irclog2html for #asterisk on 20050905

00:00.04*** join/#asterisk hans (i=fugalh@falcon.fugal.net)
00:00.12Grubslast thing - asterisk *will* be transcoding as Office users connect with PAP2 using uLaw and we send out the wire using iLBC (seems clearer than native G729 on the ATAs *shrug*).....PIII 800 still good enough for 3-4 simultaneous calls?
00:00.17hansanyone have an echo test I can use? I want to test my QoS script
00:00.27spootnick"Outbound Registration: Expiry for myisp is 60 sec (Scheduling reregistration in 45 s"
00:00.40spootnickwth? i thought it would be 60s, not 45
00:01.08*** join/#asterisk akrall_ (n=akrall@customer-201-133-125-77.prod-infinitum.com.mx)
00:01.34akrall_Guys.. can somebody give me a hand with some questions regarding E1 and a TE110?
00:01.37FuzzyCatshould be ok grubs...
00:02.17FuzzyCatspootnick, becaus eit takes time to do stuff
00:02.43spacklemmmmm, Liquid bread
00:03.03FuzzyCatooo... car football
00:03.10spootnickamazing. it's always 15s, no variation
00:03.23spootnickso actually 75s means 60
00:04.07opus_fuzzycat I can't find anywhere show= option
00:04.20FuzzyCatit doesnt exist
00:04.28FuzzyCatit was a joke
00:05.12*** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net)
00:05.37Ariel_hello everyone
00:05.43puzzledhi Ariel_
00:05.43FuzzyCatlo puzzled
00:06.04akrall_Guys.. Im having a problem I guess with my TE110.. usually with TDM cards you modprobe zaptel and then the wcm drvier..I guess with TE110 E1 you just modprobe zaptel
00:06.09akrall_but when I do a ztcfg -v
00:06.16akrall_I get this:
00:06.24akrall_Zaptel Configuration
00:06.24akrall_======================
00:06.24akrall_<PROTECTED>
00:06.24akrall_SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
00:06.24akrall_<PROTECTED>
00:06.25akrall_31 channels configured.
00:06.27akrall_<PROTECTED>
00:06.29akrall_ZT_SPANCONFIG failed on span 1: No such device or address (6)
00:06.31akrall_whats this?
00:06.44Ariel_puzzled, ;-)
00:06.55FuzzyCatmodprobe wcte11xp not zaptel
00:08.53akrall_so... with the TE110 cards you also need to run modprobe ??? nohing Ive read says that
00:09.00akrall_where can I get the driver for the card?
00:09.11FuzzyCatyou have it
00:09.18FuzzyCatif you compiled zaptel
00:09.19akrall_for example, for TDM04B cards you modprobe zaptel then modprobe te tdm driver and everyting works fine
00:09.25akrall_but this is my first TE card :(
00:09.31FuzzyCatyou don;t need to modprobe zaptel
00:09.41FuzzyCatthat will happen when you modprobe the driver
00:09.49akrall_lsmod says
00:09.55akrall_Module                  Size  Used by    Not tainted
00:09.55akrall_zaptel                183936   0
00:10.01akrall_ok
00:10.04akrall_let me check
00:10.05FuzzyCatjust modprobe wcte11xp
00:10.20akrall_modprobe wcte11xp? is this the e1/t1 single span?
00:10.43akrall_nice! error is gone
00:10.56akrall_is this ok?wcte11xp               22560   0  (unused)
00:10.56akrall_zaptel                183936   0  [wcte11xp]
00:11.13FuzzyCatnow ztcfg -vv
00:11.34akrall_all 31 channels showing on ztcfg -vv
00:11.39akrall_no errors
00:11.51akrall_Module                  Size  Used by    Not tainted
00:11.51akrall_wcte11xp               22560   0  (unused)
00:11.51akrall_zaptel                183936   0  [wcte11xp]
00:12.09*** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net)
00:12.41FuzzyCatsee, magic, when you do what ur told it works :)
00:13.27*** join/#asterisk luke-jr__ (n=luke-jr@CPE-65-26-132-140.kc.res.rr.com)
00:13.32akrall_hehehe exactly! I never saw anything regard the driver so.. I was puzzled
00:13.38akrall_since for x100 and tdm
00:13.42Ariel_FuzzyCat, why do we do what we are told??? and should we??
00:13.43akrall_cards I was used to loading the driver :)
00:13.46akrall_so... :)
00:13.49akrall_thx man
00:13.56FuzzyCatnp akrall_
00:14.03akrall_BTW, anybody using this for unicall and mfcr2?
00:14.32FuzzyCatAriel_, well if you ask a question and don't do what we say it's not our fault if you can't get it working
00:17.51akrall_Im trying to recompile asterisk with unicall but I get a lot of erros and wont compile. Anybody tackled those issues?
00:18.31bkw__lalallalalal
00:19.21opus_damn it i should file more bugs in asterisk
00:19.28*** part/#asterisk hans (i=fugalh@falcon.fugal.net)
00:19.37bkw__opus_, why?
00:19.40bkw__you find a new one?
00:19.53opus_yeah, if a macro calls itself asterisk segfaults
00:20.01bkw__haha ya thats fun
00:20.03bkw__file it boi
00:20.15*** join/#asterisk kentclaussen (n=blueboy@204.13.224.246)
00:20.26FuzzyCatRange Rover Vs Challenger Tank
00:20.51*** join/#asterisk jamesmi (i=root@JB007.hvi.lt)
00:20.52Ariel_FuzzyCat, I was just making a joke.
00:20.58FuzzyCatoh..
00:21.00FuzzyCatok...
00:21.02FuzzyCathahahahhah...
00:21.04FuzzyCat;)
00:21.05kentclaussenbkw!!!
00:21.33SwKbut doesnt everyone have  [macro-foo] exten => s,1,Macro(foo)  in their dialplans?
00:21.44*** join/#asterisk ManxPower (n=eric@69.149.125.137)
00:21.46spackleis Ditka in the range rover?
00:22.00opus_how come I don't get the 'Dial' command in the CLI?
00:22.00FuzzyCatno, clarkson
00:22.03Ariel_self looping call
00:22.06opus_is it a zap only feature?
00:22.13FuzzyCatyou don;t have a sound card
00:22.22FuzzyCator it's not configured
00:22.36opus_the server is 4 miles away anyway
00:23.14kentclaussen4 miles is better than 400..
00:24.22*** part/#asterisk kentclaussen (n=blueboy@204.13.224.246)
00:24.28*** part/#asterisk jamesmi (i=root@JB007.hvi.lt)
00:24.30*** join/#asterisk churley34M (n=cehurley@miro.voltaiccommerce.com)
00:25.37bkw__lkasdjfoej
00:25.43FuzzyCatgfchmngfrftrghyj
00:30.45FuzzyCatlol
00:30.53FuzzyCatwtf
00:30.57FuzzyCatmeh!
00:31.16opus_lets see. i know a few other ways to segfault asterisk right now
00:32.29Ariel_opus_, this is on the head or the 1.2beta1
00:33.00ManxPowerIf I spend much more time in TX I'm going to pick up the accent *shudder*
00:33.14bkw__ManxPower, you were in Waveland?
00:33.19Ariel_ManxPower, wow would that not be strange
00:33.29*** join/#asterisk lika (n=sacura_c@200.69.125.1)
00:33.31ManxPowerbkw_, I WAS in Waveland.  Evacuated before the storm.
00:33.45ManxPowerIf I start saying "varments", you have permission to shoot me.
00:33.46bkw__you going to even try to go back to see if anything is left?
00:34.08ManxPowerbkw_, ariel photos seem to show that I live in the 10% of waveland that was not leveled.
00:34.18bkw__ah
00:34.24ManxPowerno visible roof damage, no visible other damage.
00:34.27Ariel_ManxPower, you were talking of moving to another contry did not know it would be the contry of Tejas
00:34.55ManxPowerI live about as far from the ocean as you can and still be in Waveland.
00:35.06*** part/#asterisk lika (n=sacura_c@200.69.125.1)
00:35.19ManxPowerI HOPE to go back this coming week sometime.
00:35.37*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
00:36.32hugo-v6ManxPower: and everything u had is stolen or burned?
00:36.51ManxPowerhugo-v6, Hmm?  No sign of fire.
00:38.21Ariel_bkw_, your site needs a better menu or ways to get info on voip service.  Less then 3 cents...hummm.
00:40.01hugo-v6ManxPower: well the news said that the ppl there have done much plundering and burned down lots of buildings
00:40.42ManxPowerhugo-v6, In Waveland MS?
00:40.59hugo-v6hell no. new orleans. i think i missed a detail ;)
00:41.15ManxPowerhugo-v6, Um, I don't live in New Orleans, I live 50 miles east of New Orleans.
00:41.24opus_hell. i just submitted a 'crash' problem as minor, hmm.
00:41.28opus_oh well.
00:41.38*** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
00:41.41ManxPowerI just used to tell people I lived in New Orleans since before this week nobody knew where the fuck Waveland MS was.
00:42.05hugo-v6ManxPower: i still didnt know it since u told me that ;)
00:42.32ManxPowerhugo-v6, reports are that 90% of Waveland MS was destroyed.
00:42.52ManxPowerhowever, ariel photos of my area show very little descruction.
00:42.55hugo-v6that doesnt sound better
00:43.09hugo-v6well that sounds good
00:43.17ManxPowerBut as I said, I live about as far from the ocean as you can and still be within the city limits
00:44.43hugo-v6but whats about water-supply and possible dirt and epidemics?
00:44.51ManxPowerThe NOC at corporate HQ of my largest customer is currently up and running (they are about 40 miles north of New orleans), power is on, about 2/3 of the corporate wan is up and running.
00:45.15hugo-v6sounds good
00:45.17ManxPowerhugo-v6, New Orleans is the only area under water.
00:45.29ManxPowerthe MS gulf coast is dry
00:45.59ManxPowerRemember, CNN and the rest of them are only showing the destroyed areas, since that's what gets the ratings.
00:46.05hugo-v6good.
00:46.11SkramX<PROTECTED>
00:46.14SkramXoops
00:46.20*** join/#asterisk spootnick (n=julio@50.118.233.220.exetel.com.au)
00:46.56ManxPowerlarge parts of new orleans IS under water, but large parts of the city and surrounding areas are NOT under water.
00:47.32ManxPowerfor example, the New Orleans air port, where recue operations are being run out of, is only about a 30 miles from downtown new orleans and it's dry.
00:47.49*** join/#asterisk |cleric| (n=dacleric@p5482AB35.dip0.t-ipconnect.de)
00:48.24ManxPowerand our ISP, which is located downtown new orleans is up and running off of generator power and is arranging airlift of fuel for their generators.
00:48.49ManxPowerMost of our CLEC's network is also up and running.
00:49.21hugo-v6doenst sound that bad at all
00:49.38ManxPowerOn the other hand, when I checked into the hotel today, I talked to one of my former customers that is also stayinh at the same hotel and he has pics of the area where he lives and the water is up to the roof.
00:49.45*** join/#asterisk Pkunk (n=Pkunkage@mbbs.munnabhai.info)
00:50.05hardwireblah
00:50.18hugo-v6bad
00:50.53ManxPowerOne of the people that evacuated with me lives on the westbank and none of the levies broke there, so they should be high and dry.
00:51.25ManxPowerOne of the other people that evacuated with me lives in uptown new orleans and the water appears to not even cover the parked cars there.
00:51.46*** join/#asterisk SarahEmm (n=sarahemm@2.35.220-216.q9.net)
00:51.49SarahEmmsivana: you around? ;)
00:52.03ManxPowerIt CAN be tough to tell from the satalite photos, however.
00:52.14spootnickManxPower: i heard they said on cnn that around 60% of the police force deserted
00:52.27ManxPowerspootnick, Yeah.  I don't believe that.
00:52.43gambolputtyIs there a way to dial by SIP URL into a * box even if one isn't authenticated?  I got this error message so far:  Failed to authenticate user "a@a.com"
00:52.48*** join/#asterisk loud (n=ariel@cypher.punk.net)
00:52.54ManxPowerThe police force in New Orleans get paid like $25,000/yr.  They have to be pretty commited to just stay on the force in normal times.
00:53.15ManxPowermost of the have to take 2nd jobs just to pay their bills.
00:53.44*** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl)
00:54.04drraywell the bribes from corruption augment their salary
00:54.47spootnickManxPower: can't blame them if they did desert... but why don't you believe that may have happened?
00:54.49*** part/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
00:55.49spootnickgambolputty: http://www.nolata.org/wiki/Handle_request:_Failed_to_authenticate_user
00:55.49*** join/#asterisk ManxPwr (n=eric@69.149.125.137)
00:57.07ManxPwrspootnick, because the police radios are not even working.  There is so much chaos in New Orleans there is NO way to even estimate how many police offices are still on duty, how many died, and how many are on the streets helping people and not checking in.
00:57.42gambolputtythis would be a guest call into my box
00:58.12Ariel_it's a major problem. But things will get better. And I feel we need to look at the good that is going on more then the bad.
00:58.14spootnickgambolputty: i think so, but i'm not sure. i think sip debug can say that, or somebody more experienced
00:59.09spootnickAriel_: that's true. maybe the media is saying more than it's actually happening. but maybe not. i'm just amazed seeing all this happening in the US
00:59.27Ariel_gambolputty, if you setup your default context in the general area of sip.conf you can direct an statemant like exten => s,1,Play(What are you doing here)
00:59.29spootnickAriel_: not the disaster, but the government reaction
00:59.38*** part/#asterisk SarahEmm (n=sarahemm@2.35.220-216.q9.net)
01:00.07Ariel_spootnick, I lived through a major one that was not has hard due to the flooding but never the less bad.
01:00.20Ariel_And they alway say the bad and never put on the news the good stuff.
01:01.05Ariel_spootnick, besides the feds can't go into the states with troops due to our laws. The Governor needs to request this. And it's something I would like to keep in the state level
01:01.41*** join/#asterisk azrishahril (n=nasa5435@61.6.68.202)
01:01.47ManxPwrI'm stil trying to contact my landlord
01:02.06Ariel_ManxPwr, it took me 1 month to contact mine
01:02.32Ariel_back in the time of Andrew. And I lost the house completely. just a pile of sticks and stones
01:03.51spootnickAriel_: i understand. i just think that if the law doesn't allow a quick response in this kind of situation, then there should be another way of doing things
01:04.20spootnickprobably easy for me to say, i know, but still, it's shocking
01:04.22Ariel_spootnick, there is the state has the national gaurd at there beck an call.
01:04.30Kattybeep.
01:04.47Ariel_Katty, evening
01:04.56KattyAriel_: (=
01:05.13spootnickk, back to asterisk, what's the difference between the "s" and the "i" extension?
01:05.51KattyAriel_: how is family?
01:05.51Ariel_Well the s is start. the i is incomplete number like, t is for timeout.
01:06.10spootnicki thought "i" standed for an invalid extension number
01:06.20spootnickthat's how I'm using it in my dialplan, in fact
01:06.22Ariel_Katty, going krazy today.  all the girls in the house except the little is well that time of the month...
01:06.37Kattyeek!
01:06.48Ariel_spootnick, yes it does sorry incompelet is handled by t for timeout
01:07.17Ariel_ARGH just got yelled at by the wife for smilling at my screen.
01:07.44spootnickweird. voip-info.org wiki says " 's' is used when there is no known called number in the context used."
01:08.01spootnickthat sounds like an alias for "invalid" extension  =)
01:08.01hugo-v6Ariel_: did she thought u looked on pr0n?
01:08.03Ariel_spootnick, yes it's correct
01:08.10Ariel_pattern match happens first
01:08.20Ariel_s is start when no pattern match
01:08.26spootnickumm
01:09.16Ariel_spootnick, also if you put zap to imediate=yes (check spelling) it goes to the s extension.
01:09.51spootnicki'll start playing with zap in the next 5 days. so far i used asterisk for sip/iax only
01:10.24spootnickAriel_: so if immediate=yes, everything goes to s ?
01:10.30*** join/#asterisk hat (n=hat@bb220-255-134-33.singnet.com.sg)
01:11.02Ariel_spootnick, yes but it's used for like door phones and things that don't dial
01:11.38ManxPwrWell, I just got some good news.
01:11.40*** join/#asterisk techie (n=gus@70.86.57.50)
01:11.49*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
01:12.27ManxPwrprices of houses outside of New orleans at at %125 of what they were.  This is good news for me and my clients
01:12.52Ariel_ManxPwr, really nice
01:13.05Ariel_so do you have some houses your going to be selling?
01:13.15hatGood morning. i am looking for information/tutorial about digium E1 card and its asterisk configurations. Who can help?
01:13.26ManxPwrAriel_, Gads no.
01:13.39ManxPwrBut 90% of my consulting income comes from a real estate company
01:13.39Ariel_~docs
01:13.40jbotwell, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
01:13.42*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
01:13.44ManxPwrand they only lost 3 offices
01:13.57Ariel_ManxPwr, great to hear it.
01:13.59hatthanks jbot_
01:14.37*** join/#asterisk newlook_ (n=newlook@203-59-168-152.perm.iinet.net.au)
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01:15.55Grubsdoes anyone here know the inside of a Compaq DL380 3U rackmount server?
01:16.19ManxPwrAriel_, Another one of my customers just came back online
01:16.50Himekoif it is alot liek a ml370 yes
01:17.30GrubsI'm wondering if there is a fan in front of the redundant PSUs.  Each has an exhaust fan - but is there another fan in the front end also?
01:18.14Grubs(planing a cheap but near silent server)
01:18.18Himekohttp://h20000.www2.hp.com/bizsupport/CoreRedirect.jsp?targetPage=http%3A%2F%2Fh200001.www2.hp.com%2Fbc%2Fdocs%2Fsupport%2FSupportManual%2Fc00218203%2Fc00218203.pdf
01:18.21Himekomight be in there
01:18.26Ariel_ManxPwr, soon you will be up to your neck in work again.
01:18.36Grubsthx
01:18.40Himekohttp://h20000.www2.hp.com/bizsupport/CoreRedirect.jsp?targetPage=http%3A%2F%2Fh200001.www2.hp.com%2Fbc%2Fdocs%2Fsupport%2FSupportManual%2Fc00292624%2Fc00292624.pdf
01:18.44Himekothat ones is prolly better
01:18.51Himekoit is the maint and service guide
01:20.12Himekoyep 2 fans
01:20.15Himekoone fromt one back
01:20.55hatWho has experience of using TE411P digium card? I need a recommended computer server to host this card. From voip-info.org, it seems that some server specs has problem for digium card.
01:21.00GrubsIn my last two 4U servers I put a "wall" of three near silet glacialtech fans between the drives and the motherboard and then filled in the space around them with heat resistant close-cell-foam - excellent air flow and almost silent.  Now want to convert the cheap ebay compaq to something similar.
01:21.42spootnickis there a way to detach from the console without having to stop asterisk?
01:21.49GrubsThanks Himeko - much better than the user guides I found
01:21.54spootnickand without putting the process in the bg as a job
01:22.02drrayspootnick - exit
01:22.07drraythen asterisk -r
01:22.09drrayto reconnect
01:22.30*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
01:22.59*** join/#asterisk hellagony (n=egutierr@200.121.213.88)
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01:46.30Kattytoo quiet.
01:46.46Ariel_yes it is.
01:46.50Ariel_but sometimes it's nice
01:47.03harryvvyes
01:47.11Ariel_you should hear it here. baby crying mom yelling and 17 year slamming doors
01:47.43harryvvwhere apartment or next door
01:48.43harryvvariel you have a ip500?
01:51.40Ariel_harryvv, house mine
01:51.54Ariel_and yes at work I have about 15 at my location
01:54.18*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
01:56.21hugo-v6.o(snom snom snom)
01:57.49Ariel_polycom polycom polycom.
01:59.08Ariel_damm a bottle of milk for the baby hurts when it's sent at you like a shooting rockit.
01:59.11*** part/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
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01:59.31ManxPwrI never could understand why someone would want a pet human.
02:00.01hugo-v6hrhr indeed me 2. better get a dog.
02:00.07hugo-v6thats what i did :)
02:00.20hugo-v6but hell... im only 24 ;)
02:00.39*** join/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
02:00.57*** part/#asterisk vuvie (n=vuvie@bb219-74-47-90.singnet.com.sg)
02:01.49opus_right on
02:01.51opus_i filed 4 new bugs
02:02.14hugo-v6damn should get some amphetamines from a doc or local dealer *yawn* so much todo and no time.
02:02.26opus_i hvae one more in mind, but will wait a few days to see if it 'goes away'
02:02.54Hmmhesaysi've not learned how to try another server if one fails in ser
02:02.56Hmmhesaysrock
02:03.10Hmmhesayst_on_failure
02:03.14Katty...
02:04.09jskcrHmmhesays:  use srv records
02:04.22Hmmhesaysjskcr: no
02:04.26Ariel_argh kids... I need a noise cancel head set.
02:04.29DarthCluetoo bad those were pots and not pans...
02:04.42DarthClueAriel_: with time, you learn to auto-cancel
02:05.20Ariel_not when you have a house full of girls and all of them well most of them it's there time of the month.
02:05.39DarthClueAriel_: um, well...
02:05.47jskcrHmmhesays:  You can have failure routing in your script and check for the status of the server
02:05.48Ariel_that is why i am going to work in the morning....
02:06.02Hmmhesaysjskcr: yes
02:06.34jskcrHmmhesays:  rember you must also route cancels or you phones will keep ringing after someone hangs up too
02:06.59Hmmhesaysyes
02:07.18jskcrHmmhesays:  you can also have ser authenticate to the same database that asterisk uses with the realtime version
02:07.23h3xjust get a scram jet pilots headset
02:07.23h3xhaha
02:07.35spootnickjskcr: that happened with me quite a few times. people hanged up then the phone would ring right after
02:07.53jskcrspootnick:  its because you have to route cancels properly
02:08.06spootnickand "route cancels" means.. ?
02:08.12jskcrspootnick: you have to handle them bouth inside and outsite the main routing logic
02:08.13spootnickhandle hangups?
02:08.18jskcryup
02:08.43spootnickwell you know, i think i solved it accidentally, because after a recent revamp on my dialplan, that suddenly stoppep
02:08.49spootnickstopped*
02:08.51spootnickbut i was never sure why
02:08.55file[laptop]Hmmhesays!
02:08.58KattyHmmhesays: mrow?
02:09.10Hmmhesayshey Katty
02:09.30KattyHmmhesays: can i get a little yum yum, kitty kitty?
02:09.34Ariel_hay h3x I have my head set in the car.. there for my airplane and they have Noice canceling.....
02:09.44Hmmhesaysjust a little something somethin'... itty bitty
02:09.53h3xhaha
02:09.55Kattythat song is hott
02:09.56DarthClueHmmhesays: that song is just so wrong
02:10.01h3xif i wore one of those
02:10.07HmmhesaysDarthClue: wrong and good
02:10.11Hmmhesaysdoom boom is good too
02:10.12h3xit would just cancel the noise comign from the bitch i was talking to on the phone
02:10.24spootnickjskcr: so if i have this 'exten => 2,n,Dial(${EXTENSION1},25,rmtT)', then i should add '2,n,Hangup()' after?
02:10.38bkw__damn it
02:10.42bkw__haha
02:10.43bkw__no
02:10.59file[laptop]let's NOT drive people away
02:11.13Hmmhesaysmaybe, 'the special'
02:11.16DarthCluefile[laptop]: um, i know people who would pay to see that
02:11.21Ariel_2,n
02:11.29Kattya couple madonna songs
02:11.29jskcrspootnick:  you need to change your routing logic to handle cancels properly
02:11.32Kattyfrozen++
02:11.38DarthCluefile[laptop]: i know alot of people who would pay to see you and bkw dance
02:11.46file[laptop]pfft
02:11.48file[laptop]together?
02:11.56file[laptop]show me some money and I'll do it :P
02:12.00bjohnsonnaked
02:12.01DarthCluefile[laptop]: we would have to charge more for that
02:12.09file[laptop]naked? that costs a lot extra
02:12.12Ariel_I knew someone would put the naked in
02:12.14hathi, what is E1 signalling?
02:12.34opus_european T1 signalling
02:12.41Kattyi dreamt about a chic last night
02:12.44DarthCluesomething tells me that Katty will be trying to find a way to get back to Diamonds again
02:13.00KattyHmmhesays: you would have liked my dream.
02:13.10KattyHmmhesays: you should have dropped by in it :P
02:13.22puzzledis there any benefit in running Asterisk on a 64bit platform?
02:13.23KattyDarthClue: eh..it was ok.
02:13.27KattyDarthClue: too many males there.
02:13.36KattyDarthClue: and way too direct.
02:13.41KattyDarthClue: subtle++
02:13.52*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
02:13.53Hmmhesayscrank it up when you're sucking on a stick of doom boom boom
02:14.06Ariel_puzzled, depends
02:14.22KattyHmmhesays: yum.
02:14.22puzzledAriel_: on what?
02:14.35Ariel_puzzled, transcoding
02:14.44HmmhesaysKatty: Doom Boom is a good song
02:14.52KattyDarthClue: :<
02:14.53puzzledAriel_: maybe a little g729 but majority is ulaw/alaw
02:15.02Ariel_then no different
02:15.05jskcrif (!(method == "INVITE" || method == "ACK" || method == "CANCEL" || method == "REFER" || method == "BYE")) { log(1, "no invite,ack,cancel,refer->return 404\n");                 sl_send_reply("404", "Not Found");                break;        };
02:15.12puzzledAriel_: ok, thanks
02:15.15DarthClueKatty: i prefer to hide in the shadows, makes it easier to 'observe'
02:15.20KattyDarthClue: k
02:15.27Ariel_but the more transcoding yes and the more channels with meetme and things like that yes
02:15.28jskcrdo a method==CANCEL
02:15.43ManxPwr<PROTECTED>
02:15.50KattyHmmhesays: did you read the article /. awhile back about running bad vodka through a water filter?
02:16.00jskcrthen send sl_send_reply and to forward the cancel from asterisk to ser
02:16.02KattyHmmhesays: and it suddenly being good vodka?
02:16.24puzzledManxPwr: empty as in totally empty?
02:16.28HmmhesaysKatty: yeah, doesnt' work worth a shit
02:16.30DarthClueManxPwr: really?  um, i hope it was the first.
02:16.37Hmmhesaysand i'm not wasting good filters on bad vodka
02:16.46Kattyk
02:16.51ManxPwrpuzzled, no idea.  Just talked to the cab driver I know there and she said someone was taking stuff out of my place.
02:16.53Kattyi obviously wouldn't know
02:16.56Kattyseeing how i don't drink.
02:16.59ManxPwrProlly my landlord.
02:17.01Kattybut someone has to take your tail home.
02:17.13HmmhesaysKatty: true
02:17.22hugo-v6hmmm around 4am here and about 18°C. was on a walk with my dog right now :)
02:18.03hugo-v6no crying, nothing. besidethe fact that he tried to catch a cat. never seen my dog that fast in the middle of the night.
02:18.13DarthClueHmmhesays: HAC is addictive
02:18.30HmmhesaysDarthclue: oh hell yes
02:18.32KattyLacrimosa is addictive
02:18.32jskcrspootnick if (method=="CANCEL") log(1, "CANCEL message received\n"); to see if its getting the cancel messaGES
02:18.35Kattymmm, german opera
02:18.46Kattytilo wolff is yummy
02:19.04DarthClueKatty: don't mention Lacrimosa, I'm just getting over them
02:19.13DarthClueLacrimosa puts me in an evil mood.
02:19.16hugo-v6Katty: what opera?
02:19.24KattyDarthClue: strange..they put me to sleep :P
02:19.30hugo-v6DarthClue: lacrimosa is evil ;)
02:19.33Kattyhugo-v6: german opera.
02:19.47Kattyhugo-v6: lacrimosa is symphonic rock orchestra opera stuff
02:19.52hugo-v6Katty: well... yes, but whats the name of the opera?
02:19.59hugo-v6ahhh opera.
02:20.03Kattyhugo-v6: its a band...
02:20.07DarthClueKatty: well, they put me to sleep at some point too, but they do make me a bit moody
02:20.21KattyDarthClue: well it is considered gothic music :P
02:20.29KattyDarthClue: goths are generally somewhat moody
02:20.36KattyDarthClue: or extremely chipper.
02:20.49DarthCluehugo-v6: the 'Echos' and 'Lichtgestalt' albums
02:20.54Kattymopey -> boingboing
02:21.06DarthClueKatty: bipolar
02:21.06spootnickjskcr: that's a .ael script?
02:21.16hugo-v6i thought of series like enterprise ;)
02:21.22KattyDarthClue: no, i'm not bipolar
02:21.30KattyDarthClue: but i do get moody
02:21.31Hmmhesaysi am slighty
02:21.36jskcrspootnick:  thats for ser.cfg
02:21.36Hmmhesaysand an alcholic
02:21.37DarthClueKatty: no, but most goths are somewhat
02:21.38KattyDarthClue: though...i'm not completely goth. just psuedo
02:22.05DarthClueHmmhesays: you are a bipolar schitzo with paranoid delusional tendencies
02:22.12KattyHmmhesays: does that make you a bi alcholic polar bear?
02:22.18spootnickjskcr: hm, ok , i'm not using SER over here
02:22.19DarthClueand yes, they are watching you
02:22.41jskcrspootnick: ps -ef | grep ser
02:22.47jskcrand you will know real fast
02:22.57HmmhesaysKatty: no, just a slightly bipolar person brought on by excessive alcohol consumption
02:23.19Hmmhesayswho needs a shower
02:23.20KattyHmmhesays: k
02:23.21spootnickjskcr: yeah, i know, i'm not running it
02:24.06KattyHmmhesays: i just got done with my bath.
02:24.19*** join/#asterisk remmo (n=rem@smack.isp.net.au)
02:24.42Kattyi guess i could get the towel off my hair :/
02:24.44*** join/#asterisk litage (n=nick@203.201.98.84)
02:24.52spootnickspootnick: i have asterisk behind NAT. i'll be configuring a few other ones behind nat as well. but setting up nat=yes, extenip=xx solved my problems
02:25.00remmois anyone using a digium card and a ericsson tigris?
02:25.06spootnickspootnick: so i ended up never even installing it
02:25.11spootnick(dug)
02:25.12*** join/#asterisk charles___ (n=charles_@adsl-149-1-140.mia.bellsouth.net)
02:25.17charles___wow
02:25.22spootnickjskcr: so i ended up never installing it
02:25.25*** join/#asterisk Snake-Eyes (n=blog@203.201.98.84)
02:25.32charles___anybody have ever got this:
02:25.43charles___/chan_iax2.so: undefined symbol: ast_check_signature
02:26.23hugo-v6hmz. i have to remove a hdd from one box to put it into another. but i dont want to :/
02:26.36puzzlednite all
02:26.42hugo-v6gd nite puzzled
02:28.09hugo-v6Katty: good girls drink soda, bad girls drink vodka :)
02:28.26hathi, who can help me about E1 card configuration? I am new to such thing and evne don't know what is MFC R2, pri_cpe. Any information is appreciated?
02:28.29hat.
02:28.31Kattyhugo-v6: what about vegan ones?
02:28.34tzangerKatty: cosmo
02:28.39Kattytzanger: mrow?
02:28.56KattyHmmhesays: what doesn't taste like alchohol?
02:29.03hugo-v6Katty: vegan drinks? vodka is only made from corn ;)
02:29.22bjohnsonhugo-v6: other way around
02:29.32Kattyhugo-v6: and i presume all bread is vegan too then, right?
02:29.36jskcrhugo-v6:  vodka is made from potatoes
02:29.41Kattyhugo-v6: it's just flour or wheat or whatever...right?
02:30.51hugo-v6jskcr: not really, backward it was. now its made from corn everywhere (at least i dont know where its still made from potatos) mabe in a small village in .pl or .ru/.ua
02:31.15spootnickwhat about AEL? did anybody replace extensions.conf with it and lived to tell the tale?
02:31.17hugo-v6Katty: i guess... but if u ask so, it makes me think u think im false ;)
02:31.31Kattythe vodka Hmmhesays had didn't smell like anything.
02:31.38Kattywhich was kinda nice.
02:31.45Kattyalchohol smells horrible.
02:32.09charles___charles /chan_iax2.so: undefined symbol: ast_check_signature
02:32.21hugo-v6since i eat meat (at best raw) im not a vegan or veggi
02:32.55harryvvhugo want it for the protine?
02:33.06hugo-v6Katty: vodka out of the freezer is best. best vodka i know of is "absolut"
02:33.17hugo-v6harryvv: not only ;)
02:33.19harryvvIm going to try tofu again
02:33.26harryvvits loaded with it
02:33.33hugo-v6bah, tofu tastes like nothing
02:33.36Kattyharryvv: as a point of warning, tofu is best when regarded as an ingredient in the recipe
02:33.37harryvvbut
02:33.49Kattyharryvv: when it's the main ingredient, it sucks.
02:33.49harryvvput flavoring in it
02:33.50hugo-v6i love the taste of meat.
02:33.54Kattyharryvv: better to use it like flour...or a binding agent..
02:33.57harryvvmine is strawberry tofu :)
02:34.01Kattyharryvv: or to mimic scrabbled egg...
02:34.07*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
02:34.11Kattyharryvv: or to whip it into pie, etc.
02:34.15harryvvyea
02:34.16harryvv:)
02:34.21hugo-v6harryvv: if i want something similar i eat feta or something
02:34.49hugo-v6strawberry tofu? u americans are crazy :)
02:34.59harryvvI personally want to increase my endurance and streagth but dont want to look like a fricken gorilla. Some guys at the weight room look like that.
02:35.14harryvvhugo yea im american but im in canada. so its sold here.
02:35.28hugo-v6well ok ;) then it cant be that bad :)
02:36.04harryvvwish that weight room had a inversion table my back is still in corrctive recovery.
02:36.10opus_awesome
02:36.17opus_i'm converting my car to run off of firewood
02:36.28harryvvopus...i saw that on tv
02:36.40harryvvguy uses stove pellets as a fuel
02:36.48opus_do you know where he got the engine
02:36.48*** join/#asterisk funraps-as (n=funraps@cpe-66-75-82-216.socal.res.rr.com)
02:36.51opus_i'm looking on ebay
02:37.01Kattyi found a really pretty one for 250 at REI
02:37.17harryvvopus send me the link...while its interesting i will be going boidiesel soon.
02:37.23funraps-ashello all
02:37.23hugo-v6i've run my car with rape oil. since it was a diesel
02:37.36harryvvhugo how long
02:37.39hugo-v6now its dead :( rip. good ol pal :(
02:37.45Hmmhesayssublime rocks
02:37.50harryvvohh what killed it?
02:37.51Kattyyour socks
02:37.57Hmmhesaysright off
02:37.58hugo-v6harryvv: til my waterpump broke and the engine was overheated on a higway
02:38.07harryvvthat sucks
02:38.16hugo-v6damn right.
02:38.17harryvvand you never noticed it overheat?
02:38.18harryvv:)
02:38.34Hmmhesaysfunny if they removed the impeller from the water pump you wouldn't really have a problem driving on the highway
02:38.42Hmmhesaysbecause the pump is passive
02:38.50harryvvI only heard my car over heat once...the engine knock was gettign louder then the stereo :)
02:39.02hugo-v6harryvv: temp display was broken and i never thought about a worst case and saw no need to fix it
02:39.13hugo-v6now i know it better
02:39.18harryvvyea
02:39.19harryvvhehe
02:39.29harryvvit takes only that one time.
02:39.34Hmmhesaysi wouldn't drive a car with no temp gauge
02:39.59Hmmhesaysi don't even like having no oil pressure gauge, but I get over that
02:40.08harryvvI wonder why cars dont have a float in the radiator that measure water height.
02:40.11hugo-v6well that says everyone. kown i know it :(
02:40.17Hmmhesayscause it'll sound like the drummer for green day is playing basket case on my car
02:40.39Hmmhesaysoften lack of coolant is not a problem
02:40.48hugo-v6harryvv: that would be a good idea
02:40.52Kattywho let the roos out!
02:40.52Hmmhesaysno
02:41.04Hmmhesaysthat would be more or less useless
02:41.06hugo-v6but mine doesnt got such feature
02:41.29Hmmhesayswhat you need is a gauge that works
02:41.42hugo-v6Hmmhesays: yes. ure right.
02:41.47Hmmhesaysa float in the radiator would be pointless
02:41.57hugo-v6i think i need that gauge really
02:42.10Hmmhesaysif you wanted to add a feature to a conventional cooling system then they should add a flow rate meter
02:42.26harryvvHmm okay or a water height sensor.
02:42.28hugo-v6Hmmhesays: or in the watercycle
02:42.40Hmmhesaysno, not a water height sensor, again... pointless
02:42.49harryvvHmmhesays not everyone watches there water temp at all times.
02:43.00remmoi'm getting taps on my te110p zaptel interface, but all echo cancellation and detection is turned off, any ideas?
02:43.20Hmmhesaysharryvv: the problem is not the features of the current sensing system, it is the alerting system
02:43.25hugo-v6damn its nearly 5am. got to switch that hdd now.
02:43.34opus_my 99 maxima went into safe mode when i notice the heat was above normal.
02:43.35Hmmhesaysthat is why they invented idiot lights
02:43.42opus_wouldn't rev more then 600rpm
02:43.54hugo-v6opus_: nice
02:44.18opus_its the same car that, while in cruise control at 60 and shifting into neurtral it will red line the engine. an obvious bug
02:44.23Hmmhesaysheh opus_ imagine that happening on an icy north dakota highway in a manual transmission car
02:44.41Kattyowch
02:44.51Hmmhesaysfront wheels would skid before you had a chance to hit the clutch
02:45.05opus_Hmmhesays whoah shit
02:45.16hugo-v6sounds funny :) damn gotta fix that now. brb
02:45.29Hmmhesaysyou'd flip upside down into the ditch come across the other side and kill a family on their way to christmas dinner
02:45.39Hmmhesaysdog and all.
02:45.45*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
02:45.52bkw__that was the dumbest "bengay" commercial I have ever seen
02:45.54funraps-ashey folks
02:45.55opus_<nop_:#interdictor-scanner2> [#i paste: scanner2: <lvlass> I understand that the hazmet strike team is in the area because of the chemicals in the area and dead
02:45.55opus_+animals]
02:46.02Kattybkw__: mrow?
02:46.09funraps-asseems that no one is talking voip here
02:46.12funraps-asBKW
02:46.20funraps-ascan I ask a few quickies
02:46.22opus_voip voip voip
02:46.22Kattyfunraps-as: so?
02:46.26funraps-asI know how much you like em
02:46.27funraps-aslol
02:46.34Kattyfunraps-as: just because we don't /always/ talk voip doesn't mean we don't know anything ;)
02:46.39opus_or voice over kermit
02:46.41funraps-asI know
02:46.45funraps-as:-)
02:46.46Kattythere must be time for cars and alchohol and kittens and things
02:46.51Kattyoh, and shopping
02:46.54funraps-asagreeed
02:46.56Kattythere is /always/ time to talk about shopping
02:46.58funraps-ashate to butt in
02:47.01Hmmhesayshaha, i rarely talk voip in here
02:47.07KattyHmmhesays: liar.
02:47.08opus_i saw driving in my car, talking on voip,
02:47.08funraps-asis there a channel I can ask
02:47.09funraps-as?
02:47.16Kattyfunraps-as: just ask
02:47.19DarthCluevoip?  you must think this is #asterisk or something
02:47.19funraps-asthanks
02:47.21HmmhesaysKatty: well at night
02:47.31funraps-asI want to setup SER with Asterisk...
02:47.31KattyHmmhesays: oh well that's a different story
02:47.40funraps-asI have both already setup
02:47.43DarthCluecan we get that back in the topic 'Just ask the damn question'
02:47.48Hmmhesaysi'm going to rock the pool table tonight
02:47.57funraps-asboth work just fine
02:48.07funraps-asbut I want to setup openser as the proxy
02:48.10funraps-asand routing logic
02:48.15funraps-asasterisk to do the rest
02:48.24Hmmhesayscause I bought a new 4 pack of pocket t's and feel sexy kinda
02:48.29funraps-asdocs online so far that I found
02:48.31funraps-assuck!
02:48.43funraps-asso lookind for some help
02:48.46funraps-asexamples
02:48.47funraps-asetc
02:48.49funraps-asthe idea is>:
02:48.57funraps-asI have broadvioce
02:48.59funraps-aswith 100P
02:49.09*** join/#asterisk spootnick (n=irc@CPE-147-10-168-100.nsw.bigpond.net.au)
02:49.24jskcrfunraps-as:  why setup ser on such a small setup
02:49.26funraps-asand I want users on my lan to use the ast as pbx
02:49.33funraps-asthen route everything out
02:49.41funraps-asjskcr
02:49.47funraps-asbecause I want to learn :0(
02:49.55funraps-asknowledge is always good
02:49.58funraps-aseventually
02:50.05funraps-asfriends and family all over the workd
02:50.07funraps-ascan register
02:50.11funraps-asand talk to us :-)
02:50.26funraps-asI've had nat issues with *
02:50.41funraps-asan from what I've read SER fixed all them issues
02:51.16funraps-asdoes that all make sense
02:51.24funraps-asor am I talking out of my a$$
02:51.55jskcrasterisk can handle nat just fine
02:52.33funraps-asreally?
02:52.48funraps-asI've had issues where calls go through but no voice
02:53.06opus_jskcr oh yeah
02:53.08opus_http://bugs.digium.com/view.php?id=5112
02:54.07jskcrI said nat not double nat :P
02:54.45opus_:)
02:55.12jskcr~ser
02:55.13jbot[ser] Sip Express Router - see http://www.iptel.org/ser/
02:55.54jskcrtry http://www.google.com/search?hl=en&lr=&safe=off&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&q=sip+express+router++asterisk+site%3Alists.digium.com&btnG=Search
02:55.58hugo-v6puh. done. i hate that
02:56.00jskcrfor some ser - asterisk examples
02:56.10Kattyhi.
02:56.18jskcrhi Katty
02:56.50funraps-aslet me check the link..
02:57.26*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
02:57.48funraps-asthats what I want to setup
02:57.49funraps-asSER: for SIP registration and IP call routing, incoming number
02:57.49funraps-astermination, STUN, Nat traversal etc.
02:57.49funraps-asAsterisk: outgoing call routing, calling card platform, billing,
02:57.49funraps-asextended facilities e.g. voicemail etc.
02:58.24EquinoxDoes a PRI have 23 or 24 usable channels?
02:58.47Katty23
02:58.53jskcryup
02:58.57Kattyone channel is used for signalling and data stuffs
02:58.59jskcr23b+1d
02:59.04EquinoxThat's what I thought
03:00.01funraps-asjskcr
03:00.05bkw__YAY Katty
03:00.08funraps-asthose are all lists
03:00.12funraps-asemails people sent
03:00.17funraps-asgoogle search
03:00.22Kattybkw__: mew?
03:00.26funraps-asno real docs that I can see
03:00.27jskcryup youll find tons of ser.cfg examples there too
03:00.33Kattybkw__: did you insane?
03:00.35funraps-asstill looking
03:00.40jskcriptel.org for the focs
03:00.43jskcrerrr docs
03:01.31jskcrWith ser you must write you own routing logic on how to handles calls and so forth
03:02.39SkramX800 inbound services: What billing incremement is your 800 inbound?!?
03:02.41funraps-asyea
03:02.46jskcrfor more ser docs you can goto http://www.voice-sistem.ro/index.php?lang=en&page=Technology&sub_page=documentation-repository also
03:02.55funraps-asbut simple routing logic is okay
03:03.02funraps-asast will do the rest
03:03.16funraps-asnat is the thing I want to fix
03:03.27funraps-asI have two networks under my lan with nat
03:03.37charles___does anybody had ever had /chan_iax2.so: undefined symbol: ast_check_signature
03:04.24funraps-aslooking at docs now
03:04.26funraps-asthanks....
03:04.30hugo-v6charles___: nope. tried too google for "undefined symbol: ast_check_signature"?
03:04.32*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-90-1.tu.ok.cox.net)
03:04.36jskcrnp
03:04.58DaminWelp..
03:05.09DaminOhio Linuxfest has been slashdotted..
03:05.51Daminhttp://home.neo.rr.com/meffie/ohiolinux/reg-count.png
03:06.59gaggamancharles: try to load res_crypto.so in modules.conf
03:07.56charles___hugo-v6 yes but I just got the source code
03:08.04charles___hugo-v6 I already tried doing cvs and stable
03:08.23charles___hugo-v6 and already deleted /usr/lib/asterisk and /usr/include/asterisk
03:10.20opus_charles do you have openssl installed and openssl-devel?
03:10.34charles___yes I do
03:10.59opus_hmm
03:10.59charles___<PROTECTED>
03:10.59charles___libopenssl0.9.7-0.9.7c-3mdk
03:10.59charles___openssl-0.9.7c-3mdk
03:11.00charles___libopenssl0.9.7-devel-0.9.7c-3mdk
03:11.14opus_strange.
03:11.24opus_can you reproduce it on a different computer?
03:11.32charles___don't think so
03:11.38charles___on the other server IAX just goes thruu
03:12.00charles___that one is mandrake 10 the other is 10.1
03:12.18opus_yeah, no luck on google either?
03:13.13*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
03:13.18hugo-v6since i dont got that problem, i cant hel you.
03:13.37charles___yeah
03:13.43gaggamanopus doesn't he possilbly just need to load the module manually?
03:13.45charles___kind of something not detected at compiling time
03:13.54charles___gaggaman I'm doing it
03:13.58charles___I can't load IAX
03:14.00opus_its a shared module
03:14.06opus_loaded when asterisk starts
03:14.09charles___it miss ast_check_signature
03:14.15opus_try
03:14.20opus_running 'ldconfig'
03:14.32opus_then running again
03:15.27charles___opus_ wow man a simple ldconfig
03:15.46charles___opus_ I can't believe that I was getting hurt by that
03:16.40hugo-v6hrhr. ldconfig and depmod 2 things a man should run after each install/systemchange :p
03:17.21opus_they should figure out away to automatically build them
03:17.59opus_now I think the asterisk makefile should automagically run ldconfig
03:18.02opus_but apparently not
03:18.25*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
03:19.00remmoanyone have a te110p working right?
03:19.05charles___ldconfig is done on most make installs but not on ast
03:19.11remmomine keeps looping back
03:19.34*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
03:19.56shido6remmo, whats wrong?
03:20.15remmoloops back every now and then
03:20.36remmoSep  5 13:19:13 WARNING[30748]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 30: Yellow Alarm
03:20.36remmoSep  5 13:19:13 WARNING[30748]: chan_zap.c:5684 handle_init_event: Detected alarm on channel 31: Yellow Alarm
03:20.36remmoSep  5 13:19:13 WARNING[30748]: chan_zap.c:1938 pri_find_dchan: No D-channels available!  Using Primary on channel anyway 16!
03:20.37remmoSep  5 13:19:13 NOTICE[30748]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 1
03:20.38remmoSep  5 13:19:13 NOTICE[30748]: chan_zap.c:5679 handle_init_event: Alarm cleared on channel 2
03:21.07shido6what version of libpri / zaptel are you using?
03:21.17Grubsdodgy plug connection on the cable?
03:21.25remmoi have used everything from 1.0.9 to 1.2 beta 1 to cvs head
03:21.51remmogrubs: could be. thats the only thing i have not tried
03:22.18remmoi have an e100p in another machine and its working better than the te110p
03:22.33remmowhats weird is it works, but quality is bad and lots of drop outs.
03:22.43remmosame config as the e100p boxen
03:22.48shido6did you whipe out every remnance of the old before tyring to use the new?
03:22.54shido6as you dont want
03:22.56shido6version skew
03:23.15shido6brb, greg@nufone.net if u need me
03:23.16remmoi believe i have, had same problem when i first installed 1.2b1
03:23.46charles___does anybody have TELIAX ?
03:24.55hugo-v6puh. damn. i hate it to make backups from windows. well at least i can test norton ghost 9 :>
03:26.10hmodesbooored
03:28.50opus_<lvlass:#interdictor-scanner2> I'm not quite sure what support is avail with this, but we do have dogs that are wandering wild and they are attackihng
03:28.58opus_shit
03:29.05tessierMMmm...dog...
03:29.11tessierSounds like there is plenty of food down there.
03:29.19*** join/#asterisk TheCops (n=mdb@206.248.136.146)
03:29.25TheCopsHi
03:29.53TheCopsSomeone heard about a Public pay phone who is working via SIP protocol, with some pay fonctionnality?
03:30.15hugo-v6hmmm mine doesnt attack nor goes wild.
03:30.39tzangerTheCops: interesting, since IIRC the payphones control singals are all well above the 3100Hz telco bandwidth
03:30.48hugo-v6unless he sees a cat :>
03:31.04jskcranyone know how to extract the configureation information on a at&t callvantage dlink?
03:31.36TheCopstzanger, and asterisk is managing signal of >3100hz ?
03:31.42tzangerTheCops: most ATAs won't
03:31.48TheCopsduh
03:32.04tzangerand * has no way of generating singals above 4kHz due to the 8kHz sample rate
03:32.12opus_yeah it does
03:32.19opus_there is a ultrawideband patch for speex
03:32.23opus_and a new g722
03:32.35tzangerwell with those developments then yes it may be possible
03:33.06dougheckaanyone link gtalk into asterisk yet?
03:33.15opus_no
03:33.19dougheckahmm
03:33.31opus_however, people are moving to using DNS SVR with VOIP recently because of gtalk
03:33.43TheCopstzanger there's no payphone with built-in SIP functionnality ?
03:33.44opus_like, you dial by email address
03:33.49dougheckaah
03:33.50dougheckasweet
03:33.55tzangerTheCops: I don't know of any, no
03:34.00TheCopsok
03:34.06opus_yeah, there was an interesting review of gtalk about it
03:34.13tzangertell me why would they installa  DSL connection for a payphone unless they're giving it data access too
03:34.18opus_i think the future is DNS SVN but i may be wrong
03:34.26dougheckahow to install ztdummy?
03:34.38opus_doug make install, make config
03:34.43opus_modprobe ztdummy
03:34.48opus_modprobe zaptel first
03:34.51opus_then run ztcfg
03:35.02dougheckahmm
03:35.04opus_if your running fedora core, then you need to modify some udev type files
03:35.38dougheckahmm, would need to install zaptel first eh? :P
03:35.44TheCopstzanger, I have already DSL business and line, why not put a public phone for doing money ?
03:35.47dougheckaI knew how to do that, just didnt install it
03:36.06TheCopstzanger my business is in the center of the city, good spot for get some people call
03:36.24Vcopeople still use payphones?
03:36.32TheCopsyeah :P
03:36.49TheCopsbaby boomers is using it
03:37.43hugo-v6TheCops: not a bad idea ;)
03:37.49hathi, i try to configure the TE411P digium card. how to specify the signaling value in zapata.conf file? I am confused by the possible values such as pri_cpe, pri_net and mfc/r2 etc.
03:38.46Vcosounds to me like yo shoudl back away from that console.....
03:40.26TheCopshugo-v6, I hate to get idea that doesnt exist
03:40.27TheCopshehe
03:42.12Kumbanghat: what do you expect to connect to?
03:44.26*** join/#asterisk xai (n=pasta@cpe-70-112-17-10.austin.res.rr.com)
03:47.53hatKumbang, sorry?
03:48.18hati am new to E1 technology.
03:49.32shido6its not that hard
03:49.39Kumbanghat: are you expecting to connect it to telco, then you should know what type of signalling your telco is?
03:50.58*** join/#asterisk bmg505 (n=leon@rndf-146-6-74.telkomadsl.co.za)
03:51.22jdiskywlkrI've been trying to build Asterisk on a Solaris machine for a while.  I keep running into a problem when compiling streamplayer.o.  ld returns the errors undefined refrence to 'gethostbyname', 'socket', and 'connect'.  What might be causing this problem?
03:51.24hatyes. Kumbang, but what is the possible signalling for the telco?
03:52.03hati try to understand more about the terminologies
03:52.59*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
03:53.24*** join/#asterisk diegows (n=diegows@201.250.118.14)
03:53.34hatin addition, i try to get an IBM server. but i am not sure which one is better to work
03:55.54*** join/#asterisk ManxPwr (n=eric@69.149.125.137)
03:59.07shido6asterisk works on sparcs
03:59.57*** join/#asterisk iq (n=iq@71-38-67-181.omah.qwest.net)
04:00.04jdiskywlkrit is a sparc
04:02.09Vcooooh...
04:02.42Vcoi should get off my ass and try to install on a U60
04:02.48hathi shido6
04:02.52Vcotommorow....
04:03.27shido6hello
04:04.10hatshido6,i type message in a seperate window
04:04.53*** join/#asterisk chendy (n=Alex_Dot@web1.ningo.net)
04:06.30xaiManxPwr: you still awake?
04:13.30ManxPwrxai, going to sleep soon
04:13.59ManxPwrIt's quite odd bot being subscribed to the mailinglists
04:14.53*** join/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
04:15.20*** join/#asterisk santiago (n=santiago@63.245.86.163)
04:15.23*** part/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
04:15.45SwKhey waiter, bring a pitcher of beer every 7 minutes til someone passes out, then bring one every 10 minutes
04:17.46JerJeryep, its about that time to go back to school
04:17.51*** join/#asterisk spootnick (n=irc@CPE-147-10-168-100.nsw.bigpond.net.au)
04:17.53*** part/#asterisk akrall_ (n=akrall@customer-201-133-125-77.prod-infinitum.com.mx)
04:19.06*** join/#asterisk dexteruk_ (n=dexteruk@de22399.alshamil.net.ae)
04:19.22spootnickis there a way to force asterisk to re-register with a sip proxy? after a few retries, it simply won't reconnect anymore
04:19.36SwKreload chan_sip
04:20.06harpertrowis anyone using vonage ATA w/ a ZAP trunk successfully?
04:20.09spootnickSwK: ok, but let's say i'm not aware it lost the connection
04:20.50SwKwhy's it failing to register?
04:21.03spootnickit timeouts a few times
04:21.05*** join/#asterisk ptblank (n=MURDER1@68.169.160.44)
04:21.25SwKharpertow; yes several people have done that
04:22.12spootnickapparently, it tries 4 times then it stops trying. sip show registry shows me "failed" for my numbers, and it stays that way
04:22.21spootnicksip debug doesn't show any attempts to reconnect
04:22.46harpertrowSwK; I have added trunk on channel 4 (my fxs interface), and have incoming calls & outbound routing set up, but nothing seems to work
04:22.51SwKyou can try setting registerattemps
04:23.03SwKcheck sample configs
04:23.36hathello, for TDM400P card, can i assigh fxo or fxs to ports myself? Does it mean i can set the usage for this card in zaptel.conf, right?
04:23.37harpertrowmy analog extension is working fine, so the digium card itself seems to be OK
04:23.39spootnickSwK: yeah, found it now. tks
04:24.18droothhey all, looking for an asterisk developer to help me expand my IVR system.  please chat if interested
04:24.19*** join/#asterisk kusznir (n=kusznir@pool-68-238-130-44.sea.dsl-w.verizon.net)
04:24.31DarthCluedrooth: what kind of help?
04:24.53SwKhat:  that all depends on what modules you got with it
04:25.41hatSwk, how to get modules for this card?
04:25.49SwKorder them from digium
04:25.55SwKor any of the other digium resellers
04:26.06*** join/#asterisk charles___ (n=charles@fw.invosat.com)
04:26.23hat<PROTECTED>
04:26.23hat<PROTECTED>
04:26.25*** join/#asterisk caroca (n=caroca@conm200-116-120-194.epm.net.co)
04:26.50hatdo you mean i cannot configure the modules myself?
04:27.14SwKhat you can donfigure the modules yourself, but there are physical difference between FXO and FXS modules
04:27.53*** part/#asterisk caroca (n=caroca@conm200-116-120-194.epm.net.co)
04:27.54hati see. so i need to order TDM10B, TDM11B,TDM12B,TDM13B etc from distributor. right?
04:27.55*** join/#asterisk Legend (n=legend@24.244.142.133)
04:29.21*** join/#asterisk diegows (n=diegows@201.250.105.67)
04:30.12Grubshat - correct..   There are 4 slots onto which the modules plug as daughter cards.  You buy the modules that you need.
04:30.17kusznirHi all:  Does anyone know an astrisk-friendly service provider that provides basic plans (DID + incomming only, outgoing per min)?  Preferably IAX.
04:30.46GrubsWhat country? ;)
04:30.47hatGrubs, thanks very much for clean my unknown
04:30.56kusznirI'm looking for someone to interconnect with for playing and learning with asterisk.  Unfortunately, due to tight budget, any recurring expense has to have "real utility" and thus will also be a phone service.
04:31.07kusznirUSA (Washington State, 509 area code)
04:33.16Grubs* blush*
04:33.24kusznir:)
04:35.41gekkusznir: anyone but sixtel
04:35.44*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
04:35.44SwKkusznir: check out asterlink they do sip and iax and are pay as you go they dont really have DIDs but they have cheap reliable 800 service
04:35.46Vcoheh..heh..
04:36.41Vcotry les.net if you want decent customer service
04:37.48SwKasterlink has good customer service...
04:37.57SwKthey all hang out here and are quick to respond
04:38.19DarthClueasterlink employees are overly obsessed with perfection or something close to it.
04:38.43SwKnow discliam yourself darth
04:39.45DarthCluei am not being paid to promote nor endorse any service or provider.  and i never really was, it has always just been my honest opinion
04:41.55*** join/#asterisk _DAW (n=_DAW@ip68-229-153-182.lf.br.cox.net)
04:42.11DarthClueGot SIP response 481 "Call Leg Does Not Exist" back from
04:42.16DarthCluefile, what did you break?
04:42.27file[laptop]it's chan_sip behavior
04:43.04file[laptop]chan_sip has become slightly incompatible with itself
04:44.34GrubsDoes anyone know if IBM netinfinty rackmount servers were designed for narrow racks (less than 19")?   I am looking at a NetFinity 4500R for an * box  but the specs say 41.5cm wide which seems too narrow.
04:45.35*** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net)
04:46.38kusznirHmm...maby I'm missing something, but asterlink doesn't appear to have any "small" services, and they don't seem to have any rates posted on their web site.  Did I hit the right site?
04:46.43*** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim)
04:47.11SwKasterlink.com
04:47.13file[laptop]kusznir: 2 cents per minute for inbound and outbound, billed in 6 second increments
04:47.14SwKits .02/minut
04:47.18DarthCluekusznir: 2 cents a minute with a toll free did, that is what asterlink provides as the basics
04:47.35file[laptop]I need to modify the site... ugh
04:47.53file[laptop]wow, asterlink support is getting spam
04:47.55SwKits prepay... throw $10 in an account and it'll last you a while unless you just wear it the hell out
04:48.09file[laptop]I can give you a credit too to try it out
04:48.27kusznirCool...I'd like that.
04:48.57*** join/#asterisk aYCa\ (i=amor_F_@server.ivinskis.kursenai.lm.lt)
04:48.57kusznirI take it I go ahead and hit the "sign up" link to set up an account?
04:49.11file[laptop]yes
04:49.15file[laptop]just go, use your head
04:49.29*** join/#asterisk deLTa (i=user226@server.ivinskis.kursenai.lm.lt)
04:50.36kusznirahh..I was a bit afraid because the web site and plan descriptions all looked like they were for corporate or business customers (espicially when the stie started talking about terminating T1s at their facility and routing them over voIP to you, etc....I wanted to make sure I was signing up for the "right" service)
04:51.36kusznirSo, there's an inbound rate of $0.02/min.  There's no mention of DID, and someone else here said they don't really do DIDs.  So how do I get incomming minutes?
04:51.43*** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it)
04:51.45kusznirerr..incomming minute charges/usage
04:51.52DarthCluekusznir: it's toll free DID at the same rate
04:52.18DarthCluewhen you sign up, you get a tollfree number 8XX-XXX-XXXX that is your incoming number
04:52.48kusznirCool..no monthly charge for that?
04:53.06*** join/#asterisk akrall_ (n=akrall@customer-201-133-125-77.prod-infinitum.com.mx)
04:53.07SwKjust usages
04:53.10DarthCluefile[laptop] can verify, but not that i know of
04:53.23kusznirWow...That sounds almost too good to be true!!
04:53.38file[laptop]no monthly charge for your first number
04:53.43akrall_Guys.. anybody has issues compiling chan_unicall? Im gettings this error
04:53.44akrall_chan_unicall.c:36: parse error before string constant
04:53.44akrall_In file included from /usr/include/sched.h:32,
04:53.44akrall_<PROTECTED>
04:53.44akrall_<PROTECTED>
04:53.44akrall_<PROTECTED>
04:53.46SwKyeah I think they do have a charge for custom/vantiy toll frees (right file)
04:53.47file[laptop]and you get to pester me on the phone or via e-mail if you have problems!
04:53.58file[laptop]vanity numbers are a one time $25, portings are the same
04:54.02akrall_and then c file has this on the code: ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
04:54.03DarthCluefile[laptop]: what's the monthly on the second number?
04:54.14file[laptop]DarthClue: depends how many you go for
04:54.18file[laptop]the more the less
04:56.04Vcous/canada termination only i'm guessing?
04:56.12file[laptop]correct
04:56.21file[laptop]international is too high a potential for fraud
04:59.48Grubsfile[laptop] - so international's can get the US DID? - which part is the fraud risk?
05:00.09file[laptop]calling internationally
05:01.10kusznirOk, question.  What is the web site reffering to when they say "Enter the number you would like to route your tollfree number below."  Is it intending to set up forwarding from the toll-free number to some existing non-toll-free number?
05:01.23GrubsOK - better go read some more.  I thought the US DID was seding the VoIP call over the internet.
05:01.24file[laptop]ignore that
05:01.30file[laptop]the platform is system-wide
05:01.50file[laptop]okay, I'm losing it, stuff like that on the platform is system-wide
05:01.57file[laptop]so for other stuff we use that, but for Asterlink just leave it blank
05:02.45*** join/#asterisk TheCops (n=mdb@206.248.136.146)
05:03.00kusznirok
05:03.35TheCopsHi, how can I use PoE support with snom 320 ? can I buy any switch or hub that's support 802.3af
05:03.36TheCops?
05:04.05jskcrtigerdirext has poe switches cheap
05:04.24GrubsSo asterlink is for voip call termination not for DID?  I'm confusing myself
05:04.29jskcrerr tigerdirect
05:05.07TheCopsjskcr, ok, but I can buy any switch or hub that support 802.3af ?
05:05.09file[laptop]Asterlink provides outbound calling to the US and Canada
05:05.18file[laptop]we also provide a toll-free number that people can reach you on, delivered by VoIP
05:05.24jskcryup
05:05.28file[laptop]or sent to another phone number if you're using Asterlink Extreme and have that setup
05:05.34jskcrhow many snom's do you have
05:06.00GrubsI see now.   I'm in AU - was looking for another DID other than Staphone
05:06.11TheCopsjsandnes, 8
05:06.12GrubsStanafone
05:06.35*** join/#asterisk tinsan (n=tinsan@202.171.49.33)
05:06.54Grubss/f/ph
05:06.58jskcrtigerdirect has a 12 port poe / 12 port regular switch for like 450
05:07.09VcoGrubs..you need a AU DID?
05:07.45jskcrmaximum length of cable is 100m and the delivered power is 12.95W
05:08.10Grubsno.. I want a US DID to cannel into my AU asterisk box.
05:08.14TheCopsjskcr, ouch, expensive
05:08.19Grubs(cant type today)
05:08.20*** join/#asterisk LeoB (n=chatzill@pool-70-20-20-158.bstnma.fios.verizon.net)
05:08.23TheCopsI'll order if from my dealer hehe
05:08.49Grubse.g. like IPKall
05:09.16jskcrTheCops:  thats one of the cheaper ones
05:09.27Vcoahh..was going to say didx.org had AU did's for $5/mo and no incoming charges.
05:09.29jskcrTheCops:  24 port poe switches run 1000
05:09.54Grubsoh...  I will look into that however also :)
05:10.11Vco**nod**..it's an extra number to have ;)
05:10.32jskcrhaving lots of poe injectors can make a mess
05:10.46TheCopsjskcr, I have good dealer.. :p
05:10.55GrubsActually I am tossing my home fixed line completely so I need a new DID.
05:11.00LeoBhello there, how to join 2 strings without including the ""?  I'm having problems with the following expression: Set(text=${text}${new-digit}) in my dialplan... can anybody help?
05:11.13Grubs..but I also have a business PSTN coming into the house for backup
05:11.17*** join/#asterisk tinsan (n=root@202.171.49.33)
05:11.35TheCopsA good VoIP ISP 1-800 number for Canada, someone know ?
05:12.09tinsananyone know roughly how much is a WILDCARD TDM400P?
05:12.16Vco1-800-upy-ours ?
05:12.19blitzragetinsan: www.digium.com
05:12.34TheCopstinsan, 4port FXO ?
05:12.38TheCopstinsan, this is that card ?
05:12.44jskcrtinsan: around 320 for 4 fxo's
05:12.50tinsanTheCops: yes...
05:12.55TheCopstinsan, 299 from netxusa
05:12.58TheCopsUS
05:12.58tinsanjskcr: 320USD?
05:13.01file[laptop]blitzrage: what have you been up to?
05:13.05jskcryup
05:13.05blitzrageGroove Salad *thumbs up*
05:13.14blitzragefile[laptop]: went out for all u can eat sushi
05:13.20file[laptop]yum
05:13.24blitzrageyah, pretty good
05:13.37blitzragehave to go to bed soon so that I can get up a reasonable hour and program my face off tomorrow
05:13.49file[laptop]that's silly, tomorrow's a holiday
05:13.52blitzragethe second "tomorrow" was redundent :)
05:14.12blitzragefile[laptop]: holidays are dead to me now (being self employed and all :))
05:14.27file[laptop]pfft
05:15.06blitzrageneed to have a lot done by tuesday morning
05:15.21tinsanTheCops: the price is it excluding the daughter card of X100M,S100M?
05:15.54blitzragetinsan: can't you just look online, or google for the answer?
05:16.13blitzrageseems to me it'd be quicker, but what do I know
05:17.06TheCopstinsan, sorry, the price I gave you was for TDM04B, Mainboard TDM400P + 4 FXO module
05:18.19X-RobBah
05:18.21X-RobBloody work.
05:18.28X-RobDAMN CUSTOMERS WANTING ME TO DO STUFF FOR 'EM.
05:18.33Vcobastards
05:18.38X-RobBugger'em. Millenium hand'n'shrimp.
05:18.58*** part/#asterisk akrall_ (n=akrall@customer-201-133-125-77.prod-infinitum.com.mx)
05:19.11*** join/#asterisk Gronker__ (n=Gronker2@70.152.170.91)
05:19.15*** part/#asterisk Gronker__ (n=Gronker2@70.152.170.91)
05:19.19X-RobYou'll all be happy to know I've set up an IRC client on AMP (aka, A@H) that sends people to #amportal, not #asterisk.
05:19.52blitzragethank goodness :)
05:19.53tinsanTheCops:Thanks
05:20.15TheCopstinsan, do you have a compagny ?
05:20.26X-RobI'll beg and grovel and see if I can get it into 1.10.009 (due for release Real-soon-now)
05:20.30TheCopsnetxusa accept incorporated compagny as reseller
05:20.40TheCopsI pay the VoIP stuff very cheap
05:20.53X-RobI not pay the VoIP stuff at all.
05:20.58TheCopslol
05:21.01tinsanTheCops: I don't have ...
05:21.23X-Robtinsan - I dunno what it's like in the US, but you're an idiot if you don't have a company in Australia.
05:21.26TheCopsX-Rob, what are you waiting, do you want to ship me some phone ?
05:21.35X-RobCompany tax rates = 30%.
05:21.41X-RobPersonal Tax rates = 48%
05:22.05TheCopsUS I guess will be very expensive to get a compagny
05:22.11Vcoin canada just add an extra 10% or so
05:22.12TheCopstaxes in canada are around 40%
05:22.20TheCopsVco right
05:22.30Vcoit's fucking retarded
05:22.32TheCopslol
05:22.46TheCopsVco, you live in canada ?
05:22.47X-RobThey keep muttering about a flat 25% tax rate for everything
05:22.52Vcoyea..
05:22.52X-Robbut that'll never happen.
05:23.01*** join/#asterisk Pete_Largo (n=Pete_Lar@225-196.35-65.tampabay.res.rr.com)
05:23.07TheCopsVco, What's your compagny ?
05:23.30Vcodon't have one.....just wandering consulant...
05:23.57Vcodoing a boring Windows contract by day, and trying to gert more of this rolling in the evenings...
05:24.01X-RobVco - I used to be like that.
05:24.03TheCopslol
05:24.04TheCopssame here
05:24.06X-RobThen I married an accountant.
05:24.15X-RobI pay fuck all tax these days 8)
05:24.17TheCopsbut with linux/networking/telphony consulting
05:24.51X-RobTheCops/Vco - there's not much money in it now, but give it another 48 months
05:25.07X-Robyou'll be raking it in while everyone else is just starting to realise that this voip stuff is a good idea.
05:25.20FaithfulHey what do you think 50% of the savings over the 1st 3 months as a consultant fee?
05:25.23TheCopsI want to sell asterisk solutions :)
05:25.38TheCopsFaithful, flat rate own!
05:25.47X-RobFaithful - depends. A good one is 'We'll install it for nothing, but keep paying your standard phone bill'
05:25.59X-Roband then after x amount of time, they start paying the new one.
05:26.02FaithfulX-Rob: I wish
05:26.25X-RobFaithful - don't have the cash?
05:26.41Faithfulno, they won't buy it.
05:26.51X-RobYou're looking at the wrong market then.
05:27.08X-RobGo look around all the small businesses in your area.
05:27.13X-RobThat's where the market is
05:27.21TheCopsyeah
05:27.22X-Robthey might not even have a paxb
05:27.25X-Robpabx
05:27.28TheCopsa lot of money :)
05:27.38luke-jr__is VoIP stuff usually paid per job or per hour? o.o
05:27.54FaithfulThese guys do nearly $2K in landline telephony a month.
05:28.09X-Robthree phones, a cheap VIA Mini-ITX server, an ADSL line, 2 days of your time.
05:28.45DarthCluejust $2K?  that's not alot.  would be perfect for an * box.
05:28.48TheCopsluke-jr__, my rate are based on how many extension/voicemail/IVR/IVR billingual and stuff like that
05:28.49X-RobYou can either charge 'em up front, or say 'This costs $2000. You keep paying your existing phone bill until it's paid off'
05:28.54FaithfulSo on top of the cost of equipment I was looking for $3K
05:29.53FaithfulSo basically I can say to them... Including my charge it will pay it'self off in 6-9 months
05:30.15FaithfulThat's how I was thinking.
05:30.31X-Robjust because the three letters its are together does not mean it needs an apostrophie.
05:30.33harryvvthecops...ummm thats not alot. The typical pbx per seat cost for a commercial system is between 1,000-2,000 per phone so a 100 phone system can cost anywhere upwards of 100 grand or higher.
05:30.36FaithfulX-Rob: yeah... so did I after I typed it
05:30.40X-Rob8)
05:31.22harryvvthat includes support for a period of time and a share is the cost of the pbx.
05:31.45X-Robyou can also offer to take their existing system off their hands
05:31.51X-Robsell it on ebay, you'll get some money for it there.
05:32.07TheCopsX-Rob good idea
05:32.11harryvvx-rob thats probebly true
05:32.13TheCopsvery good idea...
05:32.27Ashthe 'take your existing system away' is what a lot of telco vendors do
05:32.28Ashit's fun stuff
05:32.39TheCopsX-Rob I have a potential client that will probably accept with your idea, thanks
05:32.40TheCopslol
05:32.51Vcotake it away with no agreement to replace it with anything..
05:33.06Vcojust for shits'n'giggles
05:34.54FaithfulThis company has a relatively new pbx & phones with no VoIP so * is in addition.
05:35.22harryvvWhat did thay pay for it?
05:35.52Faithfullooking at it probably $10k with phones
05:36.01harryvvhow many phones?
05:36.25Faithful15 maybe
05:36.35harryvvThats a little low
05:36.38harryvvWhat city?
05:36.40X-RobFaithful - I really wouldn't be trying to sell to someone with a new phone system
05:36.44harryvvwhat pbx
05:36.56harryvvyea
05:37.01X-Robyou want to find someone who's starting to think about getting a new system and go 'Ooh, you can do _this_'
05:37.02*** join/#asterisk KaBewM (n=DA-MAN@24-180-28-208.dhcp.psdn.ca.charter.com)
05:37.22X-RobI _always_ carry a Snom 360 around with me
05:37.25hathi, how about the price of Sangoma card and digium card? which is cheaper?
05:37.35harryvvyea...alot of people already spent there money on a expensive system and would be less inclined to give it up.
05:37.36Faithfuldunno what phone system... It's not REAL new say 5 years old...
05:37.38X-Robcoz they look hr0n, and you can plug 'em an and show 'em key-system features
05:38.16harryvvbutifull thing about asterisk is you can take the phone on the road or say..at a home office.
05:38.26Faithfulbut they will save heaps with VoIP and get nice TAPI features and least cost call routing
05:38.32harryvvcannot do that with a conventional pbx.
05:38.56harryvvFaithfull mabey.
05:39.19Faithfulharryvv: maybe what?
05:40.46FaithfulTelephony in Australia is much dearer than the US I believe
05:42.59harryvvdearer
05:42.59harryvv?
05:45.19*** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3985149.sympatico.ca)
05:46.17*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
05:46.38kusznirOk, I've got some quesions about low-end VoIP phones.  I'm kinda courious what functionality they include (becides the obvious able to place and recieve VoIP calls).  For example, message waiting indicator, programmable feature buttons, etc.
05:46.56jskcrkusznir define low end
05:47.21jskcraround 100 bucks
05:47.35*** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com)
05:47.41kusznirI've seen some of the functionallity of the (very expensive) cisco phones, but am curious what the Grandstream Budgetone 101 or the Sipura SPA-841 provide.
05:47.51kusznirYea, $100 and under.
05:48.20jskcrgxp-2000 are around 99 and have multi line/mwi/speakerphone/handset adapter/speed dial buttons
05:48.21harryvvpolycom 501 is very nice.
05:48.34Moc_harryvv, they are
05:48.54DarthCluekusznir: polycom 300/301 would be my recommendation for a low-end phone, the grandstream line(s) are a bit flaky
05:48.55Moc_ip 500 and ip 600 are the best phone I found so far..
05:49.03kusznirI'm really courious about the Budgetone 101, as its $59, too.
05:49.13Moc_do not have ip300 yet
05:49.15kusznirahh..ok.
05:49.16jskcrthe bugetone 101's are fine now the firmware has matured
05:49.19harryvvI expecially like the 500's speakerphone.
05:49.30DarthCluekusznir: flaky, even with the recent firmware, flaky
05:49.35*** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) [NETSPLIT VICTIM]
05:51.05jskcrnew 501 are like 150+ bucks
05:51.06kusznirjskcr: who makes the gxp-2000
05:51.12jskcrgrandstream
05:51.41jskcrthe newest firmware is a flaky but if you run a cron job that reboots it at midnight it will work ok
05:51.56DarthCluejskcr: that's why i recommend the low end ip301, it's about 130 shipped i think
05:52.14jskcrhttp://www.grandstream.com/user_manuals/GXP2000.pdf
05:53.11jskcragain new 301's are 130
05:53.38jskcrand they have there problems too :)
05:54.14Pete_LargoI have had pretty good luck with my GXP-2000, I like it
05:54.39jskcrPete_Largo:  It supports poe too :)
05:55.08jskcrPete_Largo:  what firmware are you running
05:55.28Pete_LargoNot sure, I have it set to update every 7 days... so I gues the latest.
05:55.46hathi, what is span? is it just trunk?
05:56.17jskcrfor 100 bucks for a multiline phone its hard to beat
05:56.31kusznirI take it phones with programable buttons like the gxp-2000 are programed via the asterisk config?
05:57.03harryvvkus, some times you have to program the phone
05:57.11kusznirIn the SIP config file, or are they configed directly on the phone via web?
05:57.12kusznirok.
05:57.20jskcrkusznir:  not yet they are configured via the web interface IE: msg button /speed dial
05:57.46jskcrhas a blinking red light in the corner that also works with mwi
06:00.26jskcra grandstream is basicly TI DSPs/ realtec ethernet / issi flash
06:01.04jskcrI know I have taken them apart and carefully removed the epoxy they put on the dsp :P
06:01.14gordonjcpmorning all
06:01.23*** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net)
06:01.33gordonjcpjskcr: are the grandstreams any good?
06:01.48jskcrThey work ok.
06:01.49*** join/#asterisk Newbie___ (i=me@211.24.146.14)
06:01.58jskcrThe firmware could be better and it will get better
06:02.17gordonjcpI haven't had a play with one yet
06:02.19Newbie___hi all, any one familiar with ANI callback + prepaid billing on asterisk ?
06:02.59jskcrgordonjcp:  I have tested about 15-20 different ata adapters and sip phones
06:03.14jskcr5 of em where grandstream
06:03.16kusznirOk, another question related to phone hardware:  reinvite.  If I understand it properly, without reinvite, the phone only talks to the asterisk server.  Therefore, all IP call traffic runs through the asterisk server.  (here's where I get questionable):  If the phone does support reinvite, then the actual IP stream doesn't necessarily go through the asterisk server; it would be possible for it to connect directly to the other
06:03.16kusznir<PROTECTED>
06:03.26DrRighteoushey all.. trying to run asterisk under a screen session: screen -mS ivr /usr/sbin/asterisk -vvvc, but it instantly terminates.. however when I insert a strace on the command line, it runs fine... can anyone assist?
06:03.43*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
06:03.46gordonjcpjskcr: have you got a list of your results from that?
06:03.47DrRighteousmaybe something with the stdout?
06:03.50jskcrDrRighteous:  are you running it as root
06:03.59DrRighteousjskcr: yes
06:04.12jskcrgordonjcp:  gordonjcp cant give it out sorry :( it was for work
06:04.16gordonjcpjskcr: ah, np
06:04.42jskcrgordonjcp:  I can tell ya for the price  the gxp-2000 is one of the bests
06:04.50kusznirjskcr: As far as lowest-cost but functional IP phone, you would recommend the gxp-2000 then?
06:04.55jskcryes
06:05.04jskcrthen polycom
06:05.12DrRighteoustake a look at the polycom 300's too
06:05.13jskcrthen cisco
06:05.16jskcrthen uniden
06:05.22DrRighteouspretty cheap, but very nice
06:05.37DrRighteousplus same OS as the higher end models.
06:05.50gordonjcpjskcr: I've found Avaya 4602s with SIP firmware to be pretty good
06:06.39DrRighteousthis one is for 114USD, but have seen them as low at 90... http://www.tritechcoa.com/product/126024.html
06:07.01gordonjcpjskcr: I'm planning on getting some "engineering samples" from a couple of Chinese manufacturers
06:07.02jskcras far as wifi ones go the f1000 utstarcom is a nice one for 169 bucks
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06:07.31jskcrdoes it support poe
06:07.49gordonjcpwhat, the 4602?
06:07.54gordonjcpyes, it needs 48v PoE
06:08.12gordonjcpI just made up an adaptor, which sits on top of my hub
06:10.26*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
06:10.47jskcras I was saying about the 300's I am not a big fan of MGCP
06:11.23DrRighteousoh,...http://www.sjtelecommunications.com/pol-2200-11330-001.html
06:11.33DrRighteousjust hunt around... you'll find them in SIP and cheap
06:11.49DrRighteousjskcr: any ideas on my screen issue
06:12.15jskcra asterisk -vvvvvvvvvvvvvvvvc does not spit out any errors
06:13.19DrRighteousworks great without the screen
06:13.32DrRighteousbut the moment I add screen, it just shutsdown
06:14.56DrRighteousbtw I can start asterisk in screen, just not automaticly on the screen command line
06:15.17jskcrhmm weird
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06:17.00ManxPowerCan anyone here recommend web and e-mail hosting service that supports IMAP?
06:17.05kusznirSo was my understanding of reinvite correct?
06:17.23DrRighteousManxPower: I can give you a box if you like...
06:17.25*** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com)
06:17.31shido6beat me to it
06:17.38DrRighteousotherwise OpenSRS has a very nice outsourced one for multiple
06:18.15ManxPowerDrRighteous, I have a box, I want to outsource it all
06:18.36*** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com)
06:18.40jskcrkusznir:  yes thats what a reinvite does
06:18.49DrRighteousManxPower: I mean do you need just IMAP for yourself or for multple boxes?
06:18.56Moc_damn google are cool... puting katrina Satelite picture
06:19.07Moc_you can see before and after... anyway hehe
06:19.08DrRighteousHey Moc... long time
06:19.28kusznirI haven't noticed any mention of reinvite support on any phone spec sheet...how does one know if its supported (or do most/all phones support it already?)
06:19.30ManxPowerDrRighteous, For myself and about 10 friends, but if I'm happy with the service I might recommend it to customers.
06:19.33*** join/#asterisk asteriskDOTbz (n=logger@pbxtech.com)
06:19.46Moc_DrRighteous, damn yea.. how things goes ?
06:19.50ManxPowergodaddy.com would be perfect, except they only support POP3
06:19.55jskcrkusznir: its part of the sip spec
06:19.57asteriskDOTbz<PROTECTED>
06:20.01ManxPoweri.e. I'm looking for a hosting provider
06:20.12shido6hosting provider for?
06:20.16DrRighteousManxPower: well look at opensrs.org, its about 20 cents a box, with antivirus and spam filtering...
06:20.20shido6just shoot me a mail man...
06:20.34kusznirjskcr: ok.  I thought I read that some sip devices can't deal with it, though...is that the case?   I mean, can you safely enable it by default?
06:21.02jskcrkusznir:  if they dont then they are not really sip devices :P
06:21.03ManxPowerDrRighteous, Um, I don't want to deal with a box.  If I wanted to deal with a box I would not be trying to outsource all of it.
06:21.51DrRighteousMoc: things were going slow... but the company is about to have major life breathed back into it... two new datacenters, multi-gige's with level3, and over 15 DS3 tdm connections!!! yeah! no slepe for me!
06:22.02ManxPowerSo the answer is "no".
06:22.06DrRighteousManxPower... its not a box, its outsourced email..
06:22.26ManxPowerDrRighteous, They don't seem to do retail, only wholesale.
06:23.17DrRighteousahh then: http://www.tuffmail.com/
06:24.45DrRighteousManxPower: is that what you needed?
06:24.51ManxPowerDrRighteous, Much closer, yes.
06:25.40ManxPowerMy requirements are complicated enough that I may have to write up an FRP.
06:25.46ManxPowerRFP, even.
06:26.15DarthClueManxPower: write one up and pass it along to me, i might be able to find you something.
06:27.06ManxPowerDarthClue, The requirements for my personal domain, my business domain, and customer domains would be pretty much the same.
06:27.50*** join/#asterisk xpasha (n=pavel@217.30.252.68)
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06:30.36DarthClueManxPower: send me some details on what you are looking for and i'll see if i have access to anything that meets your needs.
06:31.38ManxPowerDarthClue, I'll write up a description of what I have now on my box, which is what I'll require if I outsource it.
06:32.12DarthCluethat'll work.
06:39.05LeoBHELP NEEDED: Problem with Asterisk command Read ...  If I dial "0", asterisk prints "User entered nothing."  If I dial "OO", asterisk recognizes both digits...  Does anyone know what is going on?
06:40.36*** join/#asterisk eivindtr (n=eivindtr@062016241059.customer.alfanett.no)
06:45.07jskcrdo you have a wait before the read
06:45.19LeoBno
06:45.28jskcrtry putting one in
06:45.56LeoBbut "0" is the only case that fails...
06:46.24jskcrthen check you extensions to see if something else is using 0
06:46.57*** join/#asterisk kusznir (n=kusznir@pool-68-238-130-44.sea.dsl-w.verizon.net)
06:46.58LeoBnope
06:47.56LeoBjskcr, does Read() with "0" work for you?
06:48.01*** join/#asterisk dexteruk (n=dexteruk@217.165.98.22)
06:48.05jskcryup
06:48.24LeoBweird
06:48.41*** join/#asterisk razu (n=razu@fw.voicenet.ee)
06:49.22jskcrwhy are you doing a read, why not just create a 0 extension
06:50.03LeoBI'm trying to use the phone keyboard as text input...
06:50.22LeoB... in a macro
06:52.03*** join/#asterisk X2 (n=X2@NAT-home-clients-99.lgnet.ro)
06:52.26X2hi, guys
06:53.18jskcrhy
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06:57.53fugitivohello
07:01.38jskcrhya
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07:16.59krisguycan anyone tell me if there is a Cisco 7910 that can do SIP or H.323?  I'm reading conflicting info online
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07:51.56MGSsanchois asterisk multi threaded?
07:52.26MGSsanchoor would HT, dualcore, multiproc do anything to asterisk?
07:54.01DybdahlAs far as I know, music playing is done using mpg321, which runs in a separate thread
07:54.06Moc_MGSsancho it is, but not very well made..
07:54.10DybdahlThis means that mp3 decoding etc. is definitely separate
07:54.27DybdahlMoc_, can you describe how it is?
07:54.52MGSsanchodoes it branch off to a new thread for every call?
07:55.00DybdahlI guess multicore CPUs would mostly benefit CPU intensive operations, like decoding and encoding
07:55.11MGSsanchoa separate thread for each line would be cool
07:55.25MGSsanchoor things that utilize it
07:55.33X-Robasterisk is _very_ threaded
07:56.06X-Robthe more CPU's you throw at it the better
07:56.11Moc_X-Rob,
07:56.14MGSsanchoyay cool
07:56.17Moc_X-Rob, * is locking all the time ..
07:56.32X-RobMoc_, 'locking' how?
07:56.40X-RobThere's something in mantis about that
07:56.42DybdahlIt does not create new processes when you call it
07:56.45*** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net)
07:56.50DybdahlI guess it uses nptl?
07:57.03Moc_X-Rob, nothing on mantis will fix it in it current state...
07:57.11Moc_masive * redesign need to be done...
07:57.17Moc_the core wasn't well maded.
07:57.20DybdahlMoc_, what kind of operations creates locks?
07:57.28DybdahlIsn't it just switching etc.?
07:57.30Moc_Dybdahl, about everything..
07:57.37Dybdahllike encoding/decoding?
07:57.44Dybdahlrecording a message?
07:57.46DrRighteousCan anyone tell me how to play MOH while ringing an extension (aka Dial() )... I'm Answering first in the exten...
07:58.11Moc_Dybdahl, DNS query, reload, cli cmd..
07:58.15X-RobDrRighteous - type 'show application dial'
07:58.27DybdahlMoc_, ok, thanks
07:58.46Moc_it just a start
07:58.49DrRighteousX-Rob: Thanks
07:58.59X-Rob<PROTECTED>
07:58.59X-Rob<PROTECTED>
07:59.15DrRighteous
08:01.13fugitivoDrRighteous: for example Dial(SIP/1234,30,m)
08:01.30DrRighteouswhats a good asterisk sound prompt to play before transfering a call?
08:02.53DrRighteoustransfer.gsm?
08:03.18fugitivowhen do you want it to be played?
08:03.28fugitivoand who will listen the audio?
08:04.24DrRighteousactually looking for someone to setup some queues on my system... simple stuff.. anyone around interested? paypal is good :) hehe
08:04.40DrRighteousfg
08:06.21JabroniX-Rob on which version does that parameter of the Dial() is introduced??
08:06.23Jabroni1.0.8 ?
08:06.32Jabronii cant see it on 1.0.7
08:06.47fugitivoit works on 1.0.7
08:07.38Jabroni'M(x) -- Executes the macro (x) upon connect of the call
08:07.38*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
08:07.56fugitivoit's lowercase
08:07.57*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
08:08.37Jabroniguess its an undocumented parameters
08:08.50Jabroniat least it doesnt appear in show application dial
08:09.19fugitivoi don't remember, but it works on 1.0.7, i was using it
08:11.25fugitivoshit, it's 5:10am here
08:11.35*** join/#asterisk konrads (n=konrads@out.ctkom.lv)
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08:17.02h3xcanada?
08:17.09h3xheh
08:18.11*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
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08:19.26opus_Could you describe exactly what failure you're seeing in processing of the
08:19.26opus_SIP call?  Forget technical failure, because setting nat=yes causes
08:19.30opus_Asterisk to technically violate the SIP spec.  This is by design.
08:19.59opus_i'm sorry but whoever is working bug reporting isn't cutting it
08:22.50gordonjcpis there a description of exactly what nat=yes does?
08:23.10*** join/#asterisk johnm (n=johnm@gentoo/developer/johnm)
08:23.14gordonjcpis it basically "don't necessarily believe the IP addresses you see" ?
08:23.48johnmdoes anyone happent o have/know of a good echo-test number in america? (I am getting bad echo to america, and I dont particular want to ring somewhere/one who I dont know to test it :)
08:24.13gordonjcphm
08:24.26dudesI could setup a number for you to echo to
08:24.46gordonjcpI could give you a stanaphone number I have set up, but I can't guarantee that's not echoing
08:25.02johnmit's actually the digium support number i'm getting terrible echo from (although I'm in the UK so it's likely any american number... this card isn't properly tuned and I'm having a pretty awkward time with it)
08:25.14johnmanything is good for now :D
08:25.16johnm(thanks)
08:25.31gordonjcp2 secs
08:25.43dudesI'll setup a number too
08:25.51dudesgive you two places to play with
08:25.56johnmalso, does anyone know of a good (accurate and computed) method of setting gain?
08:26.17johnmalthough, on my situation I have a problem with doing that anyways. I need to alter gain on a per-span basis, which im fairly sure it can't do.
08:26.18gordonjcpwhat do you want the number to do?
08:26.28johnmechotest, echodone is fine.
08:26.45johnmI realise it will echo back, but if I'm getting genuine echo it will just sound a million times worse :)
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08:31.21asteriskDOTbz<PROTECTED>
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08:34.56AkelavlkHello, how is possible invite user into conversation, but he should just listen..
08:37.58gordonjcpjohnm: pm
08:38.12opus_hmmm
08:41.03*** join/#asterisk FITA1 (n=m_ahmed@202.5.145.50)
08:41.06FITA1hi
08:42.11FITA1I m using agi and trying to print message on the console using fprintf(stderr,"") but the message is not appearing on the * console .... any help
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08:44.34FITA1I m using agi and trying to print message on the console using fprintf(stderr,"") but the message is not appearing on the * console .... any help
08:45.47*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
08:45.48RoyK"" is a null string :P
08:45.57*** join/#asterisk clive- (n=pirch@rndf-146-13-10.telkomadsl.co.za)
08:46.46RoyKFITA1: i thought stderr was intercepted by asterisk as commands/agi interface......
08:48.18FITA1should I use somthing else to print on the console except stderr
08:49.25*** join/#asterisk Danett (n=cyrieldo@tbnb-165-195-35.telkomadsl.co.za)
08:49.27Danettheya.
08:49.53Danettdoes someone know why my connection is being killed after 5 minutes?
08:49.59Danettwhile using the DIAL command?
08:50.08DanettThere is no absoluteTimeout set
08:50.35*** join/#asterisk fulgas (n=fulgas@213.58.130.46)
08:50.42AkelavlkHas anybody idea, how is possible invite user into conversation, but he should just listen not speak.
08:51.17*** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net)
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08:51.44weazulgood morning europe!
08:52.46weazulhow can i check the users which are known by my local asterisk server?
08:54.53niZon<channel> show users
08:54.55*** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net)
08:54.59niZonin the CLI
08:55.07niZonfor example, sip show users
08:55.24niZonbtw this is a global channel, not just europe :)
08:56.20*** part/#asterisk Kumbang (n=unknown@167.205.24.5)
08:56.33*** join/#asterisk Kumbang (n=unknown@167.205.24.5)
08:57.36weazulok my humble excuses ;-)
08:58.14*** join/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
08:59.06*** part/#asterisk Poincare (n=jefffnod@dD5779BD2.access.telenet.be)
08:59.48*** join/#asterisk littleball (n=littleba@bb220-255-134-33.singnet.com.sg)
09:00.21littleballhi, what is span number when i configure the E1 line for zaptel.conf?
09:03.06AkelavlkHas anybody idea, how is possible invite user into conversation, but he should just listen not speak.
09:03.52*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
09:08.08*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
09:08.16*** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
09:08.23X-RobAkelavlk - meetme
09:11.19*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
09:13.18DanettDoes anyone know why my dial connection is killed after 5 minutes sharp?
09:15.39AkelavlkX-Rob but then I need invide every inbound call to room automaticaly..  How should I do that?
09:20.11AkelavlkWhat I need is ZapBarge functionality..
09:20.27*** join/#asterisk mutilator (n=animenod@65.111.201.79)
09:21.23*** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk)
09:21.26AkelavlkWhen user call to agent, agent should during speaking allow another agent listen that conversation.. And at least forward call to that agent.
09:25.25niZonsomone tell me why X-lite uses 30MB of ram to run
09:25.40*** join/#asterisk ogun (n=ogun@h236n2fls34o865.telia.com)
09:26.44queuetueniZon: You should see how much ram X-heavy needs...
09:26.58Grubsbahahahhahaa
09:28.04*** join/#asterisk inspired (i=mikael@213.197.167.61)
09:29.31niZonno kidding
09:29.50Grubs9.7MB here niZon
09:30.22X-RobAkelavlk - Hmm. show application monitor
09:30.37X-Robbut that just _saves_ the call
09:33.21AkelavlkX-Rob, I know that's just monitor. I need somethinkg like ZapBarge or ChanSpy.
09:34.19AkelavlkBut, Only second agent can call these commands. It's not possible dial next user and run ZapBarge on my own channel..
09:40.24DanettWhat is the best way to trap an invalid input within an IVR system?
09:40.25*** join/#asterisk paing (n=chatzill@202.188.115.100)
09:41.02*** join/#asterisk RoyK (n=roy@80.239.107.80)
09:43.09X-RobAkelavlk - your last line doesn't make sence. Please rephrase what you mean.
09:44.15*** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com)
09:45.08DelvarIv got a set of speakers, not very big though
09:45.28*** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-48-91.dialin.kamp-dsl.de)
09:46.23*** join/#asterisk voipguy (n=Robert@196.200.26.42)
09:46.58AkelavlkX-Rob. User is call to agent X. Agent X invite Agent Z just for listening.
09:47.33*** join/#asterisk asteriskph (n=yakumo@203.87.204.126)
09:47.50AkelavlkThat means that Agent X should dial to Agent Z and execute command ZapBarge, ChanSpy for Agent Z..
09:48.54voipguyanyone with suggestions for a good billing solution for *?
09:49.27*** part/#asterisk IgorG (n=gia@195.162.32.126)
09:49.29AkelavlkBTW, it's possible has two open calls for one Agent?
09:49.59niZonAkelavlk: as in a 3 way call?
09:50.16AkelavlkYes, like 3 way call..
09:50.35AkelavlkI am not sure if there is some functionaly in Asterisk for that.
09:50.49niZonperhaps you can login to two different agent accounts
09:51.01AkelavlkExcept meetme..
09:51.09*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
09:51.29niZonmeetme is fun
09:51.38AkelavlkniZon, No I mean, user is call to agent X and Agent X invite Agent Z just for listening.
09:51.39niZoni just wish there was a good manager for it
09:51.48niZonoh
09:52.04niZonwell the only thing i can think of is meetme
09:52.06AkelavlkMeetme is not quite well function..
09:52.24niZontransfer the user to meetme, agent x calls in, agent z calls in
09:53.31AkelavlkI need to tell Agent Z that he should pick up a phone.. How should agent Z know that he should pick up a phone?
09:54.10razuhi
09:54.50razui have a little problem ... i've installed asterisk on fedora core 1 and i'm getting this kind of notice -> Sep  5 12:53:19 NOTICE[2459]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/vicitest/1 of format slin since our native format has changed to ulaw
09:54.55razuhow can i get rid of this ?
09:57.05*** join/#asterisk Mother_ (n=Mother@53.Red-217-126-93.pooles.rima-tde.net)
09:57.08razuall my sip phones and iax2 trunk has allow=ulaw parameter ... but this error doesn't disapear :S
09:57.45Delvardo you have disallow=all?
09:59.35razuyes
10:10.19X-Robrazu - don't use FC1.
10:10.23X-RobUse FC3 _at least_
10:10.32X-RobFC1 has significant bugs and kernel issues.
10:10.39razuhmm
10:10.42razuok
10:10.58X-RobThat may not be your problem, but still, it's a good thing to get one problem out of the way.
10:11.13*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
10:11.21razuso that may be explain why the same configuration works on slackware 10.1
10:11.43X-RobA 'recommended' distribution is CentOS 4.1
10:11.43razuX-Rob : thanks
10:15.02*** join/#asterisk CleanerX1 (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
10:18.38*** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk)
10:20.24queuetueDid FWD over IAX come back up, by any chance?
10:22.54*** join/#asterisk opti (n=nothing9@adsl-57-65.swiftdsl.com.au)
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10:24.42*** join/#asterisk znoG_ (n=gs@200.115.218.81)
10:34.44queuetueI have a grandstream 100 that flashes that annoying blue all the time...  What does the flash mean, and how can I make it never do that?
10:35.14Delvarthe flash is message waiting, voicemail
10:35.34queuetueDelvar: This account has no voicemail in it' sinbox.
10:36.00Delvaras far as i know thats the only time it flashes
10:36.07*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
10:36.44queuetueCan I disable the feature?  I can't imagine wanting it to do that when I have VM, anyway...
10:37.12queuetueDo I have to get out a screwdriver and break the led?
10:37.15sycoflyqueuetue: is there anyway of resetting the GrandStream 100 password ..
10:37.30queuetuesycofly: I'm sure there is, but I do not know it. :)
10:37.59sycoflyqueuetue: .. hehe .. dam what a way to start off with asterisk
10:38.11*** join/#asterisk jann (n=jan@matrix.loopback.org)
10:38.15sycoflyiv'e n00bed out on my dam password
10:39.10*** join/#asterisk dCc (n=cooLgirL@213.226.30.193)
10:39.23*** join/#asterisk jeh (n=jeh@ext122.almare.com)
10:39.33jehhi folks
10:40.44queuetuesycofly: Normally, find the reset and hold it for a while.
10:41.06jannI just trie to compile chan_capi-3.5 or chan_capi-4.0 on debian sarge. with woody it did work but now I get a lot of errors. any hints where to search ?
10:41.21Mother_google!
10:41.55jehi was wondering about a thing regarding parked calls. if i want to park a call on an assumed extension 700 i could do a redirect using the AMI action Redirect?
10:42.07Mother_if you google 'google', does it's servers go into a tight loop and crash?
10:42.08queuetuesycofly: http://www.broadvoice.com/support_install_byod_gsbgt.html - search for reset.
10:44.13sycoflyi found the reset queuetue .. hope it works ..
10:44.31queuetuesycofly: I just gave you instructions...
10:46.48*** join/#asterisk pycsusz (n=mrblack@pluto.euronetrt.hu)
10:46.56queuetueWhich phone support line "hinting" again?  (So you can tell which lines are in use.)
10:47.03queuetuephone(s)
10:50.56pycsuszHi Everybody! Somebody can answer to me, that can I log into more queues with one agent?
10:52.47*** join/#asterisk optim (n=nothing9@adsl-57-65.swiftdsl.com.au)
10:53.22*** join/#asterisk TheCops (n=mdb@206.248.136.146)
10:53.41AkelavlkHow can I make 3-way call? I read, Asterisk support this functionality, but how?
10:53.44*** join/#asterisk pauldy (n=pauldy@c-67-187-62-160.hsd1.tx.comcast.net)
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11:14.38*** join/#asterisk dennis (n=dennis@200.32.215.82)
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11:17.36*** join/#asterisk clynx (n=clynx@p5088C680.dip.t-dialin.net)
11:17.53clynxhey!
11:18.12*** join/#asterisk oob (n=oob@203-173-152-25.bliink.ihug.co.nz)
11:18.35clynxcould anyone tell me is these german AVM Fritz Cards are supported by Asterisk, or are they limited in any way?
11:19.53optimthrough the capi driver
11:20.01optimhave a look on voip-info.org
11:20.10optimthere is a setup for the fritz card
11:22.34MaksimHello.
11:22.53MaksimOpenBSD 3.7-stable http://pastebin.ca/22125 Any ideas?
11:23.34gaggamanclynx: fritz card should be working fair as long as you only use 1 card.
11:24.03clynxhmm .. sounds fine ...
11:24.09gaggamanfritz can't be in NT mode ("internal" bus)
11:24.33gaggamanand fritz can't be in p2p mode ("Anlagenanschluss").
11:25.38gaggamanI wasted a lot of time with the fritz's. Finally I bought some cheap no-name cards with the cologne-chipset.
11:25.49clynxhmm .. i'll honest to you .. I've currently no real big knownlegde about pbx's and all this ISDN stuff ... I've just found asterisk, and I really would make me happy if I could get it to work ;o)
11:25.53gaggamanand used chan_zap and zaphfc
11:26.15Danettgaggaman: why kind of card is it? Pri or Bri?
11:26.35clynxso you cannot really recommend fritz cards, right?
11:26.51clive-the eicon cards are goo
11:26.55clive-so are junghanns
11:27.48Danetti heard junghanns owned
11:28.04RoyKclive-: goo?
11:28.11RoyKboo?
11:28.15clive-good..:),,,with a typo
11:28.16RoyK~fart clive-
11:28.18jbotACTION farts in clive-'s general direction
11:28.56gaggamanDannett bri
11:29.10gaggamanclynx: no. not really
11:29.25DanettThe single E1 PCI ISDN brings powerful ISDN PRI connectivity to your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. The E1 PRI port can be configured for CPE or NET operation by jumpers. This port configuration is detected by the driver automatically.
11:29.28gaggamanI'd use the cologne chipset cards.
11:29.30Danettthis is a lttiel weird right
11:29.40Danettnevermind
11:29.47clynxok.
11:29.54gaggamanDanett yes the E1
11:30.08gaggamanthat's an active card. quite costly.
11:30.18DanettWhat's the cheapo card
11:30.20AkelavlkWhat softphone is best for AIX?
11:30.30gaggamanFritzCard PCI
11:30.45gaggamanThat's a passive card
11:30.50Danettwaht's the difference?
11:30.55Danettbetween active and passive
11:30.56RoyKDanett: but does it work with zaptel_
11:31.04RoyKs/_/?/
11:31.29Danettdunno
11:31.50gaggamanThe active cards have their own uP to handle isdn protocol stuff
11:32.12RoyKgaggaman: for asterisk?
11:32.26gaggamanthe passive don't. With passive cards, the whole ISDN protocol has to be done in software.
11:32.27Danettgaggaman: well. i want zaptel to do that
11:33.08DanettI need a cheap ass gsm gateway
11:33.13DanettThey are hard to find
11:33.20*** join/#asterisk jonathh (n=asd@host217-46-145-65.in-addr.btopenworld.com)
11:33.26gaggamanNo. zaptel does not handle the ISDN protocol. That is done either by the (active) card or by the driver (capi driver).
11:33.26Danettin the form of a normal pci card
11:33.36Danetti mean that ;)
11:34.09gaggamanGSM?
11:34.15Danettyes
11:34.29Danettgsm == cellphone
11:34.34gaggamanwe're talking about ISDN.
11:35.02puzzledmorning all
11:35.05Danetti know :)
11:35.09DanettI just changed the topic
11:35.29Danettit's the power within that drives us my san
11:35.37*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
11:35.38ogunWhat is the problem here: (always evaluates as true)
11:35.48ogunexten => 300*,2,GotoIf($[ "${CHANNEL:0:3} " = "ZAP"]?5:3)
11:36.08*** part/#asterisk jonathh (n=asd@host217-46-145-65.in-addr.btopenworld.com)
11:36.19*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
11:36.22Danettogun: because there is an if/else statement
11:36.29*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
11:36.34Danettso the result will always be !false
11:37.15sycoflyqueuetue: .. you beauty mate
11:37.30queuetueMarried.
11:38.02ogundanett: So to fix this I would?
11:38.39DanettYou say
11:38.52DanettIf channel equals zap then goto 5, if not then goto 3
11:39.43ogunWhich is what I want, but it always goes to 3
11:40.34Danettremove the traling space after 0:3} ?
11:40.42Danettwell
11:40.45Danettremove the quotes
11:40.49*** join/#asterisk Nix (n=Nix@81.214.255.57)
11:40.52Danettsince ${} is a variable
11:41.01Danettso it does not need enclosement by ""
11:42.09ogunRight, thanks. I'll give it a go
11:45.13*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
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11:47.32ogunMan, I need a head exam. :)
11:47.38ogunIt is Zap, not ZAP :)
11:52.03Danettoh ;)
11:52.03Danettsorry
11:54.39ogunThansk for the help anyways, it facilitated me finding my problem at least
11:55.06*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
12:01.24*** join/#asterisk morlac (i=morlac@213.186.161.28)
12:03.25konradsHello, how do I load-balance callerids?
12:04.50konradsload balance B channels, actually
12:05.17morlacany1 knows if my assumption that all sound streams are translated to SLinear in asterisk then A-LAW or u-LAW?
12:05.26RoyKanyone here using ser?
12:05.32morlacis actaully correct?
12:07.25*** join/#asterisk _jeh (n=jeh@ext122.almare.com)
12:07.25Danettwhy would anyone use ser?
12:07.35Danettwith a nice asterisk hack you can retrieve the same
12:10.09ScaredyCatser is a proper sip proxy that's why
12:10.28X-RobI'm gunna become the next hitler and kill all the jews. And 1 clown.
12:10.48ScaredyCatthat's already on bash.org
12:11.00ScaredyCat*sigh*
12:11.06X-RobIt's the funniest one I've seen
12:11.11X-RobSorry 8)
12:11.36X-RobI'm building a Celeron 500 with Windows 2000. I HAVE TO RELEASE MY TENSIONS SOME WAY!
12:11.36ScaredyCatthere are some good ones by it appears to be a bit stale, they don;t update submitted ones enough
12:12.03bkw_asdfadsfasdf
12:12.12ScaredyCatget back to sleep bkw_ ;)
12:12.17bkw_I can't boi
12:12.26ScaredyCatwhy? too many drugs?
12:12.27bkw_I have no clue why i'm up so early
12:12.32ScaredyCatlol
12:12.41X-Robit's 7am
12:12.44X-Robit's not _that_ early.
12:12.57ScaredyCatbecause you want to have intelligent conversation before those dullards wake up ?
12:13.11bkw_X-Rob, its a holiday
12:13.16ScaredyCatdepends where you are X-Rob
12:13.16bkw_and I was up at 3am
12:13.28ScaredyCatso was I, your time ;)
12:13.40bkw_smartass...
12:13.43ScaredyCathehehe
12:13.45bkw_I knew their was a reason I liked you
12:14.03ScaredyCatfor my brain, naturally
12:14.08morlacshortest path to avoid audio codec transaltions as much as possible is to save my audio files in Slinear or a-law(g711)? am running PriISDN.
12:14.28Danettis G.726
12:14.30bkw_morlac, g711
12:14.35ScaredyCatI have 20k mp3's, you'd think that I could get a track without Sting on it wouldn't you
12:14.38Danettsoryr: Do you need to pay for G.726
12:14.40konradsHow to Load Balance ISDN channels?
12:14.43morlacthanks bkw_
12:14.44bkw_but codec translation isn't that costly
12:14.54bkw_when speaking gsm or slin
12:15.00bkw_thats the least of your worries
12:15.35X-RobScaredyCat - it's 10:17pm _here_, but bkw's IRC client has a modicum of decency and bothers to respond to a /ctcp time so I can tell what time it is for him
12:16.02bkw_Mon Sep  5 07:14:50 CDT 2005
12:16.02morlacyes, agreed....but freeing as much CPU as I can for receiving and maintaining 120 calls (they comein in as little as 1 minute) might help
12:16.18bkw_morlac, 120 calls in 1 min?
12:16.22morlacyes
12:16.25bkw_give up now
12:16.30ScaredyCatlol
12:16.35bkw_thats going to be an interesting thing to take place
12:16.39Danettallow=g726
12:16.44Danettthis would be possible right?
12:16.49morlacIt worked on version 0.5
12:16.55ScaredyCatI give it 10 seconds before it barfs
12:17.11bkw_morlac, honestly that many in 1 min... that needs to be tested better
12:17.36morlacmaybe not 1 minutes, but all our lines get busy in less than 5 minutes for sure
12:18.28morlacany tips?
12:18.41ScaredyCatbuy another box and split the calls
12:19.12Kattybeep.
12:19.14morlacdid I mention that am running IVR scripts only..
12:19.27morlacperl all over the place
12:19.32ScaredyCatomg
12:19.43morlacyes, scary
12:19.56ScaredyCatuse fastagi insstead and offload the perl shite to that
12:20.13ScaredyCatie another dedicated box
12:20.34morlacyou think it will help?
12:20.39*** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net)
12:20.39*** mode/#asterisk [+o bkw__] by ChanServ
12:20.51ScaredyCatyes
12:21.22ScaredyCatyou'll be making your box just handle calls not running perl scripts
12:21.25bkw__ok i'm back
12:21.26morlaci mean the extra interrupts caused by the network interface is not gonna be an issue
12:21.27bkw__muhahaha
12:21.42Ahrimanesmorlac: or look into res_perl
12:21.48ScaredyCator you could always rewrite them in C and make ivr apps
12:22.05morlacAhrimanes> evertime I run that my Asterisk core dumps
12:22.18ScaredyCatAhrimanes: with fastagi you can offload the cpu required to another box
12:22.20Ahrimanesmorlac: hm ok, works fine hre
12:22.43AhrimanesScaredyCat: yes but with res_perl you dont have to spawn perl all the time, so alot of cpu saved there as well
12:22.57morlacAhrimanes> it worked once, then no luck....
12:23.03RoyKanyone used this? http://www.vovida.org/applications/downloads/loadbalancer/
12:23.03ScaredyCatyes, but bkw_ wrote that, so you know it wont work properly ;)
12:23.12morlaclol
12:23.17*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
12:23.17bkw__I didn't write res_perl
12:23.22puzzledRoyK: no but it seems no longer maintained
12:23.23Ahrimaneshaha
12:23.26bkw__anthm did
12:23.30bkw__anthm is my boss
12:23.48ScaredyCatahh ok, it'll work then... just so long as bkw_ didn't fiddle with it :P
12:23.52ScaredyCatand res_perl
12:23.55bkw__*FINGER*
12:24.28ScaredyCatthat's what you're calling it now then
12:24.35*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
12:24.36puzzledhehe
12:24.45morlacok, fastagi + all audio in g711 should make it cool
12:24.58bkw__you're not getting it still
12:24.58morlacIll have to dig a good machine
12:25.07bkw__the fact that the audio is in g711 is the least of your problems
12:25.25ScaredyCatactually g711 will help matters
12:25.34ScaredyCatif it's pstn -> *
12:25.47bkw__one problem is asterisk is tested, developed and such on a single machine.. with maybe 1 or 2 calls max on it
12:25.59bkw__not many people load machines up for testing while developing code
12:26.13morlacI read that call setup is a resource hog
12:26.32bkw__yes call setup and tear down are the points where you'll have problems
12:26.32RoyKpuzzled: do you know an alternative to that?
12:27.05morlacif I can get * to hold the calls (even if audio quality is degraded) then am a happy man
12:27.06puzzledRoyK: ARA with anthm's SIP clustering patch
12:27.27RoyKpuzzled: url?
12:27.38puzzledbugs.digium.com
12:27.43puzzledARA = realtime *
12:27.46bkw__yes
12:28.01RoyKpuzzled: er... asterisk can't proxy sip, right?
12:28.10*** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3864485.sympatico.ca)
12:28.14puzzledthen use SER
12:28.26RoyKpuzzled: AFAICS, SER can't do much clustering
12:28.28DanettRoyK: it can
12:28.33Danettwith a hacjk
12:28.34Danetthack
12:28.51RoyKDanett: do you know how?
12:29.00Danetti have to look up it up somewhre
12:29.03puzzledRoyK: SER scales very well so I'd be surprised if you need more than 2 SER boxes with linux-ha stuff on it
12:29.16RoyKpuzzled: i know...
12:29.28morlacbkw__> any tips or ideas on how I might be able to accomplish it?
12:29.40RoyKpuzzled: do you know how to set this up? SER in front of asterisk etc?
12:29.48Danetti have to check my laptop at home
12:30.16puzzledRoyK: I wish I could maxout your credit card but I'm afraid I don't know
12:30.25RoyKhehe
12:31.14RoyKDanett: have you done this earlier?
12:32.52Danettno. but i have seen it somewhere
12:32.52DanettI read an article about it
12:32.53Danetti was quiet nasty
12:34.54*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
12:35.58*** join/#asterisk lehel (n=asd@82.79.20.17)
12:36.03RoyKDanett: i don't give a fuck if it's written in ADA, as long as it works
12:36.05puzzlednot bkw_ !
12:36.37*** join/#asterisk konrads (n=konrads@out.ctkom.lv)
12:37.05*** join/#asterisk konrads (n=konrads@out.ctkom.lv)
12:39.34bkw__ScaredyCat, I don't have a 4x4
12:40.30gordonjcpthere's a towrope in the back
12:41.04puzzledbkw_: does your rtp patch apply cleanly if I use the rtcp, rpid and sip jb patch?
12:41.27*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
12:41.55DanettWhen using DISA, does someone know how to trap an error? like wrong number, or unable to connect?
12:43.08queuetueWhat is the default "admin password" for the flash control panel, and where do I change it?  (using a@h, if that matters.)
12:43.46*** join/#asterisk f_meehan (n=fmeehan@whoami8.cedval.org)
12:43.53lehelhello
12:43.58puzzledhi
12:44.03*** join/#asterisk tandem1 (n=tandem1@misp.misp.tuiasi.ro)
12:44.43konradsi have in [incoming] context:
12:44.44konrads[incoming]
12:44.44konradsexten => s,1,Playback(demo-abouttotry)
12:44.44konradsexten => s,n,Dial,SIP/xlite1
12:45.02*** join/#asterisk newl (n=newlook@203.59.168.152)
12:45.03lehelpuzzled: you have any idea why i doesn't hear the ringing on a remote iax call?
12:45.23puzzlednope
12:45.25konradsDoes this mean that any call I will redirect any call that arrives to incoming will get routed to SIP/xlite1?
12:46.22tandem1hi
12:46.23Danettkonrads
12:46.31Danettuse the dial option "r"
12:46.39Danettr: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so.
12:46.41lehelpuzzled: is this ok?: exten => _000ZXXXXXXXXXX,2,Dial(IAX2/boxd-peer/${EXTEN})
12:46.50*** part/#asterisk zoo (i=nobody@ip-121-16.travedsl.de)
12:47.16tandem1quick Q: i need only one TDM400P for an office with asterisk server, right ?
12:47.26puzzledlehel: look sok to me
12:47.36puzzledlooks ok to me
12:48.00lehelshould i put an "r" somewhere.. or to define the ringing time?
12:48.11konradsDanett: where should I put the r?
12:48.24Danetthttp://www.voip-info.org/wiki-Asterisk+cmd+Dial
12:48.25puzzledlehel: I usually do. do show application Dial in the asterisk console
12:48.57konradsDanett: i would be content if it would at least playback the demo audio
12:49.49konradsDanett: but this appears not to be happening :(
12:51.13Danettthen i don't know
12:51.15tandem1second Q: i see the ISDN cards have power supply. I really need one?
12:52.02konradsDanett: CAPI: no interface for PLCI = 0x101 MN = 0x6
12:52.07konradsdoes this ring a bell maybe?
12:52.08*** join/#asterisk luke-jr__ (n=luke-jr@user-0c938q3.cable.mindspring.com)
12:52.33Danettnope
12:53.37*** join/#asterisk zotz (n=zotz@24.231.36.100)
12:53.46konradsbut in general that Playback(demo-blah) exten should give me some music
12:53.51konradsor at lease be spawned
12:53.58konradswhen any call reaches incoming context?
12:54.16*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
12:54.21tandem1guys, can you enligthen me, please ?
12:55.20Danettyou first have to answer the call?
12:55.38konradsi do?
12:56.59*** join/#asterisk TheCops (n=mdb@206.248.136.146)
12:57.47lehelpuzzled: what could be with my mysql cdr?: Sep  5 15:57:05 WARNING[21186]: cdr.c:421 ast_cdr_free: CDR on channel 'Zap/3-1' not posted
12:57.56dtwilsondoes anyone know if there's an issue with using cdr_mysql on a MySQL 4.1x server? I can't seem to connect
12:58.09lehel"... lacks end"
12:58.55*** join/#asterisk littleball (n=littleba@cm157.epsilon173.maxonline.com.sg)
12:59.10*** join/#asterisk oplog2 (n=oplog2@dsl-202-72-172-165.wa.westnet.com.au)
13:00.41Kattyif you update a kernel, but forget to update lilo.conf, and reboot...how do you get back to lilo to edit it?
13:00.46dtwilsononly asking as 4.1x uses a new authentication hash afaicr which is incompatible for older mysql client libraries
13:01.28puzzledlehel: no idea. I use cvs HEAD and don't have an issue with it
13:03.34ScaredyCatanyone recommend someone who can draw PC graphics?
13:04.02puzzledScaredyCat: tigert or jimmac over in #gimp
13:04.35ScaredyCat:D ta
13:05.09DanettHow can i check if an certain context exists?
13:05.30Kattyi see.
13:05.32ScaredyCatshow dialplan <tab>
13:05.36Kattyso no one knows eh?
13:05.47Danetti mean in the extension.conf
13:05.49Kattyno one has ever updated a kernel, forgotten to update lilo.conf and then rebooted?
13:06.00Kattyi find that hard to believe
13:06.13johnmKatty: all you need to do is edit lilo.conf later on, and then reboot again.
13:06.16Danetthmm.
13:06.18Danetti use grub
13:06.20puzzledKatty: get a rescue cd, boot it and correct the error
13:06.20johnmKatty: run: lilo of course
13:06.22Kattyjohnm: yes, dear, i know this.
13:06.26Kattybut i get a kernel panic
13:06.32ScaredyCatcat /etc/asterisk/extensions.conf | grep "[" | grep "]"
13:06.37Kattyand don'tknow how to get to a point where i can edit lilo.conf
13:06.45ScaredyCatprolly not so efficient but there you go
13:06.53Kattypuzzled: how does one get a rescue cd?
13:06.59puzzledKatty: usually you don't upgrade the kernel (so remove the old one and installed the new one) but instead install the new one next to the old one so you can revert
13:07.11littleballHi, ScaredyCat, is it better for me to convert to Sangoma  card?
13:07.20Kattypuzzled: i apt-get install kernel-image-foo
13:07.21puzzledKatty: let me find a iso. just a sec
13:07.35queuetueWhat kinds of options do I have for companywide speed dialling?
13:07.38Kattypuzzled: i was under the impression that would add to the existing kernel, and update my conf automatically (like grub does)
13:07.41ScaredyCatcat /etc/asterisk/extensions.conf | grep "\[" | grep "\]"
13:07.44Kattypuzzled: that was not the case apparently :<
13:08.06tandem1guys, i see the ISDN cards do have power supply. I really need one? or is just for power cuts?
13:08.20ScaredyCatlittleball: it's up to you really, but the config is the same, and there's supposed to be a chan_sang soon which will make things better too
13:08.27puzzledKatty: I don't use debian so can't help you with that but here is a rescue cd: http://download.fedora.redhat.com/pub/fedora/linux/core/4/i386/iso/FC4-i386-rescuecd.iso
13:08.42Kattythanks.
13:09.00puzzledScaredyCat: what's "soon"?
13:09.24puzzledwhen it is ready? :)
13:09.25*** join/#asterisk rob314[laptop] (n=rob314[l@cpe-65-185-169-238.neo.res.rr.com)
13:09.48littleballsorry, what is chan_sang?
13:09.58ScaredyCatlol puzzled
13:10.03puzzledlittleball: the chan_zap for sangoma cards
13:10.26ScaredyCatlittleball: sangoma us the zap channels atm, chan_sang would replace it for sangoma cards
13:10.37ScaredyCator what puzzled said ;)
13:11.02*** part/#asterisk rob314[laptop] (n=rob314[l@cpe-65-185-169-238.neo.res.rr.com)
13:11.19littleballok. how about the price? i need to pay 2300USD$ to bue TE411P card here
13:11.42puzzledlittleball: call the vendors and ask for a quote
13:12.06ScaredyCatwtf!
13:12.11ScaredyCat2300!!
13:12.22littleballyes
13:12.44littleballif Sangoma card is cheaper, i would like to buy 8 port E1 card
13:12.48ScaredyCatwouldn't it be cheaper to fly to australia, via the north pole
13:13.14ScaredyCatthere's no single 8 port, it would be 2 4 port card
13:13.15ScaredyCats
13:13.21puzzledScaredyCat: well, digium has the 411 w/ec for $2495 on their site
13:13.25*** join/#asterisk coppice (n=chatzill@127.143.17.210.dyn.pacific.net.hk)
13:13.33puzzledmorning coppice
13:13.56coppiceevening
13:14.15ScaredyCatahh , that echo echo echo can board...
13:15.01coppicedo you mean the echo can't board?
13:15.55puzzledok share the dirty details. does the echo can board not work as advertised?
13:17.51coppicedoes anything ever work completely as advertised? :-)
13:17.59bkw__coppice, yo
13:18.07puzzledcoppice: true :)
13:18.59puzzledbkw_: I'll build a new rpm with your patch included and then will give it a whirl
13:19.23bkw__puzzled, ok
13:19.32coppicebkw: how's life in the defendant business? :-)
13:19.49bkw__no clue yet
13:19.50bkw__no word
13:20.17*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
13:20.37coppicedoes this echo can board really work? i've heard multiple negative things, and one positive
13:21.53puzzledskypification
13:21.57coppicebkw_ don't bother. it sucks at 16k. 24k is OK
13:22.30bkw__hehe thats what I thought
13:22.33bkw__32 is fine with me
13:22.39bkw__just playing around with stuff
13:22.51bkw__coppice, still leaks btw
13:23.15coppicesome people say 32k is just like a-law/u-law. some people also say 64K MP3s sound fine :-)
13:23.25bkw__exactly
13:23.27coppicebkw_ does it leak less?
13:23.33bkw__yes it leaks less
13:24.20coppicethen I must have fixed something :-) how fast is it leaking now?
13:24.33bkw__won't really know till tommorow
13:24.37bkw__holiday and all
13:24.49bkw__its not getting used much right now
13:25.05bkw__I suspect tomorrow morning will be hell
13:25.09bkw__:P
13:25.11littleballI try to quote from Sangoma website, but i don't know what is my line protocol. such as ATM, frame-relay,x.25 etc.
13:25.43bkw__littleball, what are you doing?
13:26.00littleballa simple web callback service
13:26.16bkw__but you dont know your line protocol? what do you mean?
13:26.41littleballbecause from Sangoma website, i need to fill in "what is your line protocol?"
13:27.04bkw__you're getting a voice board?
13:27.07MaksimOpenBSD 3.7-stable http://pastebin.ca/22125 Any ideas? :)
13:27.25littleballi try to get a four port E1 line card
13:27.45bkw__EuroISDN ?
13:28.06littleballasterisk will call one number and then call another number then connect these two calls.
13:28.09*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
13:28.24bkw__littleball, you sound like you have bitten off more than you can chew
13:28.47littleballEuroISDN? yes, i just read asterisk about 1 week :)
13:28.58bkw__well this isn't asterisk stuff that we are talking about
13:29.01littleballbut buy and try it first
13:29.03bkw__its telco related
13:29.27littleballthen i will ask telco
13:29.37bkw__hehe
13:29.59tandem1what do you think about  Billion 1 Port S0 Card ? works with asterisk?
13:30.30Ahrimanesds3 should be comming? hehe
13:30.30littleballbkw_, asterisk is quite interesting. My background is pure computer science... Need some time to pick up.
13:34.30littleballis Q.931/Q.932 line protocol
13:34.32littleball?
13:35.43*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
13:38.47littleballbkw_, yes, EuroISDN
13:39.06queuetueBillion?  That's a lot of ports...
13:41.04morlacbillion? wow, you must be the biggest teleco i ever heard of
13:43.09tandem1http://shop.beronet.com/product_info.php/cPath/21_25/products_id/52?osCsid=81117910b7f72cfe6379f7f7fbc809e3
13:43.23tandem1i ment this 39E card
13:43.31tandem1" Billion 1 Port S0 Card "
13:43.32puzzledtandem1: i think it's a hfc based card so should work with bristuff & asterisk
13:43.58tandem1ah, you didnt tried
13:44.15*** join/#asterisk nicodejo (n=niclas_m@ppp-250.net-611.magic.fr)
13:44.24nicodejohello !
13:44.28tandem1one card should be enough, is it? one card and the asterisk server
13:44.31coppicehousing a billion cards would be tough, and you'd have more lines than China Telecom
13:44.33tandem1Hi, nicodejo
13:45.18nicodejoi bulding an asterisk pabx in france
13:45.20*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
13:45.27*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
13:45.39nicodejoi'm happy my bri card working
13:46.16nicodejotandem1 but i can't phone out !
13:46.37nicodejotandem1 with my sip softphone
13:47.13nicodejoasterisk can't create the capi channel
13:47.42morlacoh, the billion name is just misleading....i wonder whose idea it was
13:48.56coppiceits only billion in english. in chinese its just 100 million
13:49.11*** join/#asterisk grimse__ (n=grimse@p5481C30A.dip.t-dialin.net)
13:49.11puzzledtalk about miscommunication
13:49.43ManxPowerIn the USA "1 billion" is "1000 million" in the rest of the world.
13:49.56coppiceI don't trust companies who try to give the fake impression they are japanese
13:49.57tandem1:>
13:50.23tandem1puzzled, i see some ISDN cards do have power supply. I really need one? or is just for power cuts?
