54fearnortzanger: well, how about "remote presence"?
22:47.55FuriousGeorgeyup
22:47.59fearnorjabber-like
22:48.00tzangerfearnor: I have jabber for that
22:48.04fearnoror VMWI-like
22:48.06sivanahehe
22:48.09nvrsHogie, you are on the ball
22:48.09fearnornote, VMWI *is* remote presence.
22:48.10Hogiedid you read README.udev (or is it udev.README?) in the zaptel source dir?
22:48.21tzangerfearnor: I don't need my VOIP stack to handle mail and presence and everything
22:48.28FuriousGeorgethanks, i didnt know abut that, ill check it out
22:48.32tzangerI am a firm believer in using the right protocol for the right job
22:48.36Hogieit says so in the compile
22:48.38jhowardPA...nothing but silence on hold, though it looks like it's working.
22:48.39Hogieyou just have to watch for it
22:48.49Hogieer, says to do that
22:48.54FuriousGeorgehogie, i use gentoo, which compile
22:48.54fearnorgee, why not? its one protocol to support on your media gateway controller
22:49.05tzangerfearnor: no thanks
22:49.05Hogiedid you emerge?
22:49.15tzangerfearnor: I already have a lot of jabber rolled out
22:49.21fearnoryour MGC *needs* to know about remote presence to properly signal VMWI
22:49.24tzangerI already have a lot of FTP, SMTP and web services rolled out
22:49.26eKo1tzanger: so what would you rather use?
22:49.31fearnorshrug
22:49.37fearnortzanger: how does that scale for you ;)
22:49.41HogieI know if you compile zaptel from cvs, it will say "it looks like you are using udev, please read README.udev" or something like that
22:49.41tzangereKo1: IAX2 + Jabber really
22:49.49tzangerfearnor: it works pretty well so far
22:49.49FuriousGeorgei did en "emerge this && emerge that && emerge the other && rc-update add thisd default" overnight and went to sleep
22:49.57tzangerone protocol doesn't mean it'll scale well
22:49.57Hogielol
22:49.58eKo1I thought Jabber was just a chat client/server.
22:50.00nvrs# Section for zaptel device
22:50.01nvrsKERNEL="zapctl",     NAME="zap/ctl"
22:50.01nvrsKERNEL="zaptimer",   NAME="zap/timer"
22:50.01nvrsKERNEL="zapchannel", NAME="zap/channel"
22:50.01nvrsKERNEL="zappseudo",  NAME="zap/pseudo"
22:50.01nvrsKERNEL="zap[0-9]*",  NAME="zap/%n"
22:50.07sivana~pastebin
22:50.09jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
22:50.09bugbotpastebin is assigned nothing and reported nothing.
22:50.09tzangereKo1: presence too
22:50.10FuriousGeorgejust installed.  i have distcc running on 4 boxes and it took 8 hours ;)
22:50.12Hogieand dont forget the permissions nvrs
22:50.16nvrsstick that in the bottom of  /etc/udev/rules.d/50-udev.rules
22:50.21nvrspermissions?!~?!
22:50.26Hogieyes nvrs, lol
22:50.27nvrsOMG
22:50.31nvrsIve been doing it all wrong!!
22:50.34Hogieheh
22:50.48goldeneareKo1: sip + nat = pain --> sip + STUN + nat = ok
22:50.55Qwellnvrs: README.udev
22:51.03guugmembergoldenear, what is STUN?
22:51.22QwellSIP + /dev/urandom = pain
22:51.27eKo1STUN = SIP Tryint to Use NAT.
22:52.13Sedoroxahahah
22:52.19nvrsthe readme doesnt mention anything about permissions
22:52.25goldenearguugmember, http://www.voip-info.org/wiki-STUN
22:52.36guugmembergoldenear, im there
22:52.56eKo1Bottom line is, stun will not solve all your nat issues.
22:53.20nvrsHolgie: the readme doesnt mention anything about permissions
22:53.23Weezeywhoa, when I change my phone to g726 the at.gsm is irritating.
22:53.28goldenearstun doesn't work only for symmetric nat IFAIK
22:53.37*** join/#asterisk verge (~jfargen@rrcs-24-227-48-10.se.biz.rr.com)
22:53.37sivanaWeezey: hey
22:53.57Weezeyhey hey!
22:54.00goldenearIAX doesn't work with snat neither AFAIK
22:54.02Weezeybeen a while.
22:54.10guugmembergoldenear, whatever I want to run IAX, besides IAXy any other less expensive product?
22:54.12tzangergoldenear: it works pretty well with SNAT actually
22:54.20Hogienvrs: mine does:  http://pastebin.ca/9627
22:54.55tzangerboth endpoints natted makes it difficult to initiate a call without a port forward but otherwise it works just fine
22:55.20goldeneartzanger, you can't native bridge two IAX clients between SNAT
22:55.20nvrsHogie, different file
22:55.24Weezeyis g729 worth buying?
