| 00:05.34 | kFuQ | is there some kind of command/variable/agi that could determine what phone was using a particular trunk ? |
| 00:08.40 | tzanger | I swear |
| 00:09.08 | tzanger | my microwave's "sensor reheat" just listens for the food to go POP and splatter the inside of hte microwave and HTEN it says "yup, it's reheated!" |
| 00:10.08 | shidan | maybe u should see if u can get your sokrt device to control it for u |
| 00:11.17 | shidan | use app_microwaveOven |
| 00:11.18 | Strom_TM | shidan, don't do that. |
| 00:11.39 | tzanger | shidan: hehe |
| 00:12.46 | *** join/#asterisk paulc (paulc@S010600062586a0b4.vc.shawcable.net) |
| 00:19.53 | talkwebhosts | hmm |
| 00:27.23 | hmmhesays | woot, roomate bought a new marinater |
| 00:27.37 | hmmhesays | buffalo wings here I come |
| 00:29.13 | tzanger | marinater? |
| 00:29.23 | tzanger | don't you just dump the meat into a dish with the marinade? |
| 00:29.31 | tzanger | (well pierce it first) |
| 00:30.24 | hmmhesays | this is a vaccuum marinator |
| 00:30.56 | tzanger | College is like a woman -- you work so hard to get in, and nine months |
| 00:30.56 | tzanger | later you wish you'd never come. |
| 00:35.33 | *** join/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net) |
| 00:36.28 | Qwell | tzanger: The other day with that relay, is this kinda what you were thinking? http://www.allelectronics.com/cgi-bin/category.cgi?category=500400&item=RLY-2120&type=store |
| 00:37.30 | tzanger | Qwell: yeah that'll work, then you don't need a wall wart either, nbut you have to be careful because now you're dealing with 120VAC to switch it |
| 00:37.38 | Qwell | wall mart? |
| 00:37.52 | tzanger | wall wart |
| 00:37.56 | Qwell | wart..right, whats that? |
| 00:38.43 | tzanger | power supply that plugs into the wall and converts the 120VAC down to something "saner" like 9VDC or somethign |
| 00:39.01 | Qwell | ahh, I misunderstood where the 12xvac was needed I guess |
| 00:39.07 | Qwell | When it says "120vac", thats the coil? |
| 00:39.15 | tzanger | yes |
| 00:39.22 | tzanger | there are coil ratings and contact ratings |
| 00:40.10 | Qwell | I was told it would need at least 90vac on the rest for it to ring, how can I tell if it does that or not? |
| 00:41.02 | mikegrb | heh, the 125 volts you want is the contact rating |
| 00:41.13 | mikegrb | coil is what it takes to switch it, contacts are what it is switching |
| 00:41.22 | Qwell | yeah, none of the ones I'm looking at show the contact rating |
| 00:41.51 | Qwell | I've never really done anything like this, so you'll have to excuse my ignorance on the subject. :p |
| 00:42.06 | mikegrb | not a problem |
| 00:42.19 | mikegrb | lemme get you a link |
| 00:43.04 | tzanger | 90VAC is what you ened for the contact ratings |
| 00:43.19 | tzanger | and all but the smallest relays will have 250VAC rated contacts |
| 00:43.24 | Qwell | ahh |
| 00:43.34 | Qwell | and even those will have > 90? |
| 00:44.03 | tzanger | Qwell: you need to look at the ratings |
| 00:44.16 | Qwell | the three sites I'm looking at, don't give much information |
| 00:44.27 | tzanger | you need better sites then. |
| 00:44.34 | tzanger | I can't tell you if your sites don't have the info either |
| 00:44.49 | mikegrb | http://www.radioshack.com/product.asp?catalog%5Fname=CTLG&category%5Fname=CTLG%5F011%5F002%5F013%5F000&product%5Fid=275%2D241 |
| 00:45.14 | mikegrb | this would work and you can get it around the corner ;) |
| 00:45.26 | Qwell | hmm, radioshack is giving more info, it looks like |
| 00:45.27 | mikegrb | <PROTECTED> |
| 00:45.27 | mikegrb | <PROTECTED> |
| 00:45.29 | mikegrb | <PROTECTED> |
| 00:45.42 | tzanger | mikegrb: he needs DPDT |
| 00:45.46 | mikegrb | Size: nice and small for hiding inside the phone jack |
| 00:45.52 | mikegrb | tzanger: yup |
| 00:45.58 | mikegrb | sorry about that one qwell |
| 00:46.36 | mikegrb | http://www.radioshack.com/product.asp?catalog%5Fname=CTLG&category%5Fname=CTLG%5F011%5F002%5F013%5F000&product%5Fid=275%2D249 <- better |
| 00:46.48 | Qwell | yeah, thats the one I'm looking at now |
| 00:47.14 | mikegrb | the others are the larger ones |
| 00:47.48 | Qwell | and using an npn transistor would be really easy, just take 12vdc off the power supply? |
| 00:47.57 | Qwell | then it could be controlled by a parallel port |
| 00:48.24 | mikegrb | well, Id just take 12 vdc off the power supply and hook it to the coil |
| 00:48.39 | Qwell | I don't want it to always be on when the system is |
| 00:48.45 | mikegrb | when power supply stops outputting, relaw will switch |
| 00:48.46 | mikegrb | oh |
| 00:48.54 | tzanger | mikegrb: he doesn't need a transistor if he's not using electronic control |
| 00:49.00 | mikegrb | tzanger: right |
| 00:49.13 | mikegrb | tzanger: be he wants electronic control it sounds like |
| 00:49.22 | tzanger | and if you're using the 12VDC of the computer snub the coil |
| 00:49.25 | florz | Qwell: and don't forget a recovery diode |
| 00:49.25 | Qwell | yeah, I'd like it to be a system service or something, really |
| 00:49.35 | Qwell | florz: eh? |
| 00:49.40 | tzanger | Qwell: I told you about that |
| 00:49.46 | mikegrb | Qwell: you should be able to find wiring diagram for hooking up the parallel port, you will need a few more components |
| 00:50.25 | Qwell | yeah, need to look further at my logs |
| 00:51.03 | Qwell | tzanger: I was mostly idle when you guys were discussing it the other night, heh |
| 00:51.32 | tzanger | take a nice rectification diode like a 1N4007 |
| 00:51.36 | florz | Qwell: you need a diode with reversed polarity in parallel to the relay in order to shorten the relay coil's self induction voltage so it won't damage your parallel port |
| 00:51.48 | tzanger | leg with the band on it, hook that to +12V and one end of the coil |
| 00:52.07 | tzanger | leg with no band on it, hook that to the other end of the coil |
| 00:52.16 | tzanger | don't get it backward |
| 00:52.35 | mikegrb | florz: :D |
| 00:52.44 | mikegrb | florz: you bring back memories |
| 00:53.03 | mikegrb | I just ended job as an electronics technician |
| 00:53.11 | tzanger | florz: actually it's not the parallel port it'll damage, it's the switching transistor |
| 00:53.39 | tzanger | the high reverse-polarity current flow caused by the magnetic field collapsing when the relay is turned off can damage the transistor |
| 00:53.45 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 00:54.02 | *** part/#asterisk danfrey (user@24.229.228.66) |
| 00:54.22 | tzanger | you won't be controlling a 12V relay with a parallel port without a switching transistor :-) |
| 00:54.33 | florz | tzanger: Well, indeed, the switching transistor is much more likely to be fried =:-) |
| 00:54.35 | tzanger | well... not for long anyway |
| 00:55.17 | florz | tzanger: Well, if the relay's impedance is high enough :-) |
| 00:56.04 | tzanger | florz: I used to drive a he-ne laser with an ignition coil ... the coil was driven off a transistor on the parallel port (easy way to alter the switching frequency) -- it generated enough EMI that the PC would reboot if you changed frequencies |
| 00:56.26 | florz | tzanger: *lol* |
| 00:56.28 | mikegrb | heh |
| 00:56.49 | tzanger | if I knew then what I knew now it'd be trivial to add a little snubber circuit to eliminate that |
| 00:56.57 | tzanger | but back then who cared, I had a frickin LASER, man!! |
| 00:57.54 | libpcp | hi all |
| 00:57.56 | florz | tzanger: The only thing you cared about was your remaining eye, then? =:-) |
| 00:58.15 | tzanger | hehehe |
| 00:58.21 | tzanger | well a little he-ne couldn't damage anything |
| 00:58.34 | tzanger | it was amazing how the dot 'sparkled' off of what you figured was glossy paper |
| 00:59.19 | Sedorox | is there a _good_ howto on how to record and make menus for asterisk anywhere? |
| 00:59.29 | Strom_TM | howto? |
| 00:59.37 | Strom_TM | 1. talk into microphone |
| 00:59.50 | florz | Qwell: If wanna be really safe, use an opto-coupler. But you need an additional power supply then ... |
| 00:59.55 | gambolputty | setup recording on an extension |
| 00:59.58 | Strom_TM | 2. say "for option, press 1. for other option, press 2" |
| 01:00.07 | Strom_TM | 3. play as extension |
| 01:00.09 | Strom_TM | DONE |
| 01:00.34 | Sedorox | duhhh... no I mean like recording it as a wav or what not.. and converting it to whatever format you need to play sounds.. which I'm guessing is gsm |
| 01:01.02 | Strom_TM | Sedorox, 8khz 16bit wav |
| 01:01.07 | Strom_TM | gsm sounds like ass |
| 01:01.28 | silik0n | Sedorox: check the wiki theres info there on doing recordings |
| 01:01.40 | Strom_TM | record at 44.1khz, downsample to 8, save. |
| 01:02.02 | Sedorox | ok... and how do I reference it... just stick it in a sounds dir... then just put background(name-of-file)? |
| 01:02.09 | Strom_TM | yup |
| 01:02.29 | Strom_TM | if it's called ass.wav, then you'd have exten=> 555,1,background(ass) |
| 01:02.29 | Sedorox | /var/lib/asterisk/sounds? |
| 01:02.34 | Strom_TM | yes |
| 01:02.38 | Sedorox | lol |
| 01:02.38 | Sedorox | ok |
| 01:10.40 | *** part/#asterisk hmmhesays (~noway@24-116-232-138.cpe.cableone.net) |
| 01:20.04 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
| 01:21.50 | mrproper_ | im setting up an aix link between 2 asterisk servers to share internal extensions, i've read the guides on the wiki and when i call 1 extension to another extension on the remote asterisk server, the remote asterisk server gives me: NOTICE[49159]: chan_iax2.c:4998 socket_read: Rejected connect attempt from 192.168.x.x |
| 01:22.58 | Sedorox | check make sure the username and pass are correct that its using.. AND the context that your using |
| 01:24.26 | *** join/#asterisk jarnaud (~jarnaud@65.217.47.250) |
| 01:24.28 | jarnaud | Hi all |
| 01:25.15 | jarnaud | Seems the zaptel channels restart from time to time on my box, is it a normal behavior? |
| 01:26.12 | *** join/#asterisk toddf (kz42uidqrg@default.fries.net) |
| 01:26.16 | mrproper_ | Sedorox: all looks ok as far as i can see: http://www.pastebin.com/238629 |
| 01:26.48 | *** join/#asterisk forrestc{hm} (~forrestc@iMach.com) |
| 01:28.16 | forrestc{hm} | any native bridging/codec experts on? |
| 01:28.20 | *** join/#asterisk Rick_Hunter (~rhunter@04-187.008.popsite.net) |
| 01:28.31 | Sedorox | exten => _1XXX,1,Dial(IAX2/asterisk:1234@192.168.200.24/${EXTEN}@asterisk) |
| 01:28.35 | Sedorox | the username is asterisk1 |
| 01:28.37 | Sedorox | not asterisk |
| 01:28.41 | Sedorox | on the 200.24 box |
| 01:28.48 | Sedorox | according to what you have |
| 01:29.00 | Sedorox | and make sure the extention your dialing is in the asterisk context |
| 01:30.07 | mrproper_ | thanks ill check it out |
| 01:30.27 | Sedorox | yup |
| 01:30.55 | forrestc{hm} | Sedorox: know anything about how native bridging is supposed to work? |
| 01:32.18 | *** join/#asterisk ManxPwr (~eric@dsl-208-164-150-160.datasync.com) |
| 01:32.27 | libpcp | is it possible to have an IVR that say something like "the number you dial in not valid" once the sip caller dialed an invalid sip number? |
| 01:32.43 | forrestc{hm} | libpcp: yes. |
| 01:32.51 | forrestc{hm} | libpcp: what exactly are you trying to do? |
| 01:33.48 | file | it's called... the invalid extension |
| 01:34.07 | forrestc{hm} | Actually the invalid extension may or may not work right. |
| 01:35.00 | forrestc{hm} | I've found that including a "invalid" context at the end of my context works also. |
| 01:35.21 | file | if you're using Background it should work fine and dandy, if you're actually dialing out... then yes, use an invalid context |
| 01:39.59 | forrestc{hm} | libpcp: Try doing a search for asterisk and bogons - and dig through the results. Can't find the example page right now myself. |
| 01:40.27 | *** part/#asterisk CoderCR (~creyna@ip68-8-11-127.sd.sd.cox.net) |
| 01:40.53 | Sedorox | forrestc{hm}: sorry.. didn't see it... no... I got iax between to boxes and to iaxtel and fwd.. but I dunno how it works |
| 01:42.29 | forrestc{hm} | Sedorox: I'm trying to figure this out.. the more I dig the wierder it gets. |
| 01:43.01 | Sedorox | lol |
| 01:43.09 | forrestc{hm} | GSM and ilbc calls get natively bridged through to VoicePulse just fine. |
| 01:43.29 | forrestc{hm} | Ulaw won't... For some reason asterisk negotiates a gsm call if you are using G.711 ulaw. |
| 01:43.40 | Sedorox | that is gsm |
| 01:43.52 | hermie | ??? |
| 01:43.55 | Sedorox | I think |
| 01:44.27 | forrestc{hm} | @(#$* caps anyways. |
| 01:45.35 | forrestc{hm} | Hmmm. Guess it's time do dig some more. Can't figure out why VoicePulse won't negotiate the G.711 ulaw. |
| 01:45.38 | hermie | gsm != mulaw |
| 01:46.30 | forrestc{hm} | hermie: Voicepulse supposedly supports gsm, ilbc, ulaw, and alaw. |
| 01:46.58 | forrestc{hm} | hermie: If I originate a gsm call from my softphone into my asterisk, asterisk negotiates a gsm call with voicepulse. |
| 01:47.21 | Sedorox | dunno |
| 01:47.23 | forrestc{hm} | if I originate a ilbc call from my softphone into asterisk, asterisk negotiates a ilbc call with voicepulse. |
| 01:47.51 | forrestc{hm} | if I originate a ulaw or alaw call into asterisk, asterisk negotiates a *gsm* call with voicepulse. |
| 01:48.27 | hermie | forrestc{hm}, pastebin your sip.conf |
| 01:48.34 | hermie | I think I know what the problem is |
| 01:48.39 | forrestc{hm} | hermie: no sip... All asterisk. |
| 01:48.42 | forrestc{hm} | er IAX2. |
| 01:48.58 | forrestc{hm} | still want the sip? |
| 01:49.04 | hermie | forrestc{hm}, pastebin your iax.conf |
| 01:49.32 | forrestc{hm} | let me trim it down a bit... got all the @#()$* asterisk defaults in there still. |
| 01:49.49 | hermie | just dump it all at pastebin.ca |
| 01:52.52 | libpcp | forrestc{hm} thanks alot, i would really appreciate if you can give me an example on that |
| 01:53.08 | libpcp | file: so you have an example for an invalid extension? |
| 01:54.10 | file[laptop] | exten => i,1,Playback(invalid-extension) |
| 01:55.22 | libpcp | file[laptop]: but how can i detect that the user entered an invalid extensions? |
| 01:56.05 | forrestc{hm} | http://pastebin.ca/5368 |
| 01:56.15 | file[laptop] | Background does it automatically |
| 01:56.31 | file[laptop] | well, the logic in asterisk will do it automatically |
| 01:57.15 | forrestc{hm} | libpcp: do you understand what is going on in dialplan.conf? |
| 01:57.26 | file[laptop] | it's extensions.conf ;) |
| 01:57.41 | forrestc{hm} | Obviously. |
| 01:58.20 | forrestc{hm} | That's probably why IT's negotiating gsm with voicepulse... Because I missed something stupid. |
| 01:59.15 | libpcp | file[laptop]: im using AMP right now. im not sure if its included on the extensions*.conf but ill try to figure it out |
| 01:59.30 | file[laptop] | ah well good luck with that |
| 02:00.19 | forrestc{hm} | hermie: did you see my url for pastebin? |
| 02:00.44 | forrestc{hm} | I'm only doing outbound to voicepulse so I don't have an entry in the iax.conf for it. Only in the dialplan. |
| 02:02.20 | forrestc{hm} | Hmmm... maybe I *should* put an entry in iax.conf. |
| 02:02.35 | netsurfer | put a peer entry in |
| 02:02.43 | netsurfer | then u can specify allow all |
| 02:02.51 | forrestc{hm} | This now makes sense. |
| 02:03.03 | netsurfer | ::rolleyes:: |
| 02:03.06 | forrestc{hm} | Assuming it works, that is. |
| 02:03.10 | netsurfer | :oP |
| 02:03.23 | hermie | it will... that's almost certainly your problem |
| 02:03.52 | *** join/#asterisk devious (intrin@c68.112.146.203.stc.mn.charter.com) |
| 02:03.53 | devious | hi |
| 02:04.14 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 02:04.28 | devious | new to this |
| 02:04.36 | devious | where do i put my config's for nufone.net? |
| 02:05.38 | JerJer | follow the instructions that were sent to your email address |
| 02:06.10 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 02:06.15 | devious | ah duh |
| 02:06.16 | devious | lol |
| 02:11.09 | JerJer | GOAL! :) |
| 02:11.29 | devious | anyone know of a voip provider that supports cid/ani spoofing? |
| 02:11.29 | *** join/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
| 02:12.12 | hermie | devious <--- not a good nickname to use and try to get that info |
| 02:12.20 | devious | heh |
| 02:12.46 | samsin | anyone know of a voip provider that supports cid/ani spoofing? |
| 02:12.47 | samsin | ;/ |
| 02:13.04 | Sedorox | lol |
| 02:15.09 | Sedorox | whats a good program for re-sampling wav files? |
| 02:15.34 | JerJer | sox |
| 02:15.46 | JerJer | you should record them in the proper format to begin with |
| 02:16.18 | Sedorox | well.. used a online voice generator thingy.. so I could only download them.. and I wanna use them on asterisk |
| 02:17.35 | kaitseb | how do I define the requested codec for the pstn->iax2 call? |
| 02:18.00 | JerJer | iax.conf |
| 02:18.44 | kaitseb | JerJer: the pstn calls I want to native bridge via other server to gsm client and it doesn't happen , becaus the first server wants ulaw.. |
| 02:19.50 | kaitseb | JerJer: I allow gsm ulaw speex, how I set gsm to be default when sending pstn incomming calls to clients? |
| 02:24.44 | Luke-Jr | Are there any good basic home configurations I could start with? |
| 02:24.56 | Luke-Jr | I have one, but it seems to be too outdated to work :/ |
| 02:24.58 | Sedorox | ok.. where should the sounds file be to be able to do Background(file-name) |
| 02:24.59 | Sedorox | ? |
| 02:25.02 | cypromis | ~seen maerlowe |
| 02:25.14 | jbot | i haven't seen 'maerlowe', cypromis |
| 02:25.14 | cypromis | ~seen marlowe |
| 02:25.14 | jbot | marlowe is currently on #asterisk |
| 02:25.51 | forrestc{hm} | Ok... I can *FORCE* the calls to use ulaw, but it won't automatically pick it based on the client end.... |
| 02:26.16 | forrestc{hm} | I.E. disallow=all, allow=ulaw works |
| 02:26.34 | forrestc{hm} | disallow=all, allow=ulaw, allow=gsm does the same thing. |
| 02:26.51 | forrestc{hm} | as before that is. (I.E. ulaw from softphone gets passed on as gsm) |
| 02:26.59 | nirs | do my eyes decieve me ? |
| 02:27.03 | nirs | cypromis, are you here ? |
| 02:27.15 | nirs | and JerJer ? |
| 02:27.32 | nirs | ok, either I hadn't been here in a long time, or it's my lucky day ;-) |
| 02:29.52 | *** join/#asterisk NatRH (~Nat@dargo.trilug.org) |
| 02:31.28 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 02:31.37 | MrEntropy | yo |
| 02:31.47 | Sedorox | anyone know where background looks for its files? |
| 02:32.12 | Qwell | /var/lib/asterisk/sounds |
| 02:32.18 | MrEntropy | is there a way to Dial(Zap/ but specify a numeric range of Zap cards to fall back on if one's busy? |
| 02:32.27 | *** join/#asterisk harryO (~chatzilla@ool-18bad443.dyn.optonline.net) |
| 02:32.48 | Sedorox | I have the file in there.. but it isn't finding it.. could it be because its a .wav...? |
| 02:36.00 | Sedorox | damn it... no matter what I do... |
| 02:36.17 | sjaak538 | Is _0., allways first in extention even if _048., is above in the extention.conf |
| 02:36.20 | silik0n | try calling it a .WAV |
| 02:36.33 | Sedorox | capital? |
| 02:36.35 | silik0n | yes |
| 02:36.54 | silik0n | it makes a difference on what kinda wave file it is |
| 02:37.20 | Sedorox | Feb 6 19:37:02 WARNING[64227]: file.c:779 ast_streamfile: Unable to open 1smartserv-welcome (format alaw): No such file or directory |
| 02:37.27 | silik0n | theres 2 different types (it escapes me as to what they are right now) |
| 02:37.38 | Sedorox | and I have /var/lib/asterisk/sounds/1smartserv-welcome.WAV |
| 02:40.16 | harryO | anybody know how to fix a static (noise) problem with a tdm400? |
| 02:40.28 | JohnAB | look at IRQ sharing |
| 02:40.44 | JohnAB | cat /proc/interruts |
| 02:41.24 | harryO | i saw a reference to that on the mail list can you elaborate of point me to some info? |
| 02:41.31 | *** join/#asterisk netsurfer (netsurfer@82-133-64-79.dyn.gotadsl.co.uk) |
| 02:41.38 | JohnAB | look at /proc/interrupts |
| 02:41.48 | JohnAB | <PROTECTED> |
| 02:41.51 | JohnAB | something like that would be bad |
| 02:43.30 | harryO | 3: 21616659 XT-PIC wctdm |
| 02:44.37 | JohnAB | any other irqs sharing perhaps? |
| 02:45.41 | harryO | it appears to be one device for each irq |
| 02:45.49 | JohnAB | that's how it should be |
| 02:46.08 | JohnAB | there are other reasons why you can get static |
| 02:46.56 | JohnAB | do you only get the static when you're bridging to FXO |
| 02:47.10 | JohnAB | or also when you just call e.g. the demo |
| 02:48.50 | JohnAB | and is the t400p being used as fxs or fxo or both |
| 02:48.56 | harryO | i place oen or two calls to a sip client |
| 02:49.01 | harryO | they workk fine |
| 02:49.23 | harryO | then very load static and the device won't outdial |
| 02:49.40 | harryO | ^load^loud |
| 02:50.15 | JohnAB | yeah to be honest i think that's one to pass to digium's support, assuming you bought it from there |
| 02:50.32 | JohnAB | i'm assuming you're compiled from a recent CVS source |
| 02:50.36 | harryO | i agree |
| 02:50.38 | harryO | yes |
| 02:50.56 | harryO | i reported this issue last year sometime |
| 02:51.00 | JohnAB | there's nothing too glaringly wrong to be done with the tdm400p |
| 02:51.06 | harryO | (to digium) |
| 02:51.15 | harryO | i did not get much satisfaction |
| 02:51.27 | harryO | i had to put the project down for while |
| 02:51.36 | harryO | and now i am back and facing this issue again |
| 02:51.43 | JohnAB | what's the configuration on the tdm400 in terms of fxs and fxo modules? |
| 02:51.51 | harryO | one fxs |
| 02:52.06 | JohnAB | just one module? |
| 02:52.10 | harryO | yes |
| 02:52.17 | JohnAB | not much to go wrong then |
| 02:52.24 | JohnAB | and some x100s in the same machine? |
| 02:52.33 | harryO | not now |
| 02:52.41 | JohnAB | righit |
| 02:52.44 | harryO | i had an x100 |
| 02:52.48 | JohnAB | and if you do restart now at the asterisk cli |
| 02:52.51 | harryO | i gave it to a friend |
| 02:52.54 | JohnAB | does that resolve the problem temporarily? |
| 02:53.21 | *** join/#asterisk jesse_132 (~jesse_132@12-203-179-57.client.insightBB.com) |
| 02:53.26 | harryO | i didn't try just now |
| 02:53.32 | harryO | when it happened originally |
| 02:53.48 | harryO | i remember i tried shutting down asterisk, did not work |
| 02:53.57 | harryO | rmmod did not work |
| 02:54.06 | JohnAB | only a machine restart fixes it? |
| 02:54.11 | harryO | power bounce worked |
| 02:54.23 | JohnAB | well there's 2 things i'd try then |
| 02:54.31 | jesse_132 | I'm having a problem with pyst (agi-python) stripping out #,* of the results when I do a "get data" ... anyone run into this? |
| 02:54.35 | JohnAB | first is getting the card out, taking the module in and putting it back in |
| 02:54.37 | forrestc{hm} | harryO: there are other people on the list that had the same problem. Are you sure you're running the latest code? |
| 02:54.48 | JohnAB | second is trying it in different PCI slots |
| 02:55.05 | harryO | CVS code |
| 02:55.13 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 02:55.14 | harryO | i just replaced mobo |
| 02:55.21 | *** join/#asterisk Guest^DJ (~some@espeed24-74.brunet.bn) |
| 02:55.26 | JohnAB | and if you have a spare PSU, that would be good to check |
| 02:55.34 | Guest^DJ | ~seen ZX81 |
| 02:55.36 | jbot | zx81 <~ZX81@222-153-18-2.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 3d 14h 8m 23s ago, saying: ':)'. |
| 02:56.00 | harryO | don't have a spare psu, all other boxes are at |
| 02:56.19 | harryO | (not ATX) |
| 02:56.34 | Sedorox | bsd changes /var/lib/asterisk to /usr/local/share/asterisk |
| 02:56.40 | *** join/#asterisk netsurfer (netsurfer@82-133-64-79.dyn.gotadsl.co.uk) |
| 02:57.38 | cypromis | Guest^DJ: I think he is in italy |
| 02:57.43 | JohnAB | i get a similar problem on my tdm400p if it makes you any happier |
| 02:57.57 | Alric | Doesn't everyone? |
| 02:57.59 | Darwin35 | ? |
| 02:58.15 | harryO | well it would if you told me it has gone away |
| 02:58.16 | JohnAB | seems a bit different to yours, but i've given up using it |
| 02:59.27 | JohnAB | modprobe wcfxs lowpower=1 |
| 02:59.36 | JohnAB | you might want to try that when you load the module |
| 02:59.49 | harryO | i will try that |
| 02:59.52 | JohnAB | google brings up quite a few people with similar problems |
| 03:00.04 | JohnAB | none of which have a definitive solution :) |
| 03:00.08 | harryO | i know |
| 03:00.13 | JohnAB | blah 3am, bed time, night all |
| 03:00.23 | harryO | thanks |
| 03:02.15 | *** join/#asterisk john8675309tm (~pabt@207.177.124.87) |
| 03:02.15 | zimdog | When I do sip show peers it shows my ip500 phone that is conencted to the internaet at port 60077 instead of 5060. Where would it pick this up from? I do not see it any any configs |
| 03:02.23 | tzanger | zimdog: sounds like NAT |
| 03:03.07 | zimdog | tzanger: The server is behind a nat firewall |
| 03:03.41 | JerJer | GOAAAAAAL! |
| 03:03.49 | zimdog | I am actually getting close to working now. The phone at least registers I just don't receieve audio and can't dial it. But can dial into the switch |
| 03:03.53 | john8675309tm | Does anyone know any companies with unlimited voip->PSTN calling? |
| 03:04.09 | JerJer | no such thing as 'unlimited' |
| 03:04.21 | john8675309tm | Well as close to unlimited as possible |
| 03:04.38 | JerJer | why not simply pay for what you use? |
| 03:05.53 | john8675309tm | It is a good budgeting tool to be able to pay a for sure price even if it is more |
| 03:06.08 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
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| 03:06.59 | zimdog | tzanger: Is there a way to make the phone outside the nat to use port 5060? |
| 03:07.14 | tzanger | zimdog: first things first -- get the phone and the server on the same network and test |
| 03:07.17 | JerJer | doesn't make finical sense to me...what happens if you want to stop calling for a month? you still pay that monthly fee |
| 03:07.17 | JerJer | financial |
| 03:07.20 | tzanger | if it works, you know you have a NAT issue first |
| 03:08.07 | *** join/#asterisk amir (~amir@shield.guindehi.ch) [NETSPLIT VICTIM] |
| 03:08.07 | john8675309tm | so what is the best pre pay iax and sip provider |
| 03:08.38 | zimdog | The phone is at a different location. I do have 2 phones exactly like ti that I am using the same config file |
| 03:09.19 | john8675309tm | I guess when you sit down and do the math $20 does get you over 1000 minutes |
| 03:09.23 | john8675309tm | at most places |
| 03:10.14 | john8675309tm | is voicepulse pretty good? |
| 03:10.23 | *** join/#asterisk E|nyPRI_ (E_nste_N@S0106000bdb97681e.wp.shawcable.net) |
| 03:10.31 | E|nyPRI_ | hi |
| 03:11.28 | E|nyPRI_ | I've got to asterisk boxes. I want to dial from the dialplan in box a -> box b, do something on box b, and have it return to box a dialplan. Anyone know how to do that? |
| 03:11.56 | tzanger | zimdog: not really... depends on your NAT implementation and even then it's iffy |
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| 03:13.22 | zimdog | tzanger: Thanks guess I will just keep changing things and see what happens. It just seems like I am missing a port somewhere |
| 03:13.27 | mogorman | hey bkw you around? |
| 03:15.32 | tzanger | you're not missing a port |
| 03:15.39 | tzanger | that's how NAT works |
| 03:15.49 | E|nyPRI_ | i lost a port once. |
| 03:15.52 | tzanger | you can't have everything come in on one port |
| 03:18.12 | Sedorox | is there a option to allow people to dial extentions on the system from a menu? |
| 03:20.31 | techie | $$$ |
| 03:21.41 | *** part/#asterisk DaveCanoe (~dgilbert@66.96.20.43) |
| 03:22.43 | kaitseb | anyone running asterisk on 64 bit machine? |
| 03:23.32 | *** join/#asterisk Primer (~primus@sh.nu) |
| 03:23.52 | libpcp | anyone is using AMP for asterisk administration? |
| 03:26.43 | libpcp | i would like to add a feature that inform the caller for an invalid extension if they dial a non-existing extension in AMP |
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| 03:29.36 | *** join/#asterisk AvengerX (~h_avenger@201.19.79.52) |
| 03:29.44 | AvengerX | ppl |
| 03:30.05 | AvengerX | When using a tdm400p, a guy gets this error: |
| 03:30.06 | AvengerX | Ouch ... error while writing audio data: : broken pipe |
| 03:30.27 | AgiNamu | lol, the US$100 Virbiage USB phones? |
| 03:30.27 | AgiNamu | I just bought them, same exact thing, for under $30 |
| 03:30.32 | AvengerX | and asterisk keeps looping this error until killed |
| 03:30.33 | AgiNamu | MOH |
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| 03:30.45 | AgiNamu | that's probably the mpg123 thing |
| 03:30.51 | AgiNamu | he's probably using the wrong version |
| 03:30.57 | AvengerX | hmmm |
| 03:31.08 | AgiNamu | 30 freaking dollars! and Virbiage wants $100!!! LOL |
| 03:31.19 | AgiNamu | and that's without any bulk discount |
| 03:31.38 | AvengerX | ppl do what they can (and can not) to get money, you see... |
| 03:31.38 | AvengerX | :) |
| 03:34.55 | AvengerX | woo |
| 03:35.10 | AvengerX | byebye half-of-the-channel :P |
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| 03:37.35 | AgiNamu | every time i open Asterisk in VS2005, it uses 1GB of ram :\ |
| 03:37.53 | AvengerX | woa |
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| 03:38.29 | AvengerX | you play how much nodes? |
| 03:38.33 | AvengerX | say how much concurrent connections average |
| 03:38.48 | AvengerX | woo, there is my question, up there! :P |
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| 03:40.28 | libpcp | maybe anyone can help me on how to implement an invalid extension ivr option on AMP |
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| 03:40.41 | AgiNamu | how much nodes? |
| 03:41.39 | AgiNamu | meh, PHP and so on suck |
| 03:41.39 | AgiNamu | I'm writing an asterisk editor in C#... that's nice |
| 03:44.07 | AgiNamu | you're welcome. |
| 03:46.53 | AgiNamu | fuck, really, how fuckign hard is it to just write // some comment? |
| 03:46.53 | AgiNamu | sick and tired of shit code with no comments |
| 03:46.53 | kFuQ | <PROTECTED> |
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| 03:46.56 | kFuQ | <PROTECTED> |
| 03:46.57 | kFuQ | http://voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP |
| 03:46.58 | kFuQ | <PROTECTED> |
| 03:46.59 | kFuQ | haha |
| 03:47.05 | AgiNamu | And people using a full freaking int, when an "int var:1" would work |
| 03:47.21 | AgiNamu | and then going batshit crazy and making FLAG_MACROS_FOR_NO_REASON |
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| 03:47.47 | AgiNamu | like chan_zap. gonna go thru and make it use less memory |
| 03:48.27 | AgiNamu | Oh wait, nevermind. It's 10,000+ lines long, and the majority of the variables are not commented. so this means going thru 10000 lines of code for each damn thing |
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| 03:49.31 | AgiNamu | kFuQ what's funny? |
| 03:50.21 | AgiNamu | except that they use a cronjob instead of just droppping a .call file |
| 03:55.38 | libpcp | anyone can help me how to visualize the concept on creating an asterisk billing? i would like to work on reseller section |
| 03:56.24 | AgiNamu | what do you mean |
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| 03:58.38 | libpcp | AgiNamu: i have a postpaid billing system written in php, now i want to create a reseller |
| 03:59.03 | libpcp | i just want to know how the reseller works |
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| 03:59.19 | AgiNamu | should be as simple as adding something to keep track of which clients belong to whom |
| 03:59.20 | AgiNamu | and then calculating how much money th reseller gets |
| 03:59.37 | AgiNamu | well, that's sorta up to you |
| 03:59.44 | AgiNamu | they can get a fixed amount of money |
| 03:59.46 | AgiNamu | percentages |
| 03:59.48 | AgiNamu | whateve |
| 04:00.04 | JerJer | lol |
| 04:00.05 | JerJer | simple |
| 04:00.06 | JerJer | ok |
| 04:00.36 | AgiNamu | it's a simple thing, until you get into the details |
| 04:00.55 | Sedorox | how do I make it where someone calls in.. and the automated menu picks it up.. and they have the option to dial a system extention.. how do I allow that? |
| 04:01.05 | AgiNamu | and i cant imagine libpcp means that he wants us to write a functional spec for him |
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| 04:01.08 | AgiNamu | i mean, Vonage does it nice and easy. Get $5 for everyone who signs up. or something like that :P |
| 04:01.24 | harryvv | afi, and thats a good idea to. |
| 04:01.33 | harryvv | AgiNamu yea that works. |
| 04:01.34 | harryvv | :) |
| 04:01.42 | AgiNamu | "works" |
| 04:01.46 | AgiNamu | not gonna get you "power-resellers" |
| 04:01.55 | harryvv | do thay provide a ata? |
| 04:02.00 | libpcp | okay |
| 04:03.10 | libpcp | well ill try to figure that out to write in php so i could incorporate to my existing billing system |
| 04:03.23 | AgiNamu | who? |
| 04:03.23 | AgiNamu | anyone here know chan_zap? |
| 04:03.36 | harryvv | what about it |
| 04:03.46 | AgiNamu | i want to reduce the memory usage |
| 04:03.51 | AgiNamu | doing elegant code |
| 04:04.00 | AgiNamu | before someone hits with with the FLAG_UGLY_MACROS |
| 04:04.16 | harryvv | no clue |
| 04:04.32 | AgiNamu | unfortunately, of the tons of fields, only a few have documentation. so its hard to know which fields are ints, and which are booleans as integers because no one thought of typedefs or anything |
| 04:07.17 | harryvv | :) |
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| 04:14.00 | AvengerX | AgiNamu: u said * eats up 1g ram upon running it; so I wondered how much ppl use your * server (so, how much lines you have) to do a so-great load |
| 04:14.14 | AvengerX | curse my poor english :) |
| 04:14.20 | harryvv | AgiNamu I have not looked at asterisk code its c right |
| 04:14.21 | harryvv | ? |
| 04:14.34 | AgiNamu | Avenger, No, Asterisksource when loaded in Visual Studio 2005 eats 1GB |
| 04:14.37 | AgiNamu | yes, it's ... c |
| 04:14.47 | AvengerX | ah.. got it |
| 04:14.56 | harryvv | thats cool |
| 04:14.57 | harryvv | ;) |
| 04:16.21 | AgiNamu | wait till you try to modify the code first |
| 04:16.34 | AgiNamu | so, if I have a packed structure, and it has "int x:1", and then I pass x as an int |
| 04:16.43 | AgiNamu | C will zero-extend my value, right? |
| 04:17.07 | AvengerX | so u're building * on win32... I found VS too big... I wanted to make a 200k-app to catch mouse movement, but Vc++ and dx sdk requires more than 1gb to be installed (perhaps more than 1.5gb as dx sdk installation does some insane things during installation) |
| 04:19.32 | AgiNamu | no, im editing Asterisk with VS |
| 04:20.09 | AgiNamu | I could build it from VS |
| 04:20.14 | AgiNamu | but I haven't bothered to set it up |
| 04:20.49 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-102.modem.logical.net) |
| 04:20.56 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 04:24.16 | AvengerX | hmmm |
| 04:25.14 | Carp1 | hmmmmmm |
| 04:28.28 | *** join/#asterisk suma (~suma@81-86-228-192.dsl.pipex.com) |
| 04:28.33 | AgiNamu | hmmmmmmmmmmmmmm |
| 04:29.06 | suma | May I know what RFC standard followed by the conf file in asterisk ? |
| 04:29.37 | Silik0n | who was writting that c# lib for manager? |
| 04:29.37 | Nivex | suma: no you may not... that because it doesn't follow one. |
| 04:29.47 | Silik0n | msg me if you are around |
| 04:30.15 | suma | Nivex: what you mean ? |
| 04:30.31 | suma | Nivex: It is not following any of the existing standards ? |
| 04:30.35 | Nivex | suma: The asterisk config files follow no published standard to my knowledge. |
| 04:30.44 | suma | oh ok thanks |
| 04:30.51 | AgiNamu | later all |
| 04:31.05 | suma | but it looks stupid thing right ? |
| 04:31.12 | suma | Is it not ? |
| 04:32.35 | AvengerX | lol |
| 04:37.36 | brc_ | suma, eh? |
| 04:38.06 | brc_ | suma, teh format used by the asterisk config files has just evolved naturally...no rfc's... |
| 04:38.16 | *** join/#asterisk dolson (~dana@Sudbury-HSE-ppp3979025.sympatico.ca) |
| 04:38.17 | brc_ | and man does it suck |
| 04:38.37 | AvengerX | perhaps it becomes a rfc one day :) |
| 04:38.43 | brc_ | no |
| 04:38.45 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 04:38.54 | AvengerX | just kiddin' |
| 04:39.03 | *** join/#asterisk jterrero (~some@ool-43576e0d.dyn.optonline.net) |
| 04:39.12 | ta[i]nted | are there certain file permission restrictions for a .call file? |
| 04:39.25 | ta[i]nted | i have 744 and * isn't picking it up.. only 644 picks up |
| 04:39.27 | brc_ | writable by asterisk |
| 04:39.42 | brc_ | asterisk running as root? |
| 04:39.57 | ta[i]nted | just on the test machine yea |
| 04:40.02 | ta[i]nted | is that the problem? |
| 04:40.04 | brc_ | shouldn't matter then |
| 04:40.05 | brc_ | no |
| 04:40.10 | *** join/#asterisk talkwebhosts (~admin@c-24-130-183-95.we.client2.attbi.com) |
| 04:40.11 | ta[i]nted | weird |
| 04:40.23 | ta[i]nted | does .call have to be owned by root? |
| 04:40.27 | ta[i]nted | right now it's owned as nobody |
| 04:41.20 | AvengerX | as brc said, it must be writable by asterisk's user |
| 04:41.37 | AvengerX | if asterisk runs as root, it would be able to write anything anywhere |
| 04:42.01 | ta[i]nted | can someone verify this? |
| 04:42.29 | ta[i]nted | just try a .call file with 744 owned by nobody |
| 04:42.35 | AvengerX | yes you can by ls -l to the file |
| 04:43.16 | AvengerX | try chown root.root <file>; chmod u+rw <file> |
| 04:43.32 | ta[i]nted | i have tried that |
| 04:43.45 | AvengerX | what's the error it pops with? |
| 04:43.52 | ta[i]nted | there are no errors |
| 04:44.05 | ta[i]nted | * simply ignored the .call file in the /outgoing directory |
| 04:44.41 | ariel_ | so anyone tried the softphone from snom yet the 360? |
| 04:45.50 | AvengerX | hmmm no idea, sorry. |
| 04:46.36 | suma | brc_: atleast they should have followed some grammer for parsing the file |
| 04:46.44 | brc_ | yup |
| 04:46.52 | suma | brc_: is the grammer available ? |
| 04:47.08 | suma | or is it so simple ? |
| 04:47.11 | suma | just |
| 04:47.15 | suma | [section[ |
| 04:47.17 | suma | [section] |
| 04:47.21 | suma | name=value |
| 04:47.26 | suma | this is the format ? |
| 04:49.15 | *** join/#asterisk Evanrude (~david@ip68-96-125-184.lu.dl.cox.net) |
| 04:50.44 | brc_ | anybody kknow of a viewcvs site for asterisk? |
| 04:50.51 | brc_ | uhm |
| 04:50.56 | brc_ | yeah I guess so suma |
| 04:50.58 | libpcp | anyone is using AMP? how do i include a feature for invalid extension when someone dial a non-existing numbers? |
| 04:51.00 | brc_ | more or less |
| 04:51.13 | brc_ | libpcp, use the magic extension "i" |
| 04:51.26 | letherglov | suma, it's more like a windows 3.1 INI file |
| 04:51.31 | letherglov | but it's not really |
| 04:51.41 | suma | yes |
| 04:51.59 | *** join/#asterisk soulz22 (~Soulz-@host-137-132-45-27.imcb.nus.edu.sg) |
| 04:52.00 | brc_ | http://digium-cvs.netmonks.ca/viewcvs.cgi/asterisk/ |
| 04:52.09 | brc_ | basically, it sucks. |
| 04:52.13 | soulz22 | alo all |
| 04:52.51 | libpcp | brc_: something like? exten => i,1,Playback(invalid-extension) |
| 04:53.01 | ariel_ | libpcp, I have asterisk@home that brings amp. (I only use the amp part for the reports. I make all my own conf files over to my own setup). |
| 04:53.07 | brc_ | libpcp, yes |
| 04:54.39 | libpcp | brc_: but where should i add that line on the conf file? |
| 04:55.00 | brc_ | uh |
| 04:55.11 | brc_ | how would I know? |
| 04:55.17 | libpcp | ariel_: thats nice. well im still on the stage of learning :( |
| 04:55.29 | brc_ | do you know how contexts work? |
| 04:55.36 | brc_ | if not, go read the wiki |
| 04:55.43 | suma | brc_ : i thought you are magician to insert the line into conf file ;) |
| 04:55.49 | brc_ | nope |
| 04:55.59 | ariel_ | libpcp, look at the files in the /etc/asterisk/ directory that end with .conf. You need to read them and follow there logic. |
| 04:56.01 | brc_ | what you might do is make an [invalid] context, and then include the invalid context in all the other contexts |
| 04:56.30 | brc_ | look at /usr/src/asterisk/configs/name.conf.sample |
| 04:56.48 | ariel_ | libpcp, since I do not like there way of doing things like the macro's or not allowing you to use zap ports and requiring little things. I did away with there setups and made my own. |
| 04:56.48 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 04:57.04 | libpcp | okay let me try that brc_, i will try to add the context invalid and it to the main context |
| 04:57.59 | libpcp | ariel_: yeah thats what im thinking, because the extensions*.conf uses alot of macro's |
| 04:59.09 | *** join/#asterisk MichaelSaunders (~mick@196.40.69.228) |
| 04:59.10 | ariel_ | libpcp, if you look into the asterisk@home project you will see a very nice setup. It even loads amp. (which I redo my conf.) but the web gui/ and other items are pre-installed and there very easy to edit via http://localip/maint. |
| 04:59.46 | MichaelSaunders | anyone have asterisk succesfully working with msn |
| 05:00.16 | ariel_ | msn ???? as which version. the only one that works via sip correctly is 4.7 |
| 05:00.44 | ariel_ | but why use msn there are many better soft phones out there. |
| 05:01.11 | libpcp | ariel_: okay i will try to look at it. |
| 05:01.43 | soulz22 | hi all, i just got my hd crash for asterisk, reinstalling asterisk cvs, do i have to apply the broadvoice patch? |
| 05:02.15 | ariel_ | soulz22, no it's included. last I saw. at least it's in the stable setup. |
| 05:02.38 | MichaelSaunders | I want to block msn in the office |
| 05:03.06 | ariel_ | msn as uses a port find it and use iptables to block it. |
| 05:03.12 | soulz22 | ariel: somehow i amhaving some problems |
| 05:03.19 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 05:03.31 | libpcp | brc_: i tried adding a context invalid and add that context to the main context but its not working |
| 05:03.31 | soulz22 | ariel: what is ur extension dial/sip bv? |
| 05:03.42 | MichaelSaunders | Yes but I want a replacement for msn |
| 05:03.50 | MichaelSaunders | that has all msns features |
| 05:04.00 | brc_ | libpcp, odd... |
| 05:04.02 | brc_ | good luck |
| 05:04.12 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 05:04.27 | brc_ | just add extension i directly in whatever context you are dialing in and see if that works |
| 05:04.29 | brc_ | I'm out |
| 05:04.35 | libpcp | maybe it should be add on the macro's? |
| 05:04.46 | ariel_ | soulz22, what is the error your getting. I have not used bv in over 3 weeks. I just used them for testing. and decided to use iax providers. |
| 05:06.02 | soulz22 | ariel: ok cool, SIP/2.0 404 Not Found |
| 05:11.53 | ariel_ | well since bkw_ is not here let me say NEXT! |
| 05:13.35 | *** join/#asterisk drumkilla (~russell@12.21.241.80) |
| 05:13.35 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 05:13.51 | ariel_ | MichaelSaunders, I don't use msn for sip dialling. But if your looking at replacing it use the power of google and do a look up for msn clients. |
| 05:15.05 | ariel_ | libpcp, have you looked at the wiki for information about asterisk and dialing rules. If you need to setup things other then what amp uses it's time to learn the setups and make your own. |
| 05:16.21 | ariel_ | I am glad that they have made AMP, Asterisk@home even the rapid. There great for people to get started. But once you need to start adding more of the great features asterisk has you need to start your own conf files. |
| 05:16.34 | MichaelSaunders | ariel_: No i want an internal messenging system I just dont want people talking to there friends at work |
| 05:17.02 | brc_ | http://www.brc007.com/cgi/GlassBowl0010002.jpg |
| 05:17.17 | soulz22 | nice domain:) |
| 05:17.20 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 05:17.28 | soulz22 | brc: what happened to .org? |
| 05:17.34 | ariel_ | MichaelSaunders, ICQ allows you to setup your own biz server. Just block the outbound port and it should give you what you want. |
| 05:17.49 | brc_ | soulz22, eh? |
| 05:18.14 | *** join/#asterisk harryvv (~root@S010600055d210201.vs.shawcable.net) |
| 05:18.16 | soulz22 | brc: sorry wrong person |
| 05:18.55 | brc_ | harryvv, bad bad! |
| 05:20.45 | ariel_ | well it's late and I need to get up early. See you all later. Good night. |
| 05:21.14 | libpcp | guys, i would like to ask if anyone knows a good iax provider? i want to connect my asterisk to an iax provider with a good service |
| 05:21.23 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
| 05:21.26 | Sedorox | fwd? |
| 05:21.35 | MichaelSaunders | ariel_: Trying to find info on it where can I find it |
| 05:21.37 | ariel_ | libpcp, I use race.com |
| 05:21.55 | ariel_ | ~google ICQ server |
| 05:22.45 | libpcp | how about the nufone service. is it good? |
| 05:24.09 | MichaelSaunders | ariel_: That gave me a list of servers but I want to run my own |
| 05:27.35 | tzafrir_laptop | harryvv, I understand you set your identd responder to root as a honeypot to get cracking attempts and you don't really connect as root |
| 05:28.46 | libpcp | ariel_: is race.com lower than the other voip provider? |
| 05:30.06 | *** join/#asterisk ACiDV (Joel29@66.103.213.48) |
| 05:32.05 | libpcp | ariel_: does race.com allowed to have multiple call on every account? |
| 05:32.36 | ACiDV | I'm anxious to meet Digium Staff (don't know who will be there) tommorow in Toronto, Canada :D |
| 05:33.18 | Himeko | yikes, stalker |
| 05:38.07 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 05:38.39 | JerJer | ACiDV: what is going on in Toronto? |
| 05:41.14 | ACiDV | JerJer business talk and demonstration of a GUI for Asterisk PBX... don't have more detail... I'm a developper or the GUI and I'm just comming in toronto (mississauga) |
| 05:41.24 | harryvv | Would changing the asterisk servers hostname to what it should be other then the default loopback address localhost.host.com localhost to a domain name of my choosing interfeer with asterisk talking to a sip client? I am getting this error on CLI Warning [1468] chan_sip.c 624 __sip_xmit of 0x8133dc4 (len 462) to 192.168.10.5 returned -1: Bad file descriptor. I have shutdown the client machine with the softphone restarted asterisk and it |
| 05:41.40 | harryvv | this erorr over and over. Restarted debian and same effect. |
| 05:41.58 | harryvv | Which I dont think was really nessesary. |
| 05:42.55 | harryvv | I configured sendmail to send mail and the host and this occured. But dont know if it was the result of the changes. |
| 05:43.28 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 05:45.21 | Sedorox | night |
| 05:45.21 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 05:46.03 | harryvv | okay might have a solution for this. |
| 05:46.08 | heath__ | i have come to the conclusion that asterisk is sooo where the money is at |
| 05:46.15 | harryvv | why it did not show up before dont know why :) |
| 05:46.22 | harryvv | as long as it works :) |
| 05:46.42 | harryvv | why are you selling packages? |
| 05:46.56 | heath__ | who me? |
| 05:48.19 | harryvv | yes |
| 05:49.17 | heath__ | nah, i think the money is in big call centers |
| 05:49.51 | JerJer | ACiDV: you do realize not one GUI will be able to encapsulate the power of Asterisk |
| 05:52.20 | MichaelSaunders | JerJer: Do you know any free replacements for msn. I want an internal server and client for the lan |
| 05:52.32 | JerJer | gaim |
| 05:52.43 | MichaelSaunders | does it have a server as well |
| 05:52.45 | robl^ | gaim + jabber? |
| 05:52.47 | Qwell | jabber |
| 05:52.48 | JerJer | jabber |
| 05:55.01 | ACiDV | JerJer: Yes I know, but if the gui have good scripts editor, ivr manager, tenantings, etc and is very easy to use... currently it can reproduce 100% of all Asterisk files and the Extensions manager is already powerful to never have to leave the GUI and I can create a lot of stuff in extensions.conf... |
| 05:56.00 | ACiDV | I must go to bed... it's 1h am and have a meeting at 9h :) Good night all ... and asap I will post screenshoot/demo site of this GUI |
| 05:57.05 | Qwell | for some reason, asterisk is going extremely slow all of a sudden |
| 05:57.20 | harryvv | my asterisk is down with this error. |
| 05:58.19 | Qwell | hmm, and stopped playing sounds |
| 05:59.00 | harryvv | wow, my asterisk generated these errors with a hostname change. |
| 05:59.06 | harryvv | its now running |
| 05:59.16 | harryvv | on its default hostname |
| 05:59.22 | harryvv | what would cause that? |
| 06:00.26 | harryvv | hi firestrm |
| 06:00.35 | MichaelSaunders | robl^: Would you use jabber2 or jabber |
| 06:00.37 | firestrm | hey harryvv |
| 06:00.45 | firestrm | wahts new? |
| 06:01.05 | harryvv | ever seen my problem before? I changed the hostname from its default since it did not have one and got errors on asterisk |
| 06:01.13 | harryvv | I changed it back and no more errors. |
| 06:01.25 | firestrm | harryvv, you changed /etc/hostname ? |
| 06:01.32 | robl^ | MichaelSaunders: Jabber has less bugs and is more stable than jabber2.. but jabber2 will become the stable version soon, I think |
| 06:01.55 | harryvv | yes |
| 06:02.11 | harryvv | it was the localhost orginally |
| 06:02.15 | firestrm | harryvv, what did you change it to? |
| 06:02.59 | firestrm | harryvv, oh ya.. ok.. you need to add a reference to localhost 127.0.0.1 in your hosts file.. |
| 06:03.05 | MichaelSaunders | robl^: I tried my hardest to add users to jabber. Is thre an easy way to do it |
| 06:03.21 | firestrm | astrisk expects to be able to lookup localhost |
| 06:03.38 | firestrm | then you can change hosts to what ever you want |
| 06:03.50 | firestrm | er... localhost to whatever you want |
| 06:04.15 | harryvv | firestrm.. this is the default host that is added when debian is installed and worked fine for asterisk. I did add a localhost name for the loopback but did not finish it and worked fine. Went back to make it more complete and got the errors. |
| 06:05.34 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 06:05.43 | harryvv | #127.0.0.1 pbx_01.brightstartel.com pbx_01 |
| 06:05.43 | harryvv | #192.168.10.194 pbx_01.brightstartel.com pbx_01 |
| 06:05.43 | harryvv | # |
| 06:07.29 | firestrm | you also need 127.0.0.1 localhost.localdomain localhost |
| 06:07.42 | firestrm | otw it cant find itsself |
| 06:07.51 | harryvv | I do |
| 06:08.00 | firestrm | hmmm... wierd |
| 06:08.09 | harryvv | Actually did i must have commented it out by mistake |
| 06:08.24 | firestrm | any references to localhost in your .conf 's |
| 06:08.26 | firestrm | ? |
| 06:09.00 | firestrm | harryvv, try uncommenting.. then probbly reboot for good measure.. |
| 06:09.29 | harryvv | I know it was perhaps a mistake ;) |
| 06:10.05 | firestrm | harryvv, its allways the one char errors that screw you up for hours.. |
| 06:10.51 | firestrm | brb fone.. |
| 06:10.59 | harryvv | :) |
| 06:11.16 | harryvv | fire, I made a mental note of it and knew that was perhaps it. |
| 06:19.41 | datareactor | if you are using redhat also change hostname in /etc/sysconfig/network |
| 06:21.02 | *** join/#asterisk Pulu (~chatzilla@64.200.224.158) |
| 06:22.03 | harryvv | data this is debian |
| 06:23.05 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 06:23.12 | firestrm | harryvv, any luck? |
| 06:23.49 | harryvv | yea |
| 06:23.54 | harryvv | I know whats going on |
| 06:24.02 | firestrm | cool |
| 06:24.19 | harryvv | Its was not working because its set as a dhcp server..the host file is a little different then a static one. |
| 06:24.44 | harryvv | As long as I know thats okay |
| 06:24.45 | harryvv | ;) |
| 06:25.20 | firestrm | aaahhh yes that can screw you up too.. |
| 06:25.59 | firestrm | on a different note.. im clearing a space on my bench to try a solaris 10/asterisk machine.. |
| 06:26.24 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:26.31 | dwC- | what is a good frontend gui for asterisk? I need something easy for configuring |
| 06:26.44 | Silik0n | AMP |
| 06:26.55 | dwC- | ok |
| 06:26.58 | Silik0n | if all you want is easy extension configuring |
| 06:26.59 | dwC- | is that web based or? |
| 06:27.03 | Silik0n | web based |
| 06:27.06 | dwC- | k |
| 06:27.18 | dwC- | thanks |
| 06:27.41 | firestrm | web based asterisk configurators are the best way to a completely screwed up config that i know of.. |
| 06:27.59 | harryvv | fire yea need to set this as a static server been a little lazy about it since the ip address does not change :) |
| 06:28.02 | datareactor | dwc try ACTOS it quite easy |
| 06:28.34 | dwC- | i just want a dumbass to be able to add extensions with ease |
| 06:28.47 | dwC- | the dumbass is not me lol |
| 06:29.11 | harryvv | dwc, well what for? for someone you hire to do that? |
| 06:29.44 | harryvv | im outa here firestrm get my message? |
| 06:29.46 | dwC- | no, for a poor non-profit society that cannot hire someone every time they need a extension added |
| 06:29.56 | firestrm | dwC-, make work project.. let them screw up the system with ACTOS then you get to charge them to come in a fix it ;) |
| 06:30.12 | dwC- | lol |
| 06:30.21 | Qwell | ugh, all of a sudden asterisk hangs when I dial anything |
| 06:30.22 | *** part/#asterisk santiago (~santiago@63.245.86.104) |
| 06:31.45 | Luke-Jr | Is there a way to force ulaw *only* if the remote UA is Kphone? |
| 06:32.11 | firestrm | Luke-Jr, deny all, allow ulaw |
| 06:32.27 | firestrm | er or is that disallow all. |
| 06:32.43 | firestrm | iptables getting mixed up with asterisk :) |
| 06:32.44 | datareactor | Qwell is there any error on console ? |
| 06:35.24 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 06:35.50 | harryvv | ; |
| 06:35.52 | harryvv | :) |
| 06:47.41 | talkwebhosts | anyway to record sound to asterisk with a voip phone? |
| 06:47.46 | Qwell | datareactor: looks like its only the console that hangs, when I issue a dial from it |
| 06:48.54 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 06:49.04 | wasim | bonjour monsieur |
| 06:49.21 | Zeeek | mon dieu! |
| 06:49.46 | Zeeek | you are working on that special shipment, that's why you're around... Good man! |
| 06:49.54 | wasim | oui, mon ami |
| 06:50.11 | wasim | btw, do you have a eu vat id? |
| 06:50.19 | Zeeek | sure |
| 06:50.20 | talkwebhosts | anyway to record sound to asterisk with a voip phone? |
| 06:50.28 | wasim | talkwebhosts: record() |
| 06:50.29 | Zeeek | talk yes |
| 06:50.33 | Zeeek | see wiki |
| 06:50.40 | talkwebhosts | arggh |
| 06:50.49 | talkwebhosts | i can't read |
| 06:50.55 | talkwebhosts | so i can't see wiki |
| 06:50.59 | Zeeek | ok |
| 06:51.00 | wasim | talkwebhosts: hire someone |
| 06:51.34 | wasim | whee ... still worth a shit load more than our local money |
| 06:51.43 | talkwebhosts | lol |
| 06:51.48 | Zeeek | so how much are the beta phones now? |
| 06:51.58 | wasim | Zeeek: for you the old price applies |
| 06:51.59 | talkwebhosts | two fifty |
| 06:52.03 | Zeeek | btw are you shipping the adapters too? |
| 06:52.09 | wasim | nope, 1k euro + vat |
| 06:52.14 | djin | Beta phones -> Farfon? |
| 06:52.16 | wasim | Zeeek: yes, with power |
| 06:52.21 | talkwebhosts | wasim tell me more about these phones |
| 06:52.25 | wasim | djin: farfon beta evaluation membership |
| 06:52.26 | Zeeek | hi djin - wasim you need my vat # ? |
| 06:52.39 | wasim | Zeeek: yes please email me |
| 06:52.49 | Zeeek | and PREFERRED RESELLER STATUS comes with that package |
| 06:52.51 | Zeeek | ok |
| 06:52.55 | djin | Well, wouldn't mind to test one of those as well :$ |
| 06:53.03 | Zeeek | gotta pay to get in the club |
| 06:53.15 | talkwebhosts | i would like to know more |
| 06:53.40 | Zeeek | everytime I need to change a password I screw around with pw for an hour before I remember you can use passwd :) |
| 06:53.42 | djin | That's what I was told Saterday evening as well. |
| 06:54.10 | talkwebhosts | hmmm |
| 06:54.15 | wasim | talkwebhosts: farfon.com |
| 06:54.28 | talkwebhosts | app_record.c:117 record_exec: No extension found |
| 06:54.31 | talkwebhosts | any idea? |
| 06:54.35 | talkwebhosts | k wasim |
| 06:55.04 | talkwebhosts | nice phone |
| 06:55.05 | talkwebhosts | ! |
| 06:55.28 | wasim | we're only sending phones to a) companies that would have multiple '000 on order b) you get a 650 euro rebate on the next order |
| 06:55.33 | *** join/#asterisk dasenjo (~dasenjo@201.245.164.29) |
| 06:55.44 | wasim | c) companies who are able to give us input on features etc |
| 06:56.05 | Zeeek | wasim email sent |
| 06:56.34 | Zeeek | d) VPC |
| 06:56.39 | djin | wasim, are you from Farfon, or in the Beta Group? |
| 06:56.45 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 06:56.50 | talkwebhosts | i guess you don't want my money then |
| 06:56.54 | shido6 | ? |
| 06:56.55 | wasim | djin: farfon, its the Farfon Beta Evaluation Program |
| 06:56.57 | shido6 | whats wrong now? |
| 06:57.04 | Zeeek | Very Patient Companies |
| 06:57.08 | dasenjo | Hi, how are you ? I have a little question ? Is it possible to connect a FXO line to a laptop with a X100P like card ? |
| 06:57.19 | Zeeek | shido6 with the world? LOts of shit! |
| 06:57.21 | wasim | yes, very, very patient companies, who are looking for long term relationships |
| 06:57.23 | shido6 | heheh |
| 06:57.30 | wasim | dasenjo: as longs as your laptop has PCI cards, sure |
| 06:57.43 | djin | lol |
| 06:57.46 | Zeeek | gosh it's getting light here |
| 06:57.49 | Silik0n | miniPCI ,, |
| 06:58.00 | Silik0n | ++ even |
| 06:58.29 | *** join/#asterisk thepdakid (~vtrandal@c-24-8-106-135.client.comcast.net) |
| 06:58.40 | Zeeek | "call from , came in on , at , |
| 06:58.42 | talkwebhosts | someone point me to the record docs |
| 06:58.48 | talkwebhosts | i am looking don't see them |
| 06:58.59 | Zeeek | Starter tutorial: |
| 06:58.59 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 06:58.59 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 06:58.59 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 06:58.59 | Zeeek | THE reference of the moment: |
| 06:59.00 | Zeeek | http://www.asteriskdocs.org |
| 06:59.08 | Zeeek | there's a few |
| 06:59.17 | Zeeek | the last one will have record |
| 06:59.36 | djin | talkwebhosts, there some examples on Wike. |
| 06:59.39 | djin | Wiki |
| 06:59.51 | Zeeek | or you could pastebin your attempt |
| 07:00.04 | wasim | with OFDM :) |
| 07:00.06 | djin | My Wikes have little lights in them. |
| 07:00.07 | talkwebhosts | my attempt was just one line |
| 07:00.08 | Zeeek | Wike where they pay people like $0.05 a year |
| 07:00.08 | talkwebhosts | lol |
| 07:00.23 | talkwebhosts | recoard(filename) |
| 07:00.24 | talkwebhosts | thats all |
| 07:00.28 | djin | recoard? |
| 07:00.34 | wasim | you need to answer it first too |
| 07:00.38 | talkwebhosts | recoard dijon |
| 07:00.41 | talkwebhosts | i did answer it |
| 07:00.46 | wasim | recoard? |
| 07:00.48 | talkwebhosts | s,2,Answer |
| 07:00.50 | djin | In what accent is that? |
| 07:00.55 | talkwebhosts | record |
| 07:01.00 | wasim | hallelujah |
| 07:01.01 | talkwebhosts | small keyboard |
| 07:01.04 | Zeeek | it's recoward |
| 07:01.43 | Zeeek | so you don't want me to publish the three lines I was about to paste |
| 07:01.51 | djin | Don't clap to hard, it makes you type errors. |
| 07:02.06 | sskyles | Is there anything particular I should know about running asterisk on a machine with more than one processor? |
| 07:02.22 | djin | Damn, that won't help either. |
| 07:02.31 | djin | sskyles, no. |
| 07:02.48 | wasim | sskyles: a couple of years ago there were issues, but i think they are mostly ironed out |
| 07:02.58 | sskyles | great, so seemless then. |
| 07:03.24 | djin | no guarantees. |
| 07:03.45 | sskyles | Is 1.0.3 still the defacto standard version we should be running? |
| 07:04.00 | Zeeek | wasim you got da mail? |
| 07:04.00 | djin | Sjek the topic. |
| 07:04.07 | sskyles | for production I mean. |
| 07:04.08 | robl^ | 1.0.5 |
| 07:04.25 | Zeeek | many of us stayed with 1.0.3 for produc |
| 07:04.38 | wasim | Zeeek: nyet |
| 07:04.39 | sskyles | thanks. |
| 07:04.49 | djin | Zeeek, why is that? |
| 07:04.57 | Zeeek | wasim maybe bakshish is required to get mail? |
| 07:04.59 | sskyles | pochimoo? |
| 07:04.59 | robl^ | I am still at 1.0.1 with some patches :) |
| 07:05.28 | Zeeek | djin because users get angry about stuff that amuses the rest of us like callerID behaviour differences |
| 07:05.50 | sskyles | ladna |
| 07:05.54 | djin | Aaah. |
| 07:08.10 | djin | Question, if I need to route in incoming call to an external line, * is just logging the incoming part. Is there a way to log the outgoing as well? |
| 07:08.50 | wasim | djin: define log, you might want to try ResetCDR() |
| 07:09.02 | Zeeek | wasim the ML has someone looking for an IAX hardphone again |
| 07:09.14 | djin | I was indeed refering to the CDR. |
| 07:09.41 | wasim | Zeeek: -biz? |
| 07:09.49 | wasim | i'm only on -dev anymore |
| 07:09.56 | Zeeek | users - someone already mentioned farfon |
| 07:10.10 | wasim | bah, vaporfon |
| 07:10.12 | Zeeek | users is where you learn why not to use 1.0.5 :) |
| 07:10.17 | wasim | hehehe |
| 07:10.27 | Zeeek | those bastards are taking our money and not delivering |
| 07:10.34 | wasim | how many posts on -users? |
| 07:10.34 | Zeeek | I should have gone with virbiage! |
| 07:10.48 | Zeeek | about the phone? |
| 07:10.51 | wasim | like daily ... |
| 07:10.58 | Zeeek | too many! |
| 07:11.42 | wasim | -users was a good way to see if your mail service was working on a 5 second basis |
| 07:11.48 | Zeeek | heh |
| 07:11.58 | Zeeek | I use gmail for it |
| 07:12.16 | Zeeek | makes it easy to filter those auto-responders :) |
| 07:13.53 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 07:14.29 | Zeeek | coffee's ready - who wants an espresso while I'm over there? |
| 07:15.26 | djin | Count me in! |
| 07:15.46 | Zeeek | errr I don't have that many cups! |
| 07:16.05 | Zeeek | well, there is the old one we used as a urinal when the toilets were being fixed |
| 07:16.11 | Zeeek | let me see |
| 07:16.32 | Qwell | "room service" |
| 07:17.13 | djin | oh, found one. |
| 07:18.53 | datareactor | how can i upgrade asterisk without loosing the current configurations ? |
| 07:19.05 | wasim | for a second, i misread that as hashish flavoured :S |
| 07:19.10 | shido6 | you can upgrade by doing a cvs up if you want to run cvs head |
| 07:19.11 | Zeeek | data you wont as long as you do not do a make samples |
| 07:19.11 | Qwell | datareactor: You don't have to run make samples again |
| 07:19.17 | djin | data, don't make config. |
| 07:19.17 | shido6 | all of your /etc/asterisk stuff will remain untouched |
| 07:19.23 | shido6 | DONT do a make config |
| 07:19.28 | djin | and no make samples. |
| 07:19.30 | Qwell | but, you'll need to remove everything in /usr/lib/asterisk/modules first |
| 07:19.36 | shido6 | make config actually wont hurt |
| 07:19.40 | shido6 | a make samples will |
| 07:19.46 | Zeeek | HWOEVER real geeks also copy all the conf files to somewhere else JIC |
| 07:19.51 | shido6 | make config will make the init.d stuff |
| 07:20.16 | Zeeek | use nano! |
| 07:20.29 | djin | a real geek, will backip config, make samples and compares for changes. |
| 07:20.31 | Qwell | yeah, I thought it was symlinked :p |
| 07:21.06 | datareactor | Thanks Qwell shido6 Zeeek |
| 07:21.08 | Qwell | bed time... |
| 07:21.08 | Zeeek | a real geek will have wgets all over the place making copies on servers in different countries in case of atomic war |
| 07:21.12 | implicit | use vim |
| 07:21.24 | implicit | wtf are you people doing with all these 'fake' editors? |
| 07:21.26 | wasim | Zeeek: nope, scp, wget will open your secrets up |
| 07:21.45 | implicit | wasim: i agree |
| 07:21.46 | wasim | unless you pgp them first or something, ofcourse |
| 07:21.52 | Zeeek | wasim that's true but they are honey pot secrets that laungh revenge DDOS |
| 07:21.53 | implicit | wasim: which is a waste of time |
| 07:22.05 | wasim | francoretribution |
| 07:22.28 | Zeeek | I am considering a soft DDOS on one of our providers |
| 07:22.49 | djin | oh? |
| 07:22.52 | Zeeek | that means getting hundreds of users to call their free phone number (only for taking orders, hotlmine is expensive) |
| 07:22.58 | implicit | heh |
| 07:23.26 | Zeeek | so the hotile which is like 50c/minute keeps you a minute then said no is avail for like two days |
| 07:23.50 | Zeeek | so I used the email interface which, two days later sent the answer : call the hotline |
| 07:24.03 | Zeeek | meanwhile we were miraculously reconnected :) |
| 07:24.31 | djin | We have the answer, but you have to pay for it. Calll us! |
| 07:24.31 | Zeeek | for non-geeks, it's the same problem with cellphone companies |
| 07:24.43 | Zeeek | that coffee was decent |
| 07:25.46 | Zeeek | OMG now gmail gives you 50 invites at a time. Anyone not have it yet? In the world? |
| 07:26.23 | djin | an build up to max after hangup. |
| 07:26.41 | Zeeek | I have that too : it's a button marked "pause download" |
| 07:26.48 | Zeeek | manual QoS |
| 07:26.48 | datareactor | people says if i want to use sip with net2phone i have to change chan_sip.c from from "Asterisk" to "Cisco ATA |
| 07:27.05 | datareactor | is this bug is fixed now |
| 07:27.25 | Zeeek | maybe you can do it with useragent= in sip.conf |
| 07:27.25 | djin | sounds like a Net2phone 'feature'. |
| 07:27.47 | sskyles | Wow, here's something I never saw before. There is a company called ipVolution that makes a quad span T1 card that supports Asterisk and Mac OS X! |
| 07:27.53 | Zeeek | wasim - mail yet? |
| 07:28.16 | datareactor | Zeeek yes i will try that |
| 07:28.34 | Zeeek | woulodn't you need some verison crap after the agent though? |
| 07:28.40 | Zeeek | version |
| 07:28.51 | djin | Additional platforms such as Mac OS X and Microsoft Windows will be supported Q2 2005. |
| 07:28.56 | djin | Windows?? |
| 07:29.03 | sskyles | Right... |
| 07:29.10 | sskyles | I can wait. |
| 07:29.11 | Zeeek | why not? |
| 07:29.40 | djin | well . . . . |
| 07:29.51 | sskyles | It's my dream... a rack full of Mac X-serves running Asterisk. |
| 07:30.03 | Zeeek | I bet the asterisk intalled versions would quadruple if there was decent hardware suppport |
| 07:30.16 | Zeeek | [for windze] |
| 07:30.18 | Nix | sskyles: digium cads run on OSX too.. |
| 07:30.29 | sskyles | Where are the drivers?? |
| 07:30.42 | implicit | Nix: but they do dsp in software |
| 07:30.43 | Nix | Sangoma also from memory.. |
| 07:30.45 | implicit | Nix: which is slow |
| 07:30.48 | Nix | ahh |
| 07:30.48 | sskyles | It's all I'm waiting for man. |
| 07:30.57 | Nix | sorry.. Linux on PPC with asterisk works.. |
| 07:30.59 | wasim | Zeeek: yep got it |
| 07:31.03 | Nix | with digium cards.. |
| 07:31.05 | Zeeek | ok |
| 07:31.06 | Silik0n | where are the X100P drivers for OSX? |
| 07:31.09 | Silik0n | hah |
| 07:31.15 | Silik0n | yeah LinuxPPC only |
| 07:31.21 | sskyles | That's what I thought... |
| 07:31.24 | implicit | Silik0n: ;) |
| 07:31.25 | Zeeek | isn't the X100P phased out anyway? |
| 07:31.36 | Silik0n | yeah but I still have them why should I part with them |
| 07:31.58 | Zeeek | me too, but when they die I'll have to order modules |
| 07:32.13 | sskyles | implicit, what's slow? |
| 07:32.15 | Zeeek | From: "Wistfully T. Brink" <marcella@chathamnc.every1.net> (FWD:Curiously Exploring New Options an) |
| 07:32.22 | Silik0n | slowarisk is slow |
| 07:32.23 | implicit | slowaris |
| 07:32.31 | implicit | jinx |
| 07:32.33 | Silik0n | unless you put E series hardware under it |
| 07:32.37 | sskyles | Oh yeah, don't I know that.. |
| 07:32.41 | djin | Nice, they have a Dial E1/T1 card as well. |
| 07:32.54 | implicit | sskyles: i was talking about digium cards |
| 07:32.57 | implicit | sskyles: and sangoma |
| 07:33.09 | Silik0n | anyone using sangoma cards w/ S? |
| 07:33.09 | implicit | sskyles: ipvolution is supposed to do it in hardware when it is released |
| 07:33.10 | Zeeek | From: "Crevasses C. Quits" <gallagher@5till.com> (FWD:Alabama Model Seeking Playmate winey) |
| 07:33.12 | wasim | who? sangoma, they have a quad port as well |
| 07:33.16 | Silik0n | s/S/*/ |
| 07:33.17 | wasim | Silik0n: we are |
| 07:33.23 | implicit | wasim: is it nice? |
| 07:33.26 | Silik0n | sangoma has a nice 2 port card |
| 07:33.38 | djin | No, ipvolution. |
| 07:34.03 | Silik0n | wasim: sangoma claims they perform better in the resource depart. is that true? |
| 07:34.26 | sskyles | I will spring for anything that works with Asterisk on my Macs. |
| 07:34.52 | wasim | Silik0n: yes |
| 07:35.17 | wasim | Silik0n: about 30% less |
| 07:35.52 | implicit | wasim did you get my message? |
| 07:36.09 | Silik0n | oh really? |
| 07:36.11 | Silik0n | hmmm |
| 07:36.21 | wasim | <PROTECTED> |
| 07:37.35 | wasim | <PROTECTED> |
| 07:37.52 | wasim | its got both quad-port digium and a single port sangoma in it on test for a telco |
| 07:38.48 | *** join/#asterisk mak_ (~mak@privat.ua-online.net) |
| 07:38.53 | mak_ | hi |
| 07:40.57 | talkwebhosts | hmmm |
| 07:41.20 | talkwebhosts | i notice a distortion in sound when i call my number with two different phones at the same time |
| 07:41.27 | talkwebhosts | any suggestions |
| 07:41.36 | talkwebhosts | i am using a gigabit network card |
| 07:42.55 | Zeeek | it's not even 9AM and I'm already sick of working and I haven't done any yet! |
| 07:43.31 | mak_ | how can I run agi script via asterisk manager interface ? |
| 07:44.18 | Zeeek | hmmmmm : http://lustich.de/pics/bdwthumb.jpg |
| 07:44.50 | talkwebhosts | zeeek looks like you were having fun there |
| 07:44.50 | sskyles | WTF is going on in that pic?!?! |
| 07:45.02 | Zeeek | I'm trying to figure it out! |
| 07:45.04 | sskyles | HAHAHAAAA!!!! |
| 07:45.09 | talkwebhosts | :P |
| 07:45.17 | Zeeek | It's a german site maybe for fat people? |
| 07:45.25 | sskyles | Looks like a game of strip twister. |
| 07:45.31 | Zeeek | Every once in a while a follow back links to certain sites I run |
| 07:45.40 | Zeeek | always a few surprises |
| 07:46.34 | talkwebhosts | have you all had any distortion problems with using asterisk and an sip? |
| 07:46.47 | sskyles | Depends on the codec I guess. |
| 07:46.55 | talkwebhosts | gsm |
| 07:47.04 | wasim | your sip phone does gsm? |
| 07:47.06 | sskyles | that's pretty grainy. |
| 07:47.28 | sskyles | Also, some hardware has volume settings. |
| 07:47.31 | talkwebhosts | i use a voice over ip phone |
| 07:47.42 | talkwebhosts | what codec should i be using? |
| 07:47.56 | sskyles | ULaw perhaps. |
| 07:47.58 | talkwebhosts | asterisk is fine when i dial into it with one number |
| 07:48.05 | wasim | talkwebhosts: is it on your local lan? if so do what sskyles recommended |
| 07:48.05 | talkwebhosts | but with two it gets distorted |
| 07:48.30 | talkwebhosts | so just record,mainmsg:ulaw ? |
| 07:49.09 | sskyles | What kind of CPU are you using? |
| 07:49.10 | talkwebhosts | is that the extension for ulaw? |
| 07:49.16 | talkwebhosts | its an athlon xp |
| 07:49.26 | talkwebhosts | 256 ddr |
| 07:50.19 | sskyles | Where exactly is the distortion coming from? |
| 07:50.24 | sskyles | Your recordings? |
| 07:50.27 | talkwebhosts | yes |
| 07:50.33 | talkwebhosts | when i use two phones at the same time |
| 07:50.34 | talkwebhosts | to call |
| 07:50.36 | talkwebhosts | to test the bw |
| 07:50.44 | talkwebhosts | i hear sdistortion from the recordings |
| 07:51.21 | sskyles | Hmmm, I have no idea what could be causing that. |
| 07:51.37 | talkwebhosts | might be the wireless network eh? |
| 07:52.07 | sskyles | Well, do you notice the noise on the phones while the calls are in progress? |
| 07:52.16 | talkwebhosts | all the comps in my room are connected to a gigabit switch which is connected to a belkin bridge |
| 07:52.21 | talkwebhosts | yep |
| 07:52.30 | talkwebhosts | while the calls are in progress |
| 07:52.33 | talkwebhosts | 1 phone is ok |
| 07:52.39 | talkwebhosts | 2 sounds digital |
| 07:53.14 | sskyles | Could be latency somewhere. I just can't imagin that though if everything is hardwired directly to the switch. |
| 07:53.35 | talkwebhosts | the switch connects to a wireless bridge |
| 07:53.36 | talkwebhosts | :/ |
| 07:53.40 | sskyles | Does the noise seem to be more of a drop out type? |
| 07:53.51 | talkwebhosts | sounds digital |
| 07:53.56 | talkwebhosts | robotic |
| 07:54.04 | sskyles | Wierd. |
| 07:54.12 | talkwebhosts | might be because of the wireless bridge |
| 07:54.13 | *** join/#asterisk [Sim] (florian@clio.obsimref.com) |
| 07:54.17 | talkwebhosts | i will do some testing to see |
| 07:54.30 | [Sim] | morning |
| 07:54.40 | talkwebhosts | morning Sim |
| 07:54.44 | sskyles | Yeah, but the type of noise you describe might also be some kind of inconsistency with codecs or something. |
| 07:54.54 | talkwebhosts | hmmm |
| 07:54.55 | talkwebhosts | weird |
| 07:55.09 | sskyles | You need to experiment with it a little more. |
| 07:55.17 | talkwebhosts | sskyles you able to dial to california |
| 07:55.23 | sskyles | Try connecting things directly and see of the problem goes away. |
| 07:55.25 | talkwebhosts | i want you to hear this |
| 07:56.12 | sskyles | Yeah, but I can't talk at all... Wife and baby sleeping in the next room. |
| 07:56.22 | *** join/#asterisk dtrcka (~dtrcka@dave.poda.cz) |
| 07:56.22 | talkwebhosts | ok |
| 07:56.26 | talkwebhosts | you dont need to talk |
| 07:56.28 | talkwebhosts | just hear it |
| 07:56.39 | talkwebhosts | when you call in |
| 07:56.39 | sskyles | You can send me a copy of the file, I will listen to it. |
| 07:56.41 | talkwebhosts | i will call |
| 07:56.48 | talkwebhosts | the file is fine :) |
| 07:56.58 | talkwebhosts | its when 2 or more calls occur that is not |
| 07:57.01 | sskyles | steve@netmemory.net |
| 07:58.20 | datareactor | Feb 7 11:24:19 ERROR[1927]: chan_zap.c:9435 setup_zap: Unknown signalling metho |
| 07:58.20 | datareactor | d 'bri_cpe_ptmp' |
| 07:58.54 | datareactor | after upgrading to 1.0.5 i am unable to start asterisk |
| 08:03.02 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 08:04.53 | shido6 | wht do u see at the CLI? |
| 08:05.12 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:05.30 | datareactor | Ouch ... error while writing audio data: : Broken pipe |
| 08:07.44 | datareactor | i think this is boz of mpg123 i have upgrade it boz asterisk make install was complaining of older mpg123 version |
| 08:08.37 | Silik0n | make sure you have the right version of mpg123 |
| 08:08.46 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 08:08.56 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 08:09.02 | Silik0n | 0.59r |
| 08:09.38 | datareactor | yes it the same |
| 08:13.02 | Mother_ | hi all |
| 08:13.13 | sskyles | Hi Mom. |
| 08:13.28 | dtrcka | hi |
| 08:13.31 | Mother_ | is it possible to dial out to PSTN on a defined zap channel based on the SIP client's ID, rather than having to prefix the number to reach said channel? |
| 08:13.35 | Mother_ | hi son :) |
| 08:13.54 | sskyles | I bet I could do that in an AGI. |
| 08:14.16 | Mother_ | for the record, I am male, I am straight, the nick is from the movie "Sneakers", character played by Dan Aykroyd |
| 08:14.19 | Mother_ | hmmmkay |
| 08:14.44 | sskyles | I know Dan. |
| 08:15.06 | Mother_ | I get funny questions sometimes on that ;) |
| 08:15.30 | dtrcka | need help. i'm new to asterisk. have a problem with transfers of calls picked up by agents. just dont work. callback agents are logged in on sip channels. i can transfer direct call to sip channel (by 'transfer' button |
| 08:15.36 | sskyles | I went to his house when he was here recording nothing but trouble. |
| 08:15.57 | Mother_ | really??? |
| 08:16.00 | dtrcka | but calls picked up by logged in agent is not transferable - transfer button does nothing |
| 08:16.02 | sskyles | I ate Pita bread sandwiched with his wife. |
| 08:16.09 | Mother_ | LOL |
| 08:16.28 | sskyles | I fixed their electric gate, which was broken on the house they were renting here in florida. |
| 08:16.44 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 08:17.04 | sskyles | 3 story mansion BTW, I still have the pics somewhere. |
| 08:17.26 | Mother_ | hmmm could the 7960 be made to automatically prefix based on the line you pickup for outbound? I've checked around but haven't found a way |
| 08:17.34 | Mother_ | not bad...I want one of those too |
| 08:17.50 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:18.01 | sskyles | I also met his wifes parents and his brother in law. |
| 08:18.05 | sskyles | ...his agent... |
| 08:18.23 | sskyles | Dan and his wife are the only ones with a sense of humor. |
| 08:19.31 | Mother_ | I can't think of a movie I didn't like his performance |
| 08:20.15 | sskyles | The funny thing is, he acts that way most of the time. Most people like him because he's really down to earth. |
| 08:20.39 | Zeeek | who are you evoking? I lost connex |
| 08:20.59 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 08:21.02 | Zeeek | or is it invoking? |
| 08:21.07 | Mother_ | Zeeek: Dan Aykroyd, from whom I took the nick, movie "Sneakers" |
| 08:21.07 | Makenshi | morning |
| 08:21.13 | sskyles | Dan Akroyd. |
| 08:21.15 | Mother_ | mouning |
| 08:21.21 | Zeeek | oh ya, saw it |
| 08:21.42 | Mother_ | it's just that I get funny questions because of that, but I'm just like him, male and straight :D |
| 08:21.49 | Mother_ | well, I hope he is... |
| 08:21.51 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 08:21.56 | Zeeek | heh |
| 08:21.59 | Mother_ | since he is married lol |
| 08:22.00 | sskyles | I think he's the only really famous person I've met. |
| 08:22.01 | mAsH` | morning all |
| 08:22.03 | Zeeek | what difference in today's world? |
| 08:22.44 | Mother_ | I actually had coffee with Michael Douglas once in Mallorca, that's the closest I've been |
| 08:23.01 | sskyles | I've touched lots of famous things however. |
| 08:23.04 | Mother_ | to anyone remotely famous, he has a villa there |
| 08:23.04 | Zeeek | I'd rather have coffee with Sharon Stone, personally |
| 08:23.10 | Mother_ | hehehe me too |
| 08:23.31 | *** part/#asterisk Nix (~Nix@81.213.125.220) |
| 08:23.39 | Zeeek | I keep watching that movie when it's on cable |
| 08:23.51 | Zeeek | the ice pick |
| 08:23.59 | Zeeek | almost worth it... |
| 08:24.10 | sskyles | This may sound stupid, but just a couple of weeks ago I sat in one of the original cars used in the "Dukes of hazzard" TV series. 1 of 21 originals. |
| 08:24.28 | datareactor | chan_zap.c:9077 setup_zap: Signalling must be specified before any channels are. |
| 08:24.42 | libpcp | anyone knows a site where can i find a complete country codes with FIX/MOB numbers? |
| 08:24.47 | Mother_ | have you setup the signalling on the zap channels? |
| 08:24.52 | wasim | sskyles: didn't they trash all of those in one episode? |
| 08:25.20 | sskyles | They trashed most of them, making the remaining ones worth even more money. |
| 08:25.37 | datareactor | Mother i have upgrade now this error starts to comes |
| 08:26.01 | Mother_ | hmmmm |
| 08:26.21 | sskyles | libpcp: I just finished compiling a list of countries and all their mobile phone codes. I had such a freakin' hard time doing it that I'm not giving it away. |
| 08:26.22 | datareactor | i think something have changed in lastest release :( |
| 08:26.28 | Mother_ | and your .conf haven't been overwritten? |
| 08:26.41 | Mother_ | just in case |
| 08:26.56 | datareactor | Mother_ they are old ones |
| 08:27.23 | Zeeek | sskyles take a look at rate listings for voip providers. They often have these lists all ready to go :) |
| 08:28.04 | Mother_ | oh dear, I have to run |
| 08:28.05 | Mother_ | bbl |
| 08:28.06 | sskyles | I couldn't find them anywhere... it was like pulling teeth. I had to scrounge for weeks to build this fucking list. |
| 08:28.15 | *** join/#asterisk lohelle (~post@213.184.212.218) |
| 08:28.37 | Zeeek | datareactor: http://www.google.com/search?q=site%3Alists.digium.com+1.0.5&sourceid=mozilla-search&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official |
| 08:28.38 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 08:28.57 | libpcp | for me my problem is the FIX line of countries, some voip provider make the fix line as a general. |
| 08:29.02 | sskyles | I mean, if someone is calling a mobile phone on an International call, I need to flag it... and this is where my list came in handy. |
| 08:29.16 | dtrcka | please need help with transferring calls from queue agents |
| 08:30.05 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 08:30.29 | sskyles | for example, all cell phones in Ireland begin with 085, 086 and 087. |
| 08:30.33 | *** join/#asterisk voicomm (~voicomm@adsl-145-157.swiftdsl.com.au) |
| 08:31.02 | *** join/#asterisk MacDeath (~david@at.work.web.za) |
| 08:31.07 | MacDeath | Morning all |
| 08:31.13 | voicomm | Hi, does any one here experience the following error message when 'reload' command is issued ? |
| 08:31.16 | voicomm | Morning |
| 08:31.40 | voicomm | Feb 7 19:11:16 WARNING[22983]: chan_zap.c:9773 setup_zap: Ignoring signallingFeb 7 19:11:16 WARNING[22983]: chan_zap.c:9773 setup_zap: Ignoring signalling |
| 08:32.12 | voicomm | Feb 7 19:11:16 WARNING[22983]: chan_zap.c:9773 setup_zap: Ignoring signalling |
| 08:32.18 | ragnar | welllll.. if we want to be picky, it says warning and not error |
| 08:32.23 | ragnar | fwiw :) |
| 08:32.42 | voicomm | well, yo! |
| 08:33.08 | voicomm | and thats probably the reason why it works ok. But I want my asterisk to look clean |
| 08:33.59 | voicomm | I have been googling for a while, found a fewq cases where people have experienced similar error. No solution has been proposed though! |
| 08:34.37 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4117547.sympatico.ca) |
| 08:37.01 | lohelle | I'm going to create a web page/script with login where I can input phone numbers and conference numbers to add users to. When I add 5 phone and conference numbers 5 call files are created and the asterisk server will call these 5 numbers and add them to the connferences I selected.. before I start I was wondering if someone have a script like this I can look at.. (?) |
| 08:43.09 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 08:44.35 | MacDeath | has anyone here used voipfone.co.uk? |
| 08:44.46 | *** join/#asterisk Dickson (Dickson@203.118.44.115) |
| 08:45.03 | Dickson | hmm |
| 08:47.15 | Dickson | question on configuration of cisco pstn gateway: how to I configure it such that multiple users from asterisk can user the 2 out-going lines ? |
| 08:47.30 | Dickson | multiple = 2 simultaneous users |
| 08:51.11 | *** join/#asterisk IsMe (~some@espeed24-74.brunet.bn) |
| 08:52.26 | IsMe | hi guys, if i plan to have * sit beside an old PBX. how do they connect to each other? RS 232? |
| 08:52.56 | MacDeath | what do you mean connect to? |
| 08:52.57 | [Sim] | eh? why do you ened a data connection between them? |
| 08:53.01 | implicit | hehe |
| 08:53.09 | sskyles | IsMe: T1? |
| 08:53.17 | MacDeath | im having a small problem |
| 08:53.18 | implicit | IsMe: E1? |
| 08:53.19 | MacDeath | Feb 7 10:54:30 WARNING[327]: chan_sip.c:7018 handle_response: Forbidden - wrong password on authentication for INVITE to '"David Norton " <sip |
| 08:53.21 | [Sim] | T1/E1 make more sense |
| 08:53.28 | MacDeath | does that mean my SIP provider is not working? |
| 08:53.40 | implicit | MacDeath: it means your password is wrong |
| 08:53.51 | IsMe | sskyles, implicit : POTs |
| 08:54.10 | implicit | IsMe: then connect via analog lines |
| 08:54.13 | implicit | :) |
| 08:54.17 | sskyles | Ugh, only through the use of channel banks. |
| 08:54.32 | implicit | sskyles: dont argue with him |
| 08:54.49 | IsMe | implicit: how is the order like, POTS-> * -> RJ11 cable -> old pbx ? |
| 08:54.50 | sskyles | :) |
| 08:55.07 | implicit | sure that is possible but crap |
| 08:55.58 | sskyles | POTS -> * -> T1/E1 Channel Bank -> old PBX. |
| 08:56.08 | MacDeath | implicit : checked my password with them |
| 08:56.18 | sskyles | God, why analog?!?! |
| 08:57.05 | implicit | MacDeath: did you use a lie detector to see if they are tellin gthe truth? |
| 08:57.14 | sskyles | Does anyone have a HQ copy of that Interpol video for me to download? |
| 08:57.19 | IsMe | hehehe, sskyles:: we only have 11 analog lines |
| 08:58.02 | sskyles | IsMe, well then you really ARE probably stuck with using channel banks. But that's just... <sigh> |
| 08:58.23 | IsMe | argh! channels banks are expensive |
| 08:58.32 | sskyles | They can be. |
| 08:59.01 | sskyles | My Adit 600 cost'd me nothing though. But I had to dive into the trash to get it. |
| 08:59.43 | IsMe | hrm.. i though i could make use of the PBX FXo/FXS |
| 08:59.44 | IsMe | hehe |
| 09:00.29 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:01.20 | *** join/#asterisk Dickson (~loti@203.118.44.103) |
| 09:01.24 | Dickson | hmm |
| 09:05.20 | sskyles | Yeah, but other than that, how would you fit 11 modems in one asterisk box? |
| 09:05.45 | IsMe | true enough |
| 09:06.11 | wasim | you could use 3 TDM04s ? |
| 09:06.26 | IsMe | how about FXS ? |
| 09:06.47 | sskyles | You'll invest at least another $3000 to make it work. You are better off finding a way to eliminate the old PBX alltogether. |
| 09:06.51 | wasim | TDM40s? |
| 09:07.49 | sskyles | Yeah, but even with the TDM's how would you squeeze 11 FXO/FXS into one box? |
| 09:08.27 | IsMe | hmmmm someone once told me i could stick those POTS into * and make use of the PBX FXS |
| 09:08.37 | wasim | you can get 3 of them if you've got independent PCI busses |
| 09:09.24 | wasim | thats what we do, we tie a couple of the PBX POTS and a couple of PBX extensions into * |
| 09:09.39 | sskyles | The problem is, you're just going to run out of PCI slots. |
| 09:10.02 | IsMe | wasim: did it work for you ? |
| 09:10.08 | wasim | i've got 4 X100P and 1 TDM04 in 8 boxes |
| 09:10.21 | sskyles | unless you break out with T1/E1 and channel banks. |
| 09:10.37 | wasim | but T1+cb is the best route, if you've got the $$$ |
| 09:10.38 | IsMe | then u would u need a T1/E1 card |
| 09:10.49 | wasim | so? its only $600 |
| 09:11.04 | sskyles | It's also the cleanest solotion. |
| 09:11.16 | IsMe | i am not that advance yet, still using POTS |
| 09:11.25 | sskyles | Learn. |
| 09:11.26 | wasim | IsMe: listen to sskyles |
| 09:11.46 | wasim | or fund my uberATA :) |
| 09:11.46 | sskyles | Astrisk will force you to learn many new and interresting things. |
| 09:12.00 | IsMe | sskyles: E1 is not easily adailable in this country |
| 09:12.20 | wasim | IsMe: no, no, you get a T1+cb for the internal stuff only, you interface to the telco through FXO |
| 09:12.27 | sskyles | It doesn't need to be available. You are just using it to get to the channel banks. |
| 09:13.03 | IsMe | ok, my incoming is POTS -> * -> channel banks -> extensions ? |
| 09:13.04 | sskyles | Channel banks will convert the T1/E1 to analog POTS FXO/FXS. |
| 09:13.19 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 09:14.20 | sskyles | POTS -> * -> T1/E1 Card -> Channel Bank -> Old PBX |
| 09:14.36 | sskyles | oops, what the fuck was I thinking... |
| 09:14.43 | sskyles | That's totaly fucked up. |
| 09:14.47 | sskyles | hang on a second. |
| 09:14.49 | IsMe | :) |
| 09:14.57 | lohelle | hmm.. can I run E1 (provider) => asterisk => E1 => channel bank? |
| 09:15.22 | IsMe | i am sure u can, lohelle |
| 09:15.45 | sskyles | POTS -> Channel Bank -> T1/E1 Card -> * -> Back out the T1/E1 Card -> to the channel bank -> the old PBX. |
| 09:16.28 | IsMe | i would 2 channel banks + 2 E1/T1 cards ? |
| 09:16.47 | sskyles | No, the channel bank should be equiped with FXS and FXO cards. |
| 09:17.01 | sskyles | It's the same channel bank that I am referring to. |
| 09:17.06 | IsMe | i am i am begining to have some clue here |
| 09:17.11 | IsMe | ops |
| 09:17.14 | IsMe | i think i am begining to have some clue here |
| 09:17.31 | IsMe | 1 channels bank + quad E1/T1 card |
| 09:17.35 | lohelle | sooo... If I need 30 external lines... and I have 6 pci slots.. I can get 1 E1 (provider) => asterisk => 5 E1 => 5 channel banks.. 150 lines total.. :) (with powerful hardware).. :) |
| 09:17.41 | wasim | 1 channel bank + single T1 card |
| 09:17.52 | wasim | e1 channel banks are ridiculously priced |
| 09:18.08 | IsMe | ok wasim |
| 09:18.09 | sskyles | You'll connect your POTS lines into the FXO cards on the channel bank. You'll connect your old PBX to the FXS cards, then a T1 or E1 cables goes from the channel bank into Asterisk box. |
| 09:18.32 | IsMe | sskyles: if thats the case, why would i still need the old PBX ? |
| 09:18.54 | sskyles | Exactly... But you may want to keep the old phones for some reason. I don't know... |
| 09:19.01 | datareactor | sskyless what if we dont want to use our old pbx |
| 09:19.34 | lohelle | 60 external lines = 2xE1. 6 pci slots.. that is 4x6 E1's.. 22x30 = 660 lines.. (max with 2 external and 22 internal E1's) What hardware is needed to pull something like that? |
| 09:19.40 | sskyles | Then you could get away with 3 TDM cards full of FXO modules and IP phones! |
| 09:19.43 | IsMe | if i tell my boss to get rid of the PBX, i must be ready to have my ass toast |
| 09:19.58 | implicit | IsMe: just tell him you want to have it as a 'backup' systep |
| 09:20.01 | IsMe | lohelle: dual xeon ? |
| 09:20.03 | implicit | meaning decommission it |
| 09:20.42 | datareactor | :) |
| 09:20.57 | lohelle | has anyone heard of people using this amount of lines on a single server? |
| 09:21.15 | implicit | lohelle: yes even more |
| 09:21.43 | lohelle | more? how? |
| 09:21.49 | sskyles | I run 96 conversations on a single machine. |
| 09:21.51 | implicit | what do you mean how? |
| 09:21.52 | wasim | lohelle: get a good server board, with multiple pci busses, use IO APIC, you don't need horsepower to run tdm too much, 6 is really pushing it though, i'd always split it over lots of low cost single boxes |
| 09:22.17 | implicit | wasim: or get a lucent tnt max |
| 09:22.20 | implicit | and do it all in hardware |
| 09:22.37 | lohelle | I was thinking about limitation on number of pci slots.. (one machine) |
| 09:22.43 | sskyles | He's going to explode from information overload. |
| 09:23.01 | wasim | lohelle: its the number of pci slots per pci bus thats the key |
| 09:23.20 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:23.26 | IsMe | sskyles: still processing the information, i am only an old XT |
| 09:23.28 | wasim | lohelle: anything over two, and you're asking for trouble |
| 09:24.03 | IsMe | lol |
| 09:24.10 | Zeeek | heh |
| 09:24.12 | datareactor | implicit if i have pris connected to max tnt how it is integrated to asterisk than |
| 09:24.15 | Zeeek | good old wasim :) |
| 09:24.41 | *** part/#asterisk Dibblah (~Dibblah@82-41-243-74.cable.ubr02.dund.blueyonder.co.uk) |
| 09:24.46 | implicit | datareactor: sip |
| 09:24.54 | implicit | datareactor: they push sip out the other side |
| 09:24.54 | lohelle | If a "cheap" E1 => "alcatel reflexes" channel bank existed maybe I could convince the people at my office to replace the alcatel pbx.. :) but there are 200 phones connected to it.. |
| 09:24.57 | implicit | and they are very high quality |
| 09:25.39 | implicit | lohelle: sorry :( |
| 09:26.33 | lohelle | :) didn't think so.. Too bad they didn't take my advice 1 year ago to by avaya + 4620's and 4630's.. :\ |
| 09:27.04 | *** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net) |
| 09:27.12 | shanky | hi, good morning men |
| 09:28.59 | datareactor | implicit if sip clients wants to call pstn how i can forward to max tnt |
| 09:37.37 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 09:37.43 | firestrm | arrgh.. solaris 10 is a pain in the Ass to install.. |
| 09:37.56 | firestrm | way worse than sol9 |
| 09:38.19 | sskyles | You obviously have the OS fettish going on. |
| 09:38.34 | sskyles | I went through that phase about 8 years ago. |
| 09:39.32 | sskyles | My advice, just don't start seriously messing with Macs unless you want one... Because once you've got the Mac bug, there's no turning back. |
| 09:39.51 | firestrm | sskyles, no i just have a bunch of sun hardware that i want to see if i can get asterisk running on.. |
| 09:40.08 | sskyles | I know what you mean. |
| 09:40.28 | sskyles | I wonder what good it is though if the cards won't work. |
| 09:40.38 | sskyles | Just a VOIP gateway or something? |
| 09:40.53 | IsMe | sskyles: back to the hardware, is it possible to POTS -> * -> FXS -> FXO PBX ? trying to get all the fact ready before have my ass toast |
| 09:41.22 | firestrm | sskyles, im not sure the hardware wont work.. have you tried? |
| 09:41.24 | sskyles | It's possible with TDM cards, but not with 11 lines. |
| 09:41.35 | riksta | postel: alive? |
| 09:41.36 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:41.45 | Zeeek | why... |
| 09:41.51 | sskyles | firestrm: None of my Sun hardware has PCI slots. |
| 09:42.18 | postel | riksta: im kicking |
| 09:42.23 | riksta | :) |
| 09:42.38 | IsMe | sskyles: * will need 22 channels, 11 in and 11 out, approx an T1 |
| 09:42.54 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 09:43.04 | riksta | postel: im gonna release ADM later, i suspect. So if you had any additions/changes...could i get them please? |
| 09:43.16 | sskyles | IsMe: Yes, meaning that you'll want to go with the channel bank configuration we were discussing earlier. |
| 09:43.46 | IsMe | sskyles: yes, your suggestion seems logical and make sense |
| 09:43.50 | postel | riksta: i fixed my telnet probs, DND and speaker-out works, im cooking a frontend to get over the *silly* hardcoding of the variables, get it out there, i know where to find you ;-) |
| 09:44.01 | firestrm | sskyles, i have 3 pci slots on my ultra 5 |
| 09:44.14 | sskyles | IsMe: I could probably draw up a diagram if you need more clairity. |
| 09:44.28 | riksta | postel: yeah i was planning on finishing the frontend for the variables, i'll leave that to you |
| 09:44.48 | sskyles | firestrm: I stopped collecting the Sun stuff while the ultra machines were slightly out of my reach. |
| 09:44.53 | IsMe | tks, sskyles appreciate it |
| 09:45.51 | postel | riksta: have you started on it? if you have drop files on a cvs since you're making it public |
| 09:46.04 | firestrm | sskyles, i figure a 500mhz 64bit ultrasparc otta run asterisk nicely if i can compile it |
| 09:46.07 | riksta | no i have just played about with the glade prefswin |
| 09:46.32 | riksta | postel: i havent got a cvs set up yet, i'll do that now |
| 09:46.41 | sskyles | firestrm: I'd be interrested to know if it compiles without error. |
| 09:47.15 | firestrm | sskyles, me too |
| 09:47.34 | IsMe | sskyles: it would look something like POTS->channel bank -> TE110P -> * -> TE110P -> old pbx |
| 09:47.47 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 09:47.58 | IsMe | POTS->channel bank -> TE110P -> * -> TE110P -> channelbank -> old pbx |
| 09:48.11 | sskyles | IsMe: yes |
| 09:48.26 | sskyles | IsMe: I can still draw up the diagram though if you like. |
| 09:48.41 | IsMe | damn, my old XT brain still works, but slow |
| 09:48.43 | postel | riksta: you ltl pet is more useful than i thought it would be, get it out, im sure ppl would contribute, i need something like another weekend for the frontend, get it out there |
| 09:48.54 | sskyles | IsMe: You probably don't need it by now. |
| 09:48.56 | riksta | ok will do |
| 09:49.44 | IsMe | sskyles: i think i know what is going on, i need to look for 1 channel banks = 12FXS 12FXO and 1 T1 card |
| 09:49.46 | IsMe | yes? |
| 09:50.05 | sskyles | IsMe: Exactly. |
| 09:50.14 | IsMe | nice |
| 09:51.24 | IsMe | asterisk will be in command for most calls, old PBX is just basically transfer calls |
| 09:51.38 | sskyles | IsMe: You can get away without the 12th FXS and FXO card if that's the case, but most channel banks handle a fixed number of FXO or FXS. In other cases some cards in the channel banks can have as many as 6 or 8 FXO or FXS channels on each card. |
| 09:52.49 | letherglov | IsMe, look at Adtran's offerings |
| 09:52.52 | letherglov | you have six slots |
| 09:52.55 | letherglov | four ports per card |
| 09:53.00 | letherglov | that's 6*4 = 24 |
| 09:53.12 | sskyles | For example, I would pay between $250 to $380 for one FXO card for my channel bank, but that one card gives me 6 FXO's. |
| 09:53.14 | letherglov | you can get 3-4 port cards for fxo |
| 09:53.19 | letherglov | and 3-4 port cards for fxs |
| 09:54.11 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 09:54.16 | shanky | sorry because I'm really a newbie in asterisk's issues, I need information becuase we have a very large network on a village, and we're planning to use ATAs to offer internal and external telephony |
| 09:54.23 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
| 09:55.05 | shanky | for internal telephony I think it can be done just with the ATAs registering to a asterisk server |
| 09:55.06 | IsMe | letherglov: where do i get Adtran? ebay ? |
| 09:55.09 | wasim | how can a village have a large network? |
| 09:55.45 | IsMe | wow! a village |
| 09:56.00 | shanky | wasim: is a cable network |
| 09:56.34 | wasim | shanky: ata's would work well, thats the standard methodology these days |
| 09:56.51 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 09:57.19 | wasim | shanky: and if you're not doing voip, you can always just use g711 for best quality, since cable networks generally have a lot of free local bandwidth |
| 09:57.44 | wasim | shanky: we're working with a local cable operator who has a smallish network, 50k subs |
| 09:58.04 | wasim | shanky: they are thinking of deploying ata and fons for this service |
| 09:58.46 | shanky | wasim: but I have no idea where to look for information to connect to and from the Analog Phone Network |
| 09:58.55 | wasim | shanky: you need PRI cards from digium or sangoma |
| 10:01.02 | IsMe | wasim: sangoma is new ? |
| 10:03.04 | *** join/#asterisk pranav (dawda_pran@203.115.69.81) |
| 10:03.10 | wasim | IsMe: sangoma.com have been making cards for sometime now, their cards now work with zaptel which is what * uses |
| 10:03.16 | *** join/#asterisk speakman (~speakman@c-38aa71d5.07-39-6f73641.cust.bredbandsbolaget.se) |
| 10:03.35 | speakman | hi folks! :) |
| 10:03.48 | IsMe | wasim: tks |
| 10:06.13 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 10:11.42 | *** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net) |
| 10:11.55 | shanky | sorry but my connection went down |
| 10:13.07 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 10:13.13 | pashah | hello |
| 10:13.16 | Zeeek | I think the world will be a better place when there are farfon products everywhere |
| 10:13.43 | shanky | wasim: the last I could read from you was: 10:58 < wasim> shanky: you need PRI cards from digium or sangoma |
| 10:14.14 | pashah | how do i edit extenstions.conf, so that users have an option to press say "0101" to start recording their conversation? |
| 10:15.16 | christo | I'm having loads of problems trying to compile asterisk-addons on a slackware linux box - I'm getting errors relating to format_mp3. Is there an easy way to get this working? |
| 10:17.21 | pranav | hello i need to help for astguiclient |
| 10:17.53 | pranav | any one is there astguiclient master |
| 10:18.20 | pranav | ergent i need help so please |
| 10:19.55 | pranav | please give me ans |
| 10:20.16 | pranav | hello any one is there |
| 10:20.39 | firestrm | yes, just no one who know the answer to your Q |
| 10:21.00 | pbxjunkie | guys I've got this strange issue: Sometimes after speaking a while I get BAD interference on the SIP phones end, so bad that you can't even listen. Audio becomes horrible. What could be the problem? It happens at random times\ |
| 10:21.34 | christo | sombody's caning your bandwidth? |
| 10:21.36 | pranav | hello firestrm |
| 10:22.23 | pbxjunkie | it's not like I get low bitrate and hence I lose some packets, i start getting horrible static/interference. Could that be due to bandwifth? |
| 10:22.52 | shido6 | static? |
| 10:22.57 | shido6 | like white noise from the TV |
| 10:23.19 | pbxjunkie | ..I mean like.. digital noise. Very intense |
| 10:23.21 | shido6 | like KSHHHHHHH!!!! ? |
| 10:23.27 | Zeeek | . |
| 10:23.40 | pbxjunkie | like... choppy KSH.. like. KSHHH.. KSHSHSH ER#$%!@#TF KSHSHSHS... KSHH.. KSH .. KSH ..KSHHHH |
| 10:23.43 | pbxjunkie | :) |
| 10:23.50 | Zeeek | . |
| 10:24.32 | pranav | firestrm can you know the astguiclint |
| 10:24.34 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 10:24.39 | pbxjunkie | any ideas why?:) |
| 10:24.57 | christo | guys, I'm trying to compile asterisk-addons.. it whinged at me about format_mp3, so I have removed any references to it from the Makefile.. Now I'm getting: cdr_addon_mysql.c:269: warning: assignment makes pointer from integer without a cast. How can I fix this? |
| 10:25.22 | *** join/#asterisk Xcalibur (~gdpe@62.240.241.107) |
| 10:25.24 | djin | don't mess up Makefile. |
| 10:25.36 | sandnigg0r | djin, ;) |
| 10:25.37 | riksta | Juggie: alive? |
| 10:25.50 | sandnigg0r | christo, i would not worry much about warnings |
| 10:26.03 | sandnigg0r | christo, it will still compile |
| 10:26.30 | sandnigg0r | christo, even when you compile a linux kernel you get warnings |
| 10:26.59 | speakman | Anyone here's using X100P? |
| 10:27.08 | djin | sandsomething, don't worry about errors? |
| 10:27.10 | Zeeek | I always have bad assignments without casts in my programs... keeps me awake |
| 10:27.26 | sandnigg0r | djin, not errors but pointer warnings |
| 10:27.31 | *** join/#asterisk Poincare (~jefffnode@dD5779B07.access.telenet.be) |
| 10:27.39 | djin | ah, ok. |
| 10:27.49 | djin | some sand in my eyes ;) |
| 10:27.54 | sandnigg0r | :) |
| 10:28.33 | christo | sandnigg0r - okay. I've made a mistake.. sorry. The actual bit which is breaking the compile is: make: *** [cdr_addon_mysql.o] Error 1 and before that it can't find some things... |
| 10:28.39 | christo | cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory |
| 10:28.43 | christo | cdr_addon_mysql.c: In function `my_load_module': |
| 10:28.55 | christo | probably therein lies the problem, but wherein lies the solution? :) |
| 10:29.00 | djin | ../asterisk must exist. |
| 10:29.20 | christo | oh |
| 10:29.21 | sandnigg0r | hummm |
| 10:29.29 | sandnigg0r | djin, you da man |
| 10:29.38 | christo | I need to symlink asterisk-0.7.1 to asterisk then perhaps |
| 10:29.41 | *** join/#asterisk perkin (~polarisx@194.114-84-212.ippool.ndo.com) |
| 10:29.44 | sandnigg0r | werd |
| 10:29.49 | pashah | why can not I use application monitor for incoming calls? WARNING[1146]: pbx.c:1299 pbx_extension_helper: No application '15,Monitor' for extension (incoming, 199, 2) |
| 10:29.51 | sandnigg0r | thats what that looks like to me too |
| 10:30.00 | IsMe | bye all |
| 10:30.05 | pashah | works fine for outgoing |
| 10:30.40 | christo | okay... that got one step closer on the compile, |
| 10:30.42 | christo | ~pastbin |
| 10:30.59 | djin | Stupid question. I have a Cisco 7912 connected to asterisk (SIP) that receives a call from another asterisk by IAX2 and I want to forward an incoming call to another number? It doesn't work, where could it be wrong? |
| 10:31.02 | christo | pastebin? |
| 10:31.08 | speakman | any one knows if DTMF CallerID is solved? |
| 10:31.26 | christo | damn - no clueful bots :) can somebody throw in a pastebin link pls? |
| 10:31.33 | perkin | Anyone used the Microsoft RTC Client API to register with Asterisk? I'm having trouble registering with Asterisk. |
| 10:32.49 | Zeeek | ~pastebin |
| 10:32.50 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 10:32.57 | Zeeek | ~pastrybin |
| 10:33.01 | djin | OK, djin solved his little question :) |
| 10:33.04 | christo | lol |
| 10:33.39 | Zeeek | djin what was the answer? |
| 10:33.44 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:33.53 | RoyK | ~seen coppice |
| 10:33.55 | jbot | coppice <~chatzilla@245.195.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 1d 6h 51m 33s ago, saying: 'and port'. |
| 10:34.09 | RoyK | well expressed |
| 10:34.29 | djin | Stupid question. I have a Cisco 7912 connected to asterisk (SIP) that receives a call from another asterisk by IAX2 and I want to forward an incoming call to another number? It doesn't work, where could it be wrong? |
| 10:34.44 | djin | Btw, this is solved :) |
| 10:35.23 | christo | ok guys... this is the error I get when trying to compile asterisk-addons: http://pastebin.ca/5381 What on earth does all that mean? |
| 10:36.42 | djin | It this still you special Makefile? |
| 10:36.55 | speakman | How do I patch zaptel after downloading .diff-files from Mantis? |
| 10:37.04 | christo | yeah - well just with the format_mp3 stuff removed |
| 10:37.15 | christo | ^djin |
| 10:37.20 | pranav | hello any one is there astguiclient master |
| 10:37.33 | djin | Why don't you put it back? |
| 10:37.40 | christo | :) will try |
| 10:37.50 | sandnigg0r | christo, i will try the new makefile |
| 10:37.51 | Zeeek | . |
| 10:37.55 | djin | And what was this reference to asterisk 0.7.1 earlier? |
| 10:37.58 | sandnigg0r | christo, errr old good makefile |
| 10:38.01 | sandnigg0r | christo, or |
| 10:38.05 | christo | well now I get the format_mp3 errors again |
| 10:38.06 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 10:38.14 | sandnigg0r | christo, i would give "make -i" a shot |
| 10:38.28 | christo | wwwooah |
| 10:38.29 | djin | Do you have mysql-devel installed? |
| 10:38.32 | christo | what's that -i switch? |
| 10:38.33 | sandnigg0r | christo, these make errors look like they are not your fault |
| 10:38.39 | sandnigg0r | christo, it ignores the erros |
| 10:38.44 | sandnigg0r | christo, errors |
| 10:38.49 | sandnigg0r | christo, so its a iffy build |
| 10:38.56 | Zeeek | what is going on? |
| 10:38.57 | christo | mysql-devel perhaps that's the problem |
| 10:38.59 | sandnigg0r | christo, i would not do a make install just yet |
| 10:39.05 | christo | okay |
| 10:39.20 | sandnigg0r | christo, see if you can test everything without having to do a make install |
| 10:39.28 | christo | I'll make clean and get mysql-devel installed then I'll try again, but I still can't see how that'll fix the format_mp3 errors |
| 10:39.32 | pashah | anybody uses astguiclient? is it usefull or not? |
| 10:39.39 | christo | sandnigg0r - will test it now as is ok |
| 10:39.49 | djin | pashah, talk to pranav. |
| 10:39.54 | *** part/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 10:40.13 | pashah | djin: =) |
| 10:40.16 | pranav | hello pashah |
| 10:40.30 | sandnigg0r | christo, yeah. sometimes i have to force shit to build like that. It sucks but if you can make sure it works then i would not lose too much sleep over it |
| 10:40.30 | djin | oops, I sorry. Pashah. |
| 10:40.40 | pashah | pranav: I though you had troubles installing it? |
| 10:41.00 | sandnigg0r | christo, the make errors you are getting dont look that critical |
| 10:41.10 | sandnigg0r | but you never know right |
| 10:41.50 | christo | well.. I'll just test what I've got, but won't I have to do a 'make install' for the asterisk addons stuff to be usable |
| 10:41.57 | sandnigg0r | yeah |
| 10:42.08 | pranav | i have install it but problem on 6.2 insert phone valus(......... |
| 10:42.23 | christo | and if it's still broke, I can presumably 'make uninstall' and then try plan b..c..d.. |
| 10:42.25 | djin | I has problems compiling earlier that looked like format_mp3 errors, but it was because mysql-devel wasn't installed. |
| 10:42.31 | christo | oh |
| 10:42.32 | sandnigg0r | christo, fuck it do it |
| 10:42.33 | pranav | pashah |
| 10:42.37 | christo | :) lol |
| 10:42.41 | sandnigg0r | christo, heh |
| 10:42.45 | pranav | whats your problem |
| 10:43.02 | sandnigg0r | christo, you got the old makefile if you need it anyways |
| 10:43.12 | pashah | pranav: no problem 8) |
| 10:43.17 | christo | yeah |
| 10:44.46 | Zeeek | . |
| 10:45.09 | Delvar | morning all (just) |
| 10:45.14 | pranav | pashah :whats your problem tell me i will try |
| 10:45.32 | christo | djin - well it was looking for the asterisk direcctory under ../ |
| 10:45.47 | christo | but it didn't exist, cos it was called asterisk-0.7.1 |
| 10:45.49 | *** join/#asterisk dtrcka (~dtrcka@dave.poda.cz) |
| 10:45.52 | dtrcka | hi |
| 10:45.54 | christo | seemingly |
| 10:46.43 | djin | your working with a 0.7.1 version? |
| 10:46.55 | jerlique | Any clues with this error: |
| 10:46.59 | jerlique | WARNING[66546]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! |
| 10:47.19 | christo | yeah - well it's not my machine, so I don't want to frig with too much incase it affects things that my 'colleagues' are doing |
| 10:47.50 | djin | What version is asterisk-addons then? |
| 10:48.29 | christo | latsest I guess - I just did a cvs update for asterisk-addons |
| 10:48.45 | christo | is that gonna arse things up if they're not the same version? |
| 10:48.46 | RaYmAn-Bx | you can't just use different versions of asterisk and asterisk-addons |
| 10:48.53 | christo | aaww shit |
| 10:49.14 | christo | so where can I get an older version of asterisk-addons? |
| 10:49.24 | djin | get a newer version of asterisk |
| 10:49.40 | christo | hmmmm |
| 10:50.00 | djin | 0.7.1 -> 1.0.5 is quite an upgrade. |
| 10:50.16 | pashah | pranav: there is no problem I was just wondering if astguiclient of any use or it isnt |
| 10:52.50 | dtrcka | is it a bug or feature that calls picked up from queue by callback agents cannot be transfered? |
| 10:52.51 | pranav | pashah do you know the astguiclient point no 6.2 insert into phone values( |
| 10:54.17 | Nix | lol @ dtrcka |
| 10:56.34 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 10:57.35 | Astinus_ | Hello, i have a question about telephones...does the ordering of cables matter when making a telephone cable, i.e. do the order of colors have to match at both ends? |
| 10:57.46 | Zeeek | India has the lowest rate of prostate cancer in the world |
| 10:58.12 | Zeeek | supposedly because of curry powder |
| 10:58.17 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 10:59.00 | *** join/#asterisk dtrcka (~dtrcka@dave.poda.cz) |
| 10:59.10 | riksta | postel: alive? |
| 10:59.33 | datareactor | try ~postel |
| 11:00.06 | postel | im kicking, i have some grid probs |
| 11:00.24 | jerlique | astinus: generally yes. it all depends on what type of cable you are making. |
| 11:00.45 | *** join/#asterisk UrBaNLeGeNd (~cron@202.61.38.213) |
| 11:01.13 | Astinus_ | jerlique: a cable .. from a telephone set to the wall output |
| 11:01.28 | Astinus_ | jerlique: just wondering if it can be crossover or not. |
| 11:02.16 | jerlique | depends on telephone type, eg standard pstn generally isn't fussed. with PBX phone it is important to be correct. |
| 11:02.50 | Astinus_ | "isn't fussed" ? |
| 11:03.33 | jerlique | ie a normal pstn telephone can have the wires crossed over. (at least in AU anyway). |
| 11:03.46 | Astinus_ | ok. |
| 11:03.57 | speakman | Astinus_: how many wires? |
| 11:04.20 | Astinus_ | 2 |
| 11:04.57 | speakman | What does this mean: Reversed (or previously applied) patch detected! Assume -R? |
| 11:09.32 | Zeeek | . |
| 11:10.13 | jerlique | 2 for PSTN (centre two pins) |
| 11:12.11 | Astinus_ | jerlique: ok a tad complex thingie here, three outlets in the house they're all connected but atm to ptsn but i want to connect them to a voip box |
| 11:19.59 | jerlique | well you need a device such as Sipura SPA-2000 to allow this to happen. |
| 11:22.09 | Astinus_ | yes |
| 11:22.10 | jerlique | Agent Logout feature, the code onthe voip info site appears incorrect. Can someone offer help. |
| 11:22.11 | Astinus_ | i have that one :P |
| 11:38.25 | jerlique | how do agents logout ? |
| 11:39.35 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 11:40.32 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 11:43.01 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 12:00.05 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 12:01.11 | lohelle | anyone using the avaya 4630 phone => asterisk.. can get one (6 months old) for 100$ :) |
| 12:01.23 | dtrcka | why agents cant forward incoming call? |
| 12:01.49 | dtrcka | (I mean transfer, not forward) |
| 12:10.56 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:12.56 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 12:24.27 | *** join/#asterisk wasim (~wasim@203.81.200.8) |
| 12:24.31 | *** join/#asterisk satlink (satlink@66.178.97.50) |
| 12:24.58 | empire667 | can anyone help me with capi, modprobe capi gives no error , * says:capi not installed |
| 12:28.51 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
| 12:29.51 | riksta | Hey guys, I have just released an alpha version of ADM - Asterisk Desktop Manager if anyone would like to help test it then it's available at http://adm.hamnett.org :) |
| 12:37.25 | slePP | no go, mate |
| 12:37.33 | slePP | i just can't connect to you, apparently :> |
| 12:37.38 | *** join/#asterisk lohelle (~post@213.184.212.218) |
| 12:37.49 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 12:38.08 | *** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl) |
| 12:40.16 | jerlique | cannot get to that webpage is it up? |
| 12:42.47 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 12:43.35 | zoa | yeah that adm thing is not working |
| 12:43.37 | zoa | heya oej |
| 12:44.56 | jerlique | whats the url again was d/c |
| 12:45.28 | oej | Hey Zoa |
| 12:46.15 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:48.36 | mrempire | where should i look if i receive :capi not installed on fedora core |
| 12:50.09 | *** join/#asterisk Luhiwu (~marsosa@200.63.87.246) |
| 12:51.51 | Zeeek | mmmm |
| 13:03.02 | jerlique | whats the difference between AgentLoginCallback and AddQueueMember, dont they achieve the same task? |
| 13:03.48 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 13:05.39 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:10.26 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:18.18 | *** join/#asterisk dave_mwi_ (root@harpo.dreamhost.com) |
| 13:18.19 | Zeeek | . |
| 13:20.32 | *** join/#asterisk Martohtar (~Martohtar@82.196.217.62) |
| 13:22.01 | slePP | god damn voice recognition systems |
| 13:22.06 | slePP | make me read off a stupid package number |
| 13:22.13 | slePP | then they tell me 'In Transit' |
| 13:22.16 | slePP | and i waste all this time |
| 13:22.25 | slePP | then they make mepress 0.. go figure why you can't just say operator or something |
| 13:22.53 | Zeeek | i |
| 13:22.57 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 13:23.08 | Zeeek | wasIm |
| 13:24.22 | Zeeek | say ah |
| 13:24.35 | Zeeek | how about this: |
| 13:24.54 | Zeeek | why do they ask you to type in your SS number and then ask you again when you finally reach somoen? |
| 13:25.29 | markit | anyone here using the new native assisted transfer atxfer? |
| 13:25.32 | florz | Is there some way to (on the fly?) change the size of asterisk's logging buffer? |
| 13:25.59 | florz | when I enable pri intense debugging it looks much like some ring buffer is overflowing ... |
| 13:28.55 | *** join/#asterisk imcdona1 (~t@corp.inline.com) |
| 13:29.18 | slePP | Zeeek: good question |
| 13:29.36 | slePP | Zeeek: my favourite is entering about 239847293847 digits for an account, then they triple verify your address and stuff |
| 13:29.44 | slePP | and then get your account number again |
| 13:30.04 | slePP | anyone know anything about DNIS? |
| 13:30.34 | imcdona1 | I am getting an error when compiling festival.....the wiki points me here |
| 13:30.35 | imcdona1 | http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html |
| 13:30.55 | imcdona1 | Where can I get the patch? |
| 13:32.15 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 13:33.01 | *** join/#asterisk inspired (mikael@host-81-191-113-158.bluecom.no) |
| 13:33.05 | slePP | no idea. |
| 13:33.38 | *** join/#asterisk vaewynAFK (freeman@mail.deltamach.com) |
| 13:33.54 | *** join/#asterisk JerJer (~JerJer@dsl-107-24.che.centurytel.net) |
| 13:33.59 | vaewynAFK | morning JerJer |
| 13:34.16 | Zeeek | : |
| 13:34.20 | inspired | anyone else having problems with SpanDSP? I can't send faxes and the only fax I received was from a web-to-fax service on the internet. Faxing from fax machines on the PSTN doesn't work. |
| 13:35.56 | Zeeek | inspired ... sigh... I've had the same luck with receiving faxes |
| 13:36.12 | Zeeek | Anything from jfax come in perfectly |
| 13:36.37 | Zeeek | a real fax machine like our cust use either doesn't work at all or partial page. |
| 13:37.12 | kaitseb | anyone has solution for faxing, same problem here |
| 13:37.25 | Zeeek | there is only one solmution - a cheap fax machine |
| 13:37.45 | Zeeek | on an FXS if you have a spare line |
| 13:37.49 | inspired | yeah |
| 13:37.59 | Zeeek | I know, it sucks but there you go |
| 13:38.18 | inspired | do you think it's an architectural problem with asterisk or is it spandsp? |
| 13:39.16 | florz | I am using spandsp productively. 368 faxes so far and no problems AFAIK. Connected to the PSTN using HFC cards. |
| 13:39.23 | florz | HFC-S to be exact |
| 13:39.27 | Makenshi | how good are x100p clones with uk lines? |
| 13:39.54 | inspired | florz: ISDN? |
| 13:39.58 | florz | inspired: yep |
| 13:40.11 | inspired | Zeeek: I use a e100p. what do you use? |
| 13:40.20 | wasim | Makenshi: x100p they aren't any good with any lines |
| 13:40.49 | inspired | florz: can you show us your setup? |
| 13:40.54 | florz | inspired: mom |
| 13:41.28 | Zeeek | inspired X10P |
| 13:41.30 | Zeeek | 100P |
| 13:41.34 | imcdona1 | Anyone know where I can get the festival patch as described here: |
| 13:41.42 | imcdona1 | http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html |
| 13:42.03 | Nugget | I don't think that the festival patch is necessary any more. |
| 13:42.16 | imcdona1 | in 1.95? |
| 13:42.20 | imcdona1 | the beta of festival? |
| 13:42.29 | Nugget | I have no idea. I've just heard that mentioned here. |
| 13:42.32 | Nugget | I don't use it myself. |
| 13:43.04 | inspired | florz: mom? |
| 13:43.08 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 13:43.27 | florz | inspired: Currently a bit busy, in a few mins or so ... |
| 13:43.32 | inspired | ok, thanks |
| 13:45.32 | Zeeek | ! |
| 13:46.00 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 13:46.44 | djin | ok, was an older mpg123 version. |
| 13:47.39 | *** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net) |
| 13:49.44 | inspired | florz: what version are you using? |
| 13:49.48 | JunK-Y | lo guys. |
| 13:49.56 | Zeeek | .. |
| 13:50.11 | Zeeek | I keepgetting disconnected, please ignore my punctuation |
| 13:50.55 | florz | inspired: * 1.0.5 + bristuff 0.2.0-RC5 + my patch for zaphfc + spandsp-0.0.2pre10. You can find a short overview of the system at http://florz.dyndns.org/zaphfc/ |
| 13:51.17 | inspired | ok |
| 13:53.25 | *** join/#asterisk prh (~paul@wacka.mjr.org) |
| 13:53.36 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 13:56.59 | *** join/#asterisk olivier_ (~olivier_@AMarseille-152-1-1-32.w82-127.abo.wanadoo.fr) |
| 13:57.37 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 14:03.55 | Zeeek | ... |
| 14:04.04 | Zeeek | .. |
| 14:04.06 | Zeeek | . |
| 14:04.35 | wasim | kabooom! |
| 14:05.38 | djin | Need to find your way back home, Zeeek? |
| 14:05.52 | Zeeek | breadcrumbs..... |
| 14:05.55 | Zeeek | ->->-> |
| 14:05.56 | djin | ;) |
| 14:06.03 | Zeeek | <-<-<-<-< coming from here |
| 14:06.18 | djin | ^ |
| 14:06.23 | djin | going up |
| 14:06.28 | Zeeek | <PROTECTED> |
| 14:06.48 | Zeeek | Hooror of horrors, it's..... |
| 14:06.53 | Zeeek | DOUBLE NAT! |
| 14:07.12 | djin | Why-o-why? |
| 14:07.29 | Zeeek | but wait... there is help |
| 14:07.45 | Zeeek | http://willypick.mindsay.com/?entry=10 |
| 14:08.13 | Zeeek | he can't talk but he got SIP working with double NAT :) |
| 14:09.43 | djin | nice for a babyphone. |
| 14:11.33 | *** join/#asterisk escualis (~carlos@113-140-121.adsl.cust.tie.cl) |
| 14:12.04 | Zeeek | . |
| 14:12.07 | Zeeek | bread |
| 14:12.39 | *** join/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 14:17.56 | vaewyn | bah... that's not that big of a deal... 1 phone on double NAT has always been doable... more than one phone... now that is a trick |
| 14:17.56 | JerJer | milk |
| 14:18.03 | vaewyn | cookies |
| 14:22.12 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 14:22.31 | ctooley | VoicePulse, are you here? |
| 14:23.21 | ctooley | USERNAME/USERNAME 66.234.228.132 255.255.255.255 5060 UNREACHABLE |
| 14:24.35 | Nugget | you use sip for voicepulse? |
| 14:24.38 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 14:27.52 | ctooley | Nugget, we are for now |
| 14:28.40 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 14:29.18 | florz | inspired: plus libtiff 3.5.7 - actually, that's pretty much it as far as receiving the .tiff is concerned. |
| 14:29.34 | inspired | florz: ok, what distro? |
| 14:29.39 | florz | inspired: debian woody |
| 14:29.58 | florz | inspired: though libtiff is compiled from source |
| 14:30.16 | inspired | ok, I'm running debian testing. I guess the version works as I have received one fax |
| 14:30.43 | florz | inspired: not debian source, that is, just make && make install from IIRC the "upstream"/original tarball |
| 14:30.46 | inspired | could it be that faxes from digital lines work, but not from analog lines? |
| 14:31.13 | inspired | using Version: 3.6.1-5 |
| 14:31.17 | inspired | it should work |
| 14:31.27 | florz | inspired: I don't think so @digital/analog |
| 14:31.45 | inspired | ok, it's just so strange that the fax from a web-to-fax service worked |
| 14:31.54 | inspired | but not from OKI, Canon and Panasonic fax machines |
| 14:32.29 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
| 14:32.51 | florz | inspired: actually a newer libtiff did not work IIRC |
| 14:33.04 | inspired | hmm |
| 14:35.00 | florz | inspired: Yep, I tried compiling tiff_3.6.1-2 from then-testing. Package was built without problems - but faxes were garbled or such. Dunno exactly anymore, but for some reason I must have installed from upstream source =:-) |
| 14:35.51 | inspired | ok |
| 14:36.29 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 14:36.32 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:36.34 | fantomax1 | hi all |
| 14:37.08 | fantomax1 | Did anyone use SIPP for generating SIP bulk traffic ? |
| 14:38.46 | *** join/#asterisk zno (~zeno@ip-160-79-174-100.autorev.intellispace.net) |
| 14:39.01 | zno | top o' the mornin' to y'all |
| 14:39.06 | zno | for EST people |
| 14:39.29 | inspired | florz: you are in IT? are you willing to send me a fax? I want to check out if this is a local problem |
| 14:41.20 | *** join/#asterisk ENNE (~ennepc2@62.48.113.138) |
| 14:42.38 | inspired | florz: do you have echo cancellation turned off in zapata.conf? |
| 14:44.35 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 14:45.20 | ENNE | hi people |
| 14:47.59 | *** join/#asterisk `Sauron (sauron@rrcs-24-153-164-117.sw.biz.rr.com) |
| 14:48.04 | ENNE | I'm a customer of an Internet Provider VoIP that uses the Asterisk. How can I make to take advantage of the possibility (if this exists) to send SMS? |
| 14:48.56 | wasim | ENNE: send SMS how and to whom? |
| 14:49.23 | wasim | ENNE: the zap cards support SMS on fixed line through app_sms |
| 14:49.36 | *** join/#asterisk LarsAC (~chatzilla@pD9500E94.dip0.t-ipconnect.de) |
| 14:49.38 | wasim | ENNE: the farfons should support SMS receipt and potentially respond |
| 14:49.45 | LarsAC | florz: ping |
| 14:51.32 | blitzrage | ManxPwr: btw, I never got your email the other day |
| 14:51.36 | escualis | anyone have a good page with manuals? |
| 14:51.44 | blitzrage | asteriskdocs.org |
| 14:51.51 | escualis | thanks :-) |
| 14:52.03 | ENNE | send sms to any mobile number of my country (if this is possible), I utilize a SIP Softphone software (xlite) |
| 14:54.08 | JunK-Y | off topic question: whats the equivalent of tar -c on SCO? |
| 14:54.53 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 14:55.27 | *** join/#asterisk anthm (~anthm@69.76.83.52) |
| 14:55.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:57.00 | *** join/#asterisk |^Angel^| (~funschrip@port336.ds1-alb.adsl.cybercity.dk) |
| 14:57.59 | ManxPwr | blitzrage, Feb 5 18:51:57 bourbon postfix/smtp[7343]: DF0411B8: to=<leif@leifmadsen.com>, relay=mail4.zoneedit.com[66.223.51.63], delay=22, status=sent (250 Ok: queued as 1B60F264D5B) |
| 14:59.28 | ariel_ | good morning all |
| 15:00.49 | blitzrage | ManxPwr: odd... |
| 15:01.53 | blitzrage | ManxPwr: can you try sending it again? I don't see it on Feb 5th |
| 15:02.58 | ManxPwr | blitzrage, Subject: [Fwd: [Asterisk-Users] chan_sip errors in CVS stable] |
| 15:03.10 | ManxPwr | Are you filtering on sender of subject on your system? |
| 15:03.41 | blitzrage | ManxPwr: thats why, I was looking for something *not* from asterisk-users |
| 15:03.45 | *** join/#asterisk ToyMan (~konversat@204.8.82.238) |
| 15:03.46 | blitzrage | foudn it |
| 15:04.07 | tzanger | morning blitzrage |
| 15:04.13 | ManxPwr | blitzrage, Cool. I guess I should have changed the Subject: |
| 15:04.14 | blitzrage | tzanger: morning, how goes today? |
| 15:04.21 | tzanger | it goes |
| 15:04.23 | blitzrage | ManxPwr: yah... it filtered on the subject name :) |
| 15:04.29 | tzanger | I drive a minivan now :'-( |
| 15:05.24 | blitzrage | tzanger: you sir... suck |
| 15:05.48 | tzanger | blitzrage: yeah well.. it happens |
| 15:05.52 | vaewyn | tzanger: know the feeling... I am looking for a station wagon... ohh the horrors! |
| 15:05.52 | fantomax1 | Did anyone use SIPP for generating SIP bulk traffic ? |
| 15:06.01 | tzanger | haha |
| 15:06.07 | blitzrage | tzanger: I never recommend anyone get a minivan - motor and suspension of a car in a vehicle the weight and use of a truck |
| 15:06.08 | *** join/#asterisk ex95 (~mjc205@adsl-69-153-229-186.dsl.pnblar.swbell.net) |
| 15:06.19 | vaewyn | would get a minivan... but they have SUCKY gas mileage |
| 15:06.26 | ctooley | I have a wagon, sedan, and a minivan |
| 15:06.26 | blitzrage | vaewyn: station wagon isn't so bad |
| 15:06.34 | tzanger | blitzrage: early minivans, yes... but modern ones that's simply not true anymore |
| 15:06.47 | ariel_ | well I got an SUV for the family. argh I am a yuppie. |
| 15:06.47 | *** join/#asterisk riksta (riksta@212.85.228.176) |
| 15:06.49 | ctooley | actually the minivan gets about 38 |
| 15:06.50 | tzanger | my minivan has the pickup and snottiness of a Grand Prix |
| 15:06.52 | tzanger | and I'm not joking |
| 15:06.53 | riksta | hey postel, alive? |
| 15:06.55 | blitzrage | tzanger: well, as of at least 1999 it was still true :) |
| 15:07.02 | tzanger | My '99 transport that isn't true of :-) |
| 15:07.14 | vaewyn | ctooley: 38? you talking liters or gallons? |
| 15:07.16 | vaewyn | :} |
| 15:07.27 | ctooley | vaewyn, gallons |
| 15:07.27 | ManxPwr | tzanger, I hope you have a need for a minivan and didn't get it just because it is "cool" |
| 15:07.32 | florz | inspired: echo cancel is off |
| 15:07.36 | blitzrage | haha |
| 15:07.42 | tzanger | we test drove a shitload of them and the transport/montana/ventures were all the quietest of the lot |
| 15:07.45 | florz | LarsAC: pong |
| 15:07.50 | inspired | florz: ok |
| 15:07.51 | blitzrage | ok, I have to go and write docs, lates |
| 15:07.57 | tzanger | ManxPwr: well I drove a '94 grand cherokee ltd until this weekend, so no it wasn't an upgrade to more cool |
| 15:08.02 | blitzrage | I'll be in asterisk-doc if anyone needs me |
| 15:08.06 | tzanger | blitzrage: I have an email in composition for you for the faq |
| 15:08.18 | blitzrage | tzanger: perfect, I just got a bunch of stuff from ManxPwr |
| 15:08.25 | blitzrage | or rather I just *found* a bunch of stuff from ManxPwr |
| 15:08.45 | florz | inspired: and sending you a fax from here right now is a bit difficult |
| 15:08.50 | ManxPwr | blitzrage, take a look at the faq page in the wffs.tar as well. It's REALLY old and out of date, but there may be some things you can use. |
| 15:08.56 | LarsAC | florz: schreib dir grad ne mail... ztcfg is the evil part |
| 15:09.06 | inspired | florz: ok |
| 15:09.11 | florz | LarsAC: Hmpf?! |
| 15:09.27 | LarsAC | florz: modprobing and rmmodding is fine |
| 15:09.31 | *** join/#asterisk zoa (~zoa@213.219.141.7.adslpower.by.edpnet.be) |
| 15:09.41 | *** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net) |
| 15:09.45 | LarsAC | florz: but calling ztcfg in between makes rmmod zaphfc badly crash the machine |
| 15:10.42 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 15:10.47 | florz | LarsAC: IC. That's not necessarily a sign that it's ztcfg's fault. Actually it's quite unlikely to be a problem of ztcfg - ztcfg just calls a few of the driver's ioctls. |
| 15:11.27 | florz | LarsAC: But the driver only becomes "active" after ztcfg has been called. |
| 15:12.01 | LarsAC | florz: okay, unloading an inactive driver is probably easy |
| 15:12.12 | florz | LarsAC: Exactly. |
| 15:12.22 | LarsAC | florz: I can spot a message "empty HDLC frame received" -- is that a problem ? |
| 15:12.58 | LarsAC | florz: mail sent |
| 15:13.27 | florz | LarsAC: I don't think so. I don't know exactly what causes it but it seems to happen now and then on most machines without any ill effects. |
| 15:15.07 | florz | LarsAC: Have you tried the original zaphfc driver from bristuff 0.2.0-RC5 yet? |
| 15:15.18 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:15.50 | LarsAC | no, but I had similar problems using rc2 and rc3 but never tracked them down in detail |
| 15:15.58 | inspired | Zeeek: where are you located? |
| 15:16.12 | florz | LarsAC: rc2/3 without the patch, then? |
| 15:16.13 | riksta | is the asterisk daily news guy around? |
| 15:16.29 | Zeeek | Paris France, home of Asterisk from time to time |
| 15:16.44 | Zeeek | when MArk visits :) |
| 15:16.54 | inspired | Zeeek: I turned echotraining off. are you able to send a fax to +370 52665825 ? |
| 15:16.56 | *** join/#asterisk ChatWeazl (ChatWeazl@82-197-199-89.dsl.cambrium.nl) |
| 15:16.58 | inspired | hehe |
| 15:17.01 | LarsAC | florz: vanilla and with patch called "cleanexit" which was posted in ip-phone-forum.de |
| 15:17.23 | Zeeek | I can't send faxes from here - I'm home ATM |
| 15:17.26 | inspired | ok |
| 15:17.30 | Zeeek | wait |
| 15:17.37 | inspired | damn, gotta find someone with a fax machine :D |
| 15:17.43 | Zeeek | I could but it'ds be from J2 which is a web service so it will work |
| 15:17.48 | Zeeek | not proving anything |
| 15:18.07 | inspired | ok |
| 15:18.36 | riksta | if anyone would like to try out my new software ADM - Asterisk Desktop Manager and give me some feedback, i'd appreciate it ... http://adm.hamnett.org/ |
| 15:18.43 | Logan | One of our Sipura phones is now always appearing busy. |
| 15:18.47 | florz | LarsAC: Hmmm. How about that init script versus shell phenomenon? |
| 15:18.56 | markit | riksta: some screenshot would be apreciated also ;) |
| 15:19.18 | riksta | markit its on the website |
| 15:19.29 | LarsAC | florz: the script contains ztcfg :-) |
| 15:19.39 | LarsAC | if I remove it, the script works fine |
| 15:19.49 | florz | LarsAC: So you didn't run ztcfg on the shell? IC ... |
| 15:19.49 | markit | riksta: url? |
| 15:20.04 | riksta | markit i just said it http://adm.hamnett.org/ |
| 15:20.19 | markit | riksta: it does not show other than a .tar.gz file |
| 15:20.27 | markit | no screen shots link |
| 15:20.30 | riksta | markit clear your cache |
| 15:20.35 | markit | ah :) |
| 15:20.48 | markit | riksta: yes, definetly better! :) |
| 15:21.13 | riksta | definitely |
| 15:21.33 | LarsAC | florz: no, if I do it crashes as well |
| 15:21.40 | Logan | Nevermind, my coworkers are fools. |
| 15:22.16 | markit | my english is terrible, I know |
| 15:23.04 | markit | anyone here that can help me with "disconnect" in features.conf? |
| 15:23.05 | Logan | Where is the frequency at which MWI updates are performed defined? |
| 15:23.21 | florz | LarsAC: OK, fine =:-) |
| 15:23.32 | slePP | Logan: it has to do with the qualify interval, basically |
| 15:23.36 | LarsAC | florz: rather not so... |
| 15:23.42 | Logan | slePP: What is that? |
| 15:23.57 | slePP | er.. k, just imagine it's about 30 seconds :> |
| 15:24.02 | slePP | it's not directly configured anywhere |
| 15:24.19 | Logan | I see a qualify setting. |
| 15:24.20 | LarsAC | florz: I didn't try to simply reboot, but having a system which reproducably crashes like this as a production system is not really my favourite |
| 15:24.28 | Logan | Do phones poll for MWI, or does asterisk drive that itself? |
| 15:24.32 | slePP | Logan: yeh.. when it qualifies the phone, it also does MWI updates |
| 15:24.32 | florz | LarsAC: Actually, it is. Not that the machine crashes. But that shell vs. script doesn't make a difference is fine indeed :-) |
| 15:24.35 | slePP | and asterisk drives it |
| 15:24.44 | riksta | hey slePP |
| 15:24.45 | slePP | it sends it out as part of a SIP message |
| 15:24.46 | slePP | hey rik |
| 15:24.52 | LarsAC | florz: yes, that would be really funny |
| 15:24.59 | Logan | slePP: So asterisk chooses to call ast_app_messagecount every so often for each registered user? |
| 15:25.17 | slePP | Logan: no, not that app |
| 15:25.22 | slePP | it goes through the entire sip/iax2 peer lists |
| 15:25.26 | slePP | and checks their messages, and sends updates |
| 15:26.01 | florz | LarsAC: Have you tried a 2.4 and/or a non-SMP kernel? |
| 15:26.04 | slePP | Logan: look at transmit_notify_with_mwi |
| 15:26.22 | slePP | which gets hit by sip_send_mwi_to_peer |
| 15:26.34 | LarsAC | florz: not yet |
| 15:26.38 | *** join/#asterisk BBRodriguez (~BBRodrigu@pD956346E.dip.t-dialin.net) |
| 15:27.01 | florz | LarsAC: Could you just boot the machine with the nosmp kernel parameter? |
| 15:27.04 | slePP | Logan: in the code, it checks that the current time - the last check > 10 before doing it |
| 15:27.04 | BBRodriguez | Hi everyone, how do i make * send SIP 403 "forbidden" message from dialplan ? |
| 15:27.13 | slePP | but it doesn't necessarily mean it'll happen every 10 seconds.. just no more often than that |
| 15:27.31 | LarsAC | florz: ah, that one's new to me... |
| 15:27.33 | slePP | BBRodriguez: pretty certain you don't. |
| 15:27.59 | LarsAC | florz: it acts as the router, I'll try this later, GF is surfing web |
| 15:28.03 | slePP | that's not something for the dialplan, really |
| 15:28.35 | slePP | 8:28am. whee |
| 15:28.35 | slePP | wtf am i doing awake |
| 15:28.35 | BBRodriguez | slePP: do you have a suggestion on how to do that ? |
| 15:28.37 | slePP | BBRodriguez: why do you want to do that? |
| 15:28.45 | slePP | that's an authorization thing, not a call handling bit |
| 15:28.55 | Logan | slePP: Thanks. |
| 15:29.09 | slePP | Logan: that's done in do_monitor, btw, the timeout. |
| 15:29.11 | florz | LarsAC: OK. I'll have another look at your log excerpt, maybe I'll have some other ideas what to test ... |
| 15:29.30 | Logan | slePP: I overrided the behavior of ast_app_msesagecount to call into my own code, which queries a DB, and I want to have a way reduce the load as I add more phones. :P |
| 15:29.31 | BBRodriguez | slePP: i need to stop accepting channels after the active ones, reach a certain amount, but i need to send 403 back to my softswitch, so it'll jump to the next route |
| 15:29.39 | Logan | slePP: Yeah, I fou nd the literal "10". |
| 15:30.04 | slePP | BBRodriguez: and a 404 or something wouldn't work better? |
| 15:30.10 | slePP | or perhaps a 5xx code |
| 15:30.22 | slePP | any which way, there isn't really a way to do it from the dialplan... |
| 15:30.50 | slePP | you should just be able to keep pumping it out, if you're going into zap channels. it'll send back an unavailable response |
| 15:30.55 | BBRodriguez | slePP: how do i send 404 ? circuit busy ? |
| 15:31.11 | slePP | 404 is not found, it'd be closer to Congestion.. but i don't think 404 is that... maybe it is :> |
| 15:31.17 | slePP | try Congestion and trace the protocol |
| 15:31.36 | BBRodriguez | slePP: i'm accepting sip channels and dialing sip channels, no zapata or PRI |
| 15:32.40 | slePP | k. then try Congestion |
| 15:32.45 | slePP | if nothing else, make up a function that does it :> |
| 15:33.25 | BBRodriguez | slePP: thanks, i might just do that, make a function ;-) |
| 15:34.09 | slePP | that'll be easiest :> then at least you know what it is doing |
| 15:35.20 | Luke-Jr | Is there a way to force ulaw *only* if the remote UA is Kphone? |
| 15:35.58 | satlink | Hi, anyone knowing about an IAX softphone supporting g729? |
| 15:36.07 | satlink | -for windows |
| 15:36.32 | slePP | Luke-Jr: don't think so |
| 15:42.40 | nestAr | i wish the cli syntax was a bit more sane.. more like cisco |
| 15:43.02 | Logan | Why does ztmonitor require /dev/dsp? |
| 15:43.33 | slePP | use -v or -vv |
| 15:43.35 | slePP | and it doesn't. |
| 15:43.36 | epoch | so it can yell at you |
| 15:44.38 | inspired | Zeeek: try turning echotraining=off in zapata.conf. it solved everything here! |
| 15:46.39 | Logan | slePP: So, am I supposed to place a call over a zap channel, then run ztmonitor -v on it, and I should see some output? |
| 15:46.51 | Zeeek | I'm sure I did that when I cared |
| 15:47.02 | Zeeek | but thanks |
| 15:47.10 | eKo1 | Logan: I know chan_oss requires it but not ztmonitor. |
| 15:47.57 | eKo1 | nestAr: You've been using Cisco too long. |
| 15:48.04 | Logan | I know the call is on channel for, but ztmonitor 4 -v isn't showing anything. |
| 15:48.07 | Logan | s/for/4 |
| 15:48.31 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 15:49.11 | nestAr | eKo1: well.. the whole.. sip show X is silly.. it should be show sip X |
| 15:49.22 | nestAr | i dunno |
| 15:49.37 | nestAr | not that i really like cisco |
| 15:49.48 | nestAr | but it's got a convention that i can get used to.. |
| 15:50.08 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfkh2.dialup.mindspring.com) |
| 15:50.55 | nestAr | :shrug: |
| 15:51.06 | nestAr | i'm just used to cisco and redback |
| 15:51.07 | ariel_ | nestAr, you will get used to it. Besides you can do the same with iax2 show peers |
| 15:51.32 | nestAr | half the stuff you do show X, but then there's iax show x |
| 15:51.35 | nestAr | sillyl. |
| 15:51.38 | nestAr | -l |
| 15:51.49 | ariel_ | yes but it's something we need to live with. |
| 15:52.03 | nestAr | indeed, i'm living with it. |
| 15:52.35 | ariel_ | just like linux has -h or sometimes --help and don't forget sometimes -help will work too. argh |
| 15:52.53 | eKo1 | --help is gnu |
| 15:53.36 | anthm | that's cos the modules are supposed to have thier own command namespace you cant have sip show peers without chan_sip.so so each module that implements commands shoule always begin with the name of the module so each module has logical namspace afforded to it |
| 15:53.45 | anthm | use the source luke |
| 15:53.52 | *** join/#asterisk PakiPenguin (~info@mbl-99-53-158.dsl.net.pk) |
| 15:54.00 | PakiPenguin | hello everyone |
| 15:54.08 | PakiPenguin | anyone here tried using Linksys - PAP2 with *? |
| 15:54.14 | ariel_ | anthm, have not seen you for some time welcome back. |
| 15:54.21 | PakiPenguin | i am getting a registeration problem , the linksys never registers :( it fails |
| 15:54.50 | ariel_ | PakiPenguin, are you using the pap2-na or the one that is pre-setup for vonga? |
| 15:55.20 | PakiPenguin | its na , unlocked |
| 15:55.25 | ariel_ | k |
| 15:55.35 | ariel_ | I have and they work just like the sipura. |
| 15:55.41 | PakiPenguin | any specific config in sip.conf? |
| 15:55.55 | anthm | ariel_ thx |
| 15:56.17 | ariel_ | line 1 setup proxy ip addres then the user name. password and it should work unless you have a natted asterisk. |
| 15:56.34 | ariel_ | anthm, I miss your patch I loved it..... |
| 15:56.48 | *** join/#asterisk mindCrime (~mindCrime@bi01p1.nc.us.ibm.com) |
| 15:56.51 | anthm | oh the caching peer one ? |
| 15:56.59 | ariel_ | chanspy |
| 15:57.02 | hans | "Answer(): If the channel is ringing, answer it, otherwise do nothing." Yet it seems to be answering as much as 10 seconds after I pick up, so is ring detection skewampus? |
| 15:57.32 | PakiPenguin | * is not natted , but the ata is |
| 15:57.59 | PakiPenguin | it gives me this Feb 7 09:51:17 NOTICE[2372]: chan_sip.c:7656 handle_request: Registration from 'User <sip:1002@sip.xxxxxxxx.com>' failed for '203.xx.xx.xx' |
| 15:58.04 | ariel_ | PakiPenguin, then make sure you have the firewall forwarding your port |
| 15:58.25 | ariel_ | PakiPenguin, it's not over a sat link? |
| 15:58.35 | PakiPenguin | nopes |
| 15:58.39 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 15:58.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:58.50 | hans | oh wait, I'm not doing answer, I'm doing zapateller(answer), that's probably why. |
| 15:58.52 | PakiPenguin | yes , the firewall is forwarding the ports |
| 15:58.52 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
| 16:00.19 | ariel_ | PakiPenguin, have you seen this setup info http://www.xorcom.com/sipura_tip.html |
| 16:00.40 | PakiPenguin | nopes, am looking now |
| 16:01.10 | *** join/#asterisk Corydon-w (~tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 16:01.49 | ctooley | how many concurrent SIP/G.711u channels will fit reliably on a 100Mbit network connection? I can do the math using 90kb/s, but that just gives me theoretical, not usable. |
| 16:02.05 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 16:02.21 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
| 16:02.46 | ariel_ | ctooley, it really depends on what else shares the line. |
| 16:03.31 | ctooley | In a datacenter, going out on the internet to our provider. |
| 16:04.23 | ariel_ | oh cut your figure in 1/2. When any ftp starts it will make your life hard. |
| 16:06.44 | ctooley | well, there's not going to be any FTP servers between us and them. But what should the realistic expectations be? 10Mbit, 15, 20? |
| 16:07.07 | *** join/#asterisk jterrero (~some@66.28.34.162) |
| 16:07.34 | *** part/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 16:07.58 | ariel_ | in our test lab at the last place I worked we were able to get on a 100m line about 12 max per meg. the higher then number pass 50 it would drop to around 6 per meg. |
| 16:08.02 | jterrero | sup, can someone help me out with the following? I have sendmail on my system (running gentoo 2.6.9), when i leave a voicemail i want the voicemail to be sent via email to the user, i specified everything i was suppose to in voicemail.conf and extensions.conf, i get the following error though |
| 16:08.03 | jterrero | Spawn extension (default, 301, 2) exited non-zero on 'SIP/jterrero-0808' |
| 16:08.03 | jterrero | sendmail: Cannot open mail:25 |
| 16:08.18 | jterrero | when i do a portscan on my system 25 is open |
| 16:08.44 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 16:09.36 | florz | ctooley: I don't have any experience myself. But if it's point-to-point and full duplex, I would wonder why you should not be able to use the full bandwidth!? |
| 16:10.08 | wwalker_ | jterrero first, does the hostname mail, without a domain tacked on properly resolve to your local machine (via DNS< not via /etc/hosts? |
| 16:10.11 | ariel_ | ctooley, so on a 100meg connection we maxed out at around 90 solid connections. |
| 16:10.17 | ctooley | florz it's not point to point, it's through the internet. It's not going to go out on a shared virtual hosting connection either though. |
| 16:11.05 | jterrero | wwalker_: no |
| 16:11.10 | fantomax1 | Did anyone use SIPP for generating SIP bulk traffic ? |
| 16:11.16 | *** join/#asterisk itnomad (~jackal@199.89.146.110) |
| 16:11.18 | florz | ctooley: But logically it's only two endpoints, right? So either one receives only what the other sends? |
| 16:11.36 | jterrero | wwalker_: i didnt specify mail anywhere though, it should be mail.mydomain.com |
| 16:11.41 | jterrero | not mail alone |
| 16:12.19 | florz | ctooley: then at least no congestion can happen because of packets from different sources arriving at the same time. |
| 16:12.34 | wwalker_ | jterrero is mydomain.com in your /etc/resolv.conf in search or domainname lines? |
| 16:13.01 | ctooley | florz, no, they have multiple clients doing the same thing, and w e have multiple servers sending to the same SIP endpoint |
| 16:13.31 | ctooley | so there's probably 50 SIP originiation servers talking to their Origination Proxy. |
| 16:14.39 | florz | ctooley: Well, then it's pretty difficult to say theoretically I think. |
| 16:15.20 | jterrero | wwalker_: no, the only thing in my resolve.conf is my nameserver |
| 16:16.15 | ctooley | jterrero, can you successfully "ping mail"? |
| 16:16.58 | jterrero | no, but i can ping mail.mydomain.com |
| 16:17.57 | ariel_ | ctooley, just looked at my notes the 90 figure was on a 10mg leg not 100. We never tested a full 100mg link. |
| 16:18.36 | *** part/#asterisk satlink (satlink@66.178.97.50) |
| 16:18.45 | florz | ariel_: FD or HD? PtP or PtMP? |
| 16:19.13 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 16:19.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:19.20 | wwalker_ | jterrero add "search mydomain.com" to /etc/resolv.conf and see if that fixes it. sendmail shouldn't be truncating the name, but it may be an easy fix. |
| 16:19.37 | ctooley | ariel_, thank goodness |
| 16:19.41 | wwalker_ | ariel_ Much nicer number! Thanks!! |
| 16:19.44 | ctooley | ariel_, were were starting to get worried |
| 16:19.47 | ariel_ | florz, fd |
| 16:21.12 | ariel_ | we were expecting more but as the b/w got used we had more and more problems. |
| 16:21.15 | *** join/#asterisk IronHelix (~irc@ool-182c8f9f.dyn.optonline.net) |
| 16:22.36 | florz | ariel_: That's probably due to non-optimal synchronisation. On an FD PtP(?) link it theoretically should be possible to use the full bandwidth. |
| 16:25.24 | ariel_ | should be. But we only had a limited amount of time for the test. |
| 16:26.41 | florz | ariel_: Yep. It's probably also a matter of Asterisk's implementation details, so not necessarily easy to achieve. Maybe IAX2 with trunking could do it. |
| 16:26.52 | ariel_ | ctooley, you want to put about 900 to 1000 sip calls via ulaw. Hope it works. |
| 16:26.56 | ctooley | florz, actually, at 90 concurrent calls 87-90kbit/call that's 7.8Mbit to begin with. |
| 16:27.26 | florz | ctooley: Which is about 22 % below the full bw =:-) |
| 16:28.04 | ctooley | florz, 7.8Mbit on a 10Mbit HALF duplex line. You're going to run into issues. |
| 16:28.11 | *** join/#asterisk mildenhall (~polarisx@194.114-84-212.ippool.ndo.com) |
| 16:28.24 | florz | ctooley: FD, not HD :-) |
| 16:29.25 | ariel_ | ctooley, ours was fd. And like your figures we could only get 90 max on the 10 meg before we started to get problems with the calls. |
| 16:29.38 | ariel_ | so your figures are close. |
| 16:29.38 | florz | ctooley: On HD that would not even theoretically be possible as it would sum up to 15.6 Mbit/s for both directions. |
| 16:34.21 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 16:36.21 | jterrero | got it to work, thanks people.. another issue though. i am writing cdrs in a .db format, is there a tool out there to make cdr.db look neat / clean ? |
| 16:36.42 | Luke-Jr | Can anyone suggest any cheap way to get to one or two FXS ports from my server? (PCI, USB, Ethernet, etc) |
| 16:36.55 | *** join/#asterisk netsurfer (netsurfer@82-133-64-79.dyn.gotadsl.co.uk) |
| 16:38.28 | ariel_ | Luke-Jr, use a sipura 2000 or 1001. |
| 16:39.04 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 16:39.17 | jterrero | got it to work, thanks people.. another issue though. i am writing cdrs in a .db format, is there a tool out there to make cdr.db look neat / clean ? |
| 16:39.29 | jterrero | oops. sorry |
| 16:39.31 | jterrero | double |
| 16:42.25 | wasim | i wonder if any aid agencies have used asterisk to help the comms in the tsunami efforts |
| 16:46.16 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 16:46.53 | Slainte | anyone use the polycom IP600 and find the number pad does not respond well, i.e. you have to press a number twice, when transferring etc. |
| 16:47.15 | Slainte | or you hve to hold the button down for a while. Kinda like Asterisk is ignoring it. |
| 16:48.01 | *** join/#asterisk dolson (~dolson@63.162.2.200) |
| 16:48.11 | eKo1 | Hmm...If a call comes in through my sip provider, I should see something about it on the CLI right? |
| 16:48.19 | eKo1 | I'm not seeing anything. |
| 16:49.21 | Slainte | eKo1, debug sip shows nothing? |
| 16:49.26 | ariel_ | eKo1, yes you should. |
| 16:49.45 | eKo1 | Slainte: eh, there are like 50 calls going out through that provider so... |
| 16:50.30 | Slainte | eKo, any properly formatted SIP traffic comming in on port 5060 you will see |
| 16:51.32 | eKo1 | Slainte: There's too much traffic for me to debug it properly. |
| 16:51.50 | eKo1 | 'show channels' should show it. |
| 16:52.11 | eKo1 | Of course, it doesn't show what I'm looking for. |
| 16:53.55 | riksta | anyone tried my ADM yet? |
| 16:55.44 | dolson | are any of you knowledgeable with SER and wouldn't mind helping me? |
| 16:57.51 | *** join/#asterisk Othello (Othello@spnp221042.spnp.nus.edu.sg) |
| 16:58.57 | eKo1 | I have a machine running SER here but I didn't set it up so... |
| 17:00.18 | *** join/#asterisk escualis (~carlos@113-140-121.adsl.cust.tie.cl) |
| 17:00.50 | *** join/#asterisk eipi (eipi@29-178-89-200.fibertel.com.ar) |
| 17:00.56 | eipi | hi all |
| 17:01.24 | eipi | im newbie with asterisk, i have it working with samples conf, i added user in iax.conf, but the calls are rejected |
| 17:01.39 | eipi | <PROTECTED> |
| 17:03.32 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 17:04.02 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:04.23 | *** join/#asterisk adrianhensler_ (~chatzilla@blk-222-123-93.eastlink.ca) |
| 17:05.57 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 17:08.36 | Luke-Jr | ariel_: any idea where to buy them? |
| 17:08.42 | ManxPwr | ~docs |
| 17:08.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:09.04 | Luke-Jr | For a decent price |
| 17:09.44 | Slainte | These POlycom phones wont keep an NTP synch. |
| 17:10.59 | Slainte | every time I turn around it has a different time |
| 17:11.06 | dolson | ~ser |
| 17:11.07 | jbot | [ser] Sip Express Router - see http://www.iptel.org/ser/ |
| 17:11.15 | dolson | ~make ser work |
| 17:11.18 | jbot | make: *** No rule to make target `ser work'. Stop. |
| 17:11.52 | mikegrb | Slainte: I believe they only sync on powerup |
| 17:12.02 | jterrero | got it to work, thanks people.. another issue though. i am writing cdrs in a .db format, is there a tool out there to make cdr.db look neat / clean ? how can i manipulate this data, ? |
| 17:13.54 | *** join/#asterisk zbysio (~zbysio@chello084010031149.chello.pl) |
| 17:15.15 | Slainte | Mikegrb, do you use the 600's or 500's? I am trying to get some micrpbrowser stuff working |
| 17:15.53 | Slainte | anyone know how to make hold "beep"instead of MP3 playback. MP3 playback for me is all garbbled. It may be a conflict with the zaptel drivers |
| 17:19.59 | mikegrb | Slainte: dpm |
| 17:20.01 | mikegrb | er |
| 17:20.08 | mikegrb | I don't use polycom at all |
| 17:20.13 | djin | Slainte, what mpg123 are you using? |
| 17:20.39 | bjohnson | jterrero: check the wiki. I think I saw one that does graphs and uses web interface |
| 17:21.29 | bjohnson | jterrero: in general, and script that does SQL could access the cdr data .. you could whip something up that emails you a report for example |
| 17:22.03 | greg_work | i have some mp3 files that won't play in *, but mpg123 will play them outside of * just fine. only differences I can see between the ones that work and the ones that dont are 128kbps vs 192kbps, orginal = no vs yes, and BPF = 418 vs 627 .. otherwise, everything else is the same: MPEG 1.0, Layer: III, Freq: 44100, mode: Joint-Stereo, modext: 0, BPF : 627 Channels: 2, copyright: No, original: Yes, CRC: Yes, emphasis: 0.Bitrate: 192 |
| 17:22.03 | greg_work | <PROTECTED> |
| 17:22.15 | greg_work | any ideas, and what is BPF? |
| 17:22.25 | Zeeek | they have to be at 8000 |
| 17:22.34 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 17:22.38 | Zeeek | and one channel would be nice too |
| 17:23.05 | jterrero | bjohnson: thxz |
| 17:23.27 | greg_work | Zeeek: the working one i'm comparing it to is QuajiroPromo.mp3 (comes with * ) |
| 17:23.42 | *** join/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 17:24.05 | Zeeek | ah |
| 17:24.24 | Zeeek | but is 2 chans a good idea? |
| 17:24.27 | greg_work | another one that works is MPEG 1.0, Layer: III, Freq: 44100, mode: Stereo, modext: 0, BPF : 626, Channels: 2, copyright: No, original: Yes, CRC: Yes, emphasis: 0., Bitrate: 192 Kbits/s, Extension value: 0 |
| 17:24.39 | greg_work | i dunno. doubles the data |
| 17:24.54 | Zeeek | screws up the music too |
| 17:25.01 | greg_work | howso? |
| 17:25.29 | *** join/#asterisk xilofonte (0@80.97.190.184) |
| 17:25.33 | Zeeek | well it'd be like shorting the two chans together which could multiply artifacts |
| 17:25.58 | Zeeek | I don't kniow how they're mixed in the player though for a mono output |
| 17:26.05 | greg_work | "mpg123", "-q", "-s", "-b", "1024", "-f", "8192", "--mono", "-r", "8000", filename |
| 17:26.18 | greg_work | is how app_mp3 calls mpg123 |
| 17:26.19 | Zeeek | there's the 8000 |
| 17:26.31 | Zeeek | ok - I'm not using that version |
| 17:26.47 | Zeeek | and I don't have any console audio anyway |
| 17:26.56 | Zeeek | so I should prolly shut up :) |
| 17:27.02 | nestAr | lol |
| 17:27.12 | greg_work | well, this is on hold music |
| 17:27.33 | Zeeek | or change the subject: callerid not working on ONE phone: siemens C200 - why? It works great on PSTN? |
| 17:28.08 | Zeeek | ya but yoyu mentioned that the player plays it ok, I have no way of testng that |
| 17:28.15 | Zeeek | so I'm folding |
| 17:28.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:29.05 | Slainte | djin, sorry only noticed your note |
| 17:29.11 | Slainte | I am useing latest I think |
| 17:29.17 | Slainte | one sec |
| 17:30.00 | Slainte | 0.2.10 |
| 17:30.15 | eipi | im newbie with asterisk, i have it working with samples conf, i added user in iax.conf, but the calls are rejected |
| 17:30.17 | eipi | <PROTECTED> |
| 17:30.26 | *** join/#asterisk wankel (nobody@ohno.mrbill.net) |
| 17:30.57 | eipi | i associated the new user to context=local and demo, but nothing, the same notice message... |
| 17:33.28 | Slainte | I just forced a reinstall lemme try again |
| 17:35.37 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 17:35.48 | djin | upgrading mpg123 solved my sound issue http://www.mpg123.de/ (0.59r) |
| 17:35.52 | Zeeek | eipi since it won't connect... |
| 17:36.07 | Zeeek | upgrading to "gold"? |
| 17:37.02 | Zeeek | eipi as I was about to say: there seems to be a problem with the peer or friend entry in iax.conf then? |
| 17:38.49 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 17:39.07 | *** join/#asterisk freat[laptop] (~freat[lap@65.170.62.117) |
| 17:39.10 | dontmsgme | I have an error msg it says "Nine Six Nine Six" |
| 17:39.14 | dontmsgme | When I try to dial |
| 17:39.36 | Zeeek | the old 9696 worm! |
| 17:40.09 | outtolunc | must be the SLC-96chained worm <G> |
| 17:40.39 | Zeeek | so show us yer Dial() command dontmsgme |
| 17:40.42 | dontmsgme | Im changing voicepulse servers. |
| 17:40.59 | Zeeek | changing to what? I use VP was there a change? |
| 17:41.22 | dontmsgme | Executing Dial("SIP/12345-3b9a", "IAX2/user:password@gwiaxt01.voicepulse.com/18005551212") in new stack |
| 17:41.22 | dontmsgme | <PROTECTED> |
| 17:41.22 | dontmsgme | <PROTECTED> |
| 17:41.22 | dontmsgme | <PROTECTED> |
| 17:41.26 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 17:42.03 | Zeeek | where's the 69? |
| 17:42.24 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 17:42.29 | dontmsgme | the asterisk voice says it |
| 17:42.47 | Zeeek | why? I mean how? |
| 17:42.54 | Zeeek | from the VP end? |
| 17:44.01 | dontmsgme | I dont know |
| 17:44.13 | dontmsgme | It does say Call accepted by 66. |
| 17:44.30 | Zeeek | yeah I just dialed it - looks the same |
| 17:44.39 | dontmsgme | You dialed what? |
| 17:44.48 | Zeeek | 5551212 on VP |
| 17:45.12 | *** join/#asterisk ngb (~joshua@200.49.156.89) |
| 17:45.21 | *** join/#asterisk numbone (~numBone@c-24-129-204-233.se.client2.attbi.com) |
| 17:45.23 | Zeeek | the only diff was that I have a 45 and a t after the number |
| 17:45.24 | netsurfer | quick question... if I have 2 * boxes with identical configuration, a sipphone connected to box 1, if I pull the network cable from box1 and put it in box2 (same ip address) will box2 force the sipphone to reauthenticate ? |
| 17:45.30 | ngb | in the cvs are the last version of asterisk ???? |
| 17:45.36 | ngb | 1.0.5 ? |
| 17:45.53 | Zeeek | netsurfer how would box 2 know? |
| 17:45.57 | ngb | how i can update my files from cvs command |
| 17:46.06 | netsurfer | Zeeek - when the sipphone tries to make a call |
| 17:46.24 | Zeeek | is it registering periodically? |
| 17:46.44 | Zeeek | I mean, if box 2 never heard of the phone before waking up on a LAN, how would it know ? |
| 17:47.29 | netsurfer | Zeeek - the timeout on the phone is 600 seconds.. if the phone registers with box1 then when the cable is changed it thinks its still registered to box1 - what will box2 do? ignore it? or ask it to re-register? |
| 17:47.51 | Zeeek | dontmsgme: my responses are a tiny bit different |
| 17:47.52 | Zeeek | -- Format for call is ulaw |
| 17:47.52 | Zeeek | <PROTECTED> |
| 17:47.52 | Zeeek | <PROTECTED> |
| 17:48.10 | Zeeek | -- Format for call is ulawnote ulaw in small letters and the progress report |
| 17:48.43 | netsurfer | Zeeek - the reason im asking, I am wondering if its worth my while messing with heartbeat |
| 17:48.44 | Zeeek | netsurfer I would guess "unpredicatble behaviour and may cause problems"? |
| 17:49.03 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:49.42 | Zeeek | I'm still trying to imagine why you'd want to do this |
| 17:49.50 | netsurfer | High Dependency |
| 17:49.56 | Zeeek | for thrills? |
| 17:50.01 | netsurfer | yeah yeah |
| 17:50.03 | netsurfer | :P |
| 17:50.29 | Zeeek | seems to me the call would be refused |
| 17:51.05 | Zeeek | but more importantly, isn't there cached info on connections in asterisk (like in the astdb, for one?) |
| 17:51.07 | netsurfer | I wanted to set up 2 boxes, getting their config from a central database and use heartbeat to control which one was the active server |
| 17:52.09 | Zeeek | may be time for a girlfriend |
| 17:52.18 | netsurfer | :P |
| 17:52.35 | netsurfer | Zeeek - its R&D for a project id like to get going |
| 17:52.44 | Zeeek | you have a production server with thousands of users? |
| 17:52.48 | Zeeek | ok |
| 17:53.09 | Zeeek | hot swappable asterisk servers |
| 17:53.16 | netsurfer | yup :D |
| 17:53.33 | Zeeek | better to have them both running on different ip? |
| 17:53.44 | Zeeek | ah but the phone itserfl won't handle that |
| 17:53.51 | netsurfer | exactly! |
| 17:53.58 | netsurfer | so thats why I want to use heartbeat |
| 17:54.05 | Zeeek | so you need three asterisk boxen :) |
| 17:54.12 | netsurfer | with the virtual interface ip set as the sip gateway |
| 17:54.19 | Zeeek | one to federate as a frontal and talk to the other two |
| 17:54.41 | netsurfer | lmao nah that makes it pointless |
| 17:54.46 | Zeeek | with a liberal suppluy of canreinvite=yes |
| 17:54.46 | netsurfer | the whole idea is redundancy |
| 17:55.14 | Zeeek | yeah because if box1 of three goes down it back to no servcie |
| 17:55.20 | netsurfer | yup |
| 17:55.23 | Zeeek | wait I got it |
| 17:55.27 | Zeeek | two phones |
| 17:55.34 | Zeeek | a red one and a black one |
| 17:55.37 | netsurfer | for each desk? ;) |
| 17:55.46 | Zeeek | TWO DESKS! YES! Brilliant! |
| 17:55.55 | Zeeek | two desks for each person |
| 17:56.06 | Zeeek | when box 1 goes down, he changes desks |
| 17:56.09 | ChatWeazl | yes, well, also get a redundant office building and redundant staff ;) |
| 17:56.22 | Zeeek | except that redundant means firing in UK |
| 17:56.34 | ChatWeazl | oh... |
| 17:56.37 | Zeeek | otherwise, I was about to suggest that |
| 17:57.03 | ChatWeazl | so how would you phrase that in the UK? |
| 17:57.07 | Zeeek | but the buildings woul dneed to be on different continents for obvious reasons |
| 17:57.26 | Zeeek | don't know. I'm not in the UK - but I know "made redundant" means FIRED |
| 17:57.37 | ChatWeazl | hmmz |
| 17:57.40 | Zeeek | hey, they say shit like tyres too |
| 17:57.46 | Zeeek | and hire a car |
| 17:57.50 | Zeeek | and take the lift |
| 18:00.09 | ChatWeazl | hmm websters has a special British part for redundancy... |
| 18:01.21 | netsurfer | lol Zeeek - u gotta chmod +x Zeeek before u go to work? ;) |
| 18:01.57 | Zeeek | no shit! I was so tired I couldn't even stay on IRC today |
| 18:02.01 | Zeeek | kept falling of |
| 18:02.01 | netsurfer | im getting VERY p!ssed off at debian |
| 18:02.06 | Nugget | linux is poo. |
| 18:02.26 | Zeeek | make the change to slackware - you'll never go back to women again |
| 18:02.26 | netsurfer | cant build zaptel with standard kernel |
| 18:02.32 | Zeeek | DebbyAgain |
| 18:03.04 | netsurfer | and iv spent amost 2 days trying to get a working custom kernel |
| 18:03.30 | ngb | i need upload the hits of my site... you can help me ? |
| 18:03.45 | Zeeek | Aren't there shops where guy with funny tatoos do customized kernels for you? |
| 18:04.10 | Zeeek | ngb what do you mean? |
| 18:04.46 | netsurfer | I think they should introduce the death penalty for incorrect info posted on the wiki pages |
| 18:05.04 | Zeeek | or at least assign the guilty person a night in the barrel |
| 18:05.30 | ngb | Zeeek i need that my website have visits.. all people from this channel can visit it ? |
| 18:05.32 | Zeeek | but here is a test: find out what the units used are in txgain= |
| 18:05.49 | Zeeek | ngb are there naked girls on it? |
| 18:05.58 | ngb | no |
| 18:06.04 | ChatWeazl | Zeeek: you perv |
| 18:06.05 | Zeeek | then what point is there? |
| 18:06.06 | netsurfer | Zeeek - dB ? |
| 18:06.20 | Zeeek | netsurfer - I have seen three different answers on three sites |
| 18:06.25 | netsurfer | lol |
| 18:06.29 | Zeeek | %, ratio and db |
| 18:06.44 | netsurfer | % of what exactly |
| 18:06.52 | Zeeek | it may be db because I think if you put 6 it gets REALLLYY LOUD |
| 18:06.54 | *** join/#asterisk cypenguin (~cypenguin@200.59.172.45) |
| 18:07.10 | *** join/#asterisk Dalion (anon@Toronto-HSE-ppp3771214.sympatico.ca) |
| 18:07.28 | Zeeek | both ration and opct would be as against 1 or 100% |
| 18:07.34 | netsurfer | whats 5 like? quite quiet ? |
| 18:07.58 | harryvv | morning netsurfer |
| 18:07.59 | Zeeek | so txgain=1.5 or txgain=150% or txgain=3 |
| 18:08.10 | netsurfer | if the steps seem unequal to the ear, then i'd say its probably dB |
| 18:08.23 | Zeeek | fact is it's really hard to tell |
| 18:08.23 | netsurfer | hi harryvv |
| 18:08.44 | harryvv | Is there a way to track graphicaly the amount of times a callers ID number rings into a zap? |
| 18:08.49 | Zeeek | ngb if we visit your site what do we get? |
| 18:09.38 | netsurfer | harryvv - u could store incoming clid in a database |
| 18:09.51 | harryvv | netsurfer, yea I have mysql installed |
| 18:10.02 | netsurfer | harryvv - cool.. go for it :) |
| 18:10.16 | Zeeek | harryvv depends on how you want to report it. You can just parse the cdr too |
| 18:10.16 | harryvv | btw, still getting that res_odbc.c:392 odbc_obj_connect: res_odbc: Error SQLCon problem. |
| 18:10.28 | Zeeek | with PHP or Perl |
| 18:10.32 | ngb | Zeeek more hits, and google bot visiti my site for add int in your website list |
| 18:10.37 | netsurfer | I have a script here wouldnt be hard to modify to get asterisk to read back how many times that CLID called in a given time |
| 18:10.44 | Zeeek | ngb what is asterisk? |
| 18:10.59 | harryvv | zeek, that would mean installing apache on the asterisk system and dont want to do that. |
| 18:11.04 | ngb | asterisk is more than a pbx :) |
| 18:11.07 | *** part/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 18:11.30 | Zeeek | I like Apache :) |
| 18:11.38 | ngb | :P |
| 18:11.45 | Zeeek | what is your web site about with no naked girls, ngb |
| 18:11.55 | harryvv | netsurfer, I am going to make another asterisk setup as a demonstration for our search and rescue team. There are some 200 members that belong to it. |
| 18:12.00 | Zeeek | surely you could illustrate it with a few |
| 18:12.18 | ngb | i sale web systems and web design |
| 18:12.32 | Nugget | perhaps it's an "english as a second language" site. |
| 18:12.40 | harryvv | netsurfer, there was a crisis with the blizzard and people stuck.. people were calling sar building and no one was answering because no one was there. |
| 18:12.50 | Zeeek | naked girls are the prime motor that makes sites get hits |
| 18:13.24 | Zeeek | ngb use alexa toolbar to make people think you have traffic |
| 18:13.44 | Zeeek | just get about five people to click on that all day |
| 18:14.06 | netsurfer | harryvv - sounds like it would be useful there |
| 18:14.24 | harryvv | net, we have no system there at all. |
| 18:14.59 | netsurfer | harryvv - u need something then! - btw.. did u call me early this morning/ late last night ur time ? |
| 18:15.14 | harryvv | last night :) |
| 18:15.16 | netsurfer | :oP |
| 18:15.27 | netsurfer | I had an email about a missed call when I got up lol |
| 18:15.33 | harryvv | :) |
| 18:16.07 | harryvv | btw, is there any windows client apps that can reside in memory a vm is awaiting? |
| 18:16.22 | netsurfer | wooooooow harry I got news on speeding up ur next * install |
| 18:16.22 | harryvv | say it pops up as soon as the user logs in. |
| 18:17.21 | netsurfer | harryvv not that I know of, though I find email is the best way to notify users, nothing new for them to get used to |
| 18:17.29 | harryvv | It to bad ms could not release some code say for its login window that under the windows logon prompt says "you have three voicemails waiting" |
| 18:17.47 | netsurfer | hehe |
| 18:18.42 | Zeeek | harryvv you can do it by making a custom sound file and telling the email client to play it |
| 18:18.45 | *** join/#asterisk smargaritis (~stratis@dyn164.kif8.nas.panafonet.gr) |
| 18:19.15 | Zeeek | in fact many POP mail checkers will do that and they're free |
| 18:20.15 | harryvv | Zeek, know of any ? this would be for vm only. |
| 18:20.26 | netsurfer | make a new pop account for vm only |
| 18:20.42 | Zeeek | well that'd require either presorting to a folder or what netsurfer said |
| 18:20.56 | Zeeek | which to me makes the ost sense |
| 18:20.59 | Zeeek | most |
| 18:21.43 | Zeeek | http://www.webattack.com/freeware/comm/fwemcheck.shtml |
| 18:22.07 | harryvv | no need to make another pop account. Idea is to ontify user who logs in or already logs of vm in his/her mail box. |
| 18:22.10 | Zeeek | Inbox can alert you via a popup hint or sound and launch an application of your choice or a web page to retrieve the mail. |
| 18:22.18 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-49-89.w82-122.abo.wanadoo.fr) |
| 18:22.23 | *** join/#asterisk abbas_ (nidobas@203.81.218.2) |
| 18:22.28 | Zeeek | "ignore them, launch a program and more. You can also reply to messages, delete them from the server, play custom sound notifications and more" |
| 18:22.29 | *** join/#asterisk smargaritis (~stratis@dyn164.kif8.nas.panafonet.gr) |
| 18:22.45 | Zeeek | Poptray: freeware |
| 18:22.46 | abbas_ | Hi All |
| 18:23.31 | harryvv | I want it to limit the mail checker to vm only :) |
| 18:23.32 | abbas_ | Zeeek can u pls help me |
| 18:23.36 | abbas_ | http://pastebin.ca/5405 |
| 18:23.44 | Zeeek | "Freeware programs are either distributed for the love of humanity, for fame, or as stripped down versions of programs that do cost money. " |
| 18:24.01 | abbas_ | my ATA is unable to make calls some times get registered on * and most of the time reg fails |
| 18:24.06 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 18:24.25 | Zeeek | I see that but I odn't know why |
| 18:24.29 | AgiNamu | <PROTECTED> |
| 18:24.36 | AgiNamu | WTF? I'm bridging 2 G729 devices |
| 18:24.37 | Zeeek | client behind NAT? |
| 18:24.38 | AgiNamu | and it's doign that |
| 18:24.45 | AgiNamu | (using IAX2) |
| 18:25.29 | Zeeek | harryvv that last program is able to filter by subject I think so it does what you need |
| 18:25.41 | abbas_ | Zeeek noway NAT |
| 18:26.01 | Zeeek | no NAT? |
| 18:26.10 | Zeeek | so is nat=yes in sip.conf ? |
| 18:26.26 | abbas_ | yes |
| 18:26.35 | Zeeek | why if there is no NAT? |
| 18:26.47 | abbas_ | some times i use the same machine behind NAT |
| 18:26.48 | *** join/#asterisk Tili (~Tili@202-133-65-221-dialup.sat.net.pk) |
| 18:26.49 | abbas_ | thats y |
| 18:27.06 | *** join/#asterisk tih (~tih@athene.hamartun.priv.no) |
| 18:28.37 | cypenguin | hi all |
| 18:28.53 | abbas_ | hi |
| 18:29.12 | abbas_ | Zeeek seen the page ? |
| 18:29.13 | multrix | hi everyone, I saw first time an IAX2-phone, what do you think of this ? |
| 18:29.30 | cypenguin | i am trying to record a voicemessage but Record() cut me off after 40 sec. anybody has similar problem/solution? |
| 18:29.37 | markit | hi anthm :) do you have some spare time for a question about atxfer? |
| 18:30.21 | multrix | iaxtalk.com |
| 18:30.22 | Zeeek | multrix, what IAX2 phone? |
| 18:30.40 | multrix | Zeeek: is there another that these one ? |
| 18:30.45 | anthm | no but you can ask and i might answer if im not distracted |
| 18:31.02 | Zeeek | I don't see what phone you're talking about |
| 18:31.12 | Zeeek | there is farfon |
| 18:31.15 | Zeeek | almost |
| 18:31.33 | abbas_ | :) |
| 18:31.45 | abbas_ | Zeeek have u seeen the config page of my device? |
| 18:31.47 | tzanger | anyone familliar with multimode fiber? Is it "ok" to have a run that requires four junctions if the run is under a couple hundred feet? |
| 18:31.47 | markit | anthm: it's really an interesting feature, since it supports also monitor, but what about if I transfer the call to an extension where no one is answering? what keys do Ihave to press to go back to the caller? |
| 18:31.57 | Zeeek | abbas_ yes and I have no ideas |
| 18:32.02 | abbas_ | :( |
| 18:32.04 | tih | tzanger: shouldn't be a problem. |
| 18:32.11 | ngb | Zeeek what's protocol is beeter iax or sip ? |
| 18:32.14 | PakiPenguin | abbas_: still the same problem eh |
| 18:32.17 | PakiPenguin | ? |
| 18:32.19 | tih | tzanger: I've done that lots of times. |
| 18:32.21 | PakiPenguin | ngb: iax! |
| 18:32.25 | abbas_ | ok suggest according to the pastebin |
| 18:32.32 | abbas_ | Zeeek ok suggest according to the pastebin |
| 18:32.37 | anthm | whatever you have mapped to hangup in featuremap |
| 18:32.38 | abbas_ | yah Paki same problem |
| 18:32.38 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 18:32.39 | tzanger | tih: are there rules of thumb for how many junctions are able to be handled without losing signal integrity? Obviously we're talking proper welded connections here |
| 18:32.54 | ngb | PakiPenguin iax is more quick that sip ? |
| 18:32.57 | Zeeek | post again the pastebin and someone brighter than me will look |
| 18:32.59 | tih | tzanger: probably, but I don't know them. I'm just a sysadmin. ;-) |
| 18:33.08 | markit | anthm: is it related with "disconnect" in featuremap? seems not to work (I've mapped to *0) |
| 18:33.12 | PakiPenguin | ngb: more bandwidth friendly |
| 18:33.20 | tzanger | tih: www.mixdown.ca/~andrew/fiberplan.pdf is what I'm proposing |
| 18:33.25 | abbas_ | http://pastebin.ca/5405 |
| 18:33.26 | ngb | PakiPenguin oke thanks |
| 18:33.30 | tzanger | the run between enclosures is about 70 feet |
| 18:33.35 | Zeeek | PakiPenguin since you arrived late you have to go see ngb website |
| 18:33.59 | tih | tzanger: 404 Not Found |
| 18:34.05 | PakiPenguin | hey Zeeek :) |
| 18:34.14 | Zeeek | hello |
| 18:34.20 | tzanger | tih: bah... www.mixdown.ca/~andrew/dump/fiberplan.pdf |
| 18:34.42 | abbas_ | PakiPenguin kuch karo yaar bahut takleef ho rahee hay |
| 18:34.51 | *** join/#asterisk fizbar (~jason@dsl-232-31.speedsite.com) |
| 18:35.09 | Zeeek | I ate some of that once and was stuck in the john for three days! |
| 18:35.11 | abbas_ | meraa ATA chalao |
| 18:35.12 | anthm | you may be out of luck while its ringing atm |
| 18:35.16 | PakiPenguin | abbas_: will send you ata-186 tomorrow , i promise :p |
| 18:35.23 | abbas_ | hahaha |
| 18:35.25 | abbas_ | ok |
| 18:35.28 | tih | tzanger: I can't imagine any problems with that. We routinely do worse, with no problems. |
| 18:35.31 | letherglov | tzanger, using your patch panels as routing boxes? |
| 18:35.32 | abbas_ | thats another topic u know |
| 18:35.36 | letherglov | at least they're labelled! |
| 18:35.37 | tzanger | letherglov: kind of. :) |
| 18:35.38 | multrix | tzanger: what type of connexion runs on this mm fiber ? |
| 18:35.39 | letherglov | ;-) |
| 18:35.40 | abbas_ | is ko bhee to sahee hona chahiye naaa |
| 18:35.43 | letherglov | oh |
| 18:35.44 | PakiPenguin | yeah :) |
| 18:35.51 | letherglov | and at least gigabit laser is visible...100 mbit isn't |
| 18:35.56 | tzanger | multrix: 100baseFX |
| 18:35.59 | letherglov | don't stare at it though |
| 18:36.00 | PakiPenguin | abbas_: configure it as a sipura |
| 18:36.01 | letherglov | that's a pain. |
| 18:36.08 | abbas_ | i have tried it |
| 18:36.15 | markit | anthm: do you mean your phone is ringing? damn, I'm waiting for atxfer help since 10 days :( |
| 18:36.20 | fizbar | Has anyone ever had problems with the digium tdm400p card on a 2.6 kernel running udev? It acts like the card isn't being reset properly after a call. |
| 18:36.21 | tih | letherglov: classic warning label: "WARNING: DO NOT LOOK INTO LASER WITH REMAINING EYE!" |
| 18:36.21 | mrempire | abbas: kaise |
| 18:36.29 | abbas_ | i have upgrade it through Sipura firmware and it was successfully |
| 18:36.41 | Zeeek | heh tih |
| 18:36.46 | multrix | tzanger: of course it's ok !! we can run until 2 km and 5 to 10 db attenuation ! so you're really far from having problem, be cool :) |
| 18:37.01 | anthm | no i mean once the call is answered by something you can disconnect but you may not be able to cancel until then |
| 18:37.03 | tzanger | multrix: ok, danke |
| 18:37.04 | PakiPenguin | mrempire: urdu jantay hou? |
| 18:37.13 | abbas_ | mrempire aisay hee bus |
| 18:37.14 | abbas_ | ;) |
| 18:37.22 | mrempire | PakiPenguin:hindi |
| 18:37.23 | multrix | so nobody tried these new IAX2 phones and ATA ? |
| 18:37.37 | multrix | tzanger: bitte schön |
| 18:37.41 | ngb | /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lXext |
| 18:37.42 | ngb | collect2: ld returned 1 exit status |
| 18:37.43 | Zeeek | who makes it multrix? |
| 18:37.44 | markit | anthm: oh :( is something that can be implemented, or can't work with asterisk design? it's really needed |
| 18:37.45 | ngb | make[1]: *** [pbx_gtkconsole.so] Error 1 |
| 18:37.45 | ngb | make[1]: Leaving directory `/download/asterisk_tmp/asterisk/pbx' |
| 18:37.45 | ngb | make: *** [subdirs] Error 1 |
| 18:37.54 | ngb | 1.0.5 |
| 18:37.58 | ngb | :| |
| 18:38.03 | PakiPenguin | brb |
| 18:38.51 | anthm | well in the meantime you may want to try transferring to an ext that calls answer |
| 18:39.08 | ngb | some one... can help me ??' |
| 18:39.23 | abbas_ | mrempire u from Indi a |
| 18:39.24 | ngb | when i compile the asterisk 1.0.5 |
| 18:39.26 | abbas_ | which city ? |
| 18:39.28 | multrix | tih: gigabit isn't so dangerous, SX is ok, LX begins to be dangerous if you stay in front of it, but longhaul is really dangerous... that's what I heard ! |
| 18:39.30 | ngb | i got this error |
| 18:39.33 | anthm | once the call hits answer the disconnect will work |
| 18:40.01 | abbas_ | anthm can u pls help me? |
| 18:40.13 | mrempire | abbas_:great, great grandfather, i was born in Suriname |
| 18:40.21 | markit | anthm: no, I need to go back to the caller... it's just me and my secretary, no other to transfer to |
| 18:40.29 | mrempire | abbas: I'm hindu |
| 18:40.46 | abbas_ | ok i am Muslim from pakistan |
| 18:40.51 | ngb | root@