| 00:01.35 | aliverius | any mISDN dev around? |
| 00:02.09 | aliverius | there is a problem with a header shipped with misdn |
| 00:02.13 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 00:02.39 | aliverius | which is resolved when i use a header by the same name in the hisax linux headers |
| 00:04.50 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 00:27.13 | *** join/#asterisk WebMaxtor (n=rmaslank@pool-71-126-13-183.bflony.east.verizon.net) |
| 00:28.19 | *** part/#asterisk WebMaxtor (n=rmaslank@pool-71-126-13-183.bflony.east.verizon.net) |
| 00:29.36 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 00:33.17 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:33.37 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:48.18 | *** part/#asterisk spiri (n=spiri@S01060002553240a8.vc.shawcable.net) |
| 01:07.03 | *** join/#asterisk gazzerh (n=garryh@93-97-187-150.zone5.bethere.co.uk) |
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| 01:25.27 | *** part/#asterisk aliverius (n=aliveriu@chal530-a049.home.otenet.gr) |
| 01:31.57 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:35.17 | *** join/#asterisk wick2o (n=wick2o@72.25.60.14) |
| 01:35.19 | wick2o | hello |
| 01:39.07 | wick2o | anyone have exp install asterisks in a home setting? |
| 01:39.17 | wick2o | asterisk rather |
| 01:39.18 | *** join/#asterisk jylan (n=jylan_wy@27-140-static.skymesh.net.au) |
| 01:39.46 | *** part/#asterisk jylan (n=jylan_wy@27-140-static.skymesh.net.au) |
| 01:49.01 | *** join/#asterisk rjune (n=rjune@38.103.117.250) |
| 01:49.32 | rjune | I have an extension ringing an external phone, is there a way to pass incoming caller id information along to that phone? |
| 01:50.04 | rjune | i.e. I call in and dial 714, which forwards the call to a cell phone. but the cell phone simply says "call" no indication of number calling |
| 01:53.14 | wick2o | dont know, I seem to be the only one awake at the moment |
| 01:54.46 | rjune | yeah, I'll come back tomorrow |
| 01:55.09 | wick2o | is this a normally just a M-F channel? |
| 01:55.24 | rjune | don't think so |
| 01:55.34 | rjune | I've had questions answered on Sat befor |
| 01:56.28 | wick2o | I have the asterisk applicnce, we bought it at work to compair it to a cisco CME |
| 01:56.48 | wick2o | we ended up going with the CME so now its on my desk at home. I'm thinking about wiring it into my house |
| 01:57.18 | rjune | I've got a system with asterisk now. |
| 01:57.22 | rjune | mostly been ok |
| 01:57.40 | rjune | these guys in here have been helpful with general here's what you need |
| 01:57.47 | rjune | which lets me work out how to do it in *now |
| 01:57.58 | *** join/#asterisk Aiatek (n=munoz@190.159.121.197) |
| 01:59.58 | wick2o | I have no real use for it to be honest, I just tired of looking at it in the box |
| 02:00.36 | wick2o | I'm tring to figure out if i can have it mixed with my normal house phones without relacing them |
| 02:01.21 | rjune | good luck |
| 02:01.37 | wick2o | that bad huh? |
| 02:03.44 | rjune | no clue |
| 02:03.51 | rjune | I know some devices let you do it. |
| 02:03.58 | rjune | but I don't know if * will |
| 02:09.57 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 02:18.11 | *** join/#asterisk mrluap (n=test@d75-156-11-9.bchsia.telus.net) |
| 02:18.19 | mrluap | Quick question, |
| 02:18.32 | mrluap | How do I get extension to ring longer then the standard 7 or 8 seconds? |
| 02:19.33 | Aiatek | exten => 2500,1,Dial(SIP/2500,30) |
| 02:19.48 | Aiatek | it will ring for 30 seconds |
| 02:20.39 | mrluap | will that also allow it to forward for 30 seconds? |
| 02:21.04 | Aiatek | foward to what? |
| 02:21.20 | mrluap | to another line |
| 02:21.23 | mrluap | or to an external line? |
| 02:21.45 | Aiatek | ~book |
| 02:21.46 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:22.46 | Aiatek | <mrluap> go an read it, later come to ask |
| 02:23.25 | wick2o | Aiatek: thanks, I could use that as well |
| 02:24.15 | mrluap | Thanks |
| 02:24.27 | mrluap | so what i asked doesnt have a quick answer? |
| 02:24.40 | Aiatek | yeah, the book |
| 02:24.45 | rjune | it's a good book, I have "Hacking VOIP" too |
| 02:24.50 | Aiatek | thats a shortcut |
| 02:25.04 | *** join/#asterisk gazzerh (n=garryh@93-97-187-150.zone5.bethere.co.uk) |
| 02:30.20 | rjune | what's a shortcut? |
| 02:36.02 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.73) |
| 02:38.14 | rob0 | Well, if you're driving from Nashville to Huntsville, you can get off I-65 at the last TN exit and go through Ardmore to get AL-53 (Jordan Ln.) But it might be quicker (time) to just continue on to I-565. |
| 02:38.38 | *** part/#asterisk ScribbleJ (n=sj@99-35-164-150.lightspeed.dwgvil.sbcglobal.net) |
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| 02:50.47 | rjune | rob0, I'm not sure what that analogy means |
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| 02:53.02 | *** join/#asterisk chendy_ (n=chatzill@58.60.218.202) |
| 02:54.50 | rob0 | Not much ... just one of many answers to "what's a shortcut?" |
| 02:55.55 | Qwell | rob0: but make sure you take the *last* exit, and not the one before it |
| 02:56.31 | Qwell | I mean, unless you want to stop at the truckstop strip club... |
| 02:56.54 | rob0 | And then it's definitely NOT a shortcut! |
| 02:59.56 | rue_mohr | I would like a watchdog on my asterisk system, something that toggles a bit on either a serial or parallel port, I can have hardware re-shunt lines if there is a failure, anyone know of code in asterisk that can help me along? |
| 03:07.13 | killfill | hey, how do i enable "BUSYDETECT_TONEONLY"?.. i see there are reference for it in main/dsp.c |
| 03:07.28 | killfill | but how do i enable this?.. dont see anything in any Makefile.. :S |
| 03:13.33 | killfill | i dont know if this flags are opsolete.. maybe they are.. |
| 03:23.51 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:39.28 | *** part/#asterisk Aiatek (n=munoz@190.159.121.197) |
| 03:44.13 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
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| 04:05.24 | *** join/#asterisk panicman (i=panicman@122.102.33.66) |
| 04:06.42 | panicman | hello, i'm trying to configure PRI connection with Dahdi, but its showing some error |
| 04:06.54 | panicman | Extension '35' in context 'group1' from '100' does not exist. Rejecting call on channel 0/1, span 1 |
| 04:07.07 | panicman | any helper care to help |
| 04:07.15 | *** join/#asterisk x1nux (n=x1nux@unaffiliated/x1nux) |
| 04:07.16 | x1nux | hi |
| 04:07.23 | x1nux | i need a simple help |
| 04:07.32 | [TK]D-Fender | panicman: that error is extremely explicit in telling you whats wrong |
| 04:07.58 | x1nux | i need to call a extensions, and play a sound automatic !!! |
| 04:08.01 | x1nux | can helpme |
| 04:08.03 | [TK]D-Fender | panicman: You don't have an exten in [group1] to match "35" |
| 04:08.54 | panicman | [isdn-callin] |
| 04:08.54 | [TK]D-Fender | rue_mohr: Code in * to do what part of that? |
| 04:08.54 | panicman | exten => _x.,1,Answer() |
| 04:08.54 | panicman | exten => _x.,2,Playback(hello-world) |
| 04:08.55 | panicman | exten => _x.,3,Hangup |
| 04:08.55 | panicman | [nahidtest] |
