| 00:00.15 | samoshit | now i'm trying to see if i can DIY my office phones at work |
| 00:00.15 | drmessano | Next week, on to setting up a callcenter? |
| 00:00.15 | samoshit | before i pay someone mad money to do it |
| 00:00.15 | samoshit | drmessano if thats the task presented to me, sure |
| 00:00.19 | drmessano | awesome |
| 00:00.37 | samoshit | jameswf thanks for the link, if you google R8FXX most product images show only 4 ports so i was confused |
| 00:01.01 | jameswf | that is an old revision |
| 00:01.03 | drmessano | The 8 really confused me too.. I almost thought it was a 16 port card |
| 00:01.42 | samoshit | i got a quote from a guy to sell me an avaya 4 phone system for $2800 |
| 00:01.47 | jameswf | we use to do 2 channels to a port then thought it easier to do 8 ports |
| 00:01.48 | samoshit | i said nah i can do this myself |
| 00:02.17 | drmessano | Asterisk is pretty easy for almost everyone |
| 00:02.37 | drmessano | Maybe a few weeks to learn it all |
| 00:02.38 | samoshit | i just have to get the whole wiring thing down in my head, cause right now i'm confused on whats hooked up to what |
| 00:03.06 | jameswf | ~book |
| 00:03.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:03.11 | drmessano | Just remember "red and green" |
| 00:03.15 | jameswf | ^^^^verry helpful |
| 00:03.17 | drmessano | tip and ring |
| 00:03.18 | samoshit | christmas ? |
| 00:03.32 | samoshit | whats red and green |
| 00:03.35 | jameswf | who uses red green |
| 00:03.40 | samoshit | i have that book |
| 00:03.48 | samoshit | in PDF i mean |
| 00:04.00 | samoshit | i didn't read it, i don't like reading |
| 00:04.08 | drmessano | reading sucks |
| 00:04.19 | samoshit | i like pictures. anyone have any pictures of a small office wiring diagram ? |
| 00:04.26 | samoshit | i'm not kidding that would really help me out |
| 00:04.27 | jameswf | blue white/blue 1 orange white/orange 2 green white green 3 brown white/brown 4 |
| 00:04.39 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 00:04.44 | samoshit | rj11 ? |
| 00:04.50 | drmessano | rj12 |
| 00:05.08 | jameswf | its 5 o clock somewhere and somewhere is here |
| 00:05.21 | scooby2 | white orange/orange, white green/blue, white blue/green, white brown/brown |
| 00:05.42 | samoshit | so you guys do phones for a living ? |
| 00:05.53 | drmessano | I usually do blue/green stripe, orange/red stripe, red/blue stripe, green/brown |
| 00:06.27 | drmessano | and crimp the red to ground if you have PoE |
| 00:07.16 | drmessano | I am a refridgeration manager at a fertility clinic.. I don't mess with phones |
| 00:07.35 | samoshit | cool man |
| 00:07.46 | samoshit | maybe i should get sip phones, that would be really hip of me |
| 00:07.51 | drmessano | too much voltage |
| 00:07.55 | drmessano | Yeah |
| 00:08.01 | drmessano | SIP is cool |
| 00:12.41 | russellb | SIP is not cool |
| 00:13.30 | jblack | Is it SIP, or RTP.... |
| 00:13.55 | russellb | VoIP in general is cool |
| 00:13.59 | russellb | SIP is a painful approach to it |
| 00:14.07 | [hC] | Yep. |
| 00:14.14 | russellb | but it's the standard ... so we deal |
| 00:14.17 | [hC] | SCCP came close to being cool, but cisco had to ruin it. |
| 00:14.22 | russellb | heh |
| 00:17.14 | drmessano | samoshit, what sort of phone system do you have currently at work? |
| 00:19.08 | drmessano | samoshit ? |
| 00:24.02 | russellb | drmessano: cans and string :( |
| 00:26.32 | *** join/#asterisk Docfxit (i=ExUser@netblock-208-127-208-174.dslextreme.com) |
| 00:30.32 | drmessano | :( |
| 00:30.38 | drmessano | Dialup dropped out, I guess |
| 00:30.47 | Docfxit | I'm having a problem with the system going down about once a day. Is there someone that knows how to figure out what is happening? |
| 00:36.53 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au) |
| 00:45.40 | tzafrir_laptop | Docfxit, sometimes reading the logs help |
| 00:46.02 | tzafrir_laptop | if you want us to help, you should probably provide more details |
| 00:56.37 | Docfxit | tzafrir_laptop » I'm running AsteriskNow with Asterisk ver. 1.4.17 and a Digium card TDM2400P |
| 00:57.12 | *** join/#asterisk b1shop (n=b1shop@c-24-7-202-70.hsd1.il.comcast.net) |
| 00:57.16 | tzafrir_laptop | anything in /var/log/mesages? |
| 00:57.31 | Docfxit | I'll look. |
| 00:57.40 | tzafrir_laptop | Anything on the console of the machine? |
| 00:58.32 | scooby2 | going down like kernel panic or asterisk crashing? |
| 00:59.08 | Docfxit | I don't have that directory in the root |
| 01:03.52 | Docfxit | I do have a file called messages in /var/log |
| 01:04.04 | scooby2 | thats it |
| 01:04.12 | Docfxit | There is a lot in there. |
| 01:04.46 | Docfxit | Hi scooby2 » I didn't know your message was for me. |
| 01:05.41 | Docfxit | scooby2 » I think asterisk is crashing. I can't make a call out and the incoming calls are not answered. |
| 01:05.59 | scooby2 | sounds like it |
| 01:06.05 | tzafrir_laptop | and what do you do to fix it? |
| 01:06.15 | Docfxit | Should I look for something in the file or send it to someone? |
| 01:06.27 | Docfxit | I re-boot the system |
| 01:06.54 | Docfxit | I haven't found any other way to get it back up and running. |
| 01:07.25 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 01:07.40 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-212-195-200-179.d4.club-internet.fr) |
| 01:08.03 | husimon | so where can I find a guide on how to run asterisk on an openwrt router? |
| 01:08.30 | husimon | or is there a prepackaged openwrt+asterisk i can put on my linksys |
| 01:08.44 | tzafrir_laptop | Docfxit, does the system then respond? |
| 01:09.02 | tzafrir_laptop | (apart from asterisk) - ssh, web interface, etc. |
| 01:09.40 | Docfxit | tzafrir_laptop » yes for another day just fine. |
| 01:10.10 | Docfxit | tzafrir_laptop » I have ssh and the web interface running remotely. |
| 01:10.38 | tzafrir_laptop | I mean: when the system "crashes" |
| 01:11.35 | Docfxit | The web interface works because I can reboot from it. SSH works also. |
| 01:14.35 | Docfxit | I'm reading backwards in the messages file to see what happened just before I re-booted. |
| 01:15.43 | husimon | neat I didn't know there was a chan_skype |
| 01:17.04 | Docfxit | ata_piix: probe of 0000:00:1f.2 failed with error -16 |
| 01:17.10 | Frogzoo | husimon: openwrt has a * package |
| 01:17.36 | Docfxit | tzafrir_laptop » Is that error a problem? |
| 01:17.59 | husimon | frogzoo yeah i'm trying to decide if it's worth it, I have asterisk installed already a location with a pri, I could just register my phones remotely instead of trunking another asterisk box to that pri box with aix |
| 01:18.14 | husimon | sorry iax |
| 01:18.19 | Docfxit | ata: conflict with ide1 |
| 01:18.37 | tzafrir_laptop | Docfxit, is this from system load time? |
| 01:19.41 | Docfxit | tzafrir_laptop » I'm not sure where it started the re-boot. |
| 01:19.57 | Docfxit | I'm trying to figure that out now. |
| 01:23.32 | Docfxit | That's after re-boot so it must be ok. |
| 01:24.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:25.06 | riddlebox | is there a reason that a tdm card with all fxo ports on it would need to be plugged into the systems power before the card would be recognized? |
| 01:26.13 | Docfxit | I can't see anything logged as an error before " shutdown: shutting down for system reboot |
| 01:26.17 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 01:27.19 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:27.55 | marv[work] | In asterisk 1.2, how might I set a variable on another channel in the dialplan. Like I'm in a macro on the outbound leg because of the M argument to dial but want to set a variable on the channel that made the call |
| 01:30.52 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 01:31.02 | tzafrir_laptop | marv[work], for that you'll have to use a global variable |
| 01:33.33 | marv[work] | so you're saying there's no way to directly do what i want? aw |
| 01:33.46 | husimon | marv[work]: just make them globals and it's what you want |
| 01:34.10 | husimon | marv[work]: otherwise use astdb to store your variables |
| 01:34.16 | husimon | marv[work]: then you're fine |
| 01:34.29 | marv[work] | except they stomp on each other. Unless I put the channel name in the anme of the global or something |
| 01:34.45 | husimon | marv[work]: yeah you'd need to change the variable name |
| 01:34.50 | husimon | marv[work]: but take a look at astdb |
| 01:35.19 | marv[work] | plus they're persistent so now i'll have to clean up |
| 01:35.31 | husimon | globals? |
| 01:35.37 | marv[work] | yeah |
| 01:35.47 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 01:35.48 | marv[work] | i'm assuming so is astdb |
| 01:36.06 | husimon | yeah but i assume the variable name isn't going to change |
| 01:36.09 | husimon | so you can just leave it in the db |
| 01:36.20 | husimon | likewise with globals |
| 01:36.36 | husimon | there is no cleanup |
| 01:37.12 | marv[work] | that sounds like a bad idea. not only will i end up with 100's of globals, but when asterisk reuses the channel name the variable will already have a value, and if some code path skips over the step taht would overwrite it... |
| 01:38.42 | husimon | do you seriously use 100's of variables? |
| 01:39.20 | husimon | i guess I don't quite understand your application |
| 01:39.31 | marv[work] | 1 global variable * 100 calls = 100 global variables |
| 01:40.13 | husimon | oh you are using variables as temporary place holders during a cal |
| 01:40.14 | husimon | l |
| 01:40.42 | marv[work] | yeah. what else would you use channel variables for? |
| 01:41.17 | *** join/#asterisk RoyK (n=roy@91.149.31.29) |
| 01:42.32 | husimon | marv[work]: would it at all work to use a global variable to pass the data between the channel variables? |
| 01:42.45 | husimon | the other channel would have to "know" you were passing it and look for it |
| 01:42.48 | husimon | but that seems like one way |
| 01:43.36 | marv[work] | yeah it should work, it's just a bit ugly, and i kind of wonder what the performance will be |
| 01:44.49 | husimon | i guess if you used agi you could probably have more control over the variables |
| 01:45.19 | marv[work] | the only way I know of to set a variable on an arbitrary channel is through the manager api. |
| 01:45.53 | husimon | marv[work]: otherwise you can use astdb variables keyed on the channel and then clean up up after each call ends. |
| 01:49.52 | Docfxit | <PROTECTED> |
| 01:50.35 | Docfxit | Any other ideas on what might be causing Asterisk to stop? |
| 01:50.44 | *** part/#asterisk RoyK (n=roy@91.149.31.29) |
| 01:50.50 | scooby2 | check the asterisk logs? |
| 01:51.08 | scooby2 | usually /var/log/asterisk/messages |
| 01:53.08 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 01:53.19 | obnauticus | Hey, I have a SIP provider that is telling me to add this to my extensions |
| 01:53.20 | obnauticus | exten => _*7X.,1,Switch(user,pass,${EXTEN:2}@sip.cheapcalls.com) |
| 01:53.26 | obnauticus | but there is no Application "Switch" |
| 01:53.40 | obnauticus | So... what does it actually want me to do lol |
| 01:54.37 | *** part/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 01:55.12 | Docfxit | Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. |
| 01:55.30 | Docfxit | (You may ignore this warning if 'ael-parkedcalls' exists in extensions.conf, or is created by another module. I cannot check for those.) |
| 02:00.17 | Docfxit | Under local in Extensions.conf I see include=parkedcalls |
| 02:00.35 | Docfxit | I don't see anything else with parkedcalls in it. |
| 02:00.40 | Docfxit | Is that a problem? |
| 02:01.54 | Docfxit | pbx.c: Unable to register extension 's', priority 2 in 'voicemenu-custom-2', already in use |
| 02:01.54 | Docfxit | [Feb 8 10:56:50] WARNING[6806] pbx.c: Unable to register extension 's', priority 2 in 'voicemenu-custom-3', already in use |
| 02:02.12 | Docfxit | Is this a problem? |
| 02:03.46 | Docfxit | res_smdi.c: No SMDI interfaces were specified to listen on, not starting SDMI listener. |
| 02:03.46 | Docfxit | [Feb 8 10:56:50] WARNING[6806] chan_misdn.c: chan_misdn is not initialized properly, still reloading ? |
| 02:03.46 | Docfxit | [Feb 8 10:56:50] NOTICE[6806] app_playback.c: Reloading say.conf |
| 02:03.46 | Docfxit | [Feb 8 10:56:50] WARNING[3892] frame.c: Cannot disallow unknown format '' |
| 02:03.46 | Docfxit | [Feb 8 10:56:50] WARNING[3892] frame.c: Cannot allow unknown format '' |
| 02:03.46 | Docfxit | [Feb 8 10:56:50] WARNING[3892] frame.c: Cannot disallow unknown format '' |
| 02:04.02 | scooby2 | pastebin |
| 02:04.35 | Docfxit | Where could I find Bin ?¿ |
| 02:04.55 | scooby2 | not supposed to paste to the channel |
| 02:05.05 | Docfxit | Sorry. |
| 02:05.10 | scooby2 | http://www.pastebin.ca/ |
| 02:05.25 | scooby2 | put there then paste the url it gives |
| 02:06.39 | Docfxit | Should I put the entire log for today there just incase there is something that stands out? |
| 02:08.22 | riddlebox | does anyone know why I would need to plug power from the case to a TDM card with 4 fxo ports on it? |
| 02:11.02 | plik | riddlebox: the power is only required for FXS ports i believe |
| 02:11.24 | riddlebox | plik, the card was not recognized until I plugged the power in |
| 02:11.40 | riddlebox | its 4 red cards on it so I know they are fxo cards |
| 02:11.47 | plik | strange... |
| 02:12.44 | plik | no personal experience but I read that FXS required power for ringing, but I tought FXO was OK without... seems not. AFAIK the red modules are FXO yes |
| 02:18.16 | *** join/#asterisk Bandit (n=Bandit@adsl-76-250-139-123.dsl.dytnoh.sbcglobal.net) |
| 02:21.50 | Docfxit | The Messages log for this system up until the re-boot is at http://www.pastebin.ca/896957 |
| 02:21.59 | *** part/#asterisk Bandit (n=Bandit@adsl-76-250-139-123.dsl.dytnoh.sbcglobal.net) |
| 02:23.32 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 02:26.05 | Docfxit | I was told by Digium that the red modules don't need the power plugged in. |
| 02:26.47 | Docfxit | I will be back latter if any is looking at my log. |
| 02:26.55 | Docfxit | Thank all for your help. |
| 02:28.36 | *** join/#asterisk UserReg_CL (n=COB@pc-243-246-214-201.cm.vtr.net) |
| 02:30.08 | UserReg_CL | Buenas noches !!! |
| 02:34.20 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 02:35.25 | riddlebox | what is the absolute cheapest IP phone out there? |
| 02:36.03 | UserReg_CL | need one gsm gateway ... know ? |
| 02:36.07 | husimon | check out voip-supply.com |
| 02:36.47 | riddlebox | Docfxit, thats what I thought but it was seeing that there was something there but ztcfg -vv showed an error, until I plugged the card in |
| 02:37.29 | riddlebox | husimon, I checked there already, I have a client that just wants to have some cheapy phones for some cold calling that they do twice a week |
| 02:37.45 | husimon | so what was the cheapest? |
| 02:38.01 | husimon | riddlebox: you might just consider a headset + softphone |
| 02:38.05 | husimon | if you use it that little. |
| 02:38.16 | UserReg_CL | need one hardware gateway gsm... know one work with * ? |
| 02:38.17 | husimon | which would be about $15 for a headset |
| 02:38.34 | riddlebox | husimon, these people will not have a computer |
| 02:39.22 | husimon | riddlebox: your other option is to use existing analog phones and buy an ata |
| 02:39.47 | riddlebox | husimon, I told them that or an analog card.... |
| 02:39.54 | husimon | but for $44 |
| 02:39.59 | husimon | that gs-101 |
| 02:40.00 | riddlebox | either way they will be about the same price |
| 02:40.10 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 02:42.07 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 02:42.51 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 02:44.51 | jameswf-home | jbot: ping |
| 02:44.52 | jbot | pong |
| 02:55.49 | *** join/#asterisk joez212 (n=jhart@CPE001c101b40b5-CM0018c0d91624.cpe.net.cable.rogers.com) |
| 02:55.51 | joez212 | hello |
| 02:56.25 | joez212 | after i get softphones working is it possible to get voicemail working on the * box itself? |
| 02:57.34 | *** join/#asterisk samoshit (n=msauce@ool-18be2518.dyn.optonline.net) |
| 02:59.00 | samoshit | hi friends. another unrelated to asterisk yet related to telephony question: If I'm looking to buy a phone system for an office with 4 lines, do i have to buy 4-line phones? what exactly does a phone with multiple lines allow you to do ? |
| 02:59.06 | UserReg_CL | hi... need install one gsm gateway (hardware).. know ? |
| 02:59.17 | samoshit | i mean if an office has 16 lines, they don't have to get 16 line phones do they ?? |
| 03:00.32 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 03:01.49 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au) |
| 03:08.31 | scooby2 | samoshit: depends if you use a pbx or not |
| 03:11.47 | samoshit | gotcha |
| 03:11.51 | samoshit | pbx = 1 line is fine |
| 03:12.02 | samoshit | phone company handles switching = 2+ lines |
| 03:12.20 | scooby2 | you can use a pbx on anything from 1 line to hundreds or thousands |
| 03:12.35 | eric2 | call parking comes in handy with > 1 lines |
| 03:13.50 | scooby2 | very |
| 03:14.05 | samoshit | dunno what that is |
| 03:19.11 | *** join/#asterisk AndyGraybeal (n=andy@node61.39.251.72.1dial.com) |
| 03:24.30 | scooby2 | argh i cannot figure out why this ivr will not let me dial more than 1 digit |
| 03:26.19 | samoshit | ivr ? |
| 03:26.32 | *** join/#asterisk andresmujica (n=andresmu@190.25.96.116) |
| 03:28.49 | andresmujica | <PROTECTED> |
| 03:30.59 | scooby2 | ~book |
| 03:31.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 03:32.51 | UserReg_CL | ~gsm |
| 03:32.51 | jbot | rumour has it, gsm is a codec, operating at approx 13kbps up/down. |
| 03:42.41 | *** join/#asterisk jblack (n=jblack@pool-71-173-53-239.sctnpa.east.verizon.net) |
| 03:42.55 | jblack | drmessano: drmessano: drmessano: !!! |
| 03:45.51 | jblack | drmessano: I found a video of mrdigital! |
| 03:46.24 | jameswf-home | max headroom |
| 03:46.44 | jblack | jameswf-home: http://youtube.com/watch?v=rRC971jtvEg |
| 03:50.54 | jameswf-home | holy clay aken batman |
| 03:51.45 | jameswf-home | my brother had the rick springfield hair forever |
| 03:53.11 | jblack | You're lucky you got rickrolled. You don't want to see what getting montgomeried is like |
| 03:54.34 | jameswf-home | lmao http://youtube.com/watch?v=ULgwbvj768E |
| 03:54.36 | Strom_C | I prefer to rickroll people at the nightclub with the 12" single ;) |
| 03:55.11 | jblack | Strom_C: Like this? : http://youtube.com/watch?v=ULgwbvj768E&feature=related |
| 03:55.45 | jameswf-home | lol its just like its just like a black jesse jackson |
| 03:55.53 | jameswf-home | s/black/fat/ |
| 03:56.01 | jblack | It's just like! a Mini-Mall! |
| 03:56.07 | Strom_C | no, like this |
| 03:56.08 | Strom_C | http://www.discogs.com/viewimages?what=R&obid=221824 |
| 03:56.57 | jblack | I emailed out three rickrolls tonight. One of them promised a video of RMS getting shaved bald. |
| 03:56.57 | jameswf-home | i m going to be singing that tomorrow at the flea market |
| 03:57.23 | jblack | jameswf-home: No! Get one of those post-70s boom boxes that take the D cells, and blast it! |
| 03:58.31 | jameswf-home | do anything besides vibrators take D batteries anymore |
| 04:00.36 | jblack | vibrators take D batteries? |
| 04:00.40 | jblack | No wonder I'm single |
| 04:03.00 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 04:05.50 | jameswf-home | hmmmmm |
| 04:09.40 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 04:10.32 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 04:12.11 | scooby2 | finally caught my kernel panic. Definitely appears zaptel related. |
| 04:12.13 | scooby2 | http://www.pastebin.ca/897040 |
| 04:18.02 | *** join/#asterisk samoshit (n=msauce@ool-18be2518.dyn.optonline.net) |
| 04:18.46 | andresmujica | hi, how can i configure a sip trunk to use digest authentication???? |
| 04:19.02 | jameswf-home | spam http://www.youtube.com/watch?v=wZ7YedEopp4&feature=related |
| 04:21.32 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 04:23.44 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 04:33.33 | *** join/#asterisk techie (n=techie@adsl-76-214-31-194.dsl.lsan03.sbcglobal.net) |
| 04:39.39 | tuxfoo | I think its AUTH=MD5 |
| 04:40.46 | BBHoss | yeah, its usually lowercase though, but it might not matter |
| 04:51.42 | *** join/#asterisk AJayMN (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 04:52.07 | AJayMN | Anyone use a Netgear WGR613VAL with asterisk? |
| 04:57.36 | *** join/#asterisk erojasv (n=erojasv@201.240.194.210) |
| 05:01.07 | andresmujica | which option defines the digest username.. i'm getting up to the digest challenge, but it seems to be using 2 different usernames, one for registering in order to do the digest auth and the other one to register the extension... something like that... |
| 05:03.48 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 05:06.26 | tuxfoo | do you have a registration as well? register => 1234':'password@mysipprovider.com |
| 05:07.17 | tuxfoo | remove the tick marks as the : and the p translated to :p |
| 05:13.25 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 05:16.28 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 05:18.23 | andresmujica | yeap. when i put the register screen i'm getting the register, 401, www-auth, but after that i get an unauthorized, comparing the cpature from the softphone proveided by my ITSP i can see that i'm sending the wrong digest username... |
| 05:19.10 | andresmujica | with the softphone the digest username is 1xxxxxx but asterisk is sending the 0001xxxx. The problem lies that the softphone send for the first register the 0001xxx username.... |
| 05:22.56 | styelz | my TISP requires me to register with a line like, DID@domain : pass : DID@sip.host.com/DID |
| 05:23.45 | andresmujica | hmm i saw something like that at the wiki.. gonna try it. |
| 05:31.40 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
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| 05:38.05 | andresmujica | ok, i'm close.. i'm getting the 401 unauthorized + register.. and i'm sending the right usernames. so i'm missing the 200 - ok... i'm getting a 401 - unauthorized and the sip show registry shows Auth Sent. ... |
| 05:39.13 | *** join/#asterisk PepOSX (n=angeldav@201.243.76.220) |
| 05:43.05 | Docfxit | Does anyone know of a good file editor I can load on the system? |
| 05:47.57 | andresmujica | the itsp can validate the user agent??? |
| 05:54.15 | styelz | Docfxit: what system? |
| 05:54.45 | styelz | X or console? |
| 05:56.18 | styelz | aee, nano, pico, vi |
| 06:02.07 | Frogzoo | Docfxit: vi for admin work I think |
| 06:08.05 | Docfxit | Frogzoo » Thanks. I'll look for it. |
| 06:12.26 | sbingner | vim *nod* |
| 06:17.17 | styelz | after you spent an hour learning vi... |
| 06:17.39 | styelz | tiz good |
| 06:20.20 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
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| 06:31.00 | Docfxit | styelz » Asterisk |
| 06:31.27 | Docfxit | Isn't VI and VIM two different editors? |
| 06:31.36 | styelz | vim has x support |
| 06:31.51 | Docfxit | What is x support? |
| 06:31.59 | styelz | gnome etc.. |
| 06:32.17 | styelz | you prob dont have it |
| 06:32.40 | Docfxit | Okay |
| 06:32.48 | styelz | centos or debian ? |
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| 06:32.59 | Docfxit | rpath |
| 06:33.06 | styelz | ok i dont know |
| 06:33.55 | Docfxit | I've been trying to find the download for VI. So far all I found is a cheat sheet. |
| 06:34.55 | Corydon76-dig | Docfxit: vi or vim? |
| 06:35.11 | styelz | it should be installed |
| 06:35.17 | Frogzoo | Docfxit: vi is standard on any linux |
| 06:35.20 | styelz | type vi, or nano or pico |
| 06:35.48 | Corydon76-dig | The original vi is not available anymore. You might be able to find nvi if you REALLY want a tiny vi editor |
| 06:36.06 | Corydon76-dig | but most vi users nowadays use vim |
| 06:36.12 | drmessano | I can't yum yum |
| 06:36.17 | drmessano | :( |
| 06:36.22 | styelz | aww |
| 06:36.33 | styelz | bum bum |
| 06:37.12 | Docfxit | Oh good that's why I can't find it. I'll look for vim. |
| 06:37.22 | Docfxit | I have nano and don't like it. |
| 06:37.31 | styelz | dont blame you |
| 06:38.06 | Corydon76-dig | The original vi was by Bill Joy and isn't even available on Sun machines anymore... |
| 06:38.56 | mosty | i like nvi |
| 06:39.41 | *** join/#asterisk PepOSX (n=angeldav@201.243.76.220) |
| 06:40.17 | styelz | whats the diff? |
| 06:40.23 | styelz | looks like vi, smells like vi |
| 06:41.23 | drmessano | yum -y install notepad ? |
| 06:41.30 | styelz | heh |
| 06:42.00 | styelz | i like this http://linux.about.com/cs/linux101/g/aee.htm |
| 06:42.12 | styelz | er |
| 06:42.38 | styelz | i mean this |
| 06:42.38 | styelz | http://www.users.qwest.net/~hmahon/ |
| 06:43.05 | styelz | heh |
| 06:43.26 | styelz | "Intended to be usable with little or no instruction." |
| 06:46.48 | mosty | Docfxit, it's unusual for a linux distribution to not have some variant of vi available |
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| 06:51.19 | *** mode/#asterisk [+o russellb] by ChanServ |
| 06:51.41 | Docfxit | styelz » I'll try it. Tx |
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| 08:11.42 | obnauticus | Why shouldn't exten => _9.,1,Dial(SIP/${EXTEN:1}@ast,30,r) |
| 08:11.43 | obnauticus | work |
| 08:11.48 | obnauticus | if there is an [ast] context in sip.conf |
| 08:11.50 | obnauticus | properly configured. |
| 08:13.05 | obnauticus | Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 08:13.20 | jwh | it isn't registered? |
| 08:13.55 | obnauticus | lemme see |
| 08:14.06 | obnauticus | no, it is registered |
| 08:14.24 | obnauticus | wait |
| 08:14.43 | obnauticus | the register =; what correlation does that have with sections in sip.conf |
| 08:16.31 | *** join/#asterisk worgil (n=worgil@88.231.37.55) |
| 08:18.37 | worgil | i have installed asterisk on ubuntu back to DSL line i can using local normaly but anyone call from other line or other companies i cannot hear anything, i cannot sending noise, can anyone have idea ? |
| 08:20.29 | mosty | obnauticus, register is so that the remote host knows where your asterisk box is if it has a dynamic ip |
| 08:20.36 | mosty | for incoming calls |
| 08:20.37 | Frogzoo | obnauticus: you need to setup the phone in sip.conf |
| 08:21.16 | obnauticus | Frogzoo I did |
| 08:21.17 | obnauticus | :\ |
| 08:21.22 | obnauticus | memali 98 Registered Sat, 09 Feb 2008 03:20:42 |
| 08:21.28 | obnauticus | I'm just trying to configure outbound right now |
| 08:21.38 | obnauticus | exten => _9.,1,Dial(SIP/${EXTEN:1}@memali,30,r) |
| 08:21.41 | obnauticus | that's in the dial plan |
| 08:21.52 | obnauticus | and in sip.conf i have configuration for a peer |
| 08:22.40 | mosty | worgil, set externip and localnet in sip.conf? look those up on the wiki |
| 08:22.57 | obnauticus | and also under the [general] section in sip.conf |
| 08:22.58 | obnauticus | I have register => memali:******@sip.********.com |
| 08:24.05 | obnauticus | so when I call _9. it should dial SIP/${EXTEN:1}@memali,30,r |
| 08:24.13 | obnauticus | Instead it is saying no route to destination |
| 08:24.36 | obnauticus | <PROTECTED> |
| 08:25.35 | worgil | mosty, how can i do it * |
| 08:25.39 | worgil | ? |
| 08:30.37 | obnauticus | mosty do you know? |
| 08:32.16 | scooby2 | Invalid extension '1', but no rule 'i' in context 'did' |
| 08:32.38 | scooby2 | i have an 'i' rule but i'm not in did anymore |
| 08:34.10 | scooby2 | i am coming from the did context |
| 08:42.31 | hmodes | wow, digium switchvox? *blinks* |
| 08:42.41 | hmodes | i did not get that memo |
| 08:57.12 | *** join/#asterisk sergey (n=sergey@213.24.100.5) |
| 09:05.54 | JerJer | live in the now man |
| 09:06.51 | jblack | Now? |
| 09:06.54 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 09:07.47 | styelz | no now |
| 09:08.04 | jblack | There's no such thing as now. By the time you hear it, taste it, see it, feel it, it's already happened. |
| 09:08.11 | styelz | yes |
| 09:08.12 | jblack | Thusly, we live in the past. |
| 09:08.43 | styelz | living in the now is holding back when you need a piss |
| 09:11.41 | mosty | obnauticus, put qualify=yes in the sip peer definition, do a sip reload, then pastebin the output of "sip show peers" |
| 09:16.10 | worgil | mosty how can i solve noise problem ? |
| 09:30.39 | mosty | worgil, noise? what kind of noise, and what kind of channel is it? |
| 09:36.45 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:37.03 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 09:38.21 | Frogzoo | did now happen already? |
| 09:39.13 | worgil | mosty, i cam calling other number ex. 1011 and speaking but not caming or going sound |
| 09:40.38 | mosty | worgil, no sound at all in either direction? check port forwarding on your router. also check that you have externip and localnet set in sip.conf if you're behind nat. do you control the server too? |
| 09:41.16 | worgil | yes |
| 09:41.26 | worgil | i did DMZ |
| 09:41.33 | worgil | but not working still |
| 09:42.33 | worgil | localnet=192.168.0.0/255.255.255.0 |
| 09:44.36 | worgil | what is problem for noise ? |
| 09:46.12 | *** join/#asterisk steliosk (n=Stelios@athedsl-314352.home.otenet.gr) |
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| 09:50.33 | mosty | do you have the same codec set on the server? can you run a packet sniffer like tshark on the server? |
| 09:51.06 | worgil | i have codec |
| 09:51.11 | worgil | ulaw and alaw |
| 09:53.17 | JT | both at the same time? |
| 09:54.07 | worgil | which i need them |
| 09:54.14 | JT | ? |
| 09:54.28 | worgil | disallow = all |
| 09:54.32 | worgil | allow = alaw |
| 09:58.30 | worgil | mosty, have i problem abotu tshark ? |
| 09:59.05 | mosty | i'm sorry, i don't understand the question |
| 10:07.35 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 10:16.50 | mvanbaak | mornin all |
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| 10:56.01 | jblack | I am tired of perpetual motion machines. |
| 10:56.27 | Frogzoo | yes, they do go on forever |
| 10:58.13 | Frogzoo | any robust way to dial into an IVR service (a calling card company), and then dial in the destination number? could just put in a delay, but that's a bit... |
| 10:58.52 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 11:00.21 | *** join/#asterisk WorGiL (n=worgil@88.231.35.28) |
| 11:00.39 | Frogzoo | another way to ask the same question, is there a way to bridge 2 calls through the dialplan, without using Dial ? |
| 11:01.14 | *** join/#asterisk admin0 (n=admin@116.90.228.34) |
| 11:01.45 | jblack | I believe there's a Bridge app |
| 11:01.55 | jblack | show application bridge |
| 11:02.02 | Frogzoo | ah, I will check, or if all else fails, read the code :) |
| 11:02.56 | Frogzoo | cry - it's not there in 1.4 |
| 11:04.15 | Frogzoo | ooh - ExternalIVR looks promising... |
| 11:19.54 | *** join/#asterisk lohapuk (n=lohapuk@82-69-54-51.dsl.in-addr.zen.co.uk) |
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| 11:22.25 | *** join/#asterisk UserReg_CL (n=COB@pc-243-246-214-201.cm.vtr.net) |
| 11:22.26 | lohapuk | hi i have a question i have a ciso 7960 and have connected it to asterisk it can recieve calls fine and used to be able to make calls to other extension but not i cant make calls to other extensions at all but can recieve call fine |
| 11:22.52 | lohapuk | i have looked in the forums and mailing lists and cant find anything just wondering if anyone has an idea |
| 11:24.02 | UserReg_CL | hi... need one gateway gsm (hardware) ... know one work with * ?? |
| 11:24.57 | tzafrir_home | There are several gsm PCI cards, even |
| 11:25.16 | mvanbaak | gsm pci cards, voiceblue, chan_mobile |
| 11:25.25 | tzafrir_home | or you can get a gsm->FXO or gsm->SIP gateway |
| 11:25.45 | tzafrir_home | I don't have experince with any of them |
| 11:25.54 | UserReg_CL | mmm |
| 11:27.11 | mvanbaak | we have great success with the voiceblue |
| 11:30.57 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 11:31.53 | UserReg_CL | voiveblue: work with * ? |
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| 11:34.42 | lohapuk | that is the debug output for the call if anyone is interested |
| 11:34.42 | lohapuk | http://pastebin.com/m3724d2d8 |
| 11:35.00 | lohapuk | i starting dialing but never connects but the phone can recieve calls fine |
| 11:47.45 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 12:14.02 | *** join/#asterisk Whoopie (n=Whoopie@unaffiliated/whoopie) |
| 12:15.03 | Whoopie | Hi, I built asterisk staticly. Now, when loading it, I get: |
| 12:15.06 | Whoopie | loader.c:376 load_dynamic_module: Module 'res_musiconhold.so' did not register itself during load |
| 12:15.06 | Whoopie | : can't resolve symbol 'ast_module_unregister' |
| 12:15.18 | Whoopie | any ideas? |
| 12:16.32 | mvanbaak | is there a way to check if a sip device is registered from the dialplan |
| 12:18.56 | tzafrir_home | Whoopie, asterisk will still try to load modules from /usr/lib/asterisk/modules as well |
| 12:19.16 | tzafrir_home | in asterisk.conf set the modules dir to something else, I guess |
| 12:19.21 | Whoopie | tzafrir_home: ok, and they are there. |
| 12:19.34 | Whoopie | ah, all modules are still loaded? |
| 12:19.38 | mvanbaak | or move the modules out of the way |
| 12:19.49 | tzafrir_home | will also work |
| 12:19.58 | tzafrir_home | I prefer to just edit asterisk.conf |
| 12:20.20 | mvanbaak | astmoddir=/dev/null |
| 12:21.40 | Whoopie | well, but how can I build the asterisk binary staticly, and leave the structure as is. |
| 12:21.58 | Whoopie | it's just that I don't want to have the library dependencies |
| 12:22.52 | mvanbaak | hhmm, if I use this: ChanIsAvail(SIP/thinkpad) it returns that it's available even if ekiga on my thinkpad is not registered |
| 12:23.10 | mvanbaak | thinkpad/thinkpad (Unspecified) D 0 UNKNOWN |
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| 12:24.29 | mvanbaak | I want to run a dundi query when the thinkpad is not registered to the local pbx |
| 12:24.38 | mvanbaak | it can be at 4 machines |
| 12:24.45 | mvanbaak | how can I do this ? |
| 12:29.54 | mvanbaak | hhmm, I think I can use the DB functions to check SIP/Registry/${DEVICE} |
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| 13:11.49 | Frogzoo | is 1.6b reasonably stable?? |
| 13:12.48 | mvanbaak | it is |
| 13:13.14 | mvanbaak | it IS beta |
| 13:14.06 | Frogzoo | i know, but there's betas (we think it's ready to release) and betas (this is full of bugs we haven't fixed yet) |
| 13:14.49 | mvanbaak | a beta is never "we think it's ready to be released" |
| 13:14.54 | mvanbaak | only -rc's are |
| 13:16.49 | riddlebox | how long has the gui been in beta? |
| 13:17.01 | mvanbaak | I have no idea |
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| 13:25.08 | Mw3 | is anybody using 1.6.0-beta2 around here? |
| 13:25.25 | *** join/#asterisk RoyK (n=roy@ip-223-13-149-91.dialup.ice.no) |
| 13:25.50 | mvanbaak | Mw3: I'm running trunk :) |
| 13:26.13 | Mw3 | dtmf does not work with my cisco 7912 (sip image), works fine with 1.4.18 |
| 13:26.20 | Mw3 | did you have any issues with sip and dtmf? |
| 13:30.12 | mvanbaak | nope |
| 13:30.19 | mvanbaak | but I'm not runnign 1.6 |
| 13:42.04 | *** join/#asterisk AndyGraybeal (n=andy@node46.39.251.72.1dial.com) |
| 13:42.23 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
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| 13:53.03 | Frogzoo | suggestions to best way to dial a calling card number, then dial the destination once logged in? |
| 13:57.51 | [TK]D-Fender | Frogzoo, "core show application dial" + ("D" or "M") |
| 13:58.19 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:59.15 | Frogzoo | [TK]D-Fender: I'll check out those options, thx |
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| 14:03.59 | eric2 | is dundi the best route to go for failover? |
| 14:10.10 | BBHoss | dundi is broken in 1.6b2 |
| 14:10.27 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 14:14.59 | mvanbaak | hey [TK]D-Fender |
| 14:18.18 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 14:18.44 | BBHoss | eric2, dundi works well as a load-balancer, so you could probably use it for failover |
| 14:25.19 | *** join/#asterisk shumway (n=unf@90.4.146.251) |
| 14:25.48 | shumway | greetings |
| 14:28.16 | shumway | !j openser |
| 14:28.18 | shumway | heh |
| 14:28.27 | shumway | not used to this euro keyboard yet |
| 14:28.31 | eric2 | what's the suggested way of setting up failover? |
| 14:35.51 | BBHoss | start here: http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf |
| 14:37.59 | eric2 | I found that paper earlier today :) |
| 14:38.32 | mvanbaak | it's a great piece of documentation |
| 14:42.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:46.06 | *** join/#asterisk e}{istence (n=uvedoble@140.Red-80-38-216.staticIP.rima-tde.net) |
| 14:56.29 | e}{istence | hi |
| 14:56.31 | e}{istence | <PROTECTED> |
| 15:02.33 | Corydon76-dig | That doesn't make any sense. |
| 15:04.52 | e}{istence | i think i have problem with the config of gain, |
| 15:05.49 | e}{istence | i have one TDM2403E with hardware echocancel, and i use a Gw For landlines and a analog Gw for cells calls |
| 15:06.15 | e}{istence | but i know that the quality can be performed |
| 15:06.31 | e}{istence | but i don't know how i can do it |
| 15:06.37 | Corydon76-dig | I think you mean "improved" |
| 15:08.09 | Corydon76-dig | Have you tried running fxotune on the FXO modules yet? |
| 15:08.39 | e}{istence | sorry Corydon76-dig, yes the word is improve |
| 15:08.52 | e}{istence | i'm sorry i from spain and my english as you can see, is bad |
| 15:09.09 | e}{istence | i don't know fxotune |
| 15:09.47 | Corydon76-dig | Look in the zaptel build directory |
| 15:10.21 | Corydon76-dig | It's a utility for improving quality by pretuning the echo cancellation coefficients |
| 15:11.26 | e}{istence | thank you corydon i'm going to try it |
| 15:12.20 | mvanbaak | Corydon76-dig: you know of a way to detect wether a phone is registered or not from the dialplan |
| 15:12.54 | *** join/#asterisk janinge (i=j@ninge.net) |
| 15:12.59 | mvanbaak | I have phones that can be registered on 1 of the 4 boxen that are scattered around locations of this company |
| 15:13.02 | *** part/#asterisk janinge (i=j@ninge.net) |
| 15:13.12 | mvanbaak | I use dundi to find out where they are |
| 15:13.35 | mvanbaak | but first step would be to check wether the phone is registered locally so I can dial(sip/phone) instead of running dundi |
| 15:14.23 | Corydon76-dig | mvanbaak: What about ChanIsAvail or whatever it's called? |
| 15:14.38 | mvanbaak | chanisavail always returns the phone as avail |
| 15:14.42 | mvanbaak | even if it's not registered |
| 15:15.10 | mvanbaak | sip entries are in sip.conf |
| 15:15.22 | mvanbaak | chanisavail works if the sip entries are in realtime |
| 15:15.30 | Corydon76-dig | mvanbaak: It also sets an AVAILSTATUS |
| 15:15.36 | mvanbaak | yup |
| 15:15.39 | mvanbaak | 0 all the time |
| 15:15.47 | mvanbaak | registered or not, it's 0 |
| 15:16.15 | Corydon76-dig | 0 is unknown |
| 15:16.41 | Corydon76-dig | Do you have hints for all your phones? |
| 15:16.47 | mvanbaak | no |
| 15:16.51 | mvanbaak | none have hints |
| 15:16.53 | Corydon76-dig | Try adding hints |
| 15:17.01 | Corydon76-dig | and then check AVAILSTATUS |
| 15:17.08 | mvanbaak | ouch |
| 15:17.20 | mvanbaak | 500 hints |
| 15:17.25 | mvanbaak | is asterisk going to like that ? |
| 15:17.42 | Corydon76-dig | Sure, it's fine... hints are all kept in memory, in pretty small structures |
| 15:18.03 | mvanbaak | hhmm, you're right |
| 15:18.12 | mvanbaak | and since I wont subscribe any phone to them.... |
| 15:18.15 | mvanbaak | it should be fine |
| 15:18.22 | Corydon76-dig | but then you have a tracking structure for status |
| 15:18.57 | mvanbaak | this is going to be a nice setup |
| 15:19.03 | mvanbaak | 4 boxen on 4 locations |
| 15:19.19 | mvanbaak | nec-philips dect handsets that go from one location to the other |
| 15:20.12 | mvanbaak | ok, let's try the hint stuff |
| 15:23.27 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 15:24.45 | nextime | Hello. I notice that the latest zaptel 1.4 released can't compile against kernel 2.6.24. I know that this is fixed in the latest svn branches for 1.4. When the tarball with those fix will be released? |
| 15:24.56 | _ShrikE | Does anyone know what the best g.729 codec version is for a quad core xeon? |
| 15:25.16 | e}{istence | Corydon76-dig what is the difference fxotune -i 4 and fxotune -i 5 ? |
| 15:25.21 | BBHoss | _ShrikE, are you running X86_64? |
| 15:25.29 | _ShrikE | no 32. |
| 15:25.50 | mvanbaak | hhmm |
| 15:26.04 | Corydon76-dig | e}{istence: shouldn't matter |
| 15:26.12 | mvanbaak | there's a huge difference between 1.2 and trunk when it comes to ChanIsAvail |
| 15:26.16 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 15:26.22 | Corydon76-dig | e}{istence: it's the DTMF tone that's used as a sample |
| 15:26.36 | mvanbaak | in trunk it works fine even without the hint |
| 15:27.14 | e}{istence | ok , i put fxotune -i 4 , and i can see that write, Tuning module dev/zap1 and the same for the other modules |
| 15:27.17 | e}{istence | it's ok ? |
| 15:32.16 | Corydon76-dig | e}{istence: I don't know, did it help? |
| 15:34.03 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:34.03 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:34.29 | e}{istence | Corydon76-dig now, i can't call |
| 15:34.55 | e}{istence | Call from '100004' to extension '653829104' rejected because extension not found. |
| 15:36.09 | Corydon76-dig | e}{istence: that's not due to fxotune, that's something else |
| 15:36.23 | e}{istence | yes but i only do that |
| 15:36.36 | e}{istence | and before all run well, and after not |
| 15:37.40 | *** join/#asterisk worgil (i=worgil@88.252.186.63) |
| 15:43.37 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 15:44.07 | ZPertee | is there any way that I can do chanspy for a definite period of time? |
| 15:44.17 | ZPertee | rather I mean zapbarge |
| 15:50.20 | mvanbaak | gheh, 1.2 is funny |
| 15:51.11 | mvanbaak | exten => s,n,Verbose(1,ChanIsAvail returned with AVAILCHAN:${AVAILCHAN}, AVAILORIGCHAN:${AVAILORIGCHAN}, AVAILSTATUS:${AVAILSTATUS}) |
| 15:51.23 | mvanbaak | it transforms it into: |
| 15:51.25 | mvanbaak | <PROTECTED> |
| 15:51.55 | *** join/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 15:52.03 | *** join/#asterisk KeltusEx (n=Keltus@nat/yahoo/x-8a1ed5114cfd0d92) |
| 15:57.02 | iamhrh | Hello! I'm setting up a new asterisk installation, and trying to build paging zones. I've read through most of the information I could find on voip-info.org, and I'm curious if anyone here has had experience with A) using multiple sound cards or B) using the budgettone phone and soldering a wire to it |
| 15:59.28 | eric2 | when a DID is dialed, is there a way to send it to some context while setting an additional variable? I'd like to use GoTo but it won't accept additional data |
| 16:04.36 | *** join/#asterisk hmodes (n=hmodes@2001:470:1f04:59:0:0:0:2) |
| 16:10.09 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
| 16:10.13 | Datax | hi all |
| 16:11.02 | Datax | I'm new to asterisk and have what is probably a dumb question. what is zaptel exactly ? I'm planning on configuring a SIP+SCCP asterisk server. Do I need zaptel ? |
| 16:11.28 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:11.32 | iamhrh | zaptel is the interface for standard analog lines |
| 16:11.43 | iamhrh | well, driver really |
| 16:11.50 | Datax | ok so as long as I don't use analog lines I don't need it ? |
| 16:11.55 | iamhrh | so strictly speaking, no you won't need it |
| 16:11.59 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:11.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:12.07 | iamhrh | there is a timing module that you may want in there I think |
| 16:12.09 | iamhrh | one sec |
| 16:13.18 | iamhrh | yeah, there is a module called ztdummy that you'll want to make sure you have |
| 16:13.32 | Datax | what does it do ? you mentioned timing |
| 16:14.19 | iamhrh | "Thew ztfummy module is an interface to a device that provides timing, which in turn allows asterisk to provide timing to various applications and functions that reqwuire it" |
| 16:14.24 | *** join/#asterisk redback (n=kieran@mail.datadream.co.uk) |
| 16:14.34 | iamhrh | quoting from an orielly book |
| 16:14.34 | Datax | ok thanks |
| 16:14.52 | *** join/#asterisk IPGHOST (n=IPGHOST@203.215.176.186) |
| 16:14.55 | iamhrh | so I don't know exactly other than my system seems to work, and I have it :-) |
| 16:14.58 | Datax | they should use a spellchecking unless you just rewrote that ;) |
| 16:15.12 | iamhrh | heh I just typed it while reading over my shoulder |
| 16:15.15 | Datax | ;) |
| 16:15.16 | iamhrh | didn't spell check |
| 16:15.24 | Datax | thanks for that info |
| 16:15.28 | Datax | seems clear enough :) |
| 16:15.32 | iamhrh | sure thing |
| 16:15.51 | iamhrh | chatting about these things is the only way to get them figured out |
| 16:15.53 | Datax | I'm thinking of setting up a network of asterisk servers for my family |
| 16:15.57 | redback | Hi - for some reason when I call a certain extension the first playback sometimes starts partway through the file as though the call connected before it actually did - is there a workaround to that |
| 16:16.12 | iamhrh | I almost jumped out of my chair this morning when I finally got the cisco 7960s I just got working |
| 16:16.51 | Datax | ahh you're going to help me again ! I have a 7961 with the default SCCP firmware on it. Did you configure SCCP or did you load the SIP firmware ? |
| 16:17.05 | iamhrh | loaded the SIP firmware |
| 16:17.16 | Datax | where did you get it from ? |
| 16:17.18 | iamhrh | had to use the P0S3-06-3-00 |
| 16:17.24 | iamhrh | first |
| 16:17.26 | Datax | don't know what that is |
| 16:17.35 | iamhrh | heh, its the firmware revision I used |
| 16:17.46 | Datax | don't know if the 7960 firmware will be compatible with a 7961 either :/ |
| 16:17.47 | iamhrh | this helped me: http://www.cisco.com/warp/public/788/voip/handset_to_sip.html |
| 16:17.50 | Datax | I'll check the cisco site |
| 16:17.54 | iamhrh | ooh good point |
| 16:17.57 | lmadsen | I love when people quote the oreilly book :D |
| 16:18.16 | iamhrh | its the whole reason I have anything working at all :-D |
| 16:18.29 | iamhrh | I think i read it cover to cover in one weekend |
| 16:18.52 | iamhrh | compared to sifting through what info is on voip-info its a godsend |
| 16:18.58 | Datax | I also have a Cisco wifi 7961 handset |
| 16:19.12 | Datax | but last time I checked there wasn't a SIP firmware for it |
| 16:21.45 | redback | Has anyone else experienced problems with Playback starting before the call is connected? |
| 16:22.12 | iamhrh | did you Answer() the channel first? |
| 16:22.39 | lmadsen | redback: Playback(silence/1&file-you-want-to-play) |
| 16:22.51 | redback | iamhrh: for sure |
| 16:23.05 | lmadsen | or add a Wait(1) before the Playback() after you execute Answer() |
| 16:23.10 | redback | lmadsen: I tried silence/2 and it still does |
| 16:23.25 | redback | lmadsen: that I havent tried |
| 16:23.30 | lmadsen | then there is something else possibly going on with your phone then |
| 16:23.41 | lmadsen | because I've done that on dozens of systems and that's all you need to do |
| 16:25.10 | redback | lmadsen: the wait worked |
| 16:25.17 | redback | weird - but thankyou |
| 16:27.10 | ZPertee | is there some way to put a person on hold and then continue in the dial plan with other steps. Basically what I want to do is if a call comes in on Zap/1 I want to put them on hold and then I have an overhead paging system connected to zap/7 which I want to connect to alert me of the incoming call. |
| 16:28.11 | ZPertee | I understand how to do put someone on hold and I understand how to use paging system just not sure hwo to put someone on hold and then continue farther down in the dial plan |
| 16:33.48 | ZPertee | looks like after a little more research the answer is to park the call instead of just running MusicOnHold() application |
| 16:40.59 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
| 16:41.08 | drako | why my asterisk is not compiling app_meetme |
| 16:42.03 | Datax | drako: any error messages ? |
| 16:42.18 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 16:42.19 | drako | Datax, no well just no app_meetme |
| 16:42.20 | Datax | you might not have some of the dependencies |
| 16:42.22 | ice_croft | hi all |
| 16:42.25 | ice_croft | who can say |
| 16:42.51 | ice_croft | what kinds of fxo cards are supported by freebsd and zaptel? |
| 16:44.19 | drako | Datax, hhmm |
| 16:45.26 | lmadsen | drako: you have no installed zaptel |
| 16:45.40 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 16:45.59 | lmadsen | you have to install zaptel first (at least ztdummy module) in order for asterisk to compile in the features that require the timing source, such as app_meetme |
| 16:46.29 | lmadsen | cd zaptel && ./configure && make install && cd ../asterisk && ./configure && make install |
| 16:46.45 | lmadsen | (use make menuselect to select the modules you want / don't want) |
| 16:48.38 | drako | lmadsen, i do have it installed |
| 16:49.06 | lmadsen | it needs to be installed before you run ./configure in your asterisk directory, or it won't detect the modules as being installed, and thus, won't compile app_meetme |
| 16:49.18 | ice_croft | hmm |
| 16:49.27 | lmadsen | after the ./configure, make sure you run 'make menuselect' and that app_meetme is selected in the Applications menu |
| 16:49.28 | ice_croft | so no freebsd support, right? |
| 16:49.33 | lmadsen | just use linux |
| 16:49.47 | lmadsen | freebsd does work for some people, but you need to know more |
| 16:49.53 | lmadsen | (i.e. it's not as straight forward) |
| 16:50.02 | lmadsen | but some of the developers do use it |
| 16:50.07 | ice_croft | well it's all about drivers |
| 16:50.16 | lmadsen | ahhh... driver support -- none. |
| 16:50.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:50.26 | ice_croft | no drivers for digium at all |
| 16:50.43 | *** join/#asterisk dream_th (n=dream_th@91.187.123.66) |
| 16:50.52 | lmadsen | only linux is supported |
| 16:51.11 | ice_croft | that's no good. :( |
| 16:51.12 | lmadsen | you may or may not find some driver support from a third party... but I don't know for sure |
| 16:51.38 | lmadsen | asterisk is primary developed on linux... other OS's are supported sparsely by other developers |
| 16:52.33 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 16:54.03 | dream_th | is it possible to restrict outbound/routes that can be used only by specific extension? |
| 16:54.13 | dream_th | routes or trunk\ |
| 16:54.36 | ice_croft | that's not cos asterisk, but hardware drivers r absent |
| 16:55.12 | Docfxit | What is the fastest PC that Asterisk can run on? |
| 16:55.39 | dream_th | it should be any Pc Docfxit |
| 16:55.43 | lmadsen | Docfxit: that's an odd question.... it can run on the fastest available |
| 16:56.22 | Docfxit | Can it run on a Dule core PC. |
| 16:56.29 | lmadsen | dream_th: you would control that via the dialplan |
| 16:56.30 | lmadsen | Docfxit: dual, quad, etc.. |
| 16:56.33 | lmadsen | yes -- asterisk is multithreaded |
| 16:56.55 | dream_th | lmadsen: can you give me some example how would i do that (a simple one) |
| 16:57.30 | Docfxit | I'm having trouble with Asterisk crashing about once a day. I'm thinking it may be on a box that is too fast for the Digium card. |
| 16:58.20 | lmadsen | Docfxit: no, that would be an incorrect assumption |
| 16:58.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 16:58.35 | Docfxit | It has two CPU's running 2.4ghz |
| 16:59.08 | Docfxit | Could it be it has too much memory at 2gig |
| 16:59.22 | lmadsen | dream_th: have you read the o'reilly book? you would control it by placing extensions into a context that restricts them from other contexts that you don't want them to have access to |
| 16:59.39 | lmadsen | Docfxit: no, your approach is entirely incorrect |
| 17:00.13 | lmadsen | if you have the latest version of asterisk, and you're getting a crash, then be sure to read the doc/backtrace.txt file and you can file a bug at bugs.digium.com with the required information |
| 17:00.14 | dream_th | ok thanks lmadsen i'll try to find and read how to use contexts |
| 17:00.59 | Docfxit | I've been trying to figure out what is killing this system. I'm looking for anything right now. |
| 17:02.07 | Docfxit | I had the phone company out. They found a dead short in the Digium card. I replace the card. I haven't had the phone company out to check the new card yet. |
| 17:05.05 | Docfxit | Other people must be running without crashing. I'm trying to figure out what is different with this install. I can see most people wouldn't use a box this fast. |
| 17:05.40 | lmadsen | Docfxit: lots of people use fast boxes... I deploy clustered systems on Dell 2950's which are 2x quad-core Xeons without crashes |
| 17:06.47 | Docfxit | Is there someone I could pay to find the cause? |
| 17:07.19 | lmadsen | like I said... if you're have a crash, you may have found a bug, in which case you need to open a bug on bugs.digium.com with information about how to reproduce the error along with the backtrace |
| 17:07.38 | lmadsen | Docfxit: yes, you can hire the Custom Development department at Digium to debug the system and get the backtraces, etc... for you |
| 17:08.14 | lmadsen | you could request me if you wanted, or you could find another Asterisk consultant |
| 17:09.19 | Docfxit | I was thinking of purchasing the Business edition so I could get support. They are supposed to come out with a smaller version very soon. |
| 17:10.02 | Docfxit | Talking to the Custom Development at Digium is very expensive. |
| 17:10.09 | dream_th | lmadsen: you mean this book Asterisk: The Future of Telephony |
| 17:10.19 | lmadsen | dream_th: yes sir |
| 17:10.24 | dream_th | thank you |
| 17:10.40 | lmadsen | Docfxit: smaller version? |
| 17:10.47 | Docfxit | lmadsen» yes. |
| 17:11.13 | Docfxit | lmadsen» I think the current version supports 50 lines. |
| 17:11.32 | lmadsen | oh, you just mean smaller number of simultaneous calls |
| 17:11.50 | lmadsen | afaik that's just a function of the license limiting the number of channels |
| 17:11.59 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:12.07 | Docfxit | They are releasing a version for 40 simultaneous calls. |
| 17:12.54 | Docfxit | The only part I don't like about the business edition is that it's a very old version of the software. |
| 17:13.19 | Docfxit | They are updating the software but that won't be out for a while yet. |
| 17:14.02 | lmadsen | Docfxit: I think you have old information, because ABE C.1.x is out, and it is based on 1.4.x |
| 17:15.17 | Docfxit | lmadsen» I was talking to Chris @ Digium last week. I would have thought he would be on top of things. |
| 17:15.35 | Docfxit | Unless I didn't understand correctly. |
| 17:15.49 | lmadsen | possible |
| 17:15.55 | Docfxit | lmadsen» Are you good a figuring out problems? |
| 17:16.02 | lmadsen | I'd like to think so |
| 17:16.12 | lmadsen | if not, I shouldn't be writing books :) |
| 17:17.05 | Docfxit | I'd like to talk to you about what it would take to figure this one out. |
| 17:17.54 | lmadsen | you'll have to talk to custom development though because that's where I do all my consulting through |
| 17:18.34 | Docfxit | Doesn't that cost something like a $100/hr |
| 17:18.39 | eric2 | in extensions.conf, how can I use a GoTo and send a variable with it? |
| 17:18.57 | lmadsen | Docfxit: entirely possible, but that's what I charge too |
| 17:19.09 | lmadsen | eric2: set a channel variable before calling Goto() |
| 17:19.20 | eric2 | ok, tx lmadsen |
| 17:19.29 | lmadsen | eric2: or you can explain what you mean... |
| 17:20.12 | eric2 | I have multiple incoming did's, each did points to some context (hosted pbx) need one dial plan to control everything... goto + chan var is good I think |
| 17:20.19 | Docfxit | lmadsen» This is a small install. 9 phone lines. Maybe 20 calls a day. |
| 17:20.54 | lmadsen | Docfxit: you might want to find another consultant then, because those are my rates... |
| 17:21.20 | lmadsen | eric2: catch it with a pattern match, then you can use ${EXTEN} |
| 17:21.35 | Docfxit | lmadsen» Any idea how many hours it would take to look over the box? |
| 17:21.47 | lmadsen | Docfxit: not too sure to be honest... probably 1-2 |
| 17:22.22 | lmadsen | I have a two hour minimum anyways |
| 17:22.43 | Docfxit | 1-2 would be great. My distributor and myself have been working on this for weeks. |
| 17:23.05 | Docfxit | What phone number would I call to get this started? |
| 17:23.20 | lmadsen | I'd say the sales department at digium |
| 17:24.02 | Docfxit | How could I ask for your help? |
| 17:24.04 | lmadsen | just request Leif Madsen |
| 17:24.30 | lmadsen | that should be all you need to do |
| 17:24.35 | Docfxit | Great. |
| 17:24.41 | lmadsen | and your name sir? |
| 17:25.05 | lmadsen | so I can relate the call to this conversation in my head |
| 17:25.05 | eric2 | lmadsen: in short, what do you recommend for failover? |
| 17:25.05 | Docfxit | I don't think they are open today? |
| 17:25.05 | Docfxit | Gary |
| 17:25.22 | worgil | hello, when i want login from x-lite looking error (reistration error: 408. request time out). what can i do ? |
| 17:25.29 | lmadsen | eric2: depends what you're trying to do.... |
| 17:25.38 | worgil | hello, when i want login from x-lite looking error (registration error: 408. request time out). what can i do ? |
| 17:26.04 | eric2 | 2 box's, primary takes all traffic, when it goes down, 2nd machine is to pick up the traffic |
| 17:26.15 | eric2 | no need for clustering yet |
| 17:26.27 | Docfxit | lmadsen» Thanks for your input. |
| 17:26.38 | lmadsen | eric2: look into LinuxHA |
| 17:26.42 | lmadsen | (I think that's what it's called) |
| 17:27.04 | eric2 | ok, have looked at it.. tx |
| 17:27.22 | lmadsen | Docfxit: np... I'm off to the grocery store to get some breakfast food and should be back later |
| 17:27.48 | Docfxit | lmadsen» Is there anything I can do to help resolve this. |
| 17:28.03 | Docfxit | lmadsen» Like start any traces or debugs? |
| 17:28.14 | lmadsen | Docfxit: ya... read backtrace.txt and valgrind.txt |
| 17:28.16 | lmadsen | in the doc/ dir |
| 17:28.25 | Docfxit | lmadsen» Thanks. |
| 17:28.50 | lmadsen | np |
| 17:29.10 | *** join/#asterisk timeshell (n=Khoja@206.248.136.108) |
| 17:29.36 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:33.58 | *** join/#asterisk techie (n=techie@adsl-76-214-31-194.dsl.lsan03.sbcglobal.net) |
| 17:38.48 | *** join/#asterisk reber (n=reber@193.253.213.73) |
| 17:41.12 | worgil | hello, when i want login from x-lite to asterisk server, looking error (registration error: 408. request time out). what can i do ? |
| 17:41.34 | timeshell | ping the server |
| 17:41.46 | timeshell | <PROTECTED> |
| 17:42.22 | worgil | yes |
| 17:42.28 | worgil | pinging |
| 17:42.45 | timeshell | what's the ping time? |
| 17:43.31 | worgil | Terminated |
| 17:43.38 | worgil | 88.247.51.99 cevabý: bayt=32 süre=29ms TTL=60 |
| 17:43.39 | worgil | 88.247.51.99 cevabý: bayt=32 süre=26ms TTL=60 |
| 17:44.07 | timeshell | looks ok |
| 17:44.18 | timeshell | firewall? |
| 17:44.25 | worgil | no |
| 17:44.29 | worgil | it closed |
| 17:44.47 | timeshell | On the asterisk server? |
| 17:44.56 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 17:45.16 | worgil | yes |
| 17:45.28 | worgil | behind DSL router |
| 17:46.39 | worgil | not looking any error else |
| 17:46.54 | worgil | 408 request time out |
| 17:50.23 | worgil | can it be rtp.conf timeshell? |
| 17:51.15 | timeshell | The asterisk server is behind a DSL router? Which is before the XLite? |
| 17:51.33 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 17:51.48 | worgil | zoiper |
| 17:52.00 | Datax | nat issue ? |
| 17:52.26 | Datax | worgil: have you opened the port(s) needed ? |
| 17:52.33 | worgil | yes |
| 17:52.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:52.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:52.52 | Datax | can you see the client trying to log on to the server in the asterisk CLI ? |
| 17:53.07 | worgil | how can i see ? |
| 17:53.08 | riddlebox | is there a way to see if SIPAddHeader is working? |
| 17:53.12 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 17:53.53 | Datax | worgil: asterisk -vvvvvvr |
| 17:53.57 | Datax | on the asterisk server |
| 17:54.36 | worgil | ok |
| 17:58.01 | *** join/#asterisk erago (n=erago@236.Red-81-39-224.dynamicIP.rima-tde.net) |
| 17:58.58 | *** join/#asterisk SteveTotaro (n=root@pool-70-22-26-147.balt.east.verizon.net) |
| 17:59.01 | riddlebox | I am trying to get paging to work, I followed the instructions on grandstreams website, and setup the phones and extensions.conf, when I try to call the paging exten I get this, http://pastebin.ca/897594 |
| 17:59.09 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 18:04.15 | *** join/#asterisk ZaVoid (n=zavoid@c-68-44-91-5.hsd1.nj.comcast.net) |
| 18:06.17 | worgil | i have 4 voip modem on my asterisk server on other DSL lines, must i set externip other for voip modem ip? |
| 18:09.57 | erago | Hello, i`m trying to setup a simple asterisk box with two sip clients, both registered. When I try to make a call I get a notice "chan_sip.c handle_request_invite: Call from 1002 to extension 1001 rejected because extension not found". I have autoload=no in modules.conf. Thanks. |
| 18:11.52 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 18:15.57 | mvanbaak | Corydon76-dig: the trick with hints did not work |
| 18:16.06 | mvanbaak | prolly because the hints are acting up |
| 18:18.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:19.31 | mvanbaak | some info: http://pastebin.ca/897617 |
| 18:20.02 | mvanbaak | it's funny to see a peer that's not registered on the box shows up as Idle in the 'show hints' |
| 18:21.11 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 18:26.08 | riddlebox | is anyone doing paging or intercom'ing over the phone speakers? |
| 18:27.33 | drmessano | Hmmm |
| 18:30.10 | eric2 | riddlebox, I need to do that but haven't gotten to it yet |
| 18:30.42 | riddlebox | I am trying with the cursed grandstream gxp2000's, http://pastebin.ca/897594 |
| 18:30.49 | eric2 | I think you have to use ring groups but I could be mistaken |
| 18:31.09 | riddlebox | eric2,http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Page |
| 18:35.12 | mvanbaak | Corydon76-dig: I think I'll just use Dial(SIP/${EXTEN}) and based on DIALSTATUS run a DUNDi query |
| 18:35.48 | drmessano | Grandcentral gets a BIG FAIL |
| 18:37.48 | *** part/#asterisk Whoopie (n=Whoopie@unaffiliated/whoopie) |
| 18:38.27 | timeshell | what determines which context the Queue() function uses when calling an agent? |
| 18:39.32 | timeshell | riddlebox: I am using a Polycom |
| 18:40.03 | timeshell | I have to set the AlertInfo on the phone to a value and the RingType to 4 in the sip.cfg |
| 18:40.19 | drmessano | Damn |
| 18:40.32 | timeshell | And then set the AlertInfo in the SIPAddHeader to the same value as I did on the phone before dialing it. |
| 18:40.40 | drmessano | I need to find a single EXE DNSd for Windows so I can unlock a PAP2 lol |
| 18:41.00 | timeshell | drmessano: SimpleDNS |
| 18:41.08 | timeshell | That's what I used |
| 18:41.11 | drmessano | Does it require an install? |
| 18:41.33 | timeshell | Don't know off hand... I can check |
| 18:41.40 | drmessano | Well, thats the whole point |
| 18:41.52 | drmessano | I can run BIND from an unpacked ZIP, I think |
| 18:41.54 | *** join/#asterisk bogar (n=bogar@101.Red-81-37-161.dynamicIP.rima-tde.net) |
| 18:42.24 | drmessano | I had a box just for unlocking PAP2s, and I blew it up thinking I wouldnt run across any more of them |
| 18:42.27 | drmessano | and so I did.. lol |
| 18:42.33 | riddlebox | timeshell, I just got it working, the grandstream site tells you to create a list to do it but that list is not recognized in 1.4.17, or any 1.4 I assume |
| 18:42.52 | drmessano | So I want to create an enviornment to unlock without storing a PC for it |
| 18:43.49 | drmessano | Im thinking an old Linksys router for DHCP+switch, small DNSd and small TFTPd along with that AnalogX WWW server |
| 18:44.49 | *** join/#asterisk merkurie (n=merkurie@c-68-60-85-88.hsd1.mi.comcast.net) |
| 18:45.49 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:45.52 | drmessano | timeshell: Treewalk DNS |
| 18:45.57 | drmessano | Single EXE FTW |
| 18:46.00 | timeshell | So, I'm sending calls to a queue, which in turn is calling agents. However, the agents are defined in users.conf by asterisk-gui. The ones in that have hasvoicemail=yes the queue is dialing stdexten macro to call them which forwards them to voicemail. For users that have hasvoicemail=no, it just uses dial. I need to do just dial for both. I can't see anywhere yet where it is defined to even use stdexten macro so I need to know why it's doing that |
| 18:49.39 | timeshell | there are no contexts in extensions.conf where the macro stdexten is even called |
| 18:50.28 | eric2 | in one context I set a channel variable with the following: exten => _X.,1,Set(Var_TO=${SIP_HEADER(TO):4:10}) |
| 18:50.57 | eric2 | then use a goto and the variable is not found in the goto context |
| 18:51.13 | timeshell | but something is calling it from as it shows up in the verbose |
| 19:00.35 | timeshell | is there any other conf file that can run macros other than extensions.conf? |
| 19:02.03 | *** join/#asterisk ZPertee (n=ZPertee@198.sub-70-217-164.myvzw.com) |
| 19:02.46 | ZPertee | how does asterisk know where I parked a call at? I want to park a call move on down through the dial plan and then pick it up later. how can I do this? |
| 19:03.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 19:03.10 | timeshell | variables |
| 19:03.15 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 19:05.02 | ZPertee | so once I park a call at extension say 711 than I just dial ext 711 and I'm connected??? |
| 19:05.21 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 19:05.39 | timeshell | Something like that from what I understand |
| 19:06.02 | ZPertee | ok thanks for your help |
| 19:07.41 | eric2 | timeshell, you can include files of you choice |
| 19:08.14 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 19:08.18 | eric2 | for example: I created a utils directory.... #include <utils/faxing.conf> |
| 19:08.35 | timeshell | eric2: That's not quite what I meant. |
| 19:08.48 | merkurie | my telco (comcast) seems to not perform a battery/no battery for 10-20 seconds after the other party has hung up... anyone else ever see this? |
| 19:08.52 | supjigator | Anyone here using libss7? |
| 19:09.13 | timeshell | eric2: asterisk-gui appears to be making asterisk call stdexten when putting a queue'd call to an agent's extension and I'm tryin to figure out where it does that |
| 19:09.36 | eric2 | hmm.. that's one of the reasons why I stopped using the gui |
| 19:09.39 | supjigator | merkurie: I've never seen any signaling that was 10-20 seconds. |
| 19:10.00 | merkurie | ya |
| 19:10.01 | merkurie | strange eh? |
| 19:10.14 | eric2 | timeshell do a search for stdexten as a string in the files? |
| 19:10.16 | supjigator | merkurie: Yea. |
| 19:10.23 | supjigator | merkurie: What CPE is it? |
| 19:10.56 | merkurie | i'll call my cell phone, hang up and watch log and won't get the battery/no battery for almost 20 seconds |
| 19:11.12 | merkurie | supjigator, not sure what cpe is? still pretty new to ast... |
| 19:11.31 | supjigator | merkurie: What CPE? What cable device do you have that has the FXS port in it. |
| 19:11.46 | timeshell | eric2: Yah, I'd do the same except I'm building the system for a company that needs a simpler way to manage it than conf files |
| 19:11.53 | merkurie | supjigator, ahhh... i can go look, starts with an a i think =) brb... i'll look |
| 19:11.58 | timeshell | So, I have to figure out the best way to make it work. |
| 19:12.06 | eric2 | ah |
| 19:12.38 | merkurie | supjigator, "arris touchstone" |
| 19:12.42 | supjigator | merkurie: There should be specs for it. It controls the signaling. Its most likely packetcable or SIP but it would be doing the the FXS signaling. |
| 19:12.49 | timeshell | I mean, all the gui does is manage the conf files right? So it's got to be in there somewhere. |
| 19:13.24 | eric2 | what if its in realtime? |
| 19:13.31 | lmadsen | mvanbaak: hi! |
| 19:13.44 | mvanbaak | lmadsen: ello ello ! |
| 19:13.56 | mvanbaak | lmadsen: great work on the cli stuff m8 |
| 19:13.59 | lmadsen | how goes this glorious day? |
| 19:14.11 | lmadsen | mvanbaak: you did all the work... I just documented it :0 |
| 19:14.16 | lmadsen | we make a good team :D |
| 19:14.21 | mvanbaak | yup :) |
| 19:14.44 | lmadsen | I hope devcon is in holland this year |
| 19:14.50 | mvanbaak | it wont be |
| 19:14.54 | lmadsen | :( |
| 19:15.01 | lmadsen | haven't been there and would love to go |
| 19:15.09 | mvanbaak | ah |
| 19:15.13 | mvanbaak | well, 2 man devcon :) |
| 19:17.09 | lmadsen | haha |
| 19:17.15 | lmadsen | and we probably won't get much done |
| 19:17.15 | lmadsen | heh |
| 19:17.45 | drmessano | window sucks |
| 19:17.48 | drmessano | sorry, obvious |
| 19:18.17 | lmadsen | lol |
| 19:18.31 | lmadsen | I like WinXP... I only use it for quickbooks and azureus... but I like it :0 |
| 19:18.38 | lmadsen | damnit... I keep doing :0 instead of :) |
| 19:18.55 | drmessano | I updated a bunch of apps since I last rebooted.. All of which were previously set to not start with Windows |
| 19:19.17 | drmessano | They all changed to "start with windows" |
| 19:19.30 | drmessano | I reboot after an app breaks some things.. |
| 19:19.58 | drmessano | I have a 2.2 GHZ box with 4GB RAM.. and it takes 12 minutes for the box to slow to "responsive enough" with all these apps loading |
| 19:20.12 | drmessano | HD light just solid.. |
| 19:20.14 | drmessano | BAH |
| 19:20.34 | drmessano | I said "NO".. NO MEANS NO, not "NO now, but yeah, start on boot later" |
| 19:20.37 | drmessano | BAH2 |
| 19:20.56 | drmessano | </rant> |
| 19:21.05 | timeshell | heh, why are you running windows? |
| 19:21.13 | drmessano | Application requirements |
| 19:21.20 | timeshell | Work under WINE? |
| 19:21.24 | drmessano | Nope |
| 19:21.40 | timeshell | ah well. |
| 19:21.43 | lmadsen | WinXP + VMware Server = <3 |
| 19:22.09 | drmessano | I'm considering that.. My windows need is becoming less and less |
| 19:22.25 | lmadsen | yep... I only need it for a couple of apps, and VMware Server is rock solid |
| 19:23.27 | drmessano | What bothers me worse than Windows, is the gall of a lot of these developers of WIndows apps |
| 19:23.35 | lmadsen | plus it's nice to just suspend it when you're done, and you don't have to wait for bootup times |
| 19:24.00 | drmessano | This whole "start with windows" battle is beyond Microsoft |
| 19:24.18 | drmessano | Been fighting it since the days of 95.. and it just gets worse |
| 19:24.35 | mvanbaak | lmadsen: I'm off |
| 19:24.38 | mvanbaak | going to the bar |
| 19:24.39 | mvanbaak | latero |
| 19:24.47 | drmessano | The whole platform is like the US Congress.. |
| 19:25.48 | lmadsen | mvanbaak: awesome.. I'm going to cook dinner for a couple of friends and open a bottle of vino... peas out |
| 19:26.24 | lmadsen | thinking of making a tomato pasta sauce with quick fry steak |
| 19:26.32 | lmadsen | which means I should probably start marinating that steak right now |
| 19:26.35 | lmadsen | so I'm off! |
| 19:26.39 | drmessano | cya! |
| 19:26.46 | drmessano | hehe |
| 19:26.47 | tzanger | lmadsen: you can't cook |
| 19:26.53 | lmadsen | tzanger: I almost can now! |
| 19:26.57 | tzanger | lmadsen: excellent |
| 19:27.04 | tzanger | if I were going anywhere near downtown I'd stop by for sure |
| 19:27.07 | tzanger | I've got a strong stomach |
| 19:27.17 | lmadsen | oh you don't need it... I'm almost good now |
| 19:27.23 | lmadsen | I haven't made anything that was terrible yet |
| 19:27.32 | tzanger | :-) |
| 19:27.39 | lmadsen | maybe a couple of sauces a bit bland, but I'm learning :) |
| 19:27.42 | tzanger | I was in toronto yesterday and again on monday, but only to the airport |
| 19:27.46 | lmadsen | you should totally come over sometime and show me how to cook, heh |
| 19:27.55 | lmadsen | the airport is not in toronto |
| 19:27.57 | drmessano | One of my buddies tried to grill out last night.. invited us over.. WAY underloaded the grill with charcoal, so we it was hamburger tartar or order a pizza.. the pizza was pretty good. |
| 19:28.04 | lmadsen | it is in mississauga... and mississauga != toronto |
| 19:30.14 | tzanger | I got to meet kyron too |
| 19:30.14 | tzanger | lmadsen: yeah yeah yeah, that's a typical torontonian response |
| 19:30.14 | file | tzanger: YOU |
| 19:30.17 | tzanger | anything about 25 blocks in any direction of the CN Tower is not considered toronto :-) |
| 19:31.31 | tzanger | file: me? |
| 19:31.38 | file | tzanger: maybe. |
| 19:31.51 | tzanger | file: you had a GSM radio card for your laptop; what kind of plan were you on and what did it cost? |
| 19:32.28 | file | I was not on a plan, I had a t-mobile prepaid SIM roaming on Rogers with unfiltered access |
| 19:33.02 | tzanger | t-mobile prepaid SIM, ok I understand that |
| 19:33.05 | tzanger | roaming on rogers, I got that |
| 19:33.09 | tzanger | unfiltered access -- ?? |
| 19:33.39 | file | T-Mobile provides free limited WAP access with their prepaid, when they enabled roaming on Rogers they did not properly set it up to limit - so full data access was provided |
| 19:33.41 | file | that has since been closed |
| 19:33.44 | tzanger | how much was the sim, and for how much access? |
| 19:33.46 | tzanger | ahhhhhhh |
| 19:34.08 | tzanger | so that hole's been fixed then, has it? |
| 19:34.32 | file | port filtered now, and they push everything through a transparent proxy |
| 19:34.43 | file | at least while roaming here |
| 19:34.44 | tzanger | bugger. no ssh even? |
| 19:34.48 | file | nope |
| 19:34.51 | tzanger | fuckity |
| 19:35.15 | tzanger | why is fucking data access so expensive anyway |
| 19:35.15 | tzanger | ugh |
| 19:35.26 | file | because they want it to be. |
| 19:35.39 | Nivex | because people will pay for it |
| 19:37.52 | drmessano | You see the price comparison of TXTing vs Mobile Data? |
| 19:39.15 | drmessano | Something along the lines of if you transferred 1GB of data from a $30 mobile plan, to send the same amount of data per the rate of a single text message was $32 million |
| 19:39.50 | *** join/#asterisk ManxPower (n=manxpowe@200.sub-70-221-66.myvzw.com) |
| 19:44.29 | *** join/#asterisk shinao1 (n=shinao1@80.248.0.59) |
| 19:48.35 | eric2 | lmadsen: I'll be in toronto sunday/monday what's for diner? |
| 19:49.24 | lmadsen | eric2: depends what you bring :) |
| 19:50.27 | eric2 | steak! |
| 19:51.33 | hmmhesays | does anyone know if function VMCOUNT only returns new voicemails? |
| 19:55.47 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 19:55.57 | eric2 | http://www.voip-info.org/wiki/view/Asterisk+func+vmcount |
| 19:56.57 | eric2 | looks like it does nothing more than just that |
| 19:57.18 | arekm | hello, I need a little help with asterisk buildsystem. I need chan_zap.so to be linked with one more library. I added AST_EXT_LIB_CHECK() check for it but how to make chan_zap linked with thing in MYLIB_LIBS=? |
| 20:01.55 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 20:04.26 | *** join/#asterisk mmmToop (n=michaelt@dsl-243-255-91.telkomadsl.co.za) |
| 20:12.29 | arekm | hm, found something but now it stops building chan_zap at all :( |
| 20:13.00 | *** part/#asterisk mmmToop (n=michaelt@dsl-243-255-91.telkomadsl.co.za) |
| 20:20.20 | hmmhesays | so something seriously goofy is going on in the dialplan now when I set the callerid |
| 20:21.59 | hmmhesays | it seems it is not setting the callerid |
| 20:25.04 | hmmhesays | nope it sure is not |
| 20:35.50 | hmmhesays | figured it out |
| 20:35.54 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 20:38.47 | redback | I am trying to use conditional branching using an example on pg 151 of the 'handbook': exten => 01142997839,n,GotoIf($[${CALLERID(num)} = 8885551212]?101,1:reject) but keep getting a syntax error: syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 20:39.28 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 20:42.15 | eric2 | redback, I just did something like that... |
| 20:42.42 | eric2 | exten => s,n,Set(PbxContext=${IF($[ ${Var_TO} = 4164554120]?mgr:)}) |
| 20:45.13 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 20:50.53 | redback | eric2: so that assigns nothing or mgr to the variable PbxContext? |
| 20:51.04 | eric2 | correct |
| 20:51.27 | eric2 | if its a match, then mgr gets assigned to pbxcontext |
| 20:52.40 | redback | mmm, the if looks pretty much like mine except the extra spaces - I am gonna try and add them and see if that works |
| 20:53.38 | *** join/#asterisk shinao1 (n=shinao1@41.223.144.105) |
| 20:55.07 | lmadsen | eric2: you don't need the : in there fyi |
| 20:55.09 | redback | eric2: cool that single space did it |
| 20:55.22 | lmadsen | only when you need to pass a false value back |
| 20:55.33 | lmadsen | plus, you're going to set a NULL value sometimes... if you don't already know |
| 20:56.17 | redback | I think thats where the unexpected '=', expecting $end; Input: error came in when the CALLERID was not set and was null |
| 20:56.26 | lmadsen | I prefer to do something like: exten => s,n,Exec(${IF($[${Var_TO} = 4164554120]?Set(PbxContext=mgr):NoOp())}) |
| 20:56.52 | lmadsen | $[Var_TO} has to be non-null, otherwise you have to put double quotes around it and what you are comparing to |
| 20:57.02 | lmadsen | I prefer to do something like: exten => s,n,Exec(${IF($["${Var_TO}" = "4164554120"]?Set(PbxContext=mgr):NoOp())}) |
| 20:57.11 | lmadsen | notice the double quotes in the 2nd example |
| 20:57.57 | lmadsen | or alternatively... you can do: exten => s,n,GotoIf($[${ISNULL(${Var_TO})}]?skip_set) |
| 20:58.10 | lmadsen | exten => s,n,Exec(${IF($["${Var_TO}" = "4164554120"]?Set(PbxContext=mgr):NoOp())}) |
| 20:58.17 | lmadsen | exten => s,n(skip_set),NoOp() |
| 20:58.22 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 20:59.05 | eric2 | hmm, I think I'll use one of your suggestions lmadsen.. probably the 2nd last one |
| 20:59.43 | lmadsen | ya... I don't tend to do the GotoIf() one unless I'm doing something I really don't want to execute at all |
| 20:59.56 | lmadsen | btw: don't put extra spaces where you don't need them |
| 20:59.56 | arekm | er, one more q, what's the best way to link two asterisk instances running on single machine? TDM? |
| 21:00.03 | lmadsen | it can possibly screw up your parsing |
| 21:00.17 | eric2 | k, excellent points! |
| 21:00.32 | lmadsen | only spaces around the operator... nothing else |
| 21:00.43 | lmadsen | $[ ${VAR} = foo ] <--- I don't like |
| 21:00.52 | lmadsen | $[${VAR} = foo] <--- much better |
| 21:01.20 | redback | lmadsen: $[ ${VAR} = foo ] worked for me and $[${VAR} = foo ] did not |
| 21:01.22 | lmadsen | and remember: always double quote the values being compared if one of them could possibly be null |
| 21:01.36 | lmadsen | redback: you still have an extra space |
| 21:01.42 | lmadsen | redback: this is why spaces are BAD |
| 21:02.01 | arekm | the way that would allow faxes to be transferred correctly? anyone |
| 21:02.07 | lmadsen | unless you are checking prior to using the variable that it is not null, always double quote |
| 21:02.20 | lmadsen | or you end up with essentially this: $[ = foo] |
| 21:02.27 | lmadsen | this will give you an error |
| 21:02.45 | lmadsen | if you double quote... then at worst you end up with: $["" = "foo"] |
| 21:02.49 | lmadsen | that will NOT cause an error |
| 21:02.52 | riddlebox | wohoo got the intercoming/paging working |
| 21:04.36 | eric2 | nice! :) |
| 21:04.45 | redback | lmadsen: thanks for those tips works lovely |
| 21:05.02 | JerJer | yahoo turns down microsoft |
| 21:07.12 | eric2 | arekm I was looking at faxing last week.. only way I can see doing it correctly is to use T.38 |
| 21:07.33 | arekm | eric2: huh? both asterisk are on single network so no latency issues |
| 21:07.45 | eric2 | d'oh, my bad |
| 21:08.00 | arekm | eric2: TDMoE would work... but it needs mac addresses which won't work on single machine |
| 21:08.46 | eric2 | can TDMoE be used over the internet? |
| 21:08.53 | arekm | no, only over ethernet |
| 21:09.30 | lmadsen | fax + voip = fail |
| 21:09.36 | lmadsen | fax + tdm = ok |
| 21:09.55 | eric2 | what about t.38? |
| 21:09.56 | arekm | lmadsen: now how to get tdm working on single machine :> |
| 21:10.07 | lmadsen | t.38 will work |
| 21:10.14 | lmadsen | but asterisk will only do passthrough |
| 21:10.20 | lmadsen | so both ends needs to speak t.38 |
| 21:10.23 | eric2 | passthrough is all I need |
| 21:10.36 | arekm | two network cards connected with crossover hehehe |
| 21:14.11 | tzafrir_laptop | arekm, I figure you can also use iax to link them. I'm not sure how you'll split the TDM channels between the two instances |
| 21:15.06 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 21:15.50 | arekm | tzafrir_laptop: one will use 1 eth card, second will use 2 eth card, they will talk to each other |
| 21:18.14 | tzafrir_home | What TDM channels do you want to use, exactly? |
| 21:18.39 | arekm | all of them |
| 21:19.06 | arekm | TDMoE |
| 21:21.41 | drmessano | ouch |
| 21:23.54 | tzafrir_home | arekm, zaptel channels and spans are global |
| 21:26.03 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:26.06 | arekm | tzafrir_home: hm, can I do (a1)span1 ... tmdoe ... (a2) span2 ? then half of channels would be usable at the same time |
| 21:27.16 | arekm | but maybe simply using iax would be enough even for faxes (since this is single computer so only loopback on the way) ? |
| 21:28.31 | tzafrir_home | people use iax for iaxmodem . It seems to work on the same host |
| 21:31.27 | arekm | Now the other potential problem arises. I have three cards: 1x quadGSM and 2x B410. Will such setup work: one asterisk manages quadgsm card (zaptel) and other asterisk manages 2xB410 (also via zaptel) ? |
| 21:45.50 | *** join/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 21:47.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:49.10 | *** join/#asterisk RoyK (n=roy@ip-85-21-149-91.dialup.ice.no) |
| 21:49.37 | drmessano | HA |
| 21:49.41 | *** part/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |
| 21:52.44 | *** join/#asterisk BBHoss (n=hoss@c-71-207-173-38.hsd1.al.comcast.net) |
| 21:53.10 | Greek-Boy | Is it really necessary to use dundi within a enterprise? What about using pattern matching with iax2 alone? |
| 21:54.07 | BBHoss | Greek-Boy, what do you want to do with dundi, clustering or a e.164-type network? |
| 21:55.38 | hmmhesays | I'm having a strange agi problem, if I try and read from STDIN in two places in my perl agi script it just seems to hang there |
| 21:56.01 | Greek-Boy | I have an organization with multiple branches. I just want all the organization's extensions to be reachable from every asterisk. |
| 21:56.17 | BBHoss | Greek-Boy, how many branches? |
| 21:56.30 | Greek-Boy | 7 |
| 21:56.33 | BBHoss | hmm |
| 21:56.49 | BBHoss | and there is an asterisk server running at each? |
| 21:57.17 | hmmhesays | http://www.pastebin.ca/897855 |
| 21:57.23 | Greek-Boy | yip |
| 21:57.51 | Greek-Boy | I've never experimented with dundi. This will be the first time. |
| 21:58.05 | hmmhesays | anyone know why trying to read from stdin the second time kills the script |
| 21:58.06 | BBHoss | Greek-Boy, it would probably be just as easy to make iax connections |
| 21:58.09 | hmmhesays | it seems as though it doesn't run at all |
| 21:58.42 | BBHoss | Greek-Boy, only problem is, that you'll be hard pressed to get DUNDi support if something doesn't work like you think it should |
| 21:59.02 | Greek-Boy | ok, so I just should use pattern matching? |
| 21:59.18 | BBHoss | Greek-Boy, thats what i would do, even though i use DUNDi |
| 21:59.43 | BBHoss | now if you had like 25 branches, i could see where having that many lines would get tedious |
| 21:59.59 | jhiver | hi guys |
| 22:00.19 | *** join/#asterisk nitzer (n=nitzer@unaffiliated/nitzer) |
| 22:00.36 | jhiver | i have a question about .call files, more specifically about 'Channel' |
| 22:00.36 | Greek-Boy | ok |
| 22:00.54 | jhiver | how do you make it hunt through multiple routes for the outbound call? |
| 22:01.44 | BBHoss | jhiver, you are using asterisk and not freepbx right? |
| 22:02.10 | jhiver | yeah |
| 22:02.19 | jhiver | i'm thinking pehaps using Local/ ? |
| 22:02.30 | jhiver | and then having my hunt order there? |
| 22:02.43 | BBHoss | what do you mean by hunt, what are you trying to do with it |
| 22:02.47 | jhiver | well |
| 22:03.00 | jhiver | first try peerA, then if it's congested try peerB, etc |
| 22:03.45 | jhiver | i'd like to avoid setting up a second asterisk box just to do this :) |
| 22:05.05 | drmessano | wow |
| 22:05.09 | drmessano | This is an OLD pap2 |
| 22:05.16 | BBHoss | PAP2 V1? |
| 22:05.18 | drmessano | Can't seem to get it where I want it lol |
| 22:05.25 | drmessano | PAP2v1 with 2.x firmware on it |
| 22:05.35 | BBHoss | the PAP2 V2 are easy as hell to crack |
| 22:05.40 | drmessano | I know |
| 22:06.06 | drmessano | I seem to get this one to take 3.1.6 from the SPA FW stage |
| 22:06.13 | drmessano | can't seem |
| 22:06.50 | BBHoss | jhiver, you can use the GotoIf() application |
| 22:08.59 | Datax | Hi all, I'm setting up my first Asterisk box and am learning about dialplans and peers |
| 22:09.22 | Datax | I have a SIP host that I have been able to connect to thanks to a guide found online |
| 22:09.31 | Datax | but there's something I don't understand |
| 22:09.34 | drmessano | Damn... odd one |
| 22:09.56 | Datax | if I set my SIP provider as a friend why can't I receive calls ? |
| 22:10.15 | Datax | they only arrive on my sip box if I set up the provider as a peer |
| 22:10.41 | Agrajag- | jhiver: if you have that auto fallthrough option on, Dial will go to priority n+101 on congested/busy/unavailable/noanswer |
| 22:10.43 | Greek-Boy | BBHoss from a security point of view is it better to go with DUNDi? |
| 22:11.02 | Datax | I thought that by setting it as a friend I'd be able to configure incoming and outgoing calls |
| 22:12.07 | BBHoss | Greek-Boy, ruby encrypts the lookups and such, but it still runs over iax2 |
| 22:13.15 | Agrajag- | jhiver: err, take noanwer out of that list |
| 22:14.37 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 22:14.45 | jhiver | ok the question was about callfiles, not about hunting itself, but i think using Local channel will work =) |
| 22:14.54 | *** join/#asterisk AndyGraybeal (n=andy@node207.35.251.72.1dial.com) |
| 22:15.03 | luke-jr | Does anyone know if it's possible to find dedicated servers with pings under 50ms to all the continental US? |
| 22:15.55 | BBHoss | luke-jr, it always depends on the user's side |
| 22:16.36 | luke-jr | yes, but I mean assuming a reasonable delay on the user's end ;) |
| 22:16.43 | jhiver | plus hunting on busy would be kind of bad methinks :) |
| 22:17.01 | jhiver | if the number returns busy, there's really no point in hammering it |
| 22:17.26 | jhiver | since i'm always hunting the same number through different peers |
| 22:17.35 | jhiver | it's not like i'm ringing different phoens |
| 22:18.58 | Datax | figured it out |
| 22:19.00 | Datax | :) |
| 22:22.35 | variable_office | my asterisk server keeps asking my asterisk client for options but it is to: s@xxx.xxx.xxx.xxx there is no such user, so the client replies 404. any idea where the server is getting the s@ username? |
| 22:22.58 | variable_office | the user is registered as sipuser16 |
| 22:28.36 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 22:28.48 | iamhrh | does anyone know if the latest SIP firmware for the 7960 cisco phones supports auto answer via a sip header? |
| 22:30.06 | drmessano | That was a new one |
| 22:30.19 | drmessano | I don't know if it's the hardware version, or what |
| 22:30.46 | drmessano | But I normally 3.1.9 > SPA Firmware > 3.1.6 PAP2 FW |
| 22:31.18 | drmessano | This 2.0.10 one was 2.0.10 > SPA FW > 3.1.3 PAP2 FW > 3.1.6 PAP2 FW |
| 22:31.29 | drmessano | Weird shit |
| 22:35.05 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 22:35.41 | variable_office | the Reg. Contact for the user seems to be wrong |
| 22:35.45 | variable_office | it is s@ |
| 22:36.25 | hmodes | variable_office: does your reg line have /sipuser16 at the end? |
| 22:36.44 | variable_office | thats what i am looking at now, apparently it defaults to s |
| 22:36.49 | hmodes | yup |
| 22:37.05 | variable_office | that seems stupid, why doesnt it default to the username? |
| 22:37.34 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:37.52 | Datax | quick question, I intend to use SCCP cisco phones with my asterisk server. I get the impression that there is more than just one SCCP module. Which one is best ? |
| 22:38.12 | hmodes | well, if you're registering to things that allow the contact to differ having a default can be useful |
| 22:38.14 | hmodes | i guess... |
| 22:38.25 | hmodes | i dunno really, i always use /<contact> |
| 22:39.14 | Datax | anyone ? :) |
| 22:39.59 | Datax | what differs from skinny chan, chan_sccp and chan-sccp-b ? |
| 22:40.40 | variable_office | hmodes, i did that but it is still sending back 404 sometimes |
| 22:41.06 | hmodes | 'sometimes'? |
| 22:41.58 | variable_office | oh wait, always, OPTIONS request for the server are accepted, but OPTIONS request from the server are 404 |
| 22:43.19 | hmodes | pastebin sip debug? |
| 22:43.47 | hmodes | and the relevant sip.conf/extensions.conf too i guess |
| 22:44.06 | variable_office | what in extensions.conf would matter for this? |
| 22:44.26 | hmodes | well i'm not quite sure what you mean by asking for options |
| 22:44.34 | hmodes | so i just default to asking for everything ;p |
| 22:45.02 | hmodes | afaik options should only be exchanged as part of a registration or invite |
| 22:45.05 | variable_office | 102 options is what it says |
| 22:45.21 | *** join/#asterisk TeamINM (n=TeamINM@dsl093-197-074.mke1.dsl.speakeasy.net) |
| 22:45.22 | variable_office | i think it does it if you have qualify set to yes, but i am not sure |
| 22:45.24 | *** join/#asterisk noneo (n=ankamins@82-43-248-64.cable.ubr28.newt.blueyonder.co.uk) |
| 22:45.27 | *** join/#asterisk shido6 (n=shido6@74-130-50-233.dhcp.insightbb.com) |
| 22:45.33 | hmodes | that's entirely possible |
| 22:45.41 | hmodes | just the sip debug is all i really need to see, i think |
| 22:46.07 | TeamINM | has anyone had issues with a digium tdm800p card not loading the child cards? |
| 22:46.58 | TeamINM | channel failed error 1 |
| 22:47.23 | hmodes | personally, i avoid qualify. imo it gets used for the wrong reason a lot |
| 22:47.44 | hmodes | but then i've been waiting for a registration cache forever |
| 22:48.04 | variable_office | hmodes, http://www.pastebin.ca/897922 |
| 22:48.15 | hmodes | rapid registration w/ caching >>> notify |
| 22:49.15 | variable_office | and that dialog just keeps repeating |
| 22:51.01 | hmodes | that's... odd.. |
| 22:52.13 | variable_office | which part? |
| 22:52.26 | hmodes | oh wait, no, that is normal for qualify= |
| 22:52.58 | hmodes | i guess they use that instead of notify/info now |
| 22:53.39 | hmodes | what version of * is this? |
| 22:53.59 | variable_office | 1.4 on all involved machines |
| 22:54.54 | hmodes | and the machines are both registering to each other? |
| 22:55.19 | variable_office | no, the client is registering to the server |
| 22:55.54 | hmodes | client's sip show registry shows it as registered? |
| 22:56.51 | variable_office | yep registered as sipuser16 to the dns name of the host |
| 22:57.00 | variable_office | err. to the dns name of the server |
| 22:57.42 | hmodes | is there a reason you have nat enabled? it seems they are not natted from the messages |
| 22:57.48 | hmodes | not that it should matter |
| 22:58.09 | variable_office | it is enabled on the server by default, it doesnt hurt anything when talking to non-natted atas at least |
| 22:58.42 | hmodes | i'd try disabling it just to be sure, beyond that it looks like it might be a bug |
| 22:59.15 | hmodes | i don't see anything obviously wrong with the messages assuming the client trusts the box sending the options |
| 22:59.28 | hmodes | might want to try insecure=very? |
| 22:59.34 | *** part/#asterisk RoyK (n=roy@ip-85-21-149-91.dialup.ice.no) |
| 22:59.49 | hmodes | mebbe the client is expecting auth info or something |
| 23:01.13 | hmodes | tho' it would be sending back a 401 rather then a 404 if that were the case |
| 23:01.17 | variable_office | i have that, here let me show you my sip.conf it might be something there |
| 23:01.26 | timeshell | Is there any value that get's set when you queue a call that can be read to identify within a different context that it originated from a queue? |
| 23:03.41 | TeamINM | has anyone had issues with a digium tdm800p card not loading the child cards? I am getting a zt_chanconfig failed on channel 1: device not config |
| 23:03.41 | *** join/#asterisk Corydon76-dig (i=indigo@pdpc/supporter/bronze/Corydon76-home) |
| 23:03.41 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 23:03.55 | hmodes | does the client have a [sipuser16] peer defined? |
| 23:04.38 | variable_office | hmodes http://pastebin.ca/897941 |
| 23:05.39 | variable_office | i just added all that port stuff, it didnt help |
| 23:06.22 | hmodes | well, i guess my first question is are you actually using qualify for something? |
| 23:06.35 | hmodes | 'cause you could probably just take that out and avoid any further thought ;p |
| 23:06.50 | hmodes | if it's needed, i think i'm stumped |
| 23:07.22 | TeamINM | I am getting a zt_chanconfig failed on channel 1: device not config on my tdm800p - can anyone help me with this? |
| 23:07.45 | variable_office | no but i dont think its the qualify on the client, i think its the qualify on the server that would have to be turned off. qualify off would probably make the messages go away; but the point is that it SHOULD work |
| 23:08.05 | hmodes | yeah, i agree, it should work |
| 23:09.43 | hmodes | i dunno, i use iax2 between my boxes because i don't really trust sip between two *'s ;p |
| 23:10.08 | timeshell | After a queue gives it's call to an agent, I need a way to identify that it actually came from a queue before dialing the agent. Not which queue it came from, but that it came from a queue (true/false). Any ideas? |
| 23:10.13 | hmodes | there's a lot of assumptions about what should and should not work given the number of insanely buggy sip implementations |
| 23:10.16 | hmodes | not so much with iax |
| 23:11.22 | variable_office | ya, this is for a crazy test, i didnt want to produce any more errors than i had to |
| 23:11.36 | variable_office | the client is running newest * too |
| 23:11.59 | TeamINM | can someone just let me know that my post are posting - i'm on a beta client |
| 23:12.00 | TeamINM | thanks |
| 23:12.08 | *** join/#asterisk _darkKnight_ (n=dknight@c9065980.virtua.com.br) |
| 23:12.44 | variable_office | hmodes, the only difference i found was that the "Reg. Contact : sip:sipuser16@10.1.50.60" instead of "Reg. Contact : sip:sipuser16@10.1.50.60:5060" |
| 23:12.51 | variable_office | think that could have something to do with it? |
| 23:13.14 | _darkKnight_ | variables inside global section in AEL are not expanded?? |
| 23:13.36 | variable_office | TeamINM yep can see |
| 23:13.37 | hmodes | shouldn't |
| 23:13.48 | TeamINM | thank you! |
| 23:14.15 | *** join/#asterisk alephcom (n=Darren@h66-112-187-16.mcsnet.ca) |
| 23:14.19 | alephcom | greetings |
| 23:14.27 | timeshell | greetings |
| 23:14.33 | _darkKnight_ | globals { ATA01_DEV=SIP/sipura1; ATA02_DEV=SIP/sipura2; ATA03_DEV=SIP/sipura3; ALL_ATA=${ATA01_DEV}&${ATA02_DEV}&${ATA03_DEV} }; |
| 23:14.36 | _darkKnight_ | like this |
| 23:14.53 | hmodes | any particular reason not to just use iax variable? |
| 23:15.31 | _darkKnight_ | when I Call(${ALL_ATA}); the variables inside ALL_ATA are not expanded |
| 23:15.47 | hmodes | if so, i'd probably say ask around again and if someone can reproduce file a bug |
| 23:16.01 | hmodes | sadly i'm not equipped to try cross registering two 1.4 instances at the moment |
| 23:16.17 | _darkKnight_ | it try to call "${ATA01_DEV}&${ATA02_DEV}&${ATA03_DEV}" and not "SIP/sipura1&SIP/sipura2&SIP/sipura3" |
| 23:16.23 | alephcom | I'm looking for a little wisdom. We installed a TE210 this morning and the channels show up in ztcfg but I can't get them to show in asterisk. |
| 23:16.34 | hmodes | and if i had a reason to, they wouldn't need qualify anyway ;p |
| 23:16.42 | hmodes | so it seems like a pretty rare case |
| 23:19.25 | variable_office | hmodes, is it possible that i somehow need to register sipuser16 to itself? |
| 23:19.31 | variable_office | or something crazy like that |
| 23:19.59 | alephcom | I really need to get his resolved. I'll offer $50 USD as a bounty if anybody can help me resolve it. |
| 23:20.09 | hmodes | well, technically if you trust your network you don't need to register the boxes to each other at all |
| 23:20.23 | hmodes | and the qualify is an extra level of overkill |
| 23:20.49 | hmodes | unless they're dhcp on a private network |
| 23:21.18 | drmessano | ~ron paul |
| 23:21.18 | jbot | ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
| 23:21.28 | drmessano | jbot: forget ron paul |
| 23:21.28 | jbot | i forgot ron paul, drmessano |
| 23:21.42 | variable_office | hmodes, if i do sip show peers it says the status is "unregistered" |
| 23:21.49 | variable_office | err. unmonitored |
| 23:22.10 | hmodes | i think it'll say that if it's not getting an expected response |
| 23:22.15 | hmodes | i.e. the 404 |
| 23:22.26 | drmessano | jbot: Ron Paul called it quits in '08, ZOMG RONPAULAPPLEUBUNTU 2012! |
| 23:22.27 | variable_office | that is on the client though |
| 23:22.51 | drmessano | jbot: Ron Paul called it quits in '08, ZOMG RONPAULAPPLEUBUNTU 2012! |
| 23:22.54 | variable_office | i mean on the client if i type sip show peers it says it is unmonitored |
| 23:22.57 | drmessano | damn thing |
| 23:23.16 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:23.16 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:23.18 | hmodes | the client has qualify=yes? |
| 23:23.25 | hmodes | and a peer for the other box? |
| 23:23.37 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-216-147-93.nsw.bigpond.net.au) |
| 23:23.44 | hmodes | ugh, show me the other box's sip.conf ;p |
| 23:23.55 | drmessano | hmm |
| 23:23.59 | drmessano | bbiab |
| 23:24.07 | *** part/#asterisk _darkKnight_ (n=dknight@c9065980.virtua.com.br) |
| 23:24.27 | hmodes | we'll call it the 'server' i guess |
| 23:24.45 | variable_office | hmodes, my bad, i turned off the qualify to see if that did anything; it didnt; now i turned it on and the status is back to OK |
| 23:25.07 | variable_office | the server is all realtime, itd be hard to show that config |
| 23:25.22 | hmodes | oh dear |
| 23:25.51 | variable_office | what? |
| 23:26.20 | hmodes | adding realtime in to the mix hurts my brain ;p |
| 23:27.18 | hmodes | hrmm |
| 23:27.18 | hmodes | i just noticed in the original sip debug you removed the ip of 'server' |
| 23:27.23 | hmodes | is that because it's a public ip? |
| 23:27.34 | variable_office | yep |
| 23:27.46 | hmodes | this machine has two nics? |
| 23:27.54 | variable_office | the publics and the privates can talk directly without nat in between |
| 23:28.05 | variable_office | no one nic |
| 23:28.30 | hmodes | 'client' is sending it's traffic to 'server''s public ip and receiving the response from same then? |
| 23:28.36 | *** join/#asterisk msolomos (n=m@solomos.kef.forthnet.gr) |
| 23:28.41 | variable_office | yep |
| 23:28.46 | variable_office | each box only has one ip |
| 23:28.46 | hmodes | kk |
| 23:28.52 | hmodes | that's weird, just checking ;p |
| 23:29.41 | variable_office | whats weird |
| 23:29.43 | variable_office | ? |
| 23:29.57 | hmodes | routing between public and private ip space without nat |
| 23:30.03 | msolomos | hi all |
| 23:30.16 | variable_office | ah, ya, the private space is only natted when it is going to non-our ip space |
| 23:30.33 | hmodes | most network guys i've run across think that's tantamount to sin, but meh, i see no reason why it isn't valid ;p |
| 23:30.54 | variable_office | works good for me |
| 23:31.35 | hmodes | i could see it messing with nat/localnet settings tho' |
| 23:31.38 | blitterchip | so who is going to offer kindly some help on a strange really strange thing ? |
| 23:32.05 | blitterchip | ? |
| 23:32.30 | *** part/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 23:32.43 | blitterchip | zap show status works |
| 23:32.44 | hmodes | so yeah, sorry variable, but i'm back to 'this looks buggy' and 'i'd use iax' |
| 23:32.59 | hmodes | i'm stumped |
| 23:33.00 | blitterchip | but it doesn't bring any channels |
| 23:33.03 | variable_office | thanks for the help though |
| 23:33.32 | hmodes | np |
| 23:33.34 | blitterchip | i will keep asking for this untill someone good answer me |
| 23:33.35 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 23:33.43 | blitterchip | anyone out there? |
| 23:34.05 | blitterchip | ? |
| 23:34.12 | tzafrir_home | no |
| 23:34.57 | blitterchip | hi there, we need some specialist in pri's |
| 23:35.06 | tzafrir_home | 'zap show status' simply means that there are zaptel spans on your system |
| 23:35.23 | variable_office | hmodes, i must have changed something because now it says call from 'sipuser16' to extension 'sipuser16' rejected because extension not found |
| 23:35.24 | tzafrir_home | you have to have some channel => lines to get channels |
| 23:35.31 | blitterchip | tzafrir i can tell you that card is properly configured |
| 23:35.40 | blitterchip | but for some reason it doesn't bring us the channels |
| 23:35.45 | blitterchip | it's a TE120 |
| 23:35.57 | tzafrir_home | what is the output of: cat /proc/zaptel/* |
| 23:36.01 | variable_office | nvm, i figured it out |
| 23:36.21 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.72.49) |
| 23:36.46 | hmmhesays | anyone in here using perl for their agi? |
| 23:37.11 | blitterchip | do you want me to paste it here? |
| 23:37.48 | blitterchip | tzafrir |
| 23:38.01 | blitterchip | Span 1: WCT1/0 "Wildcard TE121 Card 0" HDB3/CCS/CRC4 |
| 23:38.01 | blitterchip | <PROTECTED> |
| 23:38.01 | blitterchip | <PROTECTED> |
| 23:38.01 | blitterchip | <PROTECTED> |
| 23:38.09 | blitterchip | and so on so on ..... |
| 23:38.29 | alephcom | hmmhesays: I am |
| 23:38.30 | tzafrir_home | So it is configured, but not (In use) |
| 23:38.46 | blitterchip | aleph can tell you more about this |
| 23:38.57 | blitterchip | we have been struggling for the last 10 hours |
| 23:39.08 | blitterchip | it's damn strange |
| 23:39.10 | tzafrir_home | blitterchip, how have you defined it in zapata.conf ? |
| 23:40.30 | hmmhesays | alephcom are you using the Asterisk:AGI package? I am not |
| 23:40.33 | hmmhesays | nor do I want to |
| 23:40.38 | TeamINM | <PROTECTED> |
| 23:40.52 | tzafrir_home | hmmhesays, alephcom isn't here |
| 23:40.54 | blitterchip | he is quited |
| 23:41.01 | blitterchip | he is my friend |
| 23:41.05 | hmmhesays | I see that now |
| 23:41.15 | hmmhesays | I'm having a hell of a time with this agi, I can't figure out what is going on |
| 23:41.21 | blitterchip | tzafrir is the "irq misses" a worrisome thing ? |
| 23:41.22 | tzafrir_home | and the package is Asterisk::AGI (part of Asterisk from CPAN) |
| 23:41.39 | tzafrir_home | If it's increasing, I guess. Not really sure |
| 23:41.41 | hmmhesays | yeah my bad |
| 23:41.48 | hmmhesays | you know what I meant obviously |
| 23:42.18 | blitterchip | so have you got any idea why although the card is configureed |
| 23:42.22 | tzafrir_home | TeamINM, pastebin your zaptel.conf, and /proc/zaptel/1 |
| 23:42.26 | blitterchip | we can't see any channels in asterisk? |
| 23:43.16 | tzafrir_home | The card has been configured by ztcfg . Now - did you add the proper config in zapata.conf? |
| 23:43.16 | blitterchip | ? |
| 23:43.22 | blitterchip | yes |
| 23:43.28 | tzafrir_home | you said it was configured. What did you mean? |
| 23:43.31 | blitterchip | we did have change it 50 times |
| 23:43.34 | blitterchip | nothing works |
| 23:43.45 | tzafrir_home | pastebin your zapata.conf |
| 23:43.57 | blitterchip | i used zttool |
| 23:44.00 | blitterchip | status is ok |
| 23:44.02 | blitterchip | with no alarms |
| 23:44.05 | tzafrir_home | also: what errors do you get in /var/log/asterisk/messages? |
| 23:44.11 | blitterchip | no errros |
| 23:44.16 | tzafrir_home | pastebin your zapata.conf |
| 23:44.54 | TeamINM | i'm working on getting my zaptel.conf |
| 23:45.13 | TeamINM | what the best web front end for asterisk? |
| 23:45.18 | TeamINM | what's |
| 23:45.23 | blitterchip | tzafrir ? |
| 23:45.27 | tzafrir_home | TeamINM, none, really |
| 23:45.40 | blitterchip | [channels] |
| 23:45.40 | blitterchip | language=en |
| 23:45.40 | blitterchip | ; include zap extensions defined in AMP |
| 23:45.40 | blitterchip | #include zapata_additional.conf |
| 23:45.40 | blitterchip | group=0 |
| 23:45.41 | blitterchip | context=from-pstn |
| 23:45.43 | blitterchip | signalling=pri_cpe |
| 23:45.45 | blitterchip | switchtype=euroisdn |
| 23:45.47 | blitterchip | callwaiting=yes |
| 23:45.49 | blitterchip | threewaycalling=yes |
| 23:45.51 | blitterchip | transfer=yes |
| 23:45.53 | blitterchip | channels => 1-15,17-31 |
| 23:45.53 | scooby2 | pastebin |
| 23:46.02 | blitterchip | sorry |
| 23:46.03 | *** join/#asterisk asteriskrules (n=Darren@h66-112-187-16.mcsnet.ca) |
| 23:46.47 | tzafrir_home | blitterchip, do you have the module chan_zap.so ? |
| 23:46.57 | tzafrir_home | ah, sure you have it |
| 23:46.59 | TeamINM | tzafrir - i think it has to do with the zaptel drivers and the tdm800p |
| 23:47.13 | tzafrir_home | do you have PRI support? 'pri' command? |
| 23:47.33 | TeamINM | this is another person that before, did he tell you its on freebsd? |
| 23:47.33 | blitterchip | http://www.pastebin.ca/897984 |
| 23:47.34 | tzafrir_home | TeamINM, pastebin the output of cat /proc/zaptel/* |
| 23:48.03 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:48.15 | blitterchip | tzafrir hav you seen it? |
| 23:48.30 | TeamINM | is this working? |
| 23:48.32 | TeamINM | ok |
| 23:48.47 | TeamINM | proc zaptel doesnt exist |
| 23:48.48 | tzafrir_home | blitterchip, looks ok |
| 23:48.59 | blitterchip | that's what i a saying |
| 23:49.03 | tzafrir_home | is it readable to asterisk? |
| 23:49.06 | blitterchip | but still no channels in asterisk cli |
| 23:49.07 | TeamINM | no |
| 23:49.18 | tzafrir_home | TeamINM, I don't know zaptel-bsd |
| 23:49.23 | hmmhesays | http://www.pastebin.ca/897987 any perl agi people care to take a look at that |
| 23:49.51 | TeamINM | i thought maybe it was a bsd zaptel problem too, but i get the same results in linux |
| 23:50.15 | tzafrir_home | try: modprobe wctdm24xxp |
| 23:50.22 | TeamINM | both systems see the card, but the x400m isnot there |
| 23:50.22 | tzafrir_home | does it give an error? |
| 23:50.35 | blitterchip | tzafrir are you talking to me? |
| 23:50.47 | tzafrir_home | no |
| 23:51.11 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 23:51.20 | tzafrir_home | blitterchip, ls -ld /etc/asterisk /etc/asterisk/zapata.conf |
| 23:51.54 | TeamINM | bsd doesnt have modprobe, kldstat is the command. it doesnt show anything for any zaptel cards |
| 23:52.40 | blitterchip | drwxrwxr-x 3 asterisk asterisk 4096 Feb 10 01:45 /etc/asterisk |
| 23:52.40 | blitterchip | -rwxrwxrwx 1 asterisk asterisk 237 Feb 10 01:45 /etc/asterisk/zapata.conf |
| 23:52.48 | TeamINM | have you heard of any problems with the x400m and zaptel? like i mentioned, it does the exact same thing in linux |
| 23:53.07 | tzafrir_home | blitterchip, world-writable files are bad for your helath, generally |
| 23:53.27 | tzafrir_home | TeamINM, I have no idea |
| 23:53.33 | blitterchip | what's that supposed to me/ |
| 23:53.42 | TeamINM | alright, thanks for your time |
| 23:53.58 | tzafrir_home | TeamINM, I figure you should get errors when you load the module, or something |
| 23:54.09 | blitterchip | tzafrir you are my last hope |
| 23:54.26 | tzafrir_home | unfortunetly I'm dead tired |
| 23:54.42 | tzafrir_home | try going over the logs. You must be missing something |
| 23:55.01 | tzafrir_home | maybe another asterisk is running and using those channels? |
| 23:55.04 | blitterchip | tzafrir if you solve this i will make you a statue |
| 23:56.46 | blitterchip | big one |
| 23:58.50 | blitterchip | tzafrir |
| 23:58.54 | blitterchip | good night then |
| 23:59.03 | blitterchip | i will built the statue for someoelse |
| 23:59.45 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |