| 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) |
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| 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) |
| 15:02.38 | TheCompWiz | anyone know much about the tlsdontverifyserver in sip.conf? .... Can that be specified per-extension? |
| 15:02.49 | TheCompWiz | or 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) |
| 15:07.54 | Kobaz | anyone know if there's a polycom setting for decreasing the no-answer time |
| 15:08.40 | Kobaz | like 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 |
| 15:09.26 | Kobaz | so 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 |
| 15:10.33 | WIMPy | Do it in the dialplan instead of the phone. |
| 15:10.41 | Kobaz | that's not the solution |
| 15:11.24 | Kobaz | polycom 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) |
| 15:13.04 | WIMPy | Then ask Polycom. |
| 15:13.19 | Kobaz | yeah |
| 15:13.24 | Kobaz | just wondering if anyone knew offhand |
| 15:13.31 | WIMPy | And start again if you try another device. |
| 15:14.45 | TheCompWiz | Kobaz: in the ring type definition, there's a timeout value that can be adjusted. |
| 15:14.55 | TheCompWiz | i.e. se.rt.4.timeout="2000" |
| 15:14.56 | Kobaz | k |
| 15:15.00 | TheCompWiz | in ms |
| 15:15.01 | Kobaz | ah that would be perfect |
| 15:15.40 | WIMPy | Oh, and BTW: Deflection os not a functional replacement for forwarding. |
| 15:16.21 | Kobaz | deflection? |
| 15:16.25 | Kobaz | do you mean diversion? |
| 15:19.41 | p3nguin | Diversion or deflection doesn't really make a big difference... a 302 is a Redirect no matter what. |
| 15:20.59 | Kobaz | yeah |
| 15:21.09 | Kobaz | i like 302s because you don't have to do anything specific in dialplan |
| 15:21.26 | Kobaz | Dial 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) |
| 15:30.40 | WIMPy | No I meant deflection. Diversion means either (both). |
| 15:31.11 | WIMPy | (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) |
| 15:53.39 | psilikon | p3nguin, that was my hunch. Thanks |
| 15:54.20 | psilikon | [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) |
| 16:12.07 | danlench | morning 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 |
| 16:13.16 | Kobaz | are your existing phones regular analog phones? |
| 16:13.23 | WIMPy | Are they POTS phones? |
| 16:13.32 | danlench | hey |
| 16:13.39 | [TK]D-Fender | danlench, Those look like proprietary Digital sets which are usable as analog phones at best. Hardly worth it. |
| 16:13.43 | danlench | i dont think so |
| 16:14.05 | [TK]D-Fender | danlench, Rip & replace |
| 16:14.09 | WIMPy | They you probably can't recycle them. |
| 16:14.20 | danlench | [TK]D-Fender: thast kinda what i'm leaning toward |
| 16:14.29 | Kobaz | danlench: the money you spend on fxs ports, you can put towards new phones and have a much better setup |
| 16:14.40 | danlench | right |
| 16:14.45 | WIMPy | Or yo keep the box as a kind of channel bank. |
| 16:14.56 | danlench | and forward compatible |
| 16:15.43 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-uhavdjokhutrrfcu) |
| 16:19.14 | danlench | next query, we have 2 operators here at any given time. any suggestion for a phone with 24 buttons? |
| 16:19.38 | danlench | real new to this whole thing but getting there |
| 16:20.04 | WIMPy | Any one that accepts a side car. |
| 16:20.23 | danlench | that makes sense ;) |
| 16:20.24 | [TK]D-Fender | danlench, Polycom IP650 + 2 expansion units |
| 16:21.38 | danlench | [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) |
| 16:38.46 | p3nguin | qualify vs. ping |
| 16:39.03 | p3nguin | ~qualify vs. ping |
| 16:39.03 | infobot | One 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. |