22:55.33eKo1Anybody here use the Mediatrix FXO gateway?
22:55.34Hogienvrs: mine was from 1.0 stable on april 11
22:55.36vergeHello #asterisk
22:55.38nvrsHolgie: which distro?
22:55.40tzangergoldenear: not without a little work
22:55.48jhowardPAI'm getting "Spawn extension (default, 888, 1) exited non-zero on 'SIP/501-26fa'" when I end a call that should be playing music on hold.  Anyone know what's up?
22:56.16Hogienvrs: cvs
22:56.19nvrsHolgie, think im doing 1.0.7
22:56.30*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
22:56.39eKo1jhowardPA: If you end a call, it will exit. This is normal.
22:56.51vergeWhen I call people they can hear me, but when people call me I can't hear them.
22:56.54goldeneartzanger, you have to have something (like an asterisk server) in the path for two IAX clients behind SNAT to work
22:56.57vergeIs that a problem with my dialplan?
22:57.17eKo1verge: We need more specifics.
22:57.20tzangergoldenear: or port forward, or do some stack magic... however a lot less than SIP would have to do to get the RTP stream to do the same
22:57.21goldeneartzanger, the same can be accomplish with sip and RTP proxy...
22:57.34jhowardPAeKo1: I assumed that a non-zero return was not a good thing...?
22:57.41tzangergoldenear: what is the difference between having an RTP proxy and having an unnatted * box in the middle?  none
22:57.55eKo1jhowardPA: eh, how else do you think commands exit.
22:58.15vergeI am using a SIP ATA. It's behind a NAT box, but the * server is not behind NAT.
22:58.20eKo1IAX was made with * in mind.
22:58.25jhowardPAeKo1: With "zero" - meaning clean exit?
22:58.47FuriousGeorgeHoagie:  im still getting the same error after editing the rules and permissions file as instructed.  do i gotta restart udev or something
22:58.50eKo1No. Zero usually means the command failed.
22:58.59*** join/#asterisk netofsickcoder (~netofsick@cpe-24-170-74-115.stx.res.rr.com)
22:59.01jhowardPAeKo1: I'm also not getting any music, still.
22:59.03vergeWhat doe you think eKo1? Do you need any more info?
22:59.16jhowardPAeKo1: I think you're mistaken.
22:59.36goldeneartzanger, right an rtp proxy is like an unnatted * box
22:59.37eKo1jhowardPA: I write * apps. I think I know what I'm talking about.
22:59.42*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
23:00.08*** part/#asterisk moy (~kvirc@201.135.105.124)
23:00.13jhowardPAeKo1: Usually, a non-zero return code implies an error state, most of those error codes identify the error state.  Is this not the case for Asterisk?
23:00.29eKo1jhowardPA: If you don't believe me, ask in the #asterisk-dev
23:01.08eKo1That or look at the code.
23:01.19eKo1You'll see that most apps return -1.
23:01.31jhowardPAAhhh, cool.
23:01.37jhowardPAThanks for clearing that up.
23:02.09vergeI am just trying to use G711.
23:02.37vergeeKo1: do you have any suggestions?
23:02.54jhowardPAI figured it would be more like I'm used to, ala 'false ; echo $?'
23:02.54FuriousGeorgeHoagie:  actually, i tried unloading and reloading the module after editing udev perms and rules, and now i get an error loading wcfxs
23:02.56eKo1verge: Make sure both phones are set to use ulaw/alaw and that these codecs are allowed in *.
23:03.06FuriousGeorgeZT_CHANCONFIG failed on channel 1: No such device or address (6)
23:03.37vergeeKo1: this was actually someone from the PSTN trying to call me.
23:03.44eKo1jhowardPA: I understand. It was confusing for me at first also.
23:03.46vergeI am connected to the PSTN via SIP as well.
23:04.03eKo1Using what?
23:04.12goldeneartzanger, using only 1 udp port in indeed a benefit in many cases (like having only 1 port to forword if you want to host an * server behing a nat/firewall)
23:04.36*** join/#asterisk bjohnson (~bjohnson@ip190-172.dsl.istop.com)
23:04.39vergeeko1: I connect ATA---->*----->Telasip.com
23:04.45vergeIt's all sip
23:04.46jhowardPAeKo1: could you suggest where I might go to research my music problem?  I can't identify a location which is failing to pipe the music into the SIP channel.
23:05.33goldeneartzanger, but this can also be a problem: CDR won't work after native briging of two endpoints ... neither call hold/forwarding :(
23:05.35jhowardPAI'm using WaitMusicOnHold(30)
23:05.36eKo1jhowardPA: Look in the wiki for the musiconhold.conf page.
23:05.57tzangergoldenear: which is why I want to make a patch where after a native bridge a callback to the dorpped server is made so it can update its CDR
23:06.04jhowardPAI did, none of it seemed to relate to the symptoms I'm seeing - ie, the fixes I identified were already correct.
23:06.42*** join/#asterisk fugitivo (~ajf@201.255.106.8)
23:07.02FengShuijhowardPA: What program are you sing for playing the MoH?
23:07.03eKo1jhowardPA: Bottom line is, mpg123 sucks; it's discontinued and will be dropped from * soon (I hope).
23:07.38goldeneartzanger: how will this work ? this looks like a crappy work around ...
23:07.40FengShuiyep.  Madplay is much better.  I've had stable MoH since I switched
23:07.47tzangergoldenear: nah
23:07.53jhowardPAI'm using mpg123 - 0.59r - Madplay's a better option?
23:07.55tzangershouldn't be bad
23:07.59jhowardPAI'll try that.
23:08.02jhowardPAThanks!
23:08.20tzangerA->B->C, A&C negotiate to drop B out, end of call A and C both issue an ACK'd IAX2 IE with CDR update
23:08.20FengShuijhowardPA: yeah, madplay doesn't spawn off all of those hung processes like mpg123 does.
23:08.29goldeneartzanger, also IAX has at the moment a pretty bad codec negocation capabilities
23:08.39tzangergoldenear: that too is being worked on
23:08.46tzangerin-call codec renegotiation
23:09.16vergeeko1? do you need anymore info?
23:09.59eKo1Make the calls and check which codecs are being used.
23:10.36goldeneartzanger and what about call hold/transfer not being possible during a native bridge call
23:10.38goldenear?
23:11.11tzangergoldenear: haven't heard about that one
23:11.41goldeneardid you try it ?
23:12.31goldenearcall hold (with music)/transfer works only when * stay in the path...
23:12.54nvrsI cant get my sound card to work.. is it really required for anything important in z
23:12.57nvrsasterisk
23:13.00tzangergoldenear: well no shit
23:13.05tzangerwhere do you think the music comes from
23:13.15tzangerif A or C is playing the music then it should work just fine
23:13.18Hogiefairies
23:13.21tzangerbut if B drops out then of course it ain't gonna work
23:13.23Hogieam I right tzanger?
23:13.30jhowardPADamn, madplay didn't fix it.
23:13.43jhowardPAI'm just not getting anything out of the phone...
23:14.43*** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net)
23:14.43jhowardPAThere it went with the "something is terribly wrong, goodbye" message.
23:15.39jhowardPAIs this the right syntax?  It looks right: "/usr/bin/madplay --mono -R 8000 --output=raw:-"
23:15.44goldeneartzanger indeed, but it would be nice if B could be back in the path when need (eg music on hold).
23:15.47Hogieif I have my handset have 2 line instances on my phone, will it ring the 2nd instance when a 2nd call comes in, or the first instance?
23:16.23FengShuijohwardPA: yeah, pretty muc.  Here's mine: /usr/local/bin/madplay -Q -z --atten
23:16.23FengShuiuate=-10 --mono -R 8000 --output=raw:-
23:18.04jhowardPAFengShui: Trying yours
23:18.09*** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
23:18.20jhowardPANo dice.  Still dead line.
23:19.55*** join/#asterisk netofsickcoder (~netofsick@cpe-24-170-74-115.stx.res.rr.com)
23:20.14FengShuijhowardPA: Do you see the process running?
23:20.48jhowardPAI do, it looks proper.
23:22.50habakukanyone using the manager originate function?
23:23.00jhowardPAstrace output looks good - it appears to be reading the mp3, and pumping the output to stdout.
23:23.31jhowardPAFor some reason, the SIP channel just isn't getting the feed.
23:24.21habakukjhowardPA: do you have a zaptel driver ?
23:24.35jhowardPAI'm using ztdummy - should I not be?
23:24.39habakukerr kernel module loaded
23:24.57jhowardPAzaptel                174048   4  [ztdummy wcusb]
23:25.49habakukjhowardPA: have you verified that otherthings like meetme work?
23:26.13jhowardPAhabakuk: No, I haven't.  How should I go about checking?
23:26.30habakuksetup a meetme room and see if that works
23:26.39jhowardPAOk, trying...
23:27.19*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
23:27.56jhowardPAYeah, meetme works.
23:28.05jhowardPAMy conf room 300 is working fine.
23:29.07habakukjhowardPA: ok so its got to be something else
23:29.27jhowardPAIs it something that'd be in my zapata.conf?
23:30.20goldeneartzanger, I wondering about you're cdr patch: what happens if one end doesn't terminate the call properly (network failure etc ...) ?
23:32.50tzangerI figure it would just take the update from one side
23:33.03tzangerI haven't written any code yet
23:33.19tzangerhmm
23:33.24tzangerI forgot onion in my salad
23:33.45goldenearbon appetit :)
23:34.36Hogieztcfg -vvv    shows what?
