| 00:01.05 | profounded | >modprobe ndiswrapper |
| 00:01.14 | profounded | says it cant find ndiswrapper |
| 00:01.25 | profounded | any ideas? |
| 00:01.42 | profounded | i think i dont have paths setup if that helps |
| 00:01.46 | profounded | errr wrong chat |
| 00:03.19 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 00:03.58 | Freman | G'day folks. I have a little trouble with my asterisk. "hanguponpolarityswitch=yes" is causing outgoing calls to be hung up as soon as the remote party answers |
| 00:06.08 | [hC] | so fix your polarity |
| 00:06.54 | Freman | I need "hanguponpolarityswitch=yes" to detect the end of an incomming call tho (the telco switches polarity to end the call) |
| 00:08.09 | [hC] | well, okay, however if you start out with inversed polarity, youve got a problem |
| 00:08.56 | JT | this is true |
| 00:10.21 | JT | Freman: make sure you got the tip and ring connections the right way around |
| 00:10.21 | Freman | where do I have to go to look at this? |
| 00:10.21 | [hC] | at your demarc |
| 00:10.22 | [hC] | where the lines go in to your pbx |
| 00:10.22 | nailbags|work | Freman: iirc, you have to ask telstra to enable polarityswitchonhangup on their lines if you want to use it in asterisk |
| 00:11.16 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.99) |
| 00:11.16 | Freman | nailbags: we have ROIC enabled nailbags, it's been working fine for incomming calls - I've just got to work out how to stop it from hanging up on incomming calls |
| 00:11.16 | JT | true nailbags|work, but there's a problem if calls are being hung up when answered |
| 00:11.16 | Freman | We ring out, the phone at the other end rings, but as soon as they pick up... it hangs up |
| 00:11.16 | JT | Freman: stop it from hanging up on outgoing calls you mean? |
| 00:11.16 | nailbags|work | k, just checking .... |
| 00:11.16 | *** join/#asterisk Skarmeth (n=Skarmeth@201009089207.user.veloxzone.com.br) |
| 00:26.10 | Freman | yes, all outboud calls are hanging up - inbound ones are working perfectly |
| 00:26.10 | Freman | *outbound |
| 00:26.10 | nailbags|work | Freman: what advantage is ROIC? |
| 00:26.11 | nailbags|work | do you have trouble with hangup detection? |
| 00:26.11 | Freman | It means that it hangs up as soon as the remote party does |
| 00:26.11 | nailbags|work | Freman: its just than mine seems to hang up instantly w/out ROIC |
| 00:26.11 | JT | what does roic stand for? |
| 00:26.11 | nailbags|work | "Reverse On Idle Condition" |
| 00:26.11 | Freman | Reverse .. that |
| 00:26.11 | JT | is it free? |
| 00:26.11 | nailbags|work | it is on business lines apparently |
| 00:26.11 | Freman | can I split the setting and set it up so zap/1 has hanguponpolarityswitch=yes and zap/2 has hanguponpolarityswitch=no? |
| 00:26.11 | JT | how many lines do you have? |
| 00:26.11 | Freman | two |
| 00:26.11 | JT | isdn2 is plentiful and priced ok in .au |
| 00:26.11 | JT | ah ok |
| 00:26.12 | *** join/#asterisk RealUser5802370 (n=noname@S01060004e2f37f26.vf.shawcable.net) |
| 00:26.12 | ki2k | anyone know if ztdummy is laggy for meetme's? |
| 00:26.12 | RealUser5802370 | ki2k: I use it all the time, works fine. |
| 00:26.12 | Freman | It's a helpdesk setup - there should be next to no outgoing calls made (They're normally voip routed), and it's important that incomming calls are hug up as quickly as possible |
| 00:26.12 | ki2k | unanmed: you do a lot of conference calls? |
| 00:26.12 | ki2k | unanmed: do you happen to know how much lag you get? |
| 00:26.13 | JT | Freman: just curious why analogue lines are being used for business voice |
| 00:26.13 | unanmed | Anyone have any experience with rx/txfax? Im trying to find out why it faxes to some and not to others(I suspect the headers but I have set it and still nothing) |
| 00:26.13 | nailbags|work | Freman: asterisk seems to detect hangups pretty much instantly for me. i've never bothered with ROIC. how much lag do u get? |
| 00:26.13 | Freman | because it's what was here. |
| 00:26.13 | unanmed | ki2k: I wouldnt say a lot, its not high use, but i find no lag at all with 3 people in it. Im sorry but I dont know the exact lag |
| 00:26.13 | *** join/#asterisk roving_prole (n=Harper@c-71-199-16-110.hsd1.co.comcast.net) |
| 00:26.13 | Freman | There's 4 analogs on a rotary for main office function, 2 extras (one is a fax) and then the helpdesk lines |
| 00:26.13 | unanmed | ki2k: I use the dynamic meetme conferences if that makes any difference |
| 00:26.14 | JT | wouldn't have any of these polarity issues with isdn |
| 00:26.14 | JT | but you have to work with what you've got sometime i guess |
| 00:26.14 | Freman | It took me a year to convince them to get this far |
| 00:26.14 | nailbags|work | Freman: so you get a noticable lag with hangup detection? |
| 00:26.14 | *** join/#asterisk Nukemizer (n=Nuke@160.7.239.13) |
| 00:26.15 | Freman | 6 busies |
| 00:26.15 | nailbags|work | using busydetect=yes? |
| 00:26.15 | Freman | yes |
| 00:26.15 | nailbags|work | what hardware? |
| 00:26.15 | Freman | If they fixed the issue with our backbone, I wouldn't need to provide pstn fall over, I could just re-route the packets |
| 00:26.15 | unanmed | anyone ever use rx/txfax? Trying to get it working properly |
| 00:26.15 | JT | heh, "fall over" |
| 00:26.16 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:26.16 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:26.16 | mountainm2k | unanmed I recommend you give that up, and check out iaxmodem and Hylafax instead |
| 00:26.16 | unanmed | mountainm2k: I tried hylaxfax, not luck there either. At least with rx/tx i was able to receive and send fine, it just seems to not like "certain" faxmachines, 70% no problem, that 30% makes no sense |
| 00:26.16 | mountainm2k | Hmm, I had exactly the oppisite problem with tx/rxfax -- couldn't get it to work |
| 00:26.16 | mountainm2k | I occasionally get "failed" messages, but not very often |
| 00:26.16 | unanmed | mountainm2k: Also not looking for high volume faxing which from what ive read is more what hylafax/asterfax is good for? |
| 00:26.16 | unanmed | mountainm2k: Hows your success with hylafax? most go through no problem? |
| 00:26.22 | mountainm2k | We're not too high a volume -- few per day... Although I did set up DID-faxing for everybody |
| 00:26.31 | mountainm2k | I don't send with it, etiher, only receive |
| 00:27.23 | unanmed | mountainm2k: Yeah its what im looking for too, assign the 5 people in the office a did and have the tiff's/pdf's emailed to them. ive got that far, but the sending out is bugging me.. its just certain faxes too, some fax 100% of the time over and over |
| 00:28.24 | unanmed | mountainm2k: Do you have any issues with faxes that demand a header? i think that might be my problem but ive tried setting one without any change |
| 00:28.38 | mountainm2k | Hmm, dunno... It received OK from my fairly old (though not thermal paper) machine at home, and our fairly new (with high speed modem) machine here at the office |
| 00:29.01 | mountainm2k | Don't know on that one -- I've never used HylaFAX to send at all... |
| 00:30.26 | unanmed | yeah im afraid of the same thing with hyla, but will revist it. |
| 00:31.25 | mountainm2k | sorry -- lots of people when they say "Anybody have any experience getting rxfax to work?" can't get it to work at all |
| 00:31.37 | mountainm2k | so my stock answer is "I tried, give it up, and use iaxmodem with hylafax" |
| 00:31.51 | unanmed | fair enough :) |
| 00:37.31 | harryvv | I need to look into hylafax some time. |
| 00:38.22 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 00:38.25 | harryvv | BTW, has anyone here done some market resurch into useing wifi to link two buildings in a downtown enviroment for voip reasons? I just talked to proxim and thay have done this in the past. |
| 00:39.03 | harryvv | Thats a really good money maker idea. No need to up the bandwith and have qos issues. |
| 00:39.48 | JT | money maker in what sense? |
| 00:40.46 | AndyCap | harryvv: wifi@2.4ghz is pretty crowded in most areas, so qos issues galore |
| 00:41.24 | harryvv | Andy, yes thats possibly true. |
| 00:42.47 | JT | not to mention the fact that it's a free-for-all at 2.4GHz with no protection to interference afforded to anyone |
| 00:43.22 | harryvv | With very directional yagi antennas that may not be a issue. |
| 00:43.39 | JT | yeah well i wouldn't bed a business on it |
| 00:43.45 | JT | s/bed/bet/ |
| 00:43.53 | *** part/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 00:43.57 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:43.58 | IOscanner | I am looking to buy DID numbers. What is the best priced vendor to purchase from? |
| 00:44.37 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 00:44.42 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 00:45.02 | harryvv | proxim has already sold some high priced units for this application. |
| 00:45.45 | JT | harryvv: licensed frequencies? |
| 00:46.06 | harryvv | unlicenced freqs |
| 00:46.34 | harryvv | But I can use the licenced freqs if I wish as long as my FCC licence supports it. |
| 00:48.22 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 00:48.31 | *** join/#asterisk niZon (n=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 00:49.14 | [hC] | harryvv: I work for a company that does that, solely. |
| 00:49.24 | [hC] | (downtown metro wifi links) |
| 00:49.28 | [hC] | We use some proxim radios, too |
| 00:49.31 | [hC] | not all though |
| 00:50.35 | *** part/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 00:51.39 | harryvv | hc, how do you rate these radios? |
| 00:52.18 | [hC] | the proxim's? |
| 00:52.26 | [hC] | which model? and what distance are you shooting? |
| 00:53.19 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 00:53.29 | *** join/#asterisk daniel_bergamini (n=daniel_b@70-41-166-149.cust.wildblue.net) |
| 00:53.37 | daniel_bergamini | re all |
| 00:54.29 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 00:54.48 | daniel_bergamini | anyone used netzerovoice with asterisk? |
| 00:54.55 | [shodan] | if you want to link just 2 buildings just use the 10 terahertz band |
| 00:56.13 | JT | indeed |
| 01:02.13 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 01:02.53 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:03.42 | *** join/#asterisk Ahmuck (i=chatzill@p114n22.ruraltel.net) |
| 01:03.50 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 01:03.53 | *** join/#asterisk FlyboySR22 (n=Richard@searsair-linksys.adnc.com) |
| 01:03.57 | harryvv | what make of transmitter would that be |
| 01:03.59 | *** part/#asterisk FlyboySR22 (n=Richard@searsair-linksys.adnc.com) |
| 01:04.22 | Ahmuck | what does it take to hook up asterisk to an old merlin legend system. can we keep the same phones ? |
| 01:04.23 | harryvv | hc, no model i have no client that I know of yet to show |
| 01:04.44 | Damin | file: You have to be alive, right? |
| 01:05.04 | file | I am alive |
| 01:05.06 | Damin | Ahmuck: Yes, but that isn't a question for the developers channel, because it doesn't deal with code.. |
| 01:05.07 | file | I'm trying to fix trunk |
| 01:05.16 | Damin | Ahmuck: It's an #asterisk question.. |
| 01:05.19 | file | well, "make it better" |
| 01:05.24 | [hC] | harryvv: they work well though, genereally, yes. |
| 01:05.29 | file | Damin: you're in #asterisk |
| 01:05.54 | Damin | file: OK.. so your p2pbridging merges.. the comment "It's easier to start anew than to fix this".. whas that related to the idea? Starting fresh with it? Or rebuilding the entire core of Asterisk kind of new? |
| 01:06.01 | Damin | What the? |
| 01:06.09 | Damin | What a retard I am! :) |
| 01:06.10 | file | let's move over to dev |
| 01:06.15 | IOscanner | I am looking to buy DID numbers. What is the best priced vendor to purchase from? |
| 01:06.31 | Damin | Wow.. |
| 01:06.48 | file | mv Damin #asterisk-dev |
| 01:06.58 | *** part/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 01:07.11 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 01:10.13 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 01:14.24 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:21.30 | *** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net) |
| 01:22.58 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:26.16 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 01:27.08 | tengulre | hi,all |
| 01:27.31 | tengulre | how to using asterisk as a h323 gatekeeper? |
| 01:30.21 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:30.21 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:31.37 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 01:33.49 | kratzers | look at GNU Gatekeeper |
| 01:35.31 | Flauto | i am trying to install webvmail |
| 01:35.35 | Flauto | but it does not work |
| 01:35.47 | Flauto | is there anything i need to do to prepare the installation |
| 01:35.57 | kratzers | define does not work |
| 01:37.04 | kratzers | make sure the script is in your cgi-bin directory and that it is executable |
| 01:37.34 | Flauto | kratzers, i tried, and it is showing in red color |
| 01:37.39 | kratzers | also make sure permissions are correct on the spool directory and files |
| 01:37.51 | kratzers | and make sure to set the correct context in the script |
| 01:38.18 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:38.23 | kratzers | have you gotten past the 'make'? |
| 01:38.52 | kratzers | anybody have experience with Lucent/Ascent TNTs and SIP/H.323? |
| 01:39.23 | Flauto | would you tell me how to edit the spool directory? |
| 01:40.41 | intralanman | /join ##linux |
| 01:40.53 | intralanman | then ask them |
| 01:40.56 | kratzers | the users that httpd is running as needs to have read access to the voicemail text and audio files to list and listen to them, and write to delete them |
| 01:41.55 | kratzers | http://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi |
| 01:50.17 | tengulre | hi,everyone! |
| 01:50.31 | kratzers | negative on the TNT+VoIP? |
| 01:50.55 | tengulre | why not register with h323 gatekeeper in asterisk, client I using openhone? |
| 01:51.37 | tengulre | why not listener port 7000 in asterisk? is it a gatekeeper port? |
| 01:53.20 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 01:54.25 | *** join/#asterisk diablopico (n=diablopi@ip68-101-128-90.sd.sd.cox.net) |
| 01:54.50 | *** join/#asterisk doolph (n=doolph@200.124.28.155) |
| 01:55.19 | diablopico | hello, is there anyone here that can help with a delay problem on h323 |
| 01:55.41 | doolph | what h323 are you using |
| 01:56.06 | diablopico | its a long story ,, but h323 that comes with asterisk v1.0.6 |
| 01:56.20 | doolph | well you should upgrade to oh323 |
| 01:57.13 | diablopico | ok . is there any particular place i should look to get the upograde ? |
| 01:57.54 | doolph | google? |
| 01:58.26 | diablopico | ok .. let me explain ,, i am not an expert ,, but i think i have graduated from novice |
| 01:58.36 | diablopico | now for the question. |
| 01:58.48 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 01:59.08 | doolph | good |
| 01:59.13 | diablopico | i use openh323 and pwlib that is requested by the files in h323 directory of asterisk before i compile. |
| 01:59.28 | diablopico | this should work ,, and does |
| 01:59.51 | diablopico | but only if i wait 20 seconds after the phone connects ,, and then i can talk as usual |
| 02:00.03 | diablopico | what caused the delay |
| 02:00.15 | diablopico | ?????? |
| 02:00.16 | doolph | maybe your comp? |
| 02:00.21 | doolph | or your provider |
| 02:01.03 | diablopico | ok ,,,, i have the same computer it was working on, and the same provider. it did work without the delay before i recompiled openh323 |
| 02:01.41 | diablopico | any ideas ? |
| 02:02.01 | doolph | um |
| 02:02.10 | doolph | did you compile it right for your os? |
| 02:02.24 | diablopico | yes |
| 02:02.31 | diablopico | i get no errors |
| 02:02.45 | doolph | sometime you need to tune it |
| 02:03.34 | doolph | and did you compile with the latest version? |
| 02:03.42 | diablopico | no |
| 02:04.02 | doolph | why not |
| 02:04.06 | diablopico | i compiled with the versions asked for by the h323 files that came with asterisk |
| 02:04.26 | doolph | the h323 that came with asterisk eh |
| 02:04.40 | doolph | why did you do that |
| 02:04.42 | diablopico | in the h323 directory ,, the readme specifies the versions of openh323 and pwlib ot be used |
| 02:05.11 | doolph | well actually that h323 is pretty old i think |
| 02:05.18 | diablopico | i get errors if i use the newest versions of openh323 and pwlib |
| 02:05.30 | diablopico | yes ,, but it worked...... |
| 02:05.46 | doolph | well my oh323 is working perfectly |
| 02:05.52 | doolph | I followed the steps in this page |
| 02:05.53 | doolph | http://www.inaccessnetworks.com/projects/asterisk-oh323 |
| 02:06.12 | diablopico | ok ,, i will give it a try |
| 02:06.18 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 02:06.26 | diablopico | got nothing to lose now ,, i have a broken system anyway |
| 02:06.32 | jart | good morning! |
| 02:06.32 | doolph | yes |
| 02:06.37 | diablopico | thanks for you help doolf |
| 02:06.38 | doolph | good luck |
| 02:08.03 | diablopico | what version should i get doolf |
| 02:08.19 | doolph | let me check |
| 02:08.52 | doolph | http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.6.7.tar.gz |
| 02:10.17 | diablopico | thanks |
| 02:11.05 | doolph | then http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries |
| 02:11.14 | doolph | read all the readmes |
| 02:19.35 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:19.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:23.00 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:42.22 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 02:57.34 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 03:00.16 | *** join/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net) |
| 03:02.04 | lordbaron | ? |
| 03:02.50 | JT | what's the ? for, lordbaron? |
| 03:03.05 | lordbaron | learning how this ui works |
| 03:03.10 | lordbaron | didn't think that would 'post' |
| 03:03.35 | lordbaron | anyone here have experience with the TDM400P ? |
| 03:04.23 | JT | a few people probably do |
| 03:05.28 | TrixVox | Is something wrong with asterisk-biz? |
| 03:05.30 | lordbaron | I am having a problem with a configuration. I have two cards in a server, 1.2.10/1.2.7 - when configd for kewlstart, outbound dialing does not work. The lines hang up almost right away, and say lines congested |
| 03:05.41 | lordbaron | incoming works just fine |
| 03:05.54 | mog | not that i know of TrixVox |
| 03:06.29 | TrixVox | There were only 10 posts today? I know I tried posting and they never showed up in the archive... |
| 03:07.18 | lordbaron | loopstart works fine, but then does not detect incoming callers hangup in the IVR |
| 03:09.36 | mog | i havent seen any posts from you as of late |
| 03:10.17 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 03:11.32 | lordbaron | does KewlStart require any options to affect/control outbound dialing? |
| 03:15.30 | Corydon76-home | Nope |
| 03:16.57 | lowlevel | jt; remember my odd analog line problems with that funny error message the other night? (something about weird mode 3??) |
| 03:18.35 | lowlevel | lordbaron: I read that the boards dont work well when you put 2 on the same pci bus. |
| 03:18.48 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 03:19.45 | lordbaron | hadn't read that..but that does seem to explain the problem |
| 03:20.03 | lowlevel | lord: the board generates ALOT of irqs apparently , and its recommended to only have one per box. |
| 03:20.44 | nevyn | it's 8000 interupts/second |
| 03:20.45 | lordbaron | to handle more than 4 pots...iax bridge 2 machines? |
| 03:20.56 | lowlevel | lord: theres a sagoma (or something like that) board thats expandable with addons that would work well I think.. or you could use the 24port card |
| 03:21.03 | nevyn | lowlevel: get E1 card? |
| 03:21.10 | lowlevel | or that if that would work for ya |
| 03:21.11 | lowlevel | ;) |
| 03:21.28 | lowlevel | 8000/second eh.. geesh |
| 03:21.43 | lordbaron | I am only guessing, but I think the spare machine is cheaper than the 24port card |
| 03:22.00 | nevyn | actually... |
| 03:22.07 | lowlevel | lord: well, I dunno, if you were to buy 2 fully loaded 4 port cards, you could have got the 24port card with 2 4 port modules |
| 03:22.10 | lowlevel | I think... |
| 03:22.20 | lowlevel | depends wher eyou get it |
| 03:22.36 | lordbaron | hmm, thats a good point |
| 03:22.53 | lowlevel | I use mine at home and I can only ever have 2 phone lines in this apartment, so I'm find on the little card |
| 03:22.54 | lowlevel | ;) |
| 03:22.56 | file | are you talking about the TDM2400? |
| 03:22.56 | lordbaron | the interupts not a problem? |
| 03:23.21 | lowlevel | file: yeah, tdm2400 with 2 4port modules seems to be about the same price as 2 fully loaded 400's |
| 03:23.26 | nevyn | file: I'm probably wrong |
| 03:23.34 | lowlevel | maybe a slight bit more |
| 03:23.45 | tengulre | hi,all |
| 03:23.48 | lowlevel | lord: well, no, because youd'e only have one card |
| 03:23.48 | lowlevel | :) |
| 03:24.07 | file | the hardware design of the TDM2400 is better then the TDM400 as well |
| 03:24.07 | tengulre | why asterisk can not register to remote gatekeeper??? |
| 03:24.11 | file | less compatibility issues |
| 03:24.22 | lowlevel | file: thats good to know |
| 03:24.31 | tengulre | how to setting gatekeeper username and secret in /etc/asterisk/h323.conf?? |
| 03:25.02 | lordbaron | since I have 8 fx? (red) cards these will fit the tdm2400? |
| 03:25.07 | lowlevel | lord: no |
| 03:25.16 | lowlevel | lord; the tdm2400 board uses different modules |
| 03:25.24 | lordbaron | ahh..too bad |
| 03:25.25 | lowlevel | lord: they combine 4 fxo's or 4fxs's on one module |
| 03:25.38 | *** join/#asterisk justnulling2 (n=justnull@ool-182e41b0.dyn.optonline.net) |
| 03:25.44 | lowlevel | (and as such, cost over 3x as much) |
| 03:25.44 | lordbaron | that makes sense..looking at pics now |
| 03:25.47 | lowlevel | yup |
| 03:25.59 | file | it's insane that it can all fit on one module |
| 03:26.11 | file | pretty close together... the components |
| 03:26.17 | lowlevel | *shrug* |
| 03:26.25 | lowlevel | its a bulky card if youaskme |
| 03:26.32 | lowlevel | but I'm not complaining |
| 03:26.34 | file | well it's a full length PCI card |
| 03:26.40 | lowlevel | :) |
| 03:26.42 | file | silly goof |
| 03:26.43 | justnulling2 | anyone has voicestick configured? i keep on getting 423 "Interval Too Brief" from them |
| 03:26.44 | lowlevel | :D |
| 03:27.40 | file | and now I run to Subway |
| 03:28.02 | lowlevel | hmm, subway |
| 03:28.06 | lowlevel | pass |
| 03:28.11 | lowlevel | back in afew |
| 03:30.32 | kratzers | digium sell the cables that connect to the TDM2400P? |
| 03:31.22 | mog | we do now |
| 03:31.25 | mog | on the website |
| 03:31.40 | kratzers | I couldn't find them, have a link? |
| 03:39.47 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 03:39.49 | lordbaron | anyone have experience with the E7520 chipset and TE412p? |
| 03:40.09 | lordbaron | I am having a lot of trouble with call disconnects under heavy usage |
| 03:40.32 | lordbaron | Digium Support is saying that the chipset is to blame, but all the recommended servers have this same chipset |
| 03:41.28 | lordbaron | server is dell poweredge 2850 |
| 03:43.18 | lowlevel | man I'm craving that spicey burrito place all of a sudden... |
| 03:45.41 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:46.37 | kratzers | what chipset? |
| 03:47.07 | lordbaron | 7525 |
| 03:47.13 | lordbaron | Intel e7525 |
| 03:47.22 | kratzers | the compatibility list is really an incompatibility list |
| 03:47.33 | kratzers | read it more closely |
| 03:47.46 | *** part/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 03:47.47 | lowlevel | heh, isn't that always the case? ;) |
| 03:47.49 | lordbaron | right-but the ast-biz server lists the 2850 as the tested server |
| 03:48.04 | lordbaron | Guess dell did a switcharoo on the chipset? |
| 03:48.32 | lordbaron | but more confusing is digium told me to get a compaq with 365 with the e7525 chipset |
| 03:48.34 | kratzers | hmm, as I prepare to deploy a 2850 myself |
| 03:48.36 | lordbaron | same chipset |
| 03:48.57 | CunningPike | lordbaron: You may be being sold a line..... |
| 03:49.06 | lordbaron | I am afraid of this |
| 03:49.32 | lordbaron | it is a long saga..but I have a pile of servers now at the recommendation of digium |
| 03:49.36 | lordbaron | including a new 2950 dell |
| 03:49.38 | CunningPike | lordbaron: Plenty of people use 2850s - I don't personally, but lots of people on this channel and the list do |
| 03:50.38 | lordbaron | ok..is it unrealistic to expect a single server to handle 4 t1's with all usres on the phone with a 'simple' dialplan? |
| 03:50.51 | lordbaron | server does nothing more |
| 03:51.05 | lowlevel | lord: yeah sounds unrealistic to me ;) |
| 03:51.18 | lowlevel | lord: especially if they all start conferencing |
| 03:51.26 | lordbaron | no, all outbound calls -- call center |
| 03:51.27 | lowlevel | and having phone orgys etc. |
| 03:51.33 | CunningPike | lordbaron: How many sets? |
| 03:51.51 | lordbaron | 100 sets, 96 users + 4 monitors (listen in) |
| 03:52.07 | CunningPike | lordbaron: Sounds reasonable - how many concurrent calls? |
| 03:52.18 | CunningPike | lordbaron: Also, are you transcoding? |
| 03:52.20 | kratzers | depends on server specs, protocols in use, echo cancellation, etc. I guess |
| 03:52.24 | lowlevel | probably 100ish. :| |
| 03:52.59 | lordbaron | no compression _u_law, hardware echo cancel, 4gb ram, dual 3ghz processor (disabled hyperthread) |
| 03:53.08 | [TK]D-Fender | Witha good card setup you could load up 2 full 4-port cards I'm sure with those needs... |
| 03:53.19 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:53.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:53.28 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 03:55.38 | [shodan] | considering the cost of the telephony hardware vs the pc hardware wouldn't it be more economical to load balance that ? |
| 03:56.15 | lordbaron | Do you have a good doc on how I would do that? I am having so many problems, this is the best option I think |
| 03:56.39 | lordbaron | I am just not sure how that would work with phone registrations, lines, etc. A good doc to explain the dial plans would be helpful |
| 03:56.41 | lowlevel | lord: think your network is adequte for all those phones traffic? |
| 03:57.13 | lordbaron | I think..gigabit back bone, 24 clients per switch, 10 switches all layer 2, going to a single layer 3 |
| 03:57.13 | CunningPike | lordbaron: What problems are you experiencing? |
| 03:57.54 | lordbaron | works fine for 2-3 days, then stops dialing. Outbound calls never connect, incoming calls never answer. The TE412P card appears to be 'hung' |
| 03:58.04 | lordbaron | ztcfg -vv usually solves it |
| 03:58.14 | CunningPike | lordbaron: Have you tried reseating it etc? |
| 03:58.16 | lordbaron | but then a reboot is often necesary |
| 03:58.20 | lowlevel | hmm, k, so it does work for some time atleast |
| 03:58.22 | lordbaron | yes |
| 03:58.36 | lordbaron | yes, 3 days, and over 65000 calls |
| 03:58.40 | CunningPike | lordbaron: Taking off the VPM? |
| 03:58.49 | lordbaron | vpm? |
| 03:59.02 | lordbaron | virtual processor? yes |
| 03:59.50 | lordbaron | zttest will show 99.9875 as average until the problems start, then it is 99.974 and as low as 99.33 |
| 04:00.27 | CunningPike | lordbaron: Interesting |
| 04:01.10 | CunningPike | lordbaron: The Voice Processing Module - the plug-in hardware echo cancellation module that makes your 410P a 412P |
| 04:01.16 | CunningPike | ~vpm |
| 04:01.45 | lordbaron | ah...so that could be physically removed to be tested? The only test I have done is echocancel=no |
| 04:04.06 | CunningPike | lordbaron: So you could rule it out as a possible failure point |
| 04:04.24 | lordbaron | sure, makes sense. Is this easily identified? |
| 04:05.40 | lordbaron | I see the pics on the digium site...guess the purple part is the vpm? |
| 04:05.52 | CunningPike | lordbaron: It's a daughterboard that plugs on to the PRI card |
| 04:06.07 | CunningPike | Ours was green, but ymmv - ours was one of the older ones |
| 04:06.15 | *** join/#asterisk tlow (i=anon@gateway/tor/x-6a1a3b2a1c4f8685) |
| 04:06.16 | *** join/#asterisk bprice20 (n=brandon@cpe-72-224-53-142.nycap.res.rr.com) |
| 04:06.23 | *** join/#asterisk dacleric (n=dacleric@p54821CEE.dip0.t-ipconnect.de) |
| 04:06.50 | lordbaron | as I understand the echo, since I am not compressing and using t1's, this should not be required anyway. Is this correct? |
| 04:06.50 | bprice20 | hey I am using 1.2.10 and when tossing calls from one asterisk box to another I always receive a 603 declined |
| 04:06.59 | bprice20 | any assistance would be greatly appreciated |
| 04:07.40 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:09.21 | CunningPike | lordbaron: It depends - if you are getting far end echo, you might need it. Without it, s/w e/c takes over. You can test this by editing your modprobe.conf to add vpmsupport=0 to the line that loads wct4xxp |
| 04:09.33 | CunningPike | lordbaron: That will disable the VPM |
| 04:09.45 | lordbaron | ok, great. Thanks, I will try this |
| 04:09.54 | CunningPike | lordbaron: Worth a shot....... |
| 04:10.22 | CunningPike | bprice20: Try carefully placing your calls instead of 'tossing' them ;) |
| 04:10.28 | lordbaron | yes it is. Do you know of any docs explaining loadbalancing a server? |
| 04:10.37 | bprice20 | gee thx |
| 04:10.52 | CunningPike | lordbaron: Are you there yet? |
| 04:10.53 | bprice20 | funny though |
| 04:10.54 | tlow | i was just configuring a load balancer, you want a foundry doc ? |
| 04:11.11 | lordbaron | yes, that would be nice |
| 04:11.13 | CunningPike | bprice20: ;) |
| 04:11.15 | tlow | http://www.netapp.com/ftp/foundry_serveriron.pdf |
| 04:12.03 | CunningPike | Now, I have a question - which of the Sipura SPA-3000 options for DTMF corresponds to RFC2388? |
| 04:12.55 | CunningPike | I have InBand, AVT and INFO |
| 04:12.59 | CunningPike | :S |
| 04:13.09 | lordbaron | we use inband |
| 04:13.12 | benjk | CunningPike, it depends whether you get head or tails call as a result of tossing the call |
| 04:13.17 | lordbaron | this worked with our provider |
| 04:13.27 | CunningPike | benjk: Heh heh |
| 04:13.52 | CunningPike | lordbaron: OK - thanks - I'll try that..... |
| 04:14.32 | lordbaron | tlow: thanks..are you doing this with asterisk? |
| 04:16.08 | tlow | with asterisk i use a cisco load balancer, i use foundry for http servers. |
| 04:16.33 | lordbaron | do you share outbound lines between more than 1 box? |
| 04:16.36 | tlow | but its the same topology mostly, one virtual server, then you add real servers. |
| 04:16.46 | tlow | i dont do tdm. |
| 04:17.21 | lordbaron | is it possible for a user registered on box a to dial out box a, but if no lines avail on a, dial out of b? |
| 04:19.03 | benjk | you need to register on both servers or allow unauthenticated calls or authenticate by ip |
| 04:19.54 | lordbaron | same sip ext listed on both servers? (same user/pass) ? Or register 2 different connections? |
| 04:20.27 | benjk | either way, you have to allow the user to place a call on both servers |
| 04:21.10 | lordbaron | ok, so once that is done, what does the dial plan look like? The zapata.conf where you setup groups..can group1 span 2 servers? |
| 04:21.52 | benjk | with TDMoE it can, but that's not exactly rock solid, lots of kernel panics and stuff |
| 04:22.23 | lordbaron | ya, I am looking for a rock solid method. My current method is 2 day solid, then need a hard reboot |
| 04:22.42 | benjk | then TDMoE is probably not what you want |
| 04:23.12 | *** join/#asterisk b1ch0 (n=ralabiso@host-206-107-150-177.acelerate.net) |
| 04:24.12 | lordbaron | so if no lines are available on group 1, can group 2 be defined as a group on server 'B'? |
| 04:24.24 | lordbaron | I feel real dumb here, but I can't find a doc on this |
| 04:25.03 | benjk | do you want to protect against server failure or just trunk failure? |
| 04:25.41 | lordbaron | well the current problem appears to be that there is too much activity on the TE412P, 96 users all the time |
| 04:25.45 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 04:25.48 | lordbaron | so I am trying to split up the load |
| 04:26.01 | lordbaron | 1 T1 is local calls, 3 are long distance |
| 04:26.11 | benjk | you could just do DNS load balancing |
| 04:27.07 | lordbaron | so if the local t1 is on 'A', then 'B' can have a plan to try 'A' first for local calls? |
| 04:27.11 | *** join/#asterisk b1ch0 (n=ralabiso@host-206-107-150-177.acelerate.net) |
| 04:27.13 | benjk | otherwise, write a dial macro that examines the number of channels in use and then places the call either locally or via an IAX peer through the other server |
| 04:27.54 | benjk | that won't load balance local calls then though |
| 04:28.10 | lordbaron | but that is few as compared to the remaining |
| 04:28.11 | benjk | all your local calls would again be concentrated on server A |
| 04:28.20 | lordbaron | a single t1 |
| 04:28.25 | benjk | fair enough |
| 04:28.35 | lordbaron | I agree..not perfect |
| 04:28.38 | benjk | anyway, you should peer the two servers |
| 04:28.46 | lordbaron | ok, I will work on this. Thank you |
| 04:28.48 | benjk | via IAX |
| 04:29.02 | benjk | you need to add a user and a peer entry in each server's iax.conf |
| 04:29.18 | benjk | user entry of A corresponding to peer entry on B and vice versa |
| 04:29.37 | lordbaron | ok, that makes sense |
| 04:29.57 | benjk | then on A, the ISDN trunks of B are available via IAX and vice versa |
| 04:30.06 | lordbaron | is there a way to define a dial group on A that is a group on B? |
| 04:30.22 | lordbaron | or is it only via dial strings? |
| 04:30.45 | benjk | you'll dial an IAX peer and the grouping is done locally on that machine |
| 04:31.05 | b1ch0 | hi, its my first time |
| 04:31.05 | benjk | like you dial IAX2/foobar@machineB |
| 04:31.22 | benjk | and on machine B it is then dialed as ZAP/g1 |
| 04:31.31 | lordbaron | ok, that makes sense |
| 04:31.45 | benjk | so whatever comes in from A via that IAX peering connection is dialed locally on B as a zap group |
| 04:31.45 | b1ch0 | does anybody tell me if * could be considerer a softswitch or not ? |
| 04:31.54 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:32.14 | lordbaron | Ok, thanks benjk...that helps a lot! |
| 04:32.16 | benjk | yes for various definitions of softswitch |
| 04:32.31 | benjk | no for various other definitions of softswitch |
| 04:32.50 | JT | lowlevel: good to hear you found a lead on that problem |
| 04:33.34 | b1ch0 | mmm, |
| 04:34.05 | b1ch0 | does a softswitch things that * cant ? |
| 04:34.22 | benjk | depends on your definition of softswitch |
| 04:34.50 | lowlevel | jt: just upgraded zaptel and asterisk to latest cvs, and the problem went away totally (for now ;) |
| 04:34.58 | JT | hm |
| 04:35.03 | JT | still with 2 cards? |
| 04:35.07 | lowlevel | nah I only had 1 |
| 04:35.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:35.25 | JT | right |
| 04:35.25 | benjk | does a vehicle do things that a <your favourite brand and make of car> can't ? |
| 04:35.35 | JT | howcome you mentioned multiple cards? |
| 04:35.36 | benjk | depends on your definiton of vehicle |
| 04:35.51 | lowlevel | jt: lordbaron was running more than one in a box |
| 04:35.53 | lowlevel | or is |
| 04:35.55 | benjk | if vehicle includes boats, then yes |
| 04:36.08 | lowlevel | jt: you need more hello kitty wake up mints. |
| 04:36.25 | lowlevel | j/k ;) |
| 04:36.35 | JT | lowlevel: oh, so you summonsed me, but did not continue on with your problem, heh |
| 04:36.44 | lowlevel | jt: exactly |
| 04:36.50 | lowlevel | jt: you didn't reply, so I didn't continue |
| 04:36.51 | lowlevel | heh |
| 04:37.10 | JT | yeah i was busy reading readmes and source for bristuff stuff |
| 04:37.21 | JT | and hadn't checked this window |
| 04:37.22 | lowlevel | no problem |
| 04:37.24 | lowlevel | ;) |
| 04:37.25 | benjk | aka BRI stuff squared |
| 04:37.40 | JT | lowlevel: so just a problem with the version? |
| 04:38.32 | JT | i'd like to nominate this as coolest idea ever, btw: http://www.junghanns.net/en/callback.html |
| 04:38.47 | JT | let's ignore the fact it's implemented in php |
| 04:38.51 | lowlevel | jt: yeah I guess, I really have no idea what was causing it... |
| 04:39.15 | JT | lowlevel: yeah not really sure, were you using testing code? |
| 04:39.43 | lowlevel | nah I was using stable code supposedlyt, and now I'm on latest cvs |
| 04:39.43 | b1ch0 | well i know the wiki definition of softswitch, my question was if i can use * (maibe with SER) working to control and switch calls between a pstn user and voip world (or maybe to another pstn user) |
| 04:39.54 | lowlevel | er, subversion |
| 04:39.57 | JT | b1ch0: yes |
| 04:40.20 | JT | lowlevel: hmm |
| 04:41.14 | [shodan] | hmm , if I get this correctly whenever I open a channel to my * box , the line "exten => s,1,*******" always get's executed first ? |
| 04:42.10 | benjk | not exactly |
| 04:43.14 | benjk | if you drop a call into a context which has an s extension and you don't specify any extension explicitly, then it will use s |
| 04:43.50 | [shodan] | oh I need to read some more , I just noticed the local context has a s,1 line but so does the demo context and it's included in local , so there are 2 s,1 lines |
| 04:44.28 | lowlevel | shodan: get that 'Asterisk - The Future of Telephony' book/pdf |
| 04:44.34 | CunningPike | ~thebook |
| 04:44.37 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:44.37 | lowlevel | shodan: it does a really good job of explaining that |
| 04:44.40 | lowlevel | ahhaah |
| 04:44.41 | CunningPike | :D |
| 04:44.45 | lowlevel | the book. |
| 04:44.47 | lowlevel | figures ;) |
| 04:44.51 | CunningPike | G'night all |
| 04:44.57 | lowlevel | night pikey |
| 04:45.03 | [shodan] | k |
| 04:45.38 | *** join/#asterisk b1ch0 (n=ralabiso@host-206-107-150-177.acelerate.net) |
| 04:46.33 | lowlevel | gawd I can't wait to ditch this stupid air conditioner |
| 04:46.39 | lowlevel | its so freaking loud |
| 04:47.57 | benjk | for the avoidance of doubt, an air conditioner is generally not considered a softswitch and it usually does things a softswitch can't do, likewise a softswitch usually does things an air conditioner can't do |
| 04:48.32 | JT | hrm, so i'm not sure if it was lowlevel, or more lordbaron, but someone was talking about TDM400Ps and TDM2400Ps |
| 04:48.53 | JT | there's a company that make a 12port analogue board, too, which takes 400P modules |
| 04:48.58 | JT | just to muddy the waters |
| 04:49.27 | [shodan] | my softswitch is a reverse air conditionner ;) |
| 04:50.09 | JT | reverse cycle with no forward cycle |
| 04:52.09 | *** join/#asterisk b1ch0 (n=ralabiso@host-206-107-150-177.acelerate.net) |
| 04:52.41 | benjk | JT, for sure you mean polarity reversal |
| 04:53.45 | [TK]D-Fender | benjk : Which clearly supports my theory that apples are nideed MUCH better than oranges... |
| 04:54.05 | benjk | depends |
| 04:54.19 | benjk | if you want to hit someone hard, they would indeed |
| 04:55.02 | benjk | they also can also be kept without rotting for much longer |
| 04:55.04 | JT | benjk: heh |
| 04:55.21 | *** join/#asterisk justnulling2 (i=justnull@ool-182e41b0.dyn.optonline.net) |
| 04:55.31 | benjk | but if you like it colourful, some oranges may be better than some apples |
| 04:55.42 | JT | if you wanted to make a stink bomb, orange would be superior |
| 04:55.58 | benjk | if you want to make cider on the other hand ... |
| 04:56.08 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 04:56.42 | JT | if you wanted to make orange fragrance dishwashing detergent, neither would do |
| 04:57.05 | benjk | now, that's a surprise |
| 04:57.24 | lowlevel | jt: insteresting (re: 12port board that takes the 4port modules from digium's TDM2400P) |
| 04:57.59 | lowlevel | jt: same connector on card ? (50pin) |
| 04:58.39 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 04:58.58 | JT | might be different |
| 04:59.01 | JT | not sure |
| 04:59.02 | JT | actually no |
| 04:59.24 | b1ch0 | hi again, does anybody integrated (or heard about) with Ericsson GSM |
| 04:59.54 | b1ch0 | i mean, * used with a GSM telephony system |
| 05:00.03 | benjk | Junghanns has a GSM PCI card |
| 05:00.09 | b1ch0 | handling all voip trafic |
| 05:00.12 | JT | i remember, 3 X RJ45s deliver the linex, lowlevel |
| 05:00.18 | JT | lines |
| 05:00.27 | lowlevel | jt; yuk |
| 05:00.41 | JT | lowlevel: yeah? connectors are easy to get though |
| 05:00.52 | JT | so what's yuck |
| 05:01.03 | benjk | once you get into that kind of density with analog, you may want to consider a T1 channel bank |
| 05:01.04 | lowlevel | I prefer the prewired cable > bix :/ |
| 05:01.46 | lowlevel | benjk: I've only had ot deal with 6 to 8 lines so far, and usually , they analog pairs are already in place |
| 05:02.57 | lowlevel | our telco's here make it ridiculously expensive to go digital anyway |
| 05:04.05 | JT | lowlevel: how is a cable going from a TDP2400P any different to 3 X RJ45s functionally? |
| 05:04.08 | JT | i don't get it |
| 05:04.55 | lowlevel | jt: well, once its hooked up its irrelevant of course |
| 05:05.21 | JT | difference being it's much easier for any person to make the appropriate RJ45 cables :) |
| 05:06.55 | JT | http://www.openvox.com.cn/products.php?genre_id=17 |
| 05:07.02 | JT | A1200P |
| 05:07.11 | bprice20 | ok I am stuck here folks, I have several asterisk boxes and am unable to send calls from any one box to any other they always reply with a 603 |
| 05:07.14 | JT | full length PCI, as you can imagine |
| 05:07.20 | bprice20 | as anyone seen this before? |
| 05:07.34 | lowlevel | jt: *shrug* really its easier to patch from bix > bix than to mess with rj45 cables for me. |
| 05:07.49 | lowlevel | jt; of course one needs the appropriate tool |
| 05:08.01 | bprice20 | and this is only after upgrading to 1.2.10 |
| 05:08.02 | *** part/#asterisk bprice20 (n=brandon@cpe-72-224-53-142.nycap.res.rr.com) |
| 05:08.14 | JT | most people into telephony or networking have a cable crimper |
| 05:08.19 | JT | you actually don't even need one |
| 05:08.25 | lowlevel | jt; yeah I have one, btu I hate to use it ;) |
| 05:08.46 | JT | you could buy premade cables, cut ends off one side, and punchdown the other sides into Krone punchdown blocks |
| 05:09.30 | lowlevel | jt: Krone? |
| 05:10.05 | JT | i have no idea what you call them over there |
| 05:10.08 | lowlevel | 110? |
| 05:10.13 | JT | krone is the main brand that make them |
| 05:10.13 | JT | yes |
| 05:10.15 | lowlevel | then I'de have to buy a stupid 110/66 tool |
| 05:10.17 | lowlevel | I dont wanna do that |
| 05:10.18 | lowlevel | ;) |
| 05:10.22 | lowlevel | :P |
| 05:10.44 | JT | err, you can buy the expensive tool, or you can even use a couple of dollar disposable tool to do punchdown |
| 05:10.56 | JT | you can even use a screwdriver but that is painful |
| 05:11.01 | lowlevel | jt: I woudlnt recommend it |
| 05:11.21 | JT | anyway, i was assuming you were wanting it terminated in a professional manner, hence 110 |
| 05:11.36 | JT | the disposable krone tools work fine |
| 05:11.44 | JT | they just don't have automatic cutters |
| 05:11.51 | JT | which are usually more annoying than good |
| 05:11.55 | lowlevel | jt; BIX is more common here, and my tool has the automatic cutters |
| 05:11.56 | lowlevel | :) |
| 05:12.11 | JT | bix? |
| 05:12.12 | lowlevel | it looks crpapy to have 6 bix blocks on the wall, and 1 110 |
| 05:12.27 | lowlevel | jt: yeah its the northern telecom punchdown |
| 05:12.43 | JT | what's the difference? |
| 05:13.16 | lowlevel | jt; size/spacing/price |
| 05:13.23 | lowlevel | jt: other than that, no diff. |
| 05:13.48 | JT | ah ok |
| 05:13.51 | lowlevel | we're talking about a peice of plastic, with metal contacts embedded in it. |
| 05:13.52 | lowlevel | heh |
| 05:14.27 | JT | yep |
| 05:14.53 | JT | anyway, 8P8C is much easier than big high density connector :P |
| 05:15.25 | lowlevel | sigh. |
| 05:15.54 | lowlevel | I'de still rather slap in a pre-wired 25pair cable, with the right cable, and snap the bix block in. |
| 05:15.55 | lowlevel | :P |
| 05:16.18 | lowlevel | er right connector |
| 05:16.19 | JT | but obviously it's the right tool for the job when you're trying to pull 24 lines out of a PCI backplate |
| 05:16.33 | lowlevel | jt: yep, its unbeatable |
| 05:16.45 | lowlevel | jt; no broken clips, etc. |
| 05:16.52 | JT | heh |
| 05:17.02 | lowlevel | I screw up those rj45/rj11 connectors too much anyway |
| 05:17.03 | lowlevel | ;) |
| 05:17.15 | x86 | for some reason I can't get any of my static queue members to ring when I exec a Queue() command in my dialplan |
| 05:17.18 | JT | don't want to know what your T1s are like if you have any :P |
| 05:17.24 | x86 | anyone have experience with queues? |
| 05:17.28 | lowlevel | jt: nah, dont have any |
| 05:17.32 | JT | heh |
| 05:18.00 | *** join/#asterisk bprice20 (n=brandon@cpe-72-224-53-142.nycap.res.rr.com) |
| 05:18.15 | x86 | using RealTime queues with MySQL |
| 05:18.24 | x86 | defining the queue was easy enough |
| 05:18.34 | *** part/#asterisk bprice20 (n=brandon@cpe-72-224-53-142.nycap.res.rr.com) |
| 05:18.39 | x86 | but i'm not sure I did the queue members and/or agents setup correctly |
| 05:19.15 | b1ch0 | . |
| 05:20.39 | *** part/#asterisk b1ch0 (n=ralabiso@host-206-107-150-177.acelerate.net) |
| 05:21.58 | JT | need to get some 25pair cable for my channel bank |
| 05:22.40 | JT | electrical wholesales screwed me over twice on the weekend |
| 05:22.56 | JT | went there on saturday, they had a stocktake on |
| 05:23.15 | JT | called on sunday, said they'd be open till 4, arrived just before 3, they were closed |
| 05:36.03 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 05:37.46 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-53.intrastar.net) |
| 05:41.01 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 05:41.06 | denon | you lost? :) |
| 05:41.16 | J4k3 | nah, I just remembered to identify before trying to join this channel ;) |
| 05:41.21 | denon | hehe |
| 05:41.55 | denon | a cable must have come lose at the top of the big stick, J4k3's thinkin' straight |
| 05:41.57 | denon | trixbox? ew |
| 05:42.04 | J4k3 | well... had no fun with it until I bothered to read the instructions |
| 05:42.32 | denon | fdisk, install fresh debian, svn co asterisk :) |
| 05:42.32 | J4k3 | well, trixbox isn't the final destination... I just wanted to try something I knew "worked" |
| 05:42.37 | J4k3 | so I knew what to expect when I built it myself later. |
| 05:42.51 | denon | spose |
| 05:43.33 | J4k3 | the machine is also trash, it won't pass 24 hour memtest |
| 05:43.40 | J4k3 | with any *good* ram stick I stick in it. |
| 05:44.04 | denon | this the same motherboard you tried to jtag with an nbd cell phone special? |
| 05:44.29 | J4k3 | I doubt I could actually compile something on it, its already kernel panic'd once |
| 05:44.39 | J4k3 | in about 20 hours. |
| 05:45.37 | *** join/#asterisk daysmen3 (n=primus@host86-139-116-74.range86-139.btcentralplus.com) |
| 05:45.41 | J4k3 | nah, this is some piece of junk gigabyte built. |
| 05:45.56 | J4k3 | kt133/686a tbird 700 junk. |
| 05:46.03 | denon | actually, there are a couple pretty good gigabyte boards |
| 05:46.08 | denon | sounds like that's not one of em though |
| 05:46.11 | J4k3 | this wasn't one of them ;) |
| 05:46.15 | denon | hehe |
| 05:46.22 | J4k3 | GA-586S |
| 05:46.31 | J4k3 | first gigabyte board I ever bought |
| 05:46.41 | denon | I think it was the GA-6vx7-4x we used in a ton of workstations back in the dya |
| 05:46.42 | denon | er day |
| 05:46.53 | denon | or at least that string of chars rings a bell |
| 05:46.55 | J4k3 | first 75 mhz fsb socket 7 board, too... quite solid (at 66, never put a 75 mhz cpu in it...) |
| 05:47.39 | denon | all time favorite mobo is still intel's vx440fx ppro boards |
| 05:47.41 | denon | solid as a rock |
| 05:47.45 | denon | I still know of some running |
| 05:47.55 | denon | er vs440fx |
| 05:48.01 | J4k3 | yep |
| 05:48.09 | denon | pretty much anything with that chipset owned |
| 05:48.12 | J4k3 | the only thing that will kill those is the caps drying out |
| 05:48.37 | *** join/#asterisk hunmonk (n=hunmonk@pool-71-97-41-106.dfw.dsl-w.verizon.net) |
| 05:48.42 | denon | in fact, I think I know of some IBM Intellistations running that same chipset still running as well |
| 05:48.45 | J4k3 | and they're not bad caps... its just that they're not designed to last this long :) |
| 05:49.22 | J4k3 | I gotta order me up a local DID |
| 05:49.35 | J4k3 | or an 800#... having only outgoing calling is kinda weak. |
| 05:49.50 | denon | or a PRI |
| 05:50.16 | denon | DIDs are more fun in blocks of 100 |
| 05:50.25 | J4k3 | I'd only consider that with the ILEC |
| 05:50.31 | J4k3 | since I can't port #'s otherwise |
| 05:50.43 | J4k3 | but they will port local #'s to a PRI |
| 05:50.49 | J4k3 | well, a PRI you buy from them |
| 05:51.04 | J4k3 | but at $650/month just for the circuit and trunking, thats painful |
| 05:52.12 | x86 | J4k3: you can do LNP with an SS7 trunk ;-) |
| 05:52.38 | denon | x86: prollly not on trixbox <G> |
| 05:52.53 | x86 | denon: hehe, chan_ss7.so is considered beta anyway ;) |
| 05:53.00 | denon | yeah |
| 05:53.21 | x86 | man |
| 05:53.28 | x86 | i wish i could get this queue shit working :( |
| 05:53.34 | x86 | i dont want the agent to have to login/logout... |
| 05:53.40 | x86 | they should always be logged in ;) |
| 05:53.50 | x86 | how can i do that? |
| 05:54.07 | denon | static agents? |
| 05:54.13 | J4k3 | denon: pft, this is going straight on openwrt-on-broadcom. |
| 05:54.14 | J4k3 | ... not |
| 05:54.17 | *** part/#asterisk hunmonk (n=hunmonk@pool-71-97-41-106.dfw.dsl-w.verizon.net) |
| 05:54.28 | x86 | denon: that's what i'm trying to do |
| 05:55.34 | x86 | denon: in agents.conf, i have group = 1, agent => 100,0000,Secret Agent, agent => 101,0000,Double Agent, and in queues.conf i tell it members => @1 |
| 05:56.01 | x86 | but an inbound call will sit in the queue forever until an agent does an AgentLogin or AgentCallbackLogin |
| 05:56.12 | x86 | or the timeout is reached, obviously ;) |
| 05:56.36 | denon | dunno, I avoid agents like the plague |
| 05:56.43 | x86 | not sure what I'm doing wrong |
| 05:56.52 | x86 | I think it might be that I'm trying to use agent groups |
| 05:56.57 | x86 | perhaps incorrectly even ;) |
| 05:57.19 | x86 | i wonder if i do member => 100, member => 101 in queues.conf if that will help |
| 06:00.29 | lowlevel | whoa, this potato salad is amazing |
| 06:02.36 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 06:04.43 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:06.06 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:11.09 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 06:14.21 | *** join/#asterisk Will_Syd (n=william@c211-30-149-97.rivrw4.nsw.optusnet.com.au) |
| 06:15.25 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 06:21.10 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:23.15 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 06:25.05 | *** join/#asterisk Dico_ (n=k@219.93.185.90) |
| 06:27.03 | *** join/#asterisk KaiHanari (n=Kai@stjhnf01-22-142163031139.nf.sympatico.ca) |
| 06:29.07 | KaiHanari | nomatter what i try, my SIP extension wont allow my softphone to authenticate.... but it looks like the phone is set up right because im using similar settings for FWD, and they work, the asterisk cli is telling me Username/auth name mismatch |
| 06:29.18 | KaiHanari | have i forgotten to do something in a conf? |
| 06:29.53 | denon | KaiHanari: double check your sip.conf, and dont forget to reload |
| 06:30.53 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 06:31.02 | KaiHanari | denon, been using freepbx to do it, so i checked the sip.conf, doesnt appear that freepbx puts anything in it... yet it did put an entry in extensions.conf.... so i manually entered the sip info in sip.conf, and it still didnt work. |
| 06:32.04 | denon | you might want to try #freepbx or somewhere that knows more about it |
| 06:32.12 | KaiHanari | agh |
| 06:32.15 | KaiHanari | noticed soemthing |
| 06:32.25 | KaiHanari | sip show users returns an empty list |
| 06:32.48 | denon | all these silly asterisk "distributions" add a level of complexity .. |
| 06:32.53 | denon | that make it so hard to troubleshoot |
| 06:33.45 | KaiHanari | denon, as i said. i tried manually entering in the sip information in the sip.conf. and freepbx is not a distribution, its only a web frontend, i have official asterisk from source code on the asterisk site |
| 06:34.09 | JT | <PROTECTED> |
| 06:34.17 | JT | it modifies the way you interact with asterisk |
| 06:34.35 | KaiHanari | that doesnt matter, the problem is what matters |
| 06:34.58 | denon | heh |
| 06:35.17 | denon | actually, it's the solution that matters |
| 06:35.29 | KaiHanari | true |
| 06:38.49 | KaiHanari | http://www.nomorepasting.com/paste.php?pasteID=67460 |
| 06:38.50 | KaiHanari | :/ |
| 06:47.11 | *** join/#asterisk vlt (n=dm@p54B34118.dip0.t-ipconnect.de) |
| 06:50.17 | vlt | Good Morning. Same Problem as yesterday. My exernal ip has changed and now my asterisk behind NAT can't register to (three different) SIP accounts. Yesterday I changed "externip=1.2.3.4" to "externhost=a.domain.name" and debugging sip tells me that the correct IP Address is used while trying to register but it does'nt work. Any idea? |
| 06:52.27 | bionoid | vlt: Can you monitor traffic on the NAT firewall? Ie observe if there is any traffic destined for your asterisk box that doesn't get forwarded? |
| 06:54.45 | vlt | bionoid: Yes I can. It's a Debian Sarge Box so I'll LOG some packets and see ... |
| 06:55.56 | bionoid | vlt: apt-get install iptraf (if you don't have it already) |
| 07:00.24 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 07:00.38 | vlt | bionoid: Thank you. I'll try that. I just began to create LOG rules in iptables and watch syslog ... ;-) |
| 07:04.50 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:19.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:22.31 | *** join/#asterisk BugKham (n=bugkham@ppp-58.8.11.174.revip2.asianet.co.th) |
| 07:23.37 | BugKham | how do we get the variable "EXTEN" or "DNID" in agi? |
| 07:23.41 | BugKham | get_variable("EXTEN") doesn't work for me |
| 07:24.14 | [shodan] | anyone knows if the grandstream GXV-3000 works with h.261 or h.263 ? |
| 07:27.20 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 07:32.52 | vlt | bionoid: WTF!?! iptraf (thanks for the hint) tells me that each UDP:5060 packet from LAN is sent out to ppp0 *from* yeserday's IP address! |
| 07:33.48 | bionoid | vlt: Sounds like an entire evening of fun ;) |
| 07:37.10 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 07:38.50 | vlt | bionoid: For better debugging I stopped asterisk and tried to send some packets manually. After about 2 minutes I restarted asterisk and registering worked immediatly. Now packets are sent by Debian NAT from my new IP ... |
| 07:40.09 | bionoid | Hmm that sounds kind of sketchy ;P I don't like when things (on a sane operating system) starts working for no particular reason.. but hey - better than broken anyway ;) |
| 07:40.31 | vlt | ;) |
| 07:41.07 | bionoid | vlt: Now, please, fix the noise issues on my TDM400P :\ |
| 07:42.56 | *** join/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
| 07:43.28 | sjobeck | hey all, hope all is well, any one familiar with sangoma cards for a question |
| 07:44.01 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 07:44.54 | BugKham | hi, anyone using phpagi here? |
| 07:49.22 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 07:54.34 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 07:58.36 | phearless | asterisk is awfully complex :( |
| 07:58.45 | sjobeck | y |
| 08:00.27 | mog | whats the prob bob |
| 08:03.41 | *** join/#asterisk Wonka (i=produzie@madwifi/support/wonka) |
| 08:04.06 | Wonka | morning... |
| 08:04.07 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 08:04.09 | tengulre | hi,all |
| 08:04.22 | tengulre | how to using asterisk to register remote openh323GK? |
| 08:04.52 | Wonka | can anyone help me with chan_capi compile problems (latest chan-capi-cm-HEAD, latest asterisk-svn, "make CC=gcc-3.4" in chan_capi)? |
| 08:05.02 | Wonka | /usr/local/include/asterisk/compat.h:23: error: syntax error before "__extension__" |
| 08:05.38 | Dico_ | Wonka, do you really need the SVN version ? |
| 08:05.40 | Wonka | line 23 is "char* strsep(char** str, const char* delims);" |
| 08:05.44 | mog | you are gonna have a lot of problems |
| 08:05.50 | mog | id wait for them to update |
| 08:05.55 | mog | or push svn of asterisk back some |
| 08:06.13 | Wonka | mh. last time i tried using *, months ago, nothing else would work at all... |
| 08:06.50 | Wonka | i want to use a HFC-S with mISDN in NT mode and a AVM B1 with chan_capi |
| 08:06.52 | Dico_ | Wonka ; try the stable version ? |
| 08:07.09 | Wonka | is there a known working chan_capi for it? |
| 08:07.09 | benjk | Wonka, use BRIstuff |
| 08:07.44 | Wonka | benjk: at least zaptel tended to very brutally hang my machine... |
| 08:07.50 | Dico_ | Wonka, sorry there is no capi in stable version |
| 08:08.11 | benjk | BRIstuff is the most straightforward method to do BRI with * |
| 08:08.26 | Wonka | mh. I need capi, my externel line is attached to that B1, which is capi. |
| 08:08.37 | benjk | anything else requires fiddling with things |
| 08:08.49 | benjk | nonsense |
| 08:08.52 | benjk | you don't need capi |
| 08:09.02 | Wonka | what then? |
| 08:09.13 | benjk | BRIstuff and the drivers that come with it |
| 08:09.20 | Wonka | for an AVM B1? |
| 08:09.43 | benjk | didn't you say HFC? |
| 08:09.55 | benjk | if it is HFC it'll work with BRIstuff |
| 08:10.00 | Wonka | the HFC works nicely with mISDN |
| 08:10.05 | Wonka | my problem is the B1 |
| 08:10.06 | *** join/#asterisk SHad|Work (n=kvirc@84.255.228.2) |
| 08:10.17 | SHad|Work | Hi |
| 08:10.40 | SHad|Work | Does anyone here have any experience with Sipura/Linksys SPA-942 ? |
| 08:10.44 | benjk | not all too familiar with AVM cards, are you saying they're not HFC based? |
| 08:10.57 | Wonka | yes |
| 08:11.04 | Wonka | it's an _active_ ISDN card |
| 08:11.18 | Wonka | one of those who can do G3 faxing in firmware, and stuff |
| 08:11.18 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 08:11.19 | benjk | HFC has nothing to do with passive and active |
| 08:11.30 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 08:11.32 | Wonka | and also, it's an ISA card... |
| 08:11.39 | benjk | there are passive and active HFC cards |
| 08:11.43 | Wonka | and definitively has no HFC chip |
| 08:11.47 | benjk | ok |
| 08:12.36 | benjk | well, the fact that the leading supplier of ISDN for Asterisk used to author chan_capi and abandoned capi for something else should tell you something |
| 08:12.44 | benjk | perhaps it is time to go PCI |
| 08:13.29 | benjk | at the very least it should tell you that this isn't going to be a smooth ride |
| 08:13.51 | Wonka | someone other took over chan_capi... |
| 08:14.02 | sjobeck | hi all, do I need "crc-ccitt" for zaptel on 2.4 ? just build wanpipe for sangoma but now zaptel wont start. |
| 08:14.48 | sjobeck | SHad: i know that phone a bit |
| 08:14.54 | BugKham | anyone using agi here? |
| 08:14.55 | benjk | I didn't say that it isn't maintained anymore, but the very fact that kapejod abandoned it for something else should tell you that this was a route not worth pursuing any further |
| 08:15.21 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:15.35 | SHad|Work | sjobeck: I've got a weird auth problem, I can call the phone, but when I try to call another one from it I get an auth failure |
| 08:15.49 | SHad|Work | sjobeck: is there something I forgot to set? |
| 08:15.58 | zoa | chan_misdn works best for us |
| 08:16.19 | sjobeck | SHad: can call in but not out |
| 08:16.23 | sjobeck | ? |
| 08:16.29 | zoa | http://www.asteriskguru.com/tutorials/bri.html -> some more info here |
| 08:16.35 | benjk | you will find that the more time passes the more "anything else" will work better than chan_capi |
| 08:17.08 | Wonka | benjk: which absolutely doesn't help me when all i have is this AVM B1, and only the ISA slot it's in |
| 08:17.30 | benjk | as I said, its not going to be a smooth ride |
| 08:18.02 | benjk | people are moving away from both ISA and capi to other technologies for a reason |
| 08:18.54 | Wonka | zoa: interesting, they mention the B1 PCI working with mISDN |
| 08:20.46 | bionoid | I'm having some weird issues with my Asterisk - I had a 100FXO (cheap-ass $10 thingie) that worked perfectly well, then installed a TDM400P. The audio quality is horrible (compared to the 100fxo), and I can't seem to get it better by adjusting the regular echo cancelling parameters etc. Any tips for that? |
| 08:21.34 | benjk | probably an interrupt issue |
| 08:22.12 | SHad|Work | sjobeck: yes |
| 08:23.49 | bionoid | benjk: I've checked the hardware interrupts, no conflicts there. |
| 08:23.57 | SHad|Work | sjobeck: does it perhaps use md5 by default? |
| 08:24.11 | phearless | <mog> whats the prob bob |
| 08:24.23 | phearless | when I make a call it is written : |
| 08:24.31 | phearless | call failed, 404 not found |
| 08:24.33 | benjk | they may not share, but it is still possible that your machine can't handle two cards |
| 08:24.35 | phearless | on the softphone |
| 08:24.43 | phearless | and I try to call my cisco voip phone |
| 08:24.47 | bionoid | benjk: There is only one card, though |
| 08:25.32 | benjk | ah, misunderstanding then, I thought you added the TDM400 |
| 08:26.29 | bionoid | Nope, replaced it :) |
| 08:26.33 | phearless | and I do not know what to do to find the problem |
| 08:26.36 | phearless | ... |
| 08:26.40 | phearless | <PROTECTED> |
| 08:27.55 | BugKham | anyone knows why $cid = $agi->parse_callerid(); is returning "Array"? |
| 08:28.18 | phearless | anybody can help me ? |
| 08:28.40 | phearless | I am trying to make asterisk work everyday since 2 weeks and half ... |
| 08:28.42 | benjk | callerid is an array |
| 08:28.51 | hads|home | Makes sense :) |
| 08:28.52 | BugKham | benjk: ic |
| 08:29.07 | benjk | made up of callerid name and callerid number |
| 08:29.21 | benjk | "Foobar" <012345> |
| 08:29.44 | BugKham | benjk: get_variable(); also returns an array right? |
| 08:30.14 | benjk | does it? I am not too familiar with perl agi stuff |
| 08:30.26 | benjk | but I know that callerid is actually two values |
| 08:30.29 | BugKham | benjk: I tried get_variable("DNID"); and it returns an array also |
| 08:30.57 | benjk | I do this in C, so my functions are not always going to be the same as yours ;) |
| 08:31.45 | benjk | DNID should be a single string, technically an array of chars |
| 08:32.50 | [shodan] | why would I get an error "Cannot find extension context 'local' " because I have a [local] context in my extensions.conf , the context= from sip.conf refers to context in extensions.conf right ? |
| 08:33.02 | benjk | bionoid, how many modules in that TDM400? |
| 08:33.40 | [shodan] | it used to work earlier, I must have screwed something up :( |
| 08:33.56 | benjk | [shodan] verify if you *really* have such a context by doing show dialplan on the CLI |
| 08:34.30 | benjk | from time to time I get things that look weird because I swear I have stuff in extensions.conf |
| 08:34.55 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 08:34.55 | benjk | but then when I check with show dialplan, I realise that what I thought was there, didn't actually end up that way in memory |
| 08:35.34 | benjk | so always verify with show dialplan first to see what Asterisk actually made of your extensions.conf |
| 08:36.02 | [shodan] | oh it didn't work , there's only parkedcalls in there ! |
| 08:36.14 | benjk | see :) |
| 08:37.12 | [shodan] | oh ! |
| 08:37.22 | [shodan] | 4.0K -rw-r----- 1 root root 3.0K Aug 22 04:36 extensions.conf |
| 08:38.24 | bionoid | benjk: 2xFXO |
| 08:38.30 | bionoid | on channel 3/4 |
| 08:39.06 | benjk | have you swapped them around and tried to use just a single module in different slots? |
| 08:40.05 | vlt | I have set up a queue in asterisk. When I dial its extension from connected SIP phones I hear moh (gsm file in native mode) and the youarenext voice. The same when I dial in from outside over a SIP peer I registered to. But when I use one of the SIP phones and dial out from asterisk over a SIP provider back to the same asterisk server I only hear ringing (though I can watch asterisk playing moh and voice in CLI). When a queue member answers they |
| 08:40.05 | vlt | <PROTECTED> |
| 08:40.32 | sjobeck | hi all: any sangoma experts out there tonight/morning? |
| 08:40.54 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 08:46.00 | bionoid | benjk: No, that I havn't tried. I only tried moving the card itself to different PCI slots |
| 08:46.53 | benjk | I'd also swap the modules around and try each of them without the other in all the slots |
| 08:47.38 | benjk | if there are no differences, I'd build another system from scratch with stock software and stock example configs |
| 08:47.57 | benjk | if that still shows no change, I'd go back to Digium and ask them to replace the card |
| 08:49.09 | benjk | or at the very least open a ticket and see if they can fix it |
| 08:50.36 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 08:51.53 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:53.26 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:53.28 | *** join/#asterisk Vec (n=Vector@dsl-165-165-183.telkomadsl.co.za) |
| 08:53.43 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 08:59.22 | phearless | hello ! |
| 08:59.29 | *** part/#asterisk Wonka (i=produzie@madwifi/support/wonka) |
| 08:59.35 | phearless | anybody understand what is the *context* in sip.conf ? |
| 09:00.02 | RoyK | ~docs |
| 09:00.07 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:00.08 | RoyK | ~book |
| 09:00.10 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:00.11 | RoyK | ~rtfm |
| 09:00.14 | jbot | methinks rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 09:00.44 | phearless | I RTFM since ages |
| 09:01.51 | phearless | each "tutorial" use another thing in "context" |
| 09:01.58 | phearless | and never explain what does that mean |
| 09:02.20 | *** part/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
| 09:03.18 | RoyK | the context is the [context] in extensions.conf |
| 09:03.22 | RoyK | where the call is placed |
| 09:04.43 | phearless | okay i will have a look |
| 09:06.07 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 09:06.12 | SHad|Work | anyone here got a Sipura-SPA942 to work with asterisk? My inbound calls work fine, but outbound results in a SIP auth error. |
| 09:09.09 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 09:10.40 | *** join/#asterisk Sonderblade (n=meh@c-c358e353.131-1-64736c10.cust.bredbandsbolaget.se) |
| 09:12.46 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:16.06 | phearless | I can't create an : |
| 09:16.11 | phearless | exten => something |
| 09:16.20 | phearless | to call between internal phones |
| 09:16.23 | phearless | .... |
| 09:16.26 | phearless | this is awful |
| 09:16.29 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 09:16.57 | hads|home | ~thebook |
| 09:16.58 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:17.22 | Sonderblade | has anyone recorded their own set of voice prompts for asterisk? |
| 09:18.41 | phearless | i got the goddamn book |
| 09:19.50 | hads|home | If you have read it you should understand this. |
| 09:20.18 | *** join/#asterisk kannan (n=kannan@125.22.67.231) |
| 09:21.59 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 09:23.50 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
| 09:27.23 | RoyK | wtf?????? Aug 22 12:16:58 WARNING[7966]: channel.c:2559 ast_request: No channel type registered for 'PUf' |
| 09:28.15 | macTijn | heh |
| 09:28.19 | macTijn | typo ? :) |
| 09:28.45 | RoyK | grepping through /etc/asterisk doesn't find a single match for PUf |
| 09:29.16 | RoyK | this happens when I Queue(somequeue) and none of the SIP peers in the queue are connected |
| 09:29.49 | phearless | grep the source maybe |
| 09:30.13 | RoyK | done it. no matches |
| 09:30.16 | RoyK | :P |
| 09:30.21 | macTijn | RoyK: haha :> |
| 09:33.06 | *** join/#asterisk LoneShadow (n=duh@59.92.147.5) |
| 09:33.26 | LoneShadow | anyone using spa3k ? |
| 09:33.49 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 09:36.29 | backblue | morning |
| 09:37.47 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:39.00 | backblue | anyone here using realtime? even with cache enabled, i have nat problems, when i do a reload, can anyone give me a hand? |
| 09:39.49 | *** join/#asterisk apardo (n=apardo@87.217.146.232) |
| 09:40.11 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 09:40.42 | vlt | I have set up a queue in asterisk. When I dial its extension from connected SIP phones I hear moh (gsm file in native mode) and the youarenext voice. The same when I dial in from outside over a SIP peer I registered to. But when I use one of the SIP phones and dial out from asterisk over a SIP provider back to the same asterisk server I only hear ringing (though I can watch asterisk playing moh and voice in CLI). When a queue member answers they |
| 09:40.42 | vlt | <PROTECTED> |
| 09:40.47 | backblue | RoyK: i have done queue() without any peer connected, and have not that behaviour. |
| 09:41.48 | backblue | vlt: using some iax trunk or some card? |
| 09:42.41 | IOscanner | I am looking to buy DID numbers. What is the best priced vendor to purchase from? |
| 09:43.06 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 09:44.16 | backblue | IOscanner: you should be asking prices in the vendor's, not here. |
| 09:45.42 | vlt | backblue: All connections are SIP. |
| 09:46.23 | vlt | backblue: Exeption: The test from complete outside to the SIP provider asterisk is registered to. |
| 09:46.42 | IOscanner | I am just trying to find a good vendor to buy did's from when using asterisk. |
| 09:47.11 | *** join/#asterisk Muck- (n=Muck@145.253.170.162) |
| 09:47.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:48.47 | IOscanner | How about outbound termination? |
| 09:51.00 | RoyK | backblue: so have I, but I keep seenig this anyway.... |
| 09:56.55 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:57.31 | *** join/#asterisk jeffjohnson (n=Jeffjohn@unaffiliated/jeffjohnson) |
| 09:58.06 | IOscanner | Yes they do. Union Datacom |
| 09:58.53 | vlt | backblue: Even when I dial out over one SIP provider to another SIP provider and the call "comes back" to * I hear no moh and voice ... |
| 10:00.45 | vlt | backblue: It's possible that there's a bit of PSTN between the two SIP providers because it's an ordinary PSTN number I call ... |
| 10:01.05 | backblue | hum? |
| 10:01.09 | backblue | it's not a sip call? |
| 10:01.38 | backblue | like sip/${exten}@server ? or sip/server/${exten} ¿ |
| 10:01.46 | backblue | ? |
| 10:02.40 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 10:02.57 | vlt | backblue: Setup: SIP phone 1 --> asterisk 1 --> SIP provider 1 --> (mybe PSTN) --> SIP provider 2 --> astersik 1 --> queue() |
| 10:03.37 | vlt | backblue: SIP/${EXTEN}@server |
| 10:04.23 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 10:05.15 | vlt | backblue: To complete it: queue() --> queue member with SIP phone 2 -- but the missing moh and voice should be heard at position queue() ... |
| 10:06.05 | vlt | backblue: SIP phone 1 --> asterisk 1 --> queue() WORKS. |
| 10:06.25 | vlt | backblue: PSTN --> SIP provider 2 --> asterisk 1 --> queue() WORKS. |
| 10:07.46 | *** join/#asterisk hlpz (i=helpas@services.tvk.lt) |
| 10:07.51 | backblue | how can you say (mybe pstn) ? |
| 10:08.07 | backblue | if you make dial as i sayed, you can only have that maybe pstn, after asterisk2. |
| 10:09.57 | hlpz | hi there, does anybody know something about g729 codec available for OpenBSD? |
| 10:16.56 | backblue | do you will use asterisk with openbsd? :o |
| 10:24.10 | *** join/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
| 10:25.01 | sjobeck | hi all, how are things: any one help me with why I might be seeing this after upgrading * & zaptel : |
| 10:25.01 | sjobeck | [chan_zap.so] => (Zapata Telephony w/PRI) |
| 10:25.01 | sjobeck | <PROTECTED> |
| 10:25.03 | sjobeck | <PROTECTED> |
| 10:25.05 | sjobeck | <PROTECTED> |
| 10:25.07 | sjobeck | [root@phonesystem zaptel-1.2.7]# Ouch ... error while writing audio data: : Broken pipe |
| 10:25.57 | hads|home | <PROTECTED> |
| 10:27.03 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 10:27.07 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.178) |
| 10:27.19 | AsteriskAlbania | Aug 22 12:27:12 WARNING[4810]: chan_zap.c:8970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 10:27.20 | Jaxxan | hey guys |
| 10:27.27 | AsteriskAlbania | what wrong with it ? |
| 10:27.33 | AsteriskAlbania | what is CPE ? |
| 10:27.36 | AsteriskAlbania | Aug 22 12:27:12 WARNING[4810]: chan_zap.c:8970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 10:27.42 | Jaxxan | your zaptel.conf is incorrect |
| 10:27.43 | sjobeck | customer premise equipment |
| 10:27.54 | sjobeck | my zaptel ? |
| 10:28.08 | Jaxxan | are you trying to connect to a DMS100 ? |
| 10:28.25 | sjobeck | jaxan: who ? |
| 10:28.26 | hlpz | backblue: asterisk is running perfect on OpenBSD and g.729a would be greate as we have license. |
| 10:28.36 | AsteriskAlbania | Jaxxan: I am trying to connect to QUINTUM |
| 10:29.01 | Jaxxan | AsteriskAlbania: i encountered that the first time i tried to establish a pri to a dms100 |
| 10:29.27 | AsteriskAlbania | Jaxxan: what is the line to check at zaptel.conf |
| 10:29.42 | Jaxxan | AsteriskAlbania: sorry, i said zaptel.conf but i meant zapata.conf |
| 10:30.01 | Jaxxan | hop over to #flood |
| 10:30.44 | Jaxxan | hop over to flood and i'll paste what i use in mine |
| 10:31.27 | Jaxxan | AsteriskAlbania: /join #flood |
| 10:31.54 | Jaxxan | is that similar to what you have ? |
| 10:36.23 | Jaxxan | i'm not familiar with a QUINTUM, but basically, your error msg explains it to a T |
| 10:36.41 | sjobeck | any one seen this one before with a full PRI: |
| 10:36.41 | sjobeck | Unable to specify channel 25: Device or resource busy |
| 10:37.08 | Jaxxan | sjobeck: means that you're trying to use a channel that doesn't exist |
| 10:37.17 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 10:37.21 | Jaxxan | sjobeck: either your 2nd pri isn't configured properly or doesn't exist |
| 10:37.40 | sjobeck | jaxxna: hrm, didnt think I changed config at all, just an upgrade |
| 10:37.51 | Jaxxan | upgrades can do it to ya |
| 10:38.10 | Jaxxan | double check your zaptel.conf and zapata.conf with the samples config files |
| 10:38.56 | sjobeck | jaxxan: card says: |
| 10:38.56 | sjobeck | Channel Base 1-24 |
| 10:40.10 | Jaxxan | stop trying to use channel 25 then |
| 10:40.34 | Jaxxan | double check your zaptel.conf and zapata.conf |
| 10:40.47 | sjobeck | in etc? or etc/asterisk ? |
| 10:41.00 | sjobeck | or one in each, yes |
| 10:41.13 | Jaxxan | yup |
| 10:43.29 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 10:43.33 | muppetmaster | Hello all |
| 10:43.43 | muppetmaster | Any opinions on Fedora 5 and Asterisk SVN TRUNK? |
| 10:43.47 | muppetmaster | Good, bad, indifferent? |
| 10:43.58 | sjobeck | jaxxan: zaptel = |
| 10:43.58 | sjobeck | # Span 1: WPT1/0 "wanpipe1 card 0" RED |
| 10:43.58 | sjobeck | # ??: 1 WPT1/0/1 FXSKS |
| 10:43.58 | sjobeck | # ??: 2 WPT1/0/2 FXSKS |
| 10:43.59 | Jaxxan | commercial or private ? |
| 10:43.59 | sjobeck | # ??: 3 WPT1/0/3 FXSKS |
| 10:44.01 | sjobeck | # ??: 4 WPT1/0/4 FXSKS |
| 10:44.03 | sjobeck | # ??: 5 WPT1/0/5 FXOKS |
| 10:44.05 | sjobeck | # ??: 6 WPT1/0/6 FXOKS |
| 10:44.07 | sjobeck | # ??: 7 WPT1/0/7 FXOKS |
| 10:44.09 | sjobeck | # ??: 8 WPT1/0/8 FXOKS |
| 10:44.11 | sjobeck | # ??: 9 WPT1/0/9 FXSKS |
| 10:44.13 | sjobeck | # ??: 10 WPT1/0/10 FXSKS |
| 10:44.15 | sjobeck | # ??: 11 WPT1/0/11 FXSKS |
| 10:44.17 | sjobeck | # ??: 12 WPT1/0/12 FXSKS |
| 10:44.19 | sjobeck | # ??: 13 WPT1/0/13 |
| 10:44.21 | sjobeck | # ??: 14 WPT1/0/14 |
| 10:44.23 | sjobeck | # ??: 15 WPT1/0/15 |
| 10:44.25 | sjobeck | # ??: 16 WPT1/0/16 |
| 10:44.27 | sjobeck | # ??: 17 WPT1/0/17 |
| 10:44.29 | sjobeck | # ??: 18 WPT1/0/18 |
| 10:44.30 | muppetmaster | Hey, should that not go in a Pastebin???/ |
| 10:44.31 | sjobeck | # ??: 19 WPT1/0/19 |
| 10:44.33 | sjobeck | # ??: 20 WPT1/0/20 |
| 10:44.35 | sjobeck | # ??: 21 WPT1/0/21 |
| 10:44.37 | sjobeck | # ??: 22 WPT1/0/22 |
| 10:44.39 | sjobeck | # ??: 23 WPT1/0/23 |
| 10:44.41 | sjobeck | # ??: 24 WPT1/0/24 |
| 10:44.43 | sjobeck | <PROTECTED> |
| 10:44.45 | sjobeck | # Span 2: WCTDM/0 "Wildcard TDM400P REV I Board 1" |
| 10:44.47 | sjobeck | fxoks=25 |
| 10:44.49 | sjobeck | fxoks=26 |
| 10:44.51 | sjobeck | fxoks=27 |
| 10:44.52 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.178) |
| 10:44.53 | sjobeck | fxoks=28 |
| 10:44.55 | sjobeck | <PROTECTED> |
| 10:44.57 | sjobeck | # Global data |
| 10:44.59 | sjobeck | <PROTECTED> |
| 10:45.01 | sjobeck | loadzone = us |
| 10:45.01 | macTijn | ehm |
| 10:45.03 | sjobeck | defaultzone = us |
| 10:45.05 | sjobeck | <PROTECTED> |
| 10:45.07 | sjobeck | span=1,1,1,esf,b8zs |
| 10:45.07 | macTijn | ~paste |
| 10:45.11 | jbot | methinks paste is see http://paste.husk.org, or http://paste-it.net |
| 10:45.11 | sjobeck | bchan=1-23 |
| 10:45.12 | sjobeck | dchan=24 |
| 10:45.19 | sjobeck | all: sorry for large paste there, larger than I thought |
| 10:45.23 | sjobeck | i know |
| 10:45.25 | macTijn | ok |
| 10:45.28 | macTijn | use it ;) |
| 10:45.31 | sjobeck | jaxxan: see that last paste? mean anything to you? |
| 10:45.38 | muppetmaster | So, anyone on Fedora Core 5? |
| 10:45.55 | Jaxxan | dude, dont paste in this channel |
| 10:46.19 | Jaxxan | use pastebin |
| 10:46.28 | Jaxxan | muppetmaster: private or commercial ? |
| 10:47.16 | muppetmaster | Jaxxan Apologies, that paste messed me up. The app is for a private (although stability is needed) scenario where we need an internet PBX between three users at three sites and some call in IVR apps. |
| 10:47.37 | muppetmaster | Was thinking about Debian Sarge 2, but my hoster is having problems with making that available, so I have a limited number to choose from. |
| 10:48.10 | muppetmaster | OpenSuSE (no go because of their kernel policies), Mandrake, Debian (just Sarge for now), CentOS |
| 10:48.12 | Jaxxan | well, i stopped using fedora core at version 2. and sarge is kewl, but .... your choice |
| 10:48.23 | Jaxxan | for maximum stability though, dont install any gui crap |
| 10:48.28 | muppetmaster | I did not like FC through 3, but thought 5 might have gotten better |
| 10:48.35 | muppetmaster | Correct, headless only |
| 10:48.48 | muppetmaster | This runs at a server farm in Copenhagen, no screens, no need for a GUI |
| 10:49.08 | Jaxxan | i'm sure fedora core 5 will be fine for you |
| 10:49.18 | Jaxxan | it's a easy install |
| 10:49.20 | Jaxxan | that's for sure |
| 10:49.33 | Jaxxan | i've heard good things about it, alot of people use it in their servers |
| 10:49.44 | Jaxxan | i'm a RHEL fan these days though |
| 10:50.01 | muppetmaster | I have never been a RHEL fan, but only because I do not like the red fedora. |
| 10:50.01 | Jaxxan | corporate support and all )= |
| 10:50.04 | muppetmaster | Like the green lizard |
| 10:50.36 | Tebi_ | anyone using trixbox with digium TE110P card? |
| 10:50.45 | muppetmaster | I do like Ubuntu, but not available at my hoster |
| 10:50.51 | muppetmaster | Ubuntu Server would be nice, then I get my debs too |
| 10:51.28 | Jaxxan | is asterisk designed not to compile without make install ? |
| 10:51.47 | Jaxxan | zaptel and libpri compile fine with just make |
| 10:51.50 | Jaxxan | but asterisk fails |
| 10:52.10 | Jaxxan | i wanna make sure everythings working before i do this upgrade on my production box |
| 10:53.19 | *** part/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
| 10:54.30 | Jaxxan | you ever try to do a make instead of a make install ? |
| 10:54.55 | Jaxxan | trying to compile 1.2.10 |
| 10:56.30 | tzafrir | Jaxxan, 'make' builds everything |
| 10:56.45 | Jaxxan | weird then )= |
| 10:57.00 | tzafrir | a plain 'make' in an unconfigured trunk sadly exists with an error. run ./configure first |
| 10:57.06 | *** join/#asterisk RaYmAn-Bx (i=rayman@kbhn-vbrg-sr0-vl212-213-185-15-16.perspektivbredband.net) |
| 10:57.28 | Jaxxan | there's no configure in asterisk though |
| 10:57.45 | Jaxxan | least, not in a traditional location |
| 10:57.50 | tzafrir | (what I wrote reffered to trunk, and not to 1.2) |
| 10:58.05 | Jaxxan | oh (= |
| 10:58.45 | tzafrir | Jaxxan, anyway, pastebin the error you get |
| 11:01.43 | Jaxxan | http://pastebin.ca/144143 |
| 11:02.05 | Jaxxan | i'm thinking it's just cause i haven't done a make install on zaptel and libpri |
| 11:02.18 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 11:02.26 | Jaxxan | i'd love to be sure though |
| 11:05.52 | bionoid | benjk: I've changed to a single module, moved it around in all positions in all pci slots, and done the same operation with an identical card (I bought one production/one backup). No difference whatsoever. I'll try a clean build next, thanks for your input. |
| 11:12.41 | *** join/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net) |
| 11:13.27 | Dr-Linux|work | anybody is using iax2 trunk between 2 servers? |
| 11:13.28 | smackus2 | i have a syntax error and I am not seeing where... can you guys double check me... been working since 8am yesterday... now 5am today. eyes are shot... |
| 11:13.29 | smackus2 | exten => s,n(loop),Set(ERROR_COUNT=${IF($["${ERROR_COUNT}" = ""]?0:${ERROR_COUNT}$) |
| 11:17.30 | smackus2 | nevermind... bad copy paste exten => s,n(loop),Set(ERROR_COUNT=${IF($["${ERROR_COUNT}" = ""]?0:${ERROR_COUNT})}) |
| 11:19.04 | bionoid | smackus2: I don't speak this language (yet), but in other languages I'd terminate the if() before the ?0:.. :P |
| 11:19.32 | smackus2 | already found my answer... my copy paste cut off the last few char: |
| 11:19.33 | smackus2 | exten => s,n(loop),Set(ERROR_COUNT=${IF($["${ERROR_COUNT}" = ""]?0:${ERROR_COUNT})}) |
| 11:24.11 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 11:24.12 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:24.14 | jhiver | hi all |
| 11:24.33 | jhiver | has anybody managed to compile chan_h323 under debian sarge? |
| 11:24.50 | jhiver | I have compiled pwlib and openh323 |
| 11:25.36 | jhiver | and when i do 'make opt' in /usr/src/asterisk/channels/h323 i get ../../include/asterisk/strings.h:280: error: declaration of C function ` |
| 11:25.36 | jhiver | <PROTECTED> |
| 11:25.36 | jhiver | /usr/include/stdlib.h:188: error: previous declaration `long long int |
| 11:25.36 | jhiver | <PROTECTED> |
| 11:25.38 | jhiver | make: *** [ast_h323.o] Erreur 1 |
| 11:25.43 | jhiver | any ideas? |
| 11:25.54 | puzzled | hi |
| 11:26.51 | jhiver | hey |
| 11:27.01 | jhiver | chan seenms pretty dead... |
| 11:27.11 | jhiver | maybe it should be called chan_h323 hehe :) |
| 11:29.53 | tzafrir | jhiver, sure. |
| 11:30.36 | tzafrir | jhiver, trying to build h323 is a formedable task. mere mortals take openh323 from pkg-voip |
| 11:30.46 | jhiver | lol |
| 11:30.49 | tzafrir | http://pkg-voip.buildserver.net |
| 11:31.07 | jhiver | why should it be so hard |
| 11:31.19 | jhiver | it works out of the box with freebsd port |
| 11:31.23 | jhiver | it's crazy :) |
| 11:31.24 | tzafrir | I've rebuilt the packages from there on sarge nad it builds just fine, if you want to make sure |
| 11:32.10 | JunK-Y | jhiver: which * version? |
| 11:32.26 | jhiver | CVS head, but that's nothing to do with asterisk at this stage |
| 11:32.55 | tzafrir | CVS head of what? |
| 11:32.56 | *** join/#asterisk daurn|laptop (n=quae@unaffiliated/daurnimator) |
| 11:32.58 | daurn|laptop | hi all |
| 11:33.00 | jhiver | so tzafrir, what line should i add in the apt-sources.list to add this repo |
| 11:33.14 | JunK-Y | cvs-head is +1 years old. |
| 11:33.23 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 11:33.33 | jhiver | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 11:33.38 | jhiver | is the command i've used |
| 11:33.40 | daurn|laptop | what is a good, easy guide for setting up asterisk? |
| 11:33.58 | jhiver | TFOT is a good start |
| 11:34.04 | jhiver | google asterisk tfot |
| 11:34.09 | tzafrir | deb http://pkg-voip.buildserver.net/debian sarge main |
| 11:34.10 | JunK-Y | daurn|laptop: theres a book on oreilly, read that book. |
| 11:34.14 | jhiver | tzafrir, thx |
| 11:34.26 | JunK-Y | ~tfot |
| 11:34.32 | jbot | methinks tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 11:34.33 | JunK-Y | ~books |
| 11:34.38 | jhiver | also about chan_h323 |
| 11:34.52 | jhiver | do you know if there is an option similar to progressinband=yes? |
| 11:34.52 | QbY | If you had a choice, would you build Asterisk on a 2x XEON 3.06 with 2gb Memory, or on a 2x Opteron 250 with 2gb Mem? |
| 11:35.02 | jhiver | to tell it to forward early rtp? |
| 11:36.28 | jhiver | ok then I can just apt-get install asterisk-h323 ? |
| 11:36.32 | jhiver | sounds cool :) |
| 11:37.52 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:38.31 | jhiver | ok my version of asterisk is Asterisk SVN-trunk-r40796 |
| 11:38.33 | jhiver | mhhh |
| 11:38.42 | jhiver | maybe that's a little bit too bleeding edge :) |
| 11:41.29 | jhiver | I think i'll go fetch release instead :) |
| 11:48.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:49.44 | QbY | Queues/Agents question... I have agents that will be logged into multiple queues. Where do I set the priority of the queues? As in, queue #1 should be answered before queue #2 |
| 11:54.13 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:57.38 | Jaxxan | you set that in your queues.conf |
| 11:57.50 | Jaxxan | erm... wait |
| 11:57.58 | Jaxxan | you set a priority on the incoming calls |
| 12:01.39 | *** join/#asterisk oej (n=oej@63.117.53.60) |
| 12:03.12 | JunK-Y | QbY: see penalty stuff in queues.conf |
| 12:05.21 | jhiver | lads |
| 12:05.31 | jhiver | i don't really understand what WaitForRing() is for |
| 12:05.36 | jhiver | ne ideas? |
| 12:10.23 | ido | i believe WaitForRing(timeout) waits for the next ring that comes after 0 or timeout seconds, when the line is ringing. this (I THINK) lets you time events based on when the user hears a ring. |
| 12:10.49 | ido | this is done before answering the channel |
| 12:10.57 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:11.25 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 12:11.49 | ido | so for example WaitForRing(20) followed by Answer() would answer after the ring succeeding a 20 second wait. |
| 12:13.28 | bionoid | I'm having some very inconsistent noise issues with my TDM400P. I've tried using only one module, two different cards, in all possible slot combinations. No difference. Now I've rebuild asterisk and zaptel from source, and still no improvement. I've tried every audio tuning options that I can find documented, including disabling software interrupts in the zap source. I can make one call and not hear a single noise for ten minutes, then hangup, |
| 12:14.12 | bionoid | Also note that the exact same PC works perfectly well, without any interference, using a 100FXO card. |
| 12:15.13 | jhiver | oh great! |
| 12:15.16 | jhiver | <PROTECTED> |
| 12:15.16 | jhiver | ERROR: Could not open H.323 listener port on 1720 |
| 12:15.16 | jhiver | Jul 18 13:33:12 ERROR[14280]: chan_h323.c:2367 load_module: Unable to create H323 listener. |
| 12:15.16 | jhiver | Jul 18 13:33:12 WARNING[14280]: loader.c:414 __load_resource: chan_h323.so: load_module failed, returning -1 |
| 12:15.16 | jhiver | <PROTECTED> |
| 12:15.22 | jhiver | any ideas what's going on? |
| 12:15.48 | SaTLaN32 | hi guys |
| 12:15.54 | SaTLaN32 | need help with this var: |
| 12:15.54 | ido | jhiver: is the port in use? |
| 12:15.55 | SaTLaN32 | ${CALLERID(rdnis)} |
| 12:15.58 | bionoid | jhiver: Is port 1720 already taken? Check with netstat -an|grep 1720 |
| 12:16.17 | jhiver | nope |
| 12:16.26 | jhiver | <PROTECTED> |
| 12:16.26 | jhiver | mantis:/etc/asterisk# |
| 12:16.31 | jhiver | nothing there |
| 12:16.41 | SaTLaN32 | how do i send it to a Audiocodes M2K (SIP to E1 gw) so it will be in the Redirected ID field? |
| 12:16.45 | bionoid | Any further information in your logs, perhaps? |
| 12:16.51 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:16.56 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 12:17.05 | jhiver | bionoid, which ones? |
| 12:17.14 | bionoid | asterisk logs, presumably |
| 12:17.39 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 12:17.47 | jhiver | well in /var/log/asterisk there is just cdr-csv cdr-custom event_log queue_log |
| 12:18.00 | *** join/#asterisk Irulka (n=irina@213-35-236-25-dsl.end.estpak.ee) |
| 12:18.06 | jhiver | and in these logs there is nothing relevant |
| 12:18.07 | caio1982 | coppice: hi steve, are you up for some faxing issues using your t38 code? :) |
| 12:18.46 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:18.46 | coppice | OK |
| 12:20.11 | *** join/#asterisk lukketto (n=lukketto@host99-159.pool876.interbusiness.it) |
| 12:20.51 | Jaxxan | how do i restart a PRI from the console ? |
| 12:21.14 | Jaxxan | say i want to restart span3 on a t400p |
| 12:21.25 | Jaxxan | without resetting the whole damn card |
| 12:21.35 | caio1982 | coppice: i'm adapting your 1.2.7 patch to 1.2.10 (http://caio.ueberalles.net/asterisk_1.2.10_t38_20060817_chansip.txt) but it seems that i did something wrong editing it and since i'm not a real C programmer, it might be the cause for it: http://caio.ueberalles.net/chan_sip_t38.txt any hints about syntax errors? |
| 12:22.12 | jhiver | ok i found the error |
| 12:22.17 | jhiver | the bindaddress=was wrong |
| 12:22.21 | jhiver | great :) |
| 12:22.45 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 12:23.29 | *** join/#asterisk Tili (n=tili@202.133.67.152) |
| 12:23.36 | caio1982 | coppice: the reason i'm editing it right now is because there was some changes between 1.2.7 and 1.2.10 and it doesn't apply ok since then |
| 12:24.19 | coppice | caio1982: no idea. the code went into SVN, for what its worth, and people have abandoned the patches since then |
| 12:24.59 | Tili | how can i get the src_ip from a sip channel in extensions.conf |
| 12:25.02 | coppice | what went into SVN will need heavy mods to be useful, though. the real world doesn't work like the specs that code was developed against :-( |
| 12:25.10 | Tili | i believe there is some variable for that. but i dont know what it is |
| 12:26.24 | caio1982 | coppice: uh that's bad... well, i'm gonna try a little bit more to work on this .diff, just to see what i can get from it in 1.2 |
| 12:26.30 | caio1982 | coppice: thanks one more time steve |
| 12:28.18 | muppetmaster | When compiling SVN TRUNK, what else does one need for the jabber stuff besides iksemel and iksemel-devel? |
| 12:28.43 | coppice | caio1982: if you want to see a working solution, you can look at the openpbx SVN. that has been modded to work with the real world, and supports T.38 termination, as well as passthrough |
| 12:28.47 | vlt | Does anyone know how to suppress the CallerID in outgoing SIP calls? I tried SetCallerID() but it doesn't work. |
| 12:29.27 | caio1982 | coppice: i did a checkout from their code to analyse, yep |
| 12:29.44 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 12:30.41 | coppice | the UDP handling is separated from RTP and UDPTL. they plug into the UDP code dynamically as needed. that way they can share a single UDP port, and live with the real world |
| 12:31.25 | muppetmaster | I am currently getting this error when I try to compile SVN TRUNK on FC5: http://pastebin.ca/144277 |
| 12:33.05 | *** join/#asterisk prog (n=vdsoft@vdsoft.kh-net.cz) |
| 12:33.12 | prog | hello asteriskgeeks |
| 12:33.33 | prog | im hope some of you is alive ;-) |
| 12:33.46 | zoa | nopez |
| 12:33.50 | zoa | everybody just died |
| 12:33.51 | zoa | sorry |
| 12:33.53 | zoa | time to move on |
| 12:34.09 | zoa | *** this is an automated message *** |
| 12:34.15 | zoa | so whats the problem ? :) |
| 12:34.29 | mitcheloc | zoa: be nice ;) |
| 12:34.46 | zoa | i am nice |
| 12:34.57 | zoa | i just asked what his problem was |
| 12:35.12 | zoa | i kind of suppose he has a problem with asterisk :) |
| 12:35.25 | zoa | or maybe he was interested in you mitcheloc |
| 12:35.33 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 12:35.55 | mitcheloc | eek, no! /me rolls over and plays dead |
| 12:36.12 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 12:38.07 | prog | had i phone call ... sorry guys :-) .... well my question is ( on next row ) |
| 12:39.48 | prog | could you (please) advise me, where to look to make possible this situation: telefon user( with analog phone) want to leave and need to switch all calls to another pho mechine ? it is usually done with for example *80[number] ... |
| 12:40.32 | *** join/#asterisk bXi (i=bluepunk@irssi.co.uk) |
| 12:40.37 | prog | thank you in advance for your hints ... i do not require all the steps but one url link is appreciated :-) thank you |
| 12:40.45 | bXi | hi |
| 12:40.50 | prog | hi bxi |
| 12:40.56 | zoa | google for ASTDB |
| 12:41.08 | bXi | can i put stuff like sip configuration in a different file ? |
| 12:41.10 | zoa | or follow me |
| 12:41.16 | zoa | bxi : yes |
| 12:41.17 | prog | ok ok zoa, thx ... what this abreviation means ? |
| 12:41.22 | zoa | its standard a different phone |
| 12:41.25 | zoa | asterisk database |
| 12:41.41 | zoa | you could set a value with forward = 1 |
| 12:41.47 | zoa | if he dials a certain extension |
| 12:41.50 | prog | zoa, aaah, yes yes |
| 12:41.50 | zoa | and when a customer calls |
| 12:41.53 | zoa | check it first |
| 12:41.56 | zoa | if its yes dial A |
| 12:42.00 | zoa | if its no, dial B |
| 12:42.15 | Aurs | or if it gets anything, dial its value |
| 12:42.15 | prog | hmmm ... it sounds intelligently ....;-) great great |
| 12:42.17 | bXi | to be more specific |
| 12:42.43 | prog | zoa ... is it done in extensions.conf, isn`t it ? |
| 12:42.43 | bXi | i want to make a sip_$user.conf which contains the sip entry and the exten entry |
| 12:42.47 | bXi | is this possible? |
| 12:42.49 | zoa | yes |
| 12:42.53 | zoa | extensions.conf |
| 12:43.01 | prog | zoa , thank you very much !!!!! |
| 12:43.06 | prog | great! |
| 12:43.30 | prog | ufff, how simple is asterisk configuration with such great people ;) |
| 12:44.22 | prog | bXi, you mean something like include ? |
| 12:44.29 | prog | bXi, include="some file" ? |
| 12:44.51 | zoa | bxi, i think it can include config files for users |
| 12:45.04 | zoa | with include in the main sip.conf file |
| 12:45.19 | bXi | i have something like that already |
| 12:45.30 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:45.36 | bXi | but i'd like to include the exten for each user in the file |
| 12:45.38 | prog | and doesn`t work ? |
| 12:45.44 | bXi | so its not spread accross files |
| 12:47.03 | e-ddie | do any of you guys have any experience with encoding videos/images to .h263? |
| 12:47.35 | jeffjohnson | no |
| 12:47.51 | prog | sorry e-ddie, no |
| 12:48.04 | Irulka | hi to all! can you help me... I want to make attended call transfer using Grandstream BT102. I get call from another terminal(for example 111), talk with person on this terminal, then press 'flash' and dial new extension, where i want the received call to be transfered(222). Now i can talk with 222 and tell that i am going to transfer the call to him. After that i press 'transfer' and get 111 and 222 connected with each other. Transfer i |
| 12:48.04 | Irulka | s successful and i can hang up. The problem is, that i am not able to do the same if i get not the direct call from another extension, but a call from the queue (agent is logged in on BT102 and it is his turn to answer the call). In this case, when i press 'transfer' BT102 is hanged up and 111 and 222 are not connected with each other.... do you know what can be the reason of that?... |
| 12:48.09 | *** join/#asterisk cbrake_ (n=cbrake@oh-69-34-21-229.sta.embarqhsd.net) |
| 12:49.30 | *** join/#asterisk I-MOD (n=opticron@gateway.digium.com) |
| 12:51.35 | zoa | bxi i dont really get what you ask for :/ |
| 12:52.48 | *** join/#asterisk saftsack (n=saftsack@p54A7D54A.dip.t-dialin.net) |
| 12:56.38 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 12:58.25 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 12:59.09 | *** join/#asterisk roving_prole (n=Harper@72-254-127-33.client.stsn.net) |
| 12:59.35 | hi365 | Hello! anyone know about Sangoma a200? after installing drivers the computer doesnt c the card :-( |
| 12:59.36 | hi365 | [root@asterisk1 ~]# wanrouter hwprobe |
| 12:59.37 | hi365 | ------------------------------- |
| 12:59.37 | hi365 | | Wanpipe Hardware Probe Info | |
| 12:59.37 | hi365 | ------------------------------- |
| 12:59.37 | hi365 | Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=0 A108=0 |
| 13:02.19 | bXi | zoa: |
| 13:02.30 | bXi | normally you'd define an extension in extension.conf |
| 13:02.35 | bXi | and a sip entry in sip.conf |
| 13:02.47 | bXi | lets say i have an user called linksys |
| 13:03.05 | bXi | i want to define the extension AND sip entry in linksys.conf |
| 13:04.44 | kmilitzer | Hi everyone ... sorry to be off topic, but does anyone know if an IRC channel for SER or OpenSER exists? |
| 13:07.01 | *** join/#asterisk kagato (n=kagato@souja.net) |
| 13:07.27 | kagato | Anyone up to answering an echo canceling question? |
| 13:08.37 | kagato | I have one of the TE4xxP cards with hardware echo cancelling. Do I need to set echocancel=yes to enable echo cancelling or is that only for software echo cancelling? |
| 13:08.56 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.254) |
| 13:09.12 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:09.42 | *** join/#asterisk meppl (i=mephisto@meppl.net) |
| 13:13.13 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 13:13.41 | champster | Is anyone using overhead paging connected to a analog port? |
| 13:13.57 | champster | If so, how are you selecting the zones? |
| 13:14.28 | champster | I could make the user do it, but I would prefer to give the 3 buttons. (Office, Shop, All) |
| 13:16.36 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 13:19.57 | hi365 | Hello! anyone know about Sangoma a200? after installing drivers the computer doesnt c the card :-( |
| 13:19.57 | hi365 | [root@asterisk1 ~]# wanrouter hwprobe |
| 13:19.57 | hi365 | ------------------------------- |
| 13:19.57 | hi365 | | Wanpipe Hardware Probe Info | |
| 13:19.57 | hi365 | ------------------------------- |
| 13:19.58 | hi365 | Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=0 A108=0 |
| 13:20.07 | [TK]D-Fender | hi365: Stop spamming |
| 13:20.33 | [TK]D-Fender | hi365: and for crying out loud use PASTBIN. And you MIGHT want to actually consider showing your CARD config while you're at it. |
| 13:20.50 | [TK]D-Fender | kagato : Yes |
| 13:21.26 | hi365 | [TK]D-Fender: cant config a card if its not "there" |
| 13:21.35 | coppice | hi365: did you move the card? the default install options tie the config to a particular PCI slot |
| 13:21.54 | [TK]D-Fender | champster: Well each analog port is its own zone when you think about it so you'd need to pull them all into a conference to do a "page all". Some people use .call files with a Meetme conference for that. |
| 13:22.30 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 13:23.35 | hi365 | coppice: no. fresh install. will re-installinhg the drivers re"tie" it? |
| 13:24.22 | [TK]D-Fender | hi365: Pastebin your card config |
| 13:24.42 | cbrake | I am trying to get asterisk to dial and automatically enter calling card account info. |
| 13:25.22 | cbrake | first I dial the 1800 #. How do I then enter the call card account # and the # I am calling after the calling card service answers? |
| 13:25.23 | [TK]D-Fender | hi365: Hrm... yeah I'd try re-seating the card and then trying another slot... |
| 13:25.23 | hi365 | [TK]D-Fender: i didnt config the card yet! do u want the zaptel+zapata? |
| 13:25.27 | coppice | hi365: its not the install of the drivers that ties things. its the generation of the config file. the config files usually say the exact slot the card is expected to be in |
| 13:25.33 | [TK]D-Fender | hi365: see the point on it just not showing up... |
| 13:25.42 | zoa | bXi: aaah, i dont think that is possible |
| 13:26.24 | [TK]D-Fender | hi365: Skip the config I see your point on it not being ID'd anywhere. LSPCI & DMESG show nothing as well? |
| 13:27.35 | hi365 | [TK]D-Fender: dmesg in a min... |
| 13:29.13 | hi365 | [TK]D-Fender: http://pastebin.ca/144318 |
| 13:29.57 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:30.00 | hi365 | [TK]D-Fender: http://pastebin.ca/144319 |
| 13:31.11 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 13:32.16 | FarrisG | If I know what I've been doing, and have managed an asterisk box for a couple of years, is there any substantial reason for/against using trixbox? |
| 13:32.28 | [TK]D-Fender | hi365: Modprobe wanpipe |
| 13:32.46 | Unistim_junky | Is anyone using Transfer() to send call to a remote box? |
| 13:33.17 | [TK]D-Fender | FarrisG: Only rason to use it is to drop it in a place where they will be 100% with the functionality out of the box and you feel lucky enough that their scripts won't break on them |
| 13:33.20 | hi365 | [TK]D-Fender: FATAL: Error inserting wanpipe (/lib/modules/2.6.9-34.0.2.ELsmp/kernel/drivers/net/wan/wanpipe.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 13:33.52 | [TK]D-Fender | FarrisG: Against..... well if you wanted a canned PBX just go out and buy one. Anything goes wrong with Trixbox you know what you're in for... noone wants to think about it. |
| 13:34.00 | zeedo | FarrisG: depends what you want to do with it, TrixBox becomes a problem when you start to get complicated/large with the setup |
| 13:34.13 | zeedo | FarrisG: but for most smallish deployments it's simple and effective |
| 13:34.28 | [TK]D-Fender | Sounds like something didn't compile right and those dmesg errors.... I'd say rebuild your drivers and make sure to point them everywhere they need to. |
| 13:34.50 | FarrisG | zeedo: It's for an office of about 100. |
| 13:34.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:35.08 | zeedo | FarrisG: thats at about the limit of what Id want to run TriixBox at |
| 13:35.10 | hi365 | [TK]D-Fender: did twice. will have another go |
| 13:35.14 | [TK]D-Fender | FarrisG: Actually the only place it becomes validated is in certain LARGER scenarios where * becomes a more sizable burdon and their staff is somewhat "challenged". |
| 13:35.21 | zeedo | FarrisG: if you're happy with managing Asterisk yourself Id stick with it |
| 13:35.36 | FarrisG | zeedo: What's the logic? What makes the limit on trixbox any lower than * proper? |
| 13:35.53 | FarrisG | Got it |
| 13:36.09 | zeedo | FarrisG: the fact that it can only run on one box, the minute you go beyond that you lose most of the benefit of TrixBox |
| 13:36.22 | phearless | I'm back |
| 13:36.25 | phearless | hello |
| 13:36.30 | phearless | exten => 200,1,Dial(sip/200) |
| 13:36.37 | phearless | I manage to make a call thanks to this |
| 13:36.54 | phearless | but it should be better if I had the same for all the extensions, no ? |
| 13:37.00 | phearless | is it possible ? |
| 13:37.15 | Unistim_junky | folks, can you please check out http://pastebin.com/773205 an tell me where I screwed up on transfer |
| 13:37.32 | Irulka | can anyone help me with making attended call transfer on Grandstream BT102, please? |
| 13:37.44 | inspired | phearless, exten => _X.,1,Dial(SIP/${EXTEN}) |
| 13:37.44 | muppetmaster | Under SVN TRUNK, how should one go about adding mysql support from Asterisk addons? |
| 13:37.59 | FarrisG | So then I guess my next question is: My current asterisk box is very old. I'm installing a new one, with latest versions of everything. I've never used any of the web interfaces, but with the office growing, I'd like some sort of web-admin functionality. Is there any reliable method to this, or is it still true that if you want any semblance of stability, you should stick with managing via console? |
| 13:38.08 | phearless | ok inspired ! I will try to decrypt your line |
| 13:38.10 | inspired | phearless, if you then dial 59129 and have a peer named 59129, it will call it |
| 13:38.28 | phearless | what is _X. ? |
| 13:38.44 | phearless | X is any number |
| 13:38.54 | phearless | but _ and . I do not know |
| 13:38.58 | inspired | phearless, _ specifies that this exten is a pattern. the X matches anything, and "." means that the exten can have a variable length |
| 13:39.01 | zeedo | FarrisG: the console is more about flexibility than stability. Adding a web interface doesnt destabilise but you do have to follow the configuration conventions of the interface |
| 13:39.17 | phearless | thanks inspired |
| 13:39.18 | champster | Is anyone using overhead paging connected to a analog port? |
| 13:39.20 | champster | If so, how are you selecting the zones? |
| 13:39.21 | champster | I could make the user do it, but I would prefer to give the 3 buttons. (Office, Shop, All) |
| 13:39.45 | inspired | phearless, and since we have an X there, it must be minimum 1 digit long |
| 13:39.59 | phearless | okay |
| 13:42.05 | [TK]D-Fender | FarrisG: Every GUI out there takes COMPLETE control of your * config so there is no "half-way" |
| 13:42.17 | [TK]D-Fender | FarrisG: Without writing it yourself.... |
| 13:42.28 | phearless | it works |
| 13:42.35 | phearless | big thanks toooooo.......... |
| 13:42.37 | [TK]D-Fender | champster: I already answered you. |
| 13:42.38 | phearless | inspired ! |
| 13:43.09 | [TK]D-Fender | phearless: That is VERY unhealthy... |
| 13:43.11 | FarrisG | [TK]D-Fender: Meaning, if you're using a GUI (FreePBX, or whatever), you'll screw stuff up but good if you go underneath it and mess with your conf files? |
| 13:43.49 | [TK]D-Fender | FarrisG: Maybe yes, maybe no. The thing is if you go into the conf files and change anything, the moment they "commit" a change from the GUI all your work becomes ERASED. |
| 13:44.03 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 13:44.12 | [TK]D-Fender | FarrisG: All of those GUI's rebuild EVERYTHING from scratch.... |
| 13:44.28 | [TK]D-Fender | FarrisG: and if you don't like the "standard" way it dials a phone well... tought luck. |
| 13:44.33 | champster | Sorry, Missed it. |
| 13:44.46 | FarrisG | [TK]D-Fender: That's gross. There's not some standardized way to fry the confs based on a DB instead of using baked? |
| 13:45.18 | [TK]D-Fender | champster: Well each analog port is its own zone when you think about it so you'd need to pull them all into a conference to do a "page all". Some people use .call files with a Meetme conference for that. |
| 13:45.21 | champster | The analog port connects to a Valcom V2003A which is a 3 zone pageing unit with one port. |
| 13:45.22 | inspired | phearless, np |
| 13:45.49 | champster | It needs a DTMF tone to select the zone. |
| 13:45.52 | [TK]D-Fender | champster: Oh the unit has 3 zones and 1 analog port? Odds are you enter the zones by DTMF after it answers the ringing port. |
| 13:46.22 | [TK]D-Fender | champster: Time to crack open that manual.... |
| 13:46.35 | champster | I have tested with dial the extension, then manually enter the port and it works OK. |
| 13:47.05 | FarrisG | To be honest, I really don't NEED a gui to manage the box. There are a few pieces I'd like to be able to delegate control via some abstraction, though. For instance, it would be really nice to give ownership of a meetme line to someone, and somehow allow them to change the password, etc. |
| 13:47.09 | champster | If I use the D option of dial, it doe not give me the ability to talk for a few seconds. |
| 13:47.51 | champster | If I use the M option and have a macro with senddtmf, it takes even longer before I get the ability to speak back. |
| 13:47.54 | *** join/#asterisk daniel_bergamini (n=daniel_b@70-41-166-149.cust.wildblue.net) |
| 13:47.56 | file | champster: well you have to do give it time to send the DTMF, have it played back, plus setup the audio stream |
| 13:48.10 | FarrisG | Also, is there any non-flakey way to (a) manage conversation recordings (b) give an operator some sort of software switchboard panel, so she can tell who's on the phone and who isn't? |
| 13:48.54 | champster | I should have shopped arround and bought 2 single zone units. |
| 13:50.16 | [TK]D-Fender | FarrisG: you can make your own "phones in use" panel rather easy. You could give the receptionist a phone with good presence support (like a Polycom IP 601 + multiple attendant modules), there are several other programs like FOP that a "general" and "passive" tools to see the running state of things |
| 13:50.42 | [TK]D-Fender | FarrisG: As for managing recordings, depends how they are named, and created. |
| 13:51.35 | [TK]D-Fender | champster: If you just call the unit direct, how long do you have to wait before entering the zone in DTMF? After that how long a wait until you can start talking? |
| 13:52.06 | file | [TK]D-Fender: ! ! ! |
| 13:52.11 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 13:52.15 | *** join/#asterisk miller7 (n=999@213.5.88.49) |
| 13:52.17 | [TK]D-Fender | file: I don't want to meet your mom! |
| 13:52.24 | file | :D |
| 13:52.25 | FarrisG | [TK]D-Fender: One of my receptionists has a Polycop 501. Will that do what I want? The other has a Grandstream GXP2000. A decent little cheap phone. Do you know if the 2000 (and/or its optional sidecar) will support presence? |
| 13:52.43 | FarrisG | s/Polycop/Polycom |
| 13:52.54 | champster | If I do it manually, 1 second or less for the unit to beep, I enter the zone digit and can talk immediately. |
| 13:52.57 | [TK]D-Fender | FarrisG: the 501 can only watch 7 other devices, which I suspect is too little. |
| 13:53.13 | [TK]D-Fender | FarrisG: The 601 can watch up to 48 w/ attendant modules. |
| 13:53.35 | [TK]D-Fender | FarrisG: Mine has 2 of them full loaded and lit up like a christmas tree :D |
| 13:54.07 | champster | I think the issue is that the ammount of time it takes to bridge is normally done durring the ringing, or immed. after. |
| 13:54.08 | FarrisG | [TK]D-Fender: 7 might be enough actually. |
| 13:54.29 | FarrisG | [TK]D-Fender: She only really needs to see management's presence |
| 13:54.48 | champster | Using the D or M option, the caller knows that the call is connected and has to wait for the bridge, with no signalling to let them know it is done. |
| 13:55.07 | hi365 | [TK]D-Fender: same problem (cant c card) im trying to remove the wanrouter and reinstall. do u know what need to be deleted ?(besides /etc/wanrpipe) |
| 13:56.11 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 13:56.39 | [TK]D-Fender | FarrisG: Well you could view them through the buddies screen. Not the friendliest, but it'd work. |
| 13:57.56 | FarrisG | [TK]D-Fender: I didn't realize there was any other way. To see the 7 you mention on the 501, does that require an extension/sidecar? |
| 13:58.48 | [TK]D-Fender | FarrisG: only the 601 supports a sidecar. No, to view them you'd use the "buddies" soft-key on the 501 which would give you a scrollable list of those your phoen has been configured to watch. |
| 13:59.05 | FarrisG | [TK]D-Fender: Ah, I see now. Thanks |
| 13:59.40 | [TK]D-Fender | FarrisG: np. But seriously... get her a 601 and at least 1 module :) Good for speed-dials, etc.... |
| 13:59.58 | *** part/#asterisk miller7 (n=999@213.5.88.49) |
| 14:00.58 | JT | how do you set asterisk up to show line/extension usage on voip phones anyway? |
| 14:01.15 | champster | the hint priority |
| 14:01.15 | FarrisG | [TK]D-Fender: Not gonna happen soon. :) |
| 14:02.14 | JT | champster: ? |
| 14:02.22 | FarrisG | [TK]D-Fender: Geez, that thing looks like kind of a waste of space. Seems like they could have put a lot more lines on one |
| 14:03.06 | Unistim_junky | Folks I have helped lots of newbies in the past now I really need your help. Please view http://pastebin.com/773235 . Free Beer (STRONGBOW) to whoever finds it. |
| 14:04.44 | Unistim_junky | drumroll ##### |
| 14:05.34 | [TK]D-Fender | FarrisG: If it was only a tiny light, yes, but you get a nice full line to write the name meaningfully. |
| 14:06.19 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:07.02 | [TK]D-Fender | JT: Look up "asterisk presence" on the WIKI |
| 14:07.44 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 14:08.49 | MrChimpy | unistim: strongbow isn't beer, it's cider |
| 14:09.21 | Unistim_junky | What it is is the wife's |
| 14:09.34 | [TK]D-Fender | Unistim_junky: Something tells me you shoudl be providing SIP debug info if you are expecting any kind of help.... |
| 14:09.38 | MrChimpy | you sure you can risk giving it away then :) |
| 14:10.22 | JT | [TK]D-Fender: ok |
| 14:11.29 | hi365 | [TK]D-Fender: Card Cnt: S508=0 S514X=0 S518=0 A101-2=0 A104=0 A300=0 A200=1 A108=0 |
| 14:11.33 | hi365 | Thanks! |
| 14:13.19 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 14:14.51 | *** join/#asterisk xorotude (n=labsoard@mail.xorotude.com) |
| 14:15.09 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 14:16.11 | xorotude | impressive user list... |
| 14:16.37 | [TK]D-Fender | hi365: Cool... get cracking on it now.. |
| 14:16.59 | *** join/#asterisk sixsens (n=Ident@dsl235-63.netsys.am) |
| 14:24.12 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 14:24.30 | jtexter3 | Has anyone had experience setting up a Grandstream GXP-2000 with sidecard? |
| 14:25.07 | *** join/#asterisk xorotude (n=labsoard@mail.xorotude.com) |
| 14:25.10 | jtexter3 | I have the config generated and placed in my /tftpboot directory. When I tell the phone to use it, it says TFTP provisioning, and shows the file downloading, then it reboots and resets to factory defaults |
| 14:25.14 | jtexter3 | very annoying |
| 14:25.55 | Irulka | can anyone help me with making attended call transfer on Grandstream BT102, please? |
| 14:26.04 | *** join/#asterisk coRnholi0 (n=vircuser@62.96.103.66) |
| 14:26.31 | hi365 | jtexter3: not with the sidecard maybe this will help: http://www.voip-info.org/wiki/view/Grandstream |
| 14:26.37 | FlatFoot | anyone used one of these ( any success ) ? http://svp.co.uk/products-solo.php?pid=1487 |
| 14:26.45 | *** join/#asterisk hatamen (n=hatamen@222.183.36.54) |
| 14:27.07 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 14:27.17 | hatamen | :) |
| 14:27.19 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 14:27.24 | hi365 | jtexter3: http://www.voip-info.org/wiki/view/GXP-2000+Extension+Unit |
| 14:28.41 | hatamen | hi,all. Do you have used a MSN asterisk Group? hehe :) |
| 14:29.05 | [TK]D-Fender | OMG, a GrandSUCK expansion module! |
| 14:29.09 | Dr-Linux|work | question, can i forward the call to two numbers on cisco 7940/60? |
| 14:29.30 | [TK]D-Fender | Irulka: I'd read its manual again if I were you.... |
| 14:29.44 | vlt | Question: How can I stop aserisk if it's running in safe-mode? `/etc/init.d/asterisk stop`or `asterisk -rx stop gracefully` don't work ... |
| 14:29.49 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 14:29.57 | vlt | ... from a script ... |
| 14:30.08 | [TK]D-Fender | Dr-Linux|work: Well you can forward the call to an exten on * that CALLS 2 numbers if you want... |
| 14:30.24 | hatamen | Add asterisk-msn@hotmail.com to your msn friend list, and "nudge" to this friend , then you will see us! it's in testing... Cool! |
| 14:30.32 | Aurs | vlt: tried with sudo? |
| 14:30.36 | [TK]D-Fender | vlt: Gracefully waits till there are no more open channels. |
| 14:31.03 | pnlarsson | Q: I'm using AMI to initate a call, first calling the agent and when he picks up, the call is placed to the customer. The prob is if the user rejects the call, the call is still placed to the customer... |
| 14:31.08 | Dr-Linux|work | [TK]D-Fender, yes sir, i know that, but from the phone am looking for this option |
| 14:31.09 | jtexter3 | yeah, that's got the basics. I've downloaded the 1.1.1.19 firmware, which I can tell the phone is now talking with the sidecar. But I still can't provision over TFTP |
| 14:31.11 | Aurs | /etc/init.d/asterisk stop should stop it even if there are open chans |
| 14:31.46 | [TK]D-Fender | Dr-Linux|work: No, you cannot forward to 2 things. Only 1. * can then take that request and CHOOSE to ring 2 things, but thats not a SIP spec.... |
| 14:32.24 | jtexter3 | D-Fender: Grandstream seem to get mixed reviews, so I bought one to check out for myself. One thing I like is that the sidecar has 56 buttons. |
| 14:32.36 | jtexter3 | I've had good luck with Polycom, but with 3 side cars, you can only have 42 extensions |
| 14:32.40 | Qb3rt | where can i remove the reminder option that makes the phone ring for 2 seconds when i have a voicemail waiting? this thing ring the phone every 15 minutes! |
| 14:32.44 | Irulka | [TK] D-Fender: the problem is that i can do transfer, if i get direct call from another terminal, but i can't do that if the call came from the queue... |
| 14:32.51 | hatamen | ha ha |
| 14:32.52 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 14:32.55 | [TK]D-Fender | jtexter3: Yeah sure it has lots of lights, but its still junk... |
| 14:33.01 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 14:33.19 | [TK]D-Fender | jtexter3: if you need more than the 48 you can get on an IP 601 well... eek... |
| 14:33.47 | Dr-Linux|work | [TK]D-Fender, Thanks sir, i'd like to ask another question as well, >> cisco 7960 allows me only 3 calls at same time, can i increase calls? |
| 14:33.50 | [TK]D-Fender | Irulka: What ver of * are you on? |
| 14:34.10 | [TK]D-Fender | Dr-Linux|work: No idea.... don't do Cisco, you know that... |
| 14:34.18 | pnlarsson | And i can't find a var the is telling me if the first leg is still up... |
| 14:34.52 | [TK]D-Fender | Irulka: There is an issue with using transfers to apss off queue calls not freeing up the agent for more Queue calls. I *think* they helped correct this in recent versions.... |
| 14:35.07 | Dr-Linux|work | [TK]D-Fender, it's okey, actually myself i never seen cisco phones, but we have a bunch of cisco phones in US, so all i can do is google or ask here .. anyway thanks for your help |
| 14:35.58 | [TK]D-Fender | Dr-Linux|work: Oh well, sorry. |
| 14:36.20 | Irulka | [TK]D-Fender, Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r |
| 14:36.37 | [TK]D-Fender | Irulka: Well it may still be an outstanding issue then... |
| 14:36.44 | [TK]D-Fender | Irulka: But a known one. |
| 14:38.42 | jbalcomb | Whats the best conference room phone around? |
| 14:38.43 | *** join/#asterisk operat0r (n=h0msar@adsl-152-157-190.asm.bellsouth.net) |
| 14:38.45 | operat0r | Hello |
| 14:39.14 | operat0r | anybody ever do anything similar to this ? http://0pencircuit.net/t0c/index.php?topic=171 |
| 14:39.48 | operat0r | I figured with a uber asterisk script I can make free phone calls |
| 14:39.51 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:40.25 | Irulka | [TK]D-Fender, ok, thank you |
| 14:40.55 | backblue | Irulka: why dont you use misdn? |
| 14:41.15 | [TK]D-Fender | jbalcomb: I'd be tempted to say a high-end Polycom... there is such a crazy range of them though. How big a room? How many people? Need ex-mics? Mobility? |
| 14:41.39 | hi365 | does anyone know what steps need to ba taken to install the Snagoma Remora card OTHER than updating the zaptel+zapata files? |
| 14:42.02 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 14:42.23 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 14:42.27 | champster | <PROTECTED> |
| 14:42.44 | tzanger | wtf is a remora card |
| 14:42.45 | champster | The person with the ad pays just like a toll free. |
| 14:42.57 | coppice | remora is another name for the A200 |
| 14:43.14 | [TK]D-Fender | hi365: wancfg and setup your wanpipe1.conf file for it, add to startup, ztcfg -vvv it and you should eb good to go. |
| 14:43.16 | benjk | I think the name refers to the stackable nature of the card |
| 14:43.20 | jbalcomb | [TK]D-Fender: no mics, stationary, 18 people tops. the Polycom IP 4000 looks nice or the Avaya 4690 if its SIP |
| 14:43.26 | Irulka | [TK]D-Fender, because we use zaptel |
| 14:44.08 | [TK]D-Fender | jbalcomb: well what I did (and typically suggest), is getting a SoundStation 2W (wireless!!!) and slapping it on an SPA ATA.... |
| 14:44.10 | *** join/#asterisk cybertrickle (n=cybertri@ip70-190-74-204.ph.ph.cox.net) |
| 14:44.42 | [TK]D-Fender | jbalcomb: Also its one of their least expensive models and requires well... no wiring (in the place you use). offers mobility and quality. |
| 14:44.44 | cybertrickle | I need the possible error messages for "Show Channels" and "Show Queue QUEUE" |
| 14:45.09 | [TK]D-Fender | cybertrickle: What kind of error messages? I've never gotten an error before... clarify. |
| 14:45.38 | cybertrickle | For example. I know of 1 for Show Queue, "Invalid Queue" |
| 14:45.59 | [TK]D-Fender | cybertrickle: And why is it you'd be entering an invalid queuename? |
| 14:46.11 | jbalcomb | cybertrickle: you can certainly download the source and look at what error messages they put in there |
| 14:46.54 | jbalcomb | [TK]D-Fender: most likely i'd think error handling in a script, perhaps using the AMI |
| 14:47.53 | tzanger | wow 0pencircuit.net is like home of the pizza-faced 13 year olds |
| 14:47.59 | tzanger | reminds me of my youth, heh |
| 14:48.06 | operat0r | tzanger hey now im 25 |
| 14:48.24 | *** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 14:48.24 | operat0r | champster it is free |
| 14:48.26 | Qb3rt | where can i remove the reminder option that makes the phone ring for 2 seconds when i have a voicemail waiting? this thing ring the phone every 15 minutes! |
| 14:48.37 | operat0r | champster how is it not free |
| 14:48.52 | tzanger | operat0r: ok? |
| 14:49.00 | jbalcomb | Qb3rt: asterisk is doing this or the phone? |
| 14:49.12 | [TK]D-Fender | jbalcomb: Anything scripted should protect you from entering in junk anyways. |
| 14:49.14 | hi365 | <[TK]D-Fender>: I whish it were that simple! http://pastebin.ca/144401 |
| 14:49.39 | Qb3rt | jbalcomb dont know! its doing this when asterisk is reloaded |
| 14:49.43 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:49.44 | jbalcomb | [TK]D-Fender: proper coding chooses to account for unknown situation as best you can, never think it /won't/ happen |
| 14:49.46 | operat0r | "tzanger> wow 0pencircuit.net is like home of the pizza-faced 13 year olds |
| 14:49.58 | jbalcomb | Qb3rt: my guess would be the phone |
| 14:50.00 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:50.00 | [TK]D-Fender | jbalcomb: I suppose.... |
| 14:50.09 | Qb3rt | jbalcomb is suspect the phone but i just want to be sure |
| 14:50.26 | Qb3rt | jbalcomb k thanks ill see the phone configurations |
| 14:50.34 | jbalcomb | Qb3rt g'luck |
| 14:51.22 | [TK]D-Fender | hi365, keep looking at your configs.... |
| 14:52.12 | [TK]D-Fender | hi365: might have a bad module.... |
| 14:52.24 | [TK]D-Fender | hi365: play areound and then try to swap taht one out. |
| 14:52.29 | hi365 | [TK]D-Fender: btw, 4 work fine |
| 14:53.00 | *** join/#asterisk Nivex (n=kjotte@user-0c8hq6g.cable.mindspring.com) |
| 14:53.06 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:53.06 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:53.06 | [TK]D-Fender | hi365: sounding like a bad or poorly seated module. inspect, test and deal :) |
| 14:53.30 | hi365 | will do! |
| 14:53.59 | champster | <operat0r> - Not all ads have that feature, that is because they must pay for it. The link you posted suggested someone getting thier own button. That would not be free since the person with the button pays for its use. |
| 14:54.11 | [TK]D-Fender | OMGZ itz m4tt! |
| 14:56.04 | operat0r | champster explane this then http://www.rmccurdy.com/upload1/Clipboard02.jpg |
| 14:56.47 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:57.32 | champster | The feature is free, the usage is not. |
| 14:57.39 | file | tzanger: voodo magic. |
| 14:57.43 | file | or rather, voodoo |
| 14:58.07 | champster | It generates a call to your hose, then a call to the company, and bridges them. |
| 14:58.17 | champster | They then charge the company for the call. |
| 14:58.25 | champster | It is just a convenience feature |
| 14:58.58 | champster | hose=house |
| 14:59.08 | champster | or home ;-) |
| 14:59.35 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 14:59.43 | rpm | has anyone successfully configured a mediatrix 1204? |
| 14:59.54 | vlt | Aurs, [TK]D-Fender: Yes, as root ;-) And "-rx stop ..." doesn't work: No such command 'stop' ... ??? |
| 15:00.26 | champster | Does anyone know how to send DTMF on an already bridged call? |
| 15:00.53 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:02.09 | hmmhesays | with a keypad |
| 15:02.10 | hmmhesays | ? |
| 15:02.14 | hmmhesays | rpm yes |
| 15:02.22 | hmmhesays | pain in the @$$ they are |
| 15:02.41 | hmmhesays | I've done extensive testing with the 1204 |
| 15:02.43 | [TK]D-Fender | vlt : You are missing quotes : asterisk -rx "stop gracefully" |
| 15:02.46 | hmmhesays | they will NOT register with astersik |
| 15:02.52 | hmmhesays | *asterisk even |
| 15:03.02 | [TK]D-Fender | hmmhesays: I found the 1124 pretty easy... 1204 sucks though? |
| 15:03.24 | hmmhesays | [TK]D-Fender: whole different beast |
| 15:03.32 | jtexter3 | okay, figured it out. Had to change to use TFTP, then unplug the power instead of a soft reboot |
| 15:03.45 | hmmhesays | the 1204 is just junk in general, it is really the weak link in the mediatrix product line |
| 15:03.52 | rpm | hmmhesays: so its garbage? im trying to get a list of hardware which will work with this pbx |
| 15:03.57 | [TK]D-Fender | hmmhesays: Entirely believeable. Their PRI series is a complete import wonder even to them and its a HORRID beast.... |
| 15:04.04 | operat0r | champster what if the company call just gets fwd they still charge for the call and what would it be under ? |
| 15:04.12 | operat0r | as in cell |
| 15:04.15 | hmmhesays | rpm: oh it will work |
| 15:04.26 | hmmhesays | depends on the scenario |
| 15:04.29 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:04.42 | hmmhesays | [TK]D-Fender: their fxs line is fantastic ... the rest of the line, not so much |
| 15:04.59 | hmmhesays | I have hundreds of 1104, 2102 and 1102 in the field |
| 15:05.03 | [TK]D-Fender | hmmhesays: Oh thats the FXO..... yeah, my hopes would be dashed.... |
| 15:05.19 | boobee2 | hmmhesays do you know 1124 (24fxs tdmoe)? |
| 15:05.30 | hmmhesays | do I know it? |
| 15:05.32 | *** join/#asterisk joaovianna (i=joaovian@ool-4354d1a8.dyn.optonline.net) |
| 15:05.33 | [TK]D-Fender | boobee2: Thats not TDMoE.... |
| 15:05.42 | [TK]D-Fender | boobee2: Its a SIP gateway. |
| 15:05.45 | hmmhesays | indeed |
| 15:05.51 | hmmhesays | sip fxs gateway with 24 ports |
| 15:05.55 | benjk | I was about to say, since when does Mediatrix do TDMoE |
| 15:05.59 | [TK]D-Fender | boobee2: And I do. Works well, fairly simple to web configure and does the basics. |
| 15:06.08 | hmmhesays | advanced config via snmp |
| 15:06.11 | boobee2 | oh well sorry.. but i'm planning using those |
| 15:06.16 | boobee2 | i have 300~ analog phones |
| 15:06.19 | hmmhesays | boobee2: i have a few in the field |
| 15:06.19 | bXi | when i start asterisk it says parsing /etc/asterisk/musiconhold.conf : found |
| 15:06.25 | boobee2 | and thought about those as a solution |
| 15:06.25 | hmmhesays | to say the least |
| 15:06.26 | bXi | then it says [chan_misdn.so] |
| 15:06.28 | bXi | and it quits |
| 15:06.32 | hmmhesays | and a good solution they are |
| 15:06.34 | bXi | any idea what could cause this? |
| 15:06.57 | [TK]D-Fender | benjk: * barely does TDMoE, and there must be like what... 3 people using it?! ;) |
| 15:07.10 | hmmhesays | i've replaced a bunch of old nortel pbx's with asterisk and 1124's |
| 15:07.11 | file | [TK]D-Fender: 2.9345 statisically! |
| 15:07.17 | backblue | bXi: increase verbose, coment chan_misdn, so you can check what's happening! |
| 15:07.27 | [TK]D-Fender | trunc(file) !!! |
| 15:07.37 | [TK]D-Fender | OMGZ! |
| 15:07.39 | joaovianna | Very simple question here... I bought a TE110P from Digium and I'm ordering a T1. What T1 should I order ? PRI/T1, Voice T1, Data T1 ??? |
| 15:07.40 | benjk | yeah, unfortunately TDMoE isn't there yet, if it ever will |
| 15:07.43 | hmmhesays | so I started working on the "gel" solo again after the accident |
| 15:07.45 | boobee2 | hmmhesays that's kewl indeed, thanks for the kind info |
| 15:07.52 | [TK]D-Fender | joaovianna: PRI |
| 15:07.54 | backblue | joaovianna: portugues? |
| 15:07.57 | jtexter3 | so, anybody else have a good solution for an operator console that can support a decent number of extensions? Looks like the only other option is the Snom 360 with 2 sidecars? |
| 15:07.59 | joaovianna | Thanks |
| 15:08.00 | [TK]D-Fender | joaovianna: And nothing else... |
| 15:08.02 | hmmhesays | boobee2: np |
| 15:08.12 | joaovianna | backblue: Sim |
| 15:08.13 | hmmhesays | drop me an email if you need any help with those |
| 15:08.21 | backblue | joaovianna: n compres da digium :P |
| 15:08.25 | bXi | backblue: if i comment it out it starts properly and it doesnt give any information |
| 15:08.27 | hmmhesays | buildroot buildroot, its a buildroot day |
| 15:08.37 | boobee2 | ok 10x :D |
| 15:08.38 | jbalcomb | [TK]D-Fender: the 2w only costs $100 USD less than the 4000. i'm not seeing the joys of a mobile conference phone |
| 15:08.48 | hmmhesays | so building for mipsel instead of mips.. thats bad mkay |
| 15:08.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.91) |
| 15:08.58 | tzanger | jbalcomb: which phone? |
| 15:09.02 | backblue | bXi: so the problem it's chan_misdn? |
| 15:09.09 | jbalcomb | tzafrir: Polycom SoundStation |
| 15:09.11 | [TK]D-Fender | jbalcomb: Then you'll love the IP 4000. One of my latest customers has one... |
| 15:09.13 | backblue | joaovianna: és do ist? |
| 15:09.29 | joaovianna | backblue: New York |
| 15:09.30 | [TK]D-Fender | jbalcomb: Sets up like an IP 301. |
| 15:09.37 | bXi | my guess is that i dont have the module |
| 15:09.41 | jbalcomb | [TK]D-Fender: also, i'd like to hang all my ATAs from a tree by there ether and practice my hatchet tossing |
| 15:09.44 | bXi | but i'm not sure |
| 15:09.47 | [TK]D-Fender | jbalcomb: PoE that puppy so you don't feel over bound by it. |
| 15:09.49 | backblue | joaovianna: em new york? heheh, boa vida! i wish. |
| 15:09.51 | *** part/#asterisk operat0r (n=h0msar@adsl-152-157-190.asm.bellsouth.net) |
| 15:10.01 | [TK]D-Fender | jbalcomb: Leave it to Cleaver! |
| 15:10.03 | jbalcomb | [TK]D-Fender: PoE for sure. |
| 15:10.05 | backblue | my city dream. |
| 15:10.32 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 15:10.55 | jbalcomb | [TK]D-Fender: I let fly the PO |
| 15:11.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.91) |
| 15:11.27 | p1p | im having some trouble compiling asterisk-addons from svn, I keep getting file not found when its working with asterisk.h. This seems to be a fairly common problem but I cant seem to figure out a way around it, anyone have any suggestions? |
| 15:12.27 | hmmhesays | updatedb; locate asterisk.h |
| 15:12.29 | p1p | http://pastebin.ca/144454 |
| 15:12.35 | p1p | theres a pastebin of my errors |
| 15:12.53 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:13.25 | p1p | hmm: I can find asterisk.h in the asterisk source directory but I cant seem to find where in the makefile its pointing to it. And what do you mean updatedb? |
| 15:13.43 | hmmhesays | updatedb just updates the locate database |
| 15:14.24 | hmmhesays | where is your asterisk.h? |
| 15:14.37 | eKo1 | find is your friend |
| 15:14.45 | p1p | its in the includes directory in the asterisk source dir |
| 15:14.57 | bXi | does somebody know a good doc on how to get asterisk working with mISDN? |
| 15:15.05 | bXi | i've read a few and they all lack some info |
| 15:15.12 | rpm | i need an external fxo, fxs, internal fxo and fxs (we have chosen sangoma hardware for the internal stuff, it seems to work quite well and you don't need to worry about multiple cards and irq problems) |
| 15:15.32 | tzanger | *sigh* |
| 15:15.33 | hmmhesays | quintum 2nd generation: i use it a lot |
| 15:15.41 | tzanger | lots of bullshit about digium and irq problems |
| 15:15.55 | tzanger | older cards had this issue, yes, but the hardware and drivers are much better than they were |
| 15:16.11 | [TK]D-Fender | rpm: AudioCodes..... |
| 15:16.16 | tzanger | Digium's got a LOT of work to do to eradicate that old image |
| 15:16.29 | file | the TDM2400 also uses a completely different hardware design, which has better compatibility |
| 15:16.43 | MrChimpy | tzanger: in my experience the te411 is still crap |
| 15:16.44 | [TK]D-Fender | tzanger: And its not like new reports don't keep flying in here daily. |
| 15:16.58 | MrChimpy | my experience being last 6 months |
| 15:17.02 | [TK]D-Fender | tzanger: Progress is a good thing though. |
| 15:17.18 | MrChimpy | te411 KILLS my app at anything over 600 calls/min |
| 15:17.29 | rpm | [TK]D-Fender: yeah, we have a couple audiocodes gateways here now, the only thing they don't support is 90 volt message waiting and lack of kewlstart signalling on the fxo gateways.. |
| 15:17.33 | [TK]D-Fender | Till then I still save them a small fortune on tech-suupport monkey calls :) |
| 15:17.41 | hmmhesays | rpm: quintum 2nd generation |
| 15:17.43 | [TK]D-Fender | rpm: Ugh. |
| 15:17.44 | file | [TK]D-Fender: :D |
| 15:17.54 | hmmhesays | I use them almost exclusively for fxo gateway applications |
| 15:18.06 | [TK]D-Fender | file: Scary isn't it? Credit where credit is due, and help for all! |
| 15:18.29 | [TK]D-Fender | hmmhesays: Whats the bad-point of Quintum if you had to pick something? |
| 15:18.36 | file | I don't care what hardware people use, as long as if they have problems with Digium hardware - they call support and try to get them solved |
| 15:19.09 | hmmhesays | [TK]D-Fender: the inability to register each port as a seperate sip user |
| 15:19.10 | backblue | rpm: sangoma it's far better then digium cards. |
| 15:19.13 | backblue | ups |
| 15:19.14 | file | if in the end it doesn't work out and they go elsewhere, at least they tried and they learned |
| 15:19.18 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 15:19.20 | backblue | MrChimpy: it was for you, not for rpm. |
| 15:19.21 | backblue | :D |
| 15:19.34 | batphone | what tool can i use to obtain information on a per-host basis for bandwidth usage for my phones going through iptables? |
| 15:19.43 | hmmhesays | [TK]D-Fender: that is the only drawback I have seen, in comparison to other fxo gateways in my opinion they are king |
| 15:19.54 | MrChimpy | yep, i've figured sangoma is way better. seems the old view still stands |
| 15:19.57 | batphone | i need like a graph showing how much traffic a specific phone is using |
| 15:19.59 | file | I want the answers now, must be all confused somehow... did you say what I heard about? |
| 15:20.17 | hmmhesays | batphone: asterisk-stat-v2 |
| 15:20.21 | [TK]D-Fender | hmmhesays: Its a plus AND a minus at the same time. So it does all the other signalling you could want? |
| 15:20.32 | hmmhesays | [TK]D-Fender: indeed it does |
| 15:20.46 | MrChimpy | digium have released new echo canelling boards, but i can't go through evaluation again giving them a 2nd chance |
| 15:20.50 | IOscanner | I am looking for good rates for inbound DIDs for asterisk. I have outbound termination with callerID modification. |
| 15:20.55 | [TK]D-Fender | hmmhesays: I should investigate.... wish I had one local to play around with. |
| 15:21.01 | IOscanner | any good vendors? |
| 15:21.04 | hmmhesays | and their engineers usually have a patch fixing any bugs I might find within a week |
| 15:21.16 | hmmhesays | IOscanner: i use sixtel |
| 15:21.27 | [TK]D-Fender | hmmhesays: Become an officaial tester there? |
| 15:21.37 | hmmhesays | no |
| 15:21.56 | hmmhesays | the company I work for just sells a lot of quintum and I don't ask their techs stupid questions so they are nice to me |
| 15:21.58 | file | FINALLY |
| 15:22.01 | file | tzanger: post is up! |
| 15:22.07 | tzanger | I don't have any kind of call volume like that, but I'd be curious if the A104 did anything different |
| 15:22.12 | hmmhesays | [TK]D-Fender: the 2 port fxs/fxo unit runs about 600 I think |
| 15:22.45 | IOscanner | Sixtel? do you have a URL? |
| 15:22.47 | hmmhesays | it has 2 fxs ports 2 fxo ports and you can use a combination for 2 calls |
| 15:23.00 | hmmhesays | www.iax.cc is their old url .. they merged with some other company now though |
| 15:23.38 | hmmhesays | dolla fiddy for a DID and 1. something cents a minute |
| 15:23.45 | tzanger | iax.cc, sixtel.net... stay away from 'em |
| 15:23.59 | hmmhesays | tzanger: i've had nothing but solid service from them |
| 15:24.13 | tzanger | hmmhesays: I've had the opposite :-) |
| 15:24.13 | hmmhesays | especially since they merged with ... bah, can't remember who exgen? |
| 15:24.15 | tzanger | it's weird how that goes |
| 15:24.24 | tzanger | my experience is older though, so they may have changed |
| 15:24.34 | hmmhesays | mine is within the last 6 months |
| 15:24.47 | hmmhesays | i've got 8 or 9 did's with them |
| 15:26.46 | hmmhesays | [TK]D-Fender: I was way off on the price of the asm200 |
| 15:27.05 | *** join/#asterisk jailbreaker (n=TY@mail.jetfinanceintl.com) |
| 15:28.33 | *** join/#asterisk bmg505 (n=leon@dsl-146-59-106.telkomadsl.co.za) |
| 15:28.53 | [TK]D-Fender | hmmhesays: Haven't found a lto of retailers for them either... very limited selection. |
| 15:28.58 | [TK]D-Fender | hmmhesays: Got a good link? |
| 15:29.22 | blitzrage | A104? |
| 15:29.27 | blitzrage | www.voipdepot.ca |
| 15:29.55 | hmmhesays | we let them go for $325 for 1 |
| 15:30.22 | bXi | hmmmm |
| 15:30.31 | bXi | i've succesfully loaded the capi stuff i think |
| 15:30.32 | [TK]D-Fender | blitzrage: ! ! ! |
| 15:30.33 | hmmhesays | shipping to canada would probably be another 30-40 bucks |
| 15:30.42 | [TK]D-Fender | blitzrage: But no, we are talking about Quintum FXO gateways |
| 15:30.44 | bXi | capi info in CLI says i have 2 free channels |
| 15:30.58 | bXi | 2 free B channels |
| 15:30.59 | blitzrage | ahhhh |
| 15:31.13 | bXi | would it be possible to call an external number now? |
| 15:31.26 | [TK]D-Fender | blitzrage: And voipdepot.ca's pricing is very unimpressive |
| 15:31.45 | [TK]D-Fender | bXi: How about you go try it... |
| 15:31.51 | blitzrage | but I get my stuff on time, and I know Nabeel |
| 15:31.58 | bXi | how does one try it is my question :) |
| 15:32.05 | blitzrage | there is something to be said about that |
| 15:32.07 | hmmhesays | [TK]D-Fender: if you want one, drop me an email I'll have a sales person send you a quote |
| 15:32.23 | blitzrage | I'd rather have someone who is responsive and sends me my stuff rather than save a couple of bucks |
| 15:33.56 | hmmhesays | that does have its advantages |
| 15:35.04 | blitzrage | and shipping comes from Hamilton which is just down the road |
| 15:35.18 | jtexter3 | okay, another phone question |
| 15:35.18 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:35.25 | jtexter3 | I have a snom 360, and I have the web interface up |
| 15:35.39 | jtexter3 | How do I set the record button to do *1? |
| 15:35.44 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:35.44 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:36.14 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 15:37.01 | [TK]D-Fender | hmmhesays: Well I don't have a personal need, jsut would like to gain experience with to see if it fits into my suggested amterials category. |
| 15:37.32 | hmmhesays | [TK]D-Fender: http://www.quintum.com/support all the docs are free to the world |
| 15:37.46 | hmmhesays | no crappy firmware licenses either |
| 15:38.26 | hmmhesays | so what happens when you try and run a binary built for mipsel on a mips machine? SEGFAULT |
| 15:40.32 | benjk | amterials |
| 15:40.35 | benjk | I like that |
| 15:40.44 | [TK]D-Fender | hmmhesays: Docs != real world experience. I'd need to actually implement a system using one. |
| 15:40.49 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 15:41.08 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 15:41.15 | hmmhesays | [TK]D-Fender: of course, of course |
| 15:41.23 | Assid | VoicePulse you there? |
| 15:42.32 | *** join/#asterisk eNEMY^x (n=eqwrweqr@c213-158-248-202.static.sdsl.no) |
| 15:42.35 | hmmhesays | [TK]D-Fender: one of the things they excel at is legacy pbx integrations. You can drop one in on the trunk side of the pbx and pick calls off you want to send voip |
| 15:43.36 | benjk | not with any of the PBXes that are the norm over here |
| 15:43.48 | benjk | all proprietary digital |
| 15:43.56 | eNEMY^x | when using EAGI with perl, I`m experiencing that several of the wavs don't get played. I`ve figured out that this is because $AGI->stream_file doesnt wait until it's finished streaming the file before it goes further to the next $AGI->stream_file statement. By adding sleep 4; it will actually spit out the wav. Is there a good way for me to actually verify that the stream has completed? |
| 15:43.59 | [TK]D-Fender | benjk: Nearly a contradiction in terms ;) |
| 15:44.14 | benjk | proprietary and digital? |
| 15:44.16 | p1p | Im currently using an 8port fxo Quintum gateway for a clients pbx and it is working flawlessly |
| 15:44.21 | benjk | most certainly not |
| 15:44.40 | blitzrage | Quintum eh? Never heard of that one |
| 15:44.41 | p1p | pulled out their legacy pbx, dropped in the quintum and an ast box and we were off and running |
| 15:45.16 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:45.23 | benjk | you can implement any number of proprietary digital protocols, there is nothing that says it has to be open if digital |
| 15:45.50 | hmmhesays | p1p: yeah they are nice |
| 15:46.05 | hmmhesays | benjk: trunk side of the pbx |
| 15:46.20 | hmmhesays | not the user/station side |
| 15:46.55 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:47.13 | benjk | trunk side is almost exclusively ISDN over here, either BRI or PRI |
| 15:47.29 | [TK]D-Fender | blitzrage: Scary Price on new monitor : http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=2433697&CatId=0 |
| 15:47.52 | hmmhesays | benjk: yes hardly proprietary |
| 15:47.56 | Qwell | [TK]D-Fender: That the sub $300 20"? |
| 15:48.08 | [TK]D-Fender | Qwell: Just over in USD$ |
| 15:48.14 | [TK]D-Fender | Qwell: And 22" :) |
| 15:48.17 | Qwell | ahh |
| 15:48.32 | benjk | still no way to sell any quintum boxes into any of those accounts |
| 15:48.37 | Qwell | file (I think?) mentioned a $350ish CAD 20" |
| 15:49.00 | [TK]D-Fender | Qwell: and now I should you a sub $400 22" :) |
| 15:49.03 | file | nope |
| 15:49.03 | Qwell | [TK]D-Fender: That's decent though |
| 15:49.05 | blitzrage | [TK]D-Fender: mmmmmmmmmmmmmmmmmmmmmmmmmmmm |
| 15:49.11 | file | I mentioned that there $390 CAD 22" |
| 15:49.16 | Qwell | oh |
| 15:49.18 | rpm | fucking mediatrix piece of shit. |
| 15:49.19 | [TK]D-Fender | I think I'm gonna ditch my 19" WS for it |
| 15:49.29 | hmmhesays | rpm, whats the problem |
| 15:49.31 | Qwell | so that is CAD, okay |
| 15:49.35 | [TK]D-Fender | rpm: No, tell us how you REALLY feel.... |
| 15:49.37 | blitzrage | I just use my laptop now instead of a desktop |
| 15:49.39 | file | ncix has it too |
| 15:49.44 | Qwell | [TK]D-Fender: Send the 19" over here |
| 15:49.45 | hmmhesays | i use my desktop for gaming |
| 15:49.51 | blitzrage | HD Discovery is amazing |
| 15:50.06 | coppice | when is the 30" Dell going to get cheaper? :-) |
| 15:50.07 | [TK]D-Fender | Qwell: You get them cheap enough as it is.... |
| 15:50.17 | Qwell | Not <= $0 |
| 15:50.24 | anthm | sweep sweep sweep under the rug http://bugs.digium.com/view.php?id=7576 |
| 15:50.25 | Qwell | I'm proposing $0 here :P |
| 15:50.47 | rpm | hmmhesays: im trying to make a call to my pbx, i've got the line plugged into fxo port 1, i have a guest account setup which sends all calls which are unauthenticated to the [default] context.. |
| 15:51.02 | hmmhesays | you got this bad boy on a public ip? |
| 15:51.17 | rpm | nope. |
| 15:51.18 | hmmhesays | or at least port forward 161? |
| 15:51.19 | Qwell | anthm: a very similar patch was committed |
| 15:51.24 | Qwell | in 7563 |
| 15:51.30 | anthm | yep |
| 15:51.41 | hmmhesays | rpm can't help you much then |
| 15:51.41 | anthm | a less functional one... |
| 15:51.45 | rpm | lemme see if i can setup the port forward. |
| 15:51.48 | Qwell | a more proper one ;) |
| 15:52.21 | Qwell | anthm: but, feel free to write a patch to add the index |
| 15:52.51 | anthm | sell out some more |
| 15:52.54 | anthm | for sale |
| 15:53.14 | anthm | nothing is proper in chan_sip it's ass from head to toe |
| 15:53.23 | hmmhesays | haha |
| 15:53.46 | blitzrage | we could all learn something here... |
| 15:54.58 | *** join/#asterisk mog (n=mogorman@gateway.digium.com) |
| 15:54.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:55.08 | blitzrage | mog: ! |
| 15:55.10 | hmmhesays | i need food in mah bellah |
| 15:55.16 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 15:55.49 | rpm | when i call this mediatrix 1204, i get a second dialtone once it picks up, although i want to to pass all traffic to the pbx and not handle stuff itself. |
| 15:56.06 | hmmhesays | rpm yeah |
| 15:56.10 | hmmhesays | thats cause you set it up wrong |
| 15:56.27 | doolph | hi |
| 15:56.34 | doolph | anyone here good with ipcop? |
| 15:57.03 | hmmhesays | bah I can't find any documentation that tells me if my target proc has built in fpu or not |
| 15:58.03 | Cresl1n | anthm: you're in a lovely mood as usual |
| 15:58.07 | coppice | what is your target proc? |
| 15:58.30 | hmmhesays | mips32 |
| 15:58.44 | Cresl1n | hmmhesays: probably not |
| 15:58.52 | coppice | Cresl1n: considering the state of chan_sip he's being tactful :-) |
| 15:59.06 | coppice | hmmhesays: very unlikely |
| 15:59.23 | Qwell | coppice: "tact" is such an interesting word... |
| 15:59.26 | hmmhesays | I realize this, but i'd like some documetation to confirm |
| 15:59.48 | hmmhesays | bah I spchell gut |
| 15:59.52 | [TK]D-Fender | coppice: You're using Chatzilla right? |
| 16:00.03 | coppice | hmmhesays: of course its true. some guy you never met in the internet told you so |
| 16:00.12 | hmmhesays | haha |
| 16:00.13 | coppice | I am using chatzilla |
| 16:00.15 | Qwell | [TK]D-Fender: chatzilla is kinda cool |
| 16:00.23 | Qwell | I prefer it over anything in Windows |
| 16:00.37 | [TK]D-Fender | coppice: I don't seem to see join/departs for some reason, do you know where I should see the option to enable them again? |
| 16:00.38 | hmmhesays | this could by why my binary is segfaulting if you look at it funny |
| 16:00.50 | Qwell | [TK]D-Fender: I couldn't find such an option :p |
| 16:00.56 | Qwell | but! |
| 16:00.59 | Assid | i gotta find another place for dedicated boxes |
| 16:01.09 | Qwell | You'll notice that you *DO* see joins/parts in the second+ tab in a server |
| 16:01.15 | coppice | there is such an option, but I can't remember where |
| 16:01.16 | Qwell | just...not the first one... |
| 16:01.25 | Aurs | vlt: asterisk -rx "stop now" |
| 16:01.30 | Qwell | or maybe it was only the first tab...I forget |
| 16:01.35 | Aurs | vlt: asterisk -rx stop now <- won't work |
| 16:02.10 | [TK]D-Fender | Qwell: I don't :( |
| 16:02.27 | Qwell | it was a little funky in that regard |
| 16:03.25 | hmmhesays | according to what I've found it could go either way with fpu on the mips32 proc |
| 16:10.15 | *** join/#asterisk intralanman (n=lanman@pool-72-82-74-171.nrflva.east.verizon.net) |
| 16:10.53 | hi365 | i alway forget: how do you get asterisk display in color? |
| 16:11.13 | hmmhesays | click your heels together and repeat "there nothing like color" |
| 16:11.23 | hmmhesays | post on youtube |
| 16:11.32 | hi365 | wow it worked! but no color... |
| 16:12.25 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 16:12.36 | TripleFFFF | anyone have a baldecenter from ibm ? |
| 16:13.16 | intralanman | no, i don't have a baldcenter, but i have a receding hairline though |
| 16:13.20 | intralanman | hahha |
| 16:13.29 | eKo1 | baldcenter? |
| 16:13.37 | coppice | hmmhesays: there is an FPU option for MIPS32, but its very uncommon on the embedded versions. I assume you are playing with an embedded chip? |
| 16:13.39 | TripleFFFF | bladecenter |
| 16:13.52 | hmmhesays | yeah on an audiocodes box |
| 16:14.25 | TripleFFFF | got Item number: 200019439844 on ebay to selel for it BM eServer Blade Optical Pass-thru Module 02R9082 |
| 16:14.32 | hmmhesays | coppice: i wonder if that would explain my segfault problem |
| 16:15.06 | *** join/#asterisk daysmen3 (n=primus@host86-139-116-74.range86-139.btcentralplus.com) |
| 16:17.23 | benjk | baldcenter doesn't explain segfaults, no |
| 16:17.59 | benjk | in fact, the balder you go, the less segfaults you should have as you are supposed to get wiser |
| 16:18.53 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:19.25 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net) |
| 16:21.28 | *** part/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 16:21.37 | hmmhesays | ok compiling without fpu support and no native so loader support |
| 16:21.42 | hmmhesays | we'll see how this goes |
| 16:25.37 | hmmhesays | bah this thing is using an ancient version of uClibc |
| 16:25.53 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 16:26.55 | doolph | why dont u install it on a normal pc |
| 16:27.23 | Dovid | just rebooted my server and my sangoma card decided not work (the fxs ports only !!) can anyone have a look at this and tell me what they think it is ? |
| 16:27.24 | Dovid | http://pastebin.ca/144597 |
| 16:28.45 | *** join/#asterisk tlow (i=unknown@gateway/tor/x-a33a87f1982126c9) |
| 16:29.24 | bkw_ | Dovid, "No such device or address" should give you a clue |
| 16:29.35 | bkw_ | the device node doesn't exist ? |
| 16:29.50 | Dovid | lol |
| 16:29.55 | Dovid | it was working 20 min ago |
| 16:30.01 | Dovid | i made no changes what so ever |
| 16:34.38 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 16:36.22 | jbalcomb | [TK]D-Fender: wheres the config for the voicemail button on the 501's web interface? |
| 16:36.27 | Dovid | bkw_: and now it started working again |
| 16:36.35 | Dovid | u think its the card ? |
| 16:37.50 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 16:38.24 | [TK]D-Fender | jbalcomb: You're KIDDING, right? |
| 16:38.51 | jbalcomb | [TK]D-Fender: bah |
| 16:39.00 | [TK]D-Fender | jbalcomb: You know I don't go in there! |
| 16:39.20 | [TK]D-Fender | jbalcomb: People going in there end up on milk cartons! |
| 16:39.34 | jbalcomb | [TK]D-Fender: I'm sorry I overestimated your overall understanding of your corporate sponsors phone |
| 16:39.56 | doolph | anyone have ipcop firewall |
| 16:41.25 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 16:41.37 | [TK]D-Fender | jbalcomb: General >User Preferences |
| 16:42.57 | [TK]D-Fender | jbalcomb: set "Bypass Instant Message" and "One Touch Voice Mail" to Enabled, and under"Message Center" in your "lines page for the contact |
| 16:43.44 | Dovid | TK: can u look at this ? |
| 16:43.48 | Dovid | http://pastebin.ca/144597 |
| 16:44.09 | Dovid | some times it works and other times it dosent. can it be the card ? |
| 16:44.26 | *** part/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 16:46.35 | *** join/#asterisk evisu (i=hIRC@bzq-88-154-45-231.red.bezeqint.net) |
| 16:46.54 | [TK]D-Fender | Dovid: Show a lot more before asking people to look. |
| 16:47.05 | Dovid | what do i need to show ? |
| 16:47.26 | [TK]D-Fender | Dovid: How about all the related configs? |
| 16:47.47 | Dovid | i didnt think of using ztcfg till then end then when i ran it, it showed it was working so i tried to run asterisk and now its working |
| 16:47.50 | Dovid | gona post em |
| 16:47.51 | Dovid | one se |
| 16:48.23 | *** join/#asterisk soylentgreen (n=fgast@nebukadnezar-em0.only640k.org) |
| 16:48.48 | *** join/#asterisk BugKham (i=CKGLOB@221.128.111.155) |
| 16:49.12 | [TK]D-Fender | Dovid: if its working, don't bother for now. |
| 16:49.13 | Corydon-w | Is People! |
| 16:49.24 | BugKham | is coppice still here? |
| 16:49.30 | coppice | no |
| 16:49.40 | Nivex | only coppicebot |
| 16:49.46 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:49.46 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:49.46 | Dovid | TK: http://pastebin.ca/144637 |
| 16:49.57 | BugKham | coppice, thanks =) |
| 16:50.15 | Dovid | ok. tryin to figure out if it will act up again. i am leavin the country and dont want the client to get screwd, if its the card then i wana replace it now |
| 16:50.34 | BugKham | coppice, where are the app_rxfax.c, app_txfax.c and apps_makefile.patch for spandsp pre30? |
| 16:50.53 | BugKham | coppice, for 1.2.x I mean |
| 16:51.04 | jbalcomb | [TK]D-Fender: Thank you |
| 16:51.17 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 16:51.22 | BugKham | coppice, sorry pre22 |
| 16:51.39 | jbalcomb | [TK]D-Fender: now I can look at the diff file and set it in the phone.conf |
| 16:51.42 | coppice | there isn't an app_rxfax or app_txfax for * for spandsp 0.0.3 |
| 16:52.38 | BugKham | coppice, oh |
| 16:52.55 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 16:52.59 | jbalcomb | [TK]D-Fender: 36 linux boxes and the Senior SysAdmin thinks his alone has no reason to run remote logging. what an emotionally retarted tool. |
| 16:53.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:53.34 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 16:53.47 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 16:54.06 | BugKham | coppice, so I will need 0.0.2 to send/recieve faxes? |
| 16:54.20 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 16:55.52 | hmmhesays | ugh yet another reason not to fly on any former soviet union aircraft |
| 16:56.13 | hmmhesays | 3rd one to explode this year |
| 16:57.02 | coppice | you can use 0.0.2pre26 with * |
| 16:57.28 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 16:57.40 | brif8 | Can AstLinux be run on a std desktop PC without USB and flash memory? and is there any advantages to this ? |
| 16:57.47 | BugKham | coppice, ok thanks |
| 16:58.06 | BugKham | coppice, what is the 0.0.3 for? may I ask? |
| 17:03.18 | coppice | for things other than * right now |
| 17:03.42 | coppice | 0.0.3 has ECM, and T.38 support |
| 17:04.45 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:05.49 | BugKham | coppice, great job =) |
| 17:07.30 | *** join/#asterisk crochat (i=crochat@84-74-146-28.dclient.hispeed.ch) |
| 17:10.20 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 17:10.22 | benjk | brif8, the advantage is that what you don't have cannot break |
| 17:10.44 | benjk | the fewer parts your installation consists of, the more robust it will be |
| 17:12.06 | *** join/#asterisk Avalone (n=Avalone_@dial-478.vl-cen-as1.avtlg.ru) |
| 17:14.16 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 17:14.53 | manopulus | hello, can i playback many files? i.e. playback(file1&file2&file3...&fileN|noanswer) |
| 17:14.54 | manopulus | ? |
| 17:15.09 | doolph | just do it with priority |
| 17:15.10 | Qwell | yes |
| 17:15.15 | Qwell | manopulus: That works just fine |
| 17:15.19 | doolph | or that way |
| 17:15.49 | manopulus | Qwell, thank you. i had to rebuild saynumber and saydigits for prepaid platform. |
| 17:16.51 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 17:17.00 | TripleFFFF | anyone heard of a pcmcia card yet ? |
| 17:17.02 | manopulus | another question. exten => .... background(you-have&ten&minutes) exten => ....dial(sip/${EXTEN}) - will it work by playing announcement before caler will hear ringing? |
| 17:17.02 | TripleFFFF | for voip ;) |
| 17:17.15 | manopulus | TripleFFFF, what card, quicknet? |
| 17:17.18 | benjk | what do you mean PCMCIA card for voip? |
| 17:17.25 | TripleFFFF | to pstn |
| 17:17.26 | *** join/#asterisk KaiHanari (n=Kai@stjhnf01-22-142163031152.nf.sympatico.ca) |
| 17:17.32 | TripleFFFF | like the 100x crap |
| 17:17.34 | benjk | there are ISDN BRI and PRI cards |
| 17:17.35 | manopulus | FXO card? |
| 17:17.35 | Qwell | TripleFFFF: There is one, but there are no asterisk drivers. :) |
| 17:17.39 | TripleFFFF | fxo yes |
| 17:17.46 | benjk | Odin makes them |
| 17:17.47 | TripleFFFF | oh cool |
| 17:17.51 | TripleFFFF | got a link ? |
| 17:17.52 | Qwell | benjk: yes |
| 17:17.56 | TripleFFFF | ill get a team for drivers for it |
| 17:18.04 | benjk | odints.com perhaps |
| 17:18.17 | Qwell | TripleFFFF: there is an issue that will make it very difficult to do that |
| 17:18.20 | benjk | Corydon is working on chan_odin |
| 17:18.25 | TripleFFFF | the ts trew me off |
| 17:18.29 | benjk | for those cards |
| 17:18.34 | TripleFFFF | wat issue |
| 17:18.49 | benjk | at least that was the status some time at the end of last year |
| 17:19.06 | benjk | might want to ask him about progress |
| 17:19.22 | TripleFFFF | http://odints.com/pages/prod/t1e1j1/t2ciapro/t2ciaprofs.htm |
| 17:19.23 | TripleFFFF | ? |
| 17:19.57 | TripleFFFF | 2 T1/E1/J1 interfaces. Software switchable between T1, E1, and J1. |
| 17:20.02 | hmmhesays | bah this ac* is a piece of sh*t |
| 17:20.03 | TripleFFFF | still to much.. not 1 port ones ? |
| 17:20.06 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 17:20.32 | fiber0pti | how can I set up my dial plan to do the following: Someone hears another phone ringing and they want to pick it up from their phone. How can I do this? |
| 17:21.01 | hmmhesays | pickupgroup |
| 17:21.05 | hmmhesays | google it |
| 17:21.07 | syzygyBSD | umm.. what if many phones are ringing, how do you know the one that is |
| 17:21.28 | hmmhesays | you have little electrodes wired to different parts of your body |
| 17:21.28 | benjk | those cards are mostly intended for onsite trouble shooting |
| 17:21.45 | hmmhesays | if you feel a tingle in your junk you know its one phone, your tonque.. another phone |
| 17:22.00 | syzygyBSD | ahh... |
| 17:22.03 | benjk | so you usually need two PRIs to go inside the loop you're analysing |
| 17:22.14 | [TK]D-Fender | fiber0pti: Lookup "pickupgroup" on the WIKI |
| 17:22.30 | syzygyBSD | [TK]D-Fender: a bit slow on that today... |
| 17:22.35 | hmmhesays | 12:22:10) hmmhesays: pickupgroup |
| 17:22.35 | hmmhesays | (12:22:14) hmmhesays: google it |
| 17:22.47 | [TK]D-Fender | syzygyBSD: chan_lag :) |
| 17:22.51 | syzygyBSD | lol |
| 17:22.52 | syzygyBSD | ahhh |
| 17:22.55 | hmmhesays | suuure |
| 17:23.23 | syzygyBSD | I think they could make some improvements to that |
| 17:24.13 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 17:26.40 | hmmhesays | so I got my eye on this hamer at the local music shop here |
| 17:26.56 | *** join/#asterisk overworked554 (n=overwork@atlantis.clearshout.com) |
| 17:28.51 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 17:29.23 | dijungal | Any detailed reporting for Asterisk..? |
| 17:29.39 | hmmhesays | what a fantastically vague question |
| 17:29.40 | dijungal | like time of call, call start, call end, time call was connected.. etc.. |
| 17:29.44 | Qwell | cdr |
| 17:29.51 | hmmhesays | asterisk generates cdrs by default |
| 17:29.54 | dijungal | CDR does not give me enough info |
| 17:29.57 | hmmhesays | what you do with them is your choice |
| 17:30.03 | Qwell | it gives you all of that |
| 17:30.08 | dijungal | oooh |
| 17:30.11 | dijungal | i'lll look at it again |
| 17:30.14 | hmmhesays | as you should |
| 17:30.18 | dijungal | thanks guys |
| 17:30.27 | benjk | dijungal, you need to write your own report writer |
| 17:30.34 | dijungal | ohoo |
| 17:30.39 | hmmhesays | you can have custom cdrs also |
| 17:30.40 | dijungal | but where do i pull the info from..? |
| 17:30.44 | dijungal | oooh |
| 17:30.47 | dijungal | tell me |
| 17:30.49 | benjk | read in the info from the CDR files and generate the reports |
| 17:30.50 | hmmhesays | read |
| 17:30.58 | dijungal | hmm... |
| 17:31.11 | dijungal | ok |
| 17:31.34 | dijungal | also i'm in Ubuntu... and i'm looking for a good softphone to connect to my asterisk box... |
| 17:31.36 | dijungal | any ideas..? |
| 17:31.45 | hmmhesays | http://www.asterisk.org/doxygen/cdr_custom.html |
| 17:31.52 | hmmhesays | idefisk works in ubuntu |
| 17:31.57 | hmmhesays | ekiga |
| 17:32.04 | overworked554 | anyone here using ragi? |
| 17:32.37 | syzygyBSD | ragu? I use that on my pasta |
| 17:32.44 | dijungal | what about twinkle..? |
| 17:32.51 | manopulus | overworked554, ruby? |
| 17:33.01 | manopulus | overworked554, just play or know well rails? |
| 17:33.11 | syzygyBSD | mis type of eagi? |
| 17:33.27 | overworked554 | well i have an app that i built and im having some troubles keeping it happy, is it possible to run ragi with lighty? |
| 17:33.38 | overworked554 | or is there a better way to deploy it other than using webrick |
| 17:33.40 | hmmhesays | ragu makes a fantastic alfredo sauce |
| 17:33.42 | manopulus | overworked554, it is basically fastagi |
| 17:34.01 | manopulus | overworked554, and i guess better to use perl (and keep rest with ruby) |
| 17:34.06 | benjk | I have a friend who is Indian, his name is Raghu |
| 17:34.30 | hmmhesays | do you put his sauce on your pasta? |
| 17:34.36 | benjk | haha |
| 17:34.51 | overworked554 | manopulus: i guess im just looking for some best deployment practices for ragi |
| 17:34.56 | hmmhesays | what the crap is ragi? |
| 17:35.05 | Qwell | hmmhesays: ruby agi |
| 17:35.10 | benjk | reverse engineered agi |
| 17:35.45 | hmmhesays | i see |
| 17:35.56 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 17:35.58 | hmmhesays | i knew a chick named ruby |
| 17:36.01 | manopulus | overworked554, i like perl |
| 17:36.16 | benjk | did you put here sauce on your pasta? |
| 17:36.26 | manopulus | overworked554, for web development i think about rails but only think :) too much load for my head now :) |
| 17:36.48 | overworked554 | my web apps work great, they take a beating on a daily basis |
| 17:36.49 | hmmhesays | yeah it was all red an bloody, it tasted like placenta |
| 17:36.56 | benjk | eeeek |
| 17:36.57 | overworked554 | it just gets very angry when i bring the ragi handler into the picture |
| 17:37.01 | AndyCap | adventure game interpreter in ruby? :) |
| 17:37.10 | hmmhesays | bet you didn't see that coming |
| 17:37.27 | *** join/#asterisk svenna_ (n=svenna@p548D41C6.dip0.t-ipconnect.de) |
| 17:37.46 | benjk | its almost like we're in the wrong channel .... ragi, lighty, webrick .... |
| 17:38.17 | manopulus | benjk, why, ragi is related to asterisk where he can ask ? only here |
| 17:38.30 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1168028839.dsl.bell.ca) |
| 17:38.34 | momelod | hello people |
| 17:38.35 | benjk | I didn't mean that as criticism |
| 17:39.28 | dijungal | anyone has an asterisk box i can try interconnecting with..? |
| 17:39.33 | momelod | is it possible to have asterisk detect the difference between a voice and fax call, and then send the call to a different extension depending on the type of call? |
| 17:39.47 | eKo1 | momelod: kinda |
| 17:39.51 | benjk | exten => fax,1,Foobar() |
| 17:40.20 | momelod | eKo1, u mean i have to hack the code or something? |
| 17:40.25 | benjk | no |
| 17:40.36 | benjk | just put the above in your dialplan |
| 17:40.48 | momelod | no way, its that easy |
| 17:40.49 | momelod | ? |
| 17:40.52 | benjk | but you have to replace Foobar() with something that makes sense |
| 17:40.59 | momelod | right.. |
| 17:41.09 | manopulus | momelod, it is not so easy :) |
| 17:41.16 | momelod | but wouldnt there be a goto if line.. |
| 17:41.17 | benjk | and it needs to be in the context that handles your incoming calls |
| 17:41.18 | manopulus | momelod, better to take different fax # |
| 17:41.40 | benjk | the exten => fax.... already is a kind of goto |
| 17:41.48 | benjk | implicit |
| 17:42.17 | momelod | so why shouldnt i try it then? is it unreliable? |
| 17:42.30 | benjk | I think it only works on zap channels |
| 17:42.39 | intralanman | indeed |
| 17:42.43 | intralanman | or that's what the docs say |
| 17:42.46 | benjk | may not work if your fax is coming in on some SIP device or so |
| 17:43.00 | momelod | i only have zap channels.. (sorry should have mentioned) |
| 17:43.15 | benjk | in that case it should work |
| 17:43.16 | momelod | no, im not interested in ip faxing.. |
| 17:43.32 | momelod | sweet, thanx soo much.. |
| 17:43.46 | benjk | well, using a SIP FXO gateway box on your LAN isn't necessarily IP faxing |
| 17:44.34 | benjk | technically it is, but some people use something like Sipura 3000s to connect their landline to Asterisk |
| 17:44.52 | momelod | oh, well i just have some clients that fax us.. if u could add those fax lines to our hunt group and have asterisk route the call to a fax device zap<->zap then i would be adding two lines to my system for no extra charge :D |
| 17:45.27 | benjk | yeah, with an all zap based setup it should work |
| 17:45.36 | momelod | thanx |
| 17:45.46 | benjk | I was testing and using it once about 3 years ago or so |
| 17:46.20 | *** join/#asterisk copantl (n=copantl@207.13.77.27) |
| 17:46.34 | benjk | the downside is that asterisk has to pick up the line first before it can decide if it is a fax or not |
| 17:46.52 | benjk | or at least back then this was the case |
| 17:47.17 | benjk | maybe with ISDN fax you can detect that it is a fax before you pick up |
| 17:47.26 | benjk | bearer capability should tell you |
| 17:48.03 | benjk | with analog you have to listen to the fax machine's noise before you know |
| 17:48.03 | momelod | dont have isdn just yet.. but when i do get pri ill keep that in mind |
| 17:48.19 | hmmhesays | this mips box is pissing me off today |
| 17:48.21 | hmmhesays | hardcore |
| 17:48.36 | benjk | so if you have a dedicated line just for fax, then you obviously know in advance that what comes in there should be a fax |
| 17:48.58 | hmmhesays | benjk: that would be logical, but most people as you know are devoid of this |
| 17:49.10 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 17:49.38 | benjk | yeah, well, I don't even run the fax through asterisk |
| 17:49.55 | hmmhesays | i've had some success with it |
| 17:49.56 | benjk | I have an analog line which I need for ADSL |
| 17:50.04 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 17:50.33 | benjk | since I have to have it for ADSL and its there anyway, I may as well hook the fax machine up to it, and so I did |
| 17:50.37 | rollergrrl | Does anyone know what the typical packet size is for iax with gsm? |
| 17:50.47 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:50.50 | benjk | just under 30K |
| 17:50.53 | hmmhesays | are you a female? |
| 17:51.06 | intralanman | hmmhesays: down boy |
| 17:51.11 | rollergrrl | very funny dork |
| 17:51.13 | Dovid | hmmheasys: this isnt sex chat |
| 17:51.27 | hmmhesays | i've read about you mythical creatures on the internets |
| 17:51.27 | rollergrrl | he just does that to annoy me |
| 17:51.36 | Dovid | although first time i saw girrrrrrrrrrrl here. |
| 17:51.40 | justinu|laptop | benjk: crunchman still looking for ya :) |
| 17:51.52 | benjk | oh dear |
| 17:51.55 | justinu|laptop | heh |
| 17:51.56 | hmmhesays | along with the sun and vaginas |
| 17:52.21 | benjk | I will contact him when I finished the hashtable lib |
| 17:52.57 | rollergrrl | although I'm not one of them... there are transgenders here too |
| 17:53.01 | docelmo | dude.. asterisk chic's rull! |
| 17:53.26 | benjk | rollergrrl, one might say this is an understatement |
| 17:53.40 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:54.11 | hmmhesays | can you help me as i'm starting to burn all aloone |
| 17:54.20 | hmmhesays | to many doses and i'm starting to get an attraction |
| 17:54.32 | intralanman | tell us about your confidence |
| 17:54.33 | Dovid | hmmhesays: whats ur problem ? |
| 17:54.55 | hmmhesays | Dovid: why do you ask? |
| 17:55.10 | Dovid | for such issues do /join #INeedLove |
| 17:55.33 | benjk | looks like it may be time to leave this channel alone for today |
| 17:55.39 | Dovid | hmmhesays: if its asterisk related i can help |
| 17:55.49 | hmmhesays | heh |
| 17:56.00 | hmmhesays | ./commence whining about being off topic |
| 17:56.09 | rollergrrl | nobody liked my question |
| 17:56.13 | benjk | yeah, his asterisk has got an attraction, whatever that may mean |
| 17:56.22 | Dovid | rollergrrl: what is ur question ? |
| 17:56.32 | Dovid | lolo |
| 17:56.35 | intralanman | i thought it was answered |
| 17:56.36 | benjk | nobody noticed you asked a question |
| 17:56.38 | hmmhesays | I believe benjk answered your question |
| 17:56.38 | AndyCap | rollergrrl: or we don't carry all samplesizes in our head. |
| 17:56.54 | rollergrrl | it's like talking to a group of guys and all they do is stare at your boobs |
| 17:57.03 | benjk | ah yes |
| 17:57.08 | benjk | just under 30K |
| 17:57.12 | hmmhesays | um benjk answered your question about 3 seconds after you asked |
| 17:57.18 | hmmhesays | you were too busy being upset with me |
| 17:57.19 | Dovid | hehe. were a bunch of nerds that dont get any (i am talking for the others ) |
| 17:57.40 | AndyCap | hmmhesays: 30k packet size? |
| 17:57.46 | rollergrrl | it's just under 10000000k too then |
| 17:57.50 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 17:57.51 | rollergrrl | just under 23094810284029384012834092834092834k |
| 17:58.02 | rollergrrl | now 30 bytes would be more like it |
| 17:58.04 | Dovid | rollergrrl: y not run ethereal and see ? |
| 17:58.08 | benjk | something like 28. something K |
| 17:58.10 | AndyCap | rollergrrl: http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2 I guess you should dig out ethereal |
| 17:58.29 | hmmhesays | I bet people would stare more if you only had 1 boob |
| 17:58.49 | AndyCap | btw. did anyone try this? http://www.unleashnetworks.com/articles/asterisk-call-analyzer-for-iax2.html is it worth it? |
| 17:58.55 | Dovid | where is a mod when u need one ? |
| 17:59.02 | rollergrrl | it looked interesting |
| 17:59.11 | hmmhesays | Dovid: lol |
| 17:59.22 | hmmhesays | chances of me getting kicked out of here are slim |
| 17:59.31 | benjk | not so sure their GSM calculation is correct though |
| 17:59.33 | hmmhesays | being I help people all the time |
| 17:59.45 | benjk | you can hold a single channel IAX call on a 32K dialup link |
| 17:59.54 | hmmhesays | indeed you can |
| 17:59.59 | rollergrrl | I'll just stop being lazy and sniff then |
| 17:59.59 | cekc | this echo makes it sound like I'm in a stadium |
| 18:00.00 | Dovid | looks cool |
| 18:00.03 | AndyCap | benjk: bit or byte then? |
| 18:00.09 | Dovid | i would get it if they had if for IAX and SIP |
| 18:00.11 | benjk | 32kbps |
| 18:00.16 | rollergrrl | benjk: packet size |
| 18:00.20 | rollergrrl | not bandwidth |
| 18:00.20 | hmmhesays | do the math |
| 18:00.46 | benjk | too late in the day for that, you'll have to calculate it yourself |
| 18:00.48 | Dovid | rollergrrl: please report back so that we can remain lazy |
| 18:00.57 | rollergrrl | heh |
| 18:01.17 | rollergrrl | gonna go eat a salad first |
| 18:01.20 | rollergrrl | ttfn |
| 18:01.23 | hmmhesays | you know the average bandwidth and packet time |
| 18:01.36 | AndyCap | Haha: This document should not be viewed as a consultative document. It is the readers' responsibility to ensure that the most appropriate telecommunications strategy is applied to his or her business. No liability is accepted by the authors for omission or error. |
| 18:01.45 | cekc | anybody know what causes echo? I get echo on my phones even when I remove power from the ATAs and their relay clicks over to the analog line |
| 18:01.46 | Dovid | :) |
| 18:02.09 | Dovid | cekc: the analog line |
| 18:02.20 | hmmhesays | So Dovid: you seem to have an extremely rigid or non-existant sense of humor |
| 18:02.25 | Dovid | cekc: try having asterisk remove the echo for u |
| 18:02.29 | justinu|laptop | delay causes echo |
| 18:02.36 | eKo1 | latency causes echo |
| 18:02.38 | AndyCap | cekc: evil telco gremlins from the kremlin. |
| 18:02.41 | Dovid | hmmhesays: I do. but there is a time and place for everything |
| 18:02.44 | Cresl1n | cekc: are you using zaptel? |
| 18:02.49 | *** join/#asterisk dood| (n=wizardon@dsl-cust-83-172-73-34.kringdata.net) |
| 18:02.51 | AndyCap | cekc: http://www.voip-info.org/wiki/view/Causes+of+Echo |
| 18:02.52 | hmmhesays | Dovid: this is IRC, not paid support |
| 18:02.54 | justinu|laptop | latency and high energy |
| 18:02.59 | Dovid | hmmhesays: dont llike when u hit on them. there is AOL for that |
| 18:03.09 | hmmhesays | hit on who? |
| 18:03.19 | AndyCap | Dovid: oh, I thought myspace was all the rage these days |
| 18:03.20 | Dovid | the girrrrrrrrl that came in the room |
| 18:03.21 | TrixVox | hmmhesays: Asstricks is serious stuff! |
| 18:03.23 | cekc | I've tried a SPA3000 and some Grandstream box. What about the analog line makes the echo? |
| 18:03.34 | cekc | the internet - serious business |
| 18:03.36 | Dovid | AndyCap: it is ? maybe go there and get some |
| 18:03.50 | hmmhesays | bwhahaha the day I hit on a girl in IRC is the day I put a gun to my head and pull the trigger |
| 18:03.52 | justinu|laptop | hybrid network reflects energy back to you |
| 18:03.53 | Dovid | cekc: u r using only analog ? |
| 18:04.27 | cekc | I removed power from my entire asterisk setup. when power cuts the ata's click a relay to connect the analog line to my internal phone. I get horrible echo |
| 18:04.43 | justinu|laptop | adjust the gains on your ATA fxo port |
| 18:04.44 | hmmhesays | and what happens when you remove the ata |
| 18:04.55 | Dovid | cekc: when u plug a regular phone in do u still have echo ? |
| 18:04.56 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-177-207.bflony.east.verizon.net) |
| 18:04.56 | cekc | when I remove the ata I hear no echo. |
| 18:05.09 | Dovid | then it seems to be the ATA |
| 18:05.20 | cekc | I've tried about 4 different kinds |
| 18:05.36 | AndyCap | told you it was gremlins. :) |
| 18:05.40 | Dovid | cekc: just get a sangome card or digium card |
| 18:05.54 | hmmhesays | you got this thing duct taped to a fluorescent light? |
| 18:06.02 | AndyCap | that's sangoma btw. |
| 18:06.14 | cekc | i duct taped it to a flux capacitor |
| 18:06.38 | hmmhesays | cekc: thats off topic, you shall die by my sword |
| 18:07.16 | hmmhesays | when you have your ata connected to the line, are other phones on that circuit affected? |
| 18:07.39 | cekc | i only have one two line phone |
| 18:07.42 | benjk | echo is caused in the same way as when you are in the mountains and you can hear echo when you shout |
| 18:08.18 | benjk | its sound waves bouncing off something and back to your ears |
| 18:08.18 | cekc | how do I remove the mountains from my phone line |
| 18:08.18 | benjk | you dont |
| 18:08.24 | hmmhesays | so when you pull the chord out of the fxo port and plug that same chord into the analog phone the echo disappears? |
| 18:08.28 | benjk | because in this case the mountain is the telco switch at the other end |
| 18:08.38 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 18:08.42 | benjk | or it may be the ATA |
| 18:08.42 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 18:08.56 | benjk | you have to filter the bouncing back waves out |
| 18:08.59 | cekc | i get the echo when I plug the phone into the fxs port which is electrically wired to the fxo port (via a relay) plugged into the analog line |
| 18:09.04 | benjk | hence echo cancelation |
| 18:09.20 | hmmhesays | 9:28) hmmhesays: so when you pull the chord out of the fxo port and plug that same chord into the analog phone the echo disappears? |
| 18:09.40 | cekc | yes |
| 18:09.52 | hmmhesays | that is freaking weird |
| 18:10.01 | justinu|laptop | yelling more softly usually helps |
| 18:10.13 | cekc | my phone is all fancy electronic, it might be doing some echo cancellation |
| 18:10.14 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 18:10.15 | hmmhesays | you get echo if you send a call fxo-->ip |
| 18:10.16 | benjk | adjust down your gain |
| 18:10.30 | cekc | how do you adjust gain when you don't have the unit powered |
| 18:10.31 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:10.31 | hmmhesays | benjk there is no gain adjustment if the ata has no power |
| 18:10.51 | benjk | how can the ATA work if it has no power |
| 18:10.58 | cekc | if I call my asterisk box and leave a recording I hear echo but the recording is fine |
| 18:11.00 | hmmhesays | spa-3000 has 1fxs 1 fxo |
| 18:11.04 | justinu|laptop | metallic bridge |
| 18:11.16 | hmmhesays | conductive |
| 18:11.25 | benjk | so you're running your call through the passthrough of the ATA then? |
| 18:11.31 | cekc | yes |
| 18:11.34 | benjk | in that case its the transformer in the ATA |
| 18:11.44 | hmmhesays | I think its just a relay |
| 18:11.57 | cekc | it's a relay, you can hear it click |
| 18:12.08 | benjk | used to electrically isolate the telco's network from your end |
| 18:12.09 | hmmhesays | you got a relay with 6 miles of wire in it? |
| 18:12.20 | cekc | I wouldn't put that past sipura |
| 18:12.34 | hmmhesays | benjk I don't think so |
| 18:12.42 | hmmhesays | if it was electrically isolated the phone wouldn't be powered |
| 18:12.52 | cekc | the phone has it's own power source |
| 18:12.53 | benjk | if its got FCC then thats what will be in there |
| 18:13.11 | hmmhesays | cekc plug in a regular phone |
| 18:13.13 | Deeewayne | Does anyone know if there is an upper verbosity limit for handling 240 simultaneous calls? |
| 18:14.06 | benjk | get rid of analog altogether and you will live happily ever after |
| 18:14.28 | cekc | just the telco or the internal phones too? (i.e. no FXS) |
| 18:14.39 | benjk | the less analog the better |
| 18:14.51 | cekc | because I'm planning on getting all IP phones |
| 18:15.01 | benjk | good move |
| 18:15.18 | benjk | but if you can, you might also want to get rid of your POTS lines |
| 18:15.31 | benjk | or only keep a single one for ADSL and fax |
| 18:15.36 | cekc | we have cablemodem |
| 18:15.40 | benjk | and as a backup |
| 18:15.55 | benjk | well, then you may not need any POTS at all |
| 18:15.55 | cekc | we have 2 voice and 2 fax lines |
| 18:16.14 | benjk | I have only got an analog line because I have to have it for ADSL |
| 18:16.27 | benjk | and I plug in the fax machine to that and nothing else |
| 18:16.29 | cekc | how much more do digital lines cost? and who provides them? |
| 18:16.31 | *** join/#asterisk ToyMan (n=stuq@74-32-65-177.dsl1.mdl.ny.frontiernet.net) |
| 18:16.40 | benjk | depends on where you are |
| 18:16.55 | cekc | our fax machines have horrible connections, I wouldn't mind switching fax to all digital |
| 18:17.14 | benjk | well, I only use the fax once or twice a year |
| 18:17.49 | Beighto | Recording a meetme conference call records in signed linear .sln Is there a way to convert this to wav? |
| 18:18.00 | benjk | usually when I need to fax my passport to some embassy to get a visa for some country that needs that kind of bureaucracy |
| 18:18.52 | AndyCap | benjk: reverse this? http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 18:19.15 | cekc | thank you all |
| 18:19.50 | Beighto | AndyCap: sox doesn't recognize the sln format when going the other way |
| 18:19.57 | benjk | what do you mean "reverse"? |
| 18:20.46 | *** join/#asterisk jmang (n=not@24.79.192.187) |
| 18:20.52 | jmang | hello everybody. |
| 18:21.20 | benjk | why not just keep it in sln |
| 18:21.45 | Beighto | benjk: I can't find anything to play it outside of asterisk |
| 18:21.53 | benjk | oh |
| 18:22.11 | jmang | I have a question, I use eyeBeam 1.5 and Asterisk. My asterisk box is connected to my provider via IAX but the provider is using a quintum SIP switch. |
| 18:22.17 | benjk | well, I can just double click it and it plays |
| 18:22.36 | Beighto | plays in what? |
| 18:22.46 | benjk | QuickTime I presume |
| 18:22.48 | jmang | I need to know how I can use Asterisk to resample the eyebeam media from the 80ms frame size, to the 20ms frame size the switch needs. |
| 18:22.58 | jmang | is this possible? |
| 18:23.12 | Beighto | haven't tried quicktime... trying now... |
| 18:23.58 | Beighto | no luck with quicktime |
| 18:24.20 | benjk | quicktime may not be complete on Linux though |
| 18:31.30 | Beighto | trying to play on a windows machine |
| 18:31.32 | sb_mx | Beighto, have you tried using play from the command line? that command is included in sox |
| 18:31.32 | benjk | quicktime may not be fully features on windows either |
| 18:31.34 | benjk | its mostly there so that you can use itunes |
| 18:31.38 | Beighto | sb_mx I have not, but even if that works, how would I convert that to another format aside from putting a microphone in front of the speakers |
| 18:31.38 | benjk | on OSX quicktime plays just about everything |
| 18:31.39 | Beighto | I thought Videolan covered just about everything, but maybe it's not the best for audio |
| 18:31.40 | hmmhesays | bah, what is libm.so |
| 18:31.41 | justinu|laptop | math |
| 18:31.42 | sb_mx | Beighto, i think you can convert slinear files with sox. at least the man page says the sample data encoding can be slinear |
| 18:31.42 | hmmhesays | yeah it makes everything segfault on my mipsel box |
| 18:31.43 | AndyCap | Beighto: so "sox -t raw -r 8000 -s -w -c 1 infile.sln outfile.wav doesn't work? |
| 18:31.44 | benjk | try -b |
| 18:31.44 | *** join/#asterisk blaylock (n=seth@snap.helixsystems.com) |
| 18:31.45 | benjk | instead of -w |
| 18:31.46 | blaylock | has anyone seen the error !! Got reject for frame <number>, retransmitting frame <number> now, updating n_r! ? |
| 18:31.48 | Beighto | I know it can convert TO sln, but not back. Every time I try it says it doesn't recognize sln format, I'll try again with that -b |
| 18:31.49 | blaylock | or have any idea what it means? |
| 18:31.50 | benjk | -w would mean 16bit, I think its more likely its 8bit |
| 18:31.52 | AndyCap | benjk: that does sound probable |
| 18:33.13 | benjk | and if it is 8 and you tell it 16, then it would have trouble making sense of every second byte and tell you it can't recognise the format |
| 18:33.14 | Beighto | It worked with the -b |
| 18:33.14 | Beighto | ! |
| 18:33.14 | benjk | voila |
| 18:33.14 | AndyCap | nothing sox can't do |
| 18:33.14 | Beighto | and the -w... |
| 18:33.14 | *** join/#asterisk kward (n=kward@71-208-147-223.hlrn.qwest.net) |
| 18:33.14 | Beighto | must not have used sox correctly when I tried |
| 18:33.14 | benjk | yeah, sometimes that's the trouble with sox |
| 18:33.18 | AndyCap | that's listed under bugs in the manpage |
| 18:33.19 | benjk | its not always obvious what you have to feed it |
| 18:33.58 | caio1982 | tzafrir: hello there, those problems with t38 hadn't nothing to do with bristuff obviously... i've found 3 stupid errors in my backported patch... it's building fine now and i'm gonna file a bug to include it in the next .deb revision. do you think it's okay to commit? |
| 18:34.00 | *** join/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 18:34.22 | kward | Hi! New to this group ... have a need for "QSIG ISO Path Replacement" in LibPri ... anyone interested in a [bounty} ??? |
| 18:34.54 | benjk | might want to ask in #asterisk-dev |
| 18:35.02 | Beighto | thanks for the help |
| 18:35.23 | *** join/#asterisk backblue (n=moo@87-196-45-57.net.novis.pt) |
| 18:35.46 | kward | #asterisk-dev ... thnx! Will go there! |
| 18:37.28 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 18:39.03 | mountainm2k | PRI... outbound caller-ID show only the extension, or nothing at all... Any recomendded ways to correct? |
| 18:39.18 | mountainm2k | (presumably I need SetCallerID() or something) |
| 18:39.48 | macTijn | re |
| 18:39.49 | benjk | you need to set CALLERIDNUM |
| 18:40.15 | mountainm2k | But if I want to set it to the full DID of the callor? |
| 18:40.32 | mountainm2k | NPA-NXX-XCALLERIDNUM |
| 18:40.47 | mountainm2k | or make that NPA-NXX-X-EXTEN |
| 18:41.05 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 18:42.39 | benjk | youll have to set it to the full directory number |
| 18:42.55 | benjk | 2125551234 |
| 18:43.10 | mountainm2k | where 234 is the extension, how do I set that? |
| 18:43.25 | benjk | depends on the version of asterisk |
| 18:43.30 | mountainm2k | ie, right before I Dial()... |
| 18:43.34 | benjk | the set command keeps changing all the time |
| 18:43.35 | mountainm2k | assume 1.0 for the moment |
| 18:43.39 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 18:43.46 | mountainm2k | heh, ABE... |
| 18:43.50 | benjk | SetCallerIDNum(2125551234) |
| 18:44.08 | dhill | I am using 1.2.9.1. Does Dial() jump priorities by default? |
| 18:44.13 | dhill | or so i need the j option? |
| 18:44.31 | benjk | you need j |
| 18:44.33 | mountainm2k | SetCallerIDNum(2125551${EXTEN}) ? |
| 18:44.40 | dhill | benjk: ok. thanks |
| 18:45.16 | benjk | that would work too, but only if all your extensions are part of your assigned block of numbers |
| 18:45.26 | mountainm2k | they are... Well, most of them are... |
| 18:45.29 | benjk | unless your telco doesn't care what you send |
| 18:45.40 | mountainm2k | They probably don't -- I should test that... |
| 18:46.01 | benjk | still you'd want to send the right numbers in case somebody wants to call you bacl |
| 18:46.46 | mountainm2k | true -- although one could play lots of games with cell-providers voicemails that don't require the password if calling from own number :-) |
| 18:47.37 | benjk | waste of time though |
| 18:48.41 | benjk | you could put all your numbers into astdb |
| 18:48.45 | mountainm2k | true... |
| 18:48.58 | mountainm2k | I have them all in Realtime already... Well, not the full number... |
| 18:49.02 | mountainm2k | just the exten |
| 18:49.16 | benjk | database put ourblockofnumbers 2125551234 234 |
| 18:49.20 | mountainm2k | Guess I need an If() is_numeric($EXTEN) |
| 18:49.25 | mountainm2k | but I'm guessing that won't work... |
| 18:49.45 | benjk | then you look up before each call if the extension trying to dial out is in that database family |
| 18:50.07 | benjk | if it is, you set the callerid accordingly, if it isn't you use the main switchboard number caller id |
| 18:50.12 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 18:50.21 | teknoprep | is there a way to increase outgoing volume for sip/iax calls? |
| 18:50.26 | dhill | what is the proper way to hangup after a Dial(SIP/${EXTEN}, 60) is not answered? Just Hangup() ? or Congestion() ? |
| 18:50.29 | mountainm2k | is there a function to tell if ${EXTEN} is all numeric? |
| 18:50.49 | benjk | I have a patch for pbx.c which adds ${ISNUM(...)} |
| 18:51.08 | mountainm2k | hah, well, that doesn't help me too much -- ABE... No source... |
| 18:51.12 | benjk | but you can always fake it with a numeric context |
| 18:51.18 | benjk | [numeric] |
| 18:51.19 | mountainm2k | ...which I'm gathering is more trouble than it's worth |
| 18:51.22 | Juggie | shuodnt that be a function? |
| 18:51.44 | benjk | exten _X,1,SetVar(ISNUMERIC=TRUE) |
| 18:51.45 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 18:51.54 | benjk | exten => _XX,1,SetVar(.... |
| 18:51.55 | mountainm2k | Now that the new ABE is out, which is based on 1.2 -- perhapps that'd help me with this crap... |
| 18:51.57 | benjk | etc etc etc |
| 18:52.07 | benjk | up to the number of digits you need |
| 18:52.10 | benjk | then |
| 18:52.26 | benjk | exten => i,1,SetVar(ISNUMERIC=FALSE) |
| 18:52.32 | Alric | Anyone want to help with a seg fault that just happened, 1.2.10? |
| 18:52.54 | doolph | how |
| 18:52.55 | Alric | Or look in to. Seems to be voicemail related, but I can't really read gdb output. |
| 18:53.08 | dhill | benjk: So if Dial() is BUSY, does it go to the next priority or does it still jump n+101 ? |
| 18:53.21 | benjk | with j it jumps |
| 18:53.29 | dhill | ok |
| 18:53.31 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 18:53.40 | *** join/#asterisk adorah (n=Administ@87.68.159.29.cable.012.net.il) |
| 18:54.50 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:58.21 | [TK]D-Fender | priority jumping = ICK |
| 19:02.34 | intralanman | ICK indeed |
| 19:03.14 | dhill | if the Dial() is unanswered after X seconds, then what do you recommend to end the call? Hangup or Congestion? |
| 19:03.28 | dhill | or should it ring forever? :) |
| 19:03.49 | *** join/#asterisk bernardovieira (n=bernardo@c911935d.static.bhz.virtua.com.br) |
| 19:06.44 | eNEMY^x | when using $AGI->stream_file, is there any good way to make my script wait for the stream to finish before trying to launch the next stream_file statement? |
| 19:06.46 | hmmhesays | if you don't use voicemail I would let it riong |
| 19:11.34 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-11.getnet.net) |
| 19:11.53 | backblue | j perl |
| 19:13.35 | *** join/#asterisk CrossRoad (n=SilentVa@209.172.67.146) |
| 19:13.47 | trelane_ | now honestly, who the fuck puts "caution wear eye protection" on an RJ45 crimping tool |
| 19:14.35 | *** part/#asterisk CrossRoad (n=SilentVa@209.172.67.146) |
| 19:14.44 | *** join/#asterisk CrossRoad (n=SilentVa@209.172.67.146) |
| 19:17.08 | CrossRoad | hi guys.. I'm fairly new to asterisk, lately I have a issue where asterisk route calls to the wrong ext, - like 262 will either ring 226 or 266, please suggest a way to handle this or point me to the right direction |
| 19:20.49 | *** join/#asterisk fumasterdk (n=fbxmaste@x1-6-00-15-e9-a2-47-b6.k259.webspeed.dk) |
| 19:21.06 | doolph | chek ur extensions.conf |
| 19:21.32 | CrossRoad | any specific thing to look for? |
| 19:21.50 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 19:21.58 | fumasterdk | Hi chan |
| 19:23.02 | fumasterdk | Does anybody know if Asterisk supports any Dialogic cards? And maybe even so the model called Dialogic DTI300SE ??? Any hints would be greately appreciated |
| 19:24.33 | jbroome | someone must have blinded themselves if it's on there |
| 19:25.09 | [TK]D-Fender | fumasterdk: A correction I received concerning this : There is SOME Dialogic support COMING. What model, how well, and WHEN are entirely debatable. If you are considering a future hardware purchase of them, think again. |
| 19:25.19 | [TK]D-Fender | fumasterdk: Its on the WIKI |
| 19:25.31 | fumasterdk | Oki link to the WIKI??? |
| 19:25.44 | fumasterdk | I have 2 old cards laying here so I wanted to play wth them |
| 19:26.01 | fumasterdk | Wiki on asterisk.org? |
| 19:26.01 | [TK]D-Fender | trelane probably the guy who squeezed to hard, snapped one in half and lost an ey to shrapnel. |
| 19:26.16 | [TK]D-Fender | fumasterdk: www.voip-info.org |
| 19:26.43 | [TK]D-Fender | CrossRoad: Sounds more like a DTMF detection error |
| 19:26.59 | fumasterdk | Thx D-fender |
| 19:28.54 | CrossRoad | D-Fender: Thx, but is there a way to check this like where its actually detecting it wrongly. I used DTMF=rfc2833 |
| 19:30.45 | sixsens | HI ALL |
| 19:31.07 | sixsens | who speak russian |
| 19:31.16 | Nivex | Nyet. |
| 19:31.25 | doolph | does exists russian |
| 19:31.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:31.36 | Deeewayne | fumasterdk: chan_dialogic is the channel driver that supports 23 different Dialogic cards. Your card is supported |
| 19:31.51 | Nivex | Ich spreche Deutsch, aber nicht so gut. |
| 19:33.15 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 19:34.59 | Deeewayne | fumasterdk: the DTI/300 card has network interfaces only. You will also need CSP-capable resources |
| 19:35.21 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:35.33 | [TK]D-Fender | CrossRoad: Describe whats on both ends of the call. |
| 19:36.26 | *** join/#asterisk hi365 (n=any@212.199.22.79.forward.012.net.il) |
| 19:36.32 | *** part/#asterisk hi365 (n=any@212.199.22.79.forward.012.net.il) |
| 19:37.01 | CrossRoad | -Fender: VoicePulse is our provider, we use polycom 501s |
| 19:37.41 | CrossRoad | D-Fender:and they are SIP |
| 19:38.19 | doolph | voicepulse sucks with dtmf |
| 19:38.52 | TrixVox | works fine here, incoming calls to ivr |
| 19:39.07 | jbroome | whoah, pretty colors |
| 19:39.09 | TrixVox | CrossRoad: plain asterisk? freepbx? trixbox? |
| 19:39.11 | *** join/#asterisk hi365 (n=any@212.199.22.79.forward.012.net.il) |
| 19:39.19 | CrossRoad | doolp: can you suggest any good provider.. |
| 19:39.28 | doolph | no |
| 19:39.33 | Nivex | jbroome? cripes they'll let anybody in here. |
| 19:40.03 | CrossRoad | TrixVox : Plain Asterisk |
| 19:40.04 | hi365 | hello! im having a problem with the FOP. when i run ./op_server.pl |
| 19:40.04 | hi365 | i get: Can't listen to port 4445 |
| 19:40.07 | *** part/#asterisk kward (n=kward@71-208-147-223.hlrn.qwest.net) |
| 19:40.28 | TrixVox | CrossRoad: codec? |
| 19:41.08 | CrossRoad | TrixVox:ulaw |
| 19:41.11 | CrossRoad | ulaw |
| 19:42.52 | hmmhesays | baaaaaaaaah |
| 19:44.13 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 19:44.18 | benjk | sounds like jitter |
| 19:44.58 | *** join/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 19:45.06 | sixsens | who speak russian |
| 19:45.51 | hi365 | <PROTECTED> |
| 19:45.59 | Telamon | I'm getting a lot of "chan_sip.c: Stopping retransmission on '<long string>'@IP" errors in my logfiles, any ideas as to why? Some of them are from 127.0.0.1 too, which is odd. |
| 19:49.36 | *** join/#asterisk kpettit (n=keith@adsl-70-241-109-36.dsl.hstntx.swbell.net) |
| 19:54.57 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 19:55.47 | *** join/#asterisk ajohnson_laptop (n=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 19:57.23 | *** join/#asterisk designDREam (n=designdr@rrcs-71-40-48-202.sw.biz.rr.com) |
| 19:59.50 | *** join/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai) |
| 20:02.11 | hi365 | Can you help with FOP? |
| 20:02.18 | *** part/#asterisk ajohnson_laptop (n=ajohnson@ip68-104-215-193.ph.ph.cox.net) |
| 20:02.33 | hi365 | im getting: cant connect top port 4445 |
| 20:07.57 | *** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-109-10.wbs.co.za) |
| 20:08.19 | The_LightSide | hi all |
| 20:08.25 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 20:08.39 | folder | trunk not out yet then? |
| 20:08.56 | The_LightSide | does any1 know anything about pri, and "audio before answer" |
| 20:11.49 | hmmhesays | grand |
| 20:12.12 | mog | hello is anyone in here having jingle problems? |
| 20:12.12 | folder | Is there any kind of timeline for svn-trunk anywhere? Like a release schedule? I will be doing my first Asterisk project soon and would like to use 1.4 with the enhanced echo-cancel/jitter buffer and whatever other new features there are. I'm sure I read a date of June 30th at some point. |
| 20:13.04 | Cresl1n | mog: looks like jingle uses asn.1 |
| 20:13.13 | mog | heh you why encode in xml |
| 20:13.17 | mog | when you can use asn.1 |
| 20:13.25 | Cresl1n | apparently it already does |
| 20:13.30 | mog | i got all these bug reports and when i started testing today i cant make it break |
| 20:13.32 | Cresl1n | looks like it's a new dependency |
| 20:13.36 | Cresl1n | libtasn |
| 20:13.41 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 20:14.00 | mog | i think everyone who has problems cant configure it |
| 20:14.29 | Cresl1n | you know that that library is for asn.1 encoding/decoding, right? |
| 20:14.29 | fiber0pti | In my featers.conf file I have set the pickup group to be *8 but it does't work. How do I configure this feature? |
| 20:15.30 | x86 | you have to also put something in your dialplan |
| 20:15.44 | x86 | not sure what, but i remember coming across the same issue ;) |
| 20:15.53 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 20:16.13 | cekc | is it possible with those digium T1 cards to get both phone lines and internet access? |
| 20:16.31 | Cresl1n | cekc: yep |
| 20:16.53 | Cresl1n | ÐÏÞÅÍÕ? |
| 20:17.06 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:17.45 | cekc | I might just cancel my 4 analog phone lines and cablemodem and get a T1 line to handle botht |
| 20:18.10 | x86 | Cresl1n: wouldn't you need to do the DSX upstream from the asterisk box? |
| 20:18.28 | cekc | what's a DSX upstream? |
| 20:18.28 | x86 | Cresl1n: iirc, the digium cards dont have data CSU/DSU on them, do they? |
| 20:18.38 | x86 | cekc: drop-and-insert |
| 20:18.46 | x86 | cekc: what you'll be doing ;) |
| 20:18.47 | cekc | I don't mind using my asterix box as a linux router |
| 20:19.12 | x86 | cekc: probably dont have a CSU/DSU though |
| 20:19.47 | *** join/#asterisk draco_710 (n=tlambeth@12-214-163-60.client.mchsi.com) |
| 20:19.53 | cekc | some company is supposed to call me back, they mentioned something about just giving me an ethernet terminal to handle both internet and phone |
| 20:19.57 | x86 | cekc: you're going to come off the smartjack with RJ48S and run right into your digium card... nothing is doing CSU/DSU for data |
| 20:20.00 | cekc | terminal/termination |
| 20:20.23 | x86 | a T1 would be a better solution |
| 20:20.52 | x86 | it's used more widely, and it's far more reliable than cable, dsl, etc ;-) |
| 20:21.03 | cekc | better than what? I thought I was talking about a T1 |
| 20:21.12 | x86 | plus they've been running voice and data over T1's for years |
| 20:21.18 | x86 | ethernet != T1 |
| 20:21.21 | cekc | ah |
| 20:21.46 | cekc | so I'm going to need a digium card for the voice, and then a csu/dsu for the internet |
| 20:21.48 | Cresl1n | look for datamode zaptel |
| 20:21.53 | Cresl1n | that should do it |
| 20:22.00 | Cresl1n | you can use the zaptel card for everything |
| 20:22.04 | x86 | Cresl1n: software CSU/DSU? |
| 20:22.24 | cekc | how well does zaptel work with asterisk? |
| 20:22.34 | Cresl1n | yeah, google for zaptel data mode |
| 20:22.37 | [TK]D-Fender | Yes you can do voiec & data on a single T1... |
| 20:22.38 | Cresl1n | that should probably do it |
| 20:23.10 | cekc | [TK]D-Fender: but what kind of equipment do I need to do that |
| 20:24.02 | jbalcomb | Is anyone capturing the call quality metrics via ${RTPAUDIOQOS}? How and are you doing anything useful with them? |
| 20:24.04 | hi365 | [TK]D-Fender: im having a problem with FOP. do you think you can help? |
| 20:24.04 | hi365 | when i run ./op_server.pl i get: Can't listen to port 4445 |
| 20:24.12 | Cresl1n | cekc: just a zap card |
| 20:24.16 | Qwell | hi365: Something is probably already running on that port |
| 20:24.17 | [TK]D-Fender | cekc: Just a Digium or Sangoma digital card. its all in software period. |
| 20:24.21 | Cresl1n | and compile zaptel with datamode enabled |
| 20:24.36 | cekc | I'd like to buy digium just because of brand name |
| 20:24.56 | jbalcomb | [TK]D-Fender: are you familiar with ${RTPAUDIOQOS}? |
| 20:25.12 | [TK]D-Fender | jbalcomb: nope. |
| 20:25.16 | hi365 | Qwell: any idea of what? fop was working. i tried to upgrede, unsuccesfuly. tried to go back and thats what i got |
| 20:25.27 | Qwell | hi365: netstat -lp |
| 20:25.54 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 20:26.32 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:26.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:27.07 | jbalcomb | [TK]D-Fender: you might wanna check it out. seems like good stuff. Its part of the RTCP patch committed to trunk in June. |
| 20:28.10 | hi365 | Qwell: http://pastebin.ca/145070 |
| 20:28.23 | *** join/#asterisk clive- (n=pirch@dsl-145-39-83.telkomadsl.co.za) |
| 20:28.56 | jbalcomb | [TK]D-Fender: there was also talk of phone passing back QoS reports with the BYE message that you can pull with SIP_HEADER |
| 20:29.06 | jbalcomb | [TK]D-Fender: i wonder if the polycoms can do this |
| 20:29.19 | jbalcomb | s/can do/do |
| 20:30.22 | *** join/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
| 20:30.30 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 20:31.05 | cekc | http://www.digium.com/en/products/hardware/te110p.php I'll probably buy this one |
| 20:31.13 | dwrecktion | anybody have an experience running Asterisk on an 2GHz AMD Opteron 246? |
| 20:31.17 | Qwell | hi365: yeah, perl is already running on that port |
| 20:31.57 | hi365 | is that a good thing? |
| 20:32.22 | Qwell | only if you want it to be already running |
| 20:32.52 | hi365 | im still a bit fresh with all this. do i need pearl for the FOP? |
| 20:32.56 | x86 | dwrecktion: you'll have better luck describing your issue than seeing if anyone has a similar setup ;) |
| 20:34.12 | dwrecktion | x86: i don't have an issue, I'm just looking to buy a server to host asterisk and i was seeing if anyone had an experience with it running on that processor |
| 20:34.22 | fiber0pti | where do you define if a phone is in a pickup group? in sip.conf? |
| 20:35.07 | *** join/#asterisk ArchimedesTwo (n=Archimed@66-100-35-23-static.dsl.oplink.net) |
| 20:35.28 | x86 | dwrecktion: the general conscience is to use Intel platforms, and stay away from 64bit like the plague |
| 20:35.40 | x86 | fiber0pti: yeah |
| 20:36.07 | dwrecktion | x86: thanks, that's the kind of info i was looking for. |
| 20:36.13 | ArchimedesTwo | Where can I find the file format I need to pass to /var/spool/asterisk/outgoing to call an outside phone number and connect to a known extension? |
| 20:36.15 | The_LightSide | hi, im looking for some information on audio before answer on a digium pri card |
| 20:37.41 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 20:43.20 | *** join/#asterisk daniel_bergamini (n=daniel_b@70-41-166-149.cust.wildblue.net) |
| 20:44.55 | *** join/#asterisk adorah (n=Administ@87.68.159.29.cable.012.net.il) |
| 20:46.04 | xnon | friends is posible use asterisk for send sms to a movil phone or PDA phone? |
| 20:46.42 | tzanger | xnon: yes |
| 20:46.54 | xnon | how to¿ |
| 20:47.08 | tzanger | either through sending the message to an email-sms gateway, using an SMS service, or using the app_sms command if you have access to the carrier's SMSC |
| 20:47.10 | xnon | have you any example for help me! |
| 20:47.48 | tzanger | xnon: emailing is as simple as sending an email to the cell carrier's SMS email gateway |
| 20:48.32 | xnon | have u any manual or link to a website with a example for this? |
| 20:48.45 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 20:49.05 | xnon | have you this service in your asterisk personal server? |
| 20:49.26 | rvhi | can * realtime read dialplan on the fly? |
| 20:49.31 | hmmhesays | haha |
| 20:49.34 | hmmhesays | that would be the point |
| 20:49.35 | *** join/#asterisk justnulling2 (i=justnull@ool-182e41b0.dyn.optonline.net) |
| 20:49.45 | rvhi | i.e. if i modify mydql dial plan, will be effect right away |
| 20:49.52 | hmmhesays | that would be the point |
| 20:50.10 | rvhi | somewhere i read that need to do a 'reload' or somehting like that |
| 20:50.26 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 20:50.40 | hmmhesays | you are mistaken |
| 20:50.46 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 20:50.49 | obiwanmikenolte | extensions reload at the cli |
| 20:51.03 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:51.51 | rvhi | that's not good, if i allow users change dial plan (indirectly though), every time there is a change, i need reload |
| 20:52.09 | rvhi | why doesn't it work like ast_data? |
| 20:52.21 | obiwanmikenolte | Why would you allow users to change the dialplan? How would you even orchestrate that? |
| 20:53.29 | rvhi | ok, not user, say our own provisioning people |
| 20:53.35 | hmmhesays | asterisk realtime doesn't require a reload |
| 20:53.43 | hmmhesays | unless you change db info |
| 20:53.48 | Qwell | static realtime does |
| 20:54.02 | hmmhesays | yes |
| 20:54.18 | rvhi | what's the difference? static realtime and realtime? |
| 20:54.37 | Qwell | rvhi: static realtime is for the actual config file, where it reads the DB at load time |
| 20:54.47 | hmmhesays | yeah |
| 20:54.48 | hmmhesays | switch => Realtime |
| 20:54.48 | hmmhesays | <PROTECTED> |
| 20:54.50 | hmmhesays | that does not |
| 20:55.42 | rvhi | i'm using ast_data, so with realtime, no need, right? |
| 20:56.24 | hmmhesays | i dunno |
| 20:56.25 | hmmhesays | never used it |
| 20:57.01 | rvhi | so realtime would so a sql query for each dial plan priority, right? |
| 20:57.51 | hmmhesays | i don't remember if it does it per call or not |
| 20:58.09 | hmmhesays | a query per priority sure seem inefficient |
| 20:58.19 | Qwell | it's per priority |
| 20:59.01 | rvhi | that's the same as ast_data, also what i want. thanks! |
| 21:00.32 | The_LightSide | i assume no1 can help me with my PRI query? |
| 21:00.58 | obiwanmikenolte | First, you have to ask a question |
| 21:01.05 | The_LightSide | i have.... |
| 21:01.08 | The_LightSide | <PROTECTED> |
| 21:01.14 | obiwanmikenolte | What information? |
| 21:01.18 | The_LightSide | im looking for some information on audio before answer on a digium pri card |
| 21:01.44 | obiwanmikenolte | Haha. |
| 21:01.44 | clive- | lightside, I have read stuff about that, hmm, google for it in the archives, there is a way to do that |
| 21:01.48 | The_LightSide | basically, the pri equipment does not generate its own ring/busy tones |
| 21:02.16 | The_LightSide | but * does not seem to be passing the audio stream thru until answer |
| 21:02.26 | x86 | you can Answer(), do something, then Ringing() |
| 21:02.53 | x86 | and it wont appear to the caller that the call is ringing until Ringing() is called |
| 21:03.02 | *** join/#asterisk c4t3l (n=c4t3l@72.54.52.46) |
| 21:03.06 | The_LightSide | ok, im the lazy type here, im using one of the preconfigured ones :/ |
| 21:03.27 | The_LightSide | with the freepbx interface ... |
| 21:03.35 | The_LightSide | not trixbox tho |
| 21:03.55 | The_LightSide | clive-, i havnt had much success with the googling |
| 21:03.59 | The_LightSide | :( |
| 21:04.04 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 21:04.07 | x86 | The_LightSide: did you read what i said? |
| 21:04.24 | The_LightSide | yeah |
| 21:04.25 | x86 | The_LightSide: you'll have to Answer, do whatever you want, then Ringing |
| 21:04.32 | x86 | that's how it's done |
| 21:05.17 | The_LightSide | and for things like billing info... the call should only be billed from the time the call is actually answered |
| 21:10.20 | jbalcomb | Anyone coding apps or functions for Asterisk should check this out: http://www.lobstertech.com/code/asterisk_module_generator/ |
| 21:12.11 | *** join/#asterisk intralanman (n=lanman@pool-71-253-195-19.nrflva.east.verizon.net) |
| 21:12.34 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:21.04 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:22.23 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:24.09 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:26.48 | *** part/#asterisk dwrecktion (n=dwreckti@71.16.158.170) |
| 21:28.10 | eKo1 | jbalcomb: that's pretty cute |
| 21:28.10 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 21:28.32 | teknoprep | i have asterisk behind a nat firewall.. do i have to edit any configs to tell my trunks my external ip and all that jazz ? |
| 21:28.40 | teknoprep | this was working great when asterisk had an external ip |
| 21:29.33 | eKo1 | why yes because nat sucks |
| 21:29.43 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:30.00 | obiwanmikenolte | But how many people are using IPv6? |
| 21:30.14 | teknoprep | eKo1 thats great.. how to make it work? |
| 21:30.34 | [TK]D-Fender | teknoprep: you need to set EXTERNIP, LOCALNET, and NAT=YES in [general] in sip.conf |
| 21:30.55 | eKo1 | [TK]D-Fender: you're assuming that SIP is being used. |
| 21:31.05 | teknoprep | ty |
| 21:31.18 | [TK]D-Fender | eKo1: Yes, but stop talking about my ass ok? ;) |
| 21:32.36 | teknoprep | yeah that didn't work |
| 21:32.37 | teknoprep | hmm |
| 21:32.39 | teknoprep | brb |
| 21:34.02 | *** join/#asterisk Alric (n=nbowyer@64.6.45.2) |
| 21:38.31 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:42.10 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 21:45.17 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 21:50.11 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 21:50.26 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 21:51.16 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:51.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 21:53.41 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:54.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:54.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:54.41 | *** join/#asterisk alawguy (n=mike@85-124-36-191.dynamic.xdsl-line.inode.at) |
| 21:55.10 | *** join/#asterisk obscurant (n=obscuran@12-32-45-95.static.blackfoot.net) |
| 21:58.06 | rvhi | can mp3 file be used for MOH? |
| 21:58.24 | obiwanmikenolte | rvhi: You so crazy |
| 21:58.37 | rvhi | ?? |
| 21:59.26 | obiwanmikenolte | All these questions, but not even so much as a Google search to find the answers |
| 22:00.59 | rvhi | ok, found the answer |
| 22:05.16 | *** join/#asterisk ki2k (n=ki2k_@207.231.83.242) |
| 22:05.47 | ki2k | anyone know what happen to conferences if your zttest is below optimal? |
| 22:12.27 | *** join/#asterisk CrossRoad (n=SilentVa@209.172.67.146) |
| 22:14.13 | *** join/#asterisk k31th (n=keith@87.117.194.66) |
| 22:16.53 | k31th | any prefered distro for asterisk ? |
| 22:17.08 | eKo1 | no |
| 22:17.28 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 22:19.34 | *** join/#asterisk annonimous (n=annonimo@201.144.136.21) |
| 22:19.55 | xnon | anybody have ur asterisk server configured for send SMS Mensages to a movil ? |
| 22:20.10 | xnon | anybodt know how it is posible? |
| 22:20.33 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net) |
| 22:22.33 | *** join/#asterisk cr_0 (n=y@toronto-HSE-ppp4334596.sympatico.ca) |
| 22:22.42 | annonimous | hiya |
| 22:23.15 | cr_0 | is there any utility that can test and report on MOS scores that i can set to run in a cron? |
| 22:23.33 | tzanger | cr_0: nope |
| 22:23.33 | tzanger | and hello again :-) |
| 22:23.37 | cr_0 | hey, howdy. |
| 22:23.45 | *** join/#asterisk [Outcast] (n=outcast@222-154-56-47.jetstream.xtra.co.nz) |
| 22:23.48 | [Outcast] | sup all |
| 22:23.54 | tzanger | MOS isn't a single value, it's a number gained through dozens if not hundreds of tests |
| 22:23.56 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 22:24.02 | tzanger | that's why there's no MOS score utility |
| 22:24.07 | cr_0 | why not? i don't know how to calculate MOS, but appart from that, it doesn't seem that hard. |
| 22:24.20 | tzanger | because it's really a combination of a poll/survey and research project |
| 22:24.22 | cr_0 | tzanger, there are commercial apps. i know. i have seen them. |
| 22:24.22 | tzanger | it's not a single test |
| 22:24.33 | tzanger | yes, but AFAIK they're not really giving you MOS |
| 22:24.40 | tzanger | coppice would be able to give more detail if he were around |
| 22:25.08 | unmanaged | http://www.techabulary.com/m/mos.html |
| 22:25.20 | tzanger | I've been hankering to write an asterisk dialplan command to report the variance between a known waveform and what was received to give a quality value |
| 22:25.23 | tzanger | but I've just not had the time |
| 22:25.46 | annonimous | is there a way to setup my asterisk with a double nat isp? |
| 22:27.15 | cr_0 | let me see if i can dig up how the guys at Bell are doing it. i've trialed the software, but at the time i only cared about the QoS queue metrics. |
| 22:28.15 | cr_0 | they do get a "MOS score" graph between "service POPs" in a full mesh. |
| 22:29.49 | cr_0 | i love how the old guys tell stories about how they used to calculate it.... with "expert listeners". |
| 22:30.48 | hads | Mean _Opinion_ Score |
| 22:30.52 | annonimous | is there a way to setup my asterisk with a double nat isp (providers who assing ips like 10.0.0.*)? |
| 22:31.17 | cr_0 | hads: yup :) |
| 22:31.32 | PESQ | how is 10.0.0.* tell you it is double nat'ed? |
| 22:32.06 | annonimous | PESQ, he, i guessed that was double natted cause the inet provider who send it send it in that way |
| 22:32.29 | cekc | he probably has a nat router of his own, or is trying to connect to a remote host behind a nat |
| 22:32.58 | annonimous | its a customer who have a cable connection instead of xdsl and i want to see if asterisk can link external ata's? |
| 22:34.03 | annonimous | well here in mexico its common to see that kind of connections of providers like cable or antennas |
| 22:36.01 | cr_0 | if there is no MOS tests, what do other people use to ensure their network is of good quality for voice? |
| 22:37.06 | cr_0 | tell me it isn't ping. |
| 22:37.41 | eKo1 | annonimous: no, you can't |
| 22:38.51 | annonimous | eKo1, so i need to tell to my custome to change for xdsl or something like that, arent he? |
| 22:39.00 | *** part/#asterisk xai (i=pasta@about/networking/0.0.0.0/xai) |
| 22:41.54 | *** join/#asterisk bethaud (n=eamonn@host-84-9-92-137.bulldogdsl.com) |
| 22:42.36 | bethaud | trying to setup an X101P, and all I get is static on incoming POTS. Any ideas? |
| 22:43.36 | eKo1 | annonimous: make them use iax |
| 22:43.44 | annonimous | iax? |
| 22:43.50 | annonimous | k |
| 22:43.56 | annonimous | let me read about iax |
| 22:44.11 | SuPrSluG | bethaud:zap show channels output? |
| 22:44.20 | bethaud | I should add that the demo app responds just fine to DTMF, so *something* is getting through ... |
| 22:44.51 | SuPrSluG | bethaud:demo uses psuedo zap |
| 22:45.29 | SuPrSluG | bethaud:output from ztcfg? |
| 22:45.39 | bethaud | SuPrSlug : Chan Extension Context Language MusicOnHold |
| 22:45.40 | bethaud | <PROTECTED> |
| 22:45.40 | bethaud | <PROTECTED> |
| 22:46.42 | SuPrSluG | bethaud:you should paste stuff at pastebin.ca or you'll get some flak |
| 22:47.00 | bethaud | SuPrSlug: looks like it does indeed - so what is that? I guess google is my friend ... I did also try setting up an extensions,conf that just answered and did an echo, but got the same results ... |
| 22:47.26 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 22:47.44 | bethaud | SuPrSlug: Apologies - IRC is a new experience ( like 10 minutes ! ) |
| 22:47.58 | *** join/#asterisk Jenocin (n=jenocin@revlookupwb.maintech1.com) |
| 22:48.13 | Jenocin | is it still possible to reflash a linksys pap2, can only find the vonage branded ones in stores around me |
| 22:48.14 | SuPrSluG | bethaud:i'ts ok that was a small one. |
| 22:49.16 | SuPrSluG | bethaud:how about the output from zap show status . should say ok |
| 22:49.57 | bethaud | SuPrSlug: OK and 3 zeros, for a Generic clone board ( it's an MD3200 chipset ) |
| 22:49.57 | SuPrSluG | bethaud:also dsl or cable? if dsl is it filtered? |
| 22:50.15 | SuPrSluG | bethaud:same as mine |
| 22:50.46 | IOscanner | Anyone know a source to buy DID's for Dallas, Texas? Everyone seem to be out. I need about 20. |
| 22:50.54 | bethaud | SuPrSlug: DSL, and I've got the card filtered ( always assuming the filter works ) |
| 22:51.34 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 22:51.58 | SuPrSluG | bethaud:try it w/out |
| 22:52.20 | bethaud | SuPrSlug: I'm in the UK, so the wxfco is set with opermode=1, which I believe is correct? |
| 22:52.54 | SuPrSluG | bethaud:not familiar w/ uk telco |
| 22:53.03 | bethaud | SuPrSlug: OK, I'll get a socket doubler in the morning. I guess that might be why it doesn't spot the line hanging up as well? |
| 22:53.37 | bethaud | SuPrSlug: Not being familiar with UK telco is no crime, as it appears most of the UK phone companies share your position ;) |
| 22:53.57 | SuPrSluG | bethaud:there should be info on that in the wiki |
| 22:53.59 | generalhan | hey guys, i have an aastra phone with the same firmware as the other 30 that i have, but for some reason, this one phone cannot send digits to the VM system? like... when he dials the VoicemailMain for his extension and it asks for his pasword, no matter what he hits * says he didnt hit anything, so of course the password is invalid. any ideas? |
| 22:54.33 | generalhan | but he can call into the autoattendant and hit digits all day long that register |
| 22:54.56 | bethaud | SuPrSlug: yep, I dug out the caller ID stuff, although the debian packages appear to be patched already as of a couple of versions ago. Didn't see anything about the DSL filters, though. |
| 22:57.46 | SuPrSluG | bethaud:look here to see if ti helps http://www.velocityreviews.com/forums/t235347-asterisk-detecting-pots-line-hangup.html |
| 22:58.27 | ki2k | anyone know what happen to conferences if your zttest is below optimal? |
| 23:00.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-25-149.cybersurf.com) |
| 23:00.39 | bethaud | SuPrSlug: Thanks for that, I dug that up earlier but still no joy. I'm guessing I need to get the basic voice stuff working as a first step. I'm just digging through a web page on the pseudo zap ( havent reached it yet :) |
| 23:03.33 | *** join/#asterisk nailbags (i=someone@c220-237-123-137.randw1.nsw.optusnet.com.au) |
| 23:06.08 | SuPrSluG | bethaud:options wcfxo opermode=UK maybe? |
| 23:10.33 | bethaud | SuPrSlug: tried that, but digging through the wcfxo source it looks like opermode=1 selects the CTR21 DAA mode. opermode=UK seems to be for the specific impedance tuning the the tdm400 cards ( and mebbe some others, I only had a quick look ) |
| 23:12.24 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 23:12.48 | dhill | If Dial() is picked up, does the extension end there? Or does it pick back up after a user hangs up? |
| 23:12.58 | bethaud | SuPrSlug: many thanks for your help, however I now need to go and get some sleep - the kiddies are up in 6 or so hours .. :) I'll try the no filter thing tomorrow . Cheers ! |
| 23:13.09 | dhill | I have an answering machine that picks up.. but when the caller hangs up.. the answering machine doesn't know... |
| 23:13.09 | SuPrSluG | cheers |
| 23:13.18 | bethaud | \quit |
| 23:17.35 | *** join/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net) |
| 23:17.44 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:18.43 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 23:19.19 | ringhals | I just had an asterisk server bomb. I am running 1.2.7 and the error was in channel.c and was about blocking a deadlock 10 retries |
| 23:19.30 | trelane | ok |
| 23:19.34 | ringhals | can anyone point me in the right direction? |
| 23:19.40 | Bullseye_Network | I've got a system using SIP to a voip provider and inbound audio is choppy but outbound is not. What would be the most probable problem. |
| 23:19.41 | trelane | have you upgraded to the recent version as there's been numerous bugfixes |
| 23:20.01 | trelane | Bullseye_Network, insufficient inbound bandwdith or borked provider |
| 23:20.02 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 23:20.07 | ringhals | no because it is in production from 5 am to 12 am :-( |
| 23:20.15 | generalhan | trelane: did you happen to see my post about a phone not sending digits to the voicemailmain ? |
| 23:20.20 | ringhals | any idea what to look at that may be causing it? |
| 23:20.25 | trelane | generalhan, check your dtmfmode |
| 23:20.29 | trelane | (I've seen that one before) |
| 23:20.35 | generalhan | trelane: what should it be set to > |
| 23:20.43 | trelane | generalhan, what phones are you using? |
| 23:21.05 | generalhan | they are Aastra SIP phones |
| 23:21.10 | trelane | ringhals, I'd look at the out of date asterisk you're running that may have bugfixes |
| 23:21.20 | trelane | generalhan, are you doing anything with dtmf= in sip.conf now? |
| 23:21.22 | Bullseye_Network | trelane, thats what I thought but bandwidth is not a problem. And ofcoarse the proveder says its not them. :( |
| 23:21.26 | generalhan | nope |
| 23:21.31 | trelane | try dtmf=inband and if that doesn't work dtmf=info |
| 23:21.37 | generalhan | ok |
| 23:21.45 | trelane | Bullseye_Network, of course it's not them, and every provider's honest, done any traceroutes to their sip gateway? |
| 23:22.59 | *** join/#asterisk smashingnick (n=smashing@ip68-14-109-129.no.no.cox.net) |
| 23:23.15 | Bullseye_Network | trelane, yes I get 45ms and no packet loss. |
| 23:23.39 | smashingnick | hello all im having a problem with ground start signaled pots lines |
| 23:23.51 | trelane | Bullseye_Network, interesting, dunno quite what to tell you |
| 23:23.56 | *** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au) |
| 23:24.03 | trelane | smashingnick, digium hardware? |
| 23:24.06 | smashingnick | yes |
| 23:24.15 | trelane | they have free support, check out "contact us" on their website |
| 23:24.16 | smashingnick | two tdm400p fxo modules |
| 23:24.28 | generalhan | trelane: ok i tried both and neither work. but all of the other 30 of these phones i have are working just fine with no dtmf settings defined |
| 23:25.07 | trelane | generalhan, there's one other possibility, hang on a tick... |
| 23:25.13 | generalhan | sure thing |
| 23:25.16 | generalhan | thanks for the help |
| 23:25.18 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 23:25.29 | generalhan | hey hey justinu |
| 23:25.41 | trelane | generalhan, try dtmfmode = instead of dtmf = |
| 23:26.39 | trelane | and if that doesn't work try it with rfc2833 |
| 23:26.41 | generalhan | trelane: but would i need that for one specific phone ? if all the others dont have that and work just fine ? |
| 23:26.45 | trelane | dtmfmode = rfc2833 |
| 23:26.48 | generalhan | this phone was working just fine until this morning ! |
| 23:26.49 | generalhan | lol |
| 23:26.51 | trelane | generalhan, are they all the same model? |
| 23:26.52 | generalhan | ok ill try that |
| 23:26.53 | generalhan | yes |
| 23:27.02 | trelane | ok well possibly a bad phone? |
| 23:27.08 | generalhan | i have 35 Aasta 9112i SIP phones and 15 Cisco 7960sw |
| 23:27.29 | generalhan | well everything else works just fine ... ONLY the VMMain doesnt acceot the digits for that phone. |
| 23:27.32 | dhill | bullseye in southfield sucks |
| 23:27.47 | trelane | generalhan, for that phone = 1 phone not 1 phone model right? |
| 23:27.50 | generalhan | i can call any auto attendant including our own and it will take digits we send to them |
| 23:28.00 | trelane | generalhan, it's a DTMF issue |
| 23:28.02 | trelane | trust me |
| 23:28.13 | generalhan | trelane: yes of the 35 Aastra's this is the only one doing this |
| 23:28.22 | trelane | generalhan, busted phone, have it RMA'd |
| 23:28.26 | generalhan | and i have one other one that wont let anyone transfer to it ... but i think thats a dial plan issue |
| 23:28.37 | generalhan | trelane: thats great i cant RMA anymore lol |
| 23:28.42 | trelane | why not? |
| 23:28.42 | generalhan | had them over a year |
| 23:28.43 | Bullseye_Network | Good thing in southfield |
| 23:29.08 | Bullseye_Network | Good thing im NOT in southfield |
| 23:29.35 | *** join/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 23:29.57 | pyrom | chan_sip.c:11988 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 483 <--- why do i get that statement? |
| 23:30.28 | generalhan | i have an RTP setting for the web portal for this phone that has an option of DTMF Method that i can chose between RTP, SIP INFO, or BOTH |
| 23:30.33 | pyrom | All sipura 3000/3102 guides/configs i've found says i should put auth=md5 under [pstn-spa3k] |
| 23:30.33 | generalhan | would that have anything to do with it >? |
| 23:31.49 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 23:31.50 | trelane | generalhan, set it to "Both" and remove the dtmfmode line from your config |
| 23:31.57 | generalhan | and an option that says "Force RFC2833 Out-of-Band DTMF " that i can enable |
| 23:33.13 | generalhan | but it is disabled by default |
| 23:33.30 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 23:33.31 | teknoprep | where do i set default on to call waiting? |
| 23:34.18 | pyrom | D**n this spa3102 is hard to setup, even tried the "online wizard" at voxilla with no use. |
| 23:34.34 | pyrom | even that one says "auth=md5" but asterisk gives me warning on that one. |
| 23:34.44 | pyrom | And the sipura fails to authenticate, so i guess this is the problem. |
| 23:35.10 | pyrom | Anyone got any advice, would realy make my day! |
| 23:38.01 | generalhan | trelane: thanks a lot man ... switching that to both worked perfectly !~ |
| 23:38.05 | generalhan | thanks again for all your help ! |
| 23:39.58 | trelane | pyrom, I don't use sipura phones at all or i'd try to help, you might stick around as many do |
| 23:40.23 | trelane | pasting your connection block to pastebin.ca might point out any obvious errors :) |
| 23:40.23 | pyrom | trelane, thanks ofr the advice, it's a sipura ATA device, linksys sipura 3102. |
| 23:41.58 | trelane | right paste the stuff from sip.conf |
| 23:42.27 | pyrom | trelane, 25 secs. |
| 23:42.30 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:43.26 | pyrom | http://pastebin.ca/145392 |
| 23:46.11 | trelane | I'd have to have my hands on one |
| 23:46.14 | trelane | that config looks right |
| 23:46.19 | trelane | perhaps someone else will be able to assist you |
| 23:46.20 | trelane | sorry |
| 23:46.25 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 23:46.53 | pyrom | Thanks anyway. |
| 23:47.06 | *** part/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 23:48.14 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 23:49.23 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 23:49.44 | TripleFFFF | guys.. with $agi->request['agi_callerid'] /// how can i get the name portion fo the callerid apssed to agi ? |
| 23:57.27 | *** join/#asterisk breno (n=breno@barovia.aureal.com.pe) |
