IRC log for #asterisk on 20070331

00:01.03*** join/#asterisk luke-jr (n=luke-jr@2002:1891:f663:0:20e:a6ff:fec4:4e5d)
00:05.05*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
00:11.54*** join/#asterisk dseeb_ (n=dcb@CPE-124-177-38-240.vic.bigpond.net.au)
00:12.34[hC]Is it possible for queue agents to continue to be run while an announcement is being played?
00:12.44[hC]i get the impression that while an announcement is being played, no agents are being rang.
00:13.56*** join/#asterisk tonyb2006 (n=tonyb@2002:4571:29c2:0:0:0:0:1)
00:14.28*** join/#asterisk luke-jr (n=luke-jr@2002:1891:f663:0:20e:a6ff:fec4:4e5d)
00:27.11*** join/#asterisk LennonNZ (n=admin@port-60-234-128-20.bitstream.orcon.net.nz)
00:27.34LennonNZHello. has anyone got asterisk to peer with a Siemens Hiq8000 properly?
00:27.37LennonNZI am having troubles
00:28.03LennonNZit registers ok, but I can only call from hiq->asterisk. but asterisk->hiq gives wrong password error
00:35.25LennonNZah. worked it out..
00:37.10*** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net)
00:37.17saint_hi all
00:37.35zoahi
00:38.01saint_so i am following the doc, and I am trying the simple SIP trunk.. with no success... I just configured the sip.conf , and should I expect asterisk to work ? or should i do something else  ?
00:42.06*** join/#asterisk wubba (n=kmurrey@cable-76-215.sssnet.com)
00:42.11zoayou need to alter extensions.conf too
00:42.49wubbaIs this discussion here just about develpment of * or do you all talk about real world installations?
00:43.05zoathis channel is more like a support channel
00:43.11zoathere is different channel for development
00:43.25wubbaok
00:43.43saint_zoa, i m doing a real world install
00:43.48saint_asterisk <--> Alcatel
00:43.51_DAWAnyone interested in helping test some SIP DID's?  I need some regular, reliable, intelligent  feedback on voice quality and am willing to give away 10 for 6 months.  Can provide number to most of the US.
00:44.11wubba_daw - I could help you
00:44.23_DAWmessage me off channel if your interested.
00:44.26wubbaSaint - tell me about your install.
00:44.27_DAWplease
00:44.53wubbasent
00:45.51saint_wubba, I have a company with a Alcatel PBX in the US
00:45.58saint_they have clients all over the world who use sip
00:46.02saint_inside it works
00:46.08saint_but the alcatel sip can't NAT
00:46.12wubbaSo your using *
00:46.30saint_so I am setting up a Asterisk box which is able to NAT sip clients, and I'll try to use a SIP trunk between it and alcatel
00:46.50wubbainteresting
00:47.04saint__DAW, are you saying that you are providing a SIP trunk ?
00:47.14saint_wubba, yeah, if it works, I'll push asterisk more to my clients, lol
00:47.19saint_I have a site with 14,000 extensions
00:47.23wubbaWe have been messing around with the Trixbox/Freepbx.  We recently installed the Asterisk Now and it looks very good.
00:47.45saint_Sweet..
00:47.46_DAWsaint_ inbound.. yes.
00:47.58saint_Alcatel came up with 7.0 and SIP trunks, but it looks like none in the US tried it yet
00:48.00wubbaWe sell key systems now - but have started to run into some great opportunities for some * boxes.
00:48.11saint_I did a 7.1 install at AT&T and they were trying sip trunking and gateway in their lab
00:48.23saint_wubba, really ?
00:48.28saint__DAW, what do you need ?
00:48.35saint_wubba, like what  ?
00:49.29wubbaWe have a school system that has about 30 sites throughout our area.  They are looking to tie two of them together, and them move the rest later on.
00:49.45saint_cool... is it in the US ?
00:49.49wubbayes
00:50.04saint_what school district  ?
00:50.21wubbaIt's not really a 'district' - its a company that does Charter Schools.
00:50.28saint_oh, ok ..
00:50.52wubbaBut they have 30 sites now and are expanding
00:50.55saint_I'm just starting with Asterisk (a couple of hours ago.) It looks pretty neat
00:51.08saint_how are they connected together ?
00:51.32wubbaThey are not now - they are all on key systems.
00:51.37saint_oh, ok
00:51.43saint_have them switch to Alcatel - Lucent :-D
00:52.03wubbaWe are going to put a data T-1 in and do sip trunks with DID to each location.
00:52.38saint_then how do you connect the key system to asterisk  ?
00:53.09wubbaWe are replacing the key systems with *
00:53.15saint_oh !
00:53.32saint_I thought key system <-> * <---- SIP ---> * <-> Key system
00:53.41saint_so what kind of phones are you going to give to the clients  ?
00:53.45wubbaPolycom
00:53.48saint_SIP ? IP ? Digitals ?
00:53.54saint_which model ?
00:53.55wubbaAll SIP
00:53.59wubba330
00:54.50saint_let me check on line
00:54.57*** join/#asterisk Fieldy (i=Yi5dPltt@gentoo/contributor/Fieldy)
00:55.00saint_soundpoint ip 330 ?
00:55.54wubbayes
00:56.23bsd_techwubba wubba
00:56.49saint_wubba, interesting...
00:56.55wubbahey bsd
00:57.02saint_the good feature with alcatel, is that they have a keyboard on the screen, and you can do dial by name
00:57.20saint_the latest ip touch phone has a color screen too , where you can feed video
00:57.26saint_like a doorcam ..
00:57.37wubbainteresting
00:57.49saint_yeah.. you can also program neat XML applications for it
00:57.50bsd_tech?
00:58.17saint_send/receive SMS, check the price of a stock, weather, other shit like that
00:58.33saint_whatever -almost- you can do in xml, you can do it on this phone
00:58.35saint_it s pretty neat
00:58.49bsd_techurl?
00:58.55saint_hold on
00:59.20bsd_techno now
00:59.24bsd_technow now now
00:59.27bsd_technow
00:59.31bsd_techlol
00:59.38saint_http://www1.alcatel-lucent.com/enterprise/en/solutions/mobility/on_site_mobility/iptouch_phones.html;jsessionid=HDX0W4OXDQLVVLAWFRUE1DNMCYWGI3GC
01:00.16saint_i have one on my desk right now :-D
01:00.31saint_+ they have bluetooth.. sswwwweeeeeeeettt
01:01.10bsd_techhow much
01:01.17saint_for you ?
01:01.23saint_or as if you were my customer ?
01:01.33wubbacost/retail
01:01.35bsd_techwich is cheaper
01:01.36*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
01:02.09bsd_techand the page does not display correctly
01:02.18saint_about 280 for my to buy, about 580 me to sell to you ... anywhere in between
01:02.38saint_but this is the top .. you have other IP phones with black & white screen
01:02.41saint_instead of the color
01:02.47saint_iptouch 4038
01:02.48Innatechkeerist. That's spendy.
01:03.09saint_yeah.... Alcatel-Lucent dude :-)
01:03.21wubbaThe phones are quite hideous looking.
01:03.58saint_nnnnaaaaa
01:04.02saint_better than ciscoc
01:04.04saint_cisco
01:04.04bsd_techthat 200% markup on price
01:04.07bsd_techouch
01:04.10saint_yup
01:04.19saint_but we give 35% to most of customers
01:04.31wubbaso your work for them?
01:04.44saint_used to. i work for a business partner now
01:05.06bsd_techdonate one for review. so I can write a article on it
01:05.13saint_hehehe
01:05.16bsd_techbut I need 30 days of testing
01:05.23wubbaI'd buy for $580 - then I have to put my markup on it - man that's expensive.
01:05.25Qwelldonate one for review. so I can burn it in effigy
01:05.27saint_you won't be able to test it without a PBX !
01:05.35InnatechNone of my clients would want all that, anyway. They just want the same functions that they're used to from various Nortel PBXes.
01:05.39wubbabsd has about 20 of them
01:05.56bsd_techI have a asterisk pbx
01:06.19saint_Innatech, because they never played with it
01:06.30saint_I went into a hospital where they changed from nortel to cisco (ip)
01:06.46Innatechsaint: very true, at least for some of them. But I service a lot of technophobes too.
01:06.47saint_I came and left a demo AFTER the cisco was installed... 6 months after, the removed the cisco to install Alcatel ..
01:06.54bsd_techthey should have saved thier money and gone polycom
01:07.04saint_I'm telling you.. gotta play with it, and check all the functionalities of the system .. it's a killer system .. running on linux !
01:07.16QwellThey should have saved their money and gone with asterisk, and reused the ciscos
01:07.29saint_yeah, could of.. but then... do you have dial by name ?
01:07.38QwellIf I write it, sure
01:07.40bsd_techasterisk is not ready for hospitals
01:07.49saint_it sounds stupid, but in a hospital connected to other hospital, where you have over 4,000 people ... dial by name suddently makes sense to have ..
01:07.51bsd_techstill work to be done
01:08.28InnatechYeah....again, my clients needs may not be the most complicated (definitely not hospitals) and for them Asterisk is just perfect.
01:08.45wubbaInnatech - what is your average install in # of phones?
01:08.53*** join/#asterisk piper69 (n=piper@unaffiliated/piper69)
01:09.07piper69hi all
01:09.11bsd_techwell I would like to get one of the phones to write a review
01:09.12Innatechhmm----probably 8 to 20.
01:09.19bsd_techget them to donate one
01:09.38bsd_techwont happen thou
01:09.39piper69i got offered a 5ESS DCS switch engineer today
01:09.52piper69am very excited
01:09.58InnatechThen I show up and add a few every so often. But they tend be in that range when I go out to install or convert.
01:10.30wubba[innatech] We are getting ready to really start pushing some of these on our current installs.  Are you using SIP?
01:10.39saint_bsd_tech, an alcatel phone ?
01:10.51Innatechwubba - yes, primarily SIP.
01:11.11wubba[innatech]  So are you replacing Key Systems?
01:11.20wubba[innatech] - What phones are you using?
01:11.36Innatechwubba - I'm either replacing key systems or setting up new small businesses.
01:11.46wubba[innatech] Excellent.
01:12.00Innatechwubba - I've had good luck with the Snoms, but I know some find them buggy or lacking in features.
01:12.22wubba[innatech] So what are you using as far as the * box - you just building something?
01:12.28piper69hopefully one day i will catch up with you guys on Asterisk
01:12.36Innatechwubba - A lot of small shops without much startup money can also be serviced by using ATAs with good old analog desktop speakerphones. This often provides better sound than a dirt cheap IP phone.
01:12.53Innatechwubba - I build a box for it when I can get it approved.
01:13.02wubbak
01:13.10wubba[innatech] Hope you don't mind the questions...
01:13.17Innatechwubba - otherwise we repurpose a WinXP workstation by adding a RAID card and installing CentOS or Ubuntu LTS>
01:13.28Innatechwubba - not at all.
01:13.48wubba[innatech] We have been installing mostly key systems - but I have had * running in our office for about 3 months now runnign with 5 SIP trunks.
01:14.38wubba[innatech]  So I have had our sales people call on a bunch of folks that we quoted lasted year that didn't do anything. Hoping that we can move them over to a full * install.
01:14.43InnatechIt's truly excellent for new/small businesses. People get very excited when they find out what they can get for next to nothing in captial investment.
01:14.51Innatechwubba - that's a very smart tactic.
01:15.20wubba[innatech] Especially since it was close to $300k in Quotes.
01:15.41Innatechwubba - very nice!
01:15.43wubba[innatech] So I have close to 100 or so quotes.
01:16.02wubba[innatech] How are you marketing?
01:16.22Innatechwubba - Heh. I wish I had leads like that. I don't do much marketing yet, mostly word of mouth in the professional communities I work with.
01:16.35Innatechwubba - marketing to attys and docs gets expensive fast.
01:16.52wubba[innatech] I paid for them... we belong to a service where we pay for leads - $29 a peice.
01:17.13wubbaBut when you sell one and make $2500 - it's well work ith
01:17.18Innatechwubba - What is your conversion rate like? If I may pry a little?
01:17.32wubbaTerrible -
01:17.36Innatechhehe.
01:17.53wubbaLike I said I have 100 that we are recalling - and I sold maybe 4.
01:18.02InnatechYeah, I end up doing work for almost everyone who calls. But it does get sloooow at times.
01:18.25wubbaAlot of people just kicking the tires
01:18.32InnatechI should get the marketing religion, but I'm hesitant to throw money at the problem.
01:18.50wubbaIt's actually just an investement.
01:19.15wubbaYou invest in it to make you more money.  It's hard at times to spend the money - but it usually helps.
01:19.23InnatechIt's true, I've heard that from everyone. I just have to make myself pick a venue and do it.
01:19.39InnatechAdWords seems like a waste, for instance.
01:19.44Innatechafk 1 min, brb.
01:19.45wubbaYes
01:20.42cpmwho's selling what?
01:22.30wubbaWe are just talking about small * installs 8-20 endpoints
01:22.30Qwell"corruped"?
01:25.22*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
01:25.44TripleFFFFanyone know why if i dont use fromuser= i get non authoriezed.. but if i use it then i loose callerid
01:30.56Innatech(BAK.) Yes, * for small or new businesses. And how to market it effectively.
01:33.12wubba[innatech] - In the US?
01:33.48Innatechwubba - Yes. Los Angeles.
01:34.18wubba[innatech] OK
01:34.47wubba[innatech] How many installs have you done?
01:35.26Innatechwubba - counting everything, including tiny installs for friends, probably 15.
01:35.43Innatechwubba - installs worth talking about-- probably 5.
01:35.57wubbaok
01:36.27Innatechwubba - essentially one every couple months for the last year.
01:36.31Innatechbrb
01:41.00*** join/#asterisk Defraz (n=t0tal@209.137.240.88)
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01:57.30Innatechbak
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02:10.41JTInnatech: so most customers can only cough up for a second hand or retasked machine?
02:11.17TripleFFFFwow i do 15 a day
02:11.18TripleFFFFlol
02:11.55JT15 what?
02:12.03TripleFFFFinstalls
02:12.12JTof what?
02:12.15TripleFFFFasterisk
02:12.15JTspecs?
02:12.16TripleFFFFlol
02:12.20JTLOLLOlollllk
02:12.20JTk
02:12.32TripleFFFFwas refering to mesage form innatech and wubba
02:12.38JTlol!
02:13.18JTand i asked for the specs, not more lols
02:13.27JTlike how many stations
02:13.32JTwhat level of features
02:14.40InnatechTriple: When they're first getting started and have no capital? Yes, that's right. Some of the others can pay for more, and I sell them dual proc dual core rack servers. :)
02:14.40Innatecherr, that should have been to JT.
02:14.40JTah ok
02:14.44InnatechTriple: I'd love to do 15 a day. If that's possible.
02:14.58JThow cheap can you make an asterisk install though, Innatech ?
02:15.00*** join/#asterisk techie (n=gus@voip.routedsystems.com)
02:15.11InnatechPractically free, save my time and cost to set up the trunks, etc.
02:15.11JTInnatech: must be 15 pretty simple installs
02:15.19InnatechJT: yeah, no kidding.
02:15.53JTInnatech: got to be some hardware cost unless you just give them VoIPoInternet and softphones
02:16.30InnatechYeah--practically free. And it is VOIP over the WAN and usually cheap ATAs with existing analog deskphones. For the cheapest installs.
02:17.09InnatechSo, the cost of the ATA plus (trunk costs / # of users) is the per seat cost, before they pay me.
02:17.42InnatechOf course, I advise them that they'll need to upgrade once cash flow happens. But its good for getting up off the ground with a new business.
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02:22.54JThrm i would never give a business VoIPoI only
02:23.14JTdoesn't matter how small they are, they're better off even with 1 analogue phone and phone line
02:23.20*** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca)
02:23.28JTunless they don't actually need their phone to work
02:23.38MrTelephonedoes anyone ever have a pri just stop working and you have to reload the drivers and restart asterisk?
02:23.57JTMrTelephone: what card?
02:24.04MrTelephonesangoma dual port pri
02:24.42MrTelephonehow often should asterisk restart the b channels?
02:25.09MrTelephonethere was no alarms showing
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02:26.17InnatechJT: Yeah, I give them the spiel about reliability, etc. Some of them care enough to spring for an actual phoneline and a TDM card, but most of them don't care. Ultimately its their decision. And most of them are good about upgrading once they are established and aren't worried about month-to-month survival.
02:26.48JThmm ok
02:27.08Innatech*digium card that is/
02:27.25JTsure
02:28.10MrTelephonehmmmm
02:28.21MrTelephonefunny stuff happening
02:28.23InnatechWhen you think about it, Asterisk gives you amazing capabilities compared to what most businesses at that level are working with.
02:28.35JTInnatech: but how much can you possibly make off a project like that?
02:28.42JTInnatech: definitely
02:29.14InnatechJT: More than you'd think. Especially since I usually set up the rest of their computer network at the same time.
02:29.29JThmm
02:29.34JTnumbers? :)
02:29.51InnatechJT: mmm.....I'd prefer to be vague about my actual rates....
02:29.56MrTelephoneas long as you can keep your pri up
02:30.02InnatechJT: but, a few hundred, usually more.
02:30.36InnatechJT: If I'm putting everything in for them (server, workstations, phones etc) its usually over 1K.
02:30.49JThmm not bad
02:31.01InnatechJT: Yeah. And my own overhead is *very* low.
02:31.09MrTelephone1 thousand dollars
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02:31.14MrTelephoneshit im wiring a hotl for 9 thousand
02:31.15Maghteridonhi all
02:31.31JTMrTelephone: how many extensions?
02:31.38MrTelephonejust runnign network cable
02:31.40MrTelephoneno phones or anything
02:31.45InnatechMrT: If I had a CAT5 license, I could clean up on that stuff. I have to farm it out at the moment.
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02:31.52JThow many extensions
02:31.54MrTelephone13 extensions pc, polycom 501 phones was like 7 grand
02:32.03InnatechMrT: I'm not into dealing with insurance and building contractors, tho.
02:32.04JTah only 13
02:32.16MrTelephoneyou need a cat5 license in the states?
02:32.20MrTelephoneahhh
02:32.23bsd_techwhere do you buy your phones MRT
02:32.39MrTelephonedeny it all if something happens
02:32.39MrTelephonej/k
02:32.46MrTelephoneingrammicro
02:32.52MrTelephonewholesaler
02:33.02TripleFFFF?
02:33.03MrTelephonethe digium card was the most expensive thing
02:33.05TripleFFFFsimple vanilla install
02:33.06bsd_techhttp://ipphone-warehouse.com/
02:33.06TripleFFFFyes
02:33.09MrTelephone24 port linksys poe switch was 600
02:33.11bsd_techthey rock
02:33.12TripleFFFFsorry was doing another ;)
02:33.16MaghteridonLet's say that you have at least 3 numbers on asterisk :x,y,z. Let's suppose X is calling Y; Y answers and then gives the communication to Z, so now X is talking with Z. How is it called this process that transfers the call once Y answered?
02:33.25bsd_techI love polycom
02:33.27MrTelephone200 for each polycome phone, 950.00 for a polycom voicestation 4000
02:33.28TripleFFFFtakes me around 15 minutes per get all libs/core+ fax up.
02:33.42TripleFFFFthen 30 min to 1 hour for all tdm stuff
02:33.44MrTelephoneso it adds up pretty quick
02:34.19bsd_techyou overpayed for the 501
02:34.19MrTelephone200 canadian
02:34.19bsd_techthey are down to amost 150 a piece now
02:34.19bsd_techok
02:34.20bsd_techCanuck
02:34.21JTTripleFFFF: what sort of faxing, fax to email, or fax over pri out to an analogue port?
02:34.22MrTelephonethey went down?
02:34.32TripleFFFFfax2pdf
02:34.34MrTelephoneim using spandsp works great
02:34.43TripleFFFFand fax to analog etc
02:34.48bsd_technvfax is also good
02:34.49JTTripleFFFF: what about sending faxes... email to fax?