| 04:08.55 | panicman | exten => 110,1,Answer() |
| 04:08.57 | panicman | exten => 110,2,Dial(Dahdi/g2/:35) |
| 04:08.59 | panicman | enten => 110,3,Hangup() |
| 04:09.01 | panicman | this is the sample config |
| 04:09.05 | [TK]D-Fender | panicman: PASTEBIN, do not spam in here |
| 04:09.07 | [TK]D-Fender | ~pb |
| 04:09.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 , or , http://bin.cakephp.org/ |
| 04:09.10 | [TK]D-Fender | ^^^^^^^ |
| 04:09.12 | panicman | sorry |
| 04:09.21 | [TK]D-Fender | panicman: look at the CONTEXT that call is landing in. |
| 04:09.39 | [TK]D-Fender | pancthe call lands in [group1] <-------- you don't seem to HAVE one. |
| 04:09.40 | panicman | trying to test it , Port 1 connect to port 2 |
| 04:14.00 | x1nux | panicman, |
| 04:14.11 | x1nux | can you helpme ? |
| 04:14.18 | x1nux | i need to make a call |
| 04:14.28 | x1nux | a extensions 101, |
| 04:14.35 | x1nux | and them play a sound |
| 04:14.50 | x1nux | But, the extensions is a client ... in line .. |
| 04:14.56 | x1nux | on line |
| 04:15.16 | [TK]D-Fender | x1nux: Go search for "call files" on the WIKI |
| 04:15.18 | [TK]D-Fender | ~wikis |
| 04:15.19 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 04:17.34 | x1nux | :w |
| 04:25.55 | [TK]D-Fender | Yup, its the weekend, when all the crazies come out... |
| 04:28.57 | jaytee | lol |
| 04:29.23 | jaytee | "spare a few modems for my FXO channel bank?" |
| 04:30.36 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 04:31.34 | [TK]D-Fender | ~soundcard |
| 04:31.35 | infobot | extra, extra, read all about it, audio is usually a codec issue. start with trying to set 'disallow=all' and 'allow=alaw' in sip.conf or the channel's config file if not using sip |
| 04:32.01 | [TK]D-Fender | carrar: Dammit, whats that infor-let you made?! |
| 04:32.30 | nkohh | you will find only one 'm' in 'damnit' |
| 04:32.57 | nkohh | zing! |
| 04:33.57 | [TK]D-Fender | nkohh: And while there is no "I" in "team", there is a "U" in "dumbass" :p |
| 04:37.34 | joat | nkohh: depends on your dialect... where I grew up, it was spelled "dammit" and we didn't use n's, l's, r's, g's, or h's.... |
| 04:37.41 | joat | heh |
| 04:37.47 | joat | d's and t's were optional |
| 04:41.23 | nkohh | <PROTECTED> |
| 04:41.31 | nkohh | there are 27 "s"s in the word "mississippi" |
| 04:41.50 | jaytee | so you can't count either |
| 04:41.51 | nkohh | there are 27 occurances of the letter 's' in the word mississippi |
| 04:44.03 | LeddyHM | there is an me in team though ;) |
| 04:48.15 | nkohh | there is also tame, mate, meat, at, and at me |
| 04:52.12 | [TK]D-Fender | nkohh : And more fomally it would be "damn it" as 2 words, wishing damnation upon whatever woeful object has offended the narrator :) |
| 04:53.09 | nkohh | yeah, but the way i see it.. i use the space bar more than any other key on my keyboard (except maybe delete lol) and I don't want to wear it out. |
| 04:53.20 | nkohh | so i omit spacestoconserve key presses |
| 04:55.56 | [TK]D-Fender | nkohh: And you're anal retentive to spare the atmosphere of the greenhouse gasses of your flatulence as well no doubt :) |
| 04:56.26 | nkohh | au contraire, i don't believe in global warming |
| 04:56.35 | nkohh | and i drive a gas guzzler |
| 04:56.40 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 04:56.46 | [TK]D-Fender | nkohh: Ok... then you're still full of shit :p |
| 04:57.03 | [TK]D-Fender | ZING |
| 04:59.31 | nkohh | i really enjoy the music of Basshunter |
| 05:37.58 | *** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:38.07 | *** join/#asterisk blaxthos (n=blaxthos@onyx.krisp.com) |
| 05:38.14 | blaxthos | anyone have advice on IP phone + asterisk + voip service providers for the house ? |
| 05:38.26 | *** join/#asterisk panicman (i=panicman@122.102.33.66) |
| 05:45.44 | AlmightyOatmeal | blaxthos: i have softphone + * + sip access with a POTS number |
| 05:46.44 | blaxthos | i need to record all inbound/outbound calls, and i'd like to have a cordless extension |
| 05:46.50 | blaxthos | either by asterisk + ip phone |
| 05:47.09 | blaxthos | or asterisk + POTS adapter + standard cordless phone |
| 05:47.25 | AlmightyOatmeal | well i'm sure that could be handled by *, i'm just not sure how |
| 05:48.28 | blaxthos | ironically, i lived next door to mark in undergrad |
| 05:48.47 | [TK]D-Fender | blaxthos: Linksys SPA-3102 + cordless phone |
| 05:50.35 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 05:51.20 | blaxthos | what service provider(s) are easy on the * ? |
| 05:51.44 | [TK]D-Fender | blaxthos: How is one "difficult"? |
| 05:52.36 | blaxthos | i'd imagine some SP's would try to force you to buy their terminal adapters, use their native stuff etc |
| 05:54.57 | [TK]D-Fender | blaxthos: that would be "Vonage" |
| 05:55.03 | [TK]D-Fender | ~itsplist-us |
| 05:55.04 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 05:56.01 | blaxthos | tnx |
| 05:57.52 | *** join/#asterisk Subdolus (n=subby@vulcan.it.cx) |
| 06:10.22 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 06:11.57 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 06:14.44 | [TK]D-Fender | ok, its late. Checkout time. LAter all |
| 06:23.20 | *** join/#asterisk astrutt (i=astrutt@pinky.ratman.org) |
| 06:26.01 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 06:41.02 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:41.07 | AlmightyOatmeal | once i dial in to *, i can't create an extion to dial another external number using sip? |
| 06:41.28 | AlmightyOatmeal | exten => 4,2,dial(SIP/${EXTEN}@sip.broadvoice.com/<ext number>) |
| 06:41.38 | AlmightyOatmeal | i get WARNING[65484]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 06:43.03 | AlmightyOatmeal | whoops, now i get a number not in service after adding ,30 after my sip.broadvoice.com |
| 07:13.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:27.59 | *** join/#asterisk AdMegaMan (n=amega@41.232.106.157) |
| 07:28.46 | AdMegaMan | hi all.. need some help to configure my openvox A1200P .. i keep getting: mylin@mylin:~$ sudo dahdi_cfg -vvvvv |
| 07:28.46 | AdMegaMan | DAHDI Tools Version - 2.0.0 |
| 07:28.46 | AdMegaMan | DAHDI Version: 2.0.0 |
| 07:28.46 | AdMegaMan | Echo Canceller(s): |
| 07:28.46 | AdMegaMan | Configuration |
| 07:28.48 | AdMegaMan | ====================== |
| 07:28.50 | AdMegaMan | Channel map: |
| 07:28.52 | AdMegaMan | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
| 07:28.54 | AdMegaMan | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 07:28.56 | AdMegaMan | 2 channels to configure. |
| 07:28.58 | AdMegaMan | Changing signalling on channel 1 from Unused to FXO Kewlstart |
| 07:29.00 | AdMegaMan | DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 07:29.02 | AdMegaMan | Did you forget that FXS interfaces are configured with FXO signalling |
| 07:29.04 | AdMegaMan | and that FXO interfaces use FXS signalling? |
| 07:29.06 | AdMegaMan | sorry ... |
| 07:30.26 | AdMegaMan | i have 2 modules on my card .. Tel1: Green Module (FXS).. Tel2: Red Module (FXO) .. as far as i understood.. for FXS module i set it up to use FXO signaling..and for FXO, i set it up with FXS signaling |
| 07:30.34 | AdMegaMan | can some one kindly help out? |
| 07:46.51 | AlmightyOatmeal | did you consult the oreilly manual? |
| 07:47.10 | AlmightyOatmeal | http://www.asteriskdocs.org/ |
| 07:58.12 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
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| 08:21.13 | tzafrir_laptop | dahdi 2.0.0 . nice. |
| 08:24.27 | AlmightyOatmeal | is that a pun? |
| 08:33.59 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:45.56 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
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| 09:05.02 | MT`AwAy | is asterisk 1.6 handling correctly video codec negociation, or do we still need to force only one codec ? |
| 09:14.02 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 09:47.13 | *** join/#asterisk luckylucy (n=luckyluc@adsl-75-13-76-221.dsl.ksc2mo.sbcglobal.net) |
| 09:49.53 | luckylucy | hello everyone, i am thinking of setting up some type of a phone system for my VERY small biz, and think this might be the way to go. i basically want the three of us to still be able to use our cell phones for work, but have anyone calling for business call a single number, and get some type of a "business like" message. can this forward calls to cell phones? |
| 09:50.19 | luckylucy | i am thinking of using that ooma thing for the phone connection and drop the phone line too... |
| 10:01.36 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 10:04.38 | *** join/#asterisk Erol_ (n=x@88.235.60.22) |
| 10:04.42 | Erol_ | hi |
| 10:04.45 | luckylucy | hi |
| 10:04.54 | Erol_ | i am new to voip and trying to understand some basics |
| 10:05.02 | luckylucy | me too! |
| 10:05.10 | Erol_ | =) |
| 10:05.19 | luckylucy | :P |
| 10:05.29 | Erol_ | there are some apliances called as voip gateway |
| 10:05.36 | Erol_ | what are they used for? |
| 10:05.49 | luckylucy | good question :) |
| 10:06.02 | luckylucy | i'm going to be no help. sorry. |
| 10:06.09 | Erol_ | for example i setup asterisk and why whould I need a voip gateway? |
| 10:06.10 | *** join/#asterisk Failrar (n=Failrar@tunnel1088.ipv6.xs4all.nl) |
| 10:11.21 | luckylucy | Erol: do you know it asterisk can do call forwarding based on an incoming call's "extension selection" |
| 10:12.48 | Erol_ | yes |
| 10:14.38 | luckylucy | to a cell phone? |
| 10:17.28 | Erol_ | you mean can asterisk forward calls to an ext to a cellphone? |
| 10:18.56 | luckylucy | well, i am thinking of getting ooma for my very small biz to ditch the phone, but i want all "my" calls to go through the main number to my cell phone as i am never in the office. |
| 10:19.47 | luckylucy | the idea is to have one business number that anyone can call, but right now we have a business number and 2 guys who only use their cells. |
| 10:19.50 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128544799.dsl.bell.ca) |
| 10:24.27 | *** join/#asterisk Subdolus (n=subby@vulcan.it.cx) |
| 10:27.51 | kaldemar | Erol_: voip gateways are used to have a voip connection to some other technique. if you only need voip, asterisk is all you need. |
| 10:28.53 | kaldemar | luckylucy: you can do pretty much anything you want with calls that come to your asterisk. play a message and send the call to a cell phone, yes. |
| 10:29.49 | luckylucy | that is fantastic, do you have to have the special analog phone cards, or can you just use a modem... excuse my lack of telephony knowledge... |
| 10:30.04 | Erol_ | kaldemar: are all of the voip phones PoE? |
| 10:30.49 | Erol_ | or PoE is extra for some voip phones? |
| 10:32.32 | luckylucy | not yet, they can be... if it is cheaper... i want to have this new ooma thing be the "front" end for the calls, with two internal phones and have other "extensions" forwarded to cell phones. i want concurrent calls and voice mail on the server if the cell doesn't pick up... i have no knowledge of any of this thus far, this is more of a "feasibility study" |
| 10:32.50 | luckylucy | ooma: http://www.newegg.com/Product/Product.aspx?Item=N82E16833888001 |
| 10:33.00 | luckylucy | looks fantastic, i hate phone bills. |
| 10:38.16 | MT`AwAy | just a question regarding asterisk 1.6.1.0 and codecs order, shouldn't video codecs also appear in "codec order" ? http://pastebin.com/m5afc9c45 |
| 10:38.32 | MT`AwAy | (output of "sip show settings") |
| 10:39.19 | AlmightyOatmeal | so how would one go about submitting a cnam/lidb listing... |
| 10:39.22 | AlmightyOatmeal | my sip provider doesn't subscribe to any cnam databases |
| 10:39.26 | AlmightyOatmeal | and i don't want to subscribe to a cnam database for queries |
| 10:40.09 | *** join/#asterisk H3XiL3D (n=H3X@unaffiliated/hexiled) |
| 10:40.10 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 10:47.17 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.102.189) |
| 10:47.27 | DelphiWorld | hello |
| 10:47.43 | DelphiWorld | please how i can setup asterisk 1.6.X to a debian box? |
| 10:47.53 | AlmightyOatmeal | DelphiWorld: heard of google.com? |
| 10:48.15 | AlmightyOatmeal | DelphiWorld: also reference http://www.asteriskdocs.org/, excellent book |
| 10:48.45 | DelphiWorld | AlmightyOatmeal: thanks, but i have a problem with accessing the Web |
| 10:48.50 | DelphiWorld | AlmightyOatmeal: i'm a blind user |
| 10:49.12 | Erol_ | what card do i need to use to connect PRI line to asterisk? |
| 10:49.12 | AlmightyOatmeal | you seem to handle irc just fine |
| 10:49.53 | DelphiWorld | AlmightyOatmeal: i'm using a screen reader |
| 10:50.01 | DelphiWorld | AlmightyOatmeal: but i have a problem accessing a web pages |
| 10:50.03 | AlmightyOatmeal | DelphiWorld: as long as you have some piece of software that will allow you to read PDF files then go to the asteriskdocs.org website and download the orielly ebook |
| 10:50.09 | DelphiWorld | AlmightyOatmeal: irc is text based no graphic |
| 10:50.20 | AlmightyOatmeal | and so is most of that book |
| 10:50.30 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 10:50.45 | DelphiWorld | AlmightyOatmeal: tel me only if i can setup asterisk using Apt-get |
| 10:52.06 | AlmightyOatmeal | DelphiWorld: yes, which is covered in that book |
| 10:52.19 | DelphiWorld | AlmightyOatmeal: ok |
| 10:52.47 | AlmightyOatmeal | DelphiWorld: here is a direct link: http://downloads.oreilly.com/books/9780596510480.pdf |
| 10:53.02 | DelphiWorld | AlmightyOatmeal: please tel me what is the actualy supported Channel in asterisk including IAX2/SIP/H.323? any other supported protocol? |
| 10:54.37 | kaldemar | Erol_: definitely not all phones use PoE |
| 10:55.03 | AlmightyOatmeal | DelphiWorld: IAX2, SIP, H.323, SCCP, MGCP, and UNISTIM |
| 10:55.26 | Erol_ | kaldemar: and what card do i need to connect asterisk a pri line= |
| 10:55.40 | kaldemar | luckylucy: you can't use a modem, you need an analog card. |
| 10:56.24 | kaldemar | Erol_: erm, a PRI card. there are plenty, for example http://www.digium.com/en/products/digital/ |
| 10:57.40 | Erol_ | kaldemar: which one is better, using a pri voip gateway connected to asterisk or connect pri directly to asterisk |
| 10:58.24 | DelphiWorld | asterisk support skype Integration? |
| 10:58.25 | kaldemar | i like apples over oranges. :) |
| 10:58.57 | AlmightyOatmeal | DelphiWorld: i believe there has been alpha support for it, it may be integrated in 1.6 but i cannot say for sure |