13:50.33coppiceManxPower 1 billion is 1000M everywhere, though in britain it used to be 1,000,000,000,000
13:50.38ManxPowerI don't trust anyone that tries to fake being part of a culture that worships Hello Kitty.
13:50.49littleballbye
13:50.51ManxPowercoppice, it's not in the USA 8-)
13:51.05puzzledtandem1: dunno
13:51.11*** part/#asterisk Akelavlk (n=jansun@82.119.239.141)
13:51.13*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
13:51.23tandem1thanks
13:51.25*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
13:51.28*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
13:51.51coppiceManxPower: I think you misunderstood the original comment. The chinese name for the company billion is actually 100M
13:52.01ManxPowerAhrimanes, OK.
13:52.05ManxPowerI'm still waking up
13:52.47*** join/#asterisk TheCops (n=mdb@206.248.136.146)
13:53.13X-Robit only takes three commands to install Gentoo
13:53.23tandem1bye, guys
13:53.34puzzledX-Rob: and 3 centuries of waiting until it finishes :)
13:53.38X-Robcfdisk /dev/hda && mkfs.xfs /dev/hda1 && mount /dev/hda1 /mnt/gentoo/ && chroot /mnt/gentoo/ && env-update && . /etc/profile && emerge sync && cd /usr/portage && scripts/bootsrap.sh && emerge system && emerge vim && vi /etc/fstab && emerge gentoo-dev-sources && cd /usr/src/linux && make menuconfig && make install modules_install && emerge gnome mozilla-firefox openoffice && emerge grub && cp /boot/grub/grub.conf.sample /boot/grub/grub.c
13:53.48X-RobThat's the first one
13:54.14X-RobI'm off to bed. It's midnight.
13:54.31*** join/#asterisk sambal (n=sambal@213.148.236.189)
13:54.50*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:55.28*** join/#asterisk ManxPower (n=eric@69.149.125.137)
13:55.46morlaccoppice> are you receiving bug reports regarding CPU spikes reaching 100% with R2 every few seconds? using 0.2c or 0.3pre4 of the libraries.
13:55.46*** join/#asterisk weazul (n=weazul@82-169-62-42-mx.xdsl.tiscali.nl)
13:56.09coppiceare you using * 1.0.x?
13:56.15morlacyes
13:56.29*** join/#asterisk ManxPower (n=eric@69.149.125.137)
13:56.42morlacon CentOS4.1, recompiled kernel
13:57.02morlacremove libmfc/unicall and everything goes ok
13:57.18weazulhi all .... i need info about radius and asterisk. There should be an radius.conf in configuration dir but there isn't how should i continue?
13:57.24*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
13:57.32clive-morlac whats "R2" ?
13:57.39*** join/#asterisk tux99 (n=w@195.78.52.6)
13:57.47coppicenope. 1.1.x used to send the CPU to 100%, though I hope the latest updates fix that.
13:57.54morlacclive> protocol MFC/R2, teleco protocol
13:58.04clive-ahh,,,ta:)
13:58.12*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
13:58.18clive-I was wondering revisiopn 2 or something
13:58.24wasimclive-: nooo ... its Ribbed and Rippled (for her pleasure)
13:58.50morlaccoppice> i can provide info if you need them ;)
13:59.12coppicethere were some issues with people using SIP phones that pumped out far too much RTP, but I think it tolerates that crap now
13:59.28morlaccoppice> some ppl contacted me about it after posting to * mail list
13:59.41bkw__coppice, whats too much RTP?
13:59.46TheCopsSomeone saw wireless headset for snom phone ?
13:59.47coppicethey could be running 1.1.x
14:00.15morlaccoppice> letme check
14:00.46*** join/#asterisk ManxPwr (n=eric@69.149.125.137)
14:00.52coppicebkw_: some phones were pumping out bursts of audio, and flooding a buffer
14:01.02bkw__ah bursts...
14:01.15morlaccoppice> no, * 1.0.9
14:01.21clive-wasim, I never knew those things where for sale in pkistan..:)
14:01.38*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
14:02.38coppicemorlac: maybe its something specific they do, like the RTP floods problem. there are people using unicall in high volume systems
14:02.48coppicewhat are you doing?
14:03.40morlaccoppice> actualy, am getting this with no cables attached to the cards. no calls, nothing....system is 100% idle
14:04.09morlacI even unload IAX and SIP just to check and i get the same trouble
14:04.47morlacand I tried profiling, but multi-thread profiling in linux is not a walk in park for me
14:06.16coppiceno cables would produce alarms. I wonder if there is a problem with alarms I missed
14:06.48morlacI did attach cables to check as well, and cross cable between differnt ports....
14:06.58*** join/#asterisk grimse (n=grimse@p5481C30A.dip.t-dialin.net)
14:07.26morlacmaybe it just hates my guts
14:07.35morlac:b
14:08.05morlacseriously,I must have tried 99% of everything possible.
14:10.15coppiceso you just start * and it goes straight to 100%?
14:11.35*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
14:11.43*** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
14:11.48coppicewhat does your zaptel.conf look like?
14:13.23weazulnobody familiar with radius icw asterisk?
14:13.34jalsothi
14:13.55jalsotdoes anybody know how can I turn off ringback tone on iaxclient?
14:14.04jalsotsorry, I mean, iaxComm
14:14.33morlaccoppice> it does not go 100% straight away....it fluctuates every few seconds, like 5 seconds, goes gradually up tell 100% then back down
14:14.51coppicestrange requirement
14:15.08coppicedisabling ringback, I mean
14:15.17coppicemorlac: weird
14:15.23johnmHas anyone here got a setup which is a 4xxP card, one span is the ISDN30 from the telco, the other an ISDN30 to an existing PBX?
14:15.27morlaccoppice> my zaptel has span=1,0,0,cas,hdb3 ...etc 4 spans....and the rest just like your website
14:15.29jalsotthe problem is that I'm getting ringback from pstn, so I hear 2 ringback tones
14:15.40jalsotin which the local is too strong
14:15.53coppicemorlac: that's wrong, for a start
14:16.01jalsotmaybe option r can be useable in dial command...
14:16.14coppicebut if you have nothing connected it would be irrelevant
14:17.16morlaccoppice> yah
14:17.44tzangercoppice: can I pick your brain for a moment?
14:18.15morlaccoppice> i followed the instruction on your website to the litter
14:18.16coppicemorlac: however, the port connected to the telco should be defined as a clock source
14:18.33tzangerno I said pick his brain, not pitch his brain :-)
14:18.35morlaccoppice> tried that as well
14:18.44coppicemorlac: that's what they all say :-)
14:19.03coppicetzanger: what is it?
14:20.03*** join/#asterisk case_ (n=case@mailhost.seeft.com)
14:20.47morlaccoppice> I know, I have been trying for like 2 weeks....I dont mind trying again and again...but as you said, if no cables are connected, then it is irrelevant
14:20.58*** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de)
14:21.15tzangerjust thinking about my line quality monitor...  I mean I can send the mW tone over and listen for the response, but in terms of quantifying the line quality, I was thinking a mixture of calculating THD and also using something ismilar to an LP click/pop removal algorithm to detect any periodic noise (which I think is more of a problem)
14:21.25coppicemorlac: do you have loggin enabled for R2? does it show anything interesting?
14:21.27lilalinuxHey guys
14:21.55tzangerDo you think THD would be enough?  I mean I know it's great at measuring continuous distortion but I dunno if a little click/pop here and there is going to increase the THD significantly :-)
14:22.04morlaccoppice> if you are talking about the loggin in unicall.conf, then it is commented out
14:22.19morlaccoppice> anything else, is the default asterisk defined
14:22.23coppiceenable that and see what gets logged
14:23.06coppicetzanger clicks and pops are not harmonic distortion :-\
14:23.12*** join/#asterisk MikeJ[Laptop] (n=ircatjer@c-67-163-80-88.hsd1.il.comcast.net)
14:23.15morlaccoppice> ok, ill give you the details
14:24.10tzangercoppice: correct, which is why I'm looking at click/pop removal algos since that's likely the more prevalent kind of distortion (periodic) on a VOIP link
14:24.42coppiceVoIP links only have distortion because of data loss
14:25.13ManxPwror jitters
14:25.49coppicesame thing. jitters only matter when they enough to loose something
14:26.17morlaccoppice> but I remember getting the farend/local end unblocked when I connect the cables, then nothing.....anyway, reinstalling latest version....maybe it is solved
14:27.24coppicepossibly. that fixes a bug that only seems to have shown up on 64 bit machines, but is really there on all of them
14:28.21morlac0.3pre5?
14:29.13morlacspandsp 0.3pre1
14:29.43*** part/#asterisk tux99 (n=w@195.78.52.6)
14:29.52coppiceunicall-0.0.3pre5 and spandsp-0.0.2pre20
14:30.09coppicespandsp-0.0.3pre1 won't hurt unless you use FAX
14:30.38morlac0.02pre20 it is then, no FAX here
14:30.52coppicethe interface to app_rxfax and app_txfax changed in spandsp-0.0.3pre1 as elements of T.38 start to go into it.
14:31.12morlacsweet
14:31.51puzzledcoppice: nice
14:32.46*** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
14:33.52*** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com)
14:38.44*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
14:39.46tzangercoppice: have you seen that post on -users about the memory leak in spandsp and/or rxfax?
14:40.16coppicehave you seen the exchange a page or two earlier? :-)
14:40.26tzangerno I didn't see the response :-)
14:40.44tzangeror rather something between the two of you earlier
14:41.06coppicebkw says the latest version has reduced, but not eliminated leaking
14:41.37tzangerahh
14:41.56coppiceit should certainly have reduced it. I fixed a definite problem
14:42.28manyi dont get it. is app_sms  for rx/tx to a remote smsc or for rx/tx to a locally connected dect fon or both?
14:42.51tzangerI need to figure out how to send an email to an SMS gateway and light/extinguish MWI on my cell phone
14:42.58puzzledmany: don't think you can use app_sms to link up to an smsc
14:43.07*** join/#asterisk tux99 (n=w@195.78.52.6)
14:43.09tzangerI can send it to a specific online SMS gateway for $0.17/msg but that's bloody expensive
14:43.19coppicemany: kind of both, but its modem is a bit flaky
14:43.38coppicei think it only supports one of the SMS protocols, though
14:44.24puzzledI thought the smsc was for telco <--> telco connections
14:44.37manyno, its for handy<->telco connection, too.
14:44.45manyor for isdn<->telco
14:44.53puzzledah right, then I retract my statement :)
14:44.57*** part/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com)
14:44.59manyi just dont get the examples.
14:45.43manymh.
14:45.48many*test*
14:47.05RoyKare anyone working on asterisk sip proxy functionality?
14:47.30Katty^- is
14:47.41puzzledRoyK: why would you need that if you have SER?
14:47.43coppiceasterisk's sip functionality is already poxy
14:47.53RoyKpuzzled: then there is NAT problems
14:48.04oejNo, Asterisk will never be a SIP proxy.
14:48.20puzzledafaik SER does not have a NAT problem. there are several SER NAT solutions
14:48.21oejWe are a multiprotocol PBX, which significantly is an opposite to a SIP proxy
14:48.50*** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net)
14:49.49RoyKcoppice: already a proxy? i thought asterisk never did anything except bridging.....
14:50.29coppiceRoyK: maybe I needed to add a :-)
14:50.49*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
14:51.26coppiceoej: why does that make * the opposite of a proxy?
14:51.32*** join/#asterisk outsidefactor (n=blah@203-206-247-109.dyn.iinet.net.au)
14:51.44*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
14:51.50oejcoppice: We always answer and initiate call, *never* forwards an untouched SIP message
14:51.58oejAsterisk is an endpoint in the SIP game
14:52.14coppicebut it doesn't have to be.
14:52.30oejcoppice: Well, if so, you make Asterisk a completely different software
14:56.49queuetueDoes anyone have a recommendation for a systemwide speed dial list?
14:57.06oejUse astdb
14:58.43queuetueoej: Can you point me towards more info?  Googling for astdb... I'm not sure what part applies to me.
14:58.48mog_homecontexts are your friend queuetue
14:59.22oejSave speed dials with dbput functions
15:00.32queuetueIs some of this documented?  I don't really know how "cobntexts are my friend" here or what dbput is ... Is there any documentation on how to implement and maintain a speed dial list?
15:00.46*** join/#asterisk Koshatul (n=evangeli@ip157.net65.ipnetworks.net.au)
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15:01.20*** join/#asterisk nitram (i=foo@superblob.com)
15:01.57*** join/#asterisk gniretar (n=mark@198.173.197.15)
15:02.14gniretarcan anyone refer me to some good documentation on how to set up agents?
15:02.21morlaccoppice> I think I might have pinpointed it
15:02.42coppiceso?
15:03.08morlaccoppice> if I start * with 'service asterisk start' then I get the cpu spikes
15:03.27morlaccoppice> if I start it liek 'asterisk -vvvvvvg &' then I dont see it
15:04.02coppicewell spotted. i wonder if this is some odd permissions thing
15:05.02*** part/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
15:05.06morlacif you are interested, I can grant you access to the system to collect some data....at least you can collect some info
15:05.25gniretarcoppice> I have only cought the tail end of your convorsation but i think that you may want to use the debian init scripts.  They are far more universal and start and stop asterisk with commands like that.
15:07.13*** join/#asterisk alk (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com)
15:07.42coppiceI don't think debian scripts are something i would want to use. i never used any scripts to start or stop *, so i have no idea what quirks there might be
15:09.24gniretarcoppice> OK then.  Everything i do is directly referencing the init scripts.
15:10.49coppicemorlac: where did your init script come from?
15:11.45*** join/#asterisk hellagony (n=egutierr@200.121.129.180)
15:12.17morlaccoppice> asterisk it self by make config
15:13.58morlaccoppice> I put safe_asterisk in rc.local and I confirm it is ok, no cpu spikes
15:14.25lilalinuxdoes mISDN_dsp work with the mISDN stack in general, or is it solely for HFC cards?
15:15.39morlaccoppice> I run at level 3 btw
15:17.11*** join/#asterisk tld_ (n=tld@253.80-203-96.nextgentel.com)
15:19.49*** join/#asterisk Steppy (n=me@82.161.245.126)
15:21.45coppiceI am running * now, using the script. it looks OK so far
15:21.51coppiceI use FC4
15:24.23morlacrun top and update set to 1
15:24.28*** join/#asterisk speck (n=root@ewersbach.net)
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15:27.15dtwilsonhmmm - if when using cdr_mysql I decide to let mysql generate its own uniqueid - does this value become available to asterisk? i.e. in CLI output?
15:27.53djspeckhello, is there someone who can help me.. I have a problem with dialing out.. when i'm using the dial command it rings only for 10 seconds then I get a I IND :TIMEOUT and the line hangs up.. I'm using mISDN and SIP
15:28.00dtwilsonor does asterisk still retain the use of it's own uniqueid values independently of what is created by MySQL?
15:28.14coppiceah, yes. after a while top starts showing little bursts of activity from *
15:28.25morlac:)
15:29.11*** join/#asterisk felipex (n=dsfdsf@85-18-136-75.fastres.net)
15:29.13morlacfor a while I feared that it might be system dependent. now I know FC4 exhibits the same behaviour
15:30.00coppiceit doesn't go to 100%, but i see bursts of 30 or 40% when it is doing nothing
15:30.33morlacI ran a logger on it, for me, it spiked to 100% and down pretty fast
15:31.08morlacbyt constantly
15:31.32*** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
15:31.50djspeckI hope there is someone who can help, i tried for days...
15:32.02MikeJ[Laptop]coppice, I think you are best off to just put everything as you propose on that bug and let it get reviewed.. the bit at a time asking for imput approcah seems to not be working
15:32.30morlacsorry, that was 'but not constantly'
15:32.59coppiceMikeJ: submitted unfinished stuff isn't helpful
15:33.23MikeJ[Laptop]well... my point being, you will likely get ne real input for a month or so
15:33.40MikeJ[Laptop]do you have suggestions on the table for how to handle the multiple frames?
15:33.55coppicethe key structural thing that needs sorting out is how to return multiple frames. then things should move forward
15:34.31MikeJ[Laptop]like I said, my suggestion is to make a proposal, and write it...
15:35.04coppiceI really don't care. if other people want this stuff, its up to them
15:35.31MikeJ[Laptop]fair enough.
15:35.45*** join/#asterisk mbranca (n=matteo@81.208.92.210)
15:37.47gambolputtyI have an * box that I am trying to allow anyone to call into via SIP URL even if they aren't an extension on the box.  I keep getting error that say failed authentication.  Any ideas?
15:38.21ManxPwrgambolputty, why not just greate a guest account without a password
15:38.32ManxPwrthen just dial guest@yourfdqn
15:38.50ManxPwryou can also try insecure=very in sip.conf
15:39.02*** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk)
15:39.04gambolputtyI have insecure=very there now
15:39.06*** part/#asterisk secure75 (n=mic@ppp-82-135-14-145.mnet-online.de)
15:39.15*** join/#asterisk Tili (n=Tili@202-133-65-229-dialup.sat.net.pk)
15:41.36gambolputtycalls still don'
15:41.39gambolputtydon't get in
15:43.23*** join/#asterisk davidinno (n=davidinn@217.141.202.50)
15:43.26davidinnohello
15:43.49davidinnoi need some information about configure asterisk... someone can help me?
15:44.04davidinnoi'm going to change sip.conf
15:45.27davidinnosomeone can say me an fwd server?
15:45.38davidinnofree
15:45.49*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
15:46.00morlaccoppice> any ideas to what might be causing the problem when using the init.d script? I wish to understand the reason.
15:46.07coppicejust as I think things are settling down for 1.2, another bunch of stuff changes AHHHHHH!
15:46.13*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc12l.dialup.mindspring.com)
15:47.41*** join/#asterisk riksta (n=rick@62.6.163.85)
15:48.05morlacjoin the club......thats my dillema at work
15:48.34*** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net)
15:49.01coppicewell I can reproduce this, and I will investigate
15:49.10tux99i've problems with the pin in meetme. it doesn't accept the pin
15:50.12tux99i've tested many combinations allready, but nothing works
15:50.22morlaccoppice> that will be great, specialy, if you document it in the README or Changelog so that we can learn something useful ;)
15:50.36davidinnocan someone say me an fwd server usable in asterisk?
15:51.11*** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net)
15:51.36*** join/#asterisk popvoxdave (i=user@dave2.toad.net)
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15:58.52rikstahi can someone remind me how to unlock the network setup menu in a 79xx cisco i can't find it anywhere
15:59.36*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
16:00.18JerJerselect unlock config and provide the password?
16:00.42loudsettings + 9
16:01.29rikstahehe DOH thanks
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16:06.56*** part/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net)
16:08.17*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
16:09.53*** join/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
16:10.28TripleFFF2sdf57 WARNING[58167]: config.c:893 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
16:10.30TripleFFF2sdfany ideas ?
16:10.32TripleFFF2sdfmysql is there
16:10.43TripleFFF2sdfdid that on other box but  fixed not knowing how
16:10.52RoyKtry connecting to it from the same box with same user/pass
16:10.57TripleFFF2sdfworks
16:11.01RoyKmysql -hHostname -uUser -pPass
16:11.07RoyKmysql -hHostname -uUser -pPass dbname
16:11.14file[laptop]got the .so loaded for mysql?
16:11.32TripleFFF2sdfyes
16:11.36TripleFFF2sdfnot sure
16:11.37TripleFFF2sdfabout .so
16:12.04*** join/#asterisk tomtom (n=tom@bender.linugen.com)
16:12.07file[laptop]well, if the .so responsible for connecting to MySQL and doing everything isn't loaded - don't you think the above error might occur?
16:12.11file[laptop]as the mysql engine would not be available?
16:12.14tomtomhi
16:12.15TripleFFF2sdf[skipping cdr_addon_mysql.so]
16:12.15TripleFFF2sdf<PROTECTED>
16:12.15TripleFFF2sdf<PROTECTED>
16:12.15TripleFFF2sdf<PROTECTED>
16:12.25file[laptop]res_config_mysql.so that's it
16:12.58puzzledlooks like it's unloaded in modules.conf
16:13.01TripleFFF2sdfoh
16:13.19RoyKtry 'load res_config_mysql.so'
16:13.32TripleFFF2sdfdarn
16:13.45TripleFFF2sdfhad a no  lod on app addon and mysql.. now wtf put that there
16:13.46TripleFFF2sdflol
16:14.08ManxPowerlooks like our CLEC's generators finally ran out of fuel
16:14.37TripleFFF2sdfthanks
16:14.40TripleFFF2sdflol
16:14.51TripleFFF2sdfManxPower .. diesel ?
16:15.00TripleFFF2sdfuse. heat gaz
16:15.06TripleFFF2sdf;) same without the dye
16:15.29Hmmhesaysanyone remember if you can use the ${EXTEN} variable in a goto cmd ?
16:15.36tomtomis it possible to combine callfiles & prepaid, ie. tell asterisk in the callfile how many minutes it may connect?
16:15.49RoyKHmmhesays: you can use any variable in any app
16:15.56ManxPowerTripleFFF2sdf, I would assume so.
16:16.03ManxPowerHmmhesays, yes.
16:16.04file[laptop]tomtom: probably
16:16.10JerJertomtom:  sure - using asterisk extension logic
16:16.22TripleFFF2sdfu should
16:16.38TripleFFF2sdfi tought astcc was using it .. and putting 999999999999 for any legth
16:16.48coppiceManxPower: is this a storm thing, or they just can't afford fuel any more? :-)
16:16.48tomtomjerjer, there are only a limited number of parameters you can pass in the callfile
16:16.53ManxPowerexten => 9411,1,Goto(91${EXTEN:1},1)
16:16.56ManxPowercoppice, storm thing
16:16.57tomtomso how would one go about realizing that?
16:17.02TripleFFF2sdflike calcing balance before dial.. then sending dal with a sec param
16:17.09ManxPowerHmmhesays, A priority is ALWAYS required in a Goto
16:17.20JerJertomtom:  you are not paying attention.    Asterisk extension logic is not a call file
16:17.34tomtomi am, i just don't see how it fits into extensions
16:17.38coppiceManxPower: how big an area is without power?
16:17.50tomtomin the end i'd have to be able to specify max nr of minutes per callfile
16:18.12ManxPowercoppice, a couple of hundred miles across and maybe 50 miles high on a map
16:18.29tomtomJerJer: r there any sites you could point me too? or provide a few config parameters for me to google for?
16:18.34JerJeryour call file sends the call to a specific context and extension
16:18.35ManxPowerNew Orleans to Biloxi at least and that's a 2 hr drive on the freeway
16:18.39JerJerproblem solved
16:19.05tomtombut then i would have to change the context/extension automatically before each call
16:19.29TripleFFF2sdftomtom what u trying to do
16:20.24tomtomTripleFFF2sdf: exactly that, initiate a number of calls, that may only take so long as the customer has a positive balance
16:20.41ManxPowerWhy not use a GotoIf?
16:21.01ManxPowerexten => _2009,1,GoToIf($[X${RDNIS} != X]?${EXTEN},4)
16:21.05JerJertomtom:  then you obviously have not normalized your asterisk dial plan
16:21.40tomtomhmm maybe :0
16:21.42tomtom:/
16:21.54*** join/#asterisk fugitivo (n=ajf@201.255.104.41)
16:22.18fugitivohello
16:22.35tomtomJerJer: that would also mean i'd have to install calling card stuff no?
16:23.08JerJeri don't know your situation
16:24.58lilalinuxI just installed chan_misdn and tried to start asterisk (with -vvvvc) but it doesnt end on the CLI but stops
16:25.07lilalinuxhttp://pastebin.com/355312
16:27.19*** join/#asterisk goof_ (n=goof@81.199.100.163)
16:29.28weazulnobody familiar with radius icw asterisk?
16:32.39*** join/#asterisk drumkilla_laptop (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
16:32.39*** mode/#asterisk [+o drumkilla_laptop] by ChanServ
16:34.19Tiliweazul: i know ic-radius and made it work with asterisk.
16:38.40ManxPower~docs
16:38.41jbot[docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:38.43ManxPower~mailinglist
16:38.44jbotfrom 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/
16:39.08JerJerTili: i'm sorry
16:40.18*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
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16:40.52TiliJerJer: for what?
16:41.03TiliI just played with it. not using it or anything
16:41.15JerJergood
16:41.22*** join/#asterisk Goshen (n=Goshen@67-40-107-29.slkc.qwest.net)
16:42.28TiliJerJer: but yes right now i am stuck where I want to build a solution to use pincodes for password of a user and use IAX2. pincodes can be 10,000 and can't do authentication with it
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16:44.01JerJerTili: radius will further complicate that process
16:44.04*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
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16:44.17*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
16:45.30*** join/#asterisk Moc_ (n=mochouin@207-199-0-72-ppp.3menatwork.com)
16:47.06TiliJerJer: I am not using radius for that. But in my case the problem lies in the core of protocol. I cannot use challenge based auth as i dont know which password/pincode is going to be used by user.
16:47.45*** join/#asterisk teapot (n=tandrews@mail.grok.org.za)
16:48.00Tilinormally we have password for each user and we can use MD5(challenge+password) to compare with what client sent. but doing this for 10,000 or more pincodes is just not possible for each registration request.
16:51.41*** join/#asterisk elriah (n=jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net)
16:51.55elriahHi all.  Anyone running the 1.0.7 package that comes with debian 3.1 (sarge)?
16:52.45elriahAnd on a scale of 1 to 10, how stable is the 1.2 beta?
16:53.33ScaredyCat-54.4609
16:53.41elriahHeh.
16:53.51elriahReally?  That bad?
16:53.59ScaredyCateh?
16:54.13elriah<elriah> And on a scale of 1 to 10, how stable is the 1.2 beta?
16:54.14ScaredyCatoh no I was adding sommat, needed a reminder :)
16:54.27elriahahh..
16:54.55ScaredyCatbut it's a beta anyho.. so I wouldn;t use it in production
16:54.58gniretarwhat is the best asterisk management console program out there?  I am willing to pay if necessary.
16:55.08ScaredyCatI am :)
16:55.32ScaredyCatyou can have me 3 hours a day for 300$
16:55.32elriahI'm having a problem getting the 1.0.7 debian * package to start on boot.  I've added the asterisk user to the dialout group, no luck ...
16:56.13JerJerTili: smells like you need to re-think your system design
16:57.07ScaredyCatmd5 - how passe
16:58.55*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
16:58.59mog_homedrumkilla!!!!!!!!!!
16:59.10TiliJerJer: i cannot see other way. i need to use pincodes as password for a user. do u have any suggestion? how do people do calling card systems normally.
16:59.16Tilihi ScaredyCat
16:59.23*** join/#asterisk secure75 (n=mic@p549A0D19.dip0.t-ipconnect.de)
16:59.54*** join/#asterisk myiagy_ (n=myiagy@200.138.215.78)
17:00.48JerJerTili:  nothing like you are talking about
17:00.53elriahI'm considering switching from debian to something else.  What's the preferred distro to run * from?
17:01.14DarthClueanything but windows
17:01.16ScaredyCatlo Tili
17:01.31elriahDarthclue; what distro do you use?
17:01.52TiliJerJer: i guess i should then change some iax2 code. i am already by passing chan_iax2 auth and doing it my centralised way.
17:02.01JerJerwonderful
17:02.03elriahI want to run FreeBSD, but I have a few apps that don't compile ...
17:02.05JerJernothing like yet another hack
17:02.07Tilithanks
17:02.27JerJergood luck supporting it
17:02.28Tilihey ScaredyCat: is there any solution to my problem.
17:02.31ScaredyCatisn't that called innovation
17:02.38Tiliyeah i badly need it
17:02.47ScaredyCatTili: I'm not sure what youre after ...
17:02.53JerJerhe doesn't even know
17:02.54ScaredyCat/msg me.. :D
17:02.55DarthClueelriah: bsd isn't the greatest thing for asterisk based on the issues associated with getting some of it working.  i use fedora but that is just a preference and not a recommendation
17:03.10coppiceScaredyCat: you are thinking of lexmark. only they offer innovation
17:03.15TiliScaredyCat: my case is simple. i want user to either enter password or pincode of calling card as password for registering.
17:03.23*** join/#asterisk azrishahril (n=azrishah@60.50.193.179)
17:03.58ScaredyCatahh ok... but why can't us just make their password their pincode and use realtime?
17:03.59JerJervery trivial:  each customer must have a globally unique id
17:04.32bkw__Radius solves the simultaneous port issue across many boxes
17:04.40JerJerlol - sure
17:04.45TiliScaredyCat: a user at any one time may have more than one pincode. they may buy a 10 dollar card and then 20 dollar card. each with its own expiry time.
17:04.56ScaredyCatahhh got it..
17:05.07JerJerthen write a calling card app
17:05.10JerJervery trivial still
17:05.22JerJerdon't authenticate with IAX2 - authenticate with your application
17:05.26JerJerproblem solved
17:05.35ScaredyCatlike a pay as you go mobile Tili...
17:05.37TiliJerJer: you mean like astcc, where we play prompts to enter pin code etc
17:05.38*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
17:05.49ScaredyCatastcc sucks
17:05.51JerJerthat is only one way
17:05.57JerJerand agi does not scale
17:06.05TiliJerJer: problem is that my client already had some solution based on H323 with pincodes as password.
17:06.12JerJersucks
17:06.21JerJerdon't fix it then
17:06.32TiliScaredyCat: yes astcc sucks. i would write my own app or extend the one i wrote last year which was extended more by you.
17:06.36*** join/#asterisk razu_ (n=razu@ip192.cab63.mus.starman.ee)
17:06.39coppicekettles scale very well :-)
17:06.45gniretarhow do i configure asterisk to communicate with other asterisk servers using IAX?  I see how to connect to a remote computer but how to set up the asterisk box that is being connected to.  I know i should RTFM but i cant find the correct document page.
17:07.56TiliJerJer: i am authenticating my own way for registrations. problem is authentication while making call from endpoint.
17:08.11JerJerthen authenticate with an application
17:08.14JerJervery simple
17:08.23JerJerdump calls into a general context
17:08.36JerJeruse an application to provide whatever authentication means you require
17:09.18*** part/#asterisk TripleFFF2sdf (n=TripleFF@modemcable131.156-131-66.mc.videotron.ca)
17:09.22TiliJerJer: oh yeah that is nice. so we tell IAX2 protocol that user is authenticated but at same time we run our own authentication in parallel
17:10.06JerJerblah
17:10.12JerJertalk about over-complication
17:10.42ScaredyCatTili: create chan_iax3 :)
17:10.57JerJerwhy force your customers into the whole calling card bullshit?   Give them a user account and let them use it and recharge it
17:11.09ScaredyCatroflmao
17:11.30JerJeror are you purposely attempting to take money from them?
17:11.33Goshengniretar: I will help you in private chat
17:11.47TiliScaredyCat: well my chan_iax2 is customised already. dumping all users to one default user.
17:12.29TiliI agree with JerJer: may be we should allow users to recharge their account and add money to it. just like cellular companies do. this way, i am out of this terrible situation.
17:12.46TiliJerJer and ScaredyCat thanks a lot.
17:12.54ScaredyCatI thought that's what you were doing?
17:13.04coppicewhat's wrong with buying a calling card in a 7-11, calling in, activating it, and generating a new account on the spot if needed.
17:13.17ScaredyCatyou meant you were getting them to pay say 10 euro for a card and it expirres when they hang up?
17:13.24Tilicoppice: each user is given a callerid, a virtual VoIP number.
17:13.36lilalinuxis there an irc channel for misdn?
