IRC log for #asterisk on 20071115

00:01.18*** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
00:02.32Strom_Mb1ch0: the PSTN uses G.711
00:08.19*** join/#asterisk BBHoss (n=hoss@146.229.191.76)
00:09.42b1ch0i know , so if * recive an internal call (from ip phone hat have enabled GSM) directed to FXO port, * transcode from GSM to G711 ... right ?
00:09.59BBHossyes
00:10.29b1ch0and that is why i have to pay licence for every transcoded call if i decide to use G729
00:10.38JTStrom_M: analogue uses analogue though, so probably SLIN
00:10.55JTb1ch0: your ip phones support sip and iax2?
00:12.17JTasdx: sip works fine behind nat when setup right most of the time
00:12.28b1ch0yes both protocols
00:12.48JTb1ch0: must be some crappy chinese designed phone ;)
00:14.45b1ch0yes they are, but chinese are not stupid ... they copy very well and often improve products
00:15.01*** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz)
00:15.36JTb1ch0: considering there are no decent IAX2 ip phones... what do they have to copy?
00:16.05*** join/#asterisk xsanchito (n=jorgito@150.138.broadband6.iol.cz)
00:16.06xsanchitohi
00:16.10xsanchitoone question
00:16.22xsanchitois asterisk able to act as skinny client to cisco call manager ?
00:16.24b1ch0knows cisco and huawey story ?
00:17.04b1ch0anyway, thanks for answering my question
00:17.05[TK]D-Fenderxsanchito, IIRC, no.  Same with MGCP.
00:17.29JTb1ch0: sorry that didn't make sense
00:17.42JTb1ch0: huawei makes second rate DSLAMs, yes, i know thi
00:17.44JTthis
00:17.56b1ch0just need to know what append when a call is transcoded .....
00:18.04JTISPs here are binning Huaweis like there's no tomorrow
00:18.12xsanchito[TK]D-Fender, ok, do you know what open source implements skinny as client ?
00:18.21JTand buying proper equipment like alcatel/lucent/ericsson
00:18.23[TK]D-Fenderxsanchito, never heard of any...
00:18.51xsanchito[TK]D-Fender, shame I need it ... anyhow thanks a lot
00:18.57JTb1ch0: and how can they copy if there's no decent IAX2 IP phone to copy off?
00:19.36b1ch0JT: shure that i will need help to enable NAT
00:19.42*** part/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net)
00:19.43JTxsanchito: why can't CCM speak SIP or H.323?
00:19.52JTb1ch0: what?
00:20.49xsanchitoJT, you must pay for SIP by default you have only trunks afaik
00:21.01b1ch0so if you are so certain that * works well with transversal nat .. i will need your help (i didnt make * with transversal nat before)
00:22.13JTxsanchito: well, weigh up the cost of paying for SIP, or writing an SCCP channel driver
00:22.24JTb1ch0: just read this.:
00:22.26JT~sipnat
00:22.27jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
00:22.30JTthe first url
00:28.55b1ch0JT: seem interesting .. i will try, i hope next week  .... it is the same problem i have ... thanks again
00:31.11JTb1ch0: which problem is that?
00:37.32*** join/#asterisk infernix (i=nix@unaffiliated/infernix)
00:38.06infernixhas anyone tried wifi sip phones with asterisk, or can anyone comment on how well (or not) wifi sip phones work in general?
00:38.26BBHossnot well
00:38.36atomicdwifi sip phones suck.
00:38.47MackesI have used alot of WiFi Phones with Asterisk
00:38.55MackesThey dont suck
00:38.56BBHossSIP DECT phones rock!
00:39.07*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b8b81fba558fc2f1)
00:39.11BBHosscompared to what?
00:39.23atomicdWhich WiFi SIP phones have you used that doen't suck?
00:39.56atomicdmackes: make / model ??
00:40.01MackesThe Hitachi is very good
00:40.01BBHossall i have used have slow interfaces, shitty range, low talk time (high battery usage), etc
00:40.12MackesThe Zultys WIP2 is great
00:40.26MackesThe Linksys WiFi300 is ok
00:40.45MackesIts your WiFi network that makes the difference
00:40.51MackesNot so much the phone
00:40.57atomicdhaha...  the Linksys WIP300 sucks donkeys...
00:41.03Mackesok
00:41.08BBHosswifi is not suited for voice IMHO
00:41.25MackesWell.. I guess you have it worked out.
00:41.30BBHosstoo much packet overhead
00:41.40BBHossjitter, latency etc
00:41.50MackesI promise that the Zultys WIP2 is great $300
00:41.51infernixthere's dect phones with ethernet base stations, but thats not wifi
00:41.53BBHossno to mention it can be insecure
00:42.09BBHossinfernix: yes thats what im talking about
00:42.17BBHossthey are loads better than wifi
00:42.24BBHossmuch better battery life too
00:42.37atomicdI've got two WIP300s on eBay right now.  (Sold another earlier this week.)  That will be the end of my WIP300 experience.
00:42.44MackesWell that is just a normal Radio Phone with an ATA
00:43.03MackesCheck out the Hitachi
00:43.35Mackeshttp://www.zultys.com/index.jsp?tab=productdetail&product=wip2&detail=datasheet-wip2&type=phones
00:43.36infernixwell if dect phones with a sip base station are cheaper, that'll do
00:43.51BBHossthey arent really cheaper, but they are better
00:44.08MackesIf thats the case, pickup a Grandstream ATA, and a phone from Wallmart
00:44.23MackesYou can get it done for less then $75
00:44.27MackesSame Result
00:44.58BBHossgive this a look: http://www.2gac.net/training.ppt
00:45.14BBHossthe training guide for Aastra's SIP-DECT solution
00:45.21BBHossit will fill you in
00:45.23stubertCan someone tell me what causes asterisk to send a 491?
00:45.38atomicdAnyone ever try the Polycom/Kirk 600v3 with the Kirk DECT handsets?  Saw it at Astricon...looks cool.
00:46.42BBHossi have never used them, because when i was looking at wireless stuff, they were not avaliavle in the us
00:46.56infernixokay, so ideally i'd need 4 dect phones which are each uniquely addressable
00:46.59BBHossi have heard from guys at digium that they are a PITA to setup
00:47.07infernixare there base stations that can handle 4 or more concurrent SIP connections?
00:47.13infernixor should I just get 4 base stations
00:47.24BBHossyes SIP-DECT can handle 8 per base station
00:47.31BBHossfrom aastra
00:47.39infernixis aastra affordable?
00:47.41BBHosstake a look at that link
00:47.46infernixcompared to ATA + normal dect
00:47.46BBHosswhat is affordable to you
00:48.04infernix$400 total was what i'm thinking :)
00:48.06BBHossif you want a kludgy solution like that, go for it
00:48.17BBHoss~cheap
00:48.18jboti guess cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
00:48.53BBHossa single indoor RFP will cost you probably $600, and the phones are about $250 a piece
00:49.44BBHossis this for a business or for home?
00:50.36infernixstartup business
00:50.43infernixso i just cant shell out $1k
00:50.46BBHossyours or someone elses?
00:51.03infernixcomplicated, but lets say ours
00:51.06BBHossok
00:51.22JTMackes: NEVER use grandstream.... what the hell, don't recommend grandstrem
00:51.36JTMackes: and it is NOT the same result
00:51.37BBHossdo the phones HAVE to be function everywhere, or just in the general area around people offices
00:51.41JTDECT > Wifi
00:51.55JTusing wifi for mobile voip is a bad idea
00:52.05infernixthe office is like 25 by 5. but it contains highly isolated cabins
00:52.12infernixfull metal cages
00:52.18BBHossheh
00:52.23BBHossforget wireless then
00:52.27BBHossof any kind
00:52.34infernixyeah you got a point there
00:52.48BBHossif wireless is needed inside the offices
00:52.57BBHossget everyone a 480i CT
00:53.32infernixi think i'll try a regular dect in the soundproof cabins first
00:53.47infernixif fail then ethernet. makes stuff that much more simple, too
00:53.52BBHossjust go to best buy and get a dect phone
00:54.03BBHosssee what range you get with it outside the offices
00:54.46*** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)
00:55.08BBHossand why do you have full metal cages?
00:55.31BBHossparanoid of NSA?
00:55.55infernixsound
00:56.02infernixwell its not full metal, but a hell of a lot
00:56.05BBHossMETAL?
00:56.08infernixsound isolated cabins
00:56.22infernixmetal panels with sound insulators, outside is full metal
00:56.32infernixinside is wood, inbetween a huge insulator layer
00:56.38BBHossyeah i doubt much signal could get through there
00:56.50infernixi should check my cellphone actually :p
00:57.14BBHossyeah see how much the signal drops
00:57.39*** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt)
00:59.21JTjust wire them
00:59.31JTmuch more features on wired phones
01:04.44*** join/#asterisk relic-se (n=root@82.96.61.130)
01:05.57*** join/#asterisk fnordus (n=dnall@24.84.160.227)
01:06.45relic-seHi, can someone help me with a config issue I am having, the documentation has me going around in circles.
01:10.18MackesGrandstream is not awful, just a pain.
01:10.33MackesWiFi phones work well, if your AP is good
01:10.53Mackesif you would like to roam between AP's then your AP's need to support
01:10.54Mackesit
01:11.05MackesLinksys makes the best ATA
01:11.12JTgrandstream phones are awful
01:11.16*** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net)
01:11.19MrTelephoneqwell
01:11.21MackesThey are cheap
01:11.27Mackesyes
01:11.31JTwifi phones if you are close to an ap, or you have tonnes of aps
01:11.39JTi don't care how much they cost
01:11.41JTthey are awful
01:11.44Mackesok
01:11.53MrTelephonewho wants to make 30 bucks
01:11.54MackesWhat do you like Mr JT
01:12.08JTand a Polycom IP320 is very similar cost to a Grandstream GXP2000, and a million times better
01:12.50MackesYes, It is a great phone. No doubt, and I am not recomending anyone go by a Grandstream phone
01:12.54relic-seis there any reason why the "s" extension would get a match when no other extensions match?
01:12.59MackesHowever, they do have a place
01:13.03relic-sewould not get a match even
01:13.47JTMackes: in the bin
01:13.59MackesMan, ok
01:14.04MrTelephonerelic-se, use i for invalid extension
01:14.43MackesI like Aastra 480i and 57 iPhones we have hunderds, I also love my Snom 370
01:14.55JTMackes: the really is no point supporting cheap shit when the market leader makes products that are very similar in price
01:15.11MackesCisco 7960's are the best for a Simple rock hard phone
01:15.16JTi haven't seen the 370, but most snoms were unimpressive
01:15.19JTlol cisco
01:15.33MackesSo Polycom or nothing huh?
01:15.45JTaastra is fine
01:16.06[hC]i understand snom underwent significant changes recently
01:16.28[hC]specifically in the physical design
01:16.34MackesSnom 370do GSM, and true OpenVPN
01:16.58relic-seMrTelephone: that does not seem to help: Extension '800' in context 'incoming' from '' does not exist.  Rejecting call on channel 0/1, span 1
01:17.09MackesAnd Cisco are rock, Rock, Rock solid... Set them up once, and never worry again
01:17.33MackesMy polycom reboots when ever it hits the smallest error with SIP
01:18.43MackesIf I call a Linksys ATA FXO, some times my Polycom reboots
01:18.50JTit must have bad firmware
01:19.00MackesIt locks first and then reboots
01:19.13JTcisco are also cunts :)
01:19.18JTand the audio quality isn't as good
01:19.36MackesReally? You really dont like Cisco?
01:19.46JTmost people don't
01:19.57JTthey have terrible business policies
01:20.07JTridiculous licensing
01:20.46MackesThey released SIP 8.1 for Cisco 7960's to the public a few weeks back
01:21.01MackesI dont love Cisco as a company
01:21.14Mackes<PROTECTED>
01:21.17*** join/#asterisk fnordus (n=dnall@24.84.160.227)
01:21.31MackesHowever, no one has ever been fired for buying Cisco
01:22.13MackesSo, Asterisk PBX, and Cisco SIP Phones.... Its going to be rock solid, and none techincal users love the Cisco phones
01:22.27Mackesbecause they see then in Sit Coms, 24, and the west wing
01:22.59MackesI think we have all wanted a decent used Cisco switch if we could get our hands on one, right?
01:23.42MackesI have 4 Cisco 7960's at home as home phones.. they have never locked... so I am not worried that my wife will call 911 some day and have the phone lock
01:24.37MackesSame with my business users, Phones just HAVE to work. This is not to say that Polycoms dont. They are great. But, Polycom is a BIG company as well.
01:26.14*** join/#asterisk craigk (n=ckowald@58.174.122.198)
01:26.59JTMackes: no, i avoid cisco switches when i can
01:28.30delmarlove my Polycom phone.
01:28.35delmarwish I had more
01:28.50Nuggetcisco phones are a royal pain in the ass.
01:28.54*** join/#asterisk Cooner750 (n=Cooner75@cpe-71-72-211-147.cinci.res.rr.com)
01:28.56Cooner750hello.
01:29.48Cooner750just got Asterisk up and running with Openfire in about an hour, appears to be working rather well so far.
01:30.06Cooner750I have a question though, it's not possible for users to make outbound calls in any way, is it?
01:30.27NuggetNo clue.  I've never even heard of openfire.
01:30.47Cooner750Oh, Openfire is the Jabber server by Jive Software, it dosen't allow the user to dial a number
01:30.51Mackesok.. well. like I said Polycom are good. Not Cool, nor fun.. But they do the job
01:31.19Cooner750I'm using Asterisk just for VoIP, no hardware involved. Does anyone else use it like this?
01:31.38Nuggetsure, many people do.
01:32.02Nuggetyou can subscribe to a voip termination service that will let you interface with the public phone network if that's what you're wanting to do
01:32.22Nuggethttp://connect.voicepulse.com/ is one, http://asterlink.com/ is another one.  there are about eleventy-million of them.
01:32.32Cooner750Oh, nah, not too interested in that. The Openfire server is private anyway, used by friends and me as a chat network
01:32.45Cooner750Interfacing with the public phone network isn't free, is it?
01:32.47NuggetI wasn't sure if that's what you meant when you said "dial a number"
01:33.58MrTelephonecan someone help me with some c code?
01:34.06JTMackes: cisco phones are well and truly overrated
01:34.16Cooner750I'll probably be using VoIP on Asterisk just to communicate with friends that can use Spark (Jive Software's jabber client, supports the Asterisk-IM plugin)
01:34.45Cooner750I'm running Trixbox with Java installed (for Openfire) on a 1GHz PIII with 256MB RAM, it's a little... cramped :P
01:34.46rob0Cooner750: Not free for outbound, but very cheap. And there are free inbound providers.
01:35.27Cooner750I probably won't ever have a need to interface with the public network, but if I do I know I can now
01:35.31rob0I pay <US$0.02/minute
01:35.40*** join/#asterisk fnordus (n=dnall@24.84.160.227)
01:35.52JTMackes: their screens flicker horribly
01:36.04MackesJT, You are very sure of yourself.
01:36.10MackesThey do
01:36.12Mackes?
01:36.24JTof course i am. i have a Cisco 7905 and 7940 in front of me
01:36.38MackesI have about 20, that are WELL used (we had a call manager) and they are all fine
01:36.50JTand a linksys, a snom, and polycom
01:36.55MackesFlicker?
01:37.06JTyes i can see a flicker on the lcd
01:37.13Mackesok then
01:37.20rob0JT, I bet you rarely talk on those. :)
01:37.24JTmust be a bad refresh rate
01:37.29MackesDid someone from Cisco hurt you once?
01:37.37JTMackes: what the fuck?
01:37.51JTMackes: you can't take the fact that cisco aren't all they're cracked up to be?
01:38.10MackesWell, i just dont agree.
01:38.10JTrob0: long story, we have a few phone systems here, so i do use them a bit
01:38.37MackesI have one of quite a few phones, and the Ciscos are consistant
01:38.55JTconsistantly not as good as some competition
01:39.05JTof course they're nowhere near as bad as grandstream
01:39.09MackesAastras 480i nice, but the firmware took a while before it was stable
01:39.56JTand because i'm not a drooling cisco wanker, someone from cisco must've hurt me? :o
01:40.50MackesOk then.
01:41.21MackesI think we have spoken before JT, however, are you in the US?
01:43.02MrTelephonei have to rewrite *__get_header() to take the last Authorization header :(
01:43.57JTMackes: no
01:44.22MackesAround were I live and work, we are swimming in Cisco..... Cisco Switches, Routers, Phones, Call Managers, everyone runs Cisco. Every consultant sells and supports Cisco, Everyone runs PIX firewalls, Every shop is mostly Cisco and Microsoft
01:44.36MackesNow I dont beleave those are the best products
01:44.48Mackeshowever they are very good
01:44.56Mackesand we are awash in them
01:45.25MackesI have so many extra Cisco switches and Routers I am tripping over them in my office
01:45.31*** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net)
01:45.41Mackesthey are just the standard in the US
01:45.50MackesThat doesnt make them the best
01:45.55Mackeshowever they are good
01:45.58Mackesand availible
01:46.33*** part/#asterisk stubert (n=stu@techtools.actusa.net)
01:46.39MackesThat said, We purchaced a very extensive Alcatel OmniPBX system, and Alcatel Siwtches
01:46.44MackesI hate them
01:46.55Mackesso hard to find support and documentation
01:47.10JTlots of places have Cisco
01:47.14*** join/#asterisk chendy (n=chendy@121.76.132.123)
01:47.15JTi fail to see your point
01:48.11MackesWell, why did so many buy Cisco if it sucks?
01:48.25katsuodohow many extensions can be set on a tdm805 card with dsl?
01:48.32JTbecause popularity is absolutely no indicator or quality
01:48.39JTworst argument ever
01:49.03Mackesall right then
01:49.28MackesI understand you point of view
01:50.04MrTelephonedon't be subdued by the nonsense of popularity doesn't mean quality
01:50.09MackesWhat sold you on Polycom? What happened that made you such a dedicated fan? I like them.. but you are in love
01:50.25MrTelephonethere was a cute girl at polycom who helped him setup a phone
01:50.31JTi'm not in love, they are just the best SIP phones available
01:50.38JTask most people here
01:50.41MrTelephonei agree that polycom are the best phones
01:50.43JTit's common knowledge
01:50.54katsuodoI too agree
01:51.09MrTelephonepolycom should beef up their style a little bit
01:51.19MrTelephonepolycom phones have nontilting stands
01:51.27MrTelephonehuge mistake by the design staff
01:51.36MrTelephonewhy no backlights?
01:51.42MrTelephonewhats wrong with these companies
01:51.51MrTelephonei have a backlight on my 1980 cordless
01:52.13JTheh, most offices have lights in them
01:52.20JTmost desk phones don't have backlights
01:52.26JTbut the IP650s do i believe
01:52.35katsuodoneed info to make right decision, how many extensions can be set on tdm805 card?
01:52.40MrTelephonei work with the lights off
01:53.11JTkatsuodo: set?
01:53.14Mackeshahahhhh.. ok
01:53.22MrTelephonepolycom have tonnes of features.. provisioning features, failover features, within the phone itself
01:53.34MackesThey all do
01:53.37katsuodoJT halo yes
01:53.37MrTelephonepolycoms are practically their own sip proxies
01:53.47MrTelephonebut I misread or the documentation about vlans is wrong
01:53.49JTkatsuodo: i don't know what you mean?
01:54.03katsuodookay let me explain
01:54.06MrTelephoneI set vlans on my phones and it sets the traffic from the PHONE to the vlan, not from the computer behind it
01:54.09JTMackes: you know cisco ip conference bridges are rebadged polycoms?
01:54.33MrTelephoneJT, how can you prove that, as if cisco would admit it?
01:54.34JTMackes: and cisco ip phones with SIP firmware use intellectual property licensed off Polycom?
01:54.44MrTelephonecisco should stick to making routers and drop everything else
01:54.52JTMrTelephone: it's BLATANTLY obvious for the conference room bridges
01:54.57JTthey look the same
01:55.00MrTelephonehaha
01:55.02JTbut have a cisco sticker
01:55.17MrTelephonepolycom uses koss electronics
01:55.19MrTelephonehahah
01:55.45MrTelephonedid you guys see that equipment floating around with the mrtelephone sticker on it? its actually linksys
01:55.50katsuodoin office an asterisk server with a tdm805 fxo card only with 8 pots.  Only 5 pots will be used.  office manager wants 12 extensions from the card
01:56.07MrTelephonehaha i cracked myself up
01:56.11MrTelephonenoone else is laughing though
01:56.13Mackesgreat. So, If it has a Cisco Sticker it will sell better then if its branded Polycom?