02:34.54TripleFFFFhehehe
02:34.56TripleFFFFnot included
02:34.57bsd_techbut needs updating for 1.4.2
02:34.59TripleFFFFbut i can do
02:35.00TripleFFFF;)
02:35.17MrTelephonenvfax is another asterisk addin?
02:35.26MrTelephonewho here uses sangoma cards?
02:35.29anthonylare you using spandsl on asterisk 1.4.x?
02:35.32anthonylspandsp*
02:35.34TripleFFFFlol
02:35.38TripleFFFFnever.. not using 1.4
02:35.46MrTelephoneim using asterisk.1.4.11
02:35.49MrTelephonei mean 1.2.11
02:35.53TripleFFFFjust done a client that he pressed save on gui and it erased all the configs from extensions
02:35.59bsd_tech1.4.2 is very nice and we have spandsp faxing now
02:36.02MrTelephonebut im trying to develop NCS in MGCP
02:36.06anthonylah word, just checking a friend of mine was having issues with it on 1.4 yesterday
02:36.10TripleFFFFalso had intermittent registrations /authorisation issues
02:36.18bsd_tech1.4.2?
02:36.20TripleFFFFso we rm -rf /asterisk
02:36.23bsd_techor 1.4.0
02:36.31TripleFFFFand hmm cough.. installed opb
02:36.31MrTelephonewhere can i find out how my pri lost sync with asterisk?
02:36.42TripleFFFFcheck the util with it
02:37.06TripleFFFFthe apps that comes with the libs for the card ;)
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02:37.18TripleFFFFk friday night beer
02:37.20*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)
02:39.10MrTelephoneMar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: OOF alarm is ON
02:39.10MrTelephoneMar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: RED alarm is OFF
02:39.10MrTelephoneMar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: T1 disconnected!
02:39.10MrTelephoneMar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: AFT communications disabled!
02:39.10MrTelephoneMar 30 19:36:42 prdc-asterisk-1 kernel: wanpipe1: Starting TDMV 1ms Timer
02:39.11MrTelephoneMar 30 19:36:53 prdc-asterisk-1 kernel: wanpipe1: T1 connected!
02:39.22MrTelephoneMar 30 19:36:53 prdc-asterisk-1 kernel: wanpipe1: AFT communications enabled!
02:39.22MrTelephoneMar 30 19:36:53 prdc-asterisk-1 kernel: wanpipe1: AFT Global TDM Intr
02:39.27MrTelephoneit was connected but never restarted the bchannels
02:39.28MrTelephonewtf
02:39.37MrTelephoneis that the phone company screwing around?
02:40.26MrTelephonei'll have to install more stable zaptel drivers
02:40.32bsd_techrum and coke
02:40.56techieand pirates of the caribbean
02:41.08[TK]D-FenderMrTelephone, and stop SPAMMING THE CHANNEL
02:41.25MrTelephonesorry skippy
02:41.37MrTelephone7 teeny lines
02:41.42JunK-Yisnt spam.
02:41.45MrTelephone8 :-/
02:41.54[TK]D-FenderJunK-Y, Sure it is.
02:42.09[TK]D-FenderAnything over 3 is highly suspect and a block like that, hell yeah
02:42.13JunK-Yspam isnt unsolited publicity?
02:42.31[TK]D-Fender*I* didn't ask for it, now did I?  :)
02:42.44JTit's very simple
02:43.03JTIF lines.numbers => 3 THEN do not send to channel
02:43.28*** join/#asterisk Defraz (n=t0tal@209.137.240.88)
02:43.39MrTelephonemy information could help others
02:43.47MrTelephonenot like i pasted a bunch of C, eek
02:44.08JTstop defending yourself, and just admit you made a newbie mistake...
02:44.10MrTelephoneOOF is a loss of signal?
02:44.11JT~pb
02:44.15jbotmethinks pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
02:44.47JTthere is 268 nicks in here, it's not 8 teeny lines, it's 268 * 8 lines
02:44.50*** join/#asterisk f1r3w0rm (n=warirc@adsl-070-145-060-050.sip.bix.bellsouth.net)
02:45.42MrTelephonei've been using irc for 20 years
02:45.59JTgreat
02:46.00*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)
02:46.02JT:)
02:46.06MrTelephonea flood is like maxchars * maxlines
02:46.19MrTelephoneusually ends up with a disconnect saying queue exceeded
02:46.20MrTelephonelol
02:46.34MrTelephoneit happened to me before many moons ago
02:46.35JTno, that's an IRCD enforced excess flood kill
02:46.49MrTelephonei understand anyways
02:46.49JTi'm talking about what humans determine to be a flood :)
02:46.58bkruse_homerussellb: you there/
02:47.01bkruse_home?
02:47.18MrTelephoneat lease give me a break and subtract 260 people as they are drones :)
02:47.32f1r3w0rmhello all
02:50.46MrTelephonewhy is 802.11b/g so flakey these days
02:50.50MrTelephoneis it the interference from others?
02:51.07bkruse_homeMrTelephone: actually, pine trees will own your wireless, seriously
02:51.07bkruse_homelol
02:51.30f1r3w0rmi know
02:51.39f1r3w0rmi live around a bunch of em
02:51.57bkruse_hometrue isnt it?
02:52.10f1r3w0rmi can go into city and i have perfect sig i go home im 10 ft from router and i got 20 % sig
02:52.12[TK]D-Fenderbkruse_home = Arborially challenged :D
02:52.18bkruse_home:P
02:52.22bkruse_homei just hate trees :P
02:52.39Strom_Mlet's send bkruse on a vacation to the pacific northwest
02:52.59bkruse_homeim down
02:53.08bkruse_homenah, i dont need a vacation, just a full 8 hours of sleep
02:53.09Strom_Mall trees
02:53.13Strom_Mnothing but trees
02:53.13bkruse_homeNO!
02:53.14InnatechPacNW has AWESOME trees. Heh.
02:53.23bkruse_home</3 trees
02:53.27bkruse_homeive had big tests this week :[
02:53.54f1r3w0rmlol u think Innatech try livin in south ms
02:53.59MrTelephonethats right i have a 2km 900mhz link that the pine trees are devastating
02:54.13MrTelephonei ordered a nice yagi to put up next week to see if it makes a differnece
02:54.45fx0what difference will it make with the pine trees ?
02:55.07MrTelephoneits at -80. 6 more db will bring me to an acceptable -75
02:55.26MrTelephonethe old antenna is 9db gain and the yagi is 14db
02:55.36MrTelephoneso i guess thats 5 db difference
02:55.39fx0i'd rather buy a chainsaw :)
02:55.43MrTelephoneme too
02:55.58MrTelephonewhats funny is that he radios are pretty high
02:56.12MrTelephone60 ft at one side and 30 ft on the other
02:56.17MrTelephonethe trees are roughly 25-30ft
02:56.33bkruse_homei think im going to play counterstrike, or work on clientside graphing for asterisk......hmm
02:56.35MrTelephoneelevation difference is nil
02:56.50*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
02:57.15f1r3w0rmi use linksys wrt 54 g
02:57.24f1r3w0rmi got a custom firmware on it
02:57.43f1r3w0rmcranked the antennae power to 126
02:57.55f1r3w0rmand i actually maintain connection now
02:58.45MrTelephonelol
02:58.51MrTelephoneby knocking out the neighborhood
02:59.05MrTelephonei installed some nice proxim access points.. 300 each or something.. and they are flawless
02:59.14MrTelephonebut who can afford that at home?
02:59.19InnatechCranking up TX isn't always a good idea. Then the access point just steps all over the client transmissions.
02:59.37MrTelephoneyour raising your noise floor
03:00.30MrTelephonei installed some firmware too and it didn't make a difference when i jacked up the xmit power.. I was watching it on netstumbler..
03:00.32MrTelephonebut who knows
03:00.36*** join/#asterisk dj-fu (n=ajc@unaffiliated/dj-fu)
03:00.49MrTelephoneqdon't give yourself cancer
03:01.13Innatechyah. I've never found anything but a modest increase in TX power on WRTs makes a difference. After that, you're just cooking nearby soft tissues. ;)
03:01.40InnatechHey, look down this waveguide! >sizzle<
03:01.53MrTelephoneyeah there is a guaard on the chip or something
03:02.03MrTelephoneFCC wouldn't allow it
03:02.23InnatechWell....in theory.
03:02.25*** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk)
03:02.32MrTelephoneim having a lot of issues with the new wrt54g's, have to power cycle them all the time?
03:02.42InnatechYeah, me too, actually.
03:02.48InnatechWTF is up with that?
03:02.57MrTelephonethe only theory i know is plug in
03:03.04MrTelephonenot sure
03:03.11MrTelephonethe last 2 i sold were shit
03:03.28MrTelephonedlink used to give me problems like that
03:03.46InnatechYeah. I've had to warranty return a bunch lately. Serious pain and a waste of time, and makes me look like a chump w/the clients.
03:03.51MrTelephonei think the wireless part is crashing the unit
03:04.02MrTelephoneyeah i hear you
03:04.11MrTelephonego with dlink for now
03:04.26InnatechI had one recently where basic switching was failing. QC issues @ the factory, maybe.
03:04.35MrTelephonemaybe
03:04.35InnatechYeah, dlink is OK.
03:04.45MrTelephonei know they changed their software platform
03:05.09MrTelephonenow if you want one that will take linux you have to buy the wRT54G-l or something
03:05.09Innatechyeah, and the board design changes too. My older WRT54s are my best ones.
03:05.25MrTelephoneand i bought a rangebooster model and it sucks too
03:05.36MrTelephonedropping connections all the time, doesn't have the qos option in the firmware
03:06.06MrTelephonei just hope their switches arn't as crappy because I'm using a lot of them
03:06.33InnatechMost of their *managed* switches are rebadged low-end Cisco gear, which is OK.
03:06.43MrTelephonei feel your pain though.. I don't sell much stuff but when I do the last thing I want to do is deal with returns
03:06.47InnatechTheir dumb switches.....not so hot.
03:07.17InnatechYeah. It's awkward. That's why I try not to resell equipment as much as possible.
03:07.18MrTelephonewell i went with some 24 port cheapos that were 110 brand new
03:07.30*** join/#asterisk pkempgen (n=pkempgen@AC9E8CD7.ipt.aol.com)
03:07.36MrTelephonetrial and error I guess
03:07.45InnatechWell, they should be OK. Honestly, tho, the linksys stuff on that level is no better than, say, TrendNet.
03:07.48MrTelephonehey do u have the same problems with the access points?
03:08.04InnatechI dunno, I use WRT54s *as* access points. :)
03:08.15MrTelephoneohhh
03:08.15MrTelephoneok
03:08.27*** part/#asterisk pkempgen (n=pkempgen@AC9E8CD7.ipt.aol.com)
03:08.44MrTelephonei installed some firmware once and couldn't get wap working so i took it off
03:08.50MrTelephoneWPA rather?
03:09.22f1r3w0rmi use the dd-wrt firmware
03:09.26f1r3w0rmmicro
03:09.31f1r3w0rmits a nice one
03:09.58f1r3w0rmwrt54G cant handle the standard >.<
03:10.07f1r3w0rmnot the older models ne wyas
03:10.40f1r3w0rmif u get a newer wrt54 u install the dd-wrt standard it had ipv6 support
03:11.29MrTelephonesweet
03:11.36MrTelephonei never even looked at ipv6 yet
03:11.45MrTelephonei don't think its supported in canada yet
03:11.46MrTelephonelol
03:11.56f1r3w0rmi use it mainly for lan
03:12.02f1r3w0rmspecially @ work
03:12.16f1r3w0rmwe jus installed 45k computers
03:12.57MrTelephonezaptel-core is zaptel-base in 1.2.16?
03:13.18MrTelephone45K extended memory?
03:13.30MrTelephonej/k
03:13.32dj-fu45,000 computers?
03:13.38dj-fuThat's a huge company
03:13.40dj-fuin one building?
03:13.45MrTelephonetheres no way
03:13.50dj-fudude :\
03:13.50coppicewas that before or after lunch?
03:14.04MrTelephoneim not understanding the concept of vlans
03:14.05f1r3w0rmthat was 3 weeks work
03:14.11MrTelephoneif you have one router routing the vlans
03:14.14JTMrTelephone: very simple
03:14.19MrTelephonethen isn't that physical link easily saturated?
03:14.33JTdepnds on the links
03:14.50dj-fuyou use a layer4 switch and loadbalance across a few routers.
03:14.54dj-fudon't be daft :0
03:14.59f1r3w0rmdj-fu
03:15.00MrTelephonei guess
03:15.04MrTelephoneseems like a pain
03:15.08dj-funot really
03:15.14dj-fuless of a pain that non vlanning
03:15.15f1r3w0rmits a call center for ATT
03:15.37fx0fully voip ?
03:15.40JTf1r3w0rm: 45k PCs at one site?
03:15.44f1r3w0rmya
03:15.50dj-futhat's a big ass building.
03:15.52f1r3w0rmya
03:15.53f1r3w0rmit is
03:15.54JThow big was the place?
03:15.55dj-fulink
03:16.10f1r3w0rmits in downtown Dallas
03:16.31dj-fuTRUE BRO?
03:16.35f1r3w0rmtrue
03:16.44dj-fuI'm from texan.
03:16.45fx0how long did you take
03:16.53JThow many employees do they have their at one time?
03:17.07JTthere, even
03:17.07f1r3w0rmiuno i was jus a tech installing the shit
03:17.21dj-fuhow big of a team to install 45k pc's?
03:17.24dj-fuwhat deployment tools?
03:17.24MrTelephoneits more like downtown world trade center building #3
03:17.26dj-fuwhat OS?
03:17.38coppicelet's see. a 45k seat call centre would emply >100k people. that's about the same as the total call centre business in Bangalore :-\
03:17.55f1r3w0rmall we did was roll out hardware
03:18.06f1r3w0rmanother team did setup
03:18.17JTi'm calling bullshit
03:18.21JTmaybe 4500 stations
03:18.24JTnot 45000
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03:19.26fx0what type of hardware did you use for asterisk, i cant even imagine
03:20.06[TK]D-Fenderfx0, Who said this had anything to do with*? :)
03:20.16blitzrageincase anyone missed it earlier: http://digg.com/software/Asterisk_Users_Weekly_Friday_Conference_Podcast_at_12_30_PM_EDT
03:20.19fx0lol dont know man, that black container from sun comes in mind, blackbox.
03:20.46JTeh, you can handle 45000 users in 1 rack cabinet
03:20.56MrTelephoneim calling that im upgradeing to zaptel-1.2.16
03:21.14MrTelephoneplease don't judge me becaused I have a sangoma card.. I bought 3K worth the digium already
03:21.18JTwell, depends what they're using though
03:21.25JT45000 telephony users...
03:22.20blitzrageMrTelephone: it's fine -- whatever works
03:22.24MrTelephonei feel like bcause i had a problem with my pri card noone cares because I didn't buy digium
03:22.54[TK]D-FenderMrTelephone, So whats the issue.  Your earlier over-paste just looked like someone pulled the plug and jacked it in 11 sec later
03:23.02fx0true, we will call jan dowse @ sales to stab you with an icepick while you are sleeping
03:23.31Strom_Mhahaha
03:23.38JTMrTelephone: no, people use both cards, just the traffic flow is high in the channel
03:23.46Strom_Mimagining jan doing that is amusing :)
03:23.57InnatechWhile everyone's talking, my oft repeated legacy question: any one have experience interfacing * with Meridian/Nortel?
03:24.36[TK]D-FenderInnatech, Over FXO, I do
03:25.12InnatechTK: Is it as a terrible pain in the rear as I'm imagining it to be?
03:25.36[TK]D-FenderInnatech, What are you trying to acheive?
03:27.48InnatechI'd like to use * to control the extensions on the meridian, and to allow it to access VOIP trunks through Asterisk. Honestly, I'm a little vague on what *is* possible. In short, one of my close family members has a big legacy meridian PBX and wants to salvage as much of it as possible while moving forward into VOIP.
03:27.55MrTelephoneok i'm just checking
03:27.55MrTelephonesome drunk guy is probably tripping over my cords
03:28.21MrTelephoneasterisk with fxs cards i guess
03:28.30*** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net)
03:28.41fx0a t1 card will do the job.
03:28.51Innatechyeah...that's more or less what I figured.
03:28.58JTMrTelephone: your server as in a place that a drunk guy can trip over it?
03:29.13fx0meridian cards are way more expensive tho.
03:29.14[TK]D-FenderInnatech, how many phones/lines?
03:29.15InnatechThey originally wanted to focus on resuing the phones, but I gently suggested that was the wrong focus. ;)
03:29.24InnatechGah. A lot of phones.....
03:29.31MrTelephonei work in a native reserve
03:29.40MrTelephonea native has keys to my office
03:29.41[TK]D-FenderInnatech, NUMBERS please.  What kind of lines, and how many.  How man ext?
03:29.48MrTelephoneanything is possible
03:29.54MrTelephonefriday night :-/
03:29.55InnatechHeh, I'm thinking.
03:29.58JTMrTelephone: i see...
03:30.16fx045k lines!
03:30.18fx0omg
03:30.25MrTelephonei don't think sangoma likes the 1.2.16 zaptel :(
03:31.01InnatechI think its around 25 lines or so. I'm not sure how they're coming into the switch. Roughly 35 extensions....at any given time some of them are unplugged.
03:31.54InnatechSome of the lines are parcelled out to subtenants, so it gets a little confusing when they try and explain it to me, especially since they don't really know either.
03:32.42InnatechBut speaking generally: is the meridian hardware nice enough to bother with holding onto? I have no basis for comparison for older systems.
03:33.17Strom_MInnatech, only 35 extensions?  you sure it's not a Norstar?
03:33.53InnatechNorstar sounds familiar, but the phones and cabinet of the PBX say Meridian.  I'll go take a look at it, its around the corner.
03:34.39Strom_MInnatech, is it a fairly large standalone cabinet, or is it fairly flat modules on the wall?
03:35.30MrTelephonenortstar nortel
03:35.32MrTelephonesame crap
03:35.36InnatechStorm--you called it. Flat modules, open the cabinet and it says Norstar.
03:35.38MrTelephonereliable but overprices
03:36.04InnatechWorth holding onto? Better to abandon?
03:36.06MrTelephonei ordere an adit 600
03:36.20InnatechIf I did a conversion, I'd be doing it for free....so.....be painfully honest. :)
03:36.23JTMrTelephone: new or second hand?
03:36.50[TK]D-FenderInnatech, Norstar is annoying pile of crap.  Ditch it... it'll cost you less.
03:36.58MrTelephonenew
03:37.06JTMrTelephone: how much are they new?
03:37.24MrTelephone1600 canadian for 16fxs and dual t1
03:37.25[TK]D-FenderInnatech, about 5K in gear + PC and its over with
03:37.39InnatechYeah, OK. That was my instinct, thanks.
03:37.50MrTelephonesupposed to be good equipment
03:38.18JTMrTelephone: dual t1? only 16 ports....
03:38.22Strom_MInnatech, yeah, ditch the Norstar
03:38.27Innatechcool.
03:38.28MrTelephoneyeah because it supports up to 48
03:38.32MrTelephoneor something
03:38.37JToh right
03:38.45JThow much does the adit cost with 48 fxs ports?
03:38.59MrTelephonenot sure i thinjk the blades are like 3 or 4 hundred each
03:39.05JThmm
03:39.07MrTelephone?
03:39.13JThow many ports a blade?
03:39.16MrTelephone8
03:39.20JToh
03:39.23MrTelephonei bought it from texas actually
03:39.29f1r3w0rmNight All
03:39.34MrTelephonevox technologys or soemthing
03:39.48JTi'm in australia, so getting channel banks is harder
03:39.51JTno-one uses them
03:40.02cpmsell ya mine
03:40.04cpm:)
03:40.11JTheh
03:40.11MrTelephonethe office nextdoor just bought a brand new meridian system
03:40.12cpmI gotta pile of 'em here that suck
03:40.19JTwhat model?
03:40.24cpmCAC
03:40.27MrTelephonei think so
03:40.31JTwhy do they suck?
03:40.33cpmCAC
03:40.34MrTelephonewith the call pilot
03:40.37cpmthe suck model
03:40.47JTi thought CAC wasn't that bad
03:40.53MrTelephonethey are good, but rather have installed a crashing asterisk box
03:40.54MrTelephonelol
03:40.55cpmCarrier Access Group, the sucky ones
03:41.20MrTelephonei need something so i can power cycle a computer
03:41.23cpmseriously, I have a closet full of these damned things
03:41.37JTMrTelephone: ip powerboard
03:41.38MrTelephoneover the internet
03:41.48JTcpm: what bit sucks?
03:42.00cpmthese are the CB1s
03:42.04MrTelephonecuz i just crashed my computer at work now i have to drive 20km to reboot it
03:42.04InnatechMy relative's company bought the Norstar back in the 90s. It was a  *ton* of money back then, and it has done alright over the years. But it's good to get some endorsement on getting rid of it.
03:42.14InnatechAnyway, thanks everyone. TTYL.
03:42.22cpmI think they were custom burned for the telco I got them from,
03:42.38JTMrTelephone: you can get ip powerboards
03:42.42cpmI have a CB2 that sucks less
03:42.47cpmthats the one I use
03:42.50MrTelephoneim gonna buy one
03:43.08JTMrTelephone: they cost about the price of a new desktop pc though
03:43.19MrTelephoneyeah i know
03:43.23MrTelephonesame with those nice ip kvms
03:43.27MrTelephone2k
03:43.29cpmI've seen'em for as little as $250 or so
03:43.30JTyeah
03:43.42MrTelephoneexpensive crap
03:43.46cpmjust for an IEC power interrupter
03:43.50MrTelephoneanyways gotta go for a drive
03:43.54cpmn'joy
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03:50.49coppicedata centre staff will interrupt your power for free.
03:51.02coppicerestoring it can cost, though :-)
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03:59.38cpmmy local power company does it for me.
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04:35.20SECGODanyone have success with Cisco 7960 phones with Asterisk ?
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04:43.31fx0thousands of times
04:44.31fx0whats up
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05:17.07[[blah]asfdi added accountcode = dialer to my extensions in sip.conf. but when I dial those extensions there is nothing added to the cdr with that account code. is that not the right way to do that?
05:22.22dc3aesim having a really weird sporatic bug here guys.. got an IAX trunk to NuFone from our PBX and completely randomly (about 30% on average) of the time it the CPN/CID is being dropped and it shows up as "unavailable" to the dialed party.. but in the nufone logs it in fact shows that we set the CID properly.. anyone heard of this?
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05:22.51[[blah]asfdI just left nufone.. i was very unhappy with their service.
05:23.17[[blah]asfdVERY unhappy!!!
05:24.27[[blah]asfdI did not think that nufone supported outbound caller id
05:26.38dc3aesthey do, and let you set it at will... however if its this sporadic whats the point?
05:27.37[[blah]asfdyeah, I had them for 4 months and had nothing but "sporadic issues" and could never get them to answer trouble tickets or calls.
05:44.27[[blah]asfdso i am setting accountcode = dialer in sip.conf, but when I do a NoOp(${ACCOUNTCODE}) it displays nothing. what am i doing wrong?
05:53.07SwKtry getting rid of the spaces aroudn the = sign
05:54.33[[blah]asfdno difference.
05:54.41[[blah]asfdstill does not display properly.
05:55.20[[blah]asfdis account code applied to inbound calls to a device? or only outbound?
06:04.50dlynes_laptop[[blah]asfd: only outbound for sip
06:05.01dlynes_laptop[[blah]asfd: when you set it in sip.conf that is
06:05.49dlynes_laptop[[blah]asfd: its purpose is for billing, which would have limited use for inbound (only on systems that charge both for outgoing and incoming calls)
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06:11.10DoDaT69asdf
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06:22.11rollergrrlAnyone sober?
06:22.27dlynes_laptopnope
06:22.38Teneyus
06:23.02rollergrrlI'm barely able to type
06:23.13rollergrrland I'm supposed to fix a serve outage
06:23.48Teneack.  that's trouble.
06:24.10rollergrrlDrinking coffee like mad
06:24.16rollergrrldamn birthday parties
06:24.47rollergrrlI got a 1080p LCD 47" today though
06:24.52rollergrrlit's happy!
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06:29.16dlynes_laptoprollergrrl: great for administering asterisk, i would imagine :0
06:34.54fx0it is a tv.
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07:06.56tzafrir_laptopanybody here with some understanding of ISDN?
07:07.22tzafrir_laptopwhat does 'pridialplan=unknown'  do?
07:08.01tzafrir_laptopI know it technically sets the Type Of Number . But what does this do in practice?
07:08.32tzafrir_laptopShould I use "unknown" or national/international/local ?
07:14.00JerJertzafrir_laptop: really depends on what the provider of the ISDN expects
07:15.23tzafrir_laptopJerJer, it is actually ISDN BRI providers, in two different countries (Italy and Australia). I tried messing with local, national, etc, but no luck: I could dial national numbers, but not to mobile phones.
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07:18.18JerJerstrange
07:19.24tzafrir_laptopI figure that that telco is more tolerant to ISDN phones, and thus tolerates "unknown"
07:19.59tzafrir_laptopBut this is just a hunch
07:21.04tzafrir_laptopBut I don't really understand why I should have this knowledge. Isn't it the job of my provider?
07:21.21tzafrir_laptopI guess I'm not familiar enough with how telcos work
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07:52.22DarKnesS_WolFwhat is group paging means ?
07:55.02piper69DarKnesS_WolF: i would think paging a group of users at once
07:57.11DarKnesS_WolFpiper69: yes what is paging i'm trying to dig it in voip-infoorg
07:58.39DarKnesS_WolFahh got it
08:05.28*** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za)
08:05.36SoftIcehi qualify=yes what does this do exactly?
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08:20.16mvanbaakit will send a SIP OPTIONS packet every now and then
08:20.26SoftIceto see latency, etc ?
08:20.30mvanbaakthat waay asterisk knows the latency between itself and the phone
08:20.43SoftIceor the latency between itself and the trunk ?
08:20.52mvanbaakif the latency is more then 2 seconds asterisk will mark the phone unreachable
08:20.56SoftIceis it advised to use when using nat and using a trunk
08:21.10SoftIcemvanbaak: but what about trunks to carriers
08:21.13mvanbaakSoftIce: yeah s/phone/trunk/
08:21.15SoftIcewith latency of 50/60ms ?
08:21.26SoftIceshould i disable qualify then ?
08:21.34mvanbaakno
08:21.44SoftIcebut i dont want it to say its unreachable
08:21.54SoftIceid prefure it to try everything first
08:22.05mvanbaakit only goes to unreachable if the latency is above 2 seconds
08:22.24SoftIceahh, 200ms
08:22.30mvanbaak2000 ms
08:22.35SoftIceahh, I see
08:22.45SoftIceso allways best to use qualify then ?
08:22.55SoftIceany other tricks to help with carriers using voip
08:22.56mvanbaakif nat is involved, yeah
08:23.07SoftIcebecasue i sometimes get all curcuits are busy
08:23.19mvanbaakthe option packets will keep the nat state open in you nat device
08:23.52SoftIceand the options packet is built into qualify ?
08:23.56SoftIcewell what qualify sends out?
08:23.58mvanbaakyeah
08:24.02SoftIcegreat
08:24.03piper69DarKnesS_WolF: to be honest with you i really don't know shit about asterisk. i am dying to learn
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08:24.12SoftIceso thats about the best effor I can give to sip with nat and using a carrier ?
08:24.35mvanbaakif you get 'all circuits busy' you can check to see if the trunk is marked 'unreachable'
08:24.35SoftIceim using a dynamic ip so I have externip=hostname and have externrefresh=10
08:24.41piper69i want to go as cheap as possiable
08:24.45mvanbaakif not, the remote end might be on the phone indeed
08:24.55SoftIcemvanbaak or else does it mean that the far end doesnt have any trunks available?
08:25.01mvanbaakor...... let's hope not.....the carrier has no more free outbound lines
08:25.02SoftIcemvanbaak: ahh
08:25.07SoftIcemvanbaak :)
08:25.08SoftIcecould be
08:25.40SoftIcemvanbaak: any idea what might cause this then, im on a call and I get a break for a second I an hear the person or visa versa then it kicks in again?
08:25.45piper69mvanbaak: what do you mean by carrier, is it the T1
08:25.55SoftIcepiper69: im using a carrier for voip
08:25.59SoftIceto other countrys
08:26.26mvanbaakSoftIce: that sounds like a bandwidth issue
08:26.44SoftIcemvanbaak: even though the call is crystal clear?
08:26.47piper69SoftIce: oh, is it possable i can know the name or wesite for them
08:26.50SoftIceand im using g729
08:26.59SoftIcepiper69: hey ?
08:27.03SoftIcewww.govoip.co.za
08:27.51mvanbaakSoftIce: I never used g729
08:27.57mvanbaakI always use g711
08:28.48mvanbaakbut we had the same problem as you describe
08:29.01mvanbaakand it turned out to be an issue with our dsl line
08:29.11mvanbaakwe enabled QOS on the ATM level
08:29.17mvanbaakand everything is fine since then
08:29.47mvanbaakwe let the ISP enable QOS, sorry
08:30.22piper69SoftIce: i don't think they provide calls to Africa, sudan
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08:34.31piper69do i have to have all the hardware to run asterisk, i am running debian and i want to install asterisk to test and see how it looks like
08:35.25SoftIcemvanbaak: ahh, luck you :)
08:35.33SoftIceto get our ISP's to even bother with that is a no go
08:37.08SoftIcewhat sound is 'busy' ?
08:37.12SoftIceI cant find busy.gsm
08:37.20SoftIcejust to do a normall engaged sound ?
08:37.33mvanbaakbusy()
08:37.38SoftIcenot playing that gastly message
08:37.54SoftIceall cuircuts, etc :P
08:38.08SoftIcemvanbaak: ye, do you know what busy() actually plays, what gsm file?
08:38.09mvanbaakyou can try: Busy()
08:38.20mvanbaakI dont think it plays a file
08:38.31mvanbaakI think it uses indications.conf to create the tones
08:38.51mvanbaakbut I'm not 100% sure
08:38.56mvanbaakyou could check
08:39.02SoftIcethanks
08:39.14mvanbaakexten => 13,1,Busy(30)
08:39.23mvanbaakthat will play the busy tones for 30 seconds
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08:43.11SoftIcemvanbaak you ever had this
08:43.17SoftIceyou dial from you box to a phone
08:43.19SoftIceyou hangup
08:43.24SoftIcebut it is not allways hanging up the call
08:43.29SoftIceso it keeps ringing but you have put down the phon?
08:43.50JTtzafrir_laptop: just a quick tip: NO-ONE uses ISDN hardphones in Australia
08:43.58JTor for values very near to none
08:44.05JTpeople use isdn
08:44.08JTnot isdn phones
08:44.19SoftIcewho has here an engaged.gsm file ?
08:44.45JTSoftIce: what do you need all that crap for? just use generated tones
08:44.58SoftIceJT: playing with stuff
08:45.04SoftIceJT: any idea on my last query ?
08:45.20SoftIcei phone from my phone through my asterisk to my cell I hang up my phone
08:45.23SoftIcethe cell keeps ringing?
08:45.29JTlook up playtones, congestion, and busy in the wiki
08:45.36JTthey use indications.conf
08:45.40SoftIcethanks
08:46.03mvanbaakhhmm
08:46.12mvanbaakSoftIce: I have no idea. I never had that
08:49.12SoftIceye, its weir
08:49.17SoftIceit doesnt happend all the time
08:49.19SoftIcebut it happens
08:49.24SoftIcemaybe a sip bridge between each other
08:49.29SoftIceor somethig to that affect
08:49.48SoftIcethats the only reason I prefure iax, is all these issues with sip/nat
08:50.15piper69what is it called when you can someone and instade of you hear the ringing tone you hear music.?
08:50.25SoftIceMOH
08:50.28SoftIcemusic on hold
08:50.44SoftIceoh, i miss understood
08:50.49piper69no MOH is when you put someone on hold
08:50.59SoftIcewell it can be an IVR
08:51.04SoftIcethat will take the call and play music
08:51.09SoftIceuntill some line becomes avaiable
08:51.12piper69nope
08:51.18piper69thats not it
08:51.26JTit's called a musical ringing indication
08:51.29JTor similar :)
08:51.32piper69i think its PVI or something like that
08:51.39JTalso known as pain in the arse
08:51.55SoftIceoh, you talking about pre-answer
08:52.06SoftIceye, just like you can generate a ringing tone with r
08:52.16SoftIceyou can do the same to do nothing orplay music
08:52.17SoftIceor what ever
08:52.41piper69SoftIce: either you type really too fast, or you need a new keyboard
08:53.14piper69lol
08:53.29SoftIcewhat do you mean ?
08:53.48JTpossiblysomethingtodowiththelackofspaces
08:54.09SoftIcespaces, you mean many lines to structure a sentence
08:54.19JT<PROTECTED>
08:54.22JT^
08:54.24JT;)
08:54.39piper69SoftIce: you miss alot or s a r's :)
08:54.48SoftIceahh, well i've been a unix admin for over 12 years :) you get used to tab completion :P
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08:55.23piper69SoftIce: hahahaha yeah
08:55.23SoftIceand swopping keyboards also make a difference, laptop UK, desktop US, etc :P
08:55.30mvanbaakSoftIce: learn to touchtype
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08:55.43SoftIceI bang type :)
08:55.46*** part/#asterisk welemon (n=welemon@221.218.209.202)
08:55.47mvanbaakthat way you can set your keyboard layout in the os ok
08:55.58mvanbaakno matter what the keys are on your hardware
08:56.12piper69you know what the funny thing, i was working in a windows machine the other day and for some reason i hit the tab com after a command and it did complete it . i was stun
08:56.20SoftIcewell, let me quote what dragon natrual speaking once said, real men dont have to type, use a head set and dictiate what you want :P
08:56.34SoftIcepiper69 mwahah , had the same awhile back :)
08:57.28piper69SoftIce: what distro are you using
08:57.30AndyCapSoftIce: so what did it type on your screen if you said that. :P
08:57.56SoftIce:P
08:58.16SoftIcepiper69: well mainly fbsd, untill i got a large contract with 2 companys that use ubuntu :P
08:58.27SoftIceim south africa, i should use it anyway :
08:58.28SoftIce:)
08:58.52piper69where in south africa? we are brother then
08:58.58piper69*s
08:59.03SoftIcei live close to durban
08:59.10SoftIcelived all over though, cape town, jhb, :P
08:59.20SoftIcedidnt like jhb, im a surfer! :) need the ocean
08:59.42piper69SoftIce: i am originaly from sudan
08:59.50piper69but i live in USA now
09:00.28SoftIcenice man
09:00.33SoftIceid love to live in the us too :)
09:01.49piper69yep you will brother, one day you will. that was my dream
09:02.53piper69what do you do for living
09:05.12piper69i am running debian etch and in the process of downgrading to sarge ao that i can insall asterisk
09:05.57SoftIcehaha
09:06.16SoftIcewhy downgrade? why not use ubuntu at least it has support up till 2011 or something for 6.06
09:07.05*** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju)
09:12.13piper69isn't ubuntu from debian thu
09:12.31*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
09:12.47SoftIcewell its very much the same, except you cant use debian in an enterprise solution because they dont have support
09:13.23SoftIcealso, ubuntu has ubuntu dapper what is version 6.06 that has a support till for another 4 or so years
09:13.33SoftIceuntill their next long term release is issued
09:14.15SoftIcealso, if you compare packages, you will see that debian packages are slightly higher(newer) than ubuntu, as ubuntu tests their stuff thoroughly
09:16.45mvanbaakSoftIce: that last one is the other way around
09:16.52mvanbaakubuntu has newer packages
09:17.02mvanbaakunless you are running debian testing
09:17.12mvanbaakthat one gets new packages quickly
09:17.15mvanbaakso ppl can test it
09:17.22mvanbaakbefore it becomes debian stable
09:17.42mvanbaakdebian stable is very picky about new packages
09:17.46piper69mvanbaak: it's called etch
09:18.11piper69will astersik work on ubuntu
09:18.13mvanbaakthat's testing
09:18.18mvanbaakpiper69: yeah it will
09:19.27SoftIcemvanbaak ye
09:19.34SoftIcewell im talking about long term relase ubuntu
09:19.38SoftIcetheir packages are not newer
09:19.41SoftIcebut edgy, etc are
09:20.39SoftIcehmm, i have this issue, i phone, right it goes through the carrier rings on the caller phone
09:20.41SoftIcethey answer
09:20.47SoftIcebut it keeps ringing on my side?
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09:25.49piper69SoftIce: i was pormoted today to be an 5ESS DCS Switch Engineer
09:25.58SoftIcenice
09:26.02SoftIceyou working saturdays?
09:26.49piper69SoftIce: nope , no sat or sunday
09:28.12piper69i haven't work sat and sunday for almost 2 yrs now , thats when i was an IT manager for the wireless company i work for, but now since i am going to be a switch eng i think i will work sat or sun as it required
09:30.56SoftIce:)
09:30.57SoftIcence bro
09:31.11piper69mvanbaak: what version of ubuntu do you think ,
09:31.43SoftIcepiper69 6.06
09:31.46mvanbaakpiper69: I dont use ubuntu
09:31.47SoftIcedapper
09:31.50mvanbaakI use debian
09:32.14SoftIcedebian is great you just cant use it in an enterprise env
09:32.26piper69mvanbaak: so what do you think , i am currently runnin etch
09:32.40mvanbaaketch is fine
09:32.44mvanbaakI'm running it too
09:32.53mvanbaakSoftIce: I'm running debian in enteprise env
09:33.07SoftIcemvanbaak: well you not understanding what im saying, you dont have support
09:33.35piper69is there is a pakage for Asterisk
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09:33.47SoftIcepiper69 apt-get install asterisk zaptel
09:34.14DarKnesS_WolFSoftIce: what version of astersiks debian has ?
09:34.24piper69but why when i did apt-cache search asterisk nothing came
09:34.56SoftIcepiper69 you need to add some repos
09:35.06SoftIceto the sources.list file
09:35.06piper69DarKnesS_WolF: i will assume a lower version of the current onw
09:35.19SoftIcepiper69 wrong, as the lower version has sip vulns
09:35.22SoftIceso they are up to date
09:35.55SoftIcegrrrrr, please can somebody sugest something to me, I phone out, pass call to carrier via sip they pass the call to my cell phone I answer but on my side it keeps ringing
09:36.53DarKnesS_WolFSoftIce: hmm do u have -r option in ur dial line ?
09:37.30SoftIceDarKnesS_WolF: yes
09:37.31piper69all what i need is VOIP phone and i can run asterisk?
09:37.40DarKnesS_WolFSoftIce: remove it and try
09:37.48DarKnesS_WolFthis r i think to foce asterisk to give ring tone.
09:38.09piper69and sign up with voip carrier
09:38.20SoftIceDarKnesS_WolF: yes but Ihave a delay with the carrier
09:38.23SoftIcethats why I have it in there
09:38.30SoftIceso people can least here its doing something
09:38.36DarKnesS_WolFSoftIce: try without it 1st
09:38.51SoftIcegreat
09:40.11piper69so can someone answear my question
09:41.27piper69please
09:42.01SoftIcewhat?
09:42.39piper69for hardware, all what i need is an VOIP phone
09:43.01piper69or ATA
09:43.24AndyCappiper69: you don't really need hardware either, but it depends on what you want to do..
09:44.16piper69all what i want is to have a phone that i can use to call my parents in africa
09:44.28piper69and maybe use it here localy
09:46.14SoftIceDarKnesS_WolF it still happens :P
09:46.42DarKnesS_WolFSoftIce: hmm no idea ;-)
09:48.04piper69<PROTECTED>
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09:49.53EmleyMoorHow do I specify the timezone that covers Shanghai in the SayUnixTime command?
09:52.09piper69Setting up asterisk-sounds-main (1.2.13~dfsg-2) ...
09:52.09piper69Setting up fxload (0.0.20020411-1) ...
09:52.09piper69Setting up zaptel (1.2.11.dfsg-1) ...
09:52.09piper69Zaptel telephony kernel driver: FATAL: Module ztdummy not found.
09:52.09piper69Notice: Configuration file is /etc/zaptel.conf
09:52.11piper69line 0: Unable to open master device '/dev/zap/ctl'
09:52.14piper691 error(s) detected
09:52.16piper69/sbin/ztcfg failed. Check /etc/zaptel.confzaptel.
09:52.19piper69Setting up asterisk-classic (1.2.13~dfsg-2) ...
09:52.21piper69Setting up asterisk (1.2.13~dfsg-2) ...
09:52.24piper69Asterisk not yet configured. Edit /etc/default/asterisk first.
09:52.32piper69!Topic
09:52.42piper69!topic
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10:08.48dlynes_laptopEmleyMoor: export TZ=UTC+8 doesn't work?
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10:36.11tzafrir_laptoppiper69, what version of Debian is it?
10:36.50tzafrir_laptopapt-get install zaptel-source build-essenials
10:37.00tzafrir_laptopm-a a-i zaptel
10:37.14tzafrir_laptop/etc/init.d zaptel start
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11:06.00tzafrir_laptopEmleyMoor, Asia/Shanghai ?
11:06.22tzafrir_laptopEmleyMoor, generally look under /usr/share/zoneinfo
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11:11.10stonyhi
11:11.20stonycan i add more than one device to a trunk ?
11:11.35stonyi need 4 misdn channels in a trunk so that asterisk can get the next free one and use it
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11:14.26tzafrir_laptopstony, in zap (bri/pri) that's pretty simple: a group. I imagine a similar concept exists with misdn
11:14.54AndyMcGee001Hi
11:16.29stonytzafrir_laptop: i read it in this moment
11:16.35stonythx anyway
11:18.26AndyMcGee001Can i do more than one "hint" before i dial more than one sip-phone and will it pickup this dial if someone picksup`s one the hinted extensions?
11:25.45EmleyMoorAndyMcGee001: What are you trying to achieve
11:25.47EmleyMoor?
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11:27.44AndyMcGee001I want to dispactch a incomming call to a group of people. When i do this all the line buttons of the people not in this group start to flash...
11:28.26AndyMcGee001When someone not in the group now hits one of the flashing keys a pickup is issued to the numbered extension programmed for that key...
11:28.56EmleyMoorHmmm... I would think it possible - but I don't know how
11:29.04AndyMcGee001but the pickup goes to nirvana because the dial is not assotioated with the extension, only with the SIP username(s)
11:31.58EmleyMoorSounds like you need a dialable thing to flash the line buttons...
11:35.56AndyMcGee001Yes, i thought about that to. But i think i need to assosiate all the extensions (101, 105, 103 etc.) with the call because thats whats programmed to the buttons inside the phones.
11:36.42AndyMcGee001and "hint" seems applicable. I just dont know if i can attach more than one hint to a call.
11:37.31EmleyMoorAm I right in thinking PoE (the standard kind) will work over 2-pair wiring, btw?
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11:40.44l1nuxhi
11:40.54AndyMcGee001Yes i think so, uses some kind of phantom powering then (dc overlay)
11:41.12l1nuxasterisk 1.4.2 not support sqlite3 ?
11:41.36tzafrir_laptopl1nux, there are a number of patches for sqlite support
11:41.58tzafrir_laptopThere is also something in asterisk_addons , though it is not really maintained
11:42.44l1nuxtzafrir_laptop, please where get patch to fix it ?
11:43.04l1nuxi try patch from http://bugs.digium.com/view.php?id=6754
11:43.15tzafrir_laptopthat i one of them
11:43.28tzafrir_laptopIt should reference the other ones, I believe
11:43.41l1nuxi get "configure: *** The SQLite installation on this system appears to be broken"
11:43.44l1nux:/
11:44.41l1nuxusing Debian testing (sqlite3)
11:46.14tzafrir_laptopat least one of them is ued by the developers of destar, who mainly use debian Etch
11:47.50AndyMcGee001Bye
11:48.30tzafrir_laptopanyway, http://bugs.digium.com/view.php?id=7149 is the one that was merged into trunk
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11:49.52tzafrir_laptopactually a working sqlite3 logging was added to the Debian asterisk package at some stage. I believe t is included in Etch. checking...
11:50.58tzafrir_laptopyup. It's in there. http://packages.debian.org/changelogs/pool/main/a/asterisk/asterisk_1.2.13~dfsg-2/changelog . apt-get install asterisk
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11:54.11l1nuxtzafrir_laptop i am using astersik 1.4.2 from asterisk.org
11:56.23l1nuxhttp://lists.digium.com/pipermail/asterisk-dev/2006-December/025266.html :/
11:56.26tzafrir_laptopright. the asterisk 1.4.2 in experimental does not have that patch applied
11:59.37tzafrir_laptopI do see cdr_sqlite.so which is sqlite2
12:01.30l1nuxyes
12:17.31*** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca)
12:17.43MrTelephonehey is there a program to freemem for linux
12:17.58MrTelephonemy ram usage slowly creeps up
12:18.49*** join/#asterisk theBong (i=jvckso@84.94.98.200.cable.012.net.il)
12:20.36theBongnewb q: agi-test.agi does not play back the command SEND TEXT.. Any suggestions/Alternatives? The log reads 200 as the return code
12:23.00theBonganyone?
12:24.28MrTelephoneare those external scripts your running/
12:24.30MrTelephone?
12:25.29theBongi copied the contents of agi-test.agi to swift.agi ( I use RAGI) and the script runs.  SAY NUMBER works but not SAY TEXT
12:26.20theBongalso , where is STDERR by default (Linux)? Thanks.
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12:28.00florztheBong: Usually on the left.
12:30.14theBongthe log read Mar 31 04:27:34 VERBOSE[8666] logger.c: AGI Rx << SEND TEXT "hello world"
12:30.14theBongMar 31 04:27:34 VERBOSE[8666] logger.c: AGI Tx >> 200 result=0
12:30.18MrTelephonemy dhcp server is whacked out
12:30.32theBongand nothing gets played
12:31.57florztheBong: Why do you think anything should get played?
12:32.23theBongSEND TEXT should play the text to the user.  SAY NUMBER works fine
12:32.42florztheBong: Why do you think anything should get played?
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12:33.33theBongsee my answer above
12:33.56florztheBong: Well, then: Why do you think SEND TEXT should play the text to the user?
12:34.16florzhow ever one would "play a text"
12:35.15theBonghow else would you recommend to say text in agi? I need a simple AGI that receives 1 param and reads it to the user
12:35.19theBongthanks again
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12:35.57florztheBong: Well, I guess you would need a speech synthesis program first.
12:36.26theBongi have flite
12:36.51florztheBong: Well, and then use it like you would use it from the dialplan
12:39.24theBongthat's where I am struggling.  RAGI (Ruby AGI) has a say method which in turn calls swift.agi.  I wanted to create this swift.agi to simply read the text sent by RAGI
12:39.24*** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net)
12:39.37theBongperhaps I should change the ragi code...
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13:04.02MrTelephoneis there a pri failover device that switches to computer 2 if something fails?
13:04.33*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
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13:08.21l1nuxtzafrir_laptop, fixed in trunk (svn) :D
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13:21.53_DAWAnyone interested in testing some SIP DID's?  I have 9 to give away for the next 6 months in exchange for regular feedback on the voice quality.  I can do most of the US.
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13:36.19anthonylim up for freevoip
13:36.24anthonyler pstn stuffs
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13:36.37anthonylwait,no nevermind
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13:36.45_DAWok
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13:57.50ligobanAddons 1.4.0 installation question
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14:07.10f1r3w0rmhmm
14:07.10f1r3w0rmanyone tried asterisk for windows ?
14:07.26sbingnerblasphemy
14:07.38JunK-Yf1r3w0rm: where did u see * for windows?
14:07.41Gido-Ef1r3w0rm are you seriuos?
14:07.48stonyanyone ever tried to get a siemens hipath running together with asterisk ?
14:07.50f1r3w0rmhttp://www.asteriskwin32.com/
14:08.08f1r3w0rmi guess im pretty damn serious
14:09.29Gido-Ef1r3w0rm: heueu dont run asterisk on a windows machine.
14:10.02f1r3w0rmdo what
14:10.08drfreezeMorning. Can someone point me to some docs on updating the firmware for Polycom 501 phones?
14:10.34JunK-Ydrfreeze: just put ur new in the directory
14:10.39zeedof1r3w0rm: I'd say you'd be better of just installing it and trying it, most people here will think it's silly - personally I'd say try it and if it's good for you then great
14:10.46[TK]D-FenderAsteriskWin32 0.56 build from Asterisk 1.0.10
14:10.49[TK]D-FenderANCIENT
14:11.01[TK]D-FenderAsteriskWin32 0.60 build from Asterisk 1.2.14
14:11.05[TK]D-FenderSlightly better...
14:11.05zeedof1r3w0rm: one thing I would point out is that they dont seem to keep up with asterisk updates, so you could be left open to security issues
14:11.19JunK-Y1.2.14 is so old too.
14:11.24f1r3w0rmthis is for a non inet connect system
14:11.25[TK]D-FenderAll * for Win is is a virtual linux env running *.
14:11.31*** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca)
14:11.31f1r3w0rmso security aint much issue
14:11.48[TK]D-FenderYou have to config it EVERY BIT the same
14:11.49zeedof1r3w0rm: security was an example, all sorts of bugs will be missed
14:11.58[TK]D-Fenderand invite trouble for the virtual aspect of it.
14:12.01drfreezeJunK-Y: I'm talking about the firmware, not the config. Are they updated in the same way?
14:12.15[TK]D-Fenderzeedo: No... they won't be missing any bugs... they're all still in there ;)
14:12.25JunK-Ydrfreeze: yes
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14:12.31drfreezeAny trixbox experts here?
14:12.35drfreezeJunK-Y: thx
14:12.37[TK]D-Fender~trixbox
14:12.44jbot[trixbox] junk - avoid.  It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY!
14:12.44JunK-Ydrfreeze: see #trixbox
14:12.57blitzragemorning all!
14:13.01zoayo yo
14:13.03JunK-Yhiya blitzrage !
14:13.14drfreezeI have a client who is using trixbox, and we could not get a phone to authorize.
14:13.27blitzrageummmm... that description of trixbox is a bit harsh
14:13.29drfreezeThe passwords on the phone and the sip_additional.conf matched however
14:13.31zeedodrfreeze: try the #trixbox channel
14:13.35JunK-Ylike we just said, goto #trixbox!
14:13.37zeedoerr sorry #freepbx
14:14.08drfreezeare there other user/passwords in asterisk that I should know about to authenticate with *
14:14.15blitzrageanyone know if whisper paging can whisper page all currently bridged channels on a system?
14:14.31blitzrageI want to use it to announce system outages to current callers
14:14.45blitzragei.e. "the system will be going down in 5 minutes"
14:15.16[TK]D-Fenderblitzrage: I suspect if the source end of the whisper is the "Page" app...
14:15.54drfreezeok guys, forget I said trixbox. :)
14:16.03Iamnach0lol
14:16.17blitzrageactually, more than likely the ChanSpy() app
14:16.34JunK-Ya process for that will be excellent!
14:16.46[TK]D-Fenderblitzrage: Use Page to do a Local channel where you call the whisperpage instead of Dial.
14:17.00blitzragethis would be for active channels -- I don't think that would work
14:17.07drfreezeOther than username and secret inside sip.conf, is there another user/pass combination used by the phone to authenticate with *
14:17.09f1r3w0rmzeedo
14:17.12f1r3w0rmu there
14:17.13blitzragei.e. already bridged
14:17.17zeedof1r3w0rm: yeh
14:17.17blitzragenot inactive channels
14:17.20f1r3w0rmpm plz
14:17.24zeedosure
14:17.33[TK]D-Fenderblitzrage: You might want to use an AGI to poll for active channels
14:17.35*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
14:17.44[TK]D-Fenderblitzrage: And then dumpt them.
14:17.59JunK-Y[TK]D-Fender: theres only one problem to that
14:18.00blitzrageI think this could be done from a separate dialplan app
14:18.06blitzrageand would be the most efficient
14:18.10JunK-Yya loop to stream to all channels right?
14:18.33blitzragean Outage() app, or an 'o' flag to the ChanSpy() application ('o' for outage)
14:18.34JunK-Yso ya must wrote ur agi in C to start x threads.
14:18.38[TK]D-FenderJunK-Y: I presume he may only want to whisper to the INESIDE channels, which would require some intelligence to the selection method...
14:19.14[TK]D-FenderJunK-Y: As in "don't tell customers why we have to hang up on them"
14:19.39JunK-Yto the inside channels???
14:19.59blitzrageI'd just be happy with keeping it more simple, and just broadcast to all active bridges
14:20.26blitzrageonce that worked, then more complex methods of determining who is "inside" and who is "outside" could occur
14:20.52blitzrageyou'd basically have to parse the sip.conf file, and say, "ok, if any of these peers are active, whisper page them"
14:21.17JunK-Ywhat do ya mean by inside and outside?
14:21.18blitzrageJunK-Y: yes, I'd say all channels, not just bridged would be the ideal
14:21.29blitzrageJunK-Y: he means local extensions vs. outside callers
14:21.50JunK-Ythat could be a later option to the outage app, right?
14:21.54blitzrageyes
14:22.00blitzragefor now you'd just say "all channels"
14:22.09blitzragejust to prove it can work
14:22.14blitzragethen you can fine tune its settings
14:22.32JunK-Ybut the stream have to be on all the channels, at the same time, right?
14:22.37blitzragethis might even be a CLI option and not an application....
14:22.41blitzrageactually, both would be ideal
14:22.44blitzrageI can think of uses for box
14:22.46blitzrageboth*
14:23.00blitzrageJunK-Y: yes at the same time
14:23.03blitzrage(or within reason)
14:23.41JunK-YOutage(filetostream) right?
14:23.50blitzragestop gracefully with warning at 5,1 mins
14:23.55blitzrageJunK-Y: yes
14:24.20blitzragestop gracefully notice 5 warn 1
14:24.36blitzragethat'd send a notice at the 5 minute mark, then warn with 1 minute left
14:24.48blitzrageand of course not accept any new calls
14:25.14JunK-Yit has to avoid Zap/pseudo, any special channels to avoid?
14:25.37blitzrageyou could even get fancy and have a Dial() option or Queue() option that could listen for DTMF so agents could extend / request a longer time out if they are in the middle of a sale
14:26.07blitzrageJunK-Y: ideally it'd work for Local and application channels (i.e. MeetMe, Queue, etc...)
14:26.27blitzragebut I'm no channel expert
14:26.55JunK-Yi will work on that this evening and see what can I do exactly.
14:27.01blitzragesounds good!
14:27.09JTMrTelephone: yes, you can get pri failover devices
14:27.14JunK-Ythat will not be kpflemming's code, but that will work :)
14:27.17blitzrageI've suggested a couple of good ones to russellb lately
14:27.33zoais chanspy stable yet ?
14:27.44JunK-Yi had in mind, for conferences servers, a way, to do that
14:27.55blitzragezoa: I'm using it in testing with no crashes thus far
14:28.00JunK-Ybut only to pseudo/zap, so all meetme users could hear that.
14:28.11zoawe have some whisper thing we use here
14:28.13blitzragezoa: although I think file is building a new version in his audiohooks branch
14:28.17zoadunno if its stable though
14:28.35zoawe dont really use it in production
14:29.22blitzrageanyone know if you can "redirect" an IAX2 channel via the dialplan like the SIP channel via the Transfer() application?
14:29.33JunK-Yzoa: it just grab all actives channels and stream a specific message?
14:29.43zoawe dont have it for all channels :)
14:29.48zoabut that shouldnt be too hard to make
14:29.52blitzragebecause if that's possible, then I can build an IAX2 registration server, and redirect calls to my Asterisk cluster....
14:29.52zoai guess
14:30.02JunK-Yzoa: do u think its a good idea?
14:30.03zoablitzrage
14:30.03zoa: i think thats possible
14:30.08zoayes it would rock in some cases
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14:30.20zoablitzrage: i was thinking to do something else
14:30.24blitzragezoa: yah? As long as its the same channel type right? (i.e. I can't redirect SIP to IAX2)
14:30.31blitzragezoa: what's your idea sir?
14:30.32zoato actually build a registration server outside of asterisk
14:30.34zoajust doing that
14:30.36zoalike SER
14:30.38blitzrageyes!
14:30.42zoaas it would be faster
14:30.53blitzrageit needs to support IAX2 and DUNDi
14:31.01zoaand it would shield the iax2 port on the asterisk
14:31.04blitzrageyes
14:31.06blitzragegood idea!
14:31.17zoasimilar to what we do now for SER and RTP on our setup
14:31.19JunK-Yblitzrage: i will change it a bit
14:31.28zoaasterisk is not visible to the outside world
14:31.31blitzrageJunK-Y: np, let me know when you have something and I will test
14:31.41JunK-YOutage(), will go at a particular cep, so we could use anything ya want.
14:31.45blitzragezoa: same with us -- we distribute calls to Asterisk via SER
14:31.51blitzragecep?
14:32.00JunK-Ycontext extension priority
14:32.03blitzrageahhh
14:32.07JunK-Yso you could stream 2 streams
14:32.09blitzrageyes, that'd be handy
14:32.15zoayeah but i dont have audio going straight either
14:32.18JunK-Yor dial, or email, or whatever ya want.
14:32.21blitzragezoa: ahhhhh I see
14:32.26blitzragevia the mediaproxy?
14:32.35zoavia our own mediaproxy
14:32.35blitzrageor whatever SER calls it
14:32.39blitzragegotcha
14:32.51JunK-Ycause if ur on iax2, u could dial ur remote server, and the transfer will do the rest
14:32.56zoawe can do 3000 bi directional calls through one proxy
14:32.59JunK-Yso ur server is no longer in the path.
14:33.03zoaand those proxys are clustered
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14:33.13blitzrageI have a business edition license here I should register so I can get priority to bug reports :)
14:33.20zoalol
14:33.38JunK-Yblitzrage: i need one too!
14:33.38JunK-Y:)
14:33.57blitzrageI had Digium send me one :)
14:34.25JunK-Ythe latest ABE is based on 1.4 or still 1.2?
14:34.32blitzrage1.2 I think
14:34.36JunK-Ykk
14:34.50JunK-Yive to jet, breakfast with brother, we'll be in touch soon.
14:35.15blitzragesounds good buddy
14:35.40JunK-Ydont say that to Corydon!
14:35.43JunK-Yhehehe
14:36.08blitzrage:)
14:36.13blitzragehey, where is that .version file?
14:37.52blitzrageoh yes, make update :)
14:37.54blitzragelets see if it works
14:38.01blitzragesometimes its the simple things that you forget
14:39.04JunK-Yroot@troy:/usr/src/asterisk-1.4# build_tools/make_version .
14:39.05JunK-YSVN-branch-1.4-r59289M
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14:46.16f1r3w0rmanyone know where i can get a cheap telephony card
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14:49.33mmartinnPlease everyone take a moment to rejoice that our PRI problem may be solved :)
14:49.51blitzragehallelujah!!!
14:50.00blitzrageCAN I GET AN "AMEN"!?
14:50.16blitzrage*may* be solved?!
14:50.17Strom_MACK
14:50.25blitzrageStrom_M: heh :)
14:50.35mmartinnWell, we're pretty sure. No problems for over 24 hours
14:50.47Strom_Mi'm not terribly good at coming up with snappy ISDN jokes
14:51.02mmartinnStrom_M: LOL... I didn't get it first, but now I do
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14:51.36mmartinnI'm now looking forward to figuring out why an identical box isn't working with the PRI
14:52.32mmartinnMy bets are (a) the interface card (b) the network/cable/switch or (c) bad hardware like the mobo/memory/cpu
14:53.24stonycan i do something like: msns=1234* in misdn.conf ?
14:55.08f1r3w0rmdamn Telephony cards arent cheap
14:55.38blitzragetelephony cards for Asterisk are relatively cheap compared to standard commercial telephony hardware
14:55.58mmartinnYou can always ebay them too if you want cheaper
14:58.31f1r3w0rmi jus wanted to make a home setup to page my staff
14:58.54stonyif you use sip internaly and a isdn-card for the extern stuff it's really cheap
14:59.26f1r3w0rmmy whole setup is internal
15:00.12f1r3w0rmim using my house to demo the setup if i like the way everything comes together im gonna do it @ my buddies bussiness
15:00.20stonygo, use voip ...
15:01.51riddleboxcan you record conference calls in asterisk?
15:02.01Qwellriddlebox: sure, show application meetme
15:02.10riddleboxok cool thanks
15:05.42*** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za)
15:06.30SoftIcehmf, anyone know if/when bristiff will release a bristuff using 1.4 asterisk ?
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15:29.30tzafrir_laptopSoftIce, no idea. There's a letter on their site where they promise to do so
15:35.00SoftIcetzafrir_laptop: ahh, thank you
15:51.24JTtzafrir_laptop: catch my note about isdn phone here? :)
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15:51.35tzafrir_laptopyes
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15:52.17JTtzafrir_laptop: what telco are you having trouble with?
15:52.19tzafrir_laptopI was going to answer: well: maybe they were hoping they will have phones when they planned the ISDN BRI service
15:52.47tzafrir_laptopJT, I was helping someone, so I'm not really sure which telco. Is there more than one?
15:53.15JTi don't know if any other than Telstra offer BRI
15:53.20JTothers offer POTS and PRI
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16:00.00MaghteridonHeelloo
16:00.05Maghteridonph [TK]D-Fender my hero
16:00.26Maghteridon*oh =) Can I bother you with a couple of questions? These are fast tbh
16:00.29Maghteridonmmmm? =)
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16:01.25*** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it)
16:01.42MaghteridonI have some problems with music on hold... I'm not able to get it work fine, even tough I followed examples...
16:01.56Maghteridonall I can hear is a "heavy noise"
16:02.04*** join/#asterisk Fieldy (i=DsxvRmR2@gentoo/contributor/Fieldy)
16:02.41*** join/#asterisk uwe (n=uwe@dogbert.palnet.com)
16:04.03Maghteridonsupposing I'd like to play "example.mp3", is musiconhold.conf all I need to modify, if the mp3 file is in /var/lib/asterisk/mohmp3/example.mp3 ?
16:07.56*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
16:09.29*** join/#asterisk aaronr (n=arussell@87.127.234.100)
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16:10.59riddleboxI have setup exten =>698,1,Answer(), exten =>698,2,MeetMe(699,DprM,1234)
16:11.52riddleboxand also setup in meetme.conf conf => 699,1234,123456
16:12.06riddleboxbut when I call 698 it says it is an invalid conference
16:18.48*** join/#asterisk eliXier (n=GTI16V@gti.twice-irc.de)
16:19.59_DAWAnyone interested in some free SIP DID's to help test?  I have 9 left I can give away for the next 6 months in exchange for regular feedback on the voice quality, message me off channel if your interested.
16:24.24*** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net)
16:30.14[TK]D-FenderMaghteridon: Yes?
16:30.43MaghteridonHow do you call the function that let you pass a call from one number to another (internal numbers) ?
16:30.56MaghteridonI don't know what to look for because I don't know the translation
16:31.00Maghteridonin english ^^
16:31.12[TK]D-FenderMaghteridon: You mean send a call on 1 phone to another?
16:31.37[TK]D-FenderMaghteridon: That would be "Transfer".  There are 2 kinds of transfers, ATTENDED, and BLIND.
16:31.46MaghteridonI mean X calls Y, Y answers, and then he passes the call to Z
16:32.36[TK]D-FenderMaghteridon: Attended transfers means YOU cann person #2.  They See YOUR CallerID. they pick up.  You ask if they want to talk to this other person.  They say "yes", tyou then COMPLETE the transfer
16:33.02*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
16:33.09[TK]D-FenderMaghteridon: Blind transfer is where you jsut tell it where you want to transfer them to and the call just GOES AWAY on your phone and starts ringing at the destination.
16:33.56MaghteridonYes, then I need the first one... is it already implemented or should I configure anything?
16:34.16[TK]D-FenderMaghteridon: It depends on what kind of phones you are using.  Most SIP phones do this INTERNALLY.
16:34.21MaghteridonOn normal phone here in italy it works like pressing the "R" key and the internal number where to transfer to
16:34.27[TK]D-FenderMaghteridon: Which takes NO configuration to do.
16:34.40[TK]D-FenderMaghteridon: So what kind of phones are you using?
16:35.10MaghteridonX-ten (not really me actually... a friend) , and a phone adapter connected to a normal cordless phone
16:35.13*** join/#asterisk gustavo (n=G@87-196-142-229.net.novis.pt)
16:35.26Maghteridona lynksys pap2 port adapter
16:35.40ManxPowerI thought X-Lite did not have an attended transfer feature.  I thought that feature was only availble in the commervial "Pro" version
16:36.11MaghteridonIt might be actually... I'm not using it now
16:36.14ManxPoweron the Linksys, "R" aka "RECALL" aka "FLASH" key should initate a transfer
16:36.31MaghteridonSo R+number?
16:36.34Maghteridonor number+R?
16:36.35[TK]D-Fenderx-ten (if you mean X-Lite) does not have a transfer feature.  You will need to learn how to use FEATURES.CONF and let * do the work for this one.
16:36.36ManxPowerthe Linksys will then handle all the magic to transfer a call.
16:36.51[TK]D-FenderMaghteridon: For your ATA, it should have its own way of doing this.  Check its manual
16:37.01Maghteridonallright then
16:37.05Maghteridonand last question...
16:37.17Maghteridonthe music on hold... all I hear is n heavy noise :|
16:37.27[TK]D-FenderMaghteridon: The PAP2 does have its own way, just read the manual for it
16:37.43ManxPowerMaghteridon: the Music On Hold files may have a sample rate asterisk can't handle.
16:37.55Maghteridonactuallt look... :
16:38.11ManxPower"Asterisk: The Perl of PBXs"  i.e. there are always several ways to do something.
16:38.15[TK]D-FenderMaghteridon: And what MODE are they using in "musiconhold.conf"?
16:38.16*** join/#asterisk mihinomenest (i=QWD2@cerebus.clandestineresearch.com)
16:38.22Maghteridon[default]
16:38.22Maghteridonmode=files
16:38.22Maghteridondirectory=/var/lib/asterisk/mohmp3/
16:38.34Maghteridonit says that cannot find the directory
16:38.42[TK]D-FenderManxPower: http://www.sofaswitch.org/d/
16:39.00[TK]D-FenderMaghteridon: Does * have UNIX rights to that folder?
16:39.06MaghteridonOfcourse
16:39.43[TK]D-FenderMaghteridon: Also remov the trailing slash
16:39.53ManxPowerMaghteridon: does the user asterisk is running as have rights to that directory and does the directoy actually conain any files?
16:40.00ManxPowerI can't seem to type today
16:40.00riddlebox[TK]D-Fender, can you help me with my meetme conference problem?
16:40.12[TK]D-Fenderriddlebox: Namely?
16:40.23Maghteridonok my bad, I didn't copy fine: here is the actualt string
16:40.26Maghteridon[default]
16:40.26Maghteridonmode=files
16:40.26Maghteridondirectory=/var/lib/asterisk/mohmp3/muse
16:40.34gustavocan someone recomend one or two reliable voip providers in europe?
16:40.39Maghteridonand the answer is: yes, good permissions and real file existing
16:40.48gustavothere are so many, it's hard to test them all
16:41.04Maghteridongustavo, where are you from btw?
16:41.14[TK]D-FenderMaghteridon: If it finds your files and they sound like crap, check the encording rate, and make sure they aren't VBR.
16:41.14gustavoMaghteridon: Portugal
16:41.19drfreezeWhile trying to install asterisk-1.4.2 I get the following: configure: error: *** termcap support not found
16:41.24Maghteridonah portugal then nothing, tought ou were italian
16:41.39Maghteridon... VBR...? =)
16:41.41drfreezeOne source says that libncurses5 should fix that
16:41.49gustavoit's not clear I should go with a local provider
16:41.52riddlebox[TK]D-Fender, let me pastebin it
16:42.10drfreeze/usr/lib/libncurses.so.5  /usr/lib/libncurses.so.5.4  /usr/lib/libncursesw.so.5  /usr/lib/libncursesw.so.5.4
16:42.27Maghteridonbtw ManxPower is that entry in musiconhold.conf  right?
16:43.37riddlebox[TK]D-Fender, http://pastebin.ca/418400
16:45.27MaghteridonManxPower, it says actually: Unable to open file '/var/lib/asterisk/mohmp3/muse/massive'
16:45.32Maghteridonno such file or directory... but it exists...
16:48.55*** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com)
16:52.10stonyhmm how can i print variables with NoOp() ?
16:52.18*** join/#asterisk J4k3- (i=J4k3@dhcp-12-197-128-42.intrastar.net)
16:52.30stonyNoOp(${CALLERID(num)} only shoes "CALLERID(num)" in the cli
16:52.44stonys/shoes/shows
16:52.59blitzragemissing a closing brace
16:53.06blitzrageand you should use Verbose()
16:53.12*** join/#asterisk boojit (n=boojit@gw.carter.to)
16:53.23blitzrageVerbose(1|${CALLERID(num)}) is better
16:53.27stonyblitzrage: yeah typo here...
16:53.37blitzrageyah... like I said :)
16:53.56stonyblitzrage: in the config it's written correctly and still not working
16:55.05blitzragesomething else is wrong... because that works
16:55.08boojithi folks
16:55.21stonyblitzrage: the verbose() function works fine ... using that...
16:55.34blitzrageNoOp() is for no operation, not really for debugging
16:55.39*** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com)
16:56.17boojitI'm doing some discovery for a client and I'm wondering if anyone here has experience doing systems for voter response.
16:56.49boojitnot using SMS, but rather just regular phone access.
16:57.09*** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com)
16:57.12riddleboxis this the correct way to do meetme http://pastebin.ca/418400 ?
16:57.16drfreezeAnyone know how to solve this problem? configure: error: *** termcap support not found
16:57.46russellbdrfreeze: install libncurses-dev
16:58.02chrisknightyum install libtermcap libtermcap-devel newt newt-devel ncurses ncurses-devel
16:58.53boojitIf anyone has any experience with such systems in a high-availability large-scale implementation, please ping me.
16:59.03drfreezechrisknight: ahh, it wants the devel packages. Trying now
16:59.15chrisknightWhen all the "default" responses dont work, how can I factory reset a cisco 7960 phone?
16:59.24drfreezechrisknight: thx. that worked
16:59.32chrisknightno prob
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17:00.54chrisknight**# does not unlock the network config...  Holding down # during reboot wont work either.  Someone has set a password on the network config as well as telnet...
17:05.48*** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir)
17:09.30ManxPowerchrisknight: Cisco has documents on their web site about how to unlock their phones
17:11.31chrisknightI know, I believe I have tried all of them.  This  thing must have some "special" load...
17:11.52drfreezeIs zaptel required for an VoIP only asterisk install
17:12.27ManxPowerdrfreeze: no.  neither is a sound card
17:14.24russellbneither are glow in the dark IDE cables
17:14.49*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)
17:18.04coppicebut glow in the dark SATA cables are this year's must have
17:21.20Qwellcrap, I better get some then
17:21.25*** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net)
17:21.25*** mode/#asterisk [+o anthm] by ChanServ
17:22.06uweum, what value does astdb use for enabling/diabling dnd ? is it YES or ON and Off or NO or by deleting the record ? i mean which does asterisk use to decide if calls should be sent to that user or not ?
17:22.58blitzrageuwe: it depends how you want to implement it in your dialplan
17:23.11Qwell(or chan_skinny)
17:23.23blitzrageGotoIf($[${DB(user/dnd)} = 1]?dnd,1)
17:23.23Qwellactually, maybe not
17:23.26blitzrageas an example
17:23.27*** part/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net)
17:24.08blitzrageGotoIf($[${DB(user/dnd)} = OFF]?dnd,1) is just has valid
17:24.16blitzrages/has/as/
17:25.11uwehmm
17:25.47uwethank you blitzrage
17:26.43Qwellblitzrage: why not like this?  GotoIf($[${DB(${ODBC_GETUSER(${EXTEN})}/dnd)} = ${ODBC_GETOFFVALUE()}]?dnd,1)
17:26.46Qwell:p
17:27.43Corydon76-homeHeh
17:27.57Qwellrussellb: that's nothing
17:28.09QwellODBC_GETUSER is a 40 line query that uses 20 tables
17:28.11*** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)
17:28.31Qwellooo
17:28.34Qwellwe need 'in'
17:28.52QwellGotoIf($[${somevar} in ${MYHASH}])
17:29.17QwellGotoIf($[${DB(${ODBC_GETUSER(${EXTEN})}/dnd)} in ${ODBC_GETOFFVALUEs()}]?dnd,1)
17:29.40Qwellwhere's murf?  heh
17:30.50blitzrageQwell: because it seems kinda dumb to use func_odbc to lookup a value in AstDB? :)
17:30.58QwellYou must be new here.
17:31.28blitzrage:)
17:31.28Qwellblitzrage: it's funny - laugh :)
17:32.05QwellGotoIf($[${ODBC_IsDND(${EXTEN})}]?dnd,1)
17:32.06Qwell:)
17:32.22blitzrage:)
17:32.36Corydon76-homeHey, that's lowercase...
17:32.42blitzrageFIRED!
17:32.46QwellCorydon76-home: yeah...
17:32.58Corydon76-homeBad convention!
17:33.10*** part/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com)
17:33.18Qwellsure, and I'm really strict when doing SQL related stuff usually..  just for readabilities sake in this case
17:33.37QwellI can't stand it when people use lowercase "select" in queries
17:33.48russellbselect *;
17:33.52Corydon76-homeheh
17:34.10Qwellrussellb: no FROM clause :D
17:34.15russellbpwned!
17:34.16Corydon76-homeActually, I use lowercase keywords when morons capitalize all their table names and column names
17:34.32QwellCorydon76-home: I use uppercase table names ALL the time :p
17:34.43fileSELECT russellb FROM world;
17:34.50russellbi got really irritated the other day when I realized I couldn't do the equivalent of "SELECT * FROM whatever;" through the realtime API
17:35.20russellbI modified res_config_odbc to let you do it, but haven't finished changing all of the others ...
17:35.30*** join/#asterisk mrbnet (n=sureal@corpmail1.mrbnetworks.com)
17:35.50russellbif it worked, then all of the "sip show peers", "voicemail show users", etc. CLI commands could work with realtime
17:36.08russellb</small rant>
17:36.42blitzragerussellb: :-O
17:37.04Qwellrussellb: You can't just set no WHERE clause?
17:37.14*** join/#asterisk stony (n=steinche@p5B15196E.dip0.t-ipconnect.de)
17:37.14russellbcorrect, you have to have at least one
17:37.17Qwelllame!
17:37.38Qwellwell
17:38.04Qwellint one=1;\nsomefunction(table, one, 1);
17:38.24Qwellthat would probably fail miserably on some non-DB things
17:40.39russellbyes, kevin said that, too
17:40.48russellbI said, what about LDAP?  :)
17:40.54Qwellyeah
17:41.00Qwellstupid ldap :p
17:41.09russellbit's not even in the tree ...
17:41.24russellbsomeone should probably take that on to get it done by 1.6 ...
17:41.40ManxPowerrussellb: it is nice of your to volunteer
17:42.02russellbheh
17:42.21russellbjust tack it on my list
17:42.52*** join/#asterisk Cresl1n (i=matt@nat/digium/x-db3b326591e9148d)
17:42.52*** mode/#asterisk [+o Cresl1n] by ChanServ
17:42.58QwellCresl1n: !!!
17:43.08Cresl1nQwell !!!!!
17:43.11Cresl1nare you in the office today?
17:43.14Qwellnah
17:43.19Cresl1nlame.....
17:43.23Cresl1neverybody cool is here :-P
17:43.38*** join/#asterisk ToyMan (n=Stuart@74-32-55-210.dsl1.mdl.ny.frontiernet.net)
17:43.38QwellYou're there by yourself? :)
17:43.43Cresl1nheh :-)
17:43.46Qwell(kidding) :)
17:43.55blitzrageEveryone cool and Cresl1n too
17:44.00Qwellblitzrage: pwned
17:44.00ManxPowerCresl1n: Does Zaptel support DACS?
17:44.02Cresl1nnah, spiceland and rick loveman are here too
17:44.04russellbbkruse and I were thinking about going there and having a jam session, heh
17:44.08QwellCresl1n: ahh
17:44.10Cresl1nManxPower: oh no you didn't!
17:44.18Cresl1nhow many times do you have to explain it to someone
17:44.20Qwellrussellb: brt, I play a MEAN cowbell
17:44.28russellbsweet!
17:44.34russellbI have a couple cowbells I could bring.
17:44.38Qwellhaha, seriously?
17:44.39russellbexcept ... they are in storage ...
17:44.42Cresl1nwell, I explained I think well enough so that next time I can just tell the person to read the archives
17:44.44QwellI was totally joking
17:44.59russellblame
17:46.01blitzrageZaptel supports DACS?
17:46.12ManxPowerCresl1n: We used to use DACS extensively.
17:46.19QwellI need DACS!  Does Zaptel support it?  What is DACS?!
17:46.32russellbI totally need DACS for my office
17:46.36russellbelse I can't work
17:46.40russellband I blame Cresl1n
17:46.44ManxPowerQwell: Sangoma did not support DACS RBS until I complained to them and they fixed it in a driver update.
17:46.58QwellManxPower: well, what'd you go and do that for?
17:47.16russellb(I don't know what DACS is ...)
17:47.32ManxPowerThat was the only significant issue we have had with Sangoma.  And yes, we have had HDLC abort errors with a Sangoma card, so we know it DOES happen.
17:48.16ManxPowerrussellb: Basically you digitally cross connect 1 T-1 channel to another T-1 channel.  Sort of a digital patch panel at the 64k/56k channel level
17:48.41russellbah, neat ...
17:48.48*** join/#asterisk Insane00 (n=aamirwah@74-128-211-33.dhcp.insightbb.com)
17:48.50Qwellused for what?
17:48.54russellbI heard Sangoma eats babies
17:49.01ManxPowerhandy for example if you have a mixed voice and data T-1.  You can DACS the data channels out another T-1 port into whatever data device you have.
17:49.04Qwellrussellb: Well, they are Canadian
17:49.09Qwell...fast
17:49.24blitzrageQwell: I can run faster than you
17:49.31Qwellprobably
17:49.33Insane00Hello
17:49.33russellbManxPower: oh, alright, makes sense
17:49.34ManxPowerIt is VERY handy when putting Asterisk between the Telco and other T-1 equipment like channel banks, routers, and PBXs with T-1 ports
17:49.35blitzrageI'm sure of it :)
17:49.42Qwellbut, luckily, I've got a 3k mile headstart :P
17:50.03ManxPowerDACS is WHY we were able to transition to Asterisk for a significant number of our phones
17:51.13ManxPowerQwell: We tried Sangoma because of the long standing issue Digium cards have had with a large enough number of motherboards as to cause us issues.
17:51.36QwellManxPower: You shouldn't have those problems anymore
17:52.04ManxPowerAs most people know, many of those issues are supposed to be resolved in recent Zaptel releases, but we were committed to Sangoma well before those Zaptel releases
17:52.24Insane00I need help to undedrstand asterisk ? any body
17:52.36ManxPowerInsane00: you're insane.
17:52.45Insane00thats why i am here
17:52.54russellblol
17:53.07russellbI can tell you Asterisk starts with an A
17:53.13russellb~thebook
17:53.25jbotit has been said that thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:53.25Qwell~book
17:53.27jbotit has been said that book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:53.28Qwelljbot: you suck
17:53.29jbotno, *you* suck!
17:53.34ManxPowerInsane00: I assume you also want to start your own VoIP company next week.
17:53.34Qwellwtf
17:53.35Insane00did i asked you about spelling
17:53.48Insane00we are actually
17:54.06Insane00but i want to jump in to know some basics
17:54.07russellbQwell: Can you be helping me be the next Vonage?
17:54.46QwellInsane00: such as?
17:54.53QwellWe can't answer such a broad question
17:55.16ManxPowerInsane00: The Book is a good start.  PBXs are complex devices.  Linux is a complex system.  Networking is a complex thing.  Telco interconnects are complex.  Asterisk requires knowledge of all of these things
17:55.50Insane00hmm wow
17:55.51Qwellcorrection: Telco interconnects are incredibly complex
17:55.52Insane00i know linux
17:56.38Insane00but i think if you guys are hel me to understand the how asterisk route the calls
17:56.44ManxPowerBTW, www.sandman.com has lots of really cool telco stuff for very good prices.  Including things that take CPC tones and turn them into a battery drops, etc
17:57.24Insane00or in other words when a user dial a local number how its terminated locally
17:57.34ManxPowerhttp://www.sandman.com/loop.html#CPCGenerator
17:58.07Cresl1nYeah, I got some cool new firmware for the TE410P/TE405P as well
17:58.12Cresl1nhot rod firmware
17:58.19Cresl1nwe're beta testing it right now
17:58.27Insane00i will really apreciate if some one explain me
17:59.02*** join/#asterisk boch (n=fran@190.48.225.254)
17:59.04ManxPowerI'll be using them for things like emergency phones at clients with only 1 local line.
18:00.22ManxPowerThe site also has things like impedance matching devices.
18:00.23drfreezeTo use tftp to provision phones, the DHCP server doesn't need to be run from the Asterisk box, right?
18:00.53*** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net)
18:00.59ManxPowerdrfreeze: no.
18:01.11bochis it possible to make Record() stop recording with any key instead # ?
18:01.14drfreezeManxPower: thx
18:01.43ManxPoweryou just have to have the DHCP server add the correct option in it's response.  drfreeze: Without the DHCP option "next server" is one of them, many phones will just default to loading their configs from the DHCP server.
18:01.50ManxPowerCiscos do this AFIK
18:03.15Cresl1nit should help with motherboard compatibilty issues as well
18:03.18ManxPowerQwell you might find this interesting.  I got a line tester out of storage.  Calling a nearby 105 Test number (104Hz tone at 1db), my test set is showing a -10 to -13 db level at my 66 block
18:03.42*** join/#asterisk ming_zym (n=ming_zym@124.254.55.207)
18:03.53ManxPowerI suspect I have an impedance issue as I get echo even on TDM bridged Zap calls.
18:03.57QwellInsane00: That isn't really a question that can be answered
18:04.09Cresl1nugh
18:04.13Cresl1nManxPower: run fxotune
18:04.38ManxPowerCresl1n: on a T-1 card.
18:04.40ManxPowerthe analog lines are coming into an Adtran
18:04.58Cresl1nManxPower: oh, that's a whole different colored horse then
18:05.32ManxPowerCresl1n: I am 11,000 ft from the CO and going thru a SLC96
18:05.43un_j:-)
18:05.57Insane00Qwell: any book you recommnad
18:06.02ManxPowerso basically the worst possible case for PBX
18:06.06QwellInsane00: The one jbot said..twice
18:06.26un_jhey how to set up timeouts for iax (drops after 60 seconds) and if I call from sip ext to sip ext it drops also after 60 seconds :-)
18:06.34blitzragesomeone wrote a book?
18:06.40blitzragethose guys must be super cool
18:06.49Qwellblitzrage: sooo cool
18:06.52ManxPowerun_j: we can't help you with FreePBX and/or AMP
18:06.55russellbomg I wish I could meet them
18:06.58Qwellsomebody should give them a lot of money
18:07.02blitzragetotally
18:07.27blitzrageI heard they are building their own space station
18:07.49ManxPowerblitzrage: for total world domination, I assume?
18:07.55un_jI don't user free pbx its 1.4.2. pure no gui
18:07.57blitzragemost likely
18:08.28ManxPowerun_j: Good.  then we can help you 8-)
18:08.59un_jplease ;-)
18:10.21ManxPowerun_j: IAX and SIP should not timeout unless they are not getting any response from the far end or you have a timeout on the Dial line
18:12.10un_jhm, its gets: Auto fallthrough, channel 'IAX2/chicago-3' status is 'CHANUNAVAIL'
18:15.24un_jhow about rtptimeout?
18:22.01zoayoyo
18:28.34*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)
18:33.37JacksLivrhow do i make a channel hangup from the console?
18:33.54blitzragesoft hangup
18:34.25JacksLivrblitzrage: THANK YOU!!!!!!!!!!!! <3
18:34.39JacksLivrso easy, just not where i was looking
18:40.52*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:41.06*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:42.06un_jhey sip is ok but iax gets: [Mar 31 13:38:38] DEBUG[11742] chan_iax2.c: Received iseqno 12 not within window 13->15
18:42.06un_j<PROTECTED>
18:42.18un_j60 sorry
18:42.30*** join/#asterisk Dovid (n=Dovid@85.159.160.207)
18:42.33*** join/#asterisk eald_home (n=eald@189.157.104.40)
18:52.40*** join/#asterisk techie (n=gus@antibala.com)
18:54.14*** join/#asterisk techie (n=gus@voip.routedsystems.com)
19:04.58robin_szsigh ... echo is beginning to bug me bigtime
19:05.17*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
19:05.22robin_szchan_mISDN is the culprit I suspect
19:06.19*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
19:11.46*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:12.23*** join/#asterisk haagenti (n=user@212.91.233.233)
19:15.17haagentican asterisk do ss7/camel yet?
19:16.11ManxPowerun_j: That might indicate a serious networking issue.
19:17.14ManxPowerhaagenti: Short Answer: No.  Long Answer: Yes, but it is from some 3rd party and (I think) is commercial.
19:17.33*** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net)
19:18.34haagentihmmm... just curious
19:19.02d00gsterguys are dial plans universal? I mean a dial plan for asterisk works on sipura?
19:19.37Qwelld00gster: no, ATAs have their own dialplans
19:19.41*** join/#asterisk piper69 (n=piper@unaffiliated/piper69)
19:19.48piper69hi all
19:19.56d00gsterI see
19:20.03Qwelld00gster: there are some good examples on the wiki though
19:21.13piper69i use debian etch i am trying to install asterisk i get Setting up asterisk-sounds-main (1.2.13~dfsg-2) ...
19:21.14piper69Setting up fxload (0.0.20020411-1) ...
19:21.16piper69Setting up zaptel (1.2.11.dfsg-1) ...
19:21.19piper69Zaptel telephony kernel driver: FATAL: Module ztdummy not found.
19:21.21piper69Notice: Configuration file is /etc/zaptel.conf
19:21.24piper69line 0: Unable to open master device '/dev/zap/ctl'
19:21.26piper691 error(s) detected
19:21.29piper69/sbin/ztcfg failed. Check /etc/zaptel.confzaptel.
19:21.31piper69Setting up asterisk-classic (1.2.13~dfsg-2) ...
19:21.34piper69Setting up asterisk (1.2.13~dfsg-2) ...
19:21.34Qwell~paste
19:21.43jbotsomebody said paste was http://rafb.net/paste/
19:21.50piper69sorry
19:22.22piper69i do apologize
19:23.04*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw)
19:24.57JunK-YQwell: so how's the newest ubuntu?
19:25.10QwellJunK-Y: incomplete
19:25.22Qwellthere are quite a few packages missing
19:25.26JunK-Ylike?
19:25.28Qwellotherwise, it's the same as edgy
19:25.37JunK-Yis it ubuntu or kubuntu?
19:25.40Qwelllike, vmware-server, some amd64 things
19:25.43Qwellubuntu
19:25.45piper69Qwell: if you have time can you help me please i pasted my issue http://rafb.net/p/x95dil59.html
19:25.49Qwellsome kernel modules
19:25.52Qwellaudio is b0rked for my card
19:26.11*** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com)
19:26.13JunK-Ykk
19:26.32Qwelloh, and compiz/beryl is hosed
19:26.51JunK-Yive heard hood things about beryl
19:32.35*** join/#asterisk kannan (n=kannan@210.211.178.104)
19:32.36Qwellare you saying it's ghetto?
19:32.49QwellI *love* typos that make the sentence mean something completely different, heh
19:33.11JunK-Ys/hood/good/
19:33.12JunK-Y:)
19:33.25*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
19:33.29JunK-Ythere ya go jbot , you rock!
19:36.29blitzrageQwell: good stuff
19:37.20*** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com)
19:37.42drfreezeDoes anyone know what version of version of BootROM and SIP SW I should be using for Polycom?
19:37.50drfreezeis 1.6.7 ok for SIP SW?
19:40.57Qwellrussellb: ping?
19:41.14Dovid1.6.7 should be good
19:41.17Dovidi use it on my phones
19:42.23drfreezeDovid: thx. and bootrom 2.6.1 ok?
19:42.39Doviddont have a phone wit me now. dont know what boot rom i am using.
19:42.57Dovidthe SIP 2.x.x is what made and issue with asterisk 1.2.x
19:43.03MaghteridonIs there a way from the CLI to record a conversation that a user has with an external line?
19:43.42DovidMaghteridon: i dont think so. but u can spy in to a conversation and record it from there
19:43.42drfreezeDovid: what about asterisk 1.4.2? is it ok with SIP 2
19:43.48Dovidor u should be able
19:44.08Doviddr: i heard with 1.4.2 it's good but i never touched 1.4.x yet
19:44.14MaghteridonDovid, do you have to be on the asterisk box to do it?
19:44.28Dovidto spy
19:44.29Dovid?
19:44.30Maghteridonyes
19:44.34Dovididk if u can record from spying
19:44.48Maghteridonhow do you spy?
19:44.49Dovidno. it's a feature on asteirsk. u can do it from any phone
19:44.51*** join/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com)
19:44.56Dovidlook on the wiki
19:45.02Dovidi think the cmd is ChanSpy
19:45.06Maghteridonah okay thanks
19:46.08lungive got asterisk 1.2.17 core dumping on three seperate machines.. is this a proper outlet to dsicuss?
19:47.20JunK-Ylung: can ya reproduce the problem at any time?
19:47.28JunK-Y<PROTECTED>
19:47.28JunK-Y<PROTECTED>
19:47.30JunK-Youps
19:48.08lungJunK-Y: only in the sense that it happens so often, but no, im not seeing what exactly is triggering it.. its happening very often across all three servers though.. at least hourly
19:48.29JunK-Ylung: read the file: README.backtrace
19:48.34JunK-Yin ur doc/ dir
19:49.27*** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud)
19:49.33lungi have many cores :p   and i compiled valgrind
19:50.15JunK-Ytry to see if theres no duplcate already existing bugs first at bugs.digium.com
19:50.21lungyeap
19:50.31zoahttp://www.autoblog.com/2007/03/30/bulgaria-gets-a-loan-to-buy-fleet-of-porsche-cayenne-ambulances/
19:50.33zoaloool
19:50.34zoastupid fucks
19:51.19Qwellzoa: Where are you from?  I always get those B's mixed up...
19:51.30zoaim from belgium
19:51.32QwellBulgaria, Bolivia, Belgium..
19:51.33zoabut living in bulgaria
19:51.34Qwellahh, okay
19:51.38zoao ho ho
19:51.44zoaim coming to the devcon!!!
19:51.45zoawhiii
19:51.58QwellI told redbull they need to sponsor us just now :P
19:52.01JunK-Yzoa: you live in bulgaria? didnt know taht.
19:52.08Qwellhow freaking awesome would free redbull be?
19:52.17JunK-Yzoa: i wont be able to afford it sadly :(
19:52.18zoai dont drink redbul
19:52.22zoaits killing me
19:52.25Qwellzoa: bah!
19:52.29Qwellquit for a month
19:52.30mvanbaakQwell: any change there will be Jolt ?
19:52.35Qwellmvanbaak: none
19:52.38zoai sometimes drink it
19:52.40QwellJolt is illegal in Georgia
19:52.44zoabut it makes my heart go crazy
19:52.45mvanbaakreally ?
19:52.47Qwellno
19:52.47mvanbaakwhy ?
19:52.57zoaJunK-Y: where do you live ?
19:53.04mvanbaakit's hard to get here in .nl
19:53.12mvanbaakI always go to Germany to get it
19:53.17Qwellmvanbaak: isn't redbull too?
19:53.19zoamvanbaak: je kan het online vinden denk ik :)
19:53.23Qwellor was that...norway?
19:53.24zoaredbull is very easy to find
19:53.25Qwellstupid N's!
19:53.25mvanbaakzoa: klopt
19:53.27JunK-Yzoa: im in montreal, quebec in canada.
19:53.28zoathey have it everywhere
19:53.31zoaaha
19:53.37mvanbaakQwell: redbull is legal here
19:53.49Qwellsomebody said it was illegal in some country that started with an N, I thought
19:53.52Qwellmog would know
19:54.00zoaQwell: everything is legal in the netherlands
19:54.05zoathey sell magic mushrooms there
19:54.06mvanbaaklol
19:54.08Doviddamn
19:54.10mvanbaakyeah!
19:54.12mvanbaakshrooms !
19:54.13Dovidzoa: when can i move ?
19:54.20zoayou dont want to
19:54.21zoa:)
19:54.32LennonNZdoes anyone have a working configure to set up sipdiscount.com to asterisk
19:54.38LennonNZall I'm getting at the moment from them is
19:54.42LennonNZGot SIP response 500 "Internal server error" back from 80.239.235.200
19:54.54mvanbaakI get that a lot from sipdiscount as well
19:54.56mvanbaakit's them
19:55.05mvanbaakbecause sometimes it works out of the blue
19:55.23LennonNZI had it working.. and then suddenly I get this all the time
19:55.30mvanbaakyeah
19:55.33mvanbaakget used to it
19:55.42mvanbaakhappens at least twice a week with them
19:56.02LennonNZI am looking at getting some free callin/out numbers
19:56.08Cresl1nredbulll.......
19:56.15LennonNZthere is a list os free providers anyplace I can get numbers/connetions from?
19:56.27QwellCresl1n: :D
19:56.33mvanbaakLennonNZ: remember that you get what you pay for
19:56.35JunK-YCresl1n: ya just heard the magic word huh?!
19:56.36Qwellsomebody tried that last year, didn't they?
19:56.49LennonNZmvanbeek: true
19:56.57Cresl1noh yeah
19:57.22zoahey look its Cresl1n
19:57.25zoaits a live
19:57.25zoaalive
19:57.30Cresl1nzoa!~!!!
19:57.35zoaCresl1n: are you taking your wife ?
19:57.44zoai showed you mine now show me yours
19:57.46zoa:P
19:57.54mvanbaakehm......
19:57.54LennonNZis there anyway I can play music and a "ring ring".. "ring ring" sound at the same time whilst calling someone bfore they answer the phone?
19:57.55mvanbaaklol
19:58.23zoaSHOW ME THE GOODIES!
19:58.24zoa:)
19:58.24LennonNZat the moment I can do either, but not both
19:58.32mvanbaakthe only way to do that is to put that ringring in your music
19:58.35mvanbaakmix it together
19:59.12LennonNZoh well
19:59.31*** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl)
20:02.02dan42JunK-Y: most of the lines in my backtrace are "in ?? ()" .. am i doing something wrong?
20:02.21blitzragedan42: did you compile with DONT_OPTIMIZE?
20:02.30JunK-Ydan42: you need make dont-optimize
20:02.34dan42blitzrage: 1.2.17 compiled valgrind
20:02.43blitzrage1.2: make dont-optimize ; 1.4: Set DONT_OPTIMIZE in menuselect
20:02.43dan42which does the same thing
20:03.05dan42valgrind: dont-optimize
20:03.23JunK-Yhave ya started with -g too?
20:03.27dan42seems like its lines that would be in the libs outside of asterisk
20:03.29dan42yes
20:03.54dan42i dont think i would have a core otherwise, right?
20:04.01dan42but yes all thew same
20:04.22dan42i have my output from gdb, but its not leading ME anywhere
20:04.36dan42not that that means anything
20:05.50*** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com)
20:11.14*** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca)
20:11.54*** join/#asterisk harleya (n=harleya@c-67-161-253-232.hsd1.ut.comcast.net)
20:12.15robin_szok, so to improve the crap echo performance of chan_mISDN I should?
20:12.30drfreezeIs there a way to have an auto provisioning file for all 301 phones and one for all 501 phones?
20:12.36*** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net)
20:12.48drfreezerobin_sz: call digium?
20:12.58robin_szdrfreeze, do they retreive their configs from a web swerever?
20:13.09drfreezedon't know
20:13.12d00gsterguys, would can I force international calls to dual a callingcard DiD before the actual number?
20:13.35robin_szd00gster, yes
20:14.04zoathe HPEC might work on misdn
20:14.08zoacheck with creslin
20:14.09d00gsterhow do I do that for oubound calls?
20:14.13zoahe should know
20:14.28Qwellzoa: do any software echo cans work with misdn?
20:14.31QwellIf so, hpec should work just fine
20:14.35Cresl1nnah, HPEC doesn't work with misdn
20:14.41Cresl1nthere'd be licensing issues anyways
20:14.47Cresl1nmisdn kernel code is only GPL
20:14.52Cresl1nit'd be ugly
20:15.13robin_szd00gster, inyour dialplan, just test for the first digits of an international number, and then put the calling card numbers before the  EXTEN
20:15.34robin_szuse includes to include that pattern before the local patterns
20:16.09robin_szthere are several examples of this in the default config
20:16.29dan42JunK-Y: thanks for the help, just wanted to make sure i did that right.. i just posted a bug.. didnt see another one like it
20:16.39robin_szzoa, HPEC?
20:16.43d00gsterrobin_sz I need that for outbound calls
20:16.54robin_szthere are several examples of this in the default config
20:17.12d00gsterok
20:18.01zoaCresl1n: doesnt have to be a problem the octasic people seem to have gotten around that
20:18.14JunK-Ydan42: <value optimized out>   so isnt dont-optimize
20:18.18*** join/#asterisk Corydon76-home (i=brown@pdpc/supporter/sustaining/Corydon76-home)
20:18.18*** mode/#asterisk [+o Corydon76-home] by ChanServ
20:18.29robin_szso, basically ... unless chan_mISDN suddenyl has a new burst of code ... I need to look for a real card, rather than this cologne card?
20:18.47JunK-Ydan42: and we need a bt full too.
20:18.49rad07Everybody: I installed Asterisk 1.4.2. Any Web interface that works with it? I didn't install samples and I need to connect Linksys ATA  SPA3102. I know where is web based config page for my ATA, but I am not sure of the steps I need to do to setup Asterisk to accept/receive calls via ATA. Any step by step guides? If I have to do it manually which files (I assume after installing Samples) should I configure?  Please help me. I asked this q
20:19.08dan42JunK-Y: its all there.. "bt", "bt full", and "thread apply all bt"
20:19.52Qwelldan42: from a non-optimized build
20:19.52robin_szrad07, configure you sip.conf ... as show in the book
20:19.52*** join/#asterisk Assid (n=assid@203.212.204.107)
20:19.53dan42Qwell: its compiled non-optimized
20:19.53JunK-Yoops, sorry, miss the first part.
20:20.46*** join/#asterisk boch (n=fran@190.48.209.242)
20:21.14rad07robin_sz: Which book? Is it the first step to install Samples? After that what I configured actually? Is it just sip.conf that I need to touch?
20:21.24Qwell~book
20:21.30jbotrumour has it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
20:21.30Qwellthat one
20:21.31Qwell...
20:21.32Qwellthat one
20:21.46dan42Qwell: i did "make valgrind" actually since thats what was in voip-info .. but that seems to just run dont-optimize from what i can tell
20:21.53Qwelldan42: yeah
20:22.01rad07Is it still relevant (accurate) for latest Asterisk version?
20:22.11Qwellrad07: mostly
20:22.31Qwella new book will be published eventually
20:22.53rad071. Install samples. 2. Configure sip.conf, Next???
20:23.06Qwellrad07: Profit!!!
20:23.13*** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-179-77.ny325.east.verizon.net)
20:23.20dan42heh
20:23.23robin_szconnfigure your ata to connect to your * server, using the name/pw you set in sip .conf
20:23.26rad07It cannot be that simple
20:23.31*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)
20:23.34dan42is that res_profit or app_profit?
20:23.49robin_szstep 1) steal underpants
20:23.53robin_szstep 2) ....
20:23.58robin_szstep 3) profit!
20:24.43Qwelldan42: chan_prophet
20:25.28robin_szso ... where was I?   a single hfc/cologne card // what are my options other than chan_mISDN
20:25.51rad07Guys. Does anybody teach Asterisk? I am willing to pay small amount of money to learn basics. I need to integrate Web based conferencing with Asterisk and I need to be able to compile some modules/packages from third party? Anybody with programming knowledge?
20:25.56Qwellrobin_sz: no hardware echo can for those cards?
20:26.37Cresl1nrobin_sz, is it not a b410p?
20:26.42Cresl1nthey have HW echo can on them
20:26.42robin_szabsolutely not
20:26.53robin_szb410p?
20:26.56Cresl1nyeah
20:26.59Cresl1nthe digium bri card
20:27.07wwq222is there a way to do outgoing call queueing in asterisk w/o manually controlling the number of files in the outgoing folder?
20:27.11robin_szits just a plain old HFC/cologne data card
20:27.15Cresl1noh
20:27.26Qwellwwq222: you can `touch` the files into the future
20:27.34robin_szecho cancel works ok say, 80% of the time
20:27.34Qwellwwq222: I don't know the syntax offhand, but `man touch`
20:27.47*** join/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net)
20:29.21*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:29.21robin_szI don't think I can afford a digium card,  out of my league
20:29.21wwq222qwell:  thanks - i'm trying to implement some kind of queuing scheme because i can only make so many outgoing calls at once - so i want to try to get the next call to go once the current one finishes
20:29.21*** join/#asterisk duki (n=duki@host-85-27-49-12.brutele.be)
20:29.22Qwellahh
20:29.22*** part/#asterisk cthorner (n=cthorner@209-234-185-130.static.twtelecom.net)
20:29.22QwellYou probably want to write something that connects via manager
20:29.22wwq222qwell:  I thought about just checking the number of files in the outgoing directory, andthen moving more in there when it drops below the limit
20:29.45*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
20:29.48d00gsterrobin_sz, I can't find the pattern in the  configs. In my sipura, If I put <999011:011> is will append 999 before outbound number that starts with 011. what's the equivilance in asterisk?
20:30.25*** join/#asterisk alejandro (n=asanchez@kde/developer/alejandro)
20:30.37Qwelld00gster: _011.,1,Dial(999${EXTEN})
20:30.42Qwellsomething like that would do
20:30.47[TK]D-Fenderd00gster, There is none, because the * dialplan does whatever you TELL it to.
20:30.59[TK]D-Fenderqwell : yes.. something LIKE that ;)
20:31.12Qwellexcept I completely hosed the tech
20:31.17[TK]D-Fendershhh!
20:31.57[TK]D-Fenderqwell : Tell not the see-rat!
20:32.07[TK]D-Fenderseek*
20:32.14*** join/#asterisk Fieldy (i=suQbustO@gentoo/contributor/Fieldy)
20:32.16alejandrohi, Im trying to configure a SPA 3102 with Asterisk. I configured in the SPA as SIP client and it's connected to the PSTN. How i can configure a basic config so softphones can call the PBX and later redirect to the PSTN ?
20:32.29d00gsterQwell, I lost you  there, what I am trying to do is to prepend a 2121111111 <Pause 2 sec> before a 011. number .... can you help me with that?
20:32.29alejandroIt's the first time that I configure Asterisk as a PBX, and it's a little hard. :-)
20:32.43*** join/#asterisk Corydon76-home (i=green@pdpc/supporter/sustaining/Corydon76-home)
20:32.43*** mode/#asterisk [+o Corydon76-home] by ChanServ
20:32.51Qwellhaha, I just got my t-mobile bill
20:32.59Qwelland they charged me for sending a txt msg to myself
20:33.04Qwellthat is awesome
20:33.13robin_szd00gster, did you find the "international"  bit of the diaplan?
20:33.24d00gsterno robin_sz
20:33.33robin_szwell, you need to find that first
20:33.38QwellI'm surprised they didn't double-charge me for it, since incoming isn't free
20:33.55zoalol
20:34.07zoaits abnormal if you ask me
20:34.08robin_szd00gster, do you know which file to look in to find the dialplan?
20:34.16Qwellooo, but I did get charged double for sending one to my wife
20:34.17zoathey should have charged you double
20:34.26d00gsternope.
20:34.40Qwelland I got charge for sending messages to an invalid number
20:35.20Qwellooo, I'm mad now :D
20:35.37robin_szd00gster, its in extensions.conf ... now, you need to read the docs. really. what you will get on here is help if you have read the docs and tried to learn but we will not just spponfeed you .. we will tell you were to look for the sppon though :)
20:35.47[TK]D-Fender~book
20:35.50jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
20:35.52robin_szand that
20:35.53[TK]D-Fender~wikis
20:35.54jbotsomebody said wikis was http://www.voip-info.org
20:35.59MahmoudQwell, lol @ double charge
20:40.32*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)
20:41.47*** join/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca)
20:42.14MrTelephonecan someone help me with the Cellular gateways? Is there something that acts like a mini cellular base station?
20:42.31zoayes there are
20:42.36zoabut i dont have any links
20:42.38zoathey are affordable
20:45.26MrTelephoneare you responding to me?
20:45.29zoayes
20:45.52MrTelephoneso they are base station then? when I read one description it sounded like it uses the preexisting cellular network
20:46.02MrTelephonei want to put up a cellular network in the bush
20:46.06zoayes there are base stations
20:46.10zoait exists
20:46.20zoalook for micro cells or so
20:46.22MrTelephonehttp://www.hyperms.com/index.asp?mainpage=prod_enlarge&prodcat=2603&prodtbl=260300&prodid=106
20:46.27blitzragehow much is a base station? That'd be fun to have a base station in downtown Toronto :)
20:46.47drfreezeCan someone explain to me what a LineKey is (reg.1.lineKeys on Polycom cfg)
20:46.51MrTelephoneit's a licensed spectrum but im putting it on native reserve land
20:47.08blitzragethat makes sense
20:47.08MrTelephonelinekey is the lcd butotns on the display
20:48.26MrTelephoneu can have up to 3 line keys on a polycom 501
20:48.26*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)
20:48.26MrTelephoneis the link I pasted a base station?
20:48.26zoai thought that was the name
20:48.26MrTelephoneso if I plug that crap in I should be able to use my cell phone?
20:48.26zoablitzrage
20:48.26drfreezeMrTelephone: how does that differ from callsperlinekey?
20:48.27zoai dont remember
20:48.27zoabut its cheap
20:48.27[TK]D-FenderYou DO have exactly 3 LineKeys on an IP 501.  What you DO with them is up to you.
20:48.27MrTelephoneyou can have more than one conversation with each line key
20:48.27drfreeze[TK]D-Fender: what is a LineKeys? A 'line'?
20:48.27MrTelephoneits like call-limit in sip.conf
20:48.37[TK]D-Fenderdrfreeze, No, not quite.
20:48.55MrTelephoneusually u want one line key.. 1 call per line key or something
20:49.14MrTelephoneor 3 line keys, 1 call per line, then when a second person calls it shows up on line key 2
20:49.36zoaor try pico cell
20:49.42MrTelephonepico cell?
20:49.53[TK]D-Fenderdrfreeze, An IP 501  can support up to 3 completely distict regitrations.  to do so you must allocate 1 linkey to each.  From there though you can support from 1-8 calls on each line key.
20:50.35drfreeze[TK]D-Fender: by registration - do you mean their own phone number?
20:51.12[TK]D-Fenderdrfreeze, Most people only use a SINGLE registration with a phone however.  With this in mind you can sake "linekeys" = 3, and callperlinekey=1, and that way when youa re on a call using the first linekey, a subsequent incoming call will ring on the SECOND.  And the tird on the thrid
20:51.32[TK]D-Fenderdrfreeze, Yes.  registration = seperate account entirely.
20:52.16MrTelephonei need a cdma base station
20:52.16[TK]D-Fenderdrfreeze, 1 same would be to use reg1 to have 2 linekeys @ 1 call max each, and reg2 having 1 key supporting 5 calls.
20:52.32MACscrhow can i resolve/verify a sip address?
20:52.47drfreeze[TK]D-Fender: so, if I want a 'traditional feel' for a phone and want it to have 3 lines, I set linekeys=3 and callsperlinekey=1 or linekeys=1 and callsperlinekey=3?
20:53.00drfreezeMACscr: sip show peers
20:53.21[TK]D-Fenderdrfreeze, The former
20:53.28drfreeze[TK]D-Fender: thx
20:53.34[TK]D-Fenderdrfreeze, And that will be a "traditional" feel
20:53.46MACscrdrfreeze : that would only work if it was a peer, what if wanted to verify and address
20:53.52MACscrwhoops, any
20:54.15[TK]D-Fenderdrfreeze, since I use my IP 501 with multiple clients, I give 1 line key to each reg w/ 5 calls each max (you use the cursor keys to scroll through calls.  Its vrey nice actually).
20:54.25MACscrbasically i can receive phone calls just fine with my number, but not by my sip address
20:54.39[TK]D-Fenderdrfreeze, Polycom's call handling capabilities are superior to every phone I have ever used.
20:54.53MACscrand i want to figure out if its my address (received from provider) or my * box
20:55.02drfreeze[TK]D-Fender: what's the advantage to having multiple lines on a linekey?
20:55.33*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
20:56.02drfreeze[TK]D-Fender: is that like call waiting, but with 5 lines?
20:56.40drfreezeMACscr: I don't think I'm the one to ask. :(
20:57.11*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:57.21[TK]D-Fenderdrfreeze, Well if you want multiple regs, this way you can support multiple calls per and conference any which way you want.  Also you can use a single line-key even for just 1 reg supporting multiple call so as to leave 2 available for use as speed-dials.  Very useful when associated for presence.
20:57.30zoa3600 downloads of idefisk 2.0 so far
20:57.32zoanot bad
20:57.41zoafor 60 hours or so
20:58.29MACscrdrfreeze : i only asked after you responded to me =P
20:58.39drfreeze[TK]D-Fender: oh, so the conf button on the phones is to connect lines on a linekey?
20:59.05drfreezeMACscr: I know. sorry about that
20:59.19[TK]D-Fender<PROTECTED>
20:59.49[TK]D-Fenderdrfreeze, If you simply have 2 distinct calls in progrees and you want to merge into a conference, that is the purpose of the "join" soft-key
21:00.06LennonNZzoa: your dev of it?
21:00.18QwellLennonNZ: he pays the bills ;)
21:00.33LennonNZhow much is the OEM vers of it?
21:00.34*** part/#asterisk MrTelephone (n=DeaLER25@bas13-toronto63-1177850949.dsl.bell.ca)
21:02.05zoaqwell see my privmsg
21:02.47VecIs it possible to set an ip address range of allowed peers in sip.conf (for each peer) I remember seeing it somewhere but can't find it anywhere i.e a ACL?
21:02.57Dovidzoa: Qwell dont like PM's
21:03.10zoasure he does
21:03.13MACscrso no one can tell me how to verify a sip address?
21:03.17drfreeze[TK]D-Fender: In a polycom.cfg file, is the reg.1 a linekey?
21:03.25Qwellzoa: I saw it :p
21:03.58zoaso reply you little bastard :p
21:04.02drfreezeMACscr: ok, let's give is a try
21:04.09drfreezewhat exactly are you trying to do?
21:04.13Assidzoa: new one out ?
21:04.37zoayes
21:04.39zoadoing sip now
21:04.41MACscrdrbreeze: im trying to verify that my sip address is correct as i cant seem to register it with enum
21:04.59bochis it possible to make Record() stop recording with any key instead # ?
21:05.02*** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it)
21:05.06*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
21:05.10Assidzoa: yeah i know.. you sent me the beta remember..
21:05.13Dovidyes boch:
21:05.14Assiddid you fix those bugs?
21:05.16Dovidsimple hang up
21:05.23drfreezeMACscr: lets get some context. your phone registering with *?
21:05.26*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
21:05.28markithi, the "voice" I get from a ISDN BRI card, what "codec" is it using?
21:05.39Qwellmarkit: g711
21:05.40drfreezedoes 'sip show peers' show unkown?
21:05.50markitQwell oh, thanks a lot
21:05.51bochDovid, what do you mean with hangup ?
21:05.55russellbQwell: pong
21:05.58zoaassid: so it got a little more stable in the mean time
21:06.15zoarussel did you see my message about the 1 port on asterisk-dev ?
21:06.21Dovidboch: my bad. didnt see what u wrote. idk of switching it rom # to another key
21:06.22zoa1 port 1 thread ?
21:06.30zoamaybe we need 2 ports :)
21:06.32russellbzoa: yeah, I sure did
21:06.33russellbheh
21:07.12zoawe could do a double register for it
21:07.16Qwellrussellb: nevermind :p
21:07.18zoawouldnt break the nat stuff
21:07.18russellbzoa: I'm still pondering it.  You have definitely made me wonder if putting in IAX2 is actually the best idea
21:07.25zoahehe :)
21:07.26zoagreat
21:07.52*** join/#asterisk Fieldy (i=TlY8ZsWD@gentoo/contributor/Fieldy)
21:08.13Qwellman, gentoo gives out cloaks like candy
21:08.14zoai think the kernel proposal might be ok
21:08.15russellbIt make more sense to go with a new protocol using TCP ...
21:08.20drfreezeMACscr: solve your problem?
21:08.21zoai also think so
21:08.40Qwellzoa: eh, there are other ways to do that..  no need for a kernel modules (that would be linux specific...)
21:09.08zoamaybe its something more for dundi ?
21:09.12zoacant we put that into dundi ?
21:09.51MACscrdrfreeze : my context works fine as far as incoming and outgoing when it comes to regular PTSN numbers.
21:09.54*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)
21:09.57zoaand move dundi into more like an asterisk signalling / clustering protocol
21:10.23Qwellwell, it WOULD be for talking to asterisk
21:10.31Qwelland, if you'll all recall what IAX stands for... :p
21:10.33russellbzoa: That's an interesting thought, I have been wondering that myself
21:10.42QwellI mean, perhaps a tcp thingie for iax
21:10.48Qwell(but NOT for audio)
21:10.50russellbbut then I was thinking use DUNDi for the discovery of event peers ... and IAX2 as the event peering protocol
21:11.15zoafor messaging stuff, iax would be fine
21:11.16Qwellthere really is no reason iax2 couldn't listen on udp and tcp
21:11.26zoaand why not we could put CDR's in there too
21:11.27Qwell...for signalling and stuff like that
21:11.31zoabut im not sure it would be ok for all cases
21:12.07Qwellbut the thing is, even with tcp, you're still gonna need to be 100% certain that the record got to the other machine
21:12.16zoayes
21:12.22[TK]D-Fenderdrfreeze, No, reg1 is not a linekey.  You have to tell it HOW MANY it will receive, and how many calls will be supported on each.  If you get reg1 2 keys with 2 calls each, it will then support up to 4 calls, with the third & forth (if received consecutively) will fall on the 2nd line-key.
21:12.24zoamy collegue will post some questions about failover with it
21:12.25Qwellserver could be flat out down/off
21:12.30zoawhat if the receiving server is down
21:12.36QwellSo you'd need to keep queueing them up
21:12.39zoaeven that should be handled
21:12.44zoawith the possibility of multiple ones
21:12.47zoawith resynching
21:12.51russellbDundi could also be the transport of events, i guess.  You'd only send events to a few servers and they would make it through the network ...
21:12.54Qwellso, IMO, I don't think we really need to even go the tcp route
21:14.00Qwellthis is probably a somewhat inappropriate place for this discussion too..
21:14.09drfreeze[TK]D-Fender: thx for that explanation. What is the purpose of reg.2 - reg.6?
21:14.11russellbwe could switch to #asterisk-dev :)
21:14.24[TK]D-Fenderdrfreeze, And there is no polycom.cfg, and you should never put registration stuff in sip.cfg either, only in the phoneXXX.cfg.
21:14.38[TK]D-Fenderdrfreeze, other completely differnt identities.
21:14.47drfreeze[TK]D-Fender: yeah, that is what I meant. phonexxx.cfg
21:15.25drfreeze[TK]D-Fender: ok, so I think I'm getting it. So they can connect to different providers.
21:15.25[TK]D-Fenderdrfreeze, I am registered to MY server at home with 1 reg (using 1 linekey @ 5 calls), and I register to REMOTE PBX's with my other 2.
21:15.29*** join/#asterisk Fieldy (i=1PCoslZL@gentoo/contributor/Fieldy)
21:15.35[TK]D-Fenderdrfreeze, NOW you're getting it...
21:15.50[TK]D-Fenderdrfreeze, This is not typical in your average home/office
21:15.57drfreeze[TK]D-Fender: is there a way for the user to tell where the calls are coming from? other than ring type?
21:16.06[TK]D-Fenderdrfreeze, But jsut another reason for Polycom's superior call handling :)
21:16.23[TK]D-Fenderdrfreeze, can you clarify that a bit...
21:16.48Assidyou really should take up polycom support man..
21:17.39drfreeze[TK]D-Fender: when a comes to your phone thru your server and when one comes thru your remote pbx. Can you tell from the LCD display on your 501?
21:17.56drfreeze*when a call
21:18.16[TK]D-Fenderdrfreeze, You mean as opposed to from those 2 other regs?
21:18.27drfreezeyes
21:18.28Assiddrfreeze: the caller id shows up as what is received
21:18.46[TK]D-Fenderdrfreeze, as I gave each reg its own line key, the one thats ringing is animated.
21:18.53drfreezeAssid: that won't tell you the src
21:19.18[TK]D-Fenderdrfreeze, So on 1 client, linkey 2 will be flashing for that incoming call.
21:19.19Assidwelll shows you the line key thats calling in
21:20.28drfreezemaybe it doesn't matter. Was just wondering if you could tell when a call came thru a provider A vs provider B
21:21.16Assiddrfreeze if its over a different line key (whichever is registered with whichever pbx), that linekey will show ringing
21:21.43[TK]D-Fender<PROTECTED>
21:21.52[TK]D-Fenderdrfreeze, Instantly
21:22.48[TK]D-Fenderdrfreeze, You have no idea what other mucking I do to accomodate the fact that MY * is reg'd to a lot of other PBX for which I use a prefix code to dial out of. :)
21:23.06drfreeze:)
21:23.38drfreezeI have an office that may need to start doing just such trickery. :)
21:23.49[TK]D-Fenderdrfreeze, I then prefix callerid coming in from each to make sure it goes back out the right system :)
21:24.09Assid[TK]D-Fender: sipbroker ?
21:24.11drfreeze[TK]D-Fender: woo, smart
21:24.26[TK]D-FenderAssid, huh?
21:24.39*** join/#asterisk `p4r14h (n=j0sh@69.92.145.178)
21:24.52Assidhave you tried sipbroker?
21:25.19[TK]D-FenderAssid, Nope.
21:25.29[TK]D-FenderAssid, What do i need ITSP's or... I have CLIENTS ;)
21:25.44[TK]D-Fenders/or/for/
21:29.53l1nuxgood night all :)
21:29.56*** part/#asterisk l1nux (n=moi@jof81-1-82-245-67-40.fbx.proxad.net)
21:31.44*** join/#asterisk ruied (n=ruied@bl7-217-252.dsl.telepac.pt)
21:33.29[TK]D-Fender~book
21:33.38jbotmethinks book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
21:40.05*** join/#asterisk FieldySnuts (i=0TnvqF3f@gentoo/contributor/Fieldy)
21:40.13MACscranyone know of a service i can use to make a test sip call to me?
21:41.28DovidMACscr: u wana test incoming sip ?
21:41.38zoaits called asterisk and you can download it for free
21:41.39Dovidu can use a sogphone w
21:41.40MACscrDavid: yes
21:41.41Dovidhehe
21:41.42zoa:)
21:42.20MACscrmy DIDs work fine, but i need to get my sip address to work so i can register with enum
21:42.49[TK]D-FenderMACscr, Gimme a URI and I'll call you
21:44.05MACscrPM sent
21:44.53MACscrmy sip provider provides URI's as welll that i should be able to use too, right?
21:47.43zoaim off to bed
21:47.44zoabbye
21:48.19[TK]D-Fenderzoa, later
21:49.47Assidalrite im off to bed as well
21:49.53Assidgnight
21:59.07*** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca)
21:59.36JunK-Y%ast core show application record
22:02.51russellb%ast originate Local/12562486000@default extension Local/12564286098
22:03.11russellb:)
22:03.26JunK-Y%ast core show channels
22:03.41[TK]D-Fender<PROTECTED>
22:03.50JunK-Yrussel: 0 active channels
22:03.51[TK]D-Fender:( pwnd
22:03.52Qwell%ast !shutdown -h now
22:03.58JunK-Y%ast stop now
22:04.01*** mode/#asterisk [+b *!*n=junky@*.185-70-69.mc.videotron.ca] by Qwell
22:04.01*** kick/#asterisk [AMUG!n=north@pdpc/sponsor/digium/Qwell] by Qwell (bot)
22:04.08russellb%ast !cat /dev/urandom /tmp/fillmydisk
22:04.13[TK]D-Fenderz0mg!
22:04.18Qwelloh, heh, JunK-Y
22:04.25*** mode/#asterisk [-b *!*n=junky@*.185-70-69.mc.videotron.ca] by Qwell
22:04.33Qwellsorry :D
22:04.36JunK-Yhehe
22:04.59JunK-Y%ast dialplan add foo
22:05.03Qwell%ast \!shutdown -h now
22:05.07*** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca)
22:05.08Qwell%ast \!shutdown -h now
22:05.20Qwellboo
22:05.23Qwell%ast !shutdown -h now
22:05.36JunK-Y%ast \!ls
22:05.46Qwell%ast !reboot
22:06.04JunK-YQwell: continue, he just told me that he likes it.
22:06.07russellb% !cat /etc/shadow
22:06.22russellb%ast !cat /etc/shadow
22:06.27JunK-Y%ast !cat /etc/passwd
22:06.46Qwell%ast dialplan add extension die,1,System,reboot
22:06.56russellblol
22:06.58JunK-Ydialplan add extension die,1,System,reboot
22:07.02JunK-Y%ast dialplan add extension die,1,System,reboot
22:07.19russellb%ast sip show users
22:07.30russellbI got a password!
22:07.31russellbscore!
22:07.34Qwell%ast dialplan add extension die,1,System,reboot into blah replace
22:07.43JunK-Ynow exploit it :)
22:07.52russellb%ast manager show users
22:08.00JunK-Yrussellb: thats why i submit a damn patch!
22:08.12russellb:-p
22:08.20russellb%ast iax2 show users
22:08.21JunK-Yhttp://bugs.digium.com/view.php?id=9273
22:08.22Qwell%ast unload chan_sip
22:08.24JunK-Ycould ya commit it?
22:08.25Qwell%ast unload chan_sip.so
22:08.40russellb%ast restart now
22:08.42JunK-Y%ast restart now
22:08.43JunK-Yhjeheje
22:08.49JunK-Yi allow a restart now so far
22:08.52JunK-Y%ast core show uptime
22:09.00JunK-Ysince, this is only a dev machine.
22:09.02russellb%ast core show channels verbose
22:09.07JunK-Y%ast core show uptime
22:09.21russellb%ast convert tt-weasels.gsm tt-weasels.wav
22:09.23russellb%ast convert tt-weasels.gsm tt-weasels1.wav
22:09.25russellb%ast convert tt-weasels.gsm tt-weasels2.wav
22:09.27russellb%ast convert tt-weasels.gsm tt-weasels3.wav
22:09.29russellb%ast convert tt-weasels.gsm tt-weasels4.wav
22:09.36Qwellno, no, no!
22:09.36JunK-Ymouahah ah
22:09.48Qwell%ast convert tt-weasels.gsm tt-weasels.ilbc
22:09.51Qwell%ast convert tt-weasels.gsm tt-weasels.lpc10
22:11.38Qwellharder codecs :P
22:11.38russellb%ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river.ilbc
22:11.38russellb%ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river1.ilbc
22:11.39russellb%ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river2.ilbc
22:11.39russellb%ast convert /var/lib/asterisk/moh/fpm-calm-river.wav /var/lib/asterisk/moh/fpm-calm-river3.ilbc
22:11.39JunK-YNo such command 'convert tt-weasels.gsm' (type 'help' for help)
22:11.39russellb:(
22:11.39Qwelltrunk, file convert
22:11.39russellbbah
22:11.39JunK-Y%ast core show uptime
22:11.39russellb%ast core show version
22:11.44russellbit should have that command!  :)
22:12.12JunK-Y%ast core show version
22:12.16JunK-Y%ast core show channels
22:12.16Qwell%ast dialplan show
22:12.24JunK-YQwell: i autorize it too
22:12.28Qwelllol, gui
22:12.30Qwellnice
22:13.08Qwell%ast include context default into DID_trunk_1
22:13.08Qwell:D
22:13.19JunK-Y%ast include context
22:13.27JunK-Ycould ya commit: http://bugs.digium.com/view.php?id=9273 ?
22:14.00*** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)
22:15.55*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net)
22:16.15*** join/#asterisk garreel (n=garreel@host48-2-dynamic.16-87-r.retail.telecomitalia.it)
22:16.45*** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net)
22:17.47*** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net)
22:18.34Qwell%ast moh show files
22:18.45Qwellboo
22:19.21JunK-Ytheres no crucial information on that machine.
22:21.26russellbwhy not
22:21.43Qwell%ast indication en add 247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16
22:21.49Qwellerm
22:21.51russellb%ast console dial demo
22:22.01russellbooh, that one is allowed
22:22.11Qwell%ast indication remove en ring
22:22.14russellb%ast dialplan show
22:22.18Qwell%ast indication add en ring 247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16
22:22.30Qwell%ast indication add en ring "247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16"
22:22.36Qwellhot
22:22.58russellb%ast console dial vmenu_record
22:24.09cr4z3dyay for ssh <3
22:24.14russellbAMUG: stfu
22:24.37Qwell%ast indication add us ring 247/500,494/328,0/78,370/94,370+247/31,247/94,0/375,185/500,220/500,220+247/16,247/484,0/16,294/484,0/16
22:24.38Qwellbetter :p
22:24.46*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)
22:25.02Qwell%ast indication show us
22:25.11Qwellsilly bot
22:25.13russellb%ast console dial 1000
22:25.44russellbdon't ask for the dialplan, it'll go for 15 minutes
22:25.47JunK-Yhes lagged, wait
22:25.49russellbone line at a time, slowly
22:26.06Qwellrussellb: yeah, I already did, heh
22:27.57garreelis there a way to show caller id even if it's hidden by default on a E1 PRI line?
22:28.12Qwellgarreel: hidden by who?
22:28.19garreelhidden by the telco
22:28.23Qwellthen no..
22:28.36garreel:-(
22:29.12Qwellwhy would they hide cid on a PRI?
22:29.29garreelbecouse we are a call center
22:29.53garreelwho work for the telco
22:30.04garreelwe make outbound
22:30.15garreelso it's hidden
22:30.40garreelI was thinking to try something such the *82 code
22:31.26garreelbut if I put a *XX code before the number I get an error
22:32.27QwellJunK-Y: Playtones(ring) \n Wait(30)
22:32.36QwellDo that, you'll like it :p
22:32.56QwellAnswer() first
22:38.56*** join/#asterisk `p4r14h (n=j0sh@69.92.145.178)
22:39.33mmartinnWill [general] / disallow = all in skinny.conf disable all?
22:39.45Qwellit should, in 1.4
22:39.51mmartinnin 1.2?
22:39.55Qwellno
22:40.02Qwelldon't use chan_skinny in 1.2
22:40.14mmartinnWe don't use skinny at all but there's a log message about it
22:40.25Qwelljust noload the module
22:41.03mmartinnokay :)
22:41.59*** join/#asterisk Jubei (n=Stormtro@147.27.46.26)
22:42.35Jubeicould somebody either explain to me or point me to a document on the web about the diferrences between TE and NT mode in ISDN BRI interfaces
22:43.54mmartinnQwell: There's no weird side effects of noload=chan_iax2, right?
22:43.55Qwellmmartinn: not if you don't use it, and it'd be chan_iax2.so
22:43.55mmartinnQwell: Right! Thanks :)
22:43.58garreelQwell[]: but if it's not possible to override the default hidden caller id state... what is *82 code meant of?
22:46.06gambolputtyjubei - look at cisco ccna documents
22:46.11gambolputtythey talk about isdn
22:47.19gambolputtyand then after that, watch Ninja Scroll
22:50.22*** join/#asterisk crudi (n=crudi@ppp-70-247-203-146.dsl.snantx.swbell.net)
22:52.29Jubeiguys mISDN speaks of a command line called "misdnportinfo". I don't have that in my system, why?:)
22:52.44Jubeiis it part of the mISDN package?
22:53.16crudiI'm having issues with AsteriskNow. I'm willing to paypal $$ if someone can help me get it working. I have a Triple span FXO digium card, GXP-2000 phones, and 2 POTS lines.
22:54.21Qwellcrudi: what issues?
22:55.02crudiwell lets start with if dial an internal extension, we can't hear each other.
22:55.23Qwellare the grandstreams on the same LAN?
22:55.28crudiyes
22:55.28Qwellsame lan as asterisk, I should say
22:55.32crudiyup
22:56.02*** join/#asterisk jdiskywlkr (n=kvirc@adsl-70-234-164-77.dsl.tul2ok.sbcglobal.net)
22:56.07QwellDo you get audio if you call something like, say, voicemail?
22:56.59crudilet me check
22:57.13jdiskywlkrIf Asterisk is used to proxy a mgcp to sip phone call, will Asterisk stay in between the two endpoints, or will the conversation be handed off to the endpoints using rtp?
22:58.36*** join/#asterisk saftsack (n=saftsack@pD9E07CF9.dip.t-dialin.net)
22:58.56saftsackis the via c3 corefusion board suitable for asterisk?
22:59.10saftsackpbx system w/o transcoding with < 20 users
22:59.17crudiyes i get audio when i call VM
22:59.30Qwellcrudi: So, it's only a problem between the grandstreams?
22:59.36crudiyup
22:59.44Qwelland did you verify that they both get audio from voicemail?
23:01.37crudiyes they both do
23:02.18crudinow i get that the other # is unavailable
23:02.36Qwellit probably isn't registered
23:02.48crudiI have registration turned off in the gui on the phone
23:02.58crudithats what i read to do
23:03.14Qwellyeah, that's so asterisk doesn't register to the phone..  the phone isn't registering to asterisk, and it needs to
23:03.48JunK-Yany way to register and set an outbound-proxy btw?
23:04.51Jubeianybody know anything about mISDN ? i've done everything as the documentation describes but then "misdnportinfo" says : "Found no card. Please be sure to load card drivers." isn't that what the mISDN start script is supposed to do?
23:04.52crudiyeah, i get
23:04.56crudiName/username              Host            Dyn Nat ACL Port     Status
23:04.56crudi02                         (Unspecified)    D          0        Unmonitored
23:04.56crudi01                         (Unspecified)    D          0        Unmonitored
23:04.56crudi2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
23:05.37*** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com)
23:17.42crudiany ideas on that qwell?
23:17.50tzafrir_laptopI'm trying to sign up for an account in iaxtel . Funny enough they have a large choices of countries to choose from
23:17.55tzafrir_laptop(USA alone)
23:18.29Qwellcrudi: You'll need to fix up your phones to register
23:18.36tzafrir_laptopJubei, misdn-init might help
23:18.42tzafrir_laptopit has some useful options
23:18.45crudiive turned them on to register. still not registering.
23:19.18Qwellcrudi: Did you setup the phones in the GUI?
23:19.25crudiyes
23:19.39tzafrir_laptopwhat phones?
23:19.42Qwellusing the Users tab?
23:19.45Qwelltzafrir_laptop: gxp's
23:19.46crudigxp-2000
23:19.52*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
23:20.02crudiUsers tab in the asterisknow gui, and the account 1 tab in the phone gui
23:20.13Qwellwhat account?
23:20.26crudithe extensions
23:20.27crudi01
23:20.28crudi02
23:20.36Qwellthose are added in the Users tab...
23:21.02crudiyes
23:21.05crudinow i have
23:21.06crudiName/username              Host            Dyn Nat ACL Port     Status
23:21.06crudi02/02                      192.168.1.202    D          5060     Unmonitored
23:21.06crudi01/01                      192.168.1.201    D          5060     Unmonitored
23:21.06crudi2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
23:21.14Qwellmuch better
23:21.28crudibut still no audio
23:22.14crudiit rings, we pick up, cant hear
23:22.20tzafrir_laptophmm... they also have a typo in the screen after the registration: "minuites" instead of "minutes"
23:22.32*** join/#asterisk |dennis| (n=dennis@shc.edu.bz)
23:22.35QwellDoes the asterisk console give any useful output, if you do a `core set verbose 50`?
23:22.44Qwell..and make the call again afterwards, of course
23:23.33garreelcrudi: try canreinvite=no
23:23.39crudiVerbosity was 3 and is now 50
23:23.39crudi<PROTECTED>
23:23.39crudi<PROTECTED>
23:23.39crudi<PROTECTED>
23:23.39crudi<PROTECTED>
23:23.39crudi<PROTECTED>
23:23.41crudi<PROTECTED>
23:23.43crudi<PROTECTED>
23:23.46crudi<PROTECTED>
23:23.46Qwellgarreel: they're on the same NAT
23:23.56Qwellcrudi: please use a pastebin next time
23:23.58Qwell~pb
23:24.00jboti heard pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
23:24.00crudisorry
23:24.01crudik
23:24.39crudii know i know, i will
23:25.09*** join/#asterisk MACscr (n=MACscr@adsl-75-23-66-89.dsl.peoril.sbcglobal.net)
23:25.58MACscrwhats the minimum about of ports i need open for rtp?
23:26.09QwellMACscr: 0, if you aren't using it
23:26.49*** join/#asterisk thevoke (n=michiel@missioncontrol.eng.gbxs.net)
23:27.18MACscrlol, true
23:27.33Jubeianybody running mISDN?
23:27.56crudihttp://www.pastebin.ca/418975
23:28.39Qwellare your phones set to use a stun or anything silly?
23:28.51Qwelland no firewall of any kind between them?
23:29.01crudinope to both
23:29.11crudistraight out of the box
23:29.15crudiboth on the same switch
23:29.23Qwellswitch or router?
23:29.27crudiswitch
23:29.34Qwellok
23:29.39crudirouter connected to switch of course
23:30.05Qwellgo into users.conf, and add a canreinvite=no for each of those phones..  you can't do it from the GUI afaik
23:31.47crudididnt work
23:31.53QwellDid you reload?
23:31.56crudiyes
23:32.10QwellDoes it still show the "Native bridging" line?
23:32.24crudino
23:32.26crudipacket2packet
23:32.59Qwellany chance I can get ssh access to the box?
23:33.38crudiits on a private lan
23:34.43QwellYou didn't edit sip.conf at all?
23:34.48crudino
23:35.00Qwellthere's gotta be a setting in the phones causing that
23:35.15QwellYou're gonna have to look through the menus and find something about stun, or nat, or something, and disable it
23:35.26crudistun is blank
23:35.33garreelcrudi: try iptraf to se if data is flowing trough the phones
23:35.34crudiso is nat ip
23:35.46crudiwhere at?
23:35.48Qwellcrudi: anything on rtp debug?
23:35.59crudirtp debug?
23:36.11crudijust turned that on
23:36.28Qwellmake a call, and see if you see the rtp packets
23:36.39crudii see a bunch
23:36.45Qwellpastebin about 20-30 lines of that
23:37.01Qwellstart from the beginning if possible
23:37.13crudihttp://www.pastebin.ca/418981
23:37.34Qwellno "Got RTP packet from "?
23:37.38mmartinnHey folks; I notice that when I use the UserEvent manager action, I see the event, but I get an "Action: UserEvent" header in the response -- I don't see it explicitly in the source; does this mean the UserEvent treated Action: as a custom header?
23:37.53crudinope
23:38.06Qwellthe len is weird
23:38.10Qwell160 vs 33
23:38.19Qwell160 would be ulaw I think..  not sure what 33 is
23:38.27Qwellthat's tiny..  gsm/g729 maybe
23:38.46crudiya we keep seeing GSM and PCMU
23:38.50crudion our phones
23:38.54Qwellfor both phones, try adding disallow=all and allow=ulaw
23:39.25Qwellnot sure why they would be negotiating different codecs...
23:39.46Qwellrussellb: re-ping
23:40.25crudithat worked
23:40.42Qwellcool
23:40.43crudibut i still dont see got rtp
23:40.58Qwellyeah, it may not show that in P2P bridging mode
23:41.23Qwelltry removing those canreinvite lines now - you shouldn't need it
23:42.24crudiya still works once i remove
23:42.32Qwellcool :)
23:42.36garreel:-)
23:42.37crudiHow do I get my two pots lines to work
23:42.45crudiive got the 4000p
23:42.46Qwellcrudi: tdm400p?
23:42.50crudi400
23:42.51crudiya
23:42.53crudi3 fxo
23:42.53Qwellthen on the setup wizard, it should've detected them
23:42.56crudiit did
23:43.25crudii get a 404 when i call out
23:43.35Qwellthe Incoming Calls and Calling Rules tabs
23:44.03MACscranyone have experience with e164.org? I cant get ti to register my hostname. It says thats its unable to verify the route. But I am able to receive sip calls on that box, so im not sure what its complaining about
23:44.18crudihttp://www.pastebin.ca/419003
23:44.42Qwelland are the phones using those dialing rules?
23:44.50crudiyes
23:44.51Qwell"Dial Plan"
23:45.14crudiDialPlan1
23:45.14crudiyeah
23:45.29*** join/#asterisk brussel (n=brussel@cpe-24-165-7-252.san.res.rr.com)
23:45.32Qwelltry unchecking the "Or More" checkbox for all of those
23:45.37Qwellit was buggy previously
23:46.41Qwelland you of course need to strip off 1 digit from those
23:46.51Qwellwell, except the 911 rule
23:48.51crudithat seemed to work. removing the or more. but only on the first line. on the 2nd one i get a "603"
23:49.04Qwellthe 911?
23:49.20crudino - Line2-Local
23:49.25crudiits my 2nd pots
23:49.31crudiI dial 8 then the number
23:50.12Qwellis the phones dialplan set to use 8?
23:50.20Qwellphones have their own dialplans too
23:50.24crudiits the same dialplan
23:50.34crudiit just says if you get an 8, use this port
23:50.44Qwellin the phone itself
23:51.36crudii have a dial plan prefix
23:51.37crudiblank
23:52.55crudiand it wont let me do a 9*73 to disable call forwarding
23:53.06Qwellbecause X doesn't match *
23:53.23QwellYou'll need another pattern for that
23:54.02*** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com)
23:54.43crudii think i got that one
23:55.19chrisknightIv'e been pulling my hair out for about a month now.  Searched the net all over...  Can anyone reset a cisco 7960 phone?  I have tried everything.
23:55.53crudiqwell: whenever i call, i get the ivr. but i didnt set one up. incoming rule is set to go straight to an extension
23:59.51garreeltrying MP3Player application... but it seems to play slooooooooooooooooooooooooowly the file... any ideas?

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