23:34.43jhowardPADamn, still no music...  I upped the volume, in case, but to no avail.
23:35.02jhowardPA0 channels configured.
23:35.06jhowardPAIs that it?
23:35.12Hogiedidn't mean to you howard
23:35.14HogieI forgot to hit tab
23:35.15Hogielol
23:35.18*** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net)
23:35.24*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
23:35.34denonPTG123: well well well? :)
23:35.54jhowardPAHogie: would that affect my music on hold?
23:36.01Hogieprob not
23:36.19jhowardPADamn...
23:36.20jhowardPA:\
23:36.45Hogiemy colo facility (computer, not telco colo) wants $900/month for a pri + cross connect:(
23:36.57denonyouch
23:37.01PTG123:)
23:37.10PTG123thats why i colo in a carrier neutral facility with no cross-connect fees
23:37.26HogieI HATE HATE HATE the dallas info mart
23:37.34Hogieno way I'd colo there
23:37.45denonPTG123: get a chance to visit our friendly local government office?
23:38.07PTG123hah my wife was suppose to do it today, when i see here tonight i'll get you the tracking # :)
23:38.20denongreat :)
23:38.34denonim headin home, cya
23:40.31FuriousGeorgecan anyone confirm when the status leds are supposed to go on, on the tdm400.  when drivers are modprobed?  or after ztcfg
23:43.23goldeneartzanger: what do you think about always maintening a link with * (only sig and info) even during native bridging with an other endpoint ?
23:44.07tzangerthat would probably be a lot more convoluted but I'd have to investigate it
23:44.27*** part/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
23:45.48tzangerjbot, furiousgeroge is Furious George, the Furious little monkey... See him come, as furious as can be...
23:45.49jbotokay, tzanger
23:45.54tzangeroh
23:45.56tzangerhe left
23:46.01tzangerI tough that was a join
23:47.05goldenearwhy a call park/transfer is not possible when * is not is the path
23:47.16fugitivohey, I did a nmap to my linksys pap2-na, look at this
23:47.17tzangeruh
23:47.19fugitivoOS details: Sipura SPA-1000 or SPA-2000 POTS<->VOIP gateway
23:47.25tzangerbecause the fucking box isn't in the loop anymore? :-)
23:47.47Qwellfugitivo: doesn't mean much
23:48.01fugitivoRunning: Sipura embedded
23:48.15QwellIt doesn't actually ASK the device what it is.  It makes a logical guess.
23:48.26tzangerif you, me and fugitivo are talking and I piss off for a smoke, how can you ask me for the time?
23:48.28fugitivoQwell: maybe yes, maybe not
23:48.34QwellI'd say knowing that its an ATA is a pretty good guess
23:48.49Qwelltzanger: piss off for a smoke?  Mind if I steal that one?
23:49.05tzangergoldenear: when B drops out, you lose access to all of B's services
23:49.08tzangerQwell: heh
23:49.10tzangergo ahead
23:49.14Qwellexcellent
23:49.18goldenearI'm test this with iaxcomm, and I can't tranfer/park the endpoint I'm talking to because of the native bridge :(
23:50.01tzangergoldenear: just say notransfer=yes in the iax.conf for the user/peer
23:50.45goldenearbut why couldn't my endpoint ask the * box to reinvite the other end ?
23:51.03tzangergoldenear: feel free to add that functionality if it's that important
23:51.15tzangerI see IAX2 as a simple stack for simple connectivity
23:51.27goldenearand it is :)
23:52.22goldenearhow does a tranfer works inside * ?
23:53.04tzangergoldenear: read the code
23:53.40goldenearbut basicaly, isn't it like a SIP reinvite ?
23:53.46*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
23:53.48*** mode/#asterisk [+o bkw_] by ChanServ
23:54.00goldenear(I'm not a skilled coder)
23:54.24tzangernor am I :-)
23:56.11TomL~seen manxpower
23:56.15jbotmanxpower <~eric@adsl-35-236-60.msy.bellsouth.net> was last seen on IRC in channel #asterisk, 4h 43m 2s ago, saying: 'EGADS!  I have a negative karma!'.
23:56.15bugbotseen manxpower is assigned nothing and reported nothing.
23:56.38MikeJ[Jayden]we take unskilled coders too ;)  no time like the pres to learn
23:56.58goldeneartzanger, but you said you're going to write a patch for * :)
23:57.07tzangerI've written several patches for *
23:57.11*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
23:57.30*** part/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
23:57.36*** join/#asterisk covici (covici@static-162-83-93-166.fred.east.verizon.net)
23:59.56seanwhen I do 'sip show peers' and the status column for my DID (sip proxy) show "unmonitored", is that normal? I want to be connected to the DID.

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