IRC log for #asterisk on 20120409

00:31.31*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
00:31.31*** mode/#asterisk [+o mjordan] by ChanServ
00:58.46*** join/#asterisk luckman212 (~irc@pool-108-6-166-193.nycmny.fios.verizon.net)
01:05.15*** join/#asterisk nix8n82 (~nate@24.143.10.144)
01:07.25*** join/#asterisk luckman212_ (~irc@pool-108-6-166-193.nycmny.fios.verizon.net)
01:07.37*** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net)
01:08.35*** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net)
15:01.07*** join/#asterisk infobot (~infobot@rikers.org)
15:01.07*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
15:02.09*** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net)
15:02.38TheCompWizanyone know much about the tlsdontverifyserver in sip.conf?  .... Can that be specified per-extension?
15:02.49TheCompWizor per-peer rather...
15:03.29*** join/#asterisk ccesario (~ccesario@189.29.37.189)
15:04.34*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
15:06.00*** join/#asterisk vinhdizzo (~vinh@dhcp-v030-181.mobile.uci.edu)
15:06.37*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)
15:07.54Kobazanyone know if there's a polycom setting for decreasing the no-answer time
15:08.40Kobazlike someone sets up a call forward using the polycom menu to go on no-answer... and i do like a Dial(sip/polycom,20)  and the Dial() returns before the forward kicks in, so the call never gets forwarded
15:09.26Kobazso i can do like a dial timeout of 30-40 seconds, but then that's annoying because you don't always have to ring that long, because not everyone has noanswer forwarding turned on all the time
15:10.33WIMPyDo it in the dialplan instead of the phone.
15:10.41Kobazthat's not the solution
15:11.24Kobazpolycom has built in call forwarding that makes use of sip 302, it should be usable
15:11.34*** part/#asterisk asterisk-Tester (~ramy@80.79.159.36)
15:13.04WIMPyThen ask Polycom.
15:13.19Kobazyeah
15:13.24Kobazjust wondering if anyone knew offhand
15:13.31WIMPyAnd start again if you try another device.
15:14.45TheCompWizKobaz: in the ring type definition, there's a timeout value that can be adjusted.
15:14.55TheCompWizi.e. se.rt.4.timeout="2000"
15:14.56Kobazk
15:15.00TheCompWizin ms
15:15.01Kobazah that would be perfect
15:15.40WIMPyOh, and BTW: Deflection os not a functional replacement for forwarding.
15:16.21Kobazdeflection?
15:16.25Kobazdo you mean diversion?
15:19.41p3nguinDiversion or deflection doesn't really make a big difference... a 302 is a Redirect no matter what.
15:20.59Kobazyeah
15:21.09Kobazi like 302s because you don't have to do anything specific in dialplan
15:21.26KobazDial just makes a local channel for you in the originating context and off you go
15:21.44*** join/#asterisk reber (~reber@tsm83-4-78-232-65-13.fbx.proxad.net)
15:30.40WIMPyNo I meant deflection. Diversion means either (both).
15:31.11WIMPy(At least in telephony terms)
15:33.52*** join/#asterisk luckman212 (~irc@pool-108-6-166-193.nycmny.fios.verizon.net)
15:35.39*** join/#asterisk nix8n82 (~nate@24.143.10.144)
15:53.39psilikonp3nguin, that was my hunch. Thanks
15:54.20psilikon[TK]D-Fender, Ok, I'm going to look into what exactly qualify really means. I was equating it to latency.
16:09.56*** join/#asterisk din3sh (din3sh@41.136.241.137)
16:12.07danlenchmorning all, the quest continues. I am working on replacing a vodavi DVX with asterisk. i understand ithe need for the FXO card to keep the POTS (4 lines). do i need an FXS for each phone line (24) if i wish to keep the existing phones or am i missing that vodavi in-9015 can communicate using another way? thanks again for your help
16:13.16Kobazare your existing phones regular analog phones?
16:13.23WIMPyAre they POTS phones?
16:13.32danlenchhey
16:13.39[TK]D-Fenderdanlench,  Those look like proprietary Digital sets which are usable as analog phones at best.  Hardly worth it.
16:13.43danlenchi dont think so
16:14.05[TK]D-Fenderdanlench, Rip & replace
16:14.09WIMPyThey you probably can't recycle them.
16:14.20danlench[TK]D-Fender: thast kinda what i'm leaning toward
16:14.29Kobazdanlench: the money you spend on fxs ports, you can put towards new phones and have a much better setup
16:14.40danlenchright
16:14.45WIMPyOr yo keep the box as a kind of channel bank.
16:14.56danlenchand forward compatible
16:15.43*** join/#asterisk shido6 (~shido6@nat/yahoo/x-uhavdjokhutrrfcu)
16:19.14danlenchnext query, we have 2 operators here at any given time. any suggestion for a phone with 24 buttons?
16:19.38danlenchreal new to this whole thing but getting there
16:20.04WIMPyAny one that accepts a side car.
16:20.23danlenchthat makes sense ;)
16:20.24[TK]D-Fenderdanlench, Polycom IP650 + 2 expansion units
16:21.38danlench[TK]D-Fender: good looking system, thx
16:23.29*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
16:24.10*** join/#asterisk timahvo1 (~rogue@41.80.67.24)
16:29.02*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
16:33.31*** join/#asterisk luckman212_ (~irc@pool-108-6-166-193.nycmny.fios.verizon.net)
16:38.46p3nguinqualify vs. ping
16:39.03p3nguin~qualify vs. ping
16:39.03infobotOne is the time for a response to a SIP OPTIONS packet from Asterisk to the other device, the other is an ICMP echo (ping) and reply (pong) round-trip time.  This is a matter of application layer vs. network layer.
16:39.05p3nguinpsilikon: ^
16:40.01WIMPyWow. That's a long tag.
16:40.16p3nguinAs opposed to something like...
16:40.27p3nguin~book
16:40.27infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
16:40.48p3nguinor...
16:40.50p3nguin~echo
16:40.50infobotecho is probably an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
16:41.04p3nguinOr what?
16:41.06[TK]D-Fender~qualify
16:41.09[TK]D-Fender~sipqualify
16:41.16[TK]D-Fenderhrm, thought there was another
16:42.02*** part/#asterisk frem (2XoDPG1vmB@noembed.com)
16:51.12*** join/#asterisk Defraz (~Defraz@69.20.176.132)
16:53.51*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
16:59.20*** join/#asterisk brdude (~brdude@12.155.183.30)
17:08.27gustohey
17:08.31gustoi have a little problem
17:09.16gustoi have found a provider who does want that i call his number and then the number i want to call ended with a #
17:09.38gustowhy do they do such bullshit?
17:16.15*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
17:18.30tzangerbecause they aren't serious about your business. Use another provider
17:19.43leifmadsenor just add "#" to the end of your dial string...
17:22.16gustono
17:23.07[ProB]CrazyManWIMPy: short question if i want to dial out +49.... is the + not valid? do i have to make 00 insted ?
17:23.20p3nguinAdding the character is rather trivial.  Add it and move on.
17:23.41gustothat does not work
17:24.41p3nguinA) You're doing it wrong.  B) They told you wrong.
17:24.46gustow8 w8
17:25.05leifmadsentry real words
17:25.17p3nguinweight weight
17:25.24gustoit does not work when when i call the number and add the other and # it does not work
17:25.25leifmadsenwat wat
17:25.38gustobut it works when i call their number and then add it after
17:25.39leifmadsenthen that is not what they really want
17:25.49leifmadsenthen add a # with SendDTMF
17:25.55leifmadsenafter the answer
17:25.59gustoyy
17:26.12leifmadsenor use a provider that doesn't do things crazy
17:26.30*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
17:30.04WIMPy[ProB]CrazyMan: + is not valid, but you can use the screen-out parameter to translate it to the right number type.
17:30.32WIMPyscreen-out unknown +% international %
17:30.36*** join/#asterisk luckman212 (~irc@pool-108-6-166-193.nycmny.fios.verizon.net)
17:30.56[ProB]CrazyManuhm...
17:31.13[ProB]CrazyManwhat do i need to call internationl calls ??
17:31.19[ProB]CrazyMan0049 also not work
17:31.32WIMPyDTAG line?
17:31.37[ProB]CrazyManyes
17:31.57[ProB]CrazyMando i have to do without 00 and + 49....
17:32.13WIMPyThat's a known DTAG "feature". Add the following line befor the other one I just posted:
17:32.23WIMPyscreen-out unknown +49% national %
17:34.01*** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net)
17:34.05gustohmm
17:34.05gustoof course, but now i sent them already 10 eur, so i want to use at least that
17:34.05gustohowever, it works when i call their number and then add the number when they picked up
17:34.05gustoand there are some ways how to do that automatically in the dialplan
17:34.09gustoi think that was something with what to do first, second and so on
17:34.09gustoyou know, when the number picks u
17:34.09gustop
17:35.16[TK]D-Fender[ProB]CrazyMan, Ask your provider what format they want it in.  We do no know what they want or expect.  Ask them yourself.
17:37.02*** join/#asterisk Unimatrix-001 (~chatzilla@cpc12-hawk14-2-0-cust183.aztw.cable.virginmedia.com)
17:37.20p3nguinYou have to call an access number to make calls?  That doesn't sound like an ITSP; that sounds like a calling card service.