| 16:39.05 | p3nguin | psilikon: ^ |
| 16:40.01 | WIMPy | Wow. That's a long tag. |
| 16:40.16 | p3nguin | As opposed to something like... |
| 16:40.27 | p3nguin | ~book |
| 16:40.27 | infobot | Asterisk: 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 |
| 16:40.48 | p3nguin | or... |
| 16:40.50 | p3nguin | ~echo |
| 16:40.50 | infobot | echo 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 |
| 16:41.04 | p3nguin | Or what? |
| 16:41.06 | [TK]D-Fender | ~qualify |
| 16:41.09 | [TK]D-Fender | ~sipqualify |
| 16:41.16 | [TK]D-Fender | hrm, thought there was another |
| 16:42.02 | *** part/#asterisk frem (2XoDPG1vmB@noembed.com) |
| 16:51.12 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
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| 16:59.20 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:08.27 | gusto | hey |
| 17:08.31 | gusto | i have a little problem |
| 17:09.16 | gusto | i have found a provider who does want that i call his number and then the number i want to call ended with a # |
| 17:09.38 | gusto | why do they do such bullshit? |
| 17:16.15 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:18.30 | tzanger | because they aren't serious about your business. Use another provider |
| 17:19.43 | leifmadsen | or just add "#" to the end of your dial string... |
| 17:22.16 | gusto | no |
| 17:23.07 | [ProB]CrazyMan | WIMPy: short question if i want to dial out +49.... is the + not valid? do i have to make 00 insted ? |
| 17:23.20 | p3nguin | Adding the character is rather trivial. Add it and move on. |
| 17:23.41 | gusto | that does not work |
| 17:24.41 | p3nguin | A) You're doing it wrong. B) They told you wrong. |
| 17:24.46 | gusto | w8 w8 |
| 17:25.05 | leifmadsen | try real words |
| 17:25.17 | p3nguin | weight weight |
| 17:25.24 | gusto | it does not work when when i call the number and add the other and # it does not work |
| 17:25.25 | leifmadsen | wat wat |
| 17:25.38 | gusto | but it works when i call their number and then add it after |
| 17:25.39 | leifmadsen | then that is not what they really want |
| 17:25.49 | leifmadsen | then add a # with SendDTMF |
| 17:25.55 | leifmadsen | after the answer |
| 17:25.59 | gusto | yy |
| 17:26.12 | leifmadsen | or 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) |
| 17:30.04 | WIMPy | [ProB]CrazyMan: + is not valid, but you can use the screen-out parameter to translate it to the right number type. |
| 17:30.32 | WIMPy | screen-out unknown +% international % |
| 17:30.36 | *** join/#asterisk luckman212 (~irc@pool-108-6-166-193.nycmny.fios.verizon.net) |
| 17:30.56 | [ProB]CrazyMan | uhm... |
| 17:31.13 | [ProB]CrazyMan | what do i need to call internationl calls ?? |
| 17:31.19 | [ProB]CrazyMan | 0049 also not work |
| 17:31.32 | WIMPy | DTAG line? |
| 17:31.37 | [ProB]CrazyMan | yes |
| 17:31.57 | [ProB]CrazyMan | do i have to do without 00 and + 49.... |
| 17:32.13 | WIMPy | That's a known DTAG "feature". Add the following line befor the other one I just posted: |
| 17:32.23 | WIMPy | screen-out unknown +49% national % |
| 17:34.01 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:34.05 | gusto | hmm |
| 17:34.05 | gusto | of course, but now i sent them already 10 eur, so i want to use at least that |
| 17:34.05 | gusto | however, it works when i call their number and then add the number when they picked up |
| 17:34.05 | gusto | and there are some ways how to do that automatically in the dialplan |
| 17:34.09 | gusto | i think that was something with what to do first, second and so on |
| 17:34.09 | gusto | you know, when the number picks u |
| 17:34.09 | gusto | p |
| 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) |
| 17:37.20 | p3nguin | You have to call an access number to make calls? That doesn't sound like an ITSP; that sounds like a calling card service. |
| 17:37.40 | leifmadsen | yes |
| 17:38.23 | gusto | p3nguin: yes, sounds like |
| 17:38.42 | leifmadsen | in which case... just pass the appropriate dtmf after answer |
| 17:39.34 | p3nguin | Anyone with basic dial plan knowledge should understand the logic needed. |
| 17:40.18 | gusto | p3nguin: same => ? |
| 17:41.10 | leifmadsen | o.O |
| 17:41.34 | Unimatrix-001 | completely new... anyone up for giving us some advice? |
| 17:42.03 | p3nguin | ~ask |
| 17:42.04 | infobot | Questions 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) |
| 17:44.19 | Unimatrix-001 | running 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? |
| 17:45.25 | WIMPy | There is no such thing as Asterisk 1.7.1. |
| 17:45.33 | WIMPy | What kind of hardware do you use? |
| 17:45.40 | Unimatrix-001 | asterixnow 1.7.1 lol |
| 17:46.46 | leifmadsen | you mean asterisknow |
| 17:46.47 | gusto | how 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? |
| 17:46.47 | Unimatrix-001 | openvox a400 p11 - which is installed on test rig and is being 'seen' by freepbx gui |
| 17:47.09 | WIMPy | If you're using a gui, try asking in #asterisk-gui or #freepbx, depending on what you use. |
| 17:47.28 | WIMPy | gusto: Usually not. |
| 17:47.34 | gusto | i mean for example a analogue telephone provider wants to exchange calls with another providers for example mobile phone provider, how do they do it? |
| 17:47.54 | WIMPy | gusto: SS7 |
| 17:48.02 | gusto | ??? |
| 17:48.19 | gusto | how SS7? through telephone lines? |
| 17:48.31 | Unimatrix-001 | gui 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. |
| 17:48.37 | WIMPy | Copper, 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 |
| 17:49.03 | gusto | so it is possible to use SS7 through IP? |
| 17:49.27 | leifmadsen | Unimatrix-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. |
| 17:49.45 | WIMPy | gusto: See SIGTRAN |
| 17:51.30 | leifmadsen | Unimatrix-001: irc etiquette dictates messages should be in the chat room not directly |
| 17:51.34 | gusto | http://en.wikipedia.org/wiki/SIGTRAN |
| 17:52.04 | *** join/#asterisk greenwolf (~greenwolf@pool-173-64-4-155.bflony.fios.verizon.net) |
| 17:52.22 | Unimatrix-001 | leifmadsen: learning - been awhile since I been on irc! |
| 17:52.42 | greenwolf | Sup guys |
| 17:53.08 | greenwolf | I'm wondering how I can put a dial plan together for dynamic caller I'd on outgoing calls |
| 17:53.47 | [ProB]CrazyMan | WIMPy: do I have to place the sceen-out on a special place ??? it doesnt remove the +49 |
| 17:53.52 | greenwolf | I 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 |
| 17:53.56 | pabelanger | greenwolf: create a gosub and pass it the callerid value |
| 17:53.57 | WIMPy | Doesn't sound hard, but dynamic in what way? |
| 17:54.25 | greenwolf | In that the user can set there caller I'd on each call they make |
| 17:54.25 | WIMPy | [ProB]CrazyMan: After the portnum/portname, |
| 17:54.34 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 17:54.48 | greenwolf | Pabelanger: plz explain further |
| 17:55.17 | leifmadsen | pabelanger: don't do it, it's a trap! |
| 17:55.26 | pabelanger | o.0 |
| 17:55.27 | gusto | what does this SS7 use for voice transport? 64kbps? is that G711 again? |
| 17:55.34 | leifmadsen | greenwolf: learn how GoSub() works then it'll make sense |
| 17:55.42 | leifmadsen | gusto: try the googles |
| 17:55.43 | greenwolf | Ok thanks guys |
| 17:55.59 | WIMPy | gusto: Yes, usually the whole PSTN is G.711 |
| 17:56.07 | leifmadsen | GoSub(mySubroutine,start,1(${myDynamicCallerId})) |
| 17:56.16 | WIMPy | Althoug G.722 has been supported in theory for 20 years. |
| 17:56.45 | greenwolf | Can I use {[EXTEN]} ? To place the values they dial into setcallerid() |
| 17:57.01 | greenwolf | For each outbound call dialed |
| 17:57.40 | greenwolf | Gotcha ya leifmadsen |
| 17:57.42 | WIMPy | 'core show function CALLERID', but otherwise, yes. |
| 17:57.44 | greenwolf | Makes sense now |
| 17:57.46 | leifmadsen | you mean ${EXTEN} and Set(CALLERID(num)=...) |
| 17:57.53 | [ProB]CrazyMan | WIMPy: doesnt work ... i killed lcr and started it again |
| 17:58.06 | p3nguin | We don't use setcallerid() for anything. If you need to set the callerid number, use Set(CALLERID(num)=number) |
| 17:58.08 | greenwolf | Should I use gosub or EXTEN to set the values. |
| 17:58.32 | greenwolf | But I want that value to be dynamic for each call placed |
| 17:58.32 | leifmadsen | EXTEN is just the value that the extension executing contains |
| 17:58.36 | leifmadsen | it's not likely what you want to use |
| 17:58.42 | WIMPy | [ProB]CrazyMan: You are calling with the + in the number? And you have the two line in the correct order? |
| 17:58.46 | p3nguin | GoSub runs a subroutine. What does that have to do with the extension number? |
| 17:58.51 | leifmadsen | unless you want the CALLERID to be what you're calling every time |
| 17:59.25 | WIMPy | Or a part of it? |
| 17:59.30 | *** join/#asterisk willianmazzardo (~textual@201-34-92-116.smace701.dsl.brasiltelecom.net.br) |
| 17:59.37 | greenwolf | Ok maybe I should be more clear |
| 17:59.39 | willianmazzardo | hi all … good afternoon... |
| 17:59.48 | greenwolf | Again I appreciate everyone's help on this |
| 17:59.55 | willianmazzardo | i have this situation … Asterisk 1.8.11.0 and Extensions in DAHDI channels ... |
| 18:00.05 | willianmazzardo | sometimes … this error ocurres and crash my asterisk |
| 18:00.06 | willianmazzardo | [Apr 9 14:46:53] WARNING[10731]: sig_analog.c:3606 analog_exception: We're DAHDI/16-1, not |
| 18:00.20 | leifmadsen | ok I gotta leave bye |
| 18:00.21 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:01.13 | [ProB]CrazyMan | WIMPy: as you told me ... http://pastebin.com/y7evue8Z |
| 18:01.43 | willianmazzardo | anyone 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) |
| 18:02.38 | greenwolf_ | Each office phone when they dial 33 will ask them to enter the num they wish to pass as their callerid |
| 18:02.47 | WIMPy | [ProB]CrazyMan: I'v got the screening stuff at the end but as long as it's after portnum, it shouldn't matter. |
| 18:02.48 | greenwolf_ | Then it will ask them to dial the num they wish to call |
| 18:03.04 | greenwolf_ | Placing that call the values from step 1 for caller I'd are passed to the trunk |
| 18:03.08 | greenwolf_ | Out to the carrier |
| 18:03.37 | WIMPy | [ProB]CrazyMan: Maybe the type is set? Do you have a lcr log? |
| 18:03.43 | greenwolf_ | Rather then setting these values manually in the extensions.conf file every time |
| 18:03.58 | greenwolf_ | Does that clear up maybe,..hopefully :) |
| 18:05.28 | WIMPy | If you want to do it interavtively, Read() is your friend. |
| 18:05.55 | [ProB]CrazyMan | WIMPY: doesnt look like http://pastebin.com/rPUbSA2A |
| 18:06.24 | gusto | what are these dahdi cards for again? |
| 18:06.46 | gusto | i can not find any info on that somehow |
| 18:06.48 | [TK]D-Fender | gusto, Analog phones/lines |
| 18:06.52 | gusto | well |
| 18:07.02 | *** join/#asterisk joshaidan (~brianj@S010698fc113e438d.tb.shawcable.net) |
| 18:07.14 | gusto | but i have an ATA adapter like PAP2T, is that not sufficient for everyone? |
| 18:07.26 | [TK]D-Fender | gusto, You can't plung PHONE LINES into that. |
| 18:07.29 | [TK]D-Fender | pulg* |
| 18:07.32 | gusto | aha |
| 18:07.42 | gusto | well, i do not need to |
| 18:07.53 | [TK]D-Fender | gusto, You != Everyone |
| 18:08.44 | gusto | of 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 |
| 18:08.59 | gusto | so ... someone needs that ... ok, but for what? |
| 18:09.37 | gusto | s/not/ / |
| 18:09.45 | greenwolf | So the right direction for me is gosub() ? |
| 18:10.08 | greenwolf | I will take time to read and learn I just need to know what I should learn to achieve that |
| 18:10.16 | [TK]D-Fender | gusto, ... for what? ... to use ANALOG LINES with Asterisk. |
| 18:10.26 | [TK]D-Fender | gusto, How is this a difficult concept? |
| 18:10.50 | [TK]D-Fender | gusto, And you don't plug "VoIP" into an analog line. |
| 18:11.05 | gusto | it is not difficult, it is just something i can not imagine right |
| 18:11.36 | [TK]D-Fender | Guess you've never used a modem before. Or a fax machine. Or an older PBX. Or an answering machine... |
| 18:11.47 | gusto | of course i did |
| 18:11.55 | gusto | i even broke some of them |
| 18:12.03 | [TK]D-Fender | I am not surprised |
| 18:12.08 | gusto | yes |
| 18:12.57 | WIMPy | [ProB]CrazyMan: Looks like that search/replace thing doesn;t work for me, either. |
| 18:12.58 | [TK]D-Fender | gusto, 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... |
| 18:13.09 | WIMPy | wonders if there's somethig wrong or if there's a bug. |
| 18:13.11 | gusto | so 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-Fender | gusto, so that ASTERISK talks to the phone line. |
| 18:13.35 | WIMPy | [ProB]CrazyMan: You can do it in the dialplan, off course. |
| 18:14.12 | [ProB]CrazyMan | WIMPy: 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 ;() |
| 18:14.21 | p3nguin | greenwolf: No, GoSub is not the right thing. You want Read(). |
| 18:14.27 | [ProB]CrazyMan | WIMPy: thx for your support the last two days |
| 18:15.21 | WIMPy | [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. |
| 18:15.48 | greenwolf | <PROTECTED> |
| 18:16.06 | gusto | [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? |
| 18:16.17 | greenwolf | Then i integrate read() with set(callerid) |
| 18:16.46 | [TK]D-Fender | gusto, It lets * use your analog lines. |
| 18:17.49 | gusto | [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? |
| 18:17.58 | greenwolf | Ok I see now again thanks penguin |
| 18:18.08 | [TK]D-Fender | gusto, People who have them will want to use them |
| 18:19.37 | gusto | [TK]D-Fender: cool |
| 18:19.47 | gusto | [TK]D-Fender: how much does such a DAHDI card cost? |
| 18:20.11 | [TK]D-Fender | gusto, As much as the company that sells them asks for and yuo pay. |
| 18:20.20 | [TK]D-Fender | you* |
| 18:20.40 | gusto | [TK]D-Fender: so a lot |
| 18:20.57 | [TK]D-Fender | gusto, If that's what they ask, and that's what you pay. |
| 18:21.12 | [TK]D-Fender | gusto, And that's what you consider "a lot" |
| 18:22.14 | gusto | [TK]D-Fender: i would never give 200++ eur for only being able to talk to my analogue phone line (or four of them) |
| 18:23.03 | p3nguin | greenwolf: http://pastebin.com/smqxhRZ4 |
| 18:23.23 | [TK]D-Fender | gusto, Shouldn't cost 200eur for 1 phone line.... |
| 18:23.40 | WIMPy | gusto: If you're paying in EUR, you probably don't want (have to) to use analog anyway. |
| 18:24.00 | gusto | WIMPy: what currency should i pay instead? |
| 18:24.30 | WIMPy | If you pay in US$, you may want to. |
| 18:24.35 | [TK]D-Fender | facepalms.... |
| 18:24.58 | gusto | WIMPy: what do they have better in the US for an analog? cheaper analogue connection fees, or what? |
| 18:25.19 | [TK]D-Fender | gusto, You clearly just don't seem to get it... |
| 18:25.38 | [TK]D-Fender | gusto, this was not a currency question |
| 18:25.40 | gusto | [TK]D-Fender: i pretty much figured out that i am not getting it |
| 18:25.42 | WIMPy | No, but better alternatives are rare and expensive there. |
| 18:25.50 | WIMPy | Unlike in Europe. |
| 18:26.18 | [TK]D-Fender | gusto, WIMPyIs alluding that outside of North America most places offer ISDN instead of analog lines and would be a better choice. |
| 18:26.19 | gusto | WIMPy: ah you mean that they do not have VoIP, or ISDN, or something? |
| 18:26.39 | [TK]D-Fender | gusto, And this has nothing to do with "VoIP" |
| 18:26.46 | [TK]D-Fender | gusto, Stop thinking that everything does. |
| 18:26.52 | gusto | ok ok |
| 18:26.54 | [TK]D-Fender | gusto, We are talking LINES here. |
| 18:27.08 | [TK]D-Fender | gusto, And there are DADHi-based cards for ISDN as well. |
| 18:27.17 | [TK]D-Fender | BRI & PRI |
| 18:28.25 | *** join/#asterisk TimeRider (~steve@92.40.225.95.threembb.co.uk) |
| 18:28.42 | gusto | well, 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 |
| 18:28.50 | WIMPy | ... and cheaper. |
| 18:29.02 | gusto | WIMPy: cheaper where? |
| 18:29.45 | WIMPy | A BRIs is usually much cheaper than two analog lines. |
| 18:29.58 | gusto | what is BRI again? |
| 18:30.10 | KNERD | not in the US |
| 18:30.14 | WIMPy | And if you have ISDN, DAHDI is only one of the possibilites to connect it to Asterisk. |
| 18:30.30 | WIMPy | KNERD: Scroll up :-) |
| 18:30.47 | WIMPy | BRI = 2 channel ISDN line. |
| 18:30.52 | KNERD | i know..EUROPE |
| 18:30.57 | KNERD | but in US much cheaper |
| 18:31.18 | gusto | cool |
| 18:31.41 | gusto | and why is it still a question of money how much analogue telephone lines you want to use? |
| 18:31.47 | KNERD | ISDN is about $90 US while you could get 2 basic analog for $40US |
| 18:32.16 | gusto | i mean there are overcapacities in germany according to http://cre.fm/cre191 |
| 18:32.21 | WIMPy | So even alalog is expensive? |
| 18:32.41 | gusto | KNERD: what is that for a price? a monthly price or what? |
| 18:33.02 | KNERD | it's about $20 a month for one analog line |
| 18:33.11 | gusto | KNERD: that is too much |
| 18:33.11 | [TK]D-Fender | Dependsing where you are. |
| 18:33.23 | KNERD | on average that is the price |
| 18:33.29 | gusto | well, i would not give away shit like more than 10 eur per one line |
| 18:33.32 | WIMPy | Ah, ok, that was for two. Ok. |
| 18:33.39 | KNERD | so 2 channel ISDN for $90 is cheaper? |
| 18:33.40 | gusto | of course |
| 18:33.41 | WIMPy | gusto: That article is not about telephony. |
| 18:34.02 | gusto | WIMPy: however, but they talk about telephone lines in the podcast |
| 18:34.05 | KNERD | while 2 analog POTS for about $40 is more? hmmmm |
| 18:34.09 | [TK]D-Fender | WIMPy, none of his neurons are firing synchronously |
| 18:34.39 | WIMPy | Ja, it's a bit jumpy... |
| 18:35.40 | [TK]D-Fender | WIMPy, 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 * |
| 18:36.58 | WIMPy | At least we had the usual reminder why anyone might want to use analog at all. |
| 18:37.39 | gusto | [TK]D-Fender: so you do not understand it either? |
| 18:37.41 | [TK]D-Fender | WIMPy, 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-Fender | gusto, No, you seem to have issues putting this together. |
| 18:38.16 | gusto | [TK]D-Fender: i always have :-d |
| 18:38.23 | gusto | :-D |
| 18:38.35 | gusto | but when i put something together, it works |
| 18:38.52 | [TK]D-Fender | Except when you break it. And you've broken all sorts of things as you've mentioned. |
| 18:39.08 | gusto | [TK]D-Fender: well |
| 18:39.19 | gusto | [TK]D-Fender: was not my fault :-D |
| 18:40.03 | WIMPy | That's what they all say. |
| 18:41.18 | gusto | yes |
| 18:41.33 | gusto | that's why i never said that when something broke |
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| 18:50.29 | greenwolf | <PROTECTED> |
| 18:50.39 | greenwolf | That was very useful for me |
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| 19:00.23 | bent_screwdriver | anyone here have any luck running asterisk on an embedded os/hardware using PRI? |
| 19:03.39 | WIMPy | bent_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? |
| 19:05.20 | bent_screwdriver | WIMPy: 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? |
| 19:05.49 | WIMPy | Yes, that would require a beefy CPU. |
| 19:07.58 | bent_screwdriver | WIMPy: 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 |
| 19:08.34 | WIMPy | Ugh, g.729 sounds evil as well. |
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| 19:09.06 | WIMPy | Connect it to another box with a PRI card and loop the channels between them. |
| 19:10.14 | bent_screwdriver | what is > than g729? |
| 19:10.27 | WIMPy | I 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. |
| 19:10.37 | WIMPy | > in what sense? |
| 19:11.10 | bent_screwdriver | when bandwith considerations and quality are of importance |
| 19:11.39 | WIMPy | Stay with G.711 as is sent and receiveed on the line. |
| 19:12.13 | WIMPy | If you can't afford the bw, G.729 may be a good choice but a CPU hungry one. |
| 19:13.38 | bent_screwdriver | wasn't there some hardware module that offloads transcoding? I only have mini pci slots after the PRI so probably not an option anyways.... |
| 19:14.25 | WIMPy | Digium have a hardware solution, yes, but I'm pretty sure that mini pci will be a show stopper. |
| 19:15.08 | WIMPy | I'm not sure if anyone has mini pci pri cards perhaps. I know there are at least 4 BRI in mini pci. |
| 19:15.34 | WIMPy | PCIe might make things easier. |
| 19:17.09 | ectospasm | Digium doesn't carry mini PCI or mini PCIe anything. |
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| 19:18.44 | WIMPy | Mini PCIe would only be a mechanical issue. |
| 19:20.25 | [TK]D-Fender | WIMPy, OpenVOX has mini PCI/PCIe |
| 19:22.05 | *** join/#asterisk luckman212 (~irc@2001:470:8abb:0:49ee:2de6:6ad6:3b46) |
| 19:22.27 | WIMPy | But no PRI as it seems. |
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| 19:27.02 | gusto | hmm |
| 19:32.05 | [TK]D-Fender | WIMPy, Yup, just found the same cruising their full catalog. |
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| 19:36.34 | *** join/#asterisk thafreak (~thafreak@unaffiliated/thafreak) |
| 19:37.01 | thafreak | Can anyone recommend something to connect to analog lines? |
| 19:37.25 | thafreak | Are the wildcards the best route, or is there something better these days? |
| 19:37.38 | thafreak | very small phone system with < 4 lines |
| 19:38.01 | WIMPy | deja vu? |
| 19:38.03 | thafreak | i saw there are these sip->analog gateway boxes with like 4 ports |
| 19:39.53 | Henchman21 | im pissed at this spa3k i recently got, wont dialout |
| 19:40.04 | Henchman21 | gave up on it |
| 19:40.10 | [TK]D-Fender | thafreak, Are you expecting to stick with analog for a while? How many to start? Expected expansion? |
| 19:40.25 | [TK]D-Fender | Henchman21, You're configuring it wrong |
| 19:40.29 | Henchman21 | still answers the houseline and sends it to asterisk |
| 19:40.54 | thafreak | no expansion, 4 analog lines will be all that will ever be used |
| 19:41.00 | Henchman21 | nah i've tried quite a few different configs but it still acts goofy |
| 19:41.38 | Henchman21 | like ill send a call out it, then call the house line line and it dials the darn number in my ear |
| 19:41.53 | Henchman21 | it drops the call or something |
| 19:41.59 | thafreak | currently using an older wildcard with 4 fxo modules, but recently it seems a few got fried |
| 19:42.25 | Henchman21 | i dunno i started listening to its syslog debug output but i cant make heads or tails of it |
| 19:43.22 | [TK]D-Fender | thafreak, I'd say aim for a Sangoma B600d |
| 19:43.45 | [TK]D-Fender | thafreak, And and get some surge supressors for your lines. |
| 19:46.51 | thafreak | so the cards are still better than something like: Audiocodes MP108-FXO gateway? |
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| 19:49.24 | [TK]D-Fender | thafreak, Offers * a little more control over the lines, and are a fair bit cheaper |
| 19:51.40 | thafreak | seems that card is actually like $100 more than the audiocodes gateway... |
| 19:53.01 | [TK]D-Fender | thafreak, Depends where you shop I guess... |
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| 19:53.22 | [TK]D-Fender | thafreak, The AudioCodes should work fine for most things as well... |
| 19:54.03 | thafreak | are the grandstream ones junk? They seem to be pretty cheap (like $250) |
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| 19:58.24 | [TK]D-Fender | ~gs |
| 19:58.24 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:58.28 | [TK]D-Fender | ~grandstream |
| 19:58.28 | infobot | methinks 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-Fender | YMMV |
| 19:59.41 | thafreak | i've used their phones and ata's |
| 19:59.59 | thafreak | their ata's aren't horrible, but could be better i guess |
| 20:10.59 | gusto | ah |
| 20:11.25 | gusto | luckily i bought a second pap2t instead of grandstream |
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| 21:06.27 | ectospasm | we don't have much good to say about grandstream here |
| 21:06.57 | tm1000 | the spa112 is the newer version of the pap2t |
| 21:07.04 | tm1000 | so i hope you didnt get ripped off for an old unit |
| 21:07.10 | tm1000 | gusto: ^^ |
| 21:09.13 | gusto | no |
| 21:09.18 | gusto | it is quite cheap |
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| 21:11.14 | pyite_mac | is 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 |
| 21:11.23 | gusto | tm1000: is SPA112 for 50 eur OK? |
| 21:13.12 | tm1000 | gusto: sounds about right. they are really good for faxing |
| 21:13.20 | tm1000 | if you wanted t.38 |
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| 21:14.23 | gusto | tm1000: ok |
| 21:14.31 | gusto | tm1000: i bought that now as well :-D |
| 21:15.11 | tm1000 | they are good little units |
| 21:15.49 | gusto | yes yes |
| 21:15.55 | gusto | pap2t is too |
| 21:16.02 | gusto | i can use SOME of them |
| 21:16.48 | gusto | i 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/ |
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| 21:34.51 | greenwolf | <PROTECTED> |
| 21:35.02 | greenwolf | Use* |
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| 21:42.57 | drudge` | anyone like or recommend a free or cheap hudlite alternative? |
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| 22:16.59 | CyBeRxIxO | hi, i got a question, thanks for ur answer in advance |
| 22:17.20 | CyBeRxIxO | i just installed asterisknow, killed my elastix box |
| 22:18.09 | CyBeRxIxO | when i get the g729 codec from digium do i need to install anything else like procesor driver or something? |
| 22:18.40 | CyBeRxIxO | once i buy it do i get the support needed on my general configuration? |
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| 23:55.57 | Cubber | I 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. |
| 23:56.28 | Cubber | if 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. |
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| 23:59.28 | Cubber | asterisk[9779]: segfault at c18 ip b726627d sp b6d04cf4 error 4 in libpthread-2.13.so[b725d000+15000] |
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