17:13.44ScaredyCat#misdn
17:13.51netsurferdoh
17:13.54JerJercoppice:  nothing, when the provider wants to purposely rape their customers
17:14.00TiliScaredyCat: no it expires after sometime like 30 days 60 days etc.
17:14.13JerJerpre-paid service should not expire
17:14.28ScaredyCatok, but if they buy another card, it replaces their previous one - ie the credit is lost?
17:14.42Tiliyeah i guess i am thinking too much. i will now start fighthing for user rights in pre-paid
17:15.22TiliScaredyCat: in original plan, no credit is lost. its just that you specify which card you are using so that we deduct amount from that and not from other card u have
17:15.24*** join/#asterisk meppl (n=mephisto@p54AADB0A.dip.t-dialin.net)
17:15.31TiliScaredyCat: credit is never lost
17:15.43ScaredyCatahh ok, so it's not like topping up your credit...
17:15.44lilalinuxwell #misdn is empty
17:15.53*** join/#asterisk pr0m (n=pr0methe@2002:184b:c446:13:0:0:0:50)
17:16.01ScaredyCatwith the card...
17:16.09ScaredyCatsounds overly complex...
17:16.26ScaredyCatrather than just added the appropriate credit to the account..
17:17.22weazulhi guys why is there so less info about asterisk and radius?
17:17.23ScaredyCatso when I buy a card off you I get a temporary number... which dies when that credit runs out?
17:17.42ScaredyCatJerJer deleted it all weazul
17:17.57puzzledlol
17:18.06redder86is there an "unregister" for IAX2 protocol?
17:18.15TiliScaredyCat: no. you register with me and get a voip number. u buy cards to make calls as long as card has money in it.
17:18.36puzzledisn't that what astcc does?
17:18.37TiliScaredyCat: we keep record of each card's balance.
17:18.47ScaredyCatok, so the credit applies to a number.
17:18.59ScaredyCatwhich means your making it harder for yourself...
17:19.05ScaredyCat.,.. for no apparent reason
17:19.07Tiliredder86: there is register release for IAX2.
17:19.32ScaredyCatpuzzled: it wastes resources and pretends to be a calling card app
17:19.46*** join/#asterisk wolfson` (n=hehe@kdh-res-4.beachlink.com)
17:19.52JerJerScaredyCat:  i don't like false accusations
17:19.56Tilipuzzled: please dont start on astcc. it has already been a nightmare.
17:19.59puzzledScaredyCat: off course it does. how else would the consultants make money :)
17:20.08Tilinevermind.
17:20.12puzzledhehe
17:20.14ScaredyCatwell, i bet you would have done it if you could JerJer
17:20.15Tilithanks a lot for your ideas guys.
17:20.24JerJerScaredyCat: why?
17:20.33ScaredyCatwhy not
17:20.37tzafrir_laptopa question about mysql_cdr if I may: http://voip-info.org/wiki-Asterisk+cdr+mysql , why isn't uniqueid an auto_increment value in the default schema?
17:20.56ScaredyCatit's generated tzafrir_laptop
17:20.58tzafrir_laptopWhat's the overhead for that?
17:21.00JerJerif they feel like blowin the kind of money we did with radius, they can
17:21.15tzafrir_laptopScaredyCat, generated where?
17:21.17puzzledtzafrir_laptop: because it is randomly generated
17:21.31ScaredyCatwithin asterisk... each call has a unique id
17:21.31Tiliredder86: I tried sending IAX_COMMAND_REGREL to asterisk but asterisk didn't seem to care.
17:21.49puzzledtzafrir_laptop: prolly somewhere in asterisk-addons mysql code
17:21.58ScaredyCatno it's in *
17:22.00tzafrir_laptopIs it the channel number?
17:22.04ScaredyCatno
17:22.05Tilibye all. thanks again JerJer.
17:22.09ScaredyCatit's a unique id
17:22.38ScaredyCatbye Tili
17:22.52tzafrir_laptopScaredyCat, what's the benefit for creating it in * rather than a simple auto-increment field?
17:22.56Tilibye ScaredyCat. keep in touch.
17:23.19ScaredyCatbecause not everyone uses mysql
17:23.43tzafrir_laptopeverybody who uses mysql_cdr uses mysql.
17:23.55coppicecertainly not now they are doing deals with SCO :-)
17:24.08ScaredyCatbut not everyone who uses * uses mysql
17:24.19ScaredyCatwhy is that so difficult to understand
17:24.22tzafrir_laptopThat page is about mysql_cdr
17:24.47JerJerthen use the ODBC crap
17:24.59tzafrir_laptopAnd uniqueid in mysql_cdr seems to require a non-default #define
17:25.17JerJeror very simply write your own CDR handler
17:25.25ScaredyCatgood point jerjer...
17:25.27JerJerit can do more than just shit CDRs into a db, ya know
17:26.16puzzledfortunately you forgot to put the batteries in
17:26.36JerJerthe cattle prod can only legally come out at PhreakNIC
17:26.42coppicehe's using those for his vibrator
17:26.49*** join/#asterisk t3t (n=t3t@galley.pangalacticgargleblaster.com)
17:26.58ScaredyCatvibrating chair actually coppice
17:27.29ScaredyCataaaalll ooovvveeerr vvvvviiiiibbbrrraaattiooonss
17:28.33*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
17:28.41tzafrir_laptopWhat's the overhead of creating the uniqueid in * (in the mysql_cdr module) ? and in mysql itself? seems that in mysql the overhead is lower
17:30.20harryvvGot a complaint about callwaiting/callerid. Seems that when a caller is on the phone talking to somone and thay hear the beep that there is a second person calling in there callerid does not show up on the display of the analog/ata. On the other hand if thay are not on the phone the callerId does show up when calling in. Any ideas what may cause this?
17:30.49ScaredyCatcalleridcallwaiting
17:31.01ScaredyCator callwaitingcallerid
17:31.03harryvvthats set to yes
17:31.03t3ttzafrir_laptop, the overhead is probably pretty similar (assuming the * code is relatively efficient and doesn't query the last ID every time it needs a new one)
17:31.23harryvvcallwaitingcallerid=yes in zapata
17:31.25t3ttzafrir_laptop, the more commonly accepted way would be to allow MySQL to do it
17:31.52ScaredyCatit'll only work if the device supports it too harryvv
17:32.23ScaredyCatt3t, and then you'd lose the uniqueid in * ..
17:32.25harryvvyou mean the ATA by sipura?
17:32.30ScaredyCatyes
17:32.36ScaredyCator the actaul phone
17:32.38harryvvokay then will have to look into it.
17:32.44ScaredyCat/s/or/and/
17:32.46harryvvphone does support it.
17:32.50t3tScaredyCat, what do you mean by 'lose'?
17:32.51harryvvworks with it off asterisk
17:33.01ScaredyCatok, so I guess it's the ata then
17:33.51Hmmhesaysargh, why am I getting ringback one minute and no ring the next
17:34.04t3tharryvv, what voip-analog adapter are you using?
17:34.13harryvvspa 1000
17:34.26ScaredyCatt3t since the internal unique id wouldn't particularly match your db version... which means any investigation into problems would be hampered
17:34.31harryvvI just looked up the 1000 specs is does support it mabey just turned off.
17:34.47t3tharryvv, hmm... don't know where to turn it on for that one...
17:35.01redder86Tili: thanks
17:35.29t3tScaredyCat, oh I didn't realize that * made the uniqueID avaliable in the logs
17:35.45*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
17:36.05ScaredyCatsince * already generates it why bother generating another
17:36.28*** join/#asterisk ManxPower (n=eric@69.149.125.137)
17:36.49*** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net)
17:37.11t3tScaredyCat, because that's traditionally a function of the db and since the DB is generating it, it's guaranteed to be unique...
17:37.45t3t...this could be worked around by just adding another column called recID that is auto_increment by the DB and have * fill in uniqueid as well
17:37.49ScaredyCatbut you'll need to change the local unique id to match then
17:37.56ScaredyCatlocal=asterisk
17:38.35t3tScaredyCat, I didn't realize that * used uniqueid for anthing other than storage in the table.  where else is it used?
17:38.36ScaredyCatwhy would *'s unique id not be unique?
17:38.53ScaredyCatit's a channel var too... ${UNIQUEID}
17:38.59ScaredyCatused to identify a call..
17:39.04ScaredyCatuniquely
17:39.44*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
17:39.55ScaredyCatyou're right you could add both to the db, but that's not decreasing the overhead...
17:40.09*** join/#asterisk Romik (n=romik_@212.143.5.146)
17:40.21t3tScaredyCat, oh... then the best way would be to create a different column in * for the record identifier if you needed a way to reference specific rows and allow * to fill in uniqueid
17:40.59ScaredyCatbut you're just replicating uniqueid so why waste your time
17:41.19ScaredyCatbesides any table your already have an id field that was uniuq
17:41.20ScaredyCate
17:41.33ScaredyCatyuor=should
17:41.48t3tScaredyCat, because if something goes wrong with * ability to generate a truly unique value, you could end up with problems depending on how you use the information
17:41.49*** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net)
17:42.00elriahAny major issues with running 1.0.7, I want to use the debian stable packages..
17:42.48ScaredyCatI presume you mean to run billing on your cdr's t3t
17:43.17ScaredyCatno elriah
17:43.20t3tScaredyCat, any system that needs a unique value in each row
17:43.34ScaredyCatwhcih is the id field...
17:43.47*** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no)
17:43.48ScaredyCatwhich every table should have anyway
17:43.50t3tI didn't see that in the table def...
17:44.14ScaredyCatif it's not there someone deserves a slap
17:44.35t3tslap away... that's why it confused me
17:44.45*** join/#asterisk bonez39 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net)
17:45.15t3tthis was the suggestion for db autonum uniqueid in the wiki:
17:45.16t3tuniqueid integer PRIMARY KEY auto_increment
17:45.39t3tthat's a pretty poor choice if uniqueid is a call identifier
17:46.00ScaredyCatbut it doesn't know wat it is...
17:46.12t3texactly
17:46.30t3tI assumed it was just an integer that either * or the db would increment
17:46.51*** join/#asterisk oej (n=Olle@apollo.webway.se)
17:46.58ScaredyCat* has nothing to do with it. it uses the unique id, which is a text field
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17:47.37*** part/#asterisk manji (n=manji@pom51-1-82-226-145-244.fbx.proxad.net)
17:47.50ScaredyCatyou could combine use unixtime+destination+srcchannel+destchannel+somehippievalue
17:49.01*** join/#asterisk neonkl (n=amit_cho@210.195.36.72)
17:49.16harryvvanyone here know if the ata3000 can ring for say 10 seconds before it can switch over to the asterisk box and play the ivr?
17:49.38t3tOk, then the advice in the wiki is way off for making a uniqueid auto_increment
17:49.43RoyKhardwire: does the call come from an asterisk server?
17:50.10t3tharryvv, that should be a function of your extensions.conf settings, not the ata
17:50.15*** join/#asterisk burnproof (n=burnproo@210.213.199.216)
17:50.40burnproofgood day ppl, can i ask a newbie question since i'm really relavitely new to asterisk
17:50.56ScaredyCatt3t, got the wiki link?
17:50.57elriahWhy is * voicemail called 'comedian mail'?
17:50.58burnproofwhat is my minimum requirements to create a voip gateway
17:51.18harryvvt3t...im trying to avoid that in this case. the outside complex audio control panel kills all dtmf digits once the condo # is called ...ie, you attempt to press a extention the audio control panel shuts off as a safety consideration.
17:51.18t3thttp://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql
17:51.19*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
17:51.21RoyKburnproof: a 486 with 8MB RAM
17:51.35*** join/#asterisk ytk (n=aaa@82.211.129.231)
17:51.45tzafrir_laptopt3t, thanks
17:51.55burnproofRoyK: how about digium cards, do you think x100p would be enough just to make some experiments
17:52.00ScaredyCatt3t - oh dear...
17:52.17tzafrir_laptopelriah, parodying Meridian Mail
17:52.28elriahAhh...
17:52.40mog_home^_^
17:52.42RoyKburnproof: just take a p500+ and stuff some card into it. it'll do for testing/experimenting
17:52.55RoyKmy initial answer was rhetoric
17:52.57t3tScaredyCat, tell me about it.  it definitely led me astray... I feel so betrayed
17:53.04ScaredyCatlol
17:53.45*** join/#asterisk ytk (n=aaa@82.211.129.231)
17:54.04RoyKburnproof: please mind that AFAIK the x100p is depreciated
17:54.06t3tScaredyCat, it doesn't make sense that a compile-time define has to be made to enable the storage of uniqueid though... that threw me off too...
17:54.10RoyKburnproof: don't you have isdn?
17:54.28ScaredyCatt3t your right - it should be in there anyway.....
17:54.31JerJert3t: it was added after the code was already widely deployed without a unique id
17:54.41ScaredyCatbut then I use a custom module anyway
17:54.50t3tharryvv, can you describe what's hooked up to what in your setup
17:55.02burnproofRoyK: sorry bro. but i don't have access on ISDN cards
17:55.14RoyKcards? or lines?
17:55.31burnproofRoyK: what i try to accomplish from soft phone i wan't to make a call from soft phone to pstn
17:55.43elriahI've gone through the wiki and googled til my fingers ache, but I'm having echo issues with my test system.  I am on a pots line with a X100P ($15.00/ebay).  I have several cards and have swapped them out.  Think it's in the copper?  I do have DSL on the same line I'm testing with.
17:56.09t3tJerJer, good to type to you again... that makes sense.  no need to break the logging for the early adopters.  It should probably be the default eventually though.  Call records without an identifier get useless pretty fast in a busy environment
17:56.45ScaredyCatit should have been there from the start...
17:56.55burnproofRoyK: how about tdm400p with a couple of fx0 modules to begin with ?
17:57.16t3tdefinitely, but now it's a member of the 'demon of backwards compatibility' club
17:57.33RoyKburnproof: if you can afford it, why not :)
17:57.45ScaredyCatyes... but why were people not useing a unique id anyway...
17:57.56ScaredyCator did they all gerneate their own :)
17:57.56*** join/#asterisk newl (n=newlook@203-59-168-152.perm.iinet.net.au)
17:58.02t3tScaredyCat, I assume that it was just forgotten at first
17:58.04RoyKburnproof: but if you've got an isdn line, a cheap hfc-pci card is prolly better and far cheaper
17:58.09SwKits fxo not fx0
17:58.13RoyKs/better and //
17:58.24RoyKPakiPenguin: failover trunk? how?
17:58.34ScaredyCatnellie the elephant
17:58.35RoyKPakiPenguin: you mean failing over active calls?
17:58.48PakiPenguinRoyK, one trunk gives 503 , channel busy , so we send the call to second trunk
17:59.03ScaredyCatjust use 2 dials then
17:59.20RoyKPakiPenguin: ah. you mean you send the call from asterisk to a SIP trunk and then get a 503?
17:59.26harryvvanyone here using a ata1000?
17:59.29burnproofRoyK: what do you think the cheapest except for having an ISDN+hfc-pci card ?
17:59.34t3tScaredyCat, the WIKI should deinitely be updated.  If it's a new install of cdr_mysql then MYSQL_LOGUNIQUEID should be defined.  If not, the DB should be updated and MYSQL_LOGUNIQUEID should be defined.  The "Alternative" method needs to go.
17:59.36RoyKPakiPenguin: isn't that similar to CHANUNAVAIL?
17:59.41ManxPowerWhy not just check the status of HANGUPCAUSE or DIALSTATUS?
17:59.53RoyKburnproof: el cheapo x100p
17:59.59elriahI've gone through the wiki and googled til my fingers ache, but I'm having echo issues with my test system.  I am on a pots line with a X100P ($15.00/ebay).  I have several cards and have swapped them out.  Think it's in the copper?  I do have DSL on the same line I'm testing with.
18:00.05RoyKburnproof: not digium's stuff, though
18:00.08*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
18:00.16ScaredyCatt3t tbh, there's no reason not to change it now... it's just making things worse
18:00.19SwKpakipenguinL check the DIALSTATUS then proceed
18:00.27*** join/#asterisk zoo (i=nobody@ip-11-16.travedsl.de)
18:01.11RoyKPakiPenguin: exten => s,2,NoOp(dial status is ${DIALSTATUS} and hangupcause is ${HANGUPCAUSE})
18:01.12RoyKor something
18:01.17burnproofRoyK: anyway, i think i don't have any choice but to buy tdm400p with a couple of fxo modules :), since AFAIK x100p is a clone with original digium card and it make trouble for somebody !
18:01.37PakiPenguinhmms
18:01.43PakiPenguini see RoyK
18:01.45burnproofRoyK: thanks
18:01.46PakiPenguinlemme working on it
18:01.46RoyKburnproof: if you can afford it, the tdm400p is a good choice
18:01.47PakiPenguinthanks
18:02.11burnproofThanks everyone :) bye
18:02.46elriahWhich is the analog line?  FXO or FSX?
18:02.55ScaredyCat!!!
18:02.56t3tboth
18:03.02ScaredyCatthey're both analog
18:03.04ScaredyCat!
18:03.10*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
18:03.17t3t... and that's fXs, not fSx
18:03.18elriahI know, but which one is the phone line that  you plug into the wall?  FXO or FXS?
18:03.28t3to
18:03.35elriahThanks.
18:03.42ScaredyCatfxo (the o=office)
18:03.51ScaredyCatfxs (the s=station)
18:04.00elriahAhh.. that will be easy to remember..
18:04.37ScaredyCatlol t3t
18:04.48elriahI'm 12, cut me some slack.. What's signaling?
18:04.48harryvvI made one minor change to my sipura ata and now the analog/ata will not ring with pstn ivr calls. I switched back the ata setting and still will not ring it. When using ext 200 to ring exten 201 the analog/ata It will ring it. Getting this message.  AGI Script dialparties.agi completed, returning 0
18:04.48harryvv<PROTECTED>
18:04.48harryvv<PROTECTED>
18:04.48harryvv<PROTECTED>
18:05.07*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
18:05.10RoyKelriah: basic input output, as in sex
18:05.16bkw__elriah, you're 12?
18:05.17*** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com)
18:05.19ScaredyCat"hello, Mr Bush...yes fine thank you. Now about my mansion in New Orleans"
18:05.21t3tharryvv, is the ata behind a nat firewall?
18:05.27harryvvohh wow, now 200 cannot ring 201
18:05.40t3tharryvv, it sounds like it lost it's registration to the server
18:05.44ScaredyCatbugger, 12 year olds encroaching!
18:05.57elriahYea, I'm building a phone system for my school.  For class and they'll use it if I can prove it will work.  I got them to buy two Polycom IP 300 phones and some server components..
18:06.03harryvvt3t...everything works 100% I was trying to make a cidcw change in the spa100 and changed it back. This is a new problem.
18:06.21bkw__elriah, need some help?
18:06.26ScaredyCatshouldn't you be outside playing football or something elriah
18:06.30ScaredyCatstop it bkw_!
18:06.32bkw__ScaredyCat, I never did
18:06.36ScaredyCatperv
18:06.39harryvvt3t. i can access vm with ext 201
18:06.41bkw__ScaredyCat, be nice
18:06.44harryvvthe effected phone.
18:06.45bkw__i'm just trying to be helpful
18:06.54ScaredyCatyou never played football bkw_?
18:07.00ScaredyCator never played outside
18:07.06bkw__both
18:07.12PakiPenguino_0
18:07.18ScaredyCatoh my...
18:07.36PakiPenguinouch
18:08.12file[laptop]weirdo
18:08.17fulgashumm
18:08.27fulgasanyone knows with asterisk register with sip:s@ip ?
18:08.36fulgas*why
18:08.38elriahThanks for the offer, Bachman.  I'm getting there.
18:08.38bkw__fugitivo,
18:08.41t3tharryvv, can you describe what's hooked to what and how (phone - ata - switch - nat - *, etc.)
18:08.42bkw__because you didn't read the docs
18:08.42ScaredyCat^---taken the ball home
18:08.50bkw__register => user:pass@host/context
18:08.53bkw__er contact
18:08.54file[laptop]fulgas: because you didn't specify what to put in the contact on the register line?
18:09.00bkw__if you don't specify the contact it will use s
18:09.00PakiPenguinsomeone stole my dvdrw from the laptop:(
18:09.06*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
18:09.11harryvvdialparties.agi: Extension 201 has call waiting disabled
18:09.13shido6wow
18:09.21ScaredyCatsnow
18:09.26harryvvis that in asterisk that its disabled? its not disabled in zapata~
18:09.27harryvv!
18:09.33netsurferhi harryvv
18:09.36harryvvhi
18:09.49*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
18:09.50*** join/#asterisk JunK-Y_ (n=junky@HSE-Montreal-ppp141551.sympatico.ca)
18:09.59file[laptop]harryvv: it's disabled in AMP...
18:10.01elriahSo if I get a TDM400P and ditch my X100P's, I might get past my echo problem?
18:10.11ScaredyCatyou might...
18:10.12bkw__elriah, doubtful
18:10.24harryvvahhh
18:10.25harryvvokay
18:10.29elriahThink it's in the wall then?  Something up with my phone line?
18:10.47ScaredyCatdepends... those x100p's a re poop
18:11.01bkw__elriah, they might be reversed
18:11.04bkw__tip and ring that is
18:11.05elriahYea, but at $15.00, easy to get started with ...
18:11.23ScaredyCatyeah, they used to be $99
18:11.25t3tScaredyCat, mine works pretty well unless it has to figure out whether a call's been hung up...
18:11.32elriahbkw: Not sure I understand..
18:11.37elriahbkw: What might be reversed?
18:11.43elriahMy copper pair?
18:11.44harryvvfile, again, call waiting beep works..just does not display the cid on the display when called party is on the phone.
18:11.44bkw__elriah, you need to get a telecom book
18:11.53bkw__elriah, yes you flip the wires
18:11.55elriahI have the big yellow * book ...
18:12.01ScaredyCatoh dear
18:12.02*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
18:12.06t3tharryvv, I'm pretty sure that cw cid is controlled by the ATA and not *
18:12.18elriahbkw: So for some simple testing, I can just reverse a pair and try it?
18:12.19ScaredyCatnot the signate one I hope
18:12.20file[laptop]t3t is rrrrright yay
18:12.28elriahYea, signate.  that's it.
18:12.30fulgasthk :)
18:12.35elriahOnly one I could find.
18:12.36ScaredyCatoh well...
18:12.39*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
18:12.43t3tthanks, file
18:13.07*** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk)
18:13.12ManxPowerin SIP pretty much everything is controlled by the ATA, which is why getting a good one is so important
18:13.35ScaredyCatt3t what do you mean it has to work out if the call's been hung up?
18:13.38harryvvt3t but would the ata send feedback to asterisk that it is disabled?
18:13.59file[laptop]if call waiting is disabled on the ATA it would send back a Busy Here probably, or Temporarily Unavailable
18:13.59harryvvI personally would not think so.
18:14.14ManxPowerBusy here is what the SIPura and polycoms send back
18:14.18bkw__elriah, the signate book?
18:14.27harryvvfile...mabey thats why im now gettign this problem of caller not avaible.
18:14.36bkw__please go buy a real telecom book... that signate book is a J O K E
18:14.51file[laptop]need the Oreilly one which isn't yet released
18:14.54ScaredyCati bet he's not laughing
18:15.06bkw__that signate book is aweful
18:15.14elriahYea, signate.
18:15.25ScaredyCatit was out of day the day it was release...
18:15.26ScaredyCatd
18:15.47ScaredyCatand the author knew nothing of sip forking, which says a lot.
18:15.56elriahSince the OReilly book isn't released, what's a good reference?  I use voip-info a lto.
18:15.58elriahlto = lot
18:16.10bkw__yes voip-info is better than the signate book
18:16.41mog_homevoip-info is awesome
18:16.57elriahGotta run, thanks for all the help.
18:17.00ScaredyCatit is, just a little slow at times
18:17.06t3tharryvv, it probably wouldn't tell * that cid cw is disabled
18:17.22*** join/#asterisk roulduke_ (i=2vuxjy3b@p508D246B.dip0.t-ipconnect.de)
18:18.20elriahOh, one more question before I go.  I'm using debian 3.1 and the * package 1.0.7.  It doesn't start on boot automatically even know I've told it to in /etc/defaults/asterisk.  I've added asterisk to the dialout group per the wiki but still no luck.  Any suggestions?
18:18.31elriahIt starts fine when I log in and just run it.
18:18.32t3tScaredyCat, my 100p doesn't detect pstn hangups until the eeee-eeee-eeee--eeee-eeee from the telco
18:18.53*** join/#asterisk amir_ (n=amir@shield.guindehi.ch)
18:19.17harryvvahh weird now 200 can ring 201
18:19.24ScaredyCatahhh t3t
18:20.04*** join/#asterisk pino (n=ep@host45-28.pool21345.interbusiness.it)
18:20.10manyelriah: what does /etc/init.d/asterisk start say?
18:21.00*** join/#asterisk ido-2 (i=wtf@85-65-229-139.barak-online.net)
18:21.03t3t/me puts his money on ' No such file or directory
18:21.23elriahStarting Asterisk: asterisk ... Doesn't work that way either... but if I just type asterisk it works fine.
18:21.32ManxPowerelriah, "asterisk -cvvv"
18:21.38ido-2any C++ developers who might be able to help me with tone detection in a pcm/amr audio wave file ?
18:21.55manyelriah: hm. maybe some exit 0 or asterisk in wrong $PATH?
18:22.05elriahYea, that works fine.  I just can't get debian 3.1 to start it for me with either /etc/init.d/asterisk or at boot .. i guess it's a problem with the init script, but don't know how to troubleshoot it.
18:22.16elriahWhere would init scripts log to?
18:22.26manyelriah: $EDITOR /etc/init.d/asterisk
18:22.29manythey dont log at all.
18:22.45RoyKelriah: see logger.conf
18:23.14RoyKelriah: if logging verbose, they'll happen to be logged by the logger
18:23.27t3telriah, try this then: cd /etc/rc2.d;ln -s ../init.d/asterisk S95asterisk
18:23.33elriahWhat am I looking for in the script, many?
18:23.46harryvvvery very odd. Shows that 201 is being rung in the cli for zap dial in but it does not ring saying person is unaviable. now it will ring with 200 rining this phone.
18:23.56manyelriah:  "exit 0"  or if [ -f
18:24.12*** part/#asterisk ido-2 (i=wtf@85-65-229-139.barak-online.net)
18:24.37t3telriah, scratch that... i missed the part about not being able to start * from the init script
18:24.54*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
18:25.00t3t... or can you?
18:25.46elriaht3t, no.  Doesn't start with the init script.
18:26.12t3tdoes it start if you call the script manually with /etc/init.d/asterisk start
18:26.15ManxPowerI thought the init scripts were for redhat/mandrake and their ilk.
18:26.17elriaht3t; nope.
18:26.23elriahOnly starts if I just run it.
18:26.42ManxPowerelriah, just put "asterisk" at the end of /etc/rc.d/rc.local
18:27.34elriahManxPoewr: Isn't that the 'wrong' way to do it?  Someone told me that everything should be starting with the init scripts ...
18:27.36harryvvwho has a spa ata 1000 would care to compare settings with me?
18:27.36t3tManxPower, sure... the simple, disgusting way to do it :)
18:28.17elriahOh well.  I'll look at it later.  Thanks all.
18:28.42t3telriah, stop *, note the time on the machine, try to start it from the init script and then look at /var/log/asterisk/messages
18:28.51elriaht3t; will do.
18:28.52ManxPowerelriah, Yes, it's the "wrong" way to do it.
18:28.59ManxPowerbut it will work
18:30.30elriahThanks ManxPower, I'll keep that in mind if I can't fix it.  Later all.
18:31.12*** join/#asterisk _-_ (n=nabudoco@200.76.231.14)
18:31.43bkw__I still find it funny people fail to understand that g723 != g723.1
18:32.11darkskiezDuh! isnt that obvious!
18:32.14harryvvI bet mabey i pressed a number on the phone that put the phone in unuvaible mode.
18:32.28*** join/#asterisk amir (n=amir@shield.guindehi.ch)
18:33.30t3tharryvv, is the ATA connected directly to your * box, through a switch, or by some other means?
18:34.01t3tbkw__: what's the difference besides the .1?
18:34.55bkw__g723+g721 became g726
18:35.01bkw__16,24,32,48k
18:35.07mog_homeits .1 better
18:35.09t3tg...
18:35.20bkw__and to confuse people the ITU gave the truespeech stuff g723.1
18:35.26opus_i don't understand this. i find a way to segfault asterisk, and its not considered a bug
18:35.34FuzzyCatlol
18:35.39bkw__opus_, I seen that
18:35.45bkw__and its somethign that can be fixed
18:35.46t3topus_, don't you know anything... that's a feature
18:35.48mog_homewhat does it do opus
18:35.52FuzzyCatis that your macro calling a macro problem opus_
18:35.55mog_homei mean what did you do
18:35.59bkw__calling a macro from within a macro
18:36.00*** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com)
18:36.05bkw__its a bug
18:36.09t3tit's a feature
18:36.10mog_homeyeah
18:36.14bkw__no its a bug
18:36.16FuzzyCatwhat was the bug number opus_ ?
18:36.17t3tfeature
18:36.22mog_homesounds like a bug to me
18:36.23*** join/#asterisk nagl (n=nagl@137.208.4.172)
18:36.32FuzzyCatundocumented feature == bug
18:36.33opus_FuzzyCat which one
18:36.35t3tok, fine. it's a bug, but it wasn't my fault
18:36.36*** join/#asterisk ahecker (n=ahecker@p5497D438.dip.t-dialin.net)
18:36.37FuzzyCatyou one
18:36.41FuzzyCatyour one opus
18:36.49aheckerhello
18:36.55opus_<http://bugs.digium.com/view.php?id=5114>
18:36.56t3tFuzzyCat, it's documented now, right?
18:37.10opus_MikeJ are you around
18:37.18aheckerhow can i change the udp port, asterisk ist listening on? i like to use it with ser
18:37.22FuzzyCatno.. it doesn;t exist apparently :)
18:37.31bkw__it should error out and not let you loop a macro
18:37.36bkw__if we are already in a macro
18:37.39bkw__don't let it call another one
18:37.47bkw__or have a macro level param
18:38.02t3tahecker, sip.conf ; bindport=<portno>
18:38.14opus_what if I want a macro that calls itself, actually I wrote one and this is how I found the bug
18:38.24aheckerit allway binds to 5060
18:38.37FuzzyCatjust have a max depth for looping macros
18:38.41t3topus_, recursion is usually dangerous
18:38.55bkw__FuzzyCat, exactly
18:39.11bkw__ahecker, do you not even bother looking at the config samples?
18:40.15aheckerok, i am silly, i always used port=   :-) thank you
18:40.15t3tbkw__: I think we should send him to * bootcamp... ahecker, what's your CC#?
18:40.44fileconfig samples are for idiots
18:41.03opus_j/k
18:41.12aheckerdoes anybody have experience in combining ms lcs ser and asterisk?
18:42.46t3ti have enough experience with MS to know that it probably won't work the way you think it will...
18:42.48*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
18:44.22aheckeri know that the problem is that lcs uses tcp and asterisk uses udp. i want to connect both via ser, i think it could be possible
18:45.28*** join/#asterisk Assid (n=assid@203.115.64.62)
18:46.18Assidheya
18:46.31aheckernoting about lcs?
18:47.03*** join/#asterisk grimse (n=grimse@p5481C30A.dip.t-dialin.net)
18:47.06Hmmhesaysok sudo is pissing me off
18:47.15Hmmhesayssudo -v
18:47.43Hmmhesaysnow I can't grab asterisk off the cvs cause it won't let me create an asterisk directory
18:48.15aheckerdid you try to conect the 3 systems in any way?