01:56.16MackesInteresting
01:56.42MackesWhy would Polycom what to soil there units with a Cisco tag
01:56.43JTMackes: the sip firmware in cisco ip phones isn't as good as that on polycom ip phones though
01:57.03JTwhy would polycom say no to millions of dollars in revennue?
01:57.05*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca)
01:57.10MackesI mean if Polycom is better, they shouldnt need to resell to Cisco?
01:57.17MrTelephonepeople grow trust with brand names.. some people are ALL cisco or ALL polycom.. but i doubt you see a mix of equipment.. if you do you know its some retard running asteirsk
01:57.19JTthey're not reselling
01:57.23JTthey're OEMing
01:57.44MackesJust so we are clearm they will sell more "units" if it says Cisco then Polycom?
01:58.22katsuodoJT understand you
01:58.25MackesWhy not leave it named Polycom, and sell Polycom..... Why boost Cisco's name?
01:58.26JTcisco can either make their own conference bridges
01:58.32JTand they won't be as good
01:58.38JTas polycom have decades of experience
01:58.43JTor they can oem from polycom
01:58.50Mackesyes yes..
01:59.16MackesSell it as a Cisco and they will sell more
01:59.23Mackeswhy would that be?
01:59.39rob0MrTelephone: We're laughing at you, not with you. ;)
01:59.57JTMackes: umm, cisco has customers already, ergo, sales
01:59.58MrTelephone:P
02:00.27Mackesok guys,
02:00.33MrTelephonepolcyoms are so diverse.. web access
02:00.41MrTelephonei love the web access
02:00.53*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-0fbba998d28ce251)
02:01.22MrTelephonesomeonw wanna write some code for me for 50 bucks :-/
02:01.25MrTelephone20 minutes?
02:01.51JTMrTelephone: what sort of code?
02:02.20*** join/#asterisk hyphenex (n=scott@60-241-65-249.tpgi.com.au)
02:02.24MrTelephonei need to rewrite get_header() in chan_sip so it looks for Authorization header but it looks for doubles and picks the last one if there is double
02:02.25hyphenexWhat's a good VoIP phone to buy?
02:02.44MackesThere is only one.
02:02.47MackesPolycom
02:02.51MackesThats it
02:03.00MackesEverything else is subpar
02:03.05JTaastra are pretty good too
02:03.15MrTelephoneim not too good at c so i thought i'd just pay someone to do it
02:03.15JTand linksys in certain countries
02:03.24hyphenexI didn't need it to do anything special, just SIP VoIP calls :)
02:03.26Mackessee... how can you say that.... they lock up like crazy!
02:03.35JTMackes: what?
02:03.42MackesAnd do you know who owns Linksys?
02:03.47JTyes
02:03.50JTwhy does it matter?
02:04.00[TK]D-Fenderhyphenex, in AU the best dollar value is likely to be the linksys SPA series
02:04.06MrTelephonejt do you code?
02:04.18JTMrTelephone: only a little bit, i fumble my way around C
02:04.24MrTelephonesame here
02:04.25hyphenexThanks [TK]D-Fender
02:04.47MrTelephoneasterisk is really ahrdcore.. so many references to other stuff
02:04.50JT[TK]D-Fender: nah polycom are a reasonable buy in AU too
02:04.54JTfor business use
02:05.00[TK]D-Fenderhyphenex, Polycom is IMO the best out there, but I know it comes at a steep premium for you on import.
02:05.16MrTelephonepolycom phones work better behind nat as well
02:05.18MackesSo if (and when) Cisco buys Polycom, will Polycom still be a good product?
02:05.18JT[TK]D-Fender: the main problem is lack of IP320/330 availability
02:05.20[TK]D-FenderJT : Really?  got a good retailer you could refer me to so I can pass that on?
02:05.26JTMackes: sure
02:05.30JT[TK]D-Fender: Westan
02:05.37[TK]D-FenderJT, links please :)
02:05.43hyphenexHmm, it might cost a bit to import then
02:05.48MackesI just bought 3 Voip Polycom Conference room phones.
02:05.55MrTelephonenice
02:05.59MrTelephonei got one of those ip4000
02:06.01[TK]D-Fenderhyphenex, I meant local distribution...
02:06.02MrTelephonebig bucks though
02:06.06MackesThey have been a bitch out of the box to set up
02:06.11rob0grandsuck
02:06.15Mackesand they dont support POE!!!
02:06.22MrTelephoneonce you get your system working its easy
02:06.28MrTelephonemackes, which ones did you buy
02:06.30MackesThey came with thier own injector?
02:06.47MrTelephoneit came with an injector
02:06.55MackesThe 4 prong star... I think they only have one
02:07.11Mackesyeah... but I dont need one, we are all POE
02:07.28[TK]D-FenderMrTelephone, I picked the SoundStation 2W (Wireless analog) + ATA for my company (got it BEFORE * knowing I'd use it with an ATA later).  Works great and the wireless rocks
02:07.30Mackesbut the phone only supports its own injector
02:07.50Mackeswhy would they do that?
02:07.55JT[TK]D-Fender: Westan ... www.whitepages.com.au
02:07.57MrTelephonenice, i don't evne know what a 2w looks like
02:08.03JTi'm sure they're googleable too
02:08.12JT[TK]D-Fender: Westan is the NSW distributor
02:08.43asdxcan i use asterisk with vonage?
02:08.50MrTelephone~googleable
02:08.55Mackeswith an ATA only
02:09.08JTMackes: it was probably designed prior to the ratification of 802.3af
02:09.36MackesIts the only conference room VOIP they offer, and its brand new
02:09.39MrTelephonecan you get a good shock from putting a poe cable in your mouth?
02:10.06*** part/#asterisk hyphenex (n=scott@60-241-65-249.tpgi.com.au)
02:10.09JTMackes: i know, what i said still applies.
02:10.21JTMrTelephone: probably not if it complies with 802.3af
02:10.43JTMrTelephone: the device must signal the switch that it is capable of taking power before power is provided
02:10.51MrTelephonenext CSI features a victim who died because someone hooked him up to a non-fcc regulated poe switch
02:11.00`Seanlol
02:11.12Mackeshttp://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation_ip4000.html
02:11.22[TK]D-FenderJT : no website?  Can end users buy direct?
02:11.46[TK]D-FenderMrTelephone, 2W looks just like every other boomerang phone they've ever made :)
02:11.51[TK]D-FenderMrTelephone, its just analog :)
02:12.05MrTelephone:P
02:12.15MrTelephoneone of the models didn't support sip either
02:12.17MrTelephoneu need an ip4000
02:12.28MrTelephonei think
02:12.38Mackeshttp://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation_ip4000.html
02:12.44*** join/#asterisk asdx (n=diego@adsl-130-56.click.com.py)
02:12.47MrTelephonefender, and you said thats wireless?
02:12.58[TK]D-Fenderto tell you the truth a lot of my users just love the IP600 speakerphone.
02:13.02[TK]D-FenderMrTelephone, yup
02:13.02JT[TK]D-Fender: minimum buy is 1 unit
02:13.12MrTelephonethe problem i had is i had to have 2 ftp accounts for the downloads because the ip4000 kept crashing with the sip image that i was using on the polycom 501's
02:13.12[TK]D-FenderJT : is that a yes?
02:13.31MrTelephonebut sip image 2.2 or something works on both phones good
02:13.35JT[TK]D-Fender: yes
02:13.39*** join/#asterisk marcan (i=1337@host214-205.cvd.fit.edu)
02:13.54MrTelephonewhats wrong with the 500 phone fender?
02:14.11[TK]D-FenderMrTelephone, Which 500?
02:14.17MrTelephone500 series
02:14.18MrTelephone:P
02:14.37[TK]D-FenderMrTelephone, Never said there was anything "wrong" with it.
02:14.48MrTelephonemr high class 600 :P
02:15.06*** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au)
02:15.29Nivexgot my PAP2T-NA from telephonydepot today.  plugged in and working great. :)
02:15.59*** join/#asterisk TJNII (n=TJNII@209.234.89.237)
02:16.03[TK]D-FenderMrTelephone, Ah.  This was for my company, and since I wanted PoE I calculated the premium of adding the PoE cable to IP 501's and then factored the fact on the 600 had the MicroBrowser at that time, a higher res screen, and more line-keys with lit indicators, and it was an easy choice :)
02:17.00MrTelephonefor browsing the web?
02:17.57*** join/#asterisk fnordus (n=dnall@24.84.160.227)
02:21.16[TK]D-FenderMrTelephone, Nope, I use the MB for queue stats for our call center, custom logo, etc.
02:21.27[TK]D-FenderMrTelephone, corporate directory, and so on.
02:23.36katsuodoHello TJNII
02:24.31katsuodoI have question?
02:24.42BBHossshoot
02:25.38asdxhow can i configure asterisk with some voip provider?
02:25.38Nuggetheh
02:25.58asdxi've just got a voip provider acocunt
02:25.59asdxaccount*
02:26.02katsuodo<PROTECTED>
02:26.50BBHossjust use IP
02:26.51katsuodothis is odd request, no?
02:27.03BBHossyeah, actually its more stupidity
02:27.16BBHossyou can't get 12 ports from an 8 port card
02:27.24katsuodocorrect
02:27.35*** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com)
02:27.47katsuodothey want use dsl
02:27.49BBHossyou could get two FXS modules and then have each of the two ports ring 6 extensions
02:28.09shmaltzOT:silly question here, how do I create an env var that has more than one value like 2 paths?
02:28.10BBHossok that doesnt matter if youre only using IP internally
02:28.26katsuodono effect bandwidth?
02:28.27BBHossyou can still use your POTS
02:28.30*** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-231.shs.nyc.wayport.net)
02:28.39BBHossnope, as long as you are still using POTS
02:28.49BBHosswell, external bandwdith
02:29.05Nuggetwell you can certainly have 12 ip extensions and only 8 pots lines, you just won't be able to support more than 8 inbound/outbound calls at the same time
02:29.08Nuggetthat's perfectly sane
02:29.09BBHossyour LAN bandwdith will be affected minimally
02:29.12*** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net)
02:29.47katsuodoNugget this must be static ip and then divide?
02:29.52BBHossi think he wants 5 POTS lines with 12 extensions? using the two spare ports as FXS
02:29.53Nuggetpardon?
02:30.19BBHossyou're not understanding us
02:30.34katsuodopardone
02:30.41Nuggetdesk phones talking to the asterisk server is one thing.
02:30.44BBHossyou dont need to worry about staic ips or anything, the box doesnt even need an internet connection, as long as its connected to your LAN
02:30.54Nuggetthe asterisk server talking to the public phone network is a second thing.
02:30.58katsuodocorrect
02:31.00Nuggetthey are almost completely unrelated to each other
02:31.35Nuggetif your desk phones are "ip phones" then they will not involve the tdm805 at all
02:31.46Nuggetthey will "talk" to the asterisk server over ethernet
02:31.58Nuggetthe tdm805 is for connecting to the public phone network
02:33.54*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
02:34.28katsuodothere is mix of ip phones and rca 2543 4 line intercom speaker phone
02:34.57asdxif i want to use asterisk + voip provider, and i want to use iax2, does the voip provider has to support this protocol?
02:35.08BBHossasdx:yes
02:35.30katsuodocorrect internet no fit in this situation it is pots
02:35.32asdxi wonder if vonage does support it
02:35.38BBHossno
02:35.41BBHossvonage sucks
02:35.50asdxBBHoss: which one is good?
02:35.55BBHossyou need an ITSP that sells SIP or IAX2 trunks
02:35.58asdxteliax maybe?
02:36.09BBHosswhat do you want internal/outbound or both?
02:36.16asdxBBHoss: both
02:36.22BBHossare you a home user
02:36.28asdxBBHoss: yes
02:36.34BBHosshmm
02:37.00BBHossdo you want to pay a flat fee per month (unlimited) or pay a cheap per minute rate?
02:37.17asdxBBHoss: flat rate is ok
02:37.28asdxs/rate/fee
02:37.35katsuodoBBHoss there will be 4 rca 2543 4 line intercom speaker phones
02:37.36asdxBBHoss: is not for me, is for a possible customer
02:37.45BBHosswhat country?
02:37.50asdxBBHoss: paraguay
02:37.56BBHosshmm, i was afraid of that
02:38.34BBHossdo they want to do a lot of international calling?
02:38.40asdxBBHoss: yes
02:38.45BBHossto what countires
02:39.17asdxBBHoss: usa mostly, i think
02:39.33BBHossdo they want a USA telephone number or a paraguay one?
02:39.36asdxBBHoss: i'm telling the customer to drop vonage
02:40.05BBHosswell if youre using asterisk, vonage doesn't work i dont think
02:40.14asdxBBHoss: they have the computer running asterisk in usa, and they want to connect from here (paraguay) and make calls to usa
02:40.24BBHossOHH i see
02:40.30BBHossok
02:40.36BBHossso essentially you want a US service
02:40.37asdxthey want to connect from here to the pbx*
02:40.44asdxyeh
02:40.52BBHossok kool, that makes this easier
02:41.15asdxBBHoss: we want IAX because the morons blocked SIP here
02:41.19BBHossohh
02:41.24BBHossso thats a requirement
02:41.29BBHosscheck out VoipStreet
02:41.34BBHossi use them as backup
02:41.39asdxBBHoss: i'm sure i can use openssh with sip/asterisk but i want IAX for now
02:41.39BBHossthey are alright
02:41.44asdxok
02:41.46BBHossthey do IAX2
02:41.49BBHossand SIP
02:41.54asdxnice
02:41.58asdxthx :)
02:42.09BBHossIAX2 is minority
02:42.23BBHossothers that do it are VoicePulse Connect (I hate them)
02:42.27BBHossand a few others
02:43.06asdxwhat about teliax
02:43.31BBHossnever used them, but i hear they can be flakey at times
02:44.01BBHossim sure for what you want to do they will work
02:44.25BBHossi set up phone systems for businesses, which CANNOT be flaky or go down EVER
02:44.34BBHosswhich i why i use services that provide SLAs
02:45.00rob0Tunneling UDP (SIP, IAX2) over ssh is possible through a kludge using nc(1). But it's ugly and generally a bad idea. OpenVPN is much better, provides a complete IP connection to the remote host.
02:45.03BBHossthats odd that SIP is blocked
02:45.09cryptnixanyone here having issues hitting mikrotik.com?
02:45.33rob0BBHoss: Not odd if the ISP is a telco. :)
02:45.46cryptnixhttp://www.mikrotik.com ... (for those of you lazy and don't wanna type) :D
02:45.51BBHossi guess i dont know how well i have it with AT&T :)
02:46.04BBHosscryptnix: not coming up for me
02:46.26cryptnixthank you
02:47.33asdxi tell my customer to switch the voip provider
02:47.43asdxhe is saying "ok i will consider that"
02:47.45BBHosswell you need to test them first
02:47.49asdx"we are looking at it"
02:47.56BBHossit is loads cheaper than vonage
02:48.11BBHossbut my question is, which server is blocking SIP?
02:48.22BBHossand how is vonage working if they block sip?
02:48.29MackesBBHoss, Use Cisco Phones.... You will be very happy with the performace
02:48.33asdxBBHoss: the telco is blocking SIP
02:48.40BBHosstelco where?
02:48.45asdxBBHoss: paraguay
02:48.52BBHossbut the server is in the US
02:48.57BBHosscorrect?
02:49.02asdxBBHoss: yes but my isp goes through the telco
02:49.09asdxBBHoss: it's a evil monopoly
02:49.10BBHossright
02:49.25BBHossbut if the server is in the US, the servers ISP won't block SIP
02:49.48BBHossit will need to be like this
02:50.05asdxBBHoss: but the SIP packets are passed through the telco and they block it...
02:50.08rob0A simple peer-to-peer openvpn between the USA colo and Paraguay, you have your SIP.
02:50.15rob0or IAX2
02:50.29asdxrob0: yeah
02:50.40BBHossParaguay Office/Home<-->IAX Connection<-->USA Server<-->SIP or IAX2 Connection to ITSP<-->ITSP
02:51.05asdxrob0: what about the router in my lan, that's doing NAT, can SIP<->NAT cohexist?
02:51.09BBHossnote, the home/office will need to have an asterisk server installed
02:51.21rob0~sipnat
02:51.22jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
02:51.27asdxok, thx
02:51.27BBHossfor the IAX2 connection to USA
02:51.37rob0If you're using an openvpn, you would not need NAT.
02:52.24BBHossOR, you could just change the port that asterisk runs sip on, then tell your phones to use that port
02:52.41*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net)
02:52.42rob0oh yeah, since you control both ends.
02:52.44BBHossbecause the telco probably isnt doing level7 blocking
02:54.02BBHossif they were that would take some beefy routers
02:54.16BBHossfilters w/e
02:58.41asdxmy customer is asking me "how can i configure asterisk"
02:58.42asdxwtf
02:59.12[TK]D-Fenderasdx, Yeah... you should be asking THEM for help ;)
02:59.34Qwells/help/money/
02:59.44asdx[TK]D-Fender: he is a ~60 year old guy
02:59.55Qwells/money/taffy/
03:00.33katsuodowith age comes much knowledge and wisdom no?
03:00.39Qwelland taffy
03:00.46*** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
03:01.08rob0mmmm taffy
03:02.34asdx~book
03:02.35jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
03:02.41asdxi will give him the link of the book
03:03.05*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)
03:05.09BBHossdamn network is taking a dump
03:07.18asdxdo you know if teliax supports IAX?
03:07.21katsuodoBook only for asterisk 1.2 new book come soon via tech support digium
03:07.21asdxIAX2*
03:07.27BBHossyeah they do
03:07.36rob0telIAX might support IAX :)
03:08.26asdxi seen a asterisk video and they recommended that
03:08.36[TK]D-Fenderkatsuodo, pay attention.  Thats the SECOND EDITION and is for 1.4 <---------
03:08.57[TK]D-Fenderkatsuodo, And that is NOT written by Digium.
03:09.13asdxJohn Todd
03:09.16*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
03:10.58*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)
03:12.41*** join/#asterisk voipomatic (n=IceChat7@rrcs-70-63-204-32.midsouth.biz.rr.com)
03:13.20katsuodo[TK]D-Fender understood
03:15.36voipomaticHI All, I am running 1.2.18 and am having trouble transferring calls to remote extensions.  If we press transfer and dial an extension which is local to our * it will transfer just fine.  But if we try to transfer to an extension that dials out over our VOIP carrier the call will not transfer.  We can dial the remote extensions from the local phones but cannot transfer a call to them.  Any ideas?
03:17.13*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
03:19.09De_Monasdx are you serious? you want to know if telIAX supports IAX?
03:19.59asdxDe_Mon: yeah, im not very familiar yet with all this stuff
03:20.20voipomaticlol De_Mon
03:20.34mostydoes the j option to Dial mean "jump" or "don't jump" in asterisk 1.2.13? the wiki page is contradictory on this
03:20.39De_Monthink about it, telIAX...
03:20.45De_Montel-IAX
03:20.45asdxohhhh
03:20.46asdxdamn...
03:20.50voipomaticlol
03:20.59voipomaticasdx it happens to the best of us
03:21.03De_Monasdx you might want to get some rest before you try anything else today
03:21.13asdxDe_Mon: indeed
03:21.14asdx:-)
03:21.57asdxi was going to sleep already...
03:22.15voipomaticasdx - TelIAX is a great choice BTW
03:22.28asdxvoipomatic: yeah, i recommended that to the customer
03:22.46asdxvoipomatic: he said he will decide with that...
03:22.55asdxs/recommended/showed
03:24.00voipomaticyeah, their support is great. (the only thing they will do is try to defer you to the per hour * support, so just be sure, if you call them, to be insistant that the problem lies on their end and they will find the problem)
03:24.52voipomaticthey do go out of their way for ya.  who you all using now?
03:27.08asdxvoipomatic: my customer was using vonage with some client, i don't know what... but i think he didn't had a pbx, but he stopped to use because the telco company blocked SIP.
03:27.17asdxvoipomatic: now he called me for an "alternative"
03:27.47voipomaticinteresting to hear that sip was being blocked
03:28.09asdxhe said he doesn't have money though
03:28.14asdxbut i will do this anyway... to learn
03:29.05asdxthey want to be ISP
03:29.06asdxlol
03:29.11asdxerr
03:29.18asdxvoip service provider
03:29.18voipomaticlol, not without any money
03:29.43CrazyTuxAnyone here ever used an SPA9000 ?
03:30.09asdxvoipomatic: i will ask him for some cash when i have everything running
03:31.09*** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net)
03:31.36asdxhe wants to use asterisk + his voip provider for now, and make his calls
03:32.07voipomaticvonage?
03:32.12asdxteliax
03:32.16asdxhe will switch
03:32.48asdxi'll try to convert him to linux
03:32.55asdxhe wants to learn asterisk and cli
03:32.55asdxlol
03:33.07*** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net)
03:33.14asdxbash, etc
03:40.21asdxis IAX a standard, or will be one?
03:40.27asdxlike SIP
03:40.37voipomaticyes
03:41.12asdxnice
03:41.29mostyit's not quite a standard like sip
03:41.46Qwelloh, that Cap'n Crunch bit on Drawn Together was SO appropriate
03:41.59asdxIETF standard
03:44.16Nuggetwhich is better?  normal cap'n cruch or peanut butter?  :)
03:44.28Nuggetcrunchberries are plain disgusting.  not even in the running.
03:44.35*** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com)
03:44.39Mackeshey
03:45.02Qwellpeanut butter, but you've gotta have standard occasionally
03:49.28asdxwhat is a good IAX hardphone
03:49.55Mackesso, does anyone use a playtone to indication the beginning of a call
03:52.34*** join/#asterisk bmg505 (n=leon@196.209.183.81)
03:53.24mostyMackes, besides the default ringing sound?
03:57.13*** join/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com)
04:00.46TJNIIasdx: Not too many make iax hardphones
04:00.59TJNIII picked one up on eBay but the DSP is crap
04:01.09JayTee52I know that X100 cards are not supported anymore and that they are not recommended but can I use 2 of them with a TDM400 card also?
04:01.24BigCanOfTunaI'm trying to pass a variable from a call file to my dialplan, my dialplan has this line: Setvar: <code=911257> and my dialplan has an operation like this: exten => _XX,1,NoOp(${code}) ................the NoOp is being called, but the output suggests that no value was passed...any thoughts?
04:02.11mostyJayTee52, yeah probably. but you should probably get a BRI or fractional PRI for the effort and call quality
04:02.45*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
04:02.51mostyBigCanOfTuna, paste that section of your dialplan on a paste site
04:03.12asdxTJNII: i see
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04:03.40JayTee52mosty, we intend to move all PSTN traffic to Digium's PRI card but for now we just need to add a couple lines and have no slots available so I need to replace 1 X100 card with a TDM400 we just bought.
04:04.45mostyJayTee52, get a sangoma PRI card with hardware echo can instead
04:05.10JayTee52I'm being squeezed by my boss who is a major [expletive deleted] and lacks the sack to ask his boss for more money until he can WOW them with some fancy dog an pony show crap using Exchange Unified Messaging.
04:05.31BigCanOfTunamosty: Here you go: http://pastebin.ca/774274
04:06.17JayTee52then we can buy some decent hardware and route our existing PRI lines through Asterisk to SIP phones and also to our legacy Nortel PBX.
04:06.49mostyBigCanOfTuna, pastebin.ca looks hosed, doesn't show me anything on that page
04:07.15BigCanOfTunamosty: You sure, looks good to me in FF.
04:07.36BigCanOfTunamosty: Looks good in Safari.
04:07.46mostyyeah, it just shows me a blank page in iceweasel/firefox
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04:08.09JayTee52so from everything I can find in Asterisk TFOT and other sources online am I right in thinking the wctdm Zaptel driver will work with both the TDM400 card and the X100 cards?
04:08.52mostyyes
04:08.58mostyfrom memory
04:09.26JayTee52mosty, thanks!
04:09.32*** part/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com)
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04:13.49JayTee52good night everyone!
04:13.59*** join/#asterisk Netgeeks-laptop (n=chris@204.11.231.198.static.etheric.net)
04:14.05JayTee52and thanks again for your help
04:14.16*** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net)
04:16.24MrTelephonei never pictured myself as a dog killer but i probably will poison the dog next door if they keep putting it out at 11am
04:17.35[TK]D-FenderMrTelephone, tameshigiri ;)
04:18.29MrTelephonewhats that
04:18.38De_Monis there a dialplan app that will tell me how many people are holding in a queue?
04:19.07MrTelephoneexten => 911,1,Dog(Bullet)
04:19.46[TK]D-FenderMrTelephone, jfgi :)
04:21.36MrTelephonehey i have to make a little powerpoint game version of deal or no deal
04:21.36MrTelephonejust the money part
04:21.36MrTelephonebut i can't remember what the denominations are
04:22.14[TK]D-FenderMrTelephone, ..... jfgi! :)
04:22.48De_Monhttp://pastebin.ca/774285
04:23.10De_Mon[TK]D-Fender there's that crazy idea you helped me with earlier this week
04:23.21[TK]D-FenderDe_Mon, Probably did....
04:24.41De_Monprobably did?
04:26.56De_MonHere's the thing, at line 36 the caller (customer) is dialing line 50 and after about 5 minuites it stops playing hold music and says Call Failed: Request terminated
04:27.28De_Monthats the sip phone, I'm tryin it again now... so its almost done
04:28.52[TK]D-FenderDe_Mon, pastebin your queue def
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04:32.34De_Monreupdated
04:32.35De_Monhttp://pastebin.ca/774295
04:32.54De_Monoh, silly pastebin made it sound like it was going to update the previous paste
04:33.02[TK]D-Fender#
04:33.03[TK]D-Fenderjoinempty = strict
04:33.03[TK]D-Fender#
04:33.03[TK]D-Fenderleavewhenempty = strict
04:33.18[TK]D-Fenderhrm... if you're not answering I'm thinking you're getting kicked if it caps at 5 min
04:33.43[TK]D-FenderDe_Mon, would also be nice to see the call...
04:34.14De_Monits helluva messy every 10 seconds I get like 10 lines in the cli
04:34.18[TK]D-FenderDe_Mon, OMG, parked agents!  lol
04:34.38[TK]D-FenderDe_Mon, sickening brilliant.  This is cute....
04:35.25De_Monit was your idea :)
04:35.46[TK]D-FenderDe_Mon, umm.... was it?
04:35.51De_Monsomething is definately weird, the caller isn't actually getting to the queue
04:36.04[TK]D-FenderDe_Mon, that doesn't sound good...
04:36.07De_MonYeah I came in with a how do i... and you came up with it
04:36.51[TK]D-FenderDe_Mon, if only I could remember all this stuff I come up with...
04:39.25De_Monyou'd be rich!
04:42.08De_Monhttp://pastebin.ca/774305
04:42.54De_MonHere's the first half, it keeps trying the members and waiting for an agent to accept call for 5min and then hangs up. I've added some extra debugging in hangup to see what I can see
04:43.21De_MonOkay I just heard the periodic announce, so I guess they are in the queue
04:44.20De_Monhttp://pastebin.ca/774307
04:44.30De_Monthats show channels output
04:48.04De_Monhttp://pastebin.ca/774311 -- And there's the end of the call, dialstatus is cancel?
04:49.14De_Monoops, forgot my s/elephant/queue/ substitution.
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04:50.53De_Mons,8 is canceling the call after 5min. wtf
04:51.31De_Monexten => s,n,Dial(Local/support@elephant-support2||G(s^pickup))
04:52.34De_Monoh thats right, Dial is holding the call in my queue...
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04:56.01[TK]D-FenderDe_Mon, looks like it recurses.
04:56.09[TK]D-FenderDe_Mon, I'm getting dizzy
04:56.57De_MonI have a pile of paper where I planned this out with all sorts of drawings trying to get it straight
04:57.14De_Monmost of them are in the trash
04:58.24De_MonThe agent is called, it goes to the remote centers queue for however long. When someone finally picks up, they press a button to actually get parked and join the queue
04:59.10De_Monso I'm not really sure why the calling party stays in the queue with 2 static members that arn't really there 99% of the time...
04:59.11[TK]D-FenderDe_Mon, Clean it up... I'm all outta dramamine :)
04:59.28De_MonBut the thing is, its the Dial command that is canceling.
04:59.31De_Monclean it up? that is clean!
04:59.33[TK]D-FenderDe_Mon, stays in queue because those are STATIS memebers and can not be kicked.
04:59.42[TK]D-Fenderstatic
05:00.31De_Monso why is dial canceling the call after 5 min
05:00.44De_MonI'm pretty sure thats the real issue
05:01.09[TK]D-FenderDe_Mon, maybe its that remote queue?
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05:01.56De_Monnope, they are still looping thru their little script after the calling party is disconnected.
05:04.02De_MonI'm testing a normal dial(sip/jon) lets see if it cancels after 5min...
05:04.51De_Mondo you really think there is cleaning up that could be done to this?
05:10.42De_MonIt's been 7min, thats not whats canceling the call.
05:14.19De_Monnow just dialing local/support@queue-support2.  It's kinda strange that it's playing music on hold isn't it?
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05:15.45De_MonDial doesn't think its done until the call is bridged, I don't think it was designed to call queues :)
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05:29.31De_Moncould the calling phone have sent the CANCEL?
05:31.28*** part/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net)
05:32.10De_MonI doubt it, but I'm testing from a different phone just to make sure
05:40.24De_Monahhhh, the softphone still thinks im "calling"
05:40.36De_MonI bet it timed out...
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06:01.30De_Monsuccess!
06:04.01[TK]D-FenderOh?
06:12.04De_Monyeah, stupid softphone. Threw an Answer() into the calling side and it didn't CANCEL the call
06:12.17De_Monit was in ringing state for 5min and canceled the call
06:12.33De_Mondialing a queue does funny things :)
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06:45.42BBHossdo any of dialogic's products work with *+
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07:04.46tzafrir_homeBBHoss, I think at least some will work with some proprietary channel driver (requires BE)
07:05.02tzafrir_homeUnless you refer to DIVA cards
07:05.16tzafrir_homewhich should work with the external chan_capi
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07:27.12Raky-2hey guys, i was wondering if it's possible to convert the extensions.conf into mysql statements
07:27.19Raky-2or at least be able to run asterisk extensions, from mysql?
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09:03.12Chris-NBhi
09:03.19Chris-NBanyone using q.sig on a pri line?
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09:08.12MicWhi
09:08.36MicWis anyone here using a siemens gigaset c470?
09:08.48MicW(sip phone)
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09:27.52TelemacHello
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09:31.09TelemacI'm trying to setup odbc in asterisk to put config in database. I've installed unixODBC, configured odbcinst.init and odbc.ini. This seems ok as I can connect with isql. Then I've added res_odbc.conf and restart asterisk. When I run 'asterisk -r' odbc commands work, so I suppose odbc modules are properly loaded bu 'odbc show' doesn't display dsn configured on res_odbc.conf. Any idea what's wrong ?
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09:35.43billybongoTelemac: what does odbc show tell you?
09:36.14cjkhi, is there a way to store peers (i am using asterisk realtime)  in more then one table without using database features like views?
09:36.16*** join/#asterisk PepOSX (n=pepOSX@190.72.153.45)
09:37.44Telemacbillybongo: nothing
09:38.11Telemaculysse*CLI> odbc show
09:38.11Telemaculysse*CLI>
09:42.16billybongoTelemac: but odbc connect seems to work?
09:42.25*** join/#asterisk blq (i=me@1.1.1.vg)
09:42.30billybongorun with more verbosity and see if it tells you more
09:42.49billybongoalso are your database logs showing connections?
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09:54.23jozuhi to all
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09:55.05Telemacbillybongo: I've already increase verbosity, and it doesn't tell me more
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09:56.32TelemacI don't think there could be any connection as firstly there is no dns (while it is configured)
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09:58.51atarihi
10:00.11billybongoTelemac: so you mean that there's no way for it to resolve the db server
10:01.20Telemacbillybong: I think. It's as if res_odbc.conf is not read whereas odbc module are loaded
10:01.42Telemacpreload => res_odbc.so
10:01.42Telemacpreload => res_config_odbc.so
10:01.53TelemacI've also tested replacing preload by load
10:04.55billybongocheck permissions on the file
10:06.21billybongoalso check the messages when restarting asterisk
10:06.33Telemacbillybongo: same perms as extensions.conf (working
10:07.04Telemacbillybong: No special message when restarting, neither in console nor in logs
10:07.31billybongowhat's in it?
10:08.39billybongoyou shold have the dsn as defined in your odbc config, username and password
10:09.05billybongoi.e. the bit in [] in odbc.ini
10:09.36billybongocan you connect OK to the dsn outside of asterisk?
10:09.39Telemacbillybongo: odbc.ini is ok, as I can connect with isql and expectedn dsn
10:09.44billybongook
10:10.07billybongoso in my odbc.ini I have
10:10.07billybongo[asterisk]
10:10.07billybongo.....
10:11.03billybongothen in my res_odbc.conf I just have
10:11.03billybongo[asterisk]
10:11.03billybongodsn => asterisk
10:11.03billybongousername => blah
10:11.03billybongopassword => secret
10:11.04billybongopre-connect => yes
10:11.25billybongoTelemac: does yours look like that?
10:12.35TelemacFor perms : -rw-r-----  1 root     asterisk   100 Nov 15 10:05 res_odbc.conf
10:13.07Telemac[ast_cnf]
10:13.07Telemacenabled => yes
10:13.07Telemacdns => asterisk
10:13.07Telemacusername => xxx
10:13.07Telemacpassword => yyy
10:13.07Telemacpre-connect => yes
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10:18.38alainrhi
10:19.07alainranybody knows if asterisk supports sha encrypted passwords in sip.conf?
10:19.37BBHossi think just md5
10:19.43billybongodns => asterisk should be
10:19.44billybongodsn => asterisk
10:20.41billybongoTelemac: ^^
10:20.54alainrhm, thats bad...
10:21.57BBHosswhy is that 'bad' the majority of web apps use md5 hashed passwords?  They may be somewhat insecure, but they're better than plaintext!
10:22.27alainrtrue, but i have ldap and it stores in sha
10:22.35BBHossoh
10:22.43BBHossheh, i see your predicament
10:23.22billybongoalainr: user openser and ldap users?
10:24.09Telemacbillybongo: Arrr, thanx. Finally I should have paste it rather writing it by hand, or should by new eyes. (Should asterisk indicates an parse error about that ?)
10:24.33BBHossIAX2 uses RSA auth
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10:24.57BBHossalso the reason MD5 is the only supported type is because thats whats in the RFC for SIP
10:25.07billybongoTelemac: probably asterisk should complain - surprised that it didn't
10:25.13BBHosslike digium really follows standards!
10:25.39alainr<billybongo> i have not tried openser, but will have a look at it
10:25.39Telemacbillybongo: Haven't any error
10:26.50billybongoalainr: there are some howtos on using ser/openser for registering and * for other stuff - could be the quickest route if you want to use ldap
10:26.51BBHosseven if you got asterisk to support SHA hashes, you would have to have a phone that supported SHA
10:28.27alainr<BBHoss> no, phone sends clear text to asterisk and asterisk authentifies with the sha password in sip.conf
10:28.53BBHosswell that would be doable, but you'd need to write a patch
10:29.58BBHosswhy dont you just put the sha hash and the phones password the same thing?
10:30.38alainr?
10:31.04alainryou can't concert sha to clear text
10:31.21alainronvert
10:31.55alainrand if user changes his password, the sip password should also be changed
10:32.07BBHossol
10:32.10BBHossok
10:33.09JTalainr: since when do phones auth with clear text?
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10:34.49anujsinghHii
10:35.16alainrdon't they? if i tell them to do so
10:35.18anujsinghhow to configure loadbalancing between two asterisk servers?
10:35.45JTalainr: not normally
10:36.19anujsinghcan i have some tutorial howto ? for load balancing between two asterisk servers ?
10:36.24anujsinghthanks in advance.
10:36.54JTanujsingh: openser perhaps
10:39.33billybongoanujsingh: you have two options, use SRV DNS records to send users to one or other asterisk or configure ser/openser to sit on the front of your pair of asterisks
10:40.59JTwell there are more options than that ;)
10:41.07anujsinghyes?
10:41.29JTLinux HA
10:41.33anujsinghi am using vicidial , heartbeat and drbd
10:41.37JThaving a group of extensions each
10:41.55anujsinghcommon mysql database
10:42.09anujsinghreplicating with drbd
10:42.30JTheh mysql
10:45.17billybongoanujsingh: OK, there are an infinity of options, of which the 2 I gave was a small subset ;-)
10:45.57anujsinghwhich one is simple ? i am n00b with aster
10:45.58billybongoanujsingh: what are you trying to achieve? Complete scalability/redundancy ?
10:47.38anujsinghyes scalability/redundancy
10:48.31JTanujsingh: that's not a simple goal
10:49.57anujsinghwithout trying nothing can be done. I want to try
10:50.24anujsinghi saw someone showing balncing with vicidial monitor tool ,.
10:50.42JTsure
10:50.56anujsinghsuch as above 15 calls next call will be transfered to anothe aster.
10:50.56JTi'm just saying that the wish for it to be simple is unreasonable
10:52.35billybongoanujsingh: if I was doing this I would have a load-balanced openser cluster on the front end, as many asterisks as you need to handle the media, and a database cluster of your choice to hold the config/voicemails etc
10:52.36anujsinghis it possible to use balance package for load balancing ?
10:53.23JTbalance package?
10:53.29anujsinghsomeone claming he has done it with balance .
10:53.31billybongoanujsingh: do your asterisk users need to find each other across different machines?
10:53.31anujsinghyes
10:54.10billybongoif so you have to think it out very carefully
10:54.36anujsinghneh both the servers will be at a single place.
10:54.42*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
10:54.46anujsinghsame users.
10:55.04anujsinghsame network  same users.
10:55.21alainrJT is it then not possible to force the phone to send cleartext or is this against sip standard?
10:55.22billybongook
10:55.47billybongoanujsingh: so if user A is registered with server 1 and user B is registered with server 2, how will they find each other?
10:56.42JTalainr: pretty sure it's not possible
10:56.46JTalainr: it's also stupid
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11:00.56alainrJT is there a problem if i would use md5 in sip.conf, does it work with all phones? or is it recommended to use cleartext in sip.conf
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11:19.19JTalainr: just use cleartext, asterisk will make the md5 hash on demand
11:20.15alainrso using md5 is also not good?
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11:20.27JTi guess it's fine
11:20.33JTplaintext is more convenient
11:20.38JTover sip it's MD5 anyway
11:20.49alainri don't like to store any password in cleartext
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11:23.00alainrso if sip is md5 anyway i just store it as md5
11:23.10JTi guess you can
11:23.49alainrbut still have to find a way to get password from ldap sha to md5
11:24.08ussrbackhi all
11:24.12alainrmaybe with storing it double when registering users
11:24.19JTalainr: that's impossible
11:24.23JTi guess you'll have to
11:24.41alainrconversion is impossible, i know
11:25.05alainrdoes asterisk work fine with radius?
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11:29.18ussrbackwhat should i install to use H323 ?
11:29.32ussrbackis it necessary to install Openh323
11:29.34ussrback?
11:29.40JTpreferably not asterisk ;)
11:30.05tzafrirussrback, no. you can also use ooh323c from addons
11:30.35tzafrirand IIRC you had a problem building it yesterday, and I recall asking you for a more complete trace
11:30.42ussrbacktzafrir: i just need to compile this addon and put so in my modules dir?
11:30.49JTchan_woomera is meant to be the best H.323 option
11:31.03tzafrir'make install' should do that
11:31.13ussrbackJT: why woomera is the best solution?
11:31.29JTussrback: the rest are very unreliable
11:32.25ussrbackbut shoud i need to install woomera on separate box?
11:32.44JTno, the box does not need to be seperate
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11:33.18JTussrback: btw, what are you connecting to H.323 for? phones, provider?
11:34.56ussrbackprovider
11:35.11ussrbackill send international calls throught the h323
11:35.16ussrbackand receive them
11:35.29JTif you don't have luck with chan_woomera you may just want to use Yate as a H.323 <---> SIP gateway
11:35.36ussrbackon the other side ill have sip users
11:36.11ussrbackwhere from can i download woomera?
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11:47.05ai-acan i use the manager port to track call transfers (attended) calls ?  and who out of all Dial(SIP/,....) finally picks up the call ?  I want to track all calls as they flow via transfers, and even parked, from ringing to termination.
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12:26.52*** join/#asterisk Haris (n=Haris@unaffiliated/haris)
12:26.54HarisHello guys
12:27.01Pondiboyhi
12:27.28HarisCan we hook up normal pstn phone sets to Avaya IP office digital lines and talk through them or do we absolutely need an IP/SIP phone to talk on the digital lines?
12:28.25*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
12:28.54PondiboyHmm donno i am a newbie there m8
12:34.14tzafrirHaris, how exactly is this a question about Asterisk?
12:34.43cpinahello everybody
12:34.47tzafrirhi
12:34.51tzafrirwhat's up?
12:35.30Haristzafrir: Its not!
12:37.27*** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net)
12:37.38cpinatzafrir: 1 moment...
12:38.02*** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net)
12:38.05tzafrirHaris, you can't simply hook them
12:38.33HarisSo that means, only sip phones can be used with it
12:38.49Harisbut normal phone sets could still be used with analog lines in it?
12:41.51cpinatzafrir: about my ztdummy
12:41.54cpina:-)
12:42.03cpinawe changed the server while we buy a hardware card :-)
12:43.36tzafrirHaris, yes, you need some sort of analog adapter for analog phones
12:43.38*** join/#asterisk harpal (n=Harpal@124.125.255.223)
12:43.49*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
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12:46.01ai-aIf the E1 card is Alaw, and the phone is Alaw, does asterisk just forward the audio stream?   we seem to be having some sort of 'auto gain' effects on our calls after moving to Asterisk from old hardware pbx.  what could be causing this?
12:46.29*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
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12:51.34JTai-a: the fact that your handsets are different?
12:52.21*** join/#asterisk killfill (n=killfill@pc-164-134-45-190.cm.vtr.net)
12:52.28killfillhi..
12:52.34killfillin show channels, i see this: Local/65@default-921 65@default:4         Ring    TrySystem(fetch -T 2 "http://l
12:52.43b1ch0hi everibody, i am planning to use G729 (0r 723) on a new installation because of all my phones support those codecs, what append if i enable an IVR (with "729 voices) ? do i have to pay any kind of licence ?
12:52.47killfillits stuck in there. how would i kill that channel?
12:53.00killfillasterisk it at 100%cpu.. i think becouse of that...
12:55.06ai-aJT: its a call from outside coming into the building and answered by a snom300 phone.
12:55.33ai-aasterisk records the call, and we can hear that when they are not talking the audio is louder.
12:56.29b1ch0or if i need to listen my voicemail ?
13:01.36*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
13:03.56cpinain one asterisk server, after type show sip channels, there is a lot of BYE state channels, with unknown codec
13:04.02cpinahow we could "remove" or destroy it?
13:08.20*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
13:09.07JTai-a: did your old pbx use Snom 300 phones?
13:09.54tzafrirkillfill, soft hangup ?
13:10.00tzafrirkill the processes?
13:10.05*** part/#asterisk Haris (n=Haris@unaffiliated/haris)
13:11.08tzafrir(pkill, or ps auxww | grep whatever | grep -v grep | awk '{print $2}' |xargs kill
13:11.08tzafrir)
13:11.08cpina:-)
13:11.08cpinai mean sip channels
13:11.18*** join/#asterisk lirakis (n=lirakis@65.200.191.253)
13:11.22cpinaIP_ADDRESS   xxxxx yyyyyyy 00102/00103  unkn  No  (d)  Rx: BYE
13:11.27cpinawe have had 180 of it
13:11.41cpinayes, it has flag "d" to destroy
13:11.44cpinai would like to destroy before
13:11.45cpina:-)
13:12.06cpinatzafrir: ops, you was answering to killfill not to me :-D
13:12.26ai-aJT: nope.
13:12.45tzafrircpina, I suppose you have tried a soft hangup, right?
13:12.55cpinatzafrir: is only for active channels, yes
13:12.56*** join/#asterisk El_Capitan (n=BluesBoy@217.6.11.154)
13:13.02ai-aold system was completly different. but using the same E1 line.  we've added asterisk pbx, and 50+ snoms.
13:13.27ai-ai cant see any auto gain in the snom or sangoma card settings.   So wondering if asterisk is performing auto gain.
13:14.04ai-acome to think of it.,. you can hear the auto gaining in the recordings which have nothing to do with the snom, so it must be either the asterisk, sangoma or e1 connection somewhere performing this auto-gain
13:14.06*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)
13:15.53JTai-a: i think you're being slightly unreasonable expecting the volumes to be completely the same
13:16.01mostyi want to accept a call on one asterisk server, and terminate it via a second asterisk server, such that the billsec in the cdr entries is the same on the two servers- is this possible?
13:16.18ai-aJT: when the customer isnt talking the volume goes up and we hear a lot of background noises.
13:16.29*** join/#asterisk rob0 (i=rob0@sorry.nodns4.us)
13:16.41ai-aTelesales are saying this makes it hard to hear the customer as it seems to auto-gain too quickly.
13:16.49JTai-a: less silence supression, or clearer phones, take your pick
13:16.51ai-aso they turn the vol down, but then the customer is quiet.
13:16.57*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:17.02ai-aJT: what does auto-gain ?
13:17.08JTnothing in asterisk
13:17.17ai-aexactly what i thought.
13:17.31ai-aand the sangoma experts say the card doesnt do auto-gain either.
13:17.44JTwhat card is it
13:17.48ai-amaybe its the customer's analogue phone that is auto gaining for them.
13:17.52ai-aA101D
13:18.02JTno, it's probably not auto gain at all
13:18.17JTprobably the old pbx supressed background noise
13:18.32ai-aI see.
13:19.19JTthe only "auto-gain" possibility i see is in the snoms
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13:19.23*** mode/#asterisk [+o blitzrage] by ChanServ
13:19.25JTtried a different phone?
13:19.46ai-anope. but if its the phone, asterisk monitor wouldnt notice it.
13:20.08JTjust try a few different phones
13:20.20*** join/#asterisk Lasse123 (n=Lasse@int-gw.algitech.com)
13:21.07ai-athing is, we have 20+ guys, and 1% complaints... but the bosses want 100% perfection.
13:21.42JTso do some tests for them? :)
13:22.06ai-ai prefer them to shut up and be happy with the new tech..  they are lo paid :)
13:22.07ai-a*low
13:22.14JTheh
13:22.53*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
13:23.48lirakisDoes anyone know of good specific resources for ISDN signaling functionality in asterisk?
13:24.03JTerr, what?
13:24.07lirakisim curious how asterisk responds to certain codews
13:24.09lirakis*codes
13:24.10*** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl)
13:24.35JTthen the best documentation is the source
13:24.59lirakisJT: .. like can i configure asterisk to send 34 when i get a specific isdn code
13:25.25*** join/#asterisk Telemac (n=telemac@213.223.113.74)
13:25.28JTprobably by modifying chan_zap.c or similar
13:25.31TelemacHello (again)
13:29.34TelemacI'm still having some trouble to setup realtime with odbc. dsn is properly resolved and connected by asterisk (odbc show), when I execute "SELECT * FROM sip_conf WHERE name='x'" within isql I get matching row, but when I check it within asterisk with "realtime load sipusers name x" nothing is found. In extconfig.conf sipusers is bound to "odbc.ast_conf.sip_conf", ast_conf is my working dsn and sip_conf is the table name. What's wrong there ?
13:33.06*** join/#asterisk sasch (n=info@host117-234-static.4-79-b.business.telecomitalia.it)
13:33.08saschhi all
13:33.11*** join/#asterisk ming_zym (n=ming_zym@124.14.233.147)
13:34.02saschanyone know a link for echo cancellation with tdm400p ??
13:34.25[TK]D-Fendersasch: All you have is Zaptel.  Go WIKI it up.
13:35.22saschcan you get me a link
13:35.23[TK]D-Fendersasch: so taht being the routines that come with Zaptel as a base, HPEC from Digium (go to their site to check out), or OLSEC ( http://www.rowetel.com/ucasterisk/oslec )
13:35.29[TK]D-Fender~wikis
13:35.30jbot[wikis] http://www.voip-info.org
13:35.33saschexcusme for my english I'm italian :-P
13:35.56saschthanks !!!
13:37.31Lasse123BUG? Incoming PSTN call to asterisk with prefered codec ALAW (but iLBC supported), asterisk forwards call to client which responds "200 OK" with sdp indicating it only supports iLBC, asterisk responds with ACK and starts P2P bridging call which results in client receiving ALAW?
13:41.35mostyis it possible to get asterisk to optimize the choice of codec in order to avoid transcoding? ie if i'm trying to forward on a call using codec X to another asterisk server which supports X, use that codec even if it's not the first priority (first allow)
13:42.23*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
13:42.42puzzledhi
13:44.26[TK]D-Fendermosty: Yes.  Make another peer with only the codec you want in it.
13:44.33robl^hrmmm -- ~wikis  is plural, yet jbot returns a single wiki.  I declare a bug!
13:45.40[TK]D-Fenderrobl^: ~wiki is a reserved jbot function for Wikipedia
13:45.59mosty[TK]D-Fender, is it possible to route calls based on codec in the dialplan? eg send g729 calls to gateway A, send gsm calls to gateway B
13:46.05robl^ahhh!!  I knew there was a simple answer ;-)
13:46.40[TK]D-Fendermosty: You can get the codec of the current channel easily enough. "show functions" <-
13:49.08*** join/#asterisk maruz (n=maumar@gw.cost.it)
13:50.41_x86_morning all
13:51.53TelemacI've configured extconfig.conf but it looks like this file was not loaded. Is it read by default or should I ensure some module to be loaded (res_config ?)
13:51.57mosty[TK]D-Fender, do you mean CHANNEL? i think that's new in 1.4, and i need to use 1.2
13:52.17*** join/#asterisk duckz (n=duckz@85.204.47.228)
13:52.20_x86_mosty: 1.2 isn't supported anymore
13:52.55mostyi'm stuck with 1.2 until there's a viable backport for debian etch, unfortunately
13:52.58[TK]D-Fendermosty:  SIPCHANINFO(item)                    Gets the specified SIP parameter from the current channel <--------
13:53.18mostyi saw SIPCHANINFO, but i need this to work with IAX
13:53.25*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
13:53.28[TK]D-Fendermosty: Thinking you're restricted to friggen PACKAGES.  How sad...
13:53.37_x86_mosty: dude, compile Asterisk from source... it's easy
13:53.48_x86_mosty: pre-compiled asterisk is utter crap
13:53.56*** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it)
13:54.01[TK]D-Fendermosty: IAXPEER(<peername|CURRENTCHANNEL>[:item] <-------
13:54.07mostyi can compile asterisk from source OK, but i run too many machines for that to be practical
13:54.27[TK]D-Fendermosty: and you upgrade * how often?
13:54.39markithi, I've heard that ztdummy has a recent bug... has it been solved in svn stable? someone with a bug # to give me? a friend of mine has problems with conference and meetme...
13:54.40_x86_mosty: compile it once the way you want it, and make a tarball of the output... copy from machine to machine...
13:54.49ZenBSDinot really mosty.. just learn to compile and create your own .deb pkgs so you can just dpkg install it =p
13:54.57[TK]D-Fendermosty: Either way, I've just given you the answer
13:55.13*** join/#asterisk shinao1 (n=shinao1@209.159.162.105)
13:55.22mostyactually i am learning to build asterisk deb's, but i'm not yet confident enough that my packages are ok
13:55.43mosty[TK]D-Fender, thanks, i will see what i can get done given my constraints
13:57.17ZenBSDianother bit of advice mosty.. seeing as how debian's developers move at a pace only seen by turtles.. you might consider switching to ubuntu server going forward so you aren't stuck waiting for those molasis asses to get things into gear =p
13:57.43ZenBSDioh and if you know any deb devels.. please quote me verbatum =p
13:58.10mostyZenBSDi, i actually like the pace of debian for 99% of the packages
13:58.33ZenBSDiif you *like* the pace of debian .. you'd *love* the pace of ubuntu =p
13:59.06ZenBSDithats why I gave up debian.. the U-crew is tearing it up!!!
13:59.10mostyi don't want anything to change unless it has to. which is good for most things, but not something that's still maturing like asterisk unfortunately
13:59.41mostyi could try ubuntu lts i suppose, what version of asterisk does that have?
14:00.19ZenBSDiasterisk is very mature as it sits despite what even the devels might say.. I just put an asterisk system into place and here it is .. 600+ uses already .. p4 3ghz with 1 gig of ram.. this thing is smoking with 30 users each
14:00.32ZenBSDibased on 1.4.10
14:00.45ZenBSDiI've got some java scripts using the fastagi interface
14:00.56ZenBSDiload is sitting pretty at 1.6 ..
14:01.04ZenBSDioh .. it's ubuntu 7.10 server too
14:01.05ZenBSDiheh
14:01.50mostyregular ubuntu releases are not stable enough for me
14:02.03ZenBSDihow long you been a linux user?
14:02.19mostymore than 10 years, i don't remember exactly
14:03.04mostyif this was just a small asterisk box like yours it would not be a problem, but i admin many asterisk boxes, and some of them are large
14:03.18[TK]D-FenderZenBSDi: Know what the worst part of a head-on collision between two turtles is?  The hours of screaming :D  aaaaaaaaaaaaaaaahhhhhhhhhhhhhhh!!!!!!!!!!!!!!!!!!!!!!!!!!!
14:03.40ZenBSDipuhlease.. you can't possibly have a decade of this crap and be afraid "stability" issues .. incase you haven't noticed even the mighty old and iron forged debian has "stability" issues. if you have a decade of linux .. you'd be able to fix it so you dont' have to deal with the other 80% of things debian does wrong.
14:03.57TelemacIs extconfig.conf supported with asterisk 1.2.21.1-r1 ?
14:04.28[TK]D-FenderZenBSDi: And turtles are pretty quick too... I'd say more akin to molasses going uphill in January ;)
14:05.24ZenBSDi[TK]D-Fender, colliding turtles.. or going uphill .. either way.. it's more development unfolding quicker than debian devels.. =p
14:05.28mostyZenBSDi, i only run debian stable on boxes that i do not want to touch if i can avoid it. i don't have the time to keep upgrading these machines when the security support ends for a certain release of something like fedora
14:06.01*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
14:06.11[TK]D-Fendermosty: "yum update" <- CentOS
14:06.50mosty[TK]D-Fender, sure but if you upgrade between releases that can break things easily
14:07.19*** join/#asterisk irule (n=irule@200.53.61.4)
14:09.00ZenBSDiwho said anything about fedora mosty.. I think you need to read what I said.. I said . and here I'll break down itno parts... Going forward .... so you are NOT stuck waiting for debians development cycles .... you should consider using ubuntu server..     so what I said is.
14:09.29PBXXi have problem with /usr/local/sbin/faxgetty when he is runnign from inittab he cannot red buffer from iaxmodem
14:09.33*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
14:09.37PBXXwhen i run it on shell everything its fine
14:09.47coppiceonce you've waited for debian release cycle, continental drift seems really fast :-)
14:10.08rob0Stop Continental Drift! Sign the petition.
14:10.28*** join/#asterisk freezey (n=freezey@maher.mercy.edu)
14:10.32ZenBSDibut enough of the distro war.. this is about asterisk .. asterisk rocks!
14:10.35mostyzenbsdi: i was just exagerating to make a point. i need a distribution with a long support life. regular ubuntu is too frequent for my situation. ubuntu LTS may be ok, i don't know
14:10.43*** join/#asterisk Haris (n=Haris@unaffiliated/haris)
14:10.45Harishello guys
14:10.47freezeyif somebody set a password to their extension's voicemail how exactly do i reset the password?
14:10.50*** join/#asterisk slima (i=slima@unaffiliated/slima)
14:10.53HarisIs there a difference between a digital phone and a sip phone?
14:11.07ZenBSDisip is a protocol ..
14:11.27mostynothing changing means nothing new breaking. my users hate it when their phones don't work
14:11.48ZenBSDiso you should look into a digital phone that supports sip =p
14:11.58[TK]D-FenderHaris: Typically the term "digital" refers to proprietary electron signalling that is tied to a single vendor and won't work with any other system.
14:12.14HarisAvaya 4406D, Avaya 5410
14:12.28Haris[TK]D-Fender: I see
14:12.31ZenBSDiakkk [TK]D-Fender propietary!!!
14:12.35Haris[TK]D-Fender: Still, is it sip capable?
14:12.39[TK]D-FenderHaris: a SIP phone is a phone that uses TCP/IP and speaks the SIP protocol to communicate with typically pretty much any SIP server.
14:12.48HarisI see
14:12.49*** join/#asterisk macros73_ (n=cs@dsl093-063-226.pit1.dsl.speakeasy.net)
14:12.54freezey[TK]D-Fender: if somebody set a password to their extension's voicemail how exactly do i reset the password?
14:13.01Harisso we can say its a normal pstn phone, but a properitery one?
14:13.05freezey[TK]D-Fender: you seem to answer most questions
14:13.29[TK]D-Fenderfreezey: vi /etc/asterisk/voicmail.conf
14:13.36ZenBSDi[TK]D-Fender is an asterisk demi-god.. so of course he answers the most q's
14:13.53Harisdemo god?
14:13.55Harisdemi+
14:14.04*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)
14:14.08Harisare they still alive?
14:14.15[TK]D-FenderZenBSDi: No, I'm not that good, but I know most of the basics of day to day use.  I don't do DB's, AGI, or other coding, and suck at linux (mostly)
14:14.37*** join/#asterisk ussrback (n=MAX@80.241.177.19)
14:14.42[TK]D-FenderHaris: what does "normal" mean if its "proprietary"?
14:14.49HarisOk, on the subject of proprietary stuff
14:14.53freezey[TK]D-Fender: ok i see which line you mean 4586111,username,email@mail.com,,|tz=eastern|attach=yes|saycid=no|review=no|operator=no|envelope=yes|delete=no so that 4586111 should be password?
14:15.03[TK]D-Fenderfreezey: Yes.
14:15.05Hariswould normal non-proprietary digital phones work with Avaya IP phone 406?
14:15.10Harisphone = Office
14:15.11ZenBSDiword.. if you start doing the agi coding you'll really own then dude.. agi coding is where you can do that stuff like call in and pay with a CC over the phone.. the script grabs that stuff and processes it
14:15.15freezey[TK]D-Fender: tried usin ghtat passowrd and still got denied
14:15.16ZenBSDithats what I'm working on now
14:15.39[TK]D-FenderHaris: You can get line cards for Avaya to support standard analog phones.
14:15.50Haris[TK]D-Fender: on digital ports?
14:15.53Harisor sip ports?
14:15.55[TK]D-Fenderfreezey: And what did you do to APPLY your changes?
14:15.57Harissip capable+ ports
14:16.04freezeydialplan reload
14:16.27_x86_[TK]D-Fender: eh?
14:16.37[TK]D-FenderHaris: Depends on your line card.  Some mfg's digital line ports ALSO support analog phones directly.  In other cases you may need to by a card DEDICATED to that
14:16.39_x86_[TK]D-Fender: you are teh suck at leenocks?
14:16.51[TK]D-Fenderfreezey: voicemail != dialplan
14:17.08[TK]D-Fenderfreezey: "module reload app_voicemail.so"
14:17.10*** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org)
14:17.13Harismeans, additional! cost
14:17.26freezey[TK]D-Fender: great thanks
14:17.43[TK]D-FenderHaris: One prime reason that proprietary PBX SUCK.  Welcome to vendor lock-in, and stock up on KY.
14:18.08b1ch0JC: hi still there ?
14:18.17freezey[TK]D-Fender: and to reset the mailbox like deleted all the voicemails in their is their a way to do it through asterisk?
14:19.08[TK]D-Fenderfreezey: to fully reset a box you would delete that box's folder entirely.  You can code something in the dialplan for this easily enough.
14:19.08b1ch0im tring to test transversal nat, and i cant understand if externip=222.222.222.222 is router address or my * natted IP
14:19.25freezey[TK]D-Fender: ty
14:19.26[TK]D-Fenderb1ch0: its your routers EXTERNAL IP
14:19.42[TK]D-Fenderb1ch0: the address it gets from your provider.
14:19.47*** join/#asterisk asdx (n=diego@adsl-147-91.click.com.py)
14:20.07b1ch0ok, thanks , the same as my * default gateway, right ?
14:20.31*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)
14:20.33[TK]D-Fenderb1ch0: Dunno what your * gets for that.  I gave you the only answer that counts.  Run with it.
14:21.10[TK]D-Fenderb1ch0: typically systems behind NAT routers get PRIVATE addresses which is NOT what is called for here.
14:21.47krdian_<PROTECTED>
14:21.47TelemacI'm trying to test realtime from odbc. My dsn seems ok, I've configured extconfig.conf but when I try realtime load in asterisk console no matching row is found. So I wonder extconfig.conf was loaded by default with asterisk 1.2, or even supported ? I'd like to have the whole sip.conf in DB, should I used "switch => ..." statement anywhere or just extconfig.conf ?
14:21.52krdian_hello
14:22.26rob0whatismyip.org (which doesn't work if your ISP uses HTTP proxying)
14:22.56[TK]D-FenderTelemac: Real-time requires you to code your contexts into extensions.conf and use the switch statement to tell * to use Realtime.  You must do this for ALL contexts.  So the CONTEXTS may be dynamic, but not the STRUCTURE.
14:23.25[TK]D-Fenderc/CONTEXTS/CONTENTS (extens)/
14:23.30b1ch0ok, i understand .... so, after IAX test (that work great behind the transversal nat i got) im going to try if SIP works well or not following http://www.aocomputing.net/?p=3 instructions
14:23.42b1ch0wish me luck
14:23.53[TK]D-Fenderb1ch0: And then we'll see how good you are at following instructions :)
14:24.18b1ch0:-)
14:24.25Telemac[TK]D-Fender: Ok, thanx I will look at that
14:26.40mosty[TK]D-Fender, it appears that ${SIPPEER(account:codec[0])} shows the prefs in sip.conf, but what i need is only those prefs in sip.conf that the caller supports. is it possible to figure that out?
14:27.06[TK]D-Fendermosty: Dunno.
14:27.16shinao1hi.. has anyone ever used elastix before? with xorcom astribanks?
14:29.34tzangercoppice: (or anyone, really) -- any good links or references for fixed-point echo cancellation for short spans?
14:30.07coppicewhat do you mean by a short span?
14:30.14Qwellone that isn't long?
14:30.18*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
14:30.38[TK]D-FenderWhat about medium spans?  Or medium + smidgeon?
14:30.57tzangeryeah, a short-length span.  sliptest is reporting between 700-900 for the span
14:31.21tzangerthat's not ms, but I'm guessing it's frames? (125us?)
14:32.17coppicehave you tried OSLEC?
14:32.40tzangerno, I haven't looked at it yet, it's on my list though
14:33.25krdian_huh, anybody had problem with Astrisk::AGI 0.10 ?
14:33.32tzangerahh it's already targeted for blackfin
14:33.47coppiceduh! of course
14:33.53tzangerI didn't realize oslec was rowetel
14:34.09b1ch0...
14:34.52b1ch0hi, how about configuring all internal phones using IAX instead of SIP in an internal LAN ?
14:35.23TelemacStill trying to setup odbc realtime. My dsn is ok (loaded/connected in asterisk), I've completed extconfig.conf, added "switch => Realtime" in extensions.conf as first for "[mycontext]" but it still fails to get matching row in asterisk console with "realtime load sipusers name user_name"... What could be wrong so ?
14:35.32ussrbackwhen im trying to make install asterisk-addons i got error http://pastebin.ca/775150. how can i fix it?
14:36.23*** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com)
14:36.24coppicehey! Asterisk on an FPGA :-)
14:36.50tzanger:-)
14:37.40tzangerhmm oslec looks interesting for sure
14:37.47[TK]D-Fenderb1ch0: What kind of phones?  And whats the point of IAX phone INSIDE a LAN?
14:39.21puzzledussrback, the Makefile has a typo. if you go to the .libs directory you can find the stuff in there
14:39.50ussrbackyes i can find
14:40.05ussrbackit means that makefile have error?
14:40.18*** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net)
14:41.48puzzledussrback, yes
14:42.40ussrbackok so how can i install ooh323c
14:42.48De_Monfix the typo
14:42.51*** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-61-178.w82-120.abo.wanadoo.fr)
14:42.58shinao1tzafrir: hey
14:43.03tzafrircoppice, I noticed OSLEC has a problem with amd64 here (causes silence). At least the variant in the Debian packages.
14:43.04shinao1can we still talk some more?
14:43.11tzafrirHaven't had time to invastigate
14:43.30shinao1how did you do it? i think im using elastix -0.8
14:43.51*** join/#asterisk Darthclue (n=e054502@fw149.nisd.net)
14:43.59tzangerhmm, I would have thought NAPI would make TDMoE worse, not better
14:44.11tzangerbut that's just theorizing, I haven't tested yet
14:44.21muirohey, is there an app to say a currency amount?
14:44.36coppicetzafrir: I haven't tried it, but I understand that works OK. are you sure the version is up to date? I remember there were leak fixes after initial release, but there might be others
14:44.38ussrbackhow? i am not programmer
14:45.20tzafrirIt's a pretty recent one
14:45.33puzzledussrback, change libchan_h323.so.1.0.1,libchan_h323.1.0.1 in the Makefile and try again
14:46.07[TK]D-Fendermuiro: Not in 1 step.  You'll have to break it up yourself.
14:46.15muiroah, ok
14:46.39muiroI'll do it and the contribute it as an addon because that seems like it should be pretty ubiquitous
14:48.23*** join/#asterisk irule (n=irule@200.53.61.4)
14:48.25[TK]D-Fendermuiro: Too specialized and not hard to di in relatively pure dialplan.
14:48.54[TK]D-Fendermuiro: This is much like I class the old "blacklist" stuff.... waste of time.  We have existing stuff that con do this easy.
14:49.08muiroyeah, I suppose
14:49.39[TK]D-Fendermuiro: What I WOULD advise if you do it in standard dialplan and contribute the code to the WIKI instead.  This will both give people the "answer", as well as possibly teach/inspire someone about programming.
14:50.01muirosounds good
14:50.41muiroI actually haven't gotten into learning how to incorporate either system applications or anything with the AGI yet so it'll be pure dialplan
14:54.36*** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it)
14:54.49ussrbackwhich is config file for chan_ooh323.so?
14:55.14[TK]D-Fenderussrback: ...
14:55.18[TK]D-Fender~book
14:55.18jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
14:55.20*** join/#asterisk CunningPike (n=arodgers@204.239.8.149)
14:55.20[TK]D-Fender^^^^^^^
14:56.00muiroit's a good book
14:58.11b1ch0well it doesnt make much sense installing iax in an inside lan, but that was just an idea (to avoid future provisioning issues) ... just wanted gurus opinion
14:58.34[TK]D-Fenderb1ch0: Unless you NEED IAX I advise against its use.
14:58.54blitzrageI prefer SIP myself
14:59.25[TK]D-FenderYay, Bell just got the HTC Touch.... time to upgrade & grab their 7$ unlim data plan :)
14:59.55blitzrage[TK]D-Fender: huh?
15:00.06[TK]D-Fenderblitzrage: http://www.bell.ca/shopping/fr_CA_QC/66393.details?tab=SPECS&colourId=undefined&contractId=term36m
15:00.36*** join/#asterisk CunningPike (n=arodgers@204.239.8.149)
15:00.40rob0On IAX v. SIP, I have 2 sites, an * server at each. I linked them using IAX2, thinking that would be best. Would there be any advantage in changing that to SIP?
15:00.47*** join/#asterisk Darthclue (n=chatzill@fw149.nisd.net)
15:00.52blitzrageBell has a $7 unlimited data plan?
15:01.00*** join/#asterisk waverly360 (n=waverly@42.sub-75-200-219.myvzw.com)
15:01.08*** part/#asterisk waverly360 (n=waverly@42.sub-75-200-219.myvzw.com)
15:01.18[TK]D-Fenderblitzrage: yup, for non-tethered use
15:01.30blitzragerob0: does it work? If so... why switch?  If it ain't broke, don't fix it -- if you don't know why there would be an advantage (or why you are at a disadvantage), then why change?
15:01.42coppice[TK]D-Fender: just got it? i thought it was about to be discontnued
15:01.58blitzrage[TK]D-Fender: hrmmmmmm.... Roger's data plans are RIDICULOUSLY expensive
15:02.00rob0yeah, I was just curious, given your comment and [TK]D-Fender's.
15:02.03[TK]D-Fendercoppice: I know, silly North Americans....
15:02.36[TK]D-Fenderrob0: If it ain't broke... but that is specifically to like 2 * systems, which is the only thing I'd use it for, and only that.
15:03.21*** join/#asterisk PodMan99a (n=keith@host81-149-176-8.in-addr.btopenworld.com)
15:03.26rob0gotcha, ty
15:04.19TelemacI've "[settings]\ sippeers => odbc.ast_cnf.sip_conf" in my extconfig.conf, so what could prevent "realtime load sipusers name _user" from finding matching row within Asterisk console ?
15:04.28coppice[TK]D-Fender: the HTC Touch is cute, and fairly cheap for an HTC phone
15:04.31rob0Both sites use SIP for external services (orig./term.), and one site has SIP clients.
15:04.32*** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com)
15:04.36[TK]D-Fenderrob0: Oh... and for use where SIP gets blocked by asshole ISP's
15:04.41rob0haha yeah
15:05.20[TK]D-Fendercoppice: yeah, its base is 2/3 that of the HTC 6800, slimmer, no WIFI, but with that data plan I don't really care.
15:05.31coppiceuse SIP++
15:05.36coppicerepeating ++ until you find a port they don't piss about with
15:06.02PodMan99ahey all ... have an issue with my asterisk server  how can i get my inbound number to dial a queue then have it dial agents .... currently the call enters queue like -- exten => 05601048894,1,Queue(example_queue) but does not get agents called
15:06.03coppice[TK]D-Fender: how much to they charge for it?
15:06.34*** join/#asterisk e` (n=e@38.102.196.202)
15:06.35[TK]D-Fendercoppice: Here non-subsidized : Touch = $400CAD, 6800 = $600$CAD
15:06.37mostyPodMan99a, how many agents are in the example_queue ?
15:07.11*** join/#asterisk fnordus (n=dnall@24.84.160.227)
15:07.39PodMan99a2
15:07.49coppice[TK]D-Fender: its about US$280 here, last time I looked
15:08.03mostyPodMan99a, and they're both idle when you do "show queue example_queue" ?
15:08.10PodMan99ayes
15:08.11[TK]D-Fendercoppice: You know vendor lock-in... OH, and this is the CDMA version.
15:08.32coppicewe only see the GSM one
15:08.37mostyPodMan99a, what does it say about callers in the queue?
15:08.43PodMan99a2 seconds
15:09.03PodMan99a<PROTECTED>
15:09.10mostybingo
15:09.13PodMan99aphone is logged in and connected as can make calls
15:09.17PodMan99ajust im not an agent
15:09.25mostyyou haven't logged in as an agent
15:09.30[TK]D-Fendercoppice: Yeah, but thats what my cell-co uses and the plan I'm on is really good for here so there's a price to pay...
15:09.52[TK]D-FenderPodMan99a: Not logged in = don't expect to be called by the queue.
15:10.01[TK]D-FenderPodMan99a: Checken & egg show you're running there....
15:10.07[TK]D-Fenderchicken*
15:10.18PodMan99a[TK]D-Fender i am logged in
15:10.19coppicethey closed down the CDMA network here. now I've seen a fresh tender thing to start one up again. weird
15:11.10*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net)
15:11.15_x86_what's the cheapest CSU that adtran makes?
15:11.19_x86_single-port
15:11.22asdxlol this dumb-asshole guy (my customer) wants me to help him to build a service provider and he wants to pay me $50 usd.
15:11.25asdxwtf
15:12.14PodMan99acan i paste someone some lines of my config for the queue?
15:12.25Darthclueasdx, tell him that's a per connection charge and charge him everytime that one of his customer's connects
15:12.27[TK]D-Fenderasdx: Ask him if he wants fries with that :p
15:12.37DarthcluePodMan99a, use pastebin
15:12.42Darthclue~pastebin
15:12.43jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well
15:12.48PodMan99agood man Darthclue
15:13.36asdxDarthclue: lol
15:13.39asdx[TK]D-Fender: lol
15:15.14PodMan99aALL: http://pastebin.com/m5de47094 any ideas ... i can see something is wrong but not sure what
15:15.17asdxi'll do it anyway to learn, he will buy a teliax account so i can play it and learn, then when i have everything up and running i can ask for some cash, and if he doesn't pay i'll just make uninstall asterisk.
15:15.33asdxor turn off the server
15:15.38*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
15:15.38*** mode/#asterisk [+o anthm] by ChanServ
15:16.01*** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net)
15:16.33DarthclueJust route all the calls to a recording that says "I'm sorry.  I've chosen to be cheap and not pay the developer.  To have your service restored, please call me at (insert his cell, home, and other numbers here)."
15:17.22asdxDarthclue: lol, that would be nice as well...
15:17.26PodMan99aDarthclue : I cant be a developer unless i play ... trial and error... have to learn somewhere
15:19.40DarthcluePodMan99a, that was directed at asdx.  If you really want to play, build a cheap box at home and hook into the pots.  Use it to send the telemarketers to the monkeys and pick up a couple of sip phones and test out things like customized rings and such.  Makes it nice to know exactly who is calling during dinner.  I don't answer when it says "Mother-In-Law Calling"
15:20.21PodMan99ai have a box at home this is what I am playing with... i just need to know how to register with a queue....
15:20.44PodMan99ahave x-lite... far from the best but its good to play with
15:22.02[TK]D-FenderPodMan99a: Keep reading, and pay attention to your dialplan, you've made a couple or really silly mistakes in there.
15:22.06*** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net)
15:22.24PodMan99ais it the inbound number bit ..?
15:24.53nestArPodMan99a: Registering with a queue, there's a couple different ways. I don't use pre-defined agents, I use AddQueueMember in my dialplan
15:28.11PodMan99anot as easy as that... any url's ??
15:29.05coppice[TK]D-Fender: Pay attention to your dialplan? When your dialplan starts talking to you, take a rest
15:29.33PodMan99alol
15:29.36[TK]D-Fendercoppice: You're just jealous because the voices only talk to ME! :p
15:30.05PodMan99a[TK]D-Fender : they talk to me... but falling pixies are not very useful when programing asterix
15:30.23[TK]D-FenderPodMan99a: Ask for some of their magic dust.
15:30.41PodMan99a[TK]D-Fender: apparently is good to sniff
15:30.45coppiceasterisk only works when you apply pixie dust
15:33.04*** join/#asterisk Cyon (n=cyon@216.179.31.170)
15:33.26*** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com)
15:33.41iratikAlright anyone familiar with freepbx?
15:34.00PodMan99a[TK]D-Fender: what url's would be good for this ... or what keywords would i enter in google... i have tried "asterisk dialplan config"
15:34.03Darthclueiratik, please go to #freepbx for help with that
15:34.15iratiksigh...... i'm not really sure if its a freepbx problem
15:35.37nestArPodMan99a: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember
15:35.38PBXXis here someone ho now something about hylafax ?
15:35.57coppicenope. nobody ever used hylafax
15:36.06Mw3i know it is a faxing software
15:37.30TelemacI've "[settings]\ sippeers => odbc.ast_cnf.sip_conf" in my extconfig.conf, so what could prevent "realtime load sipusers name _user" from finding matching row within Asterisk console ?
15:37.35Darthcluewasn't hylafax the boss on level 4 of asterisk the game?
15:37.46_x86_anyone have any recommendations for CSU's?
15:38.11_x86_need a CSU for a voice (CAS) T1, need rj48 both in from the smart jack and out to the PBX
15:39.48[TK]D-FenderDarthclue: In Soviet Russia, fax sends YOU.
15:39.54nestArlol
15:40.26Darthclue[TK]D-Fender, if it sends me someplace warm, sunny, and quiet then I'm all for it.
15:41.10asdx[TK]D-Fender: lol slashdot culture :p
15:41.35*** join/#asterisk IgorG (i=FeedomPa@host-90-188-188-119.pppoe.omsknet.ru)
15:42.09[TK]D-Fenderiratik: And what would that be?
15:43.37iratikwell my provider gave me the following settings for "extensions.conf" but when i look at extensions conf ... i'm not so sure where to put the settings
15:43.39nestAri am so bored so early.
15:43.48iratikexten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@icall)
15:43.49nestAralready replaced my furnace filter
15:43.50iratik[icall_in]
15:43.51iratikexten => 4175534249,1,Answer
15:43.58iratikand i don't know where to put that stuff
15:44.50[TK]D-Fenderiratik: Yeah, thats dialplan BS and FreePBX's problem.
15:44.53[TK]D-Fender~wglwat
15:44.54jbotit has been said that wglwat is well, good luck with all that
15:46.33jameswf~rtfm
15:46.34jbotfrom memory, rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM
15:46.36TelemacI really need some clue about extconfig.conf and odbc. I'm certainly doing something bad but I really don't know what and where ...
15:47.58Telemac(and I've rtfm :) )
15:48.18*** join/#asterisk dijungal (n=kdaniel@63.175.159.171)
15:48.43[TK]D-FenderTelemac: then try this :
15:48.46[TK]D-Fender~osmosis
15:48.47jboti guess osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
15:49.19Darthclue~book
15:49.20jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
15:50.17*** join/#asterisk pepo-- (n=pepOSX@190.78.220.149)
15:50.31PodMan99a[TK]D-Fender resolved ish... lol
15:50.46[TK]D-FenderPodMan99a: "ish".... slick.....
15:51.16puzzledtzafrir, have you also packaged oslec in your debs?
15:51.25PodMan99ai have to add the agents from the CLI
15:51.31iratikWhenever I make a call using any softphone i'm getting "Service Unavailable 503" ... what this might be a problem with?
15:52.31_x86_http://cgi.ebay.com/AdTran-T1-CSU-ACE-3G-1203022L1-AC-Power-tested-warranty_W0QQitemZ170167735456QQihZ007QQcategoryZ80226QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
15:52.38_x86_will this work for a voice CAS T1?
15:52.52*** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210)
15:54.12[TK]D-Fender_x86_: What do you want to do with your T1 exactly?
15:54.19[TK]D-Fender_x86_: And your link is dead
15:56.05_x86_http://cgi.ebay.com/AdTran-T1-CSU-ACE-3G-1203022L1-AC-Power-tested-warranty_W0QQitemZ170167735456QQihZ007QQcategoryZ80226QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
15:56.13_x86_works for me... just used it
15:56.48*** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla)
15:56.48*** mode/#asterisk [+o russellb] by ChanServ
15:57.09_x86_[TK]D-Fender: i want a test point between the smart jack and the asterisk PBX
15:57.13[TK]D-Fender_x86_: Didn't for me, but does now.
15:57.31tzangerhttp://spritesmods.com/?art=wcterror&f=had
15:57.33tzangerhahahahha
15:57.37[TK]D-Fender_x86_: What for?
15:57.44_x86_don't want to do anything crazy like split channels off, etc...
15:57.54_x86_[TK]D-Fender: by law in the US it's required
15:58.02_x86_to have a CSU before your DTE
15:58.30_x86_you're not allowed to run from smart jack to DTE directly, you need a CSU in there
15:58.50coppiceby law? that's a strange thing to legislate. I can just imagine senators debating the merits of that :-)
15:58.53[TK]D-Fender_x86_: Ummmm..... CSU is integrated into Zaptel digital cards...
15:58.58_x86_i had one from the 80's and it just died, so i need to replace it
15:59.06_x86_[TK]D-Fender: perhaps digium cards, sure... i use sangoma
15:59.22coppicepretty much any modern card integrates the CSU
15:59.28[TK]D-Fender_x86_: Same bloody thing... uses the same Xilinx chip even IIRC.
15:59.48coppicethe CSU is in the framer, not the Xilink
16:00.05[TK]D-Fender_x86_: And the claim of need for equipement before either of these is ridiculous.
16:00.12[TK]D-Fendercoppice: Noted :)
16:00.41*** join/#asterisk pepo--- (n=pepOSX@190.72.153.45)
16:06.14*** join/#asterisk harpal (n=Harpal@124.125.255.223)
16:07.35destructureanybody done any testing on the accuracy of asterisk timers?  if I wait(30) how close to 30 will it be? particularly under load
16:07.40*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)
16:08.56*** join/#asterisk dhill (n=dhill@fog.mindcry.org)
16:09.18ai-adestructure: you could do it.
16:09.29dhillis there a way to edit asterisk voicemail?  For example, I want to get rid of the option to set the unavailable message..
16:09.53ai-adhill: i would like to completly edit the whole callflow of voicemail..
16:10.05dhillyea
16:10.10destructureai-a: thanks for the encouragement!
16:10.21ai-ago destructure.. go destructure.
16:10.51[TK]D-Fenderdhill: You have the source... get coding.
16:10.54dhillI use asterisk for customers not inner-office.  Customers do not understand unavailable and busy.  I'd rather just have one.. make it easy...
16:11.17ai-aFender: is the voicemail possible to reproduce in an ivr ?
16:11.47[TK]D-Fenderai-a: Sure, been done before.
16:12.20ai-aor, better still more control over the vm from within the config file ;)
16:12.31[TK]D-Fenderai-a: Sure... see my answer for dhill above :p
16:12.36ai-aHeh :p.
16:13.28*** join/#asterisk muiro (n=dan@host-69-48-104-2.akr.choiceone.net)
16:14.44muiroquestion: What I'm trying to do is allow currency entry. I want to match any number of numbers, then a star, then two numbers the pound
16:14.56*** join/#asterisk alvariux (n=alvaro@189.158.163.157)
16:15.03alvariuxhello
16:15.11muiro_X.*XX# however isn't working
16:15.21[TK]D-Fendermuiro: All doable in an IVR.
16:15.27muiroit seems like asterisk doesn't want to match the * or the #
16:15.34[TK]D-Fendermuiro: And NO, you can't give a pattern like that.
16:15.57[TK]D-Fendermuiro: you'll have to collect each digit and parse the heck out of it.
16:16.06[TK]D-Fendermuiro: Not even that complicated actually...
16:16.09coppicehe can do anything he wants
16:16.16[TK]D-Fendermuiro: More like parse a bit ;)
16:16.17ai-aor write agi script :)
16:16.38muiroyeah, I was really hoping just to do it with pattern matching but I guess parsing it is
16:16.47muiroI got the reading out of currency amounts working smoothly, btw
16:16.56muiroit's just the entry that's a bit loopy
16:17.03alvariuxhi, somebody uses phpagi?
16:17.04[TK]D-Fendermuiro: only about 5 lines of dialplan :)
16:17.10muiroexactly
16:17.39muiroif it weren't for users I could leave it alone as it stands, but that wouldn't work
16:17.56puzzledanyone have a recommendation which function I could use best with 1.4 to see if the sipheader(to) contains a fqdn or ip address?
16:18.17muiroI mean, the prompt says input it this way, that's the way you do it. Users won't, though.
16:19.14muirothey're like drugs, really. Users.
16:19.20muiroYou need them but you hate them.
16:19.26[TK]D-Fendermuiro: What I'd advise is to tell them to entire the entire amount WIOUT a decmil indicator and dvide out the last 2 digits. (/100)
16:19.40[TK]D-Fendermuiro: dead easy that way and a no-brainer.
16:19.45muiroyeah
16:20.13muirolike an atm would do it
16:20.19[TK]D-Fendermuiro: Sure.
16:20.30[TK]D-Fendermuiro: why complicate things for morons?  Then jsut read it back to confirm.
16:20.35muiroI hesitated to do that initially because it may be confusing without that visual decimal mark
16:20.51[TK]D-Fendermuiro: VISUAL?!  We're talking about a phone IVR here.
16:20.52muiroor, at least, me being a visual thinker, I thought it might be confusing
16:21.09muiroyes, that's why I avoided that, because there was no visual thing
16:21.14muiro"like an atm", as it were
16:22.46muiroI had remembered dialing into an ivr to make a tax payment for a company I was working for one time and I remember entering the decimal as *, I liked that. I was hoping to "force" it using the matching but I guess I'll just parse it.
16:22.57muirono big deal
16:23.20*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
16:24.35[TK]D-Fendermuiro: you can, but you'd have to code about 1-2 dozen lines of IVR to supoprt this.
16:24.37muiro[TK]D-Fender: thanks for the advice
16:24.55[TK]D-FenderDarthclue: load res.psychic.so :p
16:25.18DarthclueI tried.  It kept saying, failure to connect, no signal.
16:25.34muiromore like no carrier...
16:27.01coppicetry www.spiritdsp.com
16:30.18*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:30.30*** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro)
16:30.47_x86_[TK]D-Fender: did you say anything on my CSU?
16:31.08[TK]D-Fender_x86_: Yeah I said you don't need one :)
16:31.22*** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net)
16:31.27puzzleddoes voicemail also require ztdummy? I always forget when ztdummy is needed
16:31.45*** join/#asterisk dswillia (n=me2@199.3.247.34)
16:31.46*** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
16:31.50Mw3puzzled: meetme, iax trunking
16:31.58puzzledta
16:32.14deeperroris there a dialplan variable that tells me the sip username that is making a call?
16:32.25*** join/#asterisk JayTee52 (n=jforde05@207-67-84-181.static.twtelecom.net)
16:32.44JayTee52hello all
16:32.58[TK]D-Fenderdeeperror: if it matched a sip.conf entry it'd be in the channel name...
16:33.33dswilliahey all is there a way to tell asterisk to compile with zaptel and libpri support.  I have d/led the latest zap, libpri, and Asterisk.  Compiled both lib, then zap.  When I compile asterisk "make install" it works no errors, but when i connect to the cli I have no options for zap or pri
16:33.37puzzledhow about ${SIP_HEADER(From)}
16:34.02puzzleddswillia, first compile & install zaptel, then libpri, then asterisk
16:34.31dswilliawhat is the best way to "uninstall" asterisk?
16:34.57puzzleddunno. did you try # make uninstall ?
16:35.27dswilliawow thank you
16:35.36dswilliathat was painless
16:35.45puzzledlucky guess :)
16:36.35muiroalso dswillia make sure zap is running
16:37.58deeperrorpuzzled: that did it for me thanks....fender: the channel variable didn't seem to work for my application not sure what the difference was
16:38.32[TK]D-Fenderdeeperror: clarify "didn't work"
16:38.32Kandinskyin zapata.conf, can i use multiple contexts if i want to use all the channels in every context?
16:38.52[TK]D-FenderKandinsky: that makes no sense.
16:39.04Kandinskywhy?
16:39.29Kandinskyi want to have both incoming and outgoing calls on all the channels
16:39.36[TK]D-FenderKandinsky: You do not use "zaptel channels" in DIALPLAN contexts.  In the dialplan you USE zaptel channels, either by fixed number, or by a GROUP taht you make them a member of.
16:39.49deeperrorfender: was using it for a name in mixmonitor and the recording would not save probably the slash or something?
16:40.20[TK]D-FenderKandinsky: the context you set in zapata is where * will send INCOMING calls from that channel. This can only be a SINGLE place.
16:40.34Kandinskyaha
16:40.39[TK]D-Fenderdeeperror: "show function CUT" <----
16:41.01Kandinskyso all my channels can be only in the incoming context
16:41.20Kandinskyand when i want to dial out..i include the incoming context?
16:41.21[TK]D-FenderKandinsky: you are not getting it.
16:41.26Kandinskynope :(
16:41.32Darthclue~book
16:41.33jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
16:41.35*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
16:41.38[TK]D-FenderKandinsky: this ha NOTHING to do with OUTGOING CALLS.
16:41.58Kandinskybut what are those channels in zapat.conf actually?
16:42.02Kandinskynot my isdn channels?
16:42.11Kandinskywhich i can use
16:42.22[TK]D-FenderKandinsky: You can have 1,000,000 contexts in extensions.cfon that each have extens that DIAL out your zaptel channels, but that has nothing to do with how * handles INCOMING calls
16:42.33Kandinskyi know that
16:42.47Kandinskyi am speaking strictly on the []  in the zapata.conf
16:43.13[TK]D-FenderKandinsky: there are no [] <- (context looking things) in zapata.conf
16:43.21dswilliahow would i check to see if zap is running?
16:43.25deeperrorfender: thanks for CUT
16:43.25[TK]D-FenderKandinsky: that isn't how channels are defined.
16:43.30Kandinskycontext=incoming
16:43.30Kandinskychannel => 1-2
16:43.30Kandinskychannel => 4-5
16:43.34Kandinskyso this is wrong?
16:43.43Kandinskyif i have this in zapata.conf
16:45.00Kandinskydo you happen to know any good documentation for zapata.conf when using BRI isdn with hfc-s chipsets?
16:45.13Kandinskyi understand the contexts in asterisk
16:45.23Kandinskybut not sure about the zapata thing
16:45.32Kandinskyand the link between the 2
16:46.02*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
16:46.02*** mode/#asterisk [+o anthm] by ChanServ
16:46.33Kandinsky[TK]D-Fender: u still there?
16:46.48*** part/#asterisk harpal (n=Harpal@124.125.255.223)
16:47.00[TK]D-FenderKandinsky: pastebin your zapata
16:47.02[TK]D-Fender~pb
16:47.02jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
16:49.33*** join/#asterisk bantu (n=Miranda@p54A32965.dip0.t-ipconnect.de)
16:50.00Kandinskyhttp://pastebin.com/d57674b4c
16:50.35Kandinskyi left some comments from my previous atempts
16:50.58Kandinskyso ..i have 4 isdn channels
16:51.00[TK]D-FenderYou have 4 channels defined and no groups.  All incoming calls on those channels go to the same context
16:51.09Kandinskyyes
16:51.18Kandinskytell me more about these groups pls
16:51.51Kandinskywhat they are good at and how to use them
16:52.16Kandinskywhat i want..is to have all the 4 channels available for calling or being called
16:53.14[TK]D-FenderKandinsky: You can group your channels together so that you can have * automatically choose a free channel to call out of when placing an outgoing call
16:53.40Kandinskyso i group all my channels in group 1 for instance
16:54.05Kandinskybecause that coresponds to what i want
16:54.06Kandinskyno?
16:54.51[TK]D-FenderKandinsky: Currently they are NOT grouped.
16:54.57Kandinskyi know :)
16:55.06Kandinskyi mean that is what i must do
16:55.14[TK]D-FenderKandinsky: But that is what I WOULD do typically if all of your channels can be treated as equal.
16:55.35Kandinskyok..solved that
16:55.54Kandinskyand what is the order?
16:55.56*** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net)
16:55.57Kandinskyif it matters
16:56.04Kandinskygroup ... context..channels?
16:58.13[TK]D-FenderKandinsky: if you give them all the same group membership you can dial out the group like Dial(Zap/g1/1234567890)
16:58.31[TK]D-FenderKandinsky: this has nothing to do with CONTEXT, we are only talking about GROUP right now
16:58.31Kandinskyyeah..i know that
16:58.48Kandinskybut i am talknig abouty the syntax in zapata.conf
16:59.01Kandinskyi first must have a group
16:59.04Kandinskythen a context
16:59.11Kandinskyand then the channels defined?
16:59.44[TK]D-FenderKandinsky: you already have a context set and you'll probably want to handle all incoming calls the same way, so thats fine.  what you need to do is simply set the GROUP
16:59.55Kandinskyok
17:00.09Kandinskyi was talking about a general case
17:00.15Kandinskyi just set my group
17:00.30[TK]D-FenderKandinsky: what is "a general case"?
17:00.42Kandinskyaaa.....common case
17:00.50Kandinskynever mind
17:01.03*** join/#asterisk irule (n=irule@200.53.61.4)
17:01.06Kandinskyi want to understand the logic in zapata.conf
17:01.22Kandinskyso.. in group 1 i have all my 4 channels
17:01.41Kandinskyand when incoming calls come..they will go to my incoming context in extensions.conf
17:01.43Kandinskyright?
17:01.46*** join/#asterisk rpm (n=russell@75.153.47.179)
17:01.52[TK]D-FenderKandinsky: Correct
17:01.57Kandinsky:)
17:02.08Kandinskyok...done with zapata.conf?
17:02.18Kandinskyanything else i should consider adding here?
17:02.20[TK]D-FenderKandinsky: and to pick any free channel to dial OUT of you'd use a Dial command like I showed you earlier
17:02.27Kandinskyyes
17:02.33Kandinskynow
17:02.43[TK]D-FenderKandinsky: Nope, thats it if this is how you want to handle your calls.
17:03.04Kandinskyok...other stuff now :)
17:03.08Kandinskyextensions.conf
17:03.31Kandinskyin the incoming context
17:03.47Kandinskyi must defin where all my calls will ring
17:03.52Kandinskycorrect?
17:04.17Kandinskyi am having a major problem with my incoming context
17:04.40Kandinskyi want to define 10 specific numbers (our phone office numbers)
17:05.02Kandinskyso that each user will have its own number
17:05.19Kandinskyso when someone is calling xxxx22
17:05.24Kandinskyit will ring user1
17:05.27Kandinskyso when someone is calling xxxx23
17:05.29Kandinskyit will ring user2
17:05.30Kandinskyetc
17:05.50Kandinskyuntil xxxx32 for example
17:06.01Kandinskynow...my problem
17:06.28Kandinskyif i set  exten => _X!22,1,Dial(SIP/user1,20)
17:06.39Kandinskyin the incoming context I am fine
17:06.48Kandinskybut i want to be more specific
17:06.55Kandinskyto define the whole number actually
17:07.12Kandinskyfor instance that xxxx22 is 111122
17:07.17Kandinskylocal number
17:07.33Kandinskybut if I type that..the phone dosen't ring
17:07.53Kandinskyand I don't know how asterisk actually receives the number dialed from the outside
17:08.20Kandinskyi tried to find out the number of digits * reads for 111122
17:09.14Kandinskyby setting _XXXXXX
17:09.24Kandinskyand it wouldn't ring
17:09.30Kandinskyand i kept on going
17:09.36Kandinskyincreasing the X number
17:09.43Kandinskybut no pattern would be fine
17:10.03Kandinskyany ideas how i can find the number asterisk is really trying to dial in my context?
17:10.09deeperrorI call 2 inbound did numbers from the same provider.  They both goto the same locations in my dialplan and one doesn't jump to options in the ivr the other one does.  I can see dtmf being identified in messages but it doesn't seem to make any actions.  This is running over iax protocol.  Any suggestions?
17:13.30deeperrorhere is what I see in msg's http://pastebin.ca/775538
17:14.46*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
17:16.30[TK]D-FenderKandinsky: You'll have to paste your dialplan and the CLI output of a failed call at verbose 10 so we can debug.
17:17.09Kandinskyok...thanks ...but i have to do something now...will be back in 15 min i guess
17:17.40*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
17:17.40*** mode/#asterisk [+o blitzrage] by ChanServ
17:19.18*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:22.10*** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net)
17:22.53davidcsihello all, question: I changed the listening port for the manager.conf.... how do i unload and load the manager for the port change to tale effect??
17:25.15davidcsianyone?
17:25.30deeperrorasterisk restart?
17:25.42davidcsidoesn't work
17:25.49davidcsireload doesn't work
17:26.29deeperrorasterisk -rx "restart now"
17:26.35*** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)
17:27.29rhombusThe Polycom phones can be told to reboot with a "sip notify check-cfg <registration>" on the Asterisk CLI
17:27.41rhombusis there a way to accomplish the same thing with the Aastra sets?
17:27.47rhombus(specifically, the 480i)
17:29.15davidcsideeperror: does that drops the calls?
17:29.37deeperroryes it will
17:29.45deeperrorasterisk -rx "restart when convenient"
17:29.54davidcsithen I can't do that
17:30.02davidcsiis there any other way?
17:30.05deeperrorno
17:30.19deeperrorhas to restart asterisk for that type of change
17:30.21*** join/#asterisk bantu (n=Miranda@p54A32965.dip0.t-ipconnect.de)
17:30.38davidcsido you know which module handles the manager?
17:30.55deeperrorthat i'm not sure
17:31.07deeperrorhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf
17:31.21deeperror"Simply reloading asterisk will not enable the manager. You must shut down asterisk and restart."
17:31.28*** join/#asterisk Assid (n=assid@unaffiliated/assid)
17:31.34Assidhey
17:31.42Assidanyone here have 2 fwd accounts on the same box
17:33.39*** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)
17:33.52De_Monqueues have members, what do you call the people on hold?
17:34.00De_Monqueue callers sounds gay
17:34.15coppicesuckers
17:34.25deeperrorHAHA
17:34.31De_Monyeah... thats not much better
17:34.43deeperrormember holders
17:34.57coppicelusers
17:35.10[TK]D-Fenderdeeperror: lol
17:35.12coppicefriends of the telco
17:35.24[TK]D-Fenderdeeperror: now THAT was gay
17:35.26De_Monqueue waiters
17:36.03coppicesheep
17:36.10deeperrorhaha
17:36.24De_Monall these silly answers tells me nobody has a serious answer
17:36.32deeperrorunless gender is taken into account of the member holder status
17:36.50[TK]D-FenderDe_Mon: Measn nobody bothered to invent a term for that specific scenario.
17:36.53De_Monmembers are the people the queue is calling
17:36.55Qwellthere we go - gender based queue priorities
17:36.57coppicei thought lusers and suckers were the serious answers
17:37.02davidcsideeperror: ok, thanks
17:37.06Assidfreaking odd. i registered for a new fwd account.. and that just refuses to register
17:37.31[TK]D-FenderAssid: And you're doing SO much to help yourself here too.... a shame...
17:37.32deeperrori'm hearing a lot about fwd accounts...does this give me a number?  or is it similar to grand central?
17:37.57Assid[TK]D-Fender hehe.. okay would like some help.. 1 account works. the other one doesnt..
17:38.09De_Monwell, luser is slang, and sucker is just, bad. You're fired!
17:38.14Assid[TK]D-Fender and yes i am using the right password on them
17:38.24De_Mondon't even bother asking about that last paycheck...
17:39.01De_MonI thought for sure the telephony world had these terms defined hundreds of years ago
17:43.50*** part/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net)
17:45.13rhombusDe_Mon: Just use callers and get over your silly discomfort about it
17:45.27rhombusDe_Mon: it says what it means; not everything has to be jargoned up
17:45.58rhombuscallers, agents -- end of discussion
17:46.12rhombusNow, my Aastra phones. Can I reboot them from the Asterisk CLI?
17:48.00*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
17:48.02*** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru)
17:50.01[TK]D-Fenderrhombus: look harder next time : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip_notify.conf
17:50.47rhombusUmpteen google searches didn't turn that up
17:51.20rhombushow much harder am I supposed to look? how much time am I supposed to waste? Now I know, and the next time somebody asks, I can answer also
17:51.24rhombusnm
17:51.40[TK]D-Fender~wikis
17:51.41jbotfrom memory, wikis is http://www.voip-info.org
17:51.45[TK]D-Fender^^^^^^^^^^
17:51.56rhombus[TK]D-Fender: thanks for being patronizing
17:52.02*** join/#asterisk Seldon75 (n=chatzill@69.77.161.3)
17:52.12rhombus[TK]D-Fender: I've been there... MANY times :)
17:52.24rhombustry this
17:52.30[TK]D-Fenderrhombus: rhombus And http://www.google.ca/search?hl=en&q=aastra+SIP+remote+reboot&btnG=Search&meta=
17:52.31rhombusdo a search, on google, for
17:52.40rhombus"reboot Aastra from Asterisk CLI"
17:52.41[TK]D-Fenderrhombus: was #5.  Like I said.... look harder
17:52.47rhombusand see what turns up
17:53.02Seldon75Hi, I want to write a cron-job that does a soft-hangup on any lines that are up for longer than 2 hours.
17:53.12Seldon75so how do I ask asterisk for a Zap channel's up-time?
17:53.57Seldon75...without going into the console
17:54.01[TK]D-FenderSeldon75: "show channel [channel]"
17:54.04Qwelltelepathy
17:54.20rhombus<PROTECTED>
17:54.25rhombusthat's a useless piece of advice
17:54.26rhombusthanks
17:54.43*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)
17:54.48[TK]D-Fenderrhombus: I wanted to make sure not to read how to reboot their ANALOG phones :p
17:54.57[TK]D-Fenderrhombus: Your Google-fu is weak...
17:54.57deeperrorseldon75:  why not just set a timeout for 2h then not worry about it
17:55.10*** join/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net)
17:55.21Seldon75deeperror: sounds good - would that be in zapata.conf?
17:55.49rhombusI think that "reboot Aastra from Asterisk CLI" is an adequate distinction that will protect me from instructions on how to reboot the ADSI sets
17:56.02ai-a[wrk]rhombus: why do you think its asterisk that performs this ?  Maybe its your telephone company that can reboot your phone ?
17:56.03rhombusanyway
17:56.12[TK]D-Fenderrhombus: thinking : yet another skill best left to trained professionals ;)
17:56.26deeperrorseldon: Set(TIMEOUT(absolute)=1800)    in dialplan
17:56.27rhombus[TK]D-Fender: are you always this arrogant?
17:56.28[TK]D-Fenderrhombus: Oh... and you're welcome :)
17:56.33ai-a[wrk]Heh.
17:56.34Assiderr is there a speach to text module in asterisk.. that lets you convert the voicemail into text and email it
17:56.43Seldon75how can I set a Zap timeout?
17:56.52rhombusIt's easier to be thankful to people with some largesse themselves
17:56.59Seldon75ok
17:57.10[TK]D-Fenderrhombus: Arrogant?  I was thinking more like witty, sarcastic, and jovial.
17:57.27Seldon75deeperror: will that work for i/c calls?
17:57.28coppiceAssid: you need app_magic_pixies for that
17:57.37Assidhaha
17:57.48deeperrorthat i'm not sure
17:57.50[TK]D-Fenderrhombus: I included smilies, pop-culture references and the solution!
17:57.51kmchenHi evrybody. Could someone help me to configure musiconhold for streaming radio ?
17:57.51Assidi keep telling this guy no such thing exists.. hes like no.. it does
17:58.03deeperrori use it when zap lines are making outbound calls thru sip/iax
17:58.34Seldon75deeperror: yeah I need it only to work for I/C calls...
17:58.36[TK]D-Fenderrhombus: All part of the service, please claim your KY discount coupon before you log out! ;)
17:58.39rhombus[TK]D-Fender: your memory is selective -- have a closer look at what you typed
17:58.48kmchenI configured musiconhold.conf with dir=/var/lib/asterisk/mohmp3-empty
17:58.48kmchenapplication=mpg123 -q -r 8000 -f 8192 -b 2048 -s --mono http://213.205.96.91:9915/
17:59.12kmchenbut total silence
17:59.15deeperrorjust place it in the dialplan before the calls you want to timeout
17:59.30[TK]D-Fenderrhombus: Trust me you'd know if I was seriously chewing someone out.  Heck the entire channel would.  Anyways, happy coding :)
17:59.33kmchenI did that
18:00.43Seldon75deeperror: so, like "exten => s,1,Set(TIMEOUT(absolute)=1800) " and then "exten => s,2,Goto(incoming,s,1)"...
18:00.44[TK]D-FenderSeldon75: just cron up a job to dump your channels and kill off the old ones.
18:01.12Seldon75[TK]D-Fender: so you would recommend the timeout option deeperror is suggesting?  why not?
18:01.31kmchendeeperror: exten => 1,1,SetMusicOnHold(native)
18:01.31kmchen;exten => t,1,WaitMusicOnHold(3)
18:01.31kmchenexten => t,2,Dial(SIP/keynux)
18:02.17*** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com)
18:02.40[TK]D-FenderSeldon75: Should work, but you'd have to add it to every originating exten.
18:03.18Kandinskyback
18:03.28Seldon75[TK]D-Fender: you mean to prevent the timeout from 'hanging around' once set for a particular call type
18:03.38Kandinsky[TK]D-Fender:   another thing i forgot
18:03.43Kandinskyabout zapata.conf
18:03.50[TK]D-FenderSeldon75: meaning you've have to insert that all over the place.
18:03.53Seldon75:q
18:03.59Kandinskyprilocaldialplan
18:04.05Kandinskydoes it set anything?
18:04.19[TK]D-FenderKandinsky: Not sure on this.  I don't do BRI
18:04.30Kandinskyok
18:04.48Kandinskybut do you know about international prefix and nationalprefix?
18:04.50Seldon75ok, so how can I execute a console command and get the output from within a cron-job?
18:05.30deeperrorasterisk -rx "your command"
18:05.34Seldon75thx
18:06.03Kobazis there any other info besids iax2 show registry to check the status of iax trunks
18:07.03Seldon75:qls
18:09.36[TK]D-FenderKobaz: "iax2 show peers"
18:10.58Kobazyeah and besids that one too
18:11.05Kobazeverything checks out find
18:11.06Kobazfine
18:11.20Kobazbut this voicepulse number isn't hitting asterisk
18:11.21[TK]D-FenderKobaz: What else could there possibly be?
18:11.25Kobazno idea
18:11.38Kobaziax soft phones can dial other iax trunks
18:11.45[TK]D-FenderKobaz: got the proper ports forwarded to *?
18:12.02Kobazthe box is set up as a DMZ
18:12.14[TK]D-FenderKobaz: And jsut because you're registered doesn't meant hat you haven't bungled up your DIALPLAN, etc....
18:12.27Kobazit's not related to dialplan
18:12.30[TK]D-FenderKobaz: enable iax2 debug and pastebin the CLI output of a failed call attempt
18:12.31Kobazit's not even getting that far
18:12.35Kobazit's not accepting the call
18:12.40Kobazit's not even getting the call
18:12.53[TK]D-FenderKobaz: dialplan failure CAN prevent you from getting a call.
18:12.59deeperrorVP numbers require a 1 in front of them as well on incoming
18:13.03Kobazverbosity is 6
18:13.05[TK]D-FenderKobaz: do do as I suggested.
18:13.14[TK]D-FenderKobaz: I said "iax2 debug", not "verbose"
18:13.17Kobaznothing happens when the number is dialed
18:13.19Kobazyeah
18:13.37Kobazoh
18:13.37Kobazhmm
18:13.39Kobazi did get stuff
18:13.39Kobazyay
18:14.04Kobazokay this helps
18:15.22[TK]D-FenderKobaz: You go quietly fix it now.  I suspect you are feeling rather silly now....
18:16.45Kandinsky[TK]D-Fender: can i see in asterisk wich outside number is trying to dial?
18:17.00Kobaz:(
18:17.04Kandinskywhich
18:17.05Kobazrather silly
18:17.12[TK]D-FenderKandinsky: ummmm reword that please... taht didn't add up./
18:17.39Kandinskyif someone on the outside is dialing a number
18:17.44Kandinskya customer for instance
18:17.55Kandinskyand i want to see what number asterisk receives
18:18.16deeperror{$EXTEN}
18:18.31Kandinskyto see if asterisk receives local coda+number or just number ...
18:18.36Kandinskyif u get the ideea
18:19.29[TK]D-FenderKandinsky: You get that number as the exten they have dialed.  thats what you're pattern-matching for.
18:19.44Kandinskyif i set exten => 101055,1,Dial(SIP/user,20)
18:20.00Kandinskyand someone from the same telephone network dials 101055
18:20.06Kandinskywithout the local code
18:20.06[TK]D-FenderKandinsky: in that example, you KNOW they dialed 101055 if that line is being executed
18:20.21Kandinskybut asterisk doesn't seem to know
18:20.27[TK]D-FenderKandinsky: So make a bunch of extens to match what you want.
18:20.29Kandinskybecause the phone isn't ringing
18:20.51Kandinskyit only works with something like _X!309055 for instance
18:20.52[TK]D-FenderKandinsky: enable debug and see what # * is receiving.
18:21.04Kandinskytips on how to do that pls
18:21.12Kandinskysorry
18:21.18Kandinskyit only works with something like _X!101055 for instance
18:21.19[TK]D-FenderKandinsky: And try things like using a more general pattern match as opposed to a hard-coded number like that
18:21.33deeperrorisn't this in the manual?
18:21.59Kandinskybut i need hard coded numbers :)  because i want to particularize
18:22.12[TK]D-FenderKandinsky: please re-read chapter 5 a few more times so you can see what kind of pattern matching best suits your needs and go TEST what the telco sends you.
18:22.14[TK]D-Fender~book
18:22.21jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
18:22.22[TK]D-Fender^^^^^^^^^^^^^^^
18:22.38Kandinskyok..that's what i'm reading AGAIN now
18:22.40*** join/#asterisk gabiru (n=gabiru@213.37.159.28.dyn.user.ono.com)
18:22.46Kandinskypls tell me about that debuging stuff
18:22.47[TK]D-FenderKandinsky: And realize you can have a call land on a more general place and the TEST for speciifc value to determine how to proceed
18:23.14Kandinskyto see how asterisk receives the number dialed
18:23.21[TK]D-FenderKandinsky: not sure how it works for BRI, but for pri it would be "pri debug span [span]"
18:23.54Kandinskyaha
18:23.54[TK]D-FenderKandinsky: but do a GENERAL pattern match like "_x." first and NoOp the ${EXTEN} to see what it looks like for a variety of calls.
18:24.41Kandinskyexten => _X.,1,.....???
18:25.08Kandinskycan u please tell me exactley how that should look like?
18:25.23[TK]D-FenderKandinsky: exten => _x.,1,NoOp(Incoming call from exten # "${EXTEN}")
18:25.29Kandinskyk
18:25.29Kandinsky10x
18:26.01Kandinskyand what exactley does that NoOp do?
18:26.01[TK]D-FenderKandinsky: Now stop and go back and reread the chapter about the dialplan, flow control, over and over and just TRY STUFF.
18:26.14Kandinskycan i see  some output?
18:26.29[TK]D-FenderKandinsky: It just prints a line in the * console so yuo can SEE something.  It doesnt' actually "DO" anythiner
18:26.30Kandinskyor that whould show if i have debug turned on?
18:26.33[TK]D-Fenderanything*
18:26.34Kandinskya
18:26.35Kandinskyok
18:26.37Kandinskygot it
18:26.46[TK]D-FenderKandinsky: You'll see it iver you're at verbose 3 or higher IIRC
18:26.58[TK]D-FenderKandinsky: Now go test, go read, then test some more.
18:27.06Kandinsky:)
18:27.48*** join/#asterisk Dovid (n=Dovid@bzq-88-153-142-81.red.bezeqint.net)
18:27.52*** part/#asterisk pepo--- (n=pepOSX@190.72.153.45)
18:28.00Dovidanyone here set up BLF on a snom 360 before /
18:28.02Dovid?*
18:39.40*** join/#asterisk bantu (n=Miranda@p54A316EC.dip0.t-ipconnect.de)
18:51.10Kandinsky[TK]D-Fender: I have another problem/bug :) If i restart Asterisk, the VoIP phones won't dial anything SIP related, only isdn numbers
18:51.20Kandinskyfor a minute or so
18:51.50[TK]D-FenderKandinsky: maybe becasue the phones haven't all reconnected with * because of the restart.
18:51.55KandinskyI have asterisk running as user/group
18:52.05Kandinskybut when i had asterisk running on root account
18:52.09Kandinskythere was no problem
18:52.14Kandinskyany ideas?
18:52.26Kandinskybecause it doesn't seem to be phone-related
18:58.48*** join/#asterisk Strom_M (n=strom@208.127.172.112)
18:59.12hmmhesayswhen did gmail jump to 5 gig
19:00.22*** join/#asterisk joeballsonya (n=joeballs@adsl-69-237-115-101.dsl.scrm01.pacbell.net)
19:01.15J4k3who cares
19:02.57J4k3whats 5 gigs now, $1.70 worth of hard drive space?
19:03.31*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
19:03.34J4k3yep, taking the price I saw for 1TB SATA HDs yesterday ($329) that works out to right at $1.70
19:04.17*** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41)
19:04.44J4k3you let me handle your personal info, like email, and I'll be happy to give you a lot more than 5 ghz...  ;)
19:04.47J4k3err
19:04.50J4k35 gb
19:05.47l0verb0yhey hows everyone doing
19:08.49[TK]D-Fenderl0verb0y: jUST WORKING FOR THE WEEK-END....
19:08.56*** join/#asterisk ghento (n=ghento@75.155.241.7)
19:09.11*** join/#asterisk Strom_M (n=strom@208.127.172.112)
19:09.29l0verb0yohhh
19:09.59[TK]D-FenderI R FUNNEH
19:12.36l0verb0ydoes anyone know a good forumla to figure out how many callers a machine can hold before the call quality starts going down
19:13.18[TK]D-Fenderl0verb0y: usually more a factor of bandwidth
19:13.38l0verb0yassuming bandwidth was infinite
19:13.39[TK]D-Fenderl0verb0y: Whats you server have and what about your clients?
19:14.02[TK]D-Fenderl0verb0y: Pretty big.  What kind of volume do you have in mind?  And transcoding, etc?
19:14.26l0verb0yg729 codec, pentium 4 3.0
19:14.46Darthcluehmmhesays, recently.  it's been increasing steadily over time.
19:15.37coppicewell, if you have infinite bandwidth, G.729 is a poor choice :-)
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19:15.55*** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue)
19:16.18l0verb0ycoppice: if you had a pentium 4 3.0 with inf bandwidth what codec would you use and how many callers could you handle
19:16.21l0verb0yoh also 1gb of ram
19:16.24[TK]D-Fenderl0verb0y: indeed, and you did not actually answer my question.
19:16.49l0verb0ysorry
19:17.14coppicewith infinite bandwidth, I'd use 48k samples/s PCM, 24 bits. probably stereo for good conferencing
19:17.54[TK]D-Fendercoppice: 96k with full EAX ;)
19:18.21[TK]D-Fendercoppice: So we can still give them that "trapped in a tin can" feeling ;)
19:18.31*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
19:18.32coppiceyeah, surround is good for conferencing. might as well throw in HD video too
19:18.48[TK]D-Fendercoppice: Now with extra crack!
19:21.05l0verb0ywhats the best way to test the number of concurrent calls?
19:23.46[TK]D-Fenderl0verb0y: "show channels concise"
19:23.56*** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
19:24.55*** join/#asterisk |omni| (n=rob@mail.argus-search.com)
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19:30.19De_MonI wonder if anyone has created a dialplan > flowchart script yet
19:30.28dijungalhello.. is there a channel variable that will show me who completed the call COMPLETEAGENT or COMPLETECALLER?
19:31.01De_Monhmm
19:31.13l0verb0ythanks
19:31.22De_Mondijungal how is the call being made?
19:31.36dijungalDial
19:31.59dijungalDial(SIP/${EXTEN}@provider)
19:32.20dijungalDial(SIP/${EXTEN}@provider|g)
19:33.32dijungali'm trying to do my own queue_log entry after the dial cmd
19:33.44De_MonDIALSTATUS will be available on each side of the call leg (both channels)... maybe they will be different, worth looking into
19:36.31dijungal??
19:37.24dijungalhow do you separate the DIALSTATUS on diff. side/leg of the call?
19:37.40dijungali thought the DIALSTATUS was the status of the dial command
19:38.13[TK]D-Fenderdijungal: it is, and De_Mon is just a little off on that.
19:38.29[TK]D-Fenderdijungal: You'll have to account for "h" as well.
19:38.54dijungalk.. i was getting woried for a moment there.... i was about to say that Asterisk Bootcamp course tutor lied to me!.. lol
19:39.14[TK]D-Fenderdijungal: "g" means COMPLETECALLER, "h" would be "COMPLETEAGENT"
19:39.34dijungalwhere do i get the g and h ?
19:41.16[TK]D-Fenderdijungal: "g" for the dial options, "h" for the Asterisk Standard Extension.  And if you have any trouble folloing this thought, then your ABC Tutor should kick you in the ass :p
19:41.51dijungallol... ok i remember those now :)
19:41.54dijungalnon need for that kick
19:43.43[TK]D-Fenderdijungal: first one's free!
19:44.02dijungal[TK]D-Fender:  stop the violence ....
19:44.20[TK]D-Fenderdijungal: Ok, Homicide it is!
19:44.43dijungal:-/
19:44.49dijungal:-!
19:47.06*** join/#asterisk Trionnis (n=blah@209.201.67.250)
19:49.39Trionniscan someone point me toward some resources about the res_snmp module for 1.4.x ?  I can't seem to get it to compile, even though I have all of the net-snmp packages and other related crap installed on the system.
19:53.20*** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com)
19:56.26*** join/#asterisk DataCompBoy (n=datacomp@213.187.250.34)
19:56.57dijungalu can use the 'g' parameter on the dial() command to execute more commands in the current context if the called part hangs up. what if the calling party hangsup?
19:57.08dijungalis there a parameter for this?
19:57.46DataCompBoyHi all! Have anybody reach problem, when dialplan correctly handle DTMF, but FastAGI script using GetData don't see DTMF on SIP from SoftPhone :(
19:57.57[TK]D-Fenderdijungal: Didn't we just go over this?!
19:58.05dijungalnope
19:58.12[TK]D-Fenderdijungal: "h"!@!@
19:58.17dijungalexplain again
19:58.25[TK]D-Fender~osmosis
19:58.25jbotfrom memory, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ...  or at least until your unconsciousness restores peace to the channel ...
19:58.26[TK]D-Fender^^^^^^^^^^^^^^
19:58.37Trionnislmao
19:58.48Trionnisyou just got served
19:58.55[TK]D-Fenderdijungal: When YOU hangup, the "h" ASTERISK STANDARD EXTENSIONS gets called.  Use your imagination.....
19:59.00Trionnisdon't feel bad, I get it about once a week from him
19:59.03Trionnis^^
19:59.07Assidanyone have ipkall forwarding the calls directly to your box and not via FWD ?
19:59.08[TK]D-FenderTrionnis: Hardly!
19:59.17Assidi dont see the call hit my box
19:59.37Assidi have verbose on.. and still dont see any error regarding this
19:59.48[TK]D-FenderAssid: [general] allowguest=yes , context=somewhere . Enable SIP debug, and test.
19:59.51destructureai-a[wrk]: the timing seems pretty good actually
19:59.53dijungal[TK]D-Fender: when the called part or the calling party hangs up? that's what i'm trying to determine...
19:59.54[TK]D-FenderAssid: SIP DEBUG <------
20:00.26[TK]D-Fenderdijungal: "g" accounts for when the CALLEE hangs up.
20:00.37ai-a[wrk]destructure: does that mean your closing the Accuate Timing Testing Department ?
20:00.39dijungalk
20:00.42dijungalthnks
20:01.27Assid[TK]D-Fender  too many sip users.. you wouldnt happen to know by chance the ip used by ipkall ?
20:01.40[TK]D-FenderAssid: nop.
20:02.38destructureai-a[wrk]: nope, still have more to do, need to implement now.  It'd be nice if I could get better than 1s resolution from strftime, but 1s will do, as long as it's spot on.  I ran 1000 calls through this piii (call= get time, wait, record total to astdb).  And none of them missed.  the call spool was the slow part
20:04.35*** join/#asterisk fnordus (n=dnall@24.84.160.227)
20:06.38DataCompBoyAGI debug shows "AGI Rx << GET DATA ps/CN/default/appgreet 5000 1 -- <SIP/120001-08281da0> Playing 'ps/CN/default/appgreet' (language 'en') AGI Tx >> 200 result= (timeout)" while i press 1, 2, #, * etc -- no reaction.
20:07.54Trionniscan someone point me toward some resources about the res_snmp module for 1.4.x ?  I can't seem to get it to compile, even though I have all of the net-snmp packages and other related crap installed on the system.  "configure --with-netsnmp" says that it's not installed.  Help please!?
20:09.45Trionnisthe errors --> http://pastebin.ca/775948
20:10.32ai-a[wrk]checking for snmp_register_callback in -lnetsnmp... no
20:10.50GreggB[TK]D-Fender: A few days ago you were recommending Voxee, and I just got around to checking this page which claims they're effectively out of business: http://www.voip-info.org/wiki/view/Voxee
20:10.57Trionnisyes, I saw that
20:11.12[TK]D-FenderGreggB: No I certainly did not.
20:11.16TrionnisI know that's why it's failing, what I can't figure out is *why* it's failing :)
20:11.30ai-a[wrk]triiiple: open the configure file, and see how it detects it and question why its not detecting it.
20:11.31Trionnisthere seems to be very little documentation about res_snmp
20:13.34GreggB[TK]D-Fender: Hmm, alright I thought it was you, but sorry then... I could have sworn someone said that Voxee and Teliax were both working well for them.
20:13.58*** join/#asterisk saftsack (n=saftsack@pD9E0741C.dip.t-dialin.net)
20:15.08ai-a[wrk]GreggB: i cant see anything in the logs.
20:15.33*** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org)
20:18.05*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)
20:19.02Trionnisok, so somewhat related question... can anyone recommend a way to monitor the current number of calls on an asterisk machine without using the manager api ?
20:19.44*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
20:19.53De_MonTrionnis snmp
20:19.56Trionnis...
20:20.04Trionnis......
20:20.10ai-a[wrk]Heh.
20:20.29De_MonI suggest you get it working
20:21.14Trionnisyes, I'll recode the configure script... I'll get right on that
20:21.32Trionnisoh wait, I know jack shit about those... maybe that's not such a good idea :)
20:21.35De_Monyou assume its the configure script at fault
20:21.58ai-a[wrk]the configure script is telling you something is wrong on your box.
20:22.27De_Monfailing that you could always hire something that does know what they are doing ^_^ (not me)
20:22.36ai-a[wrk]"The Net-SNMP installation on this system appears to be broken."  i would be worried about why your snmp isnt working.
20:24.31asdxyay, i've got the teliax account now... should i talk with my customer about the price of work, etc, before i start working?
20:25.32De_Monasdx ... you should always have a contract before starting work stating exactly what your doing and how much he is going to play for it
20:25.50De_Monany changes to the contract means ammending the contract including how much he is going to pay...
20:26.00GreggB[TK]D-Fender: Alright, sorry for the confusion. It was teliax and VoicePulse that you stated were mostly decent back on the 6th.
20:26.21[TK]D-FenderGreggB: better.
20:26.21GreggBI take it Voxee is either failing or out of business, despite the fact they'll still take your money on the website, and that they still "rank" in the top 10% on www.myvoipprovider.com  Sigh...  Is anyone using them right now?
20:27.06asdxDe_Mon: i see, but... the problem is that i am a noob, and i still have to learn some things and my customer ask me "how much time will it take" and i don't know how much time will take for me to learn/implement that, but i really want to do it
20:27.38asdxDe_Mon: i guess is not much work and i'll learn fast... i usually learn fast.
20:28.12De_Monits better to under promise and over deliver than the other way around
20:28.15[TK]D-Fenderasdx: You are clearly in way over your head and should not even be offering * services.
20:28.49asdxyeah, i should learn first
20:28.50De_Monare you even old enough to work at mcdonalds?
20:29.37J4k3asdx: you need to be trixbox certified!  its only $5k USD!
20:29.40Assid[TK]D-Fender : i got ipkall to connect to me.. now im getting   Looking for external-shahbc in inbound (domain domain.name.here)
20:29.41J4k3(NOT)
20:29.43Trionnislmao
20:29.58asdxJ4k3: trixbox? why that? i prefer cli
20:30.09TrionnisTrixbox certified.... the MCSE of the Asterisk world
20:30.12asdxDe_Mon: i'm 24
20:30.23asdxM$ sucks
20:30.32rob0Two dozen.
20:30.35De_Monjust wow
20:30.41rob042 inverted
20:31.02J4k3asdx: once you get past 'what sucks' you might learn something.
20:31.10asdxwell, i'll do this for free... and i'll use this experience to learn.
20:31.10J4k3but, when I was 24 my anger was a lot less focused, I understand.
20:32.02asdxJ4k3: anger?
20:32.09J4k3today I'm quite a bit more upset about, say, scorching my lunch noodles, than I am pointing at some 'successful' company and saying they suck.
20:32.16[TK]D-FenderJ4k3: Time to let go of anger and move on to full blown psychosis! ;)
20:32.26asdxJ4k3: they suck for many reasons.
20:33.08J4k3asdx: don't knock the dopeman's hustle.
20:33.16J4k3and thats pretty much my opinion of Microsoft... ;)
20:33.19*** join/#asterisk ghento (n=ghento@75.155.241.7)
20:33.37De_Monmy ghetto lingo isn't up to par with that statement
20:33.46J4k3if you're angry at microsoft, you need to re-evaluate and figure out a way to profit off the situation.
20:34.02*** join/#asterisk fnordus (n=dnall@24.84.160.227)
20:34.08De_Monoh, okay
20:34.15J4k3Microsoft sells crappy products that screw people up to always being Microsoft-users.  No different than dope.
20:34.25Trionnisor Cisco
20:34.27Assidanyone know where "425" is used ?
20:34.31Assidas in the area code
20:34.35rob0WA
20:34.37Trionnisum.. after 424?
20:34.41J4k3but, when the local dope dealer drives through in a $100k car, its hard to deny the business plan's workability.
20:34.42De_Monyou should try google
20:35.29Assidyeah.. was referring to the ipkall numbers :P
20:35.51rob0http://www.nanpa.com/area_codes/
20:36.10De_MonJ4k3 bad business practices does not a good business make. Evil always has its hour, day, year... but will eventually reap what they sow
20:36.23Trionniswhoa... that's deep
20:37.06asdx[TK]D-Fender: this guy (customer) told me that he will give me some time to learn
20:37.39[TK]D-Fenderasdx: desperate chump.  Sounds like a dangerous cutsomer.
20:37.50asdx[TK]D-Fender: i have my computer here and i compiled/installed asterisk, but i have no hardware and no voip provider...
20:38.03*** join/#asterisk bantu (n=Miranda@p54A316EC.dip0.t-ipconnect.de)
20:38.06asdx[TK]D-Fender: they are giving me a voip provider now for playing...
20:38.11[TK]D-Fenderasdx: 1 monkey short of a full set ;)
20:38.26J4k3so they're investing $1.50/month and 1.1 cents per minute into your success ;)
20:38.30J4k3cheap investment, I must say.
20:38.33De_Monrob0 where is the actual list of areacodes there?
20:38.39Assidwhats wrong with ipkall with free incoming?
20:38.53AssidJ4k3 ^
20:39.01asdxhmm
20:39.13De_Monomg an access database? kill me now
20:39.22Assidwhere where?
20:39.34J4k3ax-sex.
20:40.25AssidJ4k3 thats what it feels like once it crosses 28MB
20:41.07J4k3haha
20:41.28*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
20:41.31J4k3its acceptably bad on ethernet.
20:41.43*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
20:43.14De_MonAww they have a city/county lookup tool, I want that
20:47.03*** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60)
20:48.51hmmhesaysanyone using broadvoice with asterisk 1.4?
20:50.19*** join/#asterisk slima (i=slima@unaffiliated/slima)
21:06.04Trionnisnot if they're smart
21:06.16Trionnisbroadvoice is an evil, evil company
21:07.31J4k3I've found anything with 'broad' in it should be avoided.
21:07.41J4k3broadcom, broadwing, broadvoice, etc.
21:07.56J4k3I prefer chicks, avoid the broads.
21:09.15Trionnisbroadwing..... *growl*
21:09.33Trionnis"oh, here's your NI2 PRI"  "uh, you're passing 'plan: private' on blocked CID calls... that's not following the standard" "oh, well whatever, deal with it"
21:09.54Trionnisyeah, fuck Broadwing... fuck Broadwing sideways with a broomhandle...
21:10.33*** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com)
21:11.58*** join/#asterisk BBHoss (n=hoss@146.229.191.76)
21:12.13[TK]D-FenderJ4k3: Broadway has made me a fortune on Monopoly ;)
21:13.14Qwell[TK]D-Fender: it's really Park Place
21:13.21QwellBroadway is just a freeloader
21:13.27*** join/#asterisk kingsob (n=derek@TOROON01-1177844188.sdsl.bell.ca)
21:13.29BBHossheh
21:13.43kingsobdoes anyone know anyhting about unlimitel?
21:14.19kingsobim wondering if its possible to get a number with the ability to have like 30 imbound calls as the same time
21:15.00[TK]D-Fenderkingsob: their base is 5 calls / DID, but you can call them up and arrange something bigger.
21:15.06[TK]D-Fenderkingsob: like VPRI, etc.
21:15.46[TK]D-Fenderkingsob: then again at that many channels, and for the hassle & risks you might be better of getting your own PRI...
21:16.17kingsobI kinda want to offer a service to my friends where they can call my number
21:16.22kingsoband make free outgoing calls
21:16.26kingsobto longdistance numbers
21:16.28*** join/#asterisk galeras (n=Martin@201.244.247.149)
21:16.36kingsobby dialing a local number
21:17.38kingsobhow much does a PRI cost?
21:18.44outtoluncmore than the 'free' you'll be charging your friends <G>
21:19.41outtoluncwant fries with that? <G>
21:19.43[TK]D-Fenderkingsob: Depends where you are and what provider you're looking at
21:20.02kingsobwell maybe not free, just a small charge to cover my costs
21:20.14BBHossare you in a metro area?
21:20.20kingsobyeah, toronto
21:20.28[TK]D-Fenderkingsob: at that rate they may as well just set themselves up with SkypeOut and be done with it.
21:20.43[TK]D-Fenderkingsob: And save yourself a boatload of $
21:20.49[TK]D-Fenderkingsob: And troubble
21:20.56BBHossim not sure about canada, probably anywhere from 300-600USD a month
21:20.58kingsobyeah, but u cant use skype to call a local number on ur cellphone
21:21.12lirakishow do i determine the "return code" from a carrier? .. so i can play a different sound if it is busy 17, or some thing else for another code?  is it stored in a chan var?
21:21.33kingsobthats cool, even at like 2000/mo in costs, i would only have to sell it to like 50 people @ 20/mo to cover costs
21:21.51J4k3100 people at 20/month
21:21.55J4k3makes $2000.
21:22.01kingsoblol yes, yes it does
21:22.20J4k3but
21:22.21kingsobbut still, that should be relatively easy
21:22.31J4k3a PRI in the GTA will run you max $250/month
21:22.38kingsobthe likely hood of them all calling in at exact same time is quite small
21:22.39J4k3but that'll require a colo to get a price like that
21:22.52kingsobcolo?
21:22.53J4k3and last time I paid sub-$250 for PRIs, a rack was $750/month
21:22.55J4k3colocation
21:23.07J4k3basically putting your gear at the switch, rather than them bringing a circuit to you.
21:23.10[TK]D-FenderJ4k3: 250$ for GTA?  Thats crazy low....
21:23.11Assiddamn. i wish freecall/voipstunt etc. allowed callerid
21:23.29Assidatleast they have free sms for a bit
21:23.32[TK]D-FenderJ4k3: Best I get here in Mtl its $650
21:23.54kingsobim just trying to come up with a buisness idea using asterisk
21:24.00J4k3[TK]D-Fender: $250 is crazy low anywhere, its just a matter of finding it.  hell, I was paying $210/month per PRI and like $0.10/DID/month to Focal
21:24.09kingsobever since i started playing around with it about a year ago, i;ve fell in love
21:24.22kingsobamazing what u can do with it
21:24.23J4k3of course the service went to shit, and they attempted billing me double, when they were bought out by broadwing (hence 'avoid anything with broad in its name')
21:24.45galerasHi Sirs, my client is reporting this weird behavior: Occasionally, same call is incoming for 2 diferent ZAP channels (i.e. 27 and 84). Any idea what can be happening?
21:24.46BBHossyeah pris are still damn high around here
21:25.05[TK]D-Fenderkingsob: Then open a phone-sex line like the rest of us.....
21:25.21[TK]D-Fenderkingsob: Leave termination out of it ;)
21:25.27kingsoblol
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21:26.34kingsobmaybe ill write an ai, and allow people to call in for phone sex with my computer voice, thatd be sexy
21:28.10J4k3get a PRI
21:28.16J4k3and some old Apple IIs
21:28.25J4k3and find a copy of diversi-dial
21:28.28BBHosshaha
21:28.43[TK]D-Fenderok, time to head home.  BBIAB
21:29.18creativxyou could probably earn millions on it
21:30.15BBHossyeah me and a friend actually thought of that, but we decided the computers still aren't powerful enough to UNDERSTAND voice 100%, much less speak fluently
21:31.37J4k3shit, I just remembered I can dial anywhere in the USA same as local on my cellphone, up to 14.4k
21:32.13BBHossheh
21:32.20BBHossuse the 10-228 trick to spoof your ANI :)
21:32.32BBHossor 10-10-228
21:33.15J4k3I did that, and ended up getting a $1k+ phone bill from metromedia a few months later :E
21:33.28QwellJ4k3: that's the whole use of that number
21:33.28BBHossfun
21:33.29Qwell:p
21:33.46J4k3yeah, I mowed a lot of yards... :/
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21:42.18fbntsHi, is there a way to log on an agent from a script?  I have looked at the CLI but it only allows you to logoff or show agents
21:44.23asdxdamn this customer is desperate for his pbx, but he doesn't want to pay me...
21:44.35asdxand he is pressuring me, i don't like this
21:45.18MaliutaI thought it was simple: no payee, no workee
21:45.55asdxMaliuta: i want to learn how to connect asterisk with a voip provider :-)
21:46.31asdxbut maybe i should get my own account for that...
21:46.43Maliutaasdx: you don't have to work for someone to do that
21:47.06asdxMaliuta: yeah
21:47.21Maliutaasdx: I am going to get to do some asterisk stuff in my new job, but all the stuff I have learnt has been on my own kit
21:48.06Maliutabeneficial to me too, given that I have been living 2000kms from most of my friends and my parents live in canada
21:48.35Maliuta$0.08 calls to any landline in .au, .uk, .ca .....
21:49.40asdxMaliuta: is there free voip providers that i can use?
21:49.50asdxMaliuta: for learning purposes
21:50.01Maliutathere fwd
21:50.17MaliutaI pay for my DID and any calls I make
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22:01.43jameswfso neat whena a node poops
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22:01.44Qwellthat had to have been more than a single node
22:01.45muirothat was 4 nodes by my count
22:01.45muiroright, I'll repeat my question now:
22:01.45muiroquestion: what do I need to modify to make saynumber, saydigits, etc. use background() instead of playback()?
22:02.08muiroif it's possible
22:02.43Qwellit's not
22:02.43Qwellor, rather, not currently
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22:02.51muirohrat
22:02.55muirodrat, even
22:03.00Maliutamuiro: just write your own macro to use background to play the files
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22:07.39muiroyeah, that's whait I'm going to do Maliuta, easy enough
22:09.19rob0This channel is registered users only, right?
22:09.27Qwellyes
22:09.46rob0So when I get dumped by a netsplit, my window is lost and I have to manually rejoin.
22:10.20rob0Have you guys considered just taking away voice from unregistered users?
22:11.06QwellI don't think there's a flag for that
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22:17.41[TK]D-Fender(S)Qwell(tched)!
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22:21.51Dovidis switchvox based on asterisk ?
22:24.26hmmhesaysthe mind is a terrible thing to waste
22:26.27hmmhesaysbroadvoice is a bunch of b1tches
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22:29.37Corydon76-vcchDovid: yes
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22:33.36teknoprepwhere is the asterfax config file?
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22:57.02killfillhi..
22:57.18killfilli cannot get something from my dialplan
22:57.22killfillhttp://pastebin.ca/776208
22:57.42killfillwhen i call from extension 45 to 31, the macro "macro-stdexten" gets executed
22:58.00killfillbut from 45 to 23, i get a plain Dial("IAX2/45-1", "SIP/23&IAX2/23")
22:58.05*** join/#asterisk johndbritton (n=john@cpe-74-70-255-158.nycap.res.rr.com)
22:58.07killfillwhy is it different?
22:58.44johndbrittonim getting a registration failed for xxx.xxx.xxx.xxx no matching peer found how can i make the peer match?
22:58.55johndbrittonim trying to connect from a hardphone
23:00.27killfilli have nothing specific to 23 or 31 in the whole dialplan.
23:00.40killfillwhy do i get different behaviour?..
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23:12.34killfilldamn.. :S
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23:19.28killfilloh..
23:19.29killfillwired..
23:19.37killfillits becouse one user has voicemail and the other not.
23:19.39killfillhm....
23:20.55killfillwhen the user has voicmail, the std-exten get called..
23:21.05killfillwhere is this controlled?.. how do i change this behaviour?
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23:34.27fujinkillfill: a local channel will run through the dialplans and guess how to best contact the dialee
23:35.02*** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au)
23:36.29Zuchmirhow can i tell if my digium card is working
23:36.30slowshuttif one has a asterisk box on a static ip linking it to the internet where can one find info on possible security risks?
23:38.38killfillfujin: ok thats fine. my problem is, that i need to execute something before a phone gets ring. so i was putting things in stdext macro. This works. but for users that has voicmails...
23:39.04killfillfor users that doesnt (i.e. users that are part of a Queue), how would i catch them?
23:39.13fujinqueue is entirely different
23:39.30fujinthe only way I've been able to specifically do things for queue members, is to write a dynamic queue member system
23:39.32fujinwith a Local channel
23:39.44fujin(I did it in AEL, was probably 2-3 hours work to get it perfect)
23:39.47killfillhm...
23:41.10killfillfujin: is it possible that a user has voicmail, and he can also exist in a queue as agent?. currently, im taking hes voicmail out, becouse when the queue calls him, and is busy, voicmail apears, and not the next agent..
23:43.49*** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net)
23:44.03asdxi already got a ITSP (teliax) and i have a server with asterisk already... i want to connect to my asterisk and then make outgoing calls
23:44.15*** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net)
23:44.30asdxteliax sent to me a user configuration with:
23:44.30slowshuttyes asdx are you using sip or iax2?
23:44.32asdxhost=voip-co4.teliax.com
23:44.36asdxslowshutt: yes, iax2
23:44.53slowshuttcan you not connect?
23:45.03asdxslowshutt: i can connect if i set host=dynamic
23:45.19asdxslowshutt: but that host=voip-co4.teliax.com confuses me
23:45.20Zuchmiri have 1 FXO, i plugged it in, followed the books "minimal script" (changing 2 to 1) and when i call in, it does not answer
23:46.26slowshuttjust ask them to add your ip address to their iax.conf then ip will work they keep theirs dynamic because they don't lnow where you connecting from
23:47.08johndbrittonive got a nat problem i think... ive got a hardphone on my internal (home) network conncected via internet to my asterisk box (on the net with public ip) I am able to place a call from my hardphone, and i can hear the person im calling but they cannot hear me... any suggestions
23:47.23slowshuttthant means you are connectin to there ip hostname =voip-co4.teliax.com
23:48.01asdxslowshutt: so i need two [users] entries?
23:48.29slowshuttpaste your iax.conf and i will have a look remember to xxxxx out your password
23:48.29slowshuttno
23:48.32slowshuttjust one
23:48.58*** join/#asterisk stubert (n=stu@techtools.actusa.net)
23:49.25slowshuttiax is a bothway connection you set up your iax to connect to them and visa versa
23:51.09asdxslowshutt: http://pastebin.ca/776277
23:51.54slowshuttasdx link not active
23:51.56asdxslowshutt: do i need another host so my client here in my computer will be able to connect?
23:52.37slowshuttpastebin link dead
23:52.42asdxwait
23:53.11*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:53.53slowshuttuse pastebin.com
23:54.20asdxhttp://pastebin.com/m785a5959
23:55.46slowshuttyou dont need the register string
23:56.19asdxok
23:56.38slowshuttwhat do you get if you type iax2 show peers from asterisk console?
23:57.21johndbrittonthe person im calling from my sip hardphone cant hear me, but i can hear them
23:57.56asdxslowshutt: diego            63.211.239.2    (S)  255.255.255.255  4569          Unmonitored
23:58.05slowshuttadd username=diego and secret=foobar i your diego context in iax.conf
23:58.23*** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org)
23:58.45asdxok
23:59.23asdxdone
23:59.50slowshuttwhat do use to dial use the line for outgoing
23:59.58slowshutti presume you have a sip phone

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