17:37.40leifmadsenyes
17:38.23gustop3nguin: yes, sounds like
17:38.42leifmadsenin which case... just pass the appropriate dtmf after answer
17:39.34p3nguinAnyone with basic dial plan knowledge should understand the logic needed.
17:40.18gustop3nguin: same => ?
17:41.10leifmadseno.O
17:41.34Unimatrix-001completely new... anyone up for giving us some advice?
17:42.03p3nguin~ask
17:42.04infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
17:43.02*** join/#asterisk tomaw (tom@freenode/staff/tomaw)
17:44.19Unimatrix-001running asterix 1.7.1 with a SiP320x network phone, both connected to a router running standard 192.x net. Plus one pbx card for analog phone. No dial tone/power on analogue phone but 320x seems to be okay. Currently just trying to get anolague and 320x to 'talk' to each other on a test rig?
17:45.25WIMPyThere is no such thing as Asterisk 1.7.1.
17:45.33WIMPyWhat kind of hardware do you use?
17:45.40Unimatrix-001asterixnow 1.7.1 lol
17:46.46leifmadsenyou mean asterisknow
17:46.47gustohow can it be that the voip providers are such idiots? how do providers exchange data (calls from their custommers) between each other? does anyone know?
17:46.47Unimatrix-001openvox a400 p11 - which is installed on test rig and is being 'seen' by freepbx gui
17:47.09WIMPyIf you're using a gui, try asking in #asterisk-gui or #freepbx, depending on what you use.
17:47.28WIMPygusto: Usually not.
17:47.34gustoi mean for example a analogue telephone provider wants to exchange calls with another providers for example mobile phone provider, how do they do it?
17:47.54WIMPygusto: SS7
17:48.02gusto???
17:48.19gustohow SS7? through telephone lines?
17:48.31Unimatrix-001gui isn't the problem... Cant fibure out why there appears to be no power to analoge phone even though the openvox card is powered and recorgnised by AsterixNow and Freepbx gui.
17:48.37WIMPyCopper, fiber, IP, whatever.
17:48.43*** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net)
17:48.49_Corey_gusto: http://en.wikipedia.org/wiki/Public_switched_telephone_network
17:49.03gustoso it is possible to use SS7 through IP?
17:49.27leifmadsenUnimatrix-001: this room isn't really really meant for that kind of debugging. If you're having issues with the hardware, check with the hardware manufacturer for support.
17:49.45WIMPygusto: See SIGTRAN
17:51.30leifmadsenUnimatrix-001: irc etiquette dictates messages should be in the chat room not directly
17:51.34gustohttp://en.wikipedia.org/wiki/SIGTRAN
17:52.04*** join/#asterisk greenwolf (~greenwolf@pool-173-64-4-155.bflony.fios.verizon.net)
17:52.22Unimatrix-001leifmadsen: learning - been awhile since I been on irc!
17:52.42greenwolfSup guys
17:53.08greenwolfI'm wondering how I can put a dial plan together for dynamic caller I'd on outgoing calls
17:53.47[ProB]CrazyManWIMPy: do I have to place the sceen-out on a special place ??? it doesnt remove the +49
17:53.52greenwolfI basically want the end users to be able to dial a number from their sip device and then they must dial the outbound caller I'd for each call
17:53.56pabelangergreenwolf: create a gosub and pass it the callerid value
17:53.57WIMPyDoesn't sound hard, but dynamic in what way?
17:54.25greenwolfIn that the user can set there caller I'd on each call they make
17:54.25WIMPy[ProB]CrazyMan: After the portnum/portname,
17:54.34*** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith)
17:54.48greenwolfPabelanger: plz explain further
17:55.17leifmadsenpabelanger: don't do it, it's a trap!
17:55.26pabelangero.0
17:55.27gustowhat does this SS7 use for voice transport? 64kbps? is that G711 again?
17:55.34leifmadsengreenwolf: learn how GoSub() works then it'll make sense
17:55.42leifmadsengusto: try the googles
17:55.43greenwolfOk thanks guys
17:55.59WIMPygusto: Yes, usually the whole PSTN is G.711
17:56.07leifmadsenGoSub(mySubroutine,start,1(${myDynamicCallerId}))
17:56.16WIMPyAlthoug G.722 has been supported in theory for 20 years.
17:56.45greenwolfCan I use {[EXTEN]} ? To place the values they dial into setcallerid()
17:57.01greenwolfFor each outbound call dialed
17:57.40greenwolfGotcha ya leifmadsen
17:57.42WIMPy'core show function CALLERID', but otherwise, yes.
17:57.44greenwolfMakes sense now
17:57.46leifmadsenyou mean ${EXTEN} and Set(CALLERID(num)=...)