18:49.54Hmmhesaysi'm going to charge them extra just for making me use sudo
18:50.06t3tahecker, before you try to convert udp to tcp and back again, check to make sure that the sip messages and rtp streams are compatible with each other...
18:50.18t3tahecker, see this for one example: http://lists.digium.com/pipermail/asterisk-users/2004-December/077588.html
18:51.40jalsothi
18:53.01jalsotis there a way to get know why is an iax client unregistered from asterisk? I'm getting after some time bulk unregistration (from 24 clients 20 unregisters in some seconds and later they register again)
18:53.15droothanyone here available to be an asterisk tech for a day?  Will pay $$$
18:53.27ManxPowerjalsot, are they behind nat?
18:53.33jalsotManxPower: no
18:53.40jalsotManxPower: on the same LAN
18:53.47dudesdrooth - what are you doing?
18:54.58t3tjalsot: how are they connected to the * server?
18:55.46ManxPowerjalsot, perhaps you have a duplex mismatch.
18:55.55jalsot:o
18:56.03aheckerok, i read this. i am not an expert, but (please correct me), could it be a client problem?
18:56.06*** join/#asterisk santiago (n=santiago@63.245.86.231)
18:56.23jalsotok, so.... client machines are connected to one switch, that switch is into another on which we have asterisk
18:56.25ManxPowerI've said it before and I'll say it again.  All Softphones Suck
18:56.47ManxPowerjalsot, Are they unregistered, or just unreachable/lagged?
18:57.03t3tjalsot: so no firewall, router, 'layer-3 switch,' etc. in between?
18:57.04jalsotwhere do I see the difference?
18:57.16ManxPowerjalsot, "iax2 show peers"
18:57.18jalsott3t: no, just some SMC switches
18:57.57jalsotManxPower: I see unspecified in place of IP address
18:58.58ManxPowerjalsot, what IAX2 devices are you using?
18:59.15jalsotiaxComm
18:59.24ManxPowerI've said it before and I'll say it again.  All Softphones Suck
18:59.27jalsotmore precisely a modified iaxComm
18:59.41jalsotbut there are only visual changes, no functional change
19:00.08jalsotI did a packet dump and in ethereal I see: malformed packet
19:00.28*** join/#asterisk newlook_ (n=newlook@203-59-168-152.perm.iinet.net.au)
19:00.38jalsotin the response going back from asterisk to the softphone
19:01.24ManxPowerjalsot, can you try different switches.
19:01.32ManxPowerSMC are not exactly known to be great.
19:02.36aheckerok, i read this. i am not an expert, but (please correct me), could it be a client problem?
19:02.54*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
19:03.01FuzzyCatrtkgjgcxddhkgfdssssdfghjkljhgfdsfghj
19:03.02*** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
19:03.07t3tjalsot, if you don't have different switches, try pinging the clients continuously and look for packet loss around the time of de-registration
19:03.44t3tjalsot: was the packet dump on the * box or the client
19:04.38jalsotManxPower: I will try to connect asterisk to the same switch as clients are connected
19:04.59Hmmhesayswas chan_features part of asterisk stable?
19:05.19jalsott3t: good, idea, I will ping... I did it already, but not when the case happend [usually ping times are ok and packet loss is zero]
19:05.34*** join/#asterisk vetter (i=vetter@adsl-68-91-7-226.dsl.tulsok.swbell.net)
19:05.34jalsott3t: packet dump was made on asterisk box
19:05.53vetterhey blitz, you around today?
19:06.05vetteranyone ever dealt with aptela?
19:06.38t3tjalsot: then the malformed packet thing is really strange because ethereal should have picked it up before it hit the NIC...
19:07.23t3tjalsot: unless your version of ethereal just couldn't decode the iax2 info that was being sent
19:07.50Spacebaranybody here using ooh323c in production?
19:08.03jalsottethereal 0.10.10
19:08.29*** join/#asterisk intrepidhero (n=briand@67.189.59.49)
19:09.00jalsotNIC is: Broadcom NetXtreme BCM5721 Gigabit Ethernet PCI Express
19:09.31jalsotin a Supermicro server
19:10.40ManxPowerOh, SuperMicro.
19:10.48kajtzusupermicros rock
19:10.52ManxPowerI call those "The Spawn of Satan Boards"
19:11.10ManxPowertheir IDE controllers totally fuck up interrupt latency
19:11.24jalsotManxPower: we have 3ware sata raid in it
19:11.36jalsotso not using on-board IDE
19:11.52kajtzusupermicro and scsi works just fine
19:12.02ManxPowerjalsot, do a google search for site:lists.digium.com 3ware
19:12.08intrepidheroI want to add functionality to the zaptel driver to support a new piece of hardware. Is there some place I can go for documentation? Specifically, I'm trying to figure out how zaptel interfaces with asterisk.
19:13.18jalsotManxPower: thx, looking...
19:13.33jalsotManxPower: what is the short conclusion? use or not use 3ware?
19:13.50ManxPowerjalsot, I don't know.  But read the mailing list stuff
19:13.51doolphSpacebar i do
19:13.52Hmmhesayswhat is pbx_loopback.so?
19:14.23mog_homezaptel interfaces dont work with asterisl intrepidhero
19:14.38mog_homezaptel is the driver layer, then zapata is a translation layer
19:14.41ManxPowerI know the promise RAIDs screw things up.
19:14.46mog_homeso if your driver follows zaptel api
19:14.52mog_homeyour card could talk over it
19:14.58mog_homelike sangoma's does
19:15.03harryvvi was right..some how dnd was activated. what a pain :) now it works.
19:15.10mog_homeor anyother company piggybacking of zaptel drivers
19:16.13intrepidheroithanks mog_home
19:16.17mog_homeno prob
19:17.08jalsotanother interesting thing is that I see on console [after bulk unregistration]: chan_iax2.c:5020 register_verify: Host 192.168.74.140 failed MD5 authentication for ...
19:17.34intrepidherois there someplace where the zaptel api is documented? I mean in an easier to read way than just reading the source.
19:17.41mog_homewell
19:17.52mog_homenot really
19:17.59intrepidhero:-) ok
19:18.11*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
19:18.13intrepidherowas afraid of theat
19:18.16intrepidhero*that
19:18.18mog_homeor not that i am aware of
19:18.30tzafrir_laptopintrepidhero, not really documented anywhere
19:18.32mog_homei dont even think there is any doxygen stuff around
19:18.46mog_homebut if you come here or asterisk-dev
19:18.50mog_homesure you will get answeres
19:19.07harryvvgoing to tear my hair out. ext 201 cannot decide to be unaviable or avaible.
19:19.21harryvvavaiable that is.
19:21.00*** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
19:21.24harryvvim at a loss
19:22.10*** part/#asterisk sergey (i=sergey@195.112.98.13)
19:22.12mog_homewhats wrong harry
19:22.33harryvvWhat setting would emulate a do not distrurb or "the person at extention "number" is unaviable" when obiosly that ext is not being used?
19:22.52mog_homeyou could use a gotoif
19:22.55harryvvi have do not disturbe deactivated even though I never use it.
19:22.57mog_homeand have a db var
19:23.01mog_homeor a var
19:23.07mog_homeand set it if you didnt want a call
19:23.16harryvvmog...i dont use it
19:23.27mog_home?
19:23.33harryvvthis problem crept up when i was trying to enable the cwcid setting.
19:23.50mog_homecwcid?
19:24.00harryvvcaller waiting caller id.
19:24.31lilalinuxdoes anybody here have asterisk with chan_misdn running?
19:24.34harryvvWife was complaing that she does not see the third party calling id number when she is talking to second party.
19:25.41queuetueHas anyone heard anything about fwd's iax2 connections being down?
19:25.57doolphyes
19:26.03doolphactually it was done
19:26.04doolphdown
19:26.20harryvvso much for free
19:26.22harryvv:)
19:26.52queuetueI know it's down - I was asking if anyone has heard why or for how long. :)
19:27.21doolphnop
19:27.46harryvvI just rebooted the ATA now its working properly!
19:28.44harryvvguess everyone is away enjoying the holliday.
19:30.58*** part/#asterisk opus_ (n=opus@dahphish.org)
19:32.48zedkatufharryvvv: just had the same issue myself with my ATA :)
19:33.51zedkatufanyone had this problem: when I dial my asterisk box from elsewhere (eg my mobile or a pstn/landline), I don't hear the "ring, ring" in my earpiece from the phone that I'm dialling from, but my phones at home do ring as normal...
19:34.08zedkatuf(this only started happening recently & I've no idea why)
19:38.49*** join/#asterisk bigger (n=bigger@c-67-174-51-182.hsd1.ca.comcast.net)
19:40.01jalsotManxPower: asterisk box moved to the same switch as clients are on. however after bulk unregistration now they cannot register in.
19:40.25*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
19:40.53jalsotand now somehow they registered again, strange
19:41.11jalsotare you interested in ethereal dump?
19:41.37biggeris this the appropriate forum for a typical newbie question?
19:41.41Vcoewww asteriskat
19:41.56mog_homeyup
19:42.06biggerwhew..
19:44.07biggerI've got 2 pots lines connected to my * box & a grandsream. works fine.. The problem is I have an analog phone connected one of the lines in case of power outage
19:44.22biggerif I use that phone, it works fine
19:44.24biggerbut..
19:44.35biggermy grandstream rings
19:44.46bigger& the whole conversation will run to vm
19:44.50biggerwhy is that?
19:44.52Vcoheh..heh..
19:45.30*** join/#asterisk pbd (n=PDavidso@c-24-15-72-86.hsd1.il.comcast.net)
19:45.51*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
19:45.57*** join/#asterisk newlook_ (n=newlook@203-59-168-152.perm.iinet.net.au)
19:47.36*** part/#asterisk xai (n=pasta@cpe-70-112-17-10.austin.res.rr.com)
19:51.40*** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr)
19:52.52jskcrhy all
19:55.12mog_homewhats that bigger, you have a voicemail box on your lines?
19:55.32*** join/#asterisk hadi57 (n=al_moghr@83.136.8.206)
19:55.34*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
19:58.27*** join/#asterisk syle (n=blah@wnpgmb06dc1-44-164.dynamic.mts.net)
19:58.46*** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox)
20:01.45harryvvim suprised asterisk is not showing a red alert when the phone line is removed.
20:02.13Assidsure it does
20:02.15ManxPowerharryvv, the TDM400Ps don't do that.
20:02.17Assidi tried it earlier
20:02.22harryvvthe x100p do
20:02.27Assidatleast on the x100p clone
20:02.28harryvvit has worked in the past
20:03.21*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
20:05.38Assidhow much is a cheap FXS card?
20:05.59ManxPowerAssid, About $130
20:06.06jskcryou could probably get a sip fone cheaper
20:06.22ManxPowerWell, yes.  a SIPura ATA-1000 is prolly about $60
20:06.30Kattybeep.
20:06.36Assidscrew up is.. getting it down.. i waited too long.. and someone i know is coming tomorrow
20:07.14ManxPowerAssid, I guess it sucks to be you.
20:07.28Assidim in india.. would have saved me shiping
20:08.16Assidwhats better SIPura ATA-1000 / Sipura 841 ?
20:11.14bigger* should only be using this line for outgoing calls, but it seems using a phone sharing the line causes my main extension to ring & go to vm
20:15.20chetgeneral q, not sure if i have the lingo right, if i have incoming call from pots or pri to sip phone will callerid info come? is ani required on pstn trunk?
20:15.26Assidis there any decent handset which has sip+iax?
20:17.58*** join/#asterisk crich1999 (n=crich@port-212-202-0-21.dynamic.qsc.de)
20:18.27jskcrchet: yes caller id will come in
20:18.33*** join/#asterisk criptos (n=criptos@201.135.121.4)
20:18.38criptosHello all!
20:18.55ManxPowerI need a 4-port Digium T-1 card by tomorrow.  Can anyone help me.  Shipping to Baton Rouge LA
20:19.39CaNaBiSManxPower, thats where I am!
20:19.46CaNaBiSI moved here a month ago
20:19.55jskcrCaNaBiS:  that some bad timing
20:19.59ManxPowerCaNaBiS, Baton Rouge?
20:20.13CaNaBiSResolved CaNaBiS (68.111.51.215) to ip68-111-51-215.br.no.cox.net.
20:20.17CaNaBiSyeah, baton rouge
20:20.25ManxPowerAhrimanes, so you are a Cox Sucker.
20:20.36ManxPower..er...
20:20.36CaNaBiSI have a Definity switch and 55 phones if you know someone that needs a switch
20:20.40CaNaBiSits ready to go
20:20.41ManxPowerCaNaBiS, so you are a Cox Sucker.
20:20.48CaNaBiSheh, something like that
20:21.05ManxPowerCaNaBiS, we have everything except our CLEC is down and won't be back for a while, which is why I need a T-1 card.
20:21.22CaNaBiSah, so gonna use IP trunks?
20:21.27ManxPowerCaNaBiS, yeah.
20:21.43CaNaBiSyeah, I cant find my Sipura
20:21.47ManxPowerCaNaBiS, do you have Digium cards available.
20:21.47CaNaBiSmy gf lost it
20:21.55CaNaBiSI have some IP phones here though
20:22.01CaNaBiSif I did, I'd loan em
20:22.07ManxPowerWe have 60 IP phones at another office.
20:22.22ManxPowerI would rather just unplug the CLEC's voice t-1 and plug Asterisk T-1 card into it.
20:22.27ManxPower(into the nortel)
20:22.47CaNaBiSManxPower, where are you located?
20:22.54mrfrenzyManxPower: how much bw do you need for 60 ip phones?
20:23.14ManxPowerCaNaBiS, I'm normally in Waveland Ms.
20:23.23*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
20:23.29CaNaBiSI wish I could help
20:23.31ManxPowerI'm in Texarkana TX at the moment.  I can be in Baton Rouge this evening
20:23.47CaNaBiSI wish I could help
20:23.55CaNaBiSI wish I knew someone that could use my phone switch
20:24.23CaNaBiSI have a switch and 55 phones in my garage ready to go
20:25.20*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net)
20:26.19criptosI can use them :)
20:28.19*** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net)
20:31.07*** part/#asterisk pa (n=Paolo@unaffiliated/pa)
20:31.56ManxPowerhttp://www.tshirthell.com/katrinashirts.htm
20:32.06Kattybeep
20:32.23Hmmhesaysheh
20:32.28Hmmhesaysi'm sick of working on dialplans today
20:32.40Hmmhesaysstupid asterisk having to answer before sending the call to voipjet
20:34.58*** join/#asterisk Brixius (i=Brixius@c-24-118-215-163.hsd1.mn.comcast.net)
20:35.17BrixiusHello
20:35.51JerJermoo
20:36.06CaNaBiSanyone familiar with how to convert a Cisco 7940 back to a SCCP phone from a MGCP phone?
20:38.44*** join/#asterisk gp6 (n=none@dsl-084-058-010-050.arcor-ip.net)
20:39.09gp6hello all
20:39.29gp6could someone give me a quick hint to what this means ? "pbx.c:1957 ast_pbx_run: Timeout, but no rule 't' in context 'sipout'"
20:40.35jskcrHmmhesays:  problems
20:42.53bkw__gp6 you don't have a exten => t
20:44.48BrixiusCan someone verify as I couldn't find this on the wiki but am trying to remember, I'm trying to block calls from 222-333-4444 to exten 2000 if I put 2223334444/2000 for my exten in my dialplan will it match calls from that # to exten 2000?
20:45.09bkw__Brixius, reverse that
20:45.12bkw__exten/cid
20:45.16Brixiusthanks
20:45.44BrixiusI have a stalker that needs to be blocked.
20:45.50*** join/#asterisk fugitivo (n=ajf@201.255.105.59)
20:45.51Brixiushaha
20:45.58bkw__you can do pattern matches in the cid match part
20:46.09bkw__exten/_222333XXXX
20:46.16Brixiusah, ok
20:47.27jskcrBrixius:  a stalker?
20:47.43zedkatufBrixius: I am interested to know how to do this, so if/when u get it working, could you paste your efforts on the wiki?
20:47.45dudesor even _NXXNXXXXX[01]
20:48.16zedkatuf(wrt sales calls rather than stalkers, but just as relevant I guess!)
20:48.40gp6bkw__: I have this setup -> exten => _X.,1,SetCallerId,<MYSIP_ID>, exten => _X.,3,Dial(BLT/Logitech HS02-V07 HS), exten => _X.,4,Hangup. Maybe it could be the spaces near BLT/ ?
20:48.50BrixiusA company that has been calling me 2-3 times a day, I've been putting them on hold until they hang up. but they don't seem to get the hint.
20:48.53bkw__gp6, NO
20:49.05jskcrzedkatuf:  thats one thing nice bout pbx's at least it drops the automated calls
20:49.10dudesBrixius - the coding institute?
20:49.12dudeshehe
20:49.22bkw__zedkatuf, this is documented already
20:49.26Kattybeep.
20:49.43jskcrBrixius:  I would forward them to a extension that gives a message thank you for calling we do not want what your selling have a nice day
20:50.07jskcrKatty seems to be beeping again
20:50.14Kattybeep
20:50.18tzangerkatty
20:50.21Kattyjskcr: only when my hilight is set off.
20:50.23tzangerkatty
20:50.25tzangerkatty
20:50.29manyBLT.
20:50.31dudesPattern match the CID and put them into an extension that plays a really loud fax sound
20:50.58*** join/#asterisk l1nux (n=l1nux@lns-vlq-5-tou-82-64-211-239.adsl.proxad.net)
20:51.07l1nuxhi
20:52.15l1nuxany one get connect "linksys pap2-NA" with asterisk ?
20:52.39huslageyep
20:52.39fugitivome
20:52.39file[laptop]l1nux: it's just a regular SIP device, nothing special about it
20:52.53gp6many: do you know anything about BLT ?
20:53.04manyno, ill test that tomorrow.
20:53.13FuzzyCatit has bacon, lettuce and tomato in it
20:53.14BrixiusSent them to message "You have a very important message for me, please hold..."
20:54.03gp6ah, I'm try to test it now.
20:54.24*** part/#asterisk l1nux (n=l1nux@lns-vlq-5-tou-82-64-211-239.adsl.proxad.net)
20:54.26*** join/#asterisk l1nux (n=l1nux@lns-vlq-5-tou-82-64-211-239.adsl.proxad.net)
20:54.33manyi have two headsets (hopefully different) and all the required stuff at work
20:54.50l1nuxfile[laptop] is work fine with sjphone or x-lite !
20:55.08l1nuxbut with linksys is offline ):
20:55.25gp6many: Ah, do you have a logitech one ?
20:55.41manyhama and the other one ill see tomorrow.
20:55.44*** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
20:58.40gp6many: aha. are you going to test it with a asus-wl500 ?
20:59.53l1nuxfile[laptop] whats is need for work ?
21:00.36l1nuxplease help
21:01.01l1nuxwith softphone work fine !
21:01.11file[laptop]well first you need to figure out if it's even attempting to register or anything
21:01.18file[laptop]but I bet it's your configuration of the PAP2-NA
21:01.24file[laptop]probably not completely configured
21:01.34DrukenHMEahh shit... the brats go back to school tomorrow....
21:01.35l1nuxsip.conf
21:01.37l1nux?
21:01.45file[laptop]l1nux: I meant on the PAP2-NA
21:01.46DrukenHMEall those damn buses holding up traffic
21:02.10jskcrl1nux:  get the pdf manual for them too
21:02.17DrukenHMEfile[laptop]: have you been able to get a data call over a pap2?
21:02.30file[laptop]DrukenHME: I faxed once through it
21:02.36l1nuxjskcr where ?
21:02.38DrukenHMEi've done that...
21:02.40*** join/#asterisk zotz (n=zotz@24.231.36.100)
21:02.46file[laptop]but besides that, nah
21:02.55DrukenHMEi can even get a modem to connect, but it sure as shit don't stay connected long
21:03.00l1nuxwhy pap2 work fine with sip providers
21:03.04l1nuxlike FWD
21:03.35l1nuxProxy: 192.168.0.5
21:03.46l1nuxOutbound Proxy: 192.168.0.5
21:05.19*** join/#asterisk RoyK (n=roy@80.232.64.251)
21:06.03DrukenHMEfile[laptop]: that's the only drawback i've found so far.... i can't order my pay per view with the voip line...
21:06.12DrukenHMEand since i have no other line... i'm sol
21:06.23*** join/#asterisk Baph (n=Dave@dirobertson.plus.com)
21:06.46Assidis there a softfax ?
21:06.48jskcrDrukenHME:  the problem with the faxes will be hit or miss because most still use ulaw for the faxes and it sucks some times
21:07.01Assidthat i can use to send jpegs as fax?
21:07.10DrukenHMEeven if i use alaw... it's still a whore
21:07.20jskcrI saw the t.38 patches and they look interesting
21:07.31jskcrIm gonna test em this week.
21:07.49*** join/#asterisk root_ (n=root@p5497D438.dip.t-dialin.net)
21:07.51BaphAssid: check out dataconv.org
21:08.25Assidthanks
21:08.46jskcrRoyK:  We think it will work
21:08.47ManxPowerI need a 4-port Digium T-1 card by tomorrow.  Can anyone help me.  Shipping to Baton Rouge LA
21:08.51hugo-v6jesus christ. i hate it to fix the friednship between gf and her gf
21:08.54jskcr^stable
21:09.10*** join/#asterisk andreas_hecker (n=Andreas@p5497D438.dip.t-dialin.net)
21:09.11AssidBaph:dont get it.. hows this gonna help me send faxes?
21:09.13manygp: no, yozu or what it is called like.
21:09.27manygp6: but actually i guess its not even that, but some fscking noname.
21:09.49BaphAssid: in their 'conversion tools' section, theres several apps for fax converstions (both ways)
21:10.22andreas_heckerhello, can anybody help me solving this message: Warning: 392 192.168.1.3:5060 "Noisy feedback tells:  pid=13783 req_src_ip=192.168.1.40 req_src_port=4207 in_uri=sip:192.168.1.3 out_uri=sip:192.168.1.3 via_cnt==1"
21:10.28DrukenHMEhugo-v6: just fuck them both, and tell them to deal with it :P
21:10.51manyhaha
21:11.19hugo-v6DrukenHME: well... i dont wanna fuck her gf ;) and this would be also rather contraproductive ;))
21:11.33hugo-v6yaw many
21:13.12Baphforgive my ignorance (but since this is for someone that's paying me, I want to make sure...) I've just stumbled across asterisk... I'm looking for a software solution to run VoIP (probably with analog phones using a VoE converter) office style PBX, I'm also looking at possibly linking mobile telephones & a remote site into the PBX... would asterisk at least accomplish some of this 'out of the box'?
21:14.13andreas_heckerhello, can anybody help me solving this message: Warning: 392 192.168.1.3:5060 "Noisy feedback tells:  pid=13783 req_src_ip=192.168.1.40 req_src_port=4207 in_uri=sip:192.168.1.3 out_uri=sip:192.168.1.3 via_cnt==1"
21:14.22l1nuxDrukenHME, please where download pdf guide ?
21:14.39*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
21:15.30*** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net)
21:15.42manyya hugo
21:15.54pbdBaph- sorry, was a little distracted for a minute..
21:15.56DrukenHMEBaph: asterisk doesn't do anything 'out of the box' :)
21:16.25pbdBaph- As in most complicated software, you have to expect a certain amount of configuration.. but no, you shouldn't have to rewrite it to do what you're looking to do.
21:17.55Baphpbd, that's exactly what I'm looking for then, by 'out of the box' I expect a HEAVY amount of customisation (esp due to the fact I have to VPN link the two sites together whilst avoiding another VPN system I have no control over), tyvm :)
21:17.59DrukenHMEl1nux: http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=page%3D2%26cid%3D1115416835852%26c%3DL_Content_C1&pagename=Linksys%2FCommon%2FVisitorWrapper&SubmittedElement=Linksys%2FFormSubmit%2FProductDownloadSearch&sp_prodsku=1117044308483
21:18.31pbdbaph- there are a lot of gotchas with VoIP, not just Asterisk, that you're going to run into- but it sounds like you're in the right frame of mind.
21:19.36DrukenHMEwhen i first got into voip... i swear my cell useage went threw the roof... now almost a year later i'd say i use my cell about the same as before
21:19.46Baphpbd, I'm used to *nix/BSD systems (I get paid to install them heh), and used to having issues compiling & getting things wrong every so often... and luckily, on this contract I have pretty much unlimited time to learn how to use/config the VoIP systems :)
21:20.00l1nuxthanks DrukenHME :) but not admin guide
21:20.29l1nuxUser Guide and Data Sheet
21:20.33DrukenHMEoh.. you have a -na ?
21:20.36BaphDrukenHME: lol, I'm looking at saving cell costs (currently my client spends >£150 GBP per trip over seas... and can take 3-4 trips per month)
21:20.44PupenoLWhat is astdb exactly ?
21:20.53*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
21:21.00*** join/#asterisk santiago (n=santiago@63.245.87.180)
21:21.25pbdBaph- Sounds like a nice job. Now.. as to cell costs- are you looking to replace his cell phone with a softphone client?
21:22.04jalsotdoes anybody know why am I getting incorrect UDP checksum for packets going out from asterisk to iaxComm?
21:22.21Baphpbd, not sure yet, probably a softphone to accompany his cell, since he wont always have access to the net (but I'd like to route his remote calls through the soft-PBX solution if possible to cut call costs)
21:24.47Baphhmmm, pbd, speaking of softphones, dont suppose you know of any that are 'Apache friendly' & can run from a website (probably with a Java applet to stream audio)
21:24.50l1nuxDrukenHME, yes i hane NA !
21:24.58l1nuxhave*
21:25.17pbdBaph- Interesting.  Since the caller is remote, you've got to either set up your own 'cell site' local to the cell phone (but hosting an asterisk server), or, you have to assume some sort of IP connectivity to the softphone.
21:25.40l1nuxnot found any thing in "user guide" !!
21:25.49Baphpbd, I'm thinking cell phone calls the land line number for the office, and the office then dials out to the required number
21:26.30l1nuxany one help me, howto make "LINKSYS PAP2 NA" work with asterisk ?
21:26.33pbdBaph- that'll work.. and Asterisk is pretty well suited to it.
21:27.23pbdl1nux: you'll have to unlock it, unless you have the unlocked one already (hard to get)
21:27.29Baphgood :) that's probably an instant saving on costs then :) does Asterisk have any donations procedure (because I think my client would be pretty happy about x% of the monthly saving being donated)
21:27.33l1nuxmy "LINKSYS PAP2 NA" work fine with FWD, but not with asterisk
21:28.09l1nuxpbd, my linksys not locked !
21:28.12DrukenHMEBaph: yeah, donate to me :) sales@abss.ca via paypal
21:28.12l1nuxis NA
21:28.13DrukenHMEhehehe
21:28.45pbdBaph- Don't count your chickens here.  Yes, you can donate- check voip-info.org for open bounties.. but.. you've got a long way to go before you're happy.
21:29.04*** join/#asterisk flewid (n=flewid@CPE000024c43e1c-CM000f9fac6da2.cpe.net.cable.rogers.com)
21:29.09BaphDrukenHME: get something @asterisk.org & I'll consider it :P
21:29.16flewidsup
21:29.17fugitivol1nux: pap2-na works great with asterisk
21:29.28Baphpbd, I understand that, this is all ideas to discuss with my client before our meeting on weds
21:29.40DrukenHMEBaph: :P
21:29.46l1nuxfugitivo, please howto
21:30.20fugitivol1nux: setup it correctly and it'll work
21:31.24*** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net)
21:32.04Baphpbd/DrunkenHME, do you know of any web-portal like (Java applets etc) that will easily tie into asterisk to give a web-portal softphone?
21:32.55hugo-v6web-portal softphone sounds fucked up. why should someone want that?
21:32.57pbdBaph- What's the purpose of the portal?
21:33.30puzzledhugo-v6: pr0n :)
21:33.30*** join/#asterisk KaBewM (n=DA-MAN@24-180-28-208.dhcp.psdn.ca.charter.com)
21:33.41kajtzuso your users can communicate with you over the web without a real soft client
21:33.44hugo-v6puzzled: lol
21:33.51pbdBaph- There are a few people working on active-X softphones.. but.. quality is poor, and you're not going to do much more than provide a good place to download- VoIP is not HTTP, or vice versa.
21:33.59Baphmy client can get access to the internet from remote locations overseas, but is usually VERY restricted by the various VPN tunnels used, however, plain HTML websites (inc applets) are permitted
21:34.09l1nuxfugitivo  http://l1nux.free.fr/files/4.jpg
21:34.11fugitivoactiveX sox
21:34.16fugitivosux
21:34.18fugitivo:)
21:34.24fugitivosox is great
21:34.25fugitivolol
21:35.07DrukenHMEBaph: my reccomendation, use an overseas provider, and have your client dial into the pbx with the payphones across the pond :)
21:35.11Baph(maybe something from the old Yahoo Chat Voice applets might come in handy)
21:35.47fugitivol1nux: that doesn't look like the same pap2-na as mine
21:36.18*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
21:36.23fugitivol1nux: oh, you're in advanced
21:36.33l1nuxfugitivo  http://l1nux.free.fr/files/5.jpg
21:36.45fugitivol1nux: i have this options:
21:37.20l1nuxfugitivo, not work for me ):
21:37.24fugitivol1nux: use outbound proxy: no use ob proxy in dialog: yes register expires: 3600
21:37.31fugitivol1nux: does it register?
21:37.35BaphI'll look into ripping (errr, borrowing) some parts from the old Yahoo protocols, they were pretty good quality streams, yes they were exploitable, but I'm only looking for a small solution & can put Kerberos auth in front of it... I'll let you guys know what I find
21:37.41l1nuxfugitivo, no
21:37.48fugitivol1nux: what error?
21:37.49l1nuxfugitivo, is offline
21:37.58fugitivol1nux: show me your CLI output
21:38.20l1nuxok, 2 minute
21:38.26FuzzyCatyou show me yours and I'll show  you mine
21:40.03fugitivolol
21:40.14fugitivoyou don't wanna see my CLI output
21:40.16fugitivoit's nasty
21:41.03FuzzyCat903 active calls
21:41.50l1nuxfugitivo, nothing !
21:41.57fugitivonothing what?
21:42.04l1nuxRegistration State: Can't connect to login
21:42.09l1nuxin cli
21:42.14fugitivohuh?
21:42.25l1nux<PROTECTED>
21:42.25l1nuxAsterisk Ready.
21:42.26l1nux*CLI>
21:42.35andreas_heckerhello, can anybody help me solving this message: Warning: 392 192.168.1.3:5060 "Noisy feedback tells:  pid=13783 req_src_ip=192.168.1.40 req_src_port=4207 in_uri=sip:192.168.1.3 out_uri=sip:192.168.1.3 via_cnt==1"
21:43.26fugitivol1nux: so... start with asterisk -vvvvvvvvvvvvvv and check if you see any error message
21:43.53l1nuxasterisk -cvvvvv
21:44.00l1nux?
21:44.40*** join/#asterisk nagl (n=nagl@213.235.241.6)
21:45.02fugitivoand then sip debug
21:45.46l1nuxfugitivo, nothing in CLI  ) :
21:45.59Baphlast question (for today), assuming two sites... and I want 1 site as a PBX extension of the main office... would asterisk have to dial the other location or (subject to it being online) can the call go over IP direct to the remote office (subject to decent QoS settings)?
21:46.03fugitivois 192.168.0.8 the ip of your asterisk?
21:46.17faa_Baph yes.
21:46.34Baphyes to direct link over Ethernet?