| 10:59.20 | kaldemar | DelphiWorld: no proper support. |
| 10:59.24 | DelphiWorld | AlmightyOatmeal: Thanks |
| 10:59.40 | DelphiWorld | AlmightyOatmeal: but please only tel me if that support multiple Call? |
| 10:59.46 | kaldemar | asterisk itself has no support for skype at all. |
| 10:59.58 | AlmightyOatmeal | DelphiWorld: of course |
| 11:00.23 | AlmightyOatmeal | depends on your configuration and your service provider(s) |
| 11:00.42 | *** join/#asterisk m-i-l-a-n (n=milan@92.204.99.129) |
| 11:00.55 | DelphiWorld | AlmightyOatmeal: but multiple call in one instance of skype? |
| 11:01.36 | AlmightyOatmeal | DelphiWorld: as kaldemar said there is no proper support for skype, and i don't know anything about skype support for asterisk except there have been attempts to make patches to enable some kind of support.. that's all i know |
| 11:02.16 | DelphiWorld | AlmightyOatmeal: thank you anyway |
| 11:02.37 | DelphiWorld | AlmightyOatmeal: also i have a very big problem with SIP here in my ISP: |
| 11:03.03 | DelphiWorld | if i use Sip over TCP is work properly but over UDP is not working (Sam port, 5060) |
| 11:04.09 | *** join/#asterisk plq (n=plq@193.255.135.1) |
| 11:04.45 | luckylucy | what determines if you have the hardware to support concurrent calls? |
| 11:05.08 | luckylucy | do you need two incoming phone numbers? |
| 11:05.48 | DelphiWorld | luckylucy: me? |
| 11:05.54 | luckylucy | anyone.... |
| 11:06.22 | DelphiWorld | luckylucy: how to? |
| 11:07.01 | luckylucy | how to what? |
| 11:07.28 | DelphiWorld | luckylucy get two phone number? |
| 11:08.20 | luckylucy | no, do i need two? |
| 11:09.16 | DelphiWorld | luckylucy yes |
| 11:09.20 | luckylucy | do you need two phone lines for two phones to have concurrent calls, or does it work like dsl, multiple computers doing different things? |
| 11:09.50 | DelphiWorld | luckylucy: work in both |
| 11:09.57 | DelphiWorld | luckylucy: depond on your provider |
| 11:10.12 | DelphiWorld | is it pocible to get multiple pphone number grouped in one line |
| 11:10.12 | luckylucy | i want to get ooma for my phone. |
| 11:10.36 | luckylucy | i don't want a re-occuring phone bill. |
| 11:11.06 | DelphiWorld | luckylucy: what you need exactly |
| 11:12.40 | luckylucy | i need a single phone number for my small biz that people can call, select an extension, and then get forwarded to an internal phone or an external cell phone depending on the extension they choose. voicemail would be nice too. |
| 11:14.45 | luckylucy | possible with ooma + linux running asterisk? do i need special hardware? seems digium's hardware is really expensive. |
| 11:16.47 | DelphiWorld | luckylucy: i'm no sur |
| 11:17.06 | DelphiWorld | luckylucy: but about the phone number, that pocible if you get a line that accept multiple call |
| 11:30.28 | *** join/#asterisk Subdolus (n=subby@vulcan.it.cx) |
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| 12:29.29 | mmlj4 | luckylucy: get a VoIP number, no need for hardware... take a look at teliax.com |
| 12:30.09 | mmlj4 | and yes, asterisk can do what you want |
| 12:30.36 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
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| 13:34.29 | Tond | Hi, has anyone here used MYSQL() cmd in dialplan before? |
| 13:34.59 | Tond | I am trying to get some benchmark info on it and see how many hits it can handle per second without crashing a system |
| 13:42.16 | Tond | Hm.. |
| 13:42.43 | Tond | So no info on Asterisk connection calls to MySQL? |
| 13:45.42 | gr0mit | i use it |
| 13:46.07 | gr0mit | but very low usage |
| 13:46.33 | Tond | Ya I have used it in low uasage as well, but i want to know if i should move to higher number of calls |
| 13:49.16 | Tond | I don't know where to look as it doesn't seem that popular |
| 13:53.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:53.39 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 13:57.55 | MT`AwAy | anyone managed to get video working on SIP with asterisk 1.6 ? |
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| 14:52.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:54.28 | leifmadsen | hey all, I might work on another asterisk article today -- anyone have a suggestion? |
| 15:11.51 | xuser | changes between 1.4 and 1.6? |
| 15:28.46 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279505454.dsl.bell.ca) |
| 15:31.16 | MT`AwAy | how to get video working with asterisk 1.6 ? XD |
| 15:53.31 | nkohh | very carefully |
| 15:58.51 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 16:00.35 | *** join/#asterisk Erol_ (n=x@88.235.60.22) |
| 16:01.50 | killfill | how do i enable BUSYDETECT_* switches on current asterisk 1.4.x?.. seems to be deprecated? |
| 16:06.35 | *** join/#asterisk viyyer_ (n=viyyer@thames.mum.edu) |
| 16:08.02 | Erol_ | i see that trixbox recomends sangoma interface cards |
| 16:08.10 | Erol_ | and doesnt mention about digium cards |
| 16:08.17 | Erol_ | but its based on asterisk so isnt that weird? |
| 16:11.03 | *** join/#asterisk TerrenceLam (n=Terrence@143.89.210.186) |
| 16:11.21 | *** part/#asterisk TerrenceLam (n=Terrence@143.89.210.186) |
| 16:13.15 | [TK]D-Fender | Erol_: No relationship at all |
| 16:13.39 | Erol_ | [TK]D-Fender: sorry, couldnt understand? |
| 16:14.42 | Erol_ | [TK]D-Fender: doesnt digium and sangoma produce same type of hardware= |
| 16:15.18 | [TK]D-Fender | trixbox is not "based" on Asterisk, it USES it. The hardware they recommend also has nothing to do with Asterisk itself in that it only needs to wrok. This is not a case of brand loyalty |
| 16:15.29 | coppice | so one group recommend Pepsi and some recommend Coke |
| 16:15.52 | [TK]D-Fender | Erol_: And jsut because Rold makes sound synth modules doesn't mean I want to use one of their MIDI controllers |
| 16:16.03 | [TK]D-Fender | Roland* |
| 16:16.23 | *** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
| 16:16.23 | coppice | why not? they make really nice MIDI kit |
| 16:16.39 | Erol_ | i just ment why digium hardware isnt certified for trixbox |
| 16:16.47 | Erol_ | if its certified for asterisk |
| 16:16.50 | [TK]D-Fender | Erol_: ..... |
| 16:16.53 | coppice | they make a really excellent little flash recorder too |
| 16:17.05 | [TK]D-Fender | Erol_: You just don't get it. Hardware & software are 2 completely different things here. |
| 16:17.34 | Erol_ | yeah but doesnt asterisk part of trixbox use the hardware? |
| 16:17.53 | [TK]D-Fender | coppice: I'm seriously considering buying a Novation X-Station 61 virtual analog synth/MIDI controller to use along with my Roland XV-3080 ROMpler. |
| 16:18.20 | [TK]D-Fender | Erol_: Asterisk doesn't give a crap what hardware you use with it so long as their is a driver interface for it. |
| 16:19.15 | Erol_ | uhm but then why would a certified hardware be important? |
| 16:19.33 | Erol_ | since everycard has its driver. |
| 16:19.41 | *** join/#asterisk Errotan (n=Errotan@5403E6EB.catv.pool.telekom.hu) |
| 16:19.47 | [TK]D-Fender | Erol_: Obviously there are DIFFERENCES and COMPLICATIONS. |
| 16:19.56 | coppice | and every card has its marketing dept. too |
| 16:21.32 | Erol_ | ok |
| 16:21.49 | Erol_ | i have an interesting question. |