17:57.53[ProB]CrazyManWIMPy: doesnt work ... i killed lcr and started it again
17:58.06p3nguinWe don't use setcallerid() for anything.  If you need to set the callerid number, use Set(CALLERID(num)=number)
17:58.08greenwolfShould I use gosub or EXTEN to set the values.
17:58.32greenwolfBut I want that value to be dynamic for each call placed
17:58.32leifmadsenEXTEN is just the value that the extension executing contains
17:58.36leifmadsenit's not likely what you want to use
17:58.42WIMPy[ProB]CrazyMan: You are calling with the + in the number? And you have the two line in the correct order?
17:58.46p3nguinGoSub runs a subroutine.  What does that have to do with the extension number?
17:58.51leifmadsenunless you want the CALLERID to be what you're calling every time
17:59.25WIMPyOr a part of it?
17:59.30*** join/#asterisk willianmazzardo (~textual@201-34-92-116.smace701.dsl.brasiltelecom.net.br)
17:59.37greenwolfOk maybe I should be more clear
17:59.39willianmazzardohi all … good afternoon...
17:59.48greenwolfAgain I appreciate everyone's help on this
17:59.55willianmazzardoi have this situation … Asterisk 1.8.11.0 and Extensions in DAHDI channels ...
18:00.05willianmazzardosometimes … this error ocurres and crash my asterisk
18:00.06willianmazzardo[Apr  9 14:46:53] WARNING[10731]: sig_analog.c:3606 analog_exception: We're DAHDI/16-1, not
18:00.20leifmadsenok I gotta leave bye
18:00.21*** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
18:01.13[ProB]CrazyManWIMPy: as you told me ... http://pastebin.com/y7evue8Z
18:01.43willianmazzardoanyone have anything to solve this problem? I really dont want to get back in asterisk 1.4
18:02.32*** join/#asterisk greenwolf_ (~greenwolf@mobile-198-228-204-068.mycingular.net)
18:02.38greenwolf_Each office phone when they dial 33 will ask them to enter the num they wish to pass as their callerid
18:02.47WIMPy[ProB]CrazyMan: I'v got the screening stuff at the end but as long as it's after portnum, it shouldn't matter.
18:02.48greenwolf_Then it will ask them to dial the num they wish to call
18:03.04greenwolf_Placing that call the values from step 1 for caller I'd are passed to the trunk
18:03.08greenwolf_Out to the carrier
18:03.37WIMPy[ProB]CrazyMan: Maybe the type is set? Do you have a lcr log?
18:03.43greenwolf_Rather then setting these values manually in the extensions.conf file every time
18:03.58greenwolf_Does that clear up maybe,..hopefully :)
18:05.28WIMPyIf you want to do it interavtively, Read() is your friend.
18:05.55[ProB]CrazyManWIMPY: doesnt look like http://pastebin.com/rPUbSA2A
18:06.24gustowhat are these dahdi cards for again?
18:06.46gustoi can not find any info on that somehow
18:06.48[TK]D-Fendergusto, Analog phones/lines
18:06.52gustowell
18:07.02*** join/#asterisk joshaidan (~brianj@S010698fc113e438d.tb.shawcable.net)
18:07.14gustobut i have an ATA adapter like PAP2T, is that not sufficient for everyone?
18:07.26[TK]D-Fendergusto, You can't plung PHONE LINES into that.
18:07.29[TK]D-Fenderpulg*
18:07.32gustoaha
18:07.42gustowell, i do not need to
18:07.53[TK]D-Fendergusto, You != Everyone
18:08.44gustoof course, i am not posing rather provokative questions, but thats for clarification what may be the use of such a setup to plug voip into analogue line
18:08.59gustoso ... someone needs that ... ok, but for what?
18:09.37gustos/not/ /
18:09.45greenwolfSo the right direction for me is gosub() ?
18:10.08greenwolfI will take time to read and learn I just need to know what I should learn to achieve that
18:10.16[TK]D-Fendergusto, ... for what? ... to use ANALOG LINES with Asterisk.
18:10.26[TK]D-Fendergusto, How is this a difficult concept?
18:10.50[TK]D-Fendergusto, And you don't plug "VoIP" into an analog line.
18:11.05gustoit is not difficult, it is just something i can not imagine right
18:11.36[TK]D-FenderGuess you've never used a modem before.  Or a fax machine.  Or an older PBX.  Or an answering machine...
18:11.47gustoof course i did
18:11.55gustoi even broke some of them
18:12.03[TK]D-FenderI am not surprised
18:12.08gustoyes
18:12.57WIMPy[ProB]CrazyMan: Looks like that search/replace thing doesn;t work for me, either.
18:12.58[TK]D-Fendergusto, DAHDi card = lets you plug analog lines into your server.  If you're saying you can't understand what the use of this is for... I don't even know what to say...
18:13.09WIMPywonders if there's somethig wrong or if there's a bug.
18:13.11gustoso why would someone want to use asterisk on a analogue line, when he can rather connect that telephone directly to the analogue line. or is it just for having more phones to be able to pick the call up/make
18:13.34[TK]D-Fendergusto, so that ASTERISK talks to the phone line.
18:13.35WIMPy[ProB]CrazyMan: You can do it in the dialplan, off course.
18:14.12[ProB]CrazyManWIMPy: maybe an bug, dont care ... I did it via asterisk... its easier ... I just have to remember it ... when i make the dialplan to call out ;()
18:14.21p3nguingreenwolf: No, GoSub is not the right thing.  You want Read().
18:14.27[ProB]CrazyManWIMPy: thx for your support the last two days
18:15.21WIMPy[ProB]CrazyMan: I have to say that I'm using a modified version of the screening functio, however. But that part shouldn't be affected.
18:15.48greenwolf<PROTECTED>
18:16.06gusto[TK]D-Fender: i ve read that they developed this DAHDI cards because they wanted to replace some expensive commercial hardware with only DAHDI as only cards with circuits that do only the physical stuff and everything else is being made by the CPU, but that software that does this is still a part of DAHDI, so what does asterisk do then? answer calls?
18:16.17greenwolfThen i integrate read() with set(callerid)
18:16.46[TK]D-Fendergusto, It lets * use your analog lines.
18:17.49gusto[TK]D-Fender: i ve understood that long ago, but what would be an example setup for what ppl would want asterisk to use the analogue lines?
18:17.58greenwolfOk I see now again thanks penguin
18:18.08[TK]D-Fendergusto, People who have them will want to use them
18:19.37gusto[TK]D-Fender: cool
18:19.47gusto[TK]D-Fender: how much does such a DAHDI card cost?
18:20.11[TK]D-Fendergusto, As much as the company that sells them asks for and yuo pay.
18:20.20[TK]D-Fenderyou*
18:20.40gusto[TK]D-Fender: so a lot
18:20.57[TK]D-Fendergusto,  If that's what they ask, and that's what you pay.
18:21.12[TK]D-Fendergusto, And that's what you consider "a lot"
18:22.14gusto[TK]D-Fender: i would never give 200++ eur for only being able to talk to my analogue phone line (or four of them)
18:23.03p3nguingreenwolf: http://pastebin.com/smqxhRZ4
18:23.23[TK]D-Fendergusto,  Shouldn't cost 200eur for 1 phone line....
18:23.40WIMPygusto: If you're paying in EUR, you probably don't want (have to) to use analog anyway.
18:24.00gustoWIMPy: what currency should i pay instead?
18:24.30WIMPyIf you pay in US$, you may want to.
18:24.35[TK]D-Fenderfacepalms....
18:24.58gustoWIMPy: what do they have better in the US for an analog? cheaper analogue connection fees, or what?
18:25.19[TK]D-Fendergusto, You clearly just don't seem to get it...
18:25.38[TK]D-Fendergusto, this was not a currency question
18:25.40gusto[TK]D-Fender: i pretty much figured out that i am not getting it
18:25.42WIMPyNo, but better alternatives are rare and expensive there.
18:25.50WIMPyUnlike in Europe.
18:26.18[TK]D-Fendergusto, WIMPyIs alluding that outside of  North America most places offer ISDN instead of analog lines and would be a better choice.
18:26.19gustoWIMPy: ah you mean that they do not have VoIP, or ISDN, or something?
18:26.39[TK]D-Fendergusto, And this has nothing to do with "VoIP"
18:26.46[TK]D-Fendergusto, Stop thinking that everything does.
18:26.52gustook ok
18:26.54[TK]D-Fendergusto, We are talking LINES here.
18:27.08[TK]D-Fendergusto, And there are DADHi-based cards for ISDN as well.
18:27.17[TK]D-FenderBRI & PRI
18:28.25*** join/#asterisk TimeRider (~steve@92.40.225.95.threembb.co.uk)
18:28.42gustowell, but let's stick to the point, we were talking about DAHDI and that in europe it's not very much likely that i would need one, or when i would need one, only when i am having ISDN and wanting to make some stuff with it using asterisk for that
18:28.50WIMPy... and cheaper.
18:29.02gustoWIMPy: cheaper where?
18:29.45WIMPyA BRIs is usually much cheaper than two analog lines.
18:29.58gustowhat is BRI again?
18:30.10KNERDnot in the US
18:30.14WIMPyAnd if you have ISDN, DAHDI is only one of the possibilites to connect it to Asterisk.
18:30.30WIMPyKNERD: Scroll up :-)
18:30.47WIMPyBRI = 2 channel ISDN line.
18:30.52KNERDi know..EUROPE
18:30.57KNERDbut in US much cheaper
18:31.18gustocool
18:31.41gustoand why is it still a question of money how much analogue telephone lines you want to use?
18:31.47KNERDISDN is about $90 US while you could get 2 basic analog for $40US
18:32.16gustoi mean there are overcapacities in germany according to http://cre.fm/cre191
18:32.21WIMPySo even alalog is expensive?
18:32.41gustoKNERD: what is that for a price? a monthly price or what?
18:33.02KNERDit's about $20 a month for one analog line
18:33.11gustoKNERD: that is too much
18:33.11[TK]D-FenderDependsing where you are.
18:33.23KNERDon average that is the price
18:33.29gustowell, i would not give away shit like more than 10 eur per one line
18:33.32WIMPyAh, ok, that was for two. Ok.
18:33.39KNERDso 2 channel ISDN for $90 is cheaper?
18:33.40gustoof course
18:33.41WIMPygusto: That article is not about telephony.
18:34.02gustoWIMPy: however, but they talk about telephone lines in the podcast
18:34.05KNERDwhile 2 analog POTS for about $40 is more? hmmmm
18:34.09[TK]D-FenderWIMPy, none of his neurons are firing synchronously
18:34.39WIMPyJa, it's a bit jumpy...
18:35.40[TK]D-FenderWIMPy, Case in point : I took more than a dozen lines of Q&A to still not seem to have a grasp on the concept of why someone would want to use analog lines with *
18:36.58WIMPyAt least we had the usual reminder why anyone might want to use analog at all.
18:37.39gusto[TK]D-Fender: so you do not understand it either?
18:37.41[TK]D-FenderWIMPy, And we saw much how it took to get the idea of "Because that's what the user has" across.
18:38.05[TK]D-Fendergusto, No, you seem to have issues putting this together.
18:38.16gusto[TK]D-Fender: i always have :-d
18:38.23gusto:-D
18:38.35gustobut when i put something together, it works
18:38.52[TK]D-FenderExcept when you break it.  And you've broken all sorts of things as you've mentioned.
18:39.08gusto[TK]D-Fender: well
18:39.19gusto[TK]D-Fender: was not my fault :-D
18:40.03WIMPyThat's what they all say.
18:41.18gustoyes
18:41.33gustothat's why i never said that when something broke
18:50.09*** join/#asterisk greenwolf (~greenwolf@pool-173-64-4-155.bflony.fios.verizon.net)
18:50.29greenwolf<PROTECTED>
18:50.39greenwolfThat was very useful for me
18:53.48*** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith)
18:56.03*** join/#asterisk TimeRider (~steve@92.40.254.108.threembb.co.uk)
19:00.20*** join/#asterisk Henchman21 (~rakata@208.102.127.220)
19:00.23bent_screwdriveranyone here have any luck running asterisk on an embedded os/hardware using PRI?
19:03.39WIMPybent_screwdriver: As long as you've got a slot for the card and don;t want to do EC on the host, why should that be special?
19:05.20bent_screwdriverWIMPy: just curious of other's experience. i've been running asterisk on voyage/net6501 and wanted to compare systems. I'm running hw echo cancellation. Are you referring to sw echo cancellation?
19:05.49WIMPyYes, that would require a beefy CPU.
19:07.58bent_screwdriverWIMPy: yeah, i'm curious of how many channels it can transcode g729. Is there a good way to load up the PRI and test, to see how many channels i can run before it crawls? the 6501 has about 1.6ghz cpu and 2GB ram. we only peak at about 6 channels and it gets about %20 cpu
19:08.34WIMPyUgh, g.729 sounds evil as well.
19:09.04*** join/#asterisk greenwolf (~greenwolf@pool-173-64-4-155.bflony.fios.verizon.net)
19:09.06WIMPyConnect it to another box with a PRI card and loop the channels between them.
19:10.14bent_screwdriverwhat is > than g729?
19:10.27WIMPyI used an extension that Dial()s the called extension-1 and the one ending in 0 with MusicOnHold(), so I could just dial the number of channels to try.
19:10.37WIMPy> in what sense?
19:11.10bent_screwdriverwhen bandwith considerations and quality are of importance
19:11.39WIMPyStay with G.711 as is sent and receiveed on the line.
19:12.13WIMPyIf you can't afford the bw, G.729 may be a good choice but a CPU hungry one.
19:13.38bent_screwdriverwasn't there some hardware module that offloads transcoding? I only have mini pci slots after the PRI so probably not an option anyways....
19:14.25WIMPyDigium have a hardware solution, yes, but I'm pretty sure that mini pci will be a show stopper.
19:15.08WIMPyI'm not sure if anyone has mini pci pri cards perhaps. I know there are at least 4 BRI in mini pci.
19:15.34WIMPyPCIe might make things easier.
19:17.09ectospasmDigium doesn't carry mini PCI or mini PCIe anything.
19:18.23*** join/#asterisk gusto (~gusto@nrbg-4d0766a3.pool.mediaWays.net)
19:18.44WIMPyMini PCIe would only be a mechanical issue.
19:20.25[TK]D-FenderWIMPy, OpenVOX has mini PCI/PCIe
19:22.05*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
19:22.27WIMPyBut no PRI as it seems.
19:24.56*** join/#asterisk luckman212_ (~irc@pool-108-6-166-193.nycmny.fios.verizon.net)
19:25.41*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
19:27.02gustohmm
19:32.05[TK]D-FenderWIMPy, Yup, just found the same cruising their full catalog.
19:32.17*** part/#asterisk willianmazzardo (~textual@201-34-92-116.smace701.dsl.brasiltelecom.net.br)
19:36.34*** join/#asterisk thafreak (~thafreak@unaffiliated/thafreak)
19:37.01thafreakCan anyone recommend something to connect to analog lines?
19:37.25thafreakAre the wildcards the best route, or is there something better these days?
19:37.38thafreakvery small phone system with < 4 lines
19:38.01WIMPydeja vu?
19:38.03thafreaki saw there are these sip->analog gateway boxes with like 4 ports
19:39.53Henchman21im pissed at this spa3k i recently got, wont dialout
19:40.04Henchman21gave up on it
19:40.10[TK]D-Fenderthafreak, Are you expecting to stick with analog for a while?  How many to start?  Expected expansion?
19:40.25[TK]D-FenderHenchman21, You're configuring it wrong
19:40.29Henchman21still answers the houseline and sends it to asterisk
19:40.54thafreakno expansion, 4 analog lines will be all that will ever be used
19:41.00Henchman21nah i've tried quite a few different configs but it still acts goofy
19:41.38Henchman21like ill send a call out it, then call the house line line and it dials the darn number in my ear
19:41.53Henchman21it drops the call or something
19:41.59thafreakcurrently using an older wildcard with 4 fxo modules, but recently it seems a few got fried
19:42.25Henchman21i dunno i started listening to its syslog debug output but i cant make heads or tails of it
19:43.22[TK]D-Fenderthafreak, I'd say aim for a Sangoma B600d
19:43.45[TK]D-Fenderthafreak, And and get some surge supressors for your lines.
19:46.51thafreakso the cards are still better than something like: Audiocodes MP108-FXO gateway?
19:48.31*** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com)
19:49.24[TK]D-Fenderthafreak, Offers * a little more control over the lines, and are a fair bit cheaper
19:51.40thafreakseems that card is actually like $100 more than the audiocodes gateway...
19:53.01[TK]D-Fenderthafreak, Depends where you shop I guess...
19:53.02*** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com)
19:53.22[TK]D-Fenderthafreak, The AudioCodes should work fine for most things as well...
19:54.03thafreakare the grandstream ones junk? They seem to be pretty cheap (like $250)
19:55.17*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)
19:58.24[TK]D-Fender~gs
19:58.24infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
19:58.28[TK]D-Fender~grandstream
19:58.28infobotmethinks grandstream is the Yugo of VoIP hardware.  Run...  Run away now.  Though, therealcircut says that they're not that bad.
19:58.38[TK]D-FenderYMMV
19:59.41thafreaki've used their phones and ata's
19:59.59thafreaktheir ata's aren't horrible, but could be better i guess
20:10.59gustoah
20:11.25gustoluckily i bought a second pap2t instead of grandstream
20:17.31*** join/#asterisk shido6 (~shido6@nat/yahoo/x-kfqmfjwitlvhvdwf)
20:28.50*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)
20:29.15*** join/#asterisk dijib (~dijib@bas10-kitchener06-1279682460.dsl.bell.ca)
20:37.03*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
20:39.14*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
20:40.28*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
20:48.11*** part/#asterisk Unimatrix-001 (~chatzilla@cpc12-hawk14-2-0-cust183.aztw.cable.virginmedia.com)
20:49.12*** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net)
20:49.20*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
20:54.56*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:03.05*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:06.27ectospasmwe don't have much good to say about grandstream here
21:06.57tm1000the spa112 is the newer version of the pap2t
21:07.04tm1000so i hope you didnt get ripped off for an old unit
21:07.10tm1000gusto:  ^^
21:09.13gustono
21:09.18gustoit is quite cheap
21:10.49*** join/#asterisk pyite_mac (~dschreibe@pdpc/supporter/bronze/pyite)
21:11.14pyite_macis there any way to make asterisk retry registrations on a 403? It seems sometimes the upstream provider erroneously returns 403 when they are overloaded, which i realize is there problem, but... hoping there's a solution
21:11.23gustotm1000: is SPA112 for 50 eur OK?
21:13.12tm1000gusto:  sounds about right. they are really good for faxing
21:13.20tm1000if you wanted t.38
21:13.59*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:14.23gustotm1000: ok
21:14.31gustotm1000: i bought that now as well :-D
21:15.11tm1000they are good little units
21:15.49gustoyes yes
21:15.55gustopap2t is too
21:16.02gustoi can use SOME of them
21:16.48gustoi have one now, i ordered another pap2t and now that spa112, i am going to use it for other places /saves me a lot of money/
21:18.55*** join/#asterisk jsjc (~Adium@54.Red-83-32-88.dynamicIP.rima-tde.net)
21:20.17*** part/#asterisk pyite_mac (~dschreibe@pdpc/supporter/bronze/pyite)
21:26.41*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:30.37*** join/#asterisk greenwolf (~greenwolf@pool-173-64-4-155.bflony.fios.verizon.net)
21:31.34*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
21:31.35*** mode/#asterisk [+o malcolmd] by ChanServ
21:34.51greenwolf<PROTECTED>
21:35.02greenwolfUse*
21:36.48*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:38.36*** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com)
21:42.25*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:42.32*** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452)
21:42.57drudge`anyone like or recommend a free or cheap hudlite alternative?
21:49.52*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:51.55*** join/#asterisk shido6 (~shido6@nat/yahoo/x-ztnzjijdvefeqoky)
21:55.37*** join/#asterisk shido6 (~shido6@nat/yahoo/x-bajvtxpujfjjollh)
21:55.43*** join/#asterisk greenwolf (~greenwolf@pool-173-64-4-155.bflony.fios.verizon.net)
21:55.46*** part/#asterisk danlench (~daniell@pool-71-113-225-119.herntx.dsl-w.verizon.net)
21:55.54*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
21:58.28*** join/#asterisk greenwolf (~greenwolf@mobile-198-228-206-186.mycingular.net)
22:03.23*** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith)
22:03.53*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
22:05.00*** part/#asterisk mjordan (~mjordan@nat/digium/x-lxobxqgmdielzxvz)
22:13.21*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
22:16.07*** join/#asterisk CyBeRxIxO (~efernande@190.41.182.228)
22:16.59CyBeRxIxOhi, i got a question, thanks for ur answer in advance
22:17.20CyBeRxIxOi just installed asterisknow, killed my elastix box
22:18.09CyBeRxIxOwhen i get the g729 codec from digium do i need to install anything else like procesor driver or something?
22:18.40CyBeRxIxOonce i buy it do i get the support needed on my general configuration?
22:18.49*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
22:19.43*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net)
22:25.11*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
22:30.06*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
22:33.49*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
22:34.41*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
22:36.05*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
22:50.58*** join/#asterisk mintos (mvaliyav@nat/redhat/x-qrppwjkyjudsxgpo)
23:01.58*** join/#asterisk Bullmoose (~Bullmoose@71-33-1-137.bois.qwest.net)
23:11.17*** join/#asterisk nix8n82 (~nate@24.143.10.144)
23:30.19*** join/#asterisk KNERD (~KNERD@adsl-99-153-147-234.dsl.hrlntx.sbcglobal.net)
23:32.11*** join/#asterisk MaliutaLap (~buggeroff@202.124.75.91)
23:40.00*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
23:40.00*** mode/#asterisk [+o mjordan] by ChanServ
23:42.15*** join/#asterisk luckman212_ (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46)
23:46.47*** part/#asterisk bminish (~bminish@pdpc/supporter/professional/bminish)
23:54.10*** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com)
23:55.57CubberI am running asterisk 10.3.0 on gentoo using googletalk for a trunk.  Previously I was running 1.8.8.2 with no issues but upgrading to any other version causes segfaults when I try to access the CLI when the jabber.conf file is in my /etc/asterisk directory.  It loads all of the presence information and then segfaults.
23:56.28Cubberif I do not hit the CLI the server runs fine it just happens when I try to access CLI.  If I remove jabber.conf and dont load it everything works as expected.
23:56.36*** join/#asterisk xpot-mobile (~xpot@166-70-100-198.ip.xmission.com)
23:59.28Cubberasterisk[9779]: segfault at c18 ip b726627d sp b6d04cf4 error 4 in libpthread-2.13.so[b725d000+15000]
23:59.29*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.