21:47.08faa_yes
21:47.13Baphk, tyvm :)
21:47.41Baphpbd & faa_, you've been a great help :) and DrukenHME ... err... yea :P
21:47.47l1nuxfugitivo, yes
21:48.14l1nuxfugitivo, is work fine with x-lite
21:48.22l1nuxfugitivo, or sjphone
21:48.58*** join/#asterisk newlook_ (n=newlook@203-59-168-152.perm.iinet.net.au)
21:50.32*** join/#asterisk mike_jh (n=mike@81.187.90.205)
21:54.22l1nuxfugitivo, any idea ?
21:54.35*** join/#asterisk CaNaBiS_ (i=canabis@ip68-111-51-215.br.no.cox.net)
21:54.50fugitivouse tcpdump to see if the pap2 is trying to register to asterisk
21:55.41l1nuxfugitivo, howto get linksy debug ?
21:56.41l1nuxSIP Debug Option: full
21:56.57criptosfugitivo/ l1nux, better use ethereal.. is much more easier to use and understand thatn tcpdump...
21:57.30jalsothi
21:57.39fugitivotcpdump is easy! its like looking to the matrix code
21:57.42l1nuxcriptos, in asterisk machine, i dont have X server
21:57.50jalsothmmm, I'm getting strange IAX packets
21:58.26jalsotfor PING asterisk responds with INVAL [where timestamp and seq. numbers are 0], any idea?
21:59.04ManxPoweranyone know how to get ahold of a real person at Teliax?
21:59.25JerJerdefine real
21:59.50jalsotJerJer: that's for me?
22:00.16DrukenHMEManxPower: threaten to cancel your service?
22:01.05FuzzyCathehehe * doesn;t like it if you throw 3000 calls at it at once
22:01.22JerJerthen deploy a load balancing solution
22:02.11FuzzyCatit was a deliberate attempt
22:02.29*** join/#asterisk Legend (n=legend@24.244.142.133)
22:03.13criptosl1nux, make your self the life easy, put X, debug, fix and then remove X :) but, if not, then use tcpdump, and save it to a file, later in other machine, you can upload tcpdump files to ethereal...
22:03.33fugitivohe doesn't need X
22:03.43fugitivohe only needs to see any packet comming from the pap2
22:03.51*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
22:04.03l1nuxtcpdump -i eth0 | grep 192.168.0.8
22:04.23DrukenHMEwhy not do a factory reset on the pap2?
22:04.24fugitivono, not 8
22:04.34DrukenHMEthat way it'll loose and shitty provisioning it may have
22:04.38fugitivothe ip of your pap2
22:04.54l1nuxyes
22:04.58DrukenHMEs/and/any
22:04.58fugitivoI agree with DrukenHME
22:05.13fugitivol1nux: whats the ip of your pap2 and the ip of your asterisk box?
22:05.18ManxPowerDruken, I might if I could contact them, but I just need the DID fixed.
22:05.21l1nuxfugitivo, no pzacket from linksys(192.168.0.8)
22:05.37DrukenHMEwhen ever i have problems with any of my pap2's or rt31p2's i usually have to do a factory reset to fix it
22:05.37fugitivohold on
22:05.42fugitivohold on
22:05.46l1nuxpap2 = 192.168.0.3
22:05.55l1nuxasterisk = 192.168.0.8
22:05.58fugitivook
22:06.04fugitivothen grep for 192.168.0.3, not 8
22:06.20DrukenHMEManxPower: if they are a cdn company, they won't be in today...
22:06.21l1nuxoohh yes
22:06.25l1nuxsorry
22:07.16DrukenHMEl1nux: http://192.168.0.3/admin/reset :)
22:07.34DrukenHMEthen setup the pap2 again, and i bet ya it'll work... unless you have a firewall issue on your ast box
22:07.37l1nuxDrukenHME, ehh, why
22:07.39l1nux?
22:07.42fugitivol1nux: do a hard reset before like DrukenHME said
22:07.51bkw_because sometimes its needed
22:07.57bkw_to clear up fucked up settings
22:08.10DrukenHMEbecause pap2's are like that... hehehe
22:08.12bkw_Starting Subversion commit 638 / 6503
22:08.13ManxPowerDruken, they are a dallas company
22:08.16bkw_this is going to take some time
22:08.26l1nuxmy linksys work fine with freeworldialup, and sipphone !!
22:08.29ManxPowermy poor cat has been hiding under the bed covers all day
22:08.59DrukenHMEManxPower: you've had pussy under covers all day? bastard
22:09.13bkw_I knew someone would go there
22:09.18DrukenHMEhehehe
22:09.42DrukenHMEguilty
22:09.43l1nuxfugitivo, howto get log from linksys ?
22:10.15l1nuxohhh
22:10.18l1nux22:40:13.517132 IP 192.168.0.3.5064 > zhlap.sip: UDP, length: 425
22:10.18l1nux22:40:13.517456 IP zhlap > 192.168.0.3: icmp 461: zhlap udp port sip unreachable
22:10.20FuzzyCatpuzzled, wait 10 seconds
22:11.06l1nuxfugitivo, any idea ?
22:11.17puzzledFuzzyCat: yeah I know. except that the stoneage udev version on centos does play nice with perms other than root
22:11.26puzzleddoes not
22:11.33fugitivol1nux: hard reset
22:12.08FuzzyCatoooo.... nasty ...
22:12.14puzzledyup
22:12.20FuzzyCati assume you read the udev readme
22:12.46*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
22:12.46DrukenHMEwho reads readme's ?
22:12.55puzzledyes. it works fine on FC4 (with some adjustments). the README.udev file prolly applies to debian or something. no go on FC4 & CentOS4
22:13.43FuzzyCatDrukenHME, the desperate...
22:13.54DrukenHMEtoo true
22:14.44*** join/#asterisk kusznir (n=kusznir@pool-68-238-130-44.sea.dsl-w.verizon.net)
22:15.33puzzledFuzzyCat: what's even more desparate is that you appareantly need to reboot the box for udev to pick up changes in config files...
22:15.43puzzledyuck
22:15.55DrukenHMEreminds me of windows...
22:16.01puzzledexactly
22:16.05FuzzyCatomg!
22:16.51pbdWho's brainchild was udev anyway.  More of a PITA than anything else.
22:16.57FuzzyCatwell, tbh I usually do that after setting up * and all the crap as a matter of course.. just to check everything's ok
22:17.18puzzledpbd: mainly RH
22:17.52puzzledFuzzyCat: yes that makes sense. but rebooting the thing each time I make a change. What were they thinking
22:19.27FuzzyCatthey were on acid again
22:19.48puzzledhehe acid is sooo old school
22:20.18FuzzyCatyes, but so are they... wit their beards with bits off food in it
22:21.01DrukenHMEthat's just nasty
22:21.27FuzzyCatcruel to be kind
22:21.34kusznirHi all: is the gnome asterisk console very useful at this time?
22:22.04fugitivowhat gnome asterisk console?
22:22.05*** part/#asterisk pbd (n=PDavidso@c-24-15-72-86.hsd1.il.comcast.net)
22:22.13stkn_puzzled: udevstart, will apply the changes w/o rebooting
22:22.16kusznir(My asterisk install didn't include it (gentoo ebuild asterisk 1.2 beta1), and I haven't figured out how yet.  I was wondering if its worth digging into)
22:22.30puzzledstkn_: thanks!
22:22.39kusznirFrom modules.conf:
22:22.41kusznir; If you want, load the GTK console right away.
22:22.41kusznir; Don't load the KDE console since
22:22.41kusznir; it's not as sophisticated right now.
22:23.02kusznirThe module (pbx_gtkconsole.so) wasn't provided in my installation.
22:23.24fugitivowhy do i need a sophisticated console???
22:23.31*** part/#asterisk criptos (n=criptos@201.135.121.4)
22:23.40chetdoes anyone have a cisco phone running mgcp?
22:24.03kusznirI'm trying to learn asterisk, and I thought I'd start with grasping the functionality through an easier console, then focus on learning the lower-level syntax.
22:24.04stkn_kusznir: the ebuild has a gtk flag, did you set that?
22:24.18cheta friend is trying to upgrade, and it is looking for a mgcp.cnf file, his is gone and im wondering what that file consists of
22:24.39flewidchet: check the asterisk/configs directory for a sample
22:24.40kuszniryep.  It was set
22:24.53chetthanks flewid
22:24.56FuzzyCat[Call 5632, Time 5631, CPS 15.00]  :>
22:25.00flewidchet: np
22:26.32*** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net)
22:27.53kusznirJust out of curiosity, should I be running the 1.2 beta or the 1.0.9?  This is for learning/experimentation, not "real-world use".  Not yet, anyway :)
22:28.16fugitivocvs-head
22:28.17hugo-v6kusznir: use cvs or head
22:28.29hugo-v6but not 1.2 beta
22:28.32FuzzyCatpersonally 1.0.9 because you can go live wiht it later
22:28.40FuzzyCatdon't use head
22:28.56*** join/#asterisk RoyK (n=roy@ti211310a080-12184.bb.online.no)
22:29.03puzzledhave no phear, head is here
22:29.22flewidi'd say head since if you're experiementing, you'll get to play with the upcoming features
22:29.29hugo-v6aww. i mix head and stable everytime.
22:29.30fugitivoand help to find bugs
22:29.33*** join/#asterisk CaNaBiS_ (i=canabis@ip68-111-51-215.br.no.cox.net)
22:30.26DrukenHMEwhat new features have been added lately? (last 3 months)
22:30.26kusznirFuzzyCat: This install will never really go live to support anything more than my personal phone service, as its on my personal desktop with a sub-optimal net connection ;)
22:30.37fugitivothen head
22:30.39puzzledkusznir: use head
22:30.43flewidhead
22:30.52puzzledor tail
22:30.56fugitivolol
22:31.10DrukenHMEhead :)
22:31.22fugitivokusznir: any doubt? :)
22:31.24FuzzyCatthen use cvs head then...
22:32.07kusznirnope :)
22:33.28puzzledugh, the stuff that tastes like freshly laid tarmac
22:33.53DrukenHMEnot sure i want to know why you know what tarmac tastes like
22:33.53RoyKpuzzled: it's nice, in small amounts...
22:34.07puzzledLaphroaig or Oban must be the stuff that killed the dinosaurs
22:34.37RoyKpuzzled: Oban is lite
22:34.48RoyKpuzzled: Laphroaig is heavy
22:34.57hugo-v6whats that ure talking about?
22:35.13puzzledRoyK: in my book it doesn't matter if they taste like tarmac :)
22:35.15*** part/#asterisk secure75 (n=mic@p549A0D19.dip0.t-ipconnect.de)
22:35.18RoyKpuzzled: the one that killed the dinosaurs must be the whisky from the Isle of Jura
22:35.27fugitivowhisky, mmm
22:35.44jalsothmmm, I'm getting strange IAX packets; for PING asterisk responds with INVAL [where timestamp and seq. numbers are 0], any idea?
22:35.53fugitivohmmm, whisky
22:35.59FuzzyCatmmmmm Jura...
22:36.29hugo-v6hmmmmm whisky
22:36.50fugitivoit's early yet...
22:36.52fugitivoi can wait
22:37.01RoyKpuzzled: the Isle of Jura is something like 500 yards from the Isle of Islay where Lagavulin, Bowmore, Laphroaig, Ardbeg, Coal Ila, Bunnahabain and Bruchladich are brewn
22:37.07fugitivohmmmm whisky
22:37.12tzangermmm whisky
22:37.25FuzzyCatmmmmm hotdogs
22:37.37tzangerno, not mmm hotdogs
22:37.41tzangerwhen you've got whisky what else is there
22:37.43puzzledRoyK: I stay away from that area and its products like it were Tsjernobyl
22:37.57RoyKpuzzled: rotfl
22:38.07fugitivommmm whisky
22:38.10FuzzyCathotdogs  tzafrir_laptop
22:38.12fugitivommmmmm more whisky
22:38.14FuzzyCaterrm tzanger
22:38.42fugitivotzanger: more whisky
22:38.58RoyK(pronounced ooben)
22:39.09puzzledRoyK: if you sit too long on an island you go bonkers and start to make weird concoctions and call them whisky. Look at FuzzyCat. He's escaped a while back but still...
22:39.51tzangerpuzzled: :-)
22:40.17puzzledI think I actually liked that one
22:40.38*** join/#asterisk Felagund (n=nothing@wnpgmb11dc1-165-185.dynamic.mts.net)
22:40.50Felagundevening all
22:40.53Felagundhad a quick question
22:40.58ManxPowerI think I'm getting the Heartburn of Doom
22:41.07Felagundis there a way to integrate asterisk into ms exchange or lotus notes or groupwise?
22:41.13Felagundto give a unified messaging solution?
22:41.25fugitivoFelagund: whisky
22:41.54FuzzyCatcheese
22:42.10tzangerFelagund: there's a rightclick->dial this contact for Outlook somewhere
22:42.18Felagundno I mean
22:42.24Felagundreceiving voicemail as email
22:42.28fugitivoyes
22:42.39Felagundlike with a unified messaging system you can receive voicemail as email
22:42.40YoYodid you read voicemail.conf ?
22:42.41tzangerFelagund: you don't need to do anything fancy there, it's already supported
22:42.45FuzzyCatmuwhahahhahahahahahah [Call 42582, Time 42581, CPS 18.00]
22:42.49Felagundthe question is though
22:42.55Felagundhow does asterisk know that you've received the voicemail
22:42.57fugitivoFelagund: voicemail sends emails with the voicemail attached
22:43.05*** join/#asterisk _-Jon-_ (i=jon@CPE00112f6dfbee-CM00111a232a80.cpe.net.cable.rogers.com)
22:43.05fugitivoFelagund: /etc/asterisk/voicemail.conf
22:43.08_-Jon-_Hey everyone
22:43.12drraythere is a voicemail app in asterisk
22:43.19Felagundb/c with unified messaging systems when you play the voicemail attachment it sends a message to the voicemail system that the voicemail was played
22:43.21fugitivoFelagund: it doesn't know, it just sends the email
22:43.34YoYoFelagund, what you're asking for isn't possible, but why would asterisk care?
22:43.48fugitivoFelagund: i think there's a project to develop something like that
22:43.48Felagundso it would know if the voicemail has been read or not
22:44.10_-Jon-_I'm wondering if this situation is possible..  someone calls my DID, dials an extension which simply rings my cell phone so that I can call my toll-free number back and get connected with the calling party
22:44.19Felagundthat's really the difference between full blown unified messaging and simple SMTP attachment sending of voicemail
22:44.28Felagundfugitivo where?
22:44.41fugitivoFelagund: i don't remember, but it isn't developed yet
22:44.52fugitivoFelagund: search in voip-info.org
22:45.57YoYoJon: why would you want something so complicated?
22:46.12*** join/#asterisk drbrown (n=chatzill@63.238.118.90)
22:46.15*** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net)
22:46.44_-Jon-_YoYo, my cell phone company jacked up their long distance rates, so if I can pay 2.4c/min vs 20c/min i will
22:47.31YoYowhen someone calls me, it rings like 5 phones simultaneously, including my cell phone
22:47.56YoYoand my cell phone is a local call for my * box, so there's no toll charges at all
22:48.27YoYojust counts against the minutes on my cell plan, of which I use maybe 50 out of 600 in a month
22:48.44_-Jon-_YoYo, sometimes I'm out of my local calling area so even on incoming calls I get charged LD
22:48.57YoYoJon: then you need to find a new wireless provider
22:49.17YoYothere are SO many that don't charge roaming fees
22:49.31YoYounless you're in some regressive part of the world...
22:49.42_-Jon-_Canada's wireless providers kind of suck
22:49.43EquinoxAnyone have a FWD number I can test to?  My call audio doesn't seem to work for some reason
22:49.54_-Jon-_All 3 of them
22:50.02eKo1hah! i knew _-Jon-_ was going to say canada
22:50.10_-Jon-_How'd ya know :P
22:50.12*** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net)
22:50.17YoYodamnit, when will canadians get sick of that crap and vote to become part of the union?
22:50.36YoYoJon: well, the *rogers.com was a slight clue :)
22:50.50tzanger_-Jon-_: well Telus is eager but their network is lacking (odd, since they piggyback with Bell over in Ontario), Bell works *very* well but their customer service is lacking in a big way, and rogers just doesn't have the coverage
22:50.50_-Jon-_That'll give it away
22:50.51kusznirWhen making from cvs, if I don't have any zaptel stuff, should I still make zaptel and libri?
22:51.15eKo1kusznir: well no. duh
22:51.28_-Jon-_Rogers coverage is great (better than I thought) but the overloaded network just kills it
22:51.36kusznirI knew I wasn't going to use them; I just wasn't shure if my asterisk build would die without them.
22:51.45YoYokusznir: depends on what you need from *... might need zaptel for ztdummy
22:51.52kusznirwithout doing any makefile editing and such.
22:52.05kusznirwhere is ztdummy used?
22:52.15hugo-v6kusznir: nope. but ztdummy is neede for meetme iirc
22:52.19fugitivomeetme
22:52.19flewid_-jon: tried telus?
22:52.26tzangerflewid: don't try telus
22:52.26YoYodun remember... meetme and moh?
22:52.33kusznirwhat is meetme, anyway...I've never managed to figure that out
22:52.45fugitivokusznir: conferences
22:52.48flewidtzanger: i don't mind telus that much, ive' been with them for like 10 years almost
22:52.56flewidgood support, good coverage, not _that_ bad of rates
22:53.00tzangerI'm in my living room and there's a telus tower in view.  works great.  15km down the road there's a telus tower and I drop out there every time
22:53.01flewidand better than rogers or bell :)
22:53.10tzangereven though with Bell there I get perfect reception
22:53.19flewidyou have one of the tri-mode phones?
22:53.25tzangerwhich baffles me... if the telus netowrk is down or off, the damn thing should be on Bell's towers
22:53.29flewidit should use the bell/rogers network as well should it not?
22:53.29tzangerflewid: yes
22:53.36tzangerflewid: no just Bell
22:53.44flewidoh i thought rogers was part of that deal now too
22:53.46tzangerapparently they have a receip roaming agreement with them
22:53.49YoYohow the hell does canada only have 3 wireless providers?
22:53.49flewidahh
22:53.53tzangerflewid: no, rogers is GSM.  Telus/Bell is CDMA
22:53.57flewidyoyo: telus, bell, rogers, fido
22:53.58*** join/#asterisk guest (n=guest@adsl-69-109-40-118.dsl.irvnca.pacbell.net)
22:54.05_-Jon-_Rogers own Fido now though
22:54.08tzangerYoYo: well there's Fido too.  We had Clearnet/PCS but Telus bought 'em
22:54.09flewidyeah
22:54.12flewidand sprint
22:54.26guest<PROTECTED>
22:54.33YoYoomg, we have a geust
22:54.36YoYoguest even
22:54.40flewidyoyo: i bet it's due to the spread out nature of canada that there's only three
22:54.48guesti am newbie at linux and asterisk
22:54.50flewidonly the "big guys" can say coast to coast, like the little guys can in the states
22:55.17_-Jon-_Hey wait, does anyone else in here have Fido>?
22:55.22YoYoflewid: spread out?  you guys are all bunched up against our border...
22:55.24tzangernot me
22:55.24flewida guy at my office has fido
22:55.27flewidhe loves it
22:55.27guesti am wondering if this is a good place to find help understading asterisk
22:55.38fugitivo_-Jon-_: Fidonet?
22:55.40flewidbut he goes to india to buy new cell phones and fido supports any phone i guess
22:55.43flewidso that's why he likes it
22:55.53YoYofugitivo: no... fido wireless... it's a canadian thing
22:55.55tzangerflewid: any gsm phone, yes
22:56.01YoYothey never had BBSes in Canada
22:56.02flewidyoyo: yeah, try and give some service to nunavut
22:56.04fugitivogreat name
22:56.10kusznirguest: yes, this is.
22:56.11tzangerfugitivo: it's a metropolitain carrier
22:56.17tzangerYoYo: hahaha yeah that's it :-)
22:56.20_-Jon-_I've been having the strangest problem with Fido..  Any time I call any of my DID's, it never shows the right number
22:56.27YoYowhich is it?
22:57.00tzanger_-Jon-_: I've got one more interesting.  When I call my DID from my Telus phone, the ANI2 seems to get set to a local number (local to where I'm calling FROM) -- I can tell where my phone is (roughly) by looking at ANI2
22:57.01Baphguest, I'm new to asterisk as well... and so far, this place has been the only help I've needed
22:57.13flewidjon: hmm, when my friend uses his cell and calls our DID's (in the USA), it shows unknown/unknown
22:57.20_-Jon-_tanger, you mean the area code changes depending on where you are?
22:57.34_-Jon-_flewid, that's even worse
22:57.46flewidi get that from telus as well
22:58.01_-Jon-_I've had "asterisk" show up on my cid once
22:58.01flewidtried it on two phones, one with no name display, and one with
22:58.14flewidyeah
22:58.14_-Jon-_very strange
22:58.34flewidi was talking to our PRI provider, and he said sometimes that happens just due to "the states networks not liking canadian networks"
22:58.48flewidand with it bouncing it's way around somewhere the cid is dissolved
22:59.47*** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net)
23:00.12*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
23:00.25tzanger_-Jon-_: no, I'm saying I get an ANI2 of some local-to-where-i'm-calling-from number...  it's nothing to do wtih my cell #
23:00.43tzangerflewid: that is just calling presentation
23:01.13kusznirI noticed pbx_gtkconsole was not built with cvs head; I tried to compile it manually (cd pbx; make pbx_gtkconsole) and I got more errors than my scrollback buffer can hold.  Is it broke, or did I screw something up?
23:01.20Ariel_hello everyone
23:01.25flewidtzanger: weird, like an "outdial" of sorts happening?
23:01.32Ariel_has anyone used the new fxotune for the zaptel?
23:01.49fugitivoAriel_: no, what is that?
23:01.50tzangerflewid: yeah it's weird.  CID is fine, ANI is fine, ANI2 changes with where I'm calling from
23:02.05tzangerANI2 is almost always '0' though for all other calls
23:02.11flewidinteresting
23:02.22Ariel_fugitivo, it's part of the new echo cansole routing in the zaptel head KB1
23:02.37flewidone day we'll be blanketed by a wimax network and cells will disappear :P
23:02.56DrukenHMEflewid: i can't wait
23:03.33DrukenHMEof course, if i could get my hands on some cellular dslams, i'd have my own voip cell site :)
23:03.36*** part/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com)
23:03.49flewidwhat's a dslam ?
23:04.00drumkillaDSL Access Multiplexer
23:04.00DrukenHMEtelco equipment
23:04.38flewidthis would allow you to provide your own local cell connectivity via cdma to a voip (asterisk) gateway for free?
23:04.39DrukenHMEdrumkilla: yes, that's it.... course, then i guess cells wouldn't use one... so what the hell am i thinking?
23:04.42guestgush this looks intimidating at a general level
23:04.56drumkillaDrukenHME: that's what I was wondering.  :-p
23:05.04guesta little though to keep up
23:05.38DrukenHMEdrumkilla: i haven't worked in three days, the brain is still on break
23:05.39guesti need a little help on installing asteriks in red hat 9
23:05.58drumkillaDrukenHME: I feel your pain
23:06.02drumkillaexcept I had class today
23:06.11kusznirguest: what have you done so far, and where are you stuck?
23:06.27guesti try to do the make
23:06.36drumkillaI think file[laptop] has a script to tickle me anytime I start talking on IRC :)
23:06.42_-Jon-_tzanger, i have the exact same problem with Fido..  I'm in 416 so I get a random 416 number but if I'm in like the 613 area code I get a random 613 number
23:06.42file[laptop]that's a lie
23:06.59kusznirguest: and what happens?
23:07.03flewidi'm going to try that next time i'm in TO
23:07.05flewidfrom telus
23:07.14tzanger_-Jon-_: well remember I'm talking about ANI2 not CID
23:07.17tzangermy CID is always right
23:07.18guestfirst i try cd /the path to the folder of asterisk
23:07.24guestand then I did make
23:07.38guestI am not sure if asterisk is install in my sys or not
23:07.43_-Jon-_tzanger, oohh
23:08.32kusznirguest:  the only time you use "make" is if you're compiling it from source.  Since you're on a redhat system, your system uses rpm by default.  If you install the asterisk rpm's, then you wouldn't use make; asterisk would already be compiled and installed.
23:08.43Ariel_I keep getting this error with fxotune: Could not fill input buffer
23:08.43Ariel_Tuning module 1.Failure!
23:08.51Ariel_does anyone know about this error
23:10.14guesthold on
23:10.21l1nuxfugitivo, are you here ?
23:10.25guestlet me see if I find that file .rpm
23:11.52guesti downloaded asterisk-1.0.9 should I find the .rpm file there?
23:12.00kusznirguest:  Check out this page.  Its got specific info for varous versions of RedHat and a link to an install guide specific for redhat9: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Redhat
23:13.04guesti am checking it
23:13.15*** join/#asterisk pr0m (n=pr0methe@2002:184b:c446:13:0:0:0:50)
23:16.46guestam i still connected?
23:16.52kuszniryep
23:16.54DrukenHMEto ?
23:17.09guesti notice chatting activity stop
23:17.29drumkillacan you hear me now?!
23:18.25flewidi'm over here
23:18.28guest:-)
23:18.33flewiddick larente is dead
23:18.43flewid(obscure lynch reference) :P
23:18.46*** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net)
23:19.02guesti went to the link kusznir
23:19.13guesti open the installation guide
23:19.27guestdoes asterisk have a special files fro red hat 9?
23:19.46mog_homemake config
23:19.51mog_homewioll add start up scripts
23:19.59*** join/#asterisk CaNaBiS_ (i=canabis@ip68-111-51-215.br.no.cox.net)
23:20.43guestok i got it now
23:20.45kusznirguest:  I actualy use gentoo linux, so I can't say I've installed on RedHat 9.  You might have some difficulty, since RedHat 9 has been unsupported for some time now.
23:20.50guesti am going to try that later
23:21.01drumkillawhy are you using redhat 9?  that is ancient.
23:21.03flewidtry make config or make samples
23:21.07guestgreat help kuznir
23:21.20kuszniryour welcome guest
23:21.22guestthe irc is amazing
23:21.30guesti am not alone
23:21.34guestvery cool
23:21.41guestsee you later kusznir
23:21.48guestsee you later everybody
23:22.10Romikanybody can advice which format best for MOH ? and how to create .g729 files?
23:22.18guesti am going to change my nick to linuxwaa for next time I log in
23:22.34DrukenHMEsad.... when you think your not alone because of irc.....
23:22.52*** part/#asterisk guest (n=guest@adsl-69-109-40-118.dsl.irvnca.pacbell.net)
23:23.48pr0mtruly.
23:24.20*** join/#asterisk zamsler (n=zamsler@c-67-184-243-204.hsd1.il.comcast.net)
23:24.21Ariel_along again naturally
23:25.30DrukenHMEdoes anyone here have any experince with cell sites?
23:25.42*** join/#asterisk Rob- (n=robbie@haylott.plus.com)
23:30.01*** join/#asterisk kiwnix (n=egarcia@82.158.158.62)
23:35.30*** join/#asterisk zamsler_ (n=zamsler@c-67-184-243-204.hsd1.il.comcast.net)
23:35.42FuzzyCatbkw_, ....
23:38.38hugo-v6dunno why but i always rea pussycat
23:40.35FuzzyCatmeh!!
23:42.28fugitivolol
23:43.11hugo-v6no i wont say what im thinking right now :]
23:43.41kusznirhow do I tell asterisk console to show me every request recieved from remote clients/trunks?
23:44.04jskcrsip clients?
23:44.13kusznire.g., when I dial a number from a connected softphone that is invalid, the console shows nothing, but the softphone gives me a busy tone.
23:44.26jskcrsip debug
23:44.39kusznir(the problem here is that I'm dialing something I think is valid, so I'm wanting to see debugging output to se why astrisk doesn't think that number is valid)
23:44.40jskcrand asterisk -vvvvvvvvvvvvvvvvvvvvvvvvc
23:44.48kusznirI did 7 v's.
23:45.11*** join/#asterisk linuxwaa (n=linuxwaa@adsl-69-109-40-118.dsl.irvnca.pacbell.net)
23:45.30jskcruse sip debug too
23:45.31Ariel_kusznir, do you want to debug sip /zap or Iax2?
23:45.34kusznirI also have an iax2 trunk that asterisk does show registering to, but when I call in through that trunk, I get endless ringing and the console doesn't show anything.
23:45.34file[laptop]sip debug is good
23:45.46kusznirI guess I'm looking sip and iax2 (independant problems)
23:45.47Ariel_iax2 debug
23:46.04Ariel_once you finish then iax2 no debug
23:46.07Ariel_same with sip
23:48.32kusznirhmm..that wasn't showing me what I'm looking for.  I guess I'm looking to debug the dialplan rather than sip right now...the debug did reliably show that asterisk was recieving the call request and responding with a 404 error.
23:48.52jskcrwell theres your problem ;)
23:49.01*** join/#asterisk tugalone (n=tugalone@pcp0010318441pcs.avenel01.nj.comcast.net)
23:49.02kusznirI want to know *why* asterisk believes that to be a 404...is there any debugging for the dialplan or anything else that would show this?
23:49.29kusznirjskcr: I knew that already :) (the sip phone was showing me a 404).
23:50.16kusznirFor exaple, when I *am* in a context (dialing 1000 with the sample config files, for example), the console spits out info reguarding what context its in, etc.  I'd like to se something of that sort for calls that aparently arn't ending up where I belive they should.
23:50.39jskcrcan you dail anything in the pbx from the sip phone like a echo test?
23:50.39DrukenHMEkusznir: set verbose 12
23:51.35kusznirhmm...still no output
23:51.45*** join/#asterisk akrall_ (n=akrall@201.144.58.186)
23:51.55jskcrsounds like the phone may not even be getting registering to the server
23:51.58kusznirYea, I can dile the demo stuff.
23:52.00*** join/#asterisk liberie_ (n=liberie@dsl027-160-248.atl1.dsl.speakeasy.net)
23:52.03akrall_Hi People....
23:52.15kusznirI can dial digium (500), voicemail, etc.
23:52.18akrall_anybody gotten unicall to compile under cvs-head?
23:52.24jskcris it another sip phone your trying to dial
23:52.35kusznirI've set up an iaxtel account, and tried modifiying the dialplan to route tollfree numbers via it.
23:52.53jskcrahh then your problem is in your dial plan
23:52.55jskcrpaste bin it
23:52.58kusznirWhen I try and dial a toll free number, I just get a 404 and no output on the console as to why.
23:52.59jskcrpastebin.ca
23:53.26jskcrso it could be a problem with your iax or a problem with your extensions.conf
23:53.59*** join/#asterisk clyrrad (n=ddd@CPE0050bae8d02c-CM0011aea484a4.cpe.net.cable.rogers.com)
23:54.51*** join/#asterisk elwin (n=yellowsn@dsl-213-134-245-123.solcon.nl)
23:54.52KaBewMi've not been able to register with iaxtel for months
23:55.07jskcriax show peers and iax show registry
23:55.08KaBewMi now use sipphone.com to route my toll free
23:55.21KaBewMuse to work
23:55.25KaBewMnot sure what happened
23:56.33kusznirhttp://pastebin.ca/22198
23:57.09kusznir(most of it is still the standard demo...The only things I've really modified were the IAXTEL username/pass (which I removed before pasting), and the tollfree context.
23:58.51*** join/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com)
23:58.53kusznirMy sip phone (xten-lite) is registered as 1234.  asterisk sees that and is happy, and lets me dial into the demo.
23:59.50HmmhesaysUgh I suppose I should start writing this dialplan
23:59.58veteranis there any way to check to see if a queue's agents are online in a dialplan?

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