| 16:22.07 | Erol_ | when a company switches from analog to voip.. |
| 16:22.23 | Erol_ | the employees can use their work phones even at home. |
| 16:22.59 | Erol_ | doesnt this get abused like using the work phone for making personal calls from home? |
| 16:23.04 | [TK]D-Fender | Erol_: "switch to voip" doesn't imply anything specific. its a question of what PARTS of the solution involve VoIP. |
| 16:23.21 | [TK]D-Fender | Erol_: And abused from home? How would YOU limit them? |
| 16:23.31 | Erol_ | [TK]D-Fender : |
| 16:23.36 | Erol_ | [TK]D-Fender : for example.. |
| 16:23.36 | Erol_ | [TK]D-Fender : for example. |
| 16:23.49 | Erol_ | [TK]D-Fender : i dont use my own house phone |
| 16:24.04 | Erol_ | [TK]D-Fender : and make even my personal calls from my work phone |
| 16:24.33 | [TK]D-Fender | Erol_: Yes, now please THINK for a little and see how you can conceive of ways to limit this. |
| 16:24.33 | Errotan | hi guys |
| 16:24.40 | Errotan | Anybody can tell how to make dial() application to send sip header like this:To: 0036701234567 <sip:0036701234567@sip.provider.eu>Tried with dial(SIP/provider/0036701234567) and dial(SIP/0036701234567@provider) but these give me this:To: <sip:0036701234567@sip.provider.eu> |
| 16:25.17 | *** join/#asterisk ramindia (n=Administ@202.63.96.10) |
| 16:25.27 | Erol_ | [TK]D-Fender: you need to follow the employees outgoing calls and fire them if they do personal calls? |
| 16:26.04 | [TK]D-Fender | Erol_: Keep exercising that brain of yours, I'm sure its in there somewhere.... |
| 16:26.21 | ramindia | how can i use asterisk to use a calling Platform to send call out ? any one can help me in this regard |
| 16:26.26 | Erol_ | [TK]D-Fender: putting a time limit? |
| 16:26.39 | smps | Erol_, putting connection limit |
| 16:26.49 | smps | Erol_, you limit connection only to phones from inside of company |
| 16:26.50 | [TK]D-Fender | ramindia: go lookup "call files" and "AMI Originate" on the WIKI, and in the BOOK |
| 16:26.52 | [TK]D-Fender | ~wikis |
| 16:26.53 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 16:26.54 | [TK]D-Fender | ~book |
| 16:26.55 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:27.11 | Erol_ | smps: but this also drops one of its benefits |
| 16:27.28 | ramindia | any one here some help on the subject |
| 16:27.37 | [TK]D-Fender | Erol_: tracking on repeat dialed numbers, CDR record analysis, TIME OF DAY CALLED (not allowed outside business hours) |
| 16:27.47 | killfill | anyone of you guys has a analog line working ok without "disconnect supervision" i.e. polarity switching |
| 16:27.54 | killfill | ? |
| 16:28.28 | ramindia | I want to use calling card platform instead of User and and pasword to send call out.. how can i do this ? |
| 16:28.31 | [TK]D-Fender | killfill: "callprogress=yes", but thats somewhat synonymous with "disconnectmycallsatrandom=yes" |
| 16:28.32 | Erol_ | [TK]D-Fender: yeah, time of day called logical |
| 16:28.52 | [TK]D-Fender | ramindia: "core show application dial" <- D() |
| 16:29.01 | killfill | heh.. |
| 16:29.04 | gr0mit | killfill, most telcos do this as standard |
| 16:30.02 | killfill | [TK]D-Fender: yup, tests it but doest seem to solve any problem.. :S |
| 16:30.02 | killfill | gr0mit: i guess im not lucky then.. :) |
| 16:30.02 | MT`AwAy | http://pastebin.com/m750feb7f <- can anyone spot a reason why I wouldn't get any video? Called extension (3369) is only Answer, then Echo. On Asterisk 1.4 it was ok, on 1.6 it's not working anymroe |
| 16:30.02 | ramindia | [TK]D-Fender: can u point me some URL Documentation ? |
| 16:30.02 | Erol_ | i need a pri card and found this A101 Single Port T1/E1/J1 Card |
| 16:30.02 | Erol_ | does this card support pri? |
| 16:30.12 | gr0mit | killfill, which telco is this? |
| 16:30.25 | killfill | gr0mit: im from sudamerica |
| 16:30.34 | [TK]D-Fender | Erol_: Yes, but I highly recommend getting one with HWEC |
| 16:30.38 | killfill | only public phones seems to have that in here.. |
| 16:30.46 | killfill | (for the coin stuff) |
| 16:30.51 | gr0mit | have you asked yoiur telco? |
| 16:30.54 | [TK]D-Fender | ramindia: I JUST DID |
| 16:31.01 | killfill | gr0mit: yup. |
| 16:31.27 | ramindia | [TK]D-Fender: i did not see, the mesasge ? |
| 16:31.34 | [TK]D-Fender | MT`AwAy: Try a "playback(silence/2)" right after your answer |
| 16:31.43 | [TK]D-Fender | ramindia: Scroll up and pay attention |
| 16:32.20 | Erol_ | [TK]D-Fender: what is HWEC? |
| 16:32.29 | gr0mit | killfill, well you might have prbs then |
| 16:32.30 | ramindia | [TK]D-Fender: can u do it for me once again ? |
| 16:32.32 | MT`AwAy | [TK]D-Fender> ok |
| 16:33.02 | killfill | gr0mit, yup. im trying to see how does only live withouth polarity switching in asterisk... |
| 16:33.03 | [TK]D-Fender | ramindia: Scroll up and read INFOBOT's links |
| 16:33.23 | killfill | currently there is ony of thouse linksys things, wich does detect hangsup well |
| 16:33.25 | [TK]D-Fender | Erol_: HardWare Echo Cancellation |
| 16:33.42 | gr0mit | if you can convert you line to BRI it woul save awhole lot of trouble |
| 16:33.43 | MT`AwAy | [TK]D-Fender> not much better, still no video |
| 16:33.50 | ramindia | [TK]D-Fender: seconds edition ? |
| 16:33.56 | gr0mit | but i have only used E1 in LatAm |
| 16:33.56 | killfill | gr0mit: yup, unfortunatly i cannot.. |
| 16:34.04 | [TK]D-Fender | ramindia: yes |
| 16:34.08 | Erol_ | [TK]D-Fender: it says that this card has HWEC |
| 16:34.10 | killfill | yah, there are only E1 here |
| 16:34.45 | [TK]D-Fender | Erol_: A101 = NO EC, A101d = HWEC |
| 16:35.27 | ramindia | [TK]D-Fender: hope u understand my question, iam not looking to create a calling card plat form, i want to use Asterisk, when dial out to Termination provider, it need to dial number.. then pin and call the number and connect the same to Asterisk user |
| 16:36.27 | [TK]D-Fender | ramindia: "core show application dial" <- D() <- read the damn instructions |
| 16:37.35 | gr0mit | killfill, change telco? or is there only one in yiour area? |
| 16:38.05 | killfill | gr0mit, noone of the telco's does provide this thing... |
| 16:38.23 | gr0mit | which telco is this? which country? |
| 16:38.40 | killfill | i already asks two.. ill ask on monday the other. but i doublt it |
| 16:38.45 | killfill | Chile |
| 16:38.55 | gr0mit | ok,santiago? |
| 16:39.01 | killfill | exactly |
| 16:39.44 | Errotan | anybody got some working template for a voip provider who use openser ? i can't make it work with asterisk; x-lite works but asterisk only allows inbound calls. i always got 603 declined SIP headers when i want to dial out |
| 16:40.09 | [TK]D-Fender | Off for a bit, BBL |
| 16:41.14 | MT`AwAy | [TK]D-Fender> got any idea where I could start looking at to find out why asterisk 1.6.1.1 doesn't seem to support video while it was working fine on 1.4 ? |
| 16:41.15 | killfill | hm.. there is a opermode in wctdm kernerl driver... seen to know only about newzealand amd autralia tho |
| 16:42.11 | smps | Errotan, do sip debug and look whats exactly happening |
| 16:42.31 | gr0mit | killfill, can you note get a voip account and port your number over? |
| 16:42.34 | Errotan | i done it for while (1 hours) |
| 16:42.51 | smps | Errotan, you dont need that much, only when you are making outgoing call |
| 16:43.10 | killfill | gr0mit, oh you mean getting a DID via not the local company, so over a vopi provider? |
| 16:43.19 | gr0mit | yup. |
| 16:43.20 | smps | Errotan, what does sip show registry say ? |
| 16:43.37 | gr0mit | or get your number ported from your local telco to a lcal voip provider |
| 16:43.37 | barbacha | any suggestion for a sip *video* client under linux ? |
| 16:43.42 | Errotan | the only difference between asterisk an x-lite is that x-lite sends To: 0036701234567 <sip:0036701234567@sip.provider.eu> and asterisk sends To: <sip:0036701234567@sip.provider.eu> |
| 16:44.01 | smps | Errotan, its not that much difference |
| 16:44.06 | killfill | no, cannot. its a for a client that already has a number, and people knows that number. In here, 'moving numbers' is not a posibility |
| 16:44.17 | Errotan | but thats why it won't work :( |
| 16:44.20 | gr0mit | thinks |
| 16:44.32 | killfill | telco's own numbers, and done release them.. :S |
| 16:44.33 | smps | Errotan, i doubt |
| 16:44.37 | killfill | dont |
| 16:44.57 | killfill | i think i cannot workaround the problem.. only face it.. :P |
| 16:45.18 | smps | Errotan, can you paste your trunk config and register string on pasterbin.com (without passwords) |
| 16:45.26 | gr0mit | how about purchasing call diversion, and call forward the number to avoip number? |
| 16:45.30 | Errotan | sure one moment |
| 16:46.47 | killfill | gr0mit, yah, that could work. but unfortunatly, i cannot think the internet connection will be stable. it could be down, and if so, i cannot just not revieve any calls... |
| 16:47.07 | killfill | (and it does get down...) |
| 16:47.23 | gr0mit | cant believe the internet ever fails ;-) |
| 16:47.32 | killfill | heh |
| 16:49.13 | gr0mit | killfill, this is one area which all telcos are equal! |
| 16:49.53 | gr0mit | and equally bad. |
| 16:50.36 | killfill | exactly.. :P |
| 16:51.01 | ramindia | [TK]D-Fender: i have refering the Book of 2nd edition i dont see one word in chapter5 .. other than that i dont see any document given what iam looking |
| 16:52.16 | Errotan | smps: http://asterisk.pastebin.com/m10436bec |
| 16:52.32 | Errotan | and thanks for helping |
| 16:55.08 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:56.27 | smps | Errotan, this is your username : 40123456789 ? |
| 16:57.08 | killfill | http://www.sofsis.cl/config/regional.html |
| 16:57.22 | killfill | gr0mit: if you go to tab "PSTN Line" |
| 16:57.38 | killfill | PSTN Disconnect Detection |
| 16:58.01 | killfill | i think thats what makes the device detect the hangsup/disconnections. |
| 16:58.15 | killfill | if i could only make something similar with asterisk.. |
| 16:59.08 | Errotan | smps: no it is just a sample |
| 16:59.31 | Errotan | smps: it is secret ;) |
| 17:02.29 | smps | Errotan, try this out: http://asterisk.pastebin.com/d4ea05fbe |
| 17:02.36 | gr0mit | well you can |
| 17:02.54 | gr0mit | there are 2 methods for signalling calling party clear. |
| 17:03.13 | gr0mit | 1- K-break - a short line current interruption |
| 17:03.20 | gr0mit | 2- polarity reversal |
| 17:03.38 | gr0mit | the linksys can detect either or both |
| 17:04.02 | gr0mit | if neither is available on your line, you have to detect silence or a tone which is applied |
| 17:04.17 | gr0mit | but it often leads to random droped calls. |
| 17:04.21 | killfill | oh well polarity reversal i see the lines doesnt have, becouse of zaptel logs and well have test it with that volimeter |
| 17:04.53 | gr0mit | best to check with a voltmeter |
| 17:05.25 | gr0mit | K-break is also known as CPC |
| 17:05.47 | gr0mit | it is set to 200ms in your linksys box |
| 17:05.52 | killfill | Ah cpc.. |
| 17:06.07 | gr0mit | CPC = Calling Party Clear |
| 17:06.18 | killfill | what about "Detect Disconnect Tone: yes"? |
| 17:06.39 | gr0mit | if your telco puts a tone on the line you can detect this |
| 17:07.12 | killfill | ok, how do i translate CPC and detect disconn tone in asterisk terminology? is that callprogress=yes? |
| 17:07.23 | gr0mit | 1 sec let me look |
| 17:07.58 | killfill | all i can see in zapte logs is " RING on 1/1! ", and " NO RING on 1/1!" |
| 17:08.29 | gr0mit | http://www.sinet.bt.com/351v4p5.pdf is a description of how it is done here. See 7.2 - Network innitiated clearing |
| 17:08.32 | ramindia | hi any help "iam not looking to create a calling card plat form, i want to use Asterisk, when dial out to Termination provider, it need to dial number.. then pin and call the number and connect the same to Asterisk user" |
| 17:10.15 | Errotan | smps: it don't work but i found more information in sip headers when x-lite calls the Via: SIP/2.0/UDP 10.0.2.15:5060 but with asterisk Via: SIP/2.0/UDP 192.168.1.50:5060 something is wrong that is my lan address |
| 17:10.42 | Errotan | Contact: <sip:40215691820@192.168.1.50> |
| 17:10.46 | smps | Errotan, and what is 10.0.2.15 ? |
| 17:11.13 | Errotan | i think the providers server ip behind openser proxy |
| 17:11.37 | killfill | hm.. sound like i cannot see with a voltimeter is the line is K-break or not.. |
| 17:11.39 | Errotan | Server: Sip EXpress router (0.9.6 (i386/freebsd)) |
| 17:11.41 | smps | Errotan, oh wait , you are in local network ? i mean asterisk ? |
| 17:11.53 | Errotan | yes it is behind a router |
| 17:13.38 | Errotan | and in x-lite i have to set proxy sip.geotel.eu |
| 17:13.40 | smps | Errotan, ok add this in your sip.conf : externip=your_public_ip_here |
| 17:13.40 | smps | localnet=your_local_net_here/netmask |
| 17:14.01 | Errotan | public ip is dynamic :( |
| 17:14.07 | smps | Errotan, and add outboundproxy= to you "toprovider" section |
| 17:14.31 | smps | Errotan, do it now for test only , later you could set some dnydns and use it |
| 17:18.00 | killfill | heh.. how motivating for me: "Hello, I'm getting interested to purchase an item that may solve my analog FXO/PSTN interfacing problems, as my national telco doesn't give CPC or Battery Reversal (and is not likely to, ever)." |
| 17:18.00 | Errotan | smps: it don't work, but thanks for all the help! i will wait for the support response for my problem |
| 17:18.03 | killfill | :P |
| 17:18.08 | gr0mit | http://www.sandman.com/cpcbull.html killfill |
| 17:18.12 | smps | Errotan, ok |
| 17:19.59 | gr0mit | get out your multimeter or oscilloscope |
| 17:20.19 | gr0mit | and see what is _actually_ happening on the line |
| 17:21.19 | gr0mit | it may be that the sals dept just has not got a clue what you are talking about |
| 17:21.25 | gr0mit | sales dept |
| 17:22.27 | killfill | yes i think so. ITs sad to have the same impression on the tecnical/support guys of the telco. |
| 17:22.44 | killfill | i also talk with a "ingenieer". he didnt know about this either. |
| 17:23.23 | gr0mit | hands killfill a Fluke multimeter and wishes him well |
| 17:23.31 | killfill | heh.. |
| 17:24.46 | killfill | Ok.. but is there something like "DetectDisconnect Tone" somewher ein asterisk or zaptel configuration? |
| 17:24.58 | killfill | callprogress=yes has that logic inside? |
| 17:26.52 | gr0mit | yes but only works on US tones |
| 17:27.28 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:27.36 | killfill | well i specify "progzone=cl" |
| 17:27.53 | killfill | there is a [cl] in indications.conf |
| 17:28.23 | gr0mit | not sure this is usec for tone detection, only generation |
| 17:33.11 | kaldemar | killfill: have you tried ks signaling and busydetect=yes? it has worked in some occasions. |
| 17:38.22 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 17:38.23 | killfill | kaldemar: yup. doesnt work |
| 17:39.30 | blaxthos | anyone recommend a cheap/easy/*-friendly IP/POTS term adapter ? |
| 17:39.32 | Erol_ | what should I need to be carefull about buying switches for Voip phoneS? |
| 17:39.42 | blaxthos | linksys spa-3102 is more $ than I wanted to spend |
| 17:41.34 | rob0 | Those are among the cheapest choices for FXO. |
| 17:41.44 | rob0 | ~cheap |
| 17:43.13 | infobot | somebody said cheap was 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. |
| 17:47.49 | jaytee | well, there's always the "two modems and a soldering iron approach" :-) |
| 17:47.49 | *** join/#asterisk SlipperyChicken (n=andrew@CPE0013f7c51659-CM0013f7c51655.cpe.net.cable.rogers.com) |
| 17:47.50 | rob0 | Wow, that's some lag. I thought I got the factoid wrong! |
| 17:47.50 | rob0 | ~botsnack |
| 17:47.50 | infobot | :), rob0 |
| 17:49.34 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 17:50.08 | gr0mit | in my experience, forget any analogue PSTN interface. |
| 17:50.36 | gr0mit | it is a sure way of putting you off |
| 17:52.14 | rob0 | I don't have need for FXO anyway. Soon I won't even have POTS service at all. |
| 17:52.53 | gr0mit | ok so get some ceap ip phones off ebay |
| 17:52.57 | gr0mit | cheap |
| 17:53.15 | rob0 | um, I wasn't the one asking |
| 17:54.06 | gr0mit | oops |
| 17:55.59 | gr0mit | Well, to be honest there is still almost never needs to use an analog telephone line feeding an asterisk system these days |
| 17:56.29 | gr0mit | either use a BRI or Port your analog number over to a VoIP provider |
| 17:57.54 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:58.09 | gr0mit | expects a barrage of criticism ... |
| 18:01.01 | killfill | how do i enable the DBUSYDETECT_COMPARE_TONE_AND_SILENCE flag theese days? |
| 18:02.13 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 18:14.10 | *** part/#asterisk ramindia (n=Administ@202.63.96.10) |
| 18:22.05 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 18:22.09 | Kernel_Core | hi all |
| 18:23.39 | Kernel_Core | I have 2 E1 links connected to my Box , I have two TE110P card , both of them work perfectly after 3-4 hours , first E1 fails , and when I issue pri show span X , I get "Window Lengh = 7/7 " for the Faulty card ! |
| 18:23.44 | Kernel_Core | what can be wrong ?! |
| 18:24.12 | Kernel_Core | when I shutdown the asterisk and issue loop on faulty E1 , after that it starts working .... |
| 18:25.06 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:28.11 | Errotan | smps: i found the answer !!! |
| 18:28.24 | smps | Errotan, what is it ? |
| 18:28.57 | Errotan | the voip provider don't want their users to use asterisk ! |
| 18:29.32 | Errotan | putting useragent=X-Lite in sip.conf [general] is the workaround :D |
| 18:30.04 | Errotan | found the answer here: http://lists.digium.com/pipermail/asterisk-users/2005-November/128467.html |
| 18:30.20 | Errotan | lol i hate them :) |
| 18:32.35 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-169-134.telkomadsl.co.za) |
| 18:32.52 | *** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 18:33.16 | smps | hehe crazy |
| 18:33.38 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:39.08 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.106.206) |
| 18:39.14 | DelphiWorld | hello asterisk users |
| 18:39.25 | DelphiWorld | please any SIP provider that use TCP a a transport? |
| 18:39.27 | Errotan | hello DelphiWorld |
| 18:39.33 | DelphiWorld | my ISP a blocked the SIP over UDP |
| 18:39.52 | DelphiWorld | i want to use this Provider a a gateway to other Provider |
| 18:40.03 | DelphiWorld | my friend: please try to help me |
| 18:40.55 | Errotan | openvpn to another server ? |
| 18:41.10 | DelphiWorld | Errotan: thanks, but each server? |
| 18:41.44 | Errotan | DelphiWorld: well if you have more than one it is a bit complicated |
| 18:42.48 | Errotan | DelphiWorld: but why they blocked SIP over UDP ? |
| 18:43.15 | DelphiWorld | Errotan: realy |
| 18:43.37 | DelphiWorld | Errotan: wel if you want to i give you the wireshark trace i give it to you |
| 18:44.09 | DelphiWorld | Errotan: i tryed Sip over TCP with iptel.org and is working |
| 18:44.22 | DelphiWorld | Errotan: but UDP with any provider is not working |
| 18:44.23 | Errotan | DelphiWorld: and the packets are droped or rejected ? |
| 18:44.39 | DelphiWorld | Errotan: i'm no sur about that |
| 18:44.49 | DelphiWorld | Errotan: bicose i'm blind and i'm unable to view this action |
| 18:46.17 | DelphiWorld | Errotan: also Iptel.org is stoped now bicose of server error |
| 18:47.13 | smps | DelphiWorld, do you have any firewalls ? |
| 18:47.41 | DelphiWorld | smps: no, i have only a small ADSL router: ZXDSL831EE |
| 18:48.01 | DelphiWorld | smps: is tested in other internet connection with hte sam result |
| 18:49.00 | Errotan | DelphiWorld: any other UDP stuff works ? ( like teamspeak,skype,ec.) |
| 18:50.28 | DelphiWorld | Errotan: yes, including Gizmo5! |
| 18:51.54 | Errotan | have you talked to your ISP support maybe it is a default settings and can be switched off ( like when the isp blocks tcp port 25 beacuse of pam ) |
| 18:52.06 | Errotan | spam* |
| 18:52.44 | DelphiWorld | Errotan: yes, but is blocking sip a 100% no 25 or ... |
| 18:55.21 | Errotan | DelphiWorld: i think you find more voip providers with iax2 than sip over tcp ( but i'm not an expert ) |
| 18:56.27 | DelphiWorld | Errotan: ok, but i want to connect it to a Actual SIP acount |
| 18:57.16 | Errotan | DelphiWorld: i see; hmmm thats a big problem |
| 18:58.40 | DelphiWorld | i want a Service provider that have a SIP Proxy capability using OpenSer or Kamailio |
| 19:00.25 | *** join/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net) |
| 19:06.08 | DelphiWorld | i used Voipuser.org now and is working! |
| 19:06.15 | DelphiWorld | but over TCP no UDP |
| 19:08.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:16.08 | *** join/#asterisk salvia (i=aurax@bzq-84-108-166-31.cablep.bezeqint.net) |
| 19:16.58 | salvia | Hello all |
| 19:17.17 | salvia | I'm trying to configure asterisk with drbd/heartbeat. anyone here is experienced with such configuration ? |
| 19:20.07 | *** join/#asterisk Unixdawg (n=UnixDawg@pool-96-235-13-58.pitbpa.east.verizon.net) |
| 19:20.13 | Unixdawg | hey guys getting a issue |
| 19:20.53 | Unixdawg | [Jun 7 15:18:28] WARNING[8823]: app_dial.c:1468 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 19:20.53 | Unixdawg | <PROTECTED> |
| 19:21.47 | Unixdawg | whats this cause |
| 19:25.29 | smps | Unixdawg, it could be that you have call limit on your sip trunk |
| 19:25.41 | smps | Unixdawg, for example 2 channels max at same time |
| 19:26.24 | [TK]D-Fender | Unixdawg: You should be looking at SIP DEBUG |
| 19:26.50 | Unixdawg | this is a fresh setup and no limits set |
| 19:27.06 | Unixdawg | I am on the sip debug but tis not helping with the uissue |
| 19:28.10 | [TK]D-Fender | Unixdawg: Show us the complete call attempt |
| 19:28.37 | Unixdawg | hold on have to have him try again |
| 19:29.41 | *** join/#asterisk bbryant1 (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 19:30.07 | DelphiWorld | [TK]D-Fender: hello |
| 19:30.23 | DelphiWorld | [TK]D-Fender: please cool you help me by giving me any SIP provider that support TCP? |
| 19:33.20 | gr0mit | DelphiWorld, jooi, why do u need tcp? |
| 19:33.40 | Unixdawg | http://pastebin.ca/1451061 |
| 19:35.07 | smps | Unixdawg, this is not sip debug |
| 19:35.29 | smps | Unixdawg, do this in asterisk cli: sip set debug |
| 19:35.55 | [TK]D-Fender | Unixdawg: Go dump each peer at CLI |
| 19:36.30 | DelphiWorld | gr0mit: i'm unable to use Sip over UDP in my ISP |
| 19:36.58 | gr0mit | eew - where are you? |
| 19:37.46 | DelphiWorld | gr0mit: algeria |
| 19:38.05 | gr0mit | aah ok. sounds like time for a vpn then! |
| 19:38.52 | gr0mit | They block UDP? |
| 19:39.06 | DelphiWorld | gr0mit: yes |
| 19:39.17 | DelphiWorld | gr0mit: i tryed Voipuser and is working only with TCP |
| 19:39.17 | gr0mit | What ports for UDP are open? |
| 19:39.29 | DelphiWorld | gr0mit: i'm not sur |
| 19:39.36 | DelphiWorld | gr0mit: gizmo5 is working |
| 19:40.45 | gr0mit | So if these are working, what is the problem? |
| 19:41.14 | DelphiWorld | <gr0mit: sip not working |
| 19:41.37 | gr0mit | Are you sure it is a problem with UDP ports? |
| 19:42.18 | gr0mit | I have never known an ISP block UDP |
| 19:42.35 | gr0mit | but then, I'm not in Algeria! |
| 19:42.43 | DelphiWorld | gr0mit: i'm very sur |
| 19:42.50 | DelphiWorld | gr0mit: cool you drop me a SIP call? |
| 19:43.04 | gr0mit | I can do, yes |
| 19:43.21 | gr0mit | PM me your sip address |
| 19:43.38 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 19:43.45 | DelphiWorld | gr0mit: OK |
| 19:45.44 | *** join/#asterisk ingenius (n=alektro@host176.190-230-72.telecom.net.ar) |
| 19:53.34 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:56.28 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.106.206) |
| 19:56.31 | *** part/#asterisk rene- (n=renemend@200.34.66.137) |
| 20:25.26 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 20:35.17 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 20:36.21 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 20:43.43 | HaMF | hi, I'm trying to figure out which version of asterisk we should use and what server requirements we'll have. We'll definitely need something to talk to analog telephones (via digium wildcards) and ARA (+odbc->postresql) to define the sip friends and to manage the dialplan. Furthermore it should be easy to upgrade asterisk and its modules without breaking anything. |
| 20:48.36 | HaMF | (I'm talking about 170-Sip-Phones that will connect to asterisk and asterisk relays via PRI to the PSTN. There'll be about 10 to 20 simultaneous calls (including voicemail).) |
| 20:51.59 | HaMF | It would really be cool if anyone had some experience with this kind of setup and could tell me about the problems he encountered and about things not-to-forget. |
| 21:27.00 | carrar | why odbc? |
| 21:27.15 | carrar | asterisk supports psotgresql |
| 21:28.46 | *** part/#asterisk Errotan (n=Errotan@5403E6EB.catv.pool.telekom.hu) |
| 21:32.10 | rob0 | I read here that someone authoritative (can't recall who) thinks that the ODBC support in * might be better and more robust. |
| 21:35.17 | carrar | I heard you can get viruses from handling money |
| 21:35.49 | RyanRR | i heard you're dumb |
| 21:36.18 | carrar | Thats why I use postgresql without a additional driver needlessly in the way |
| 21:37.07 | carrar | You're SMRT |
| 21:38.39 | *** join/#asterisk troy- (n=troy@worldnet.tauri.ca) |
| 21:44.34 | rob0 | So anyway, there might be good reasons, I don't know, and now I don't even care. I lurk here to try to learn some things. |
| 21:57.52 | *** join/#asterisk JoEMoMMa (n=JoE@24-171-58-216.dhcp.stls.mo.charter.com) |
| 21:59.16 | JoEMoMMa | i am new to the asterisk world and was wondering is it possiable to purchase a DID number and pipe it right into a asterisk server box over my current T1 to be used to create a Fax server? |
| 22:01.06 | Juggie | when bridgeing two sip channels any reason i would get no rtp audio working if i dont do an Answer() before the dial? |
| 22:06.04 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 22:06.31 | carrar | JoEMoMMa, yes it is possible |
| 22:06.50 | carrar | data T1 or voice T1? |
| 22:08.21 | JoEMoMMa | data T1 |
| 22:10.34 | JoEMoMMa | Carrar....I am trying to figure out if it is better to go this route or just use a dail-up modem hooked into the serial port of a standard centos 5 box and a program simalar to hylafax |
| 22:12.20 | carrar | using t.38? |
| 22:12.25 | carrar | or just g711 |
| 22:12.42 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 22:13.10 | carrar | g711 may work, but you'll find out long faxes won't since timing is critical |
| 22:13.27 | carrar | depending on your path and QoS |
| 22:14.27 | *** join/#asterisk haps (n=hapoteh@75-119-232-149.dsl.teksavvy.com) |
| 22:14.30 | haps | word |
| 22:14.41 | haps | i'm trying to set up an asterisk install on a friend's vmware machine |
| 22:14.55 | haps | all static local ips i'm trying to connect to the echo test program |
| 22:15.12 | haps | it registers fine but with core set debug 10 I don't get any messages about extensions |
| 22:15.24 | haps | so, when i dial 600 (the echo test extension) it just times out. |
| 22:15.25 | haps | any ideas? |
| 22:16.38 | haps | sip show peers tells me I'm registered, I have qualify set to 30000 so it shows the status as ok |
| 22:18.22 | JoEMoMMa | Carrar....the modem i am looking at does state anything about g711 or t.38 |
| 22:20.26 | haps | oh, btw, i'm using express talk softphone. (this is XP to a linux VM) |
| 22:25.32 | *** part/#asterisk JoEMoMMa (n=JoE@24-171-58-216.dhcp.stls.mo.charter.com) |
| 22:51.00 | *** join/#asterisk troubled_ (n=troubled@unaffiliated/troubled) |
| 22:53.07 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:53.21 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 23:08.48 | haps | looks like it was softphone troubles |
| 23:08.49 | haps | hasta |
| 23:08.51 | *** part/#asterisk haps (n=hapoteh@75-119-232-149.dsl.teksavvy.com) |
| 23:11.48 | *** join/#asterisk Whitor (n=Whitor@cpe-74-76-185-31.nycap.res.rr.com) |
| 23:14.52 | *** join/#asterisk jorts (n=jortbloe@210.48.105.162) |
| 23:15.48 | jorts | Can anyone tell me how to tell Asterisk to log the queue_log somewhere else, without recompiling? |
| 23:16.04 | jorts | please? |
| 23:19.41 | jorts | is anyone actually here? |
| 23:20.21 | layne | yes, you may have to wait a bit for a response/help |
| 23:20.44 | jorts | nods |
| 23:32.36 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:36.06 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 23:37.16 | rue_mohr | does anyone know how hard it would be to get asterisk to accept ulaw from a serial port? |
| 23:37.32 | rue_mohr | hmm have to send too |
| 23:54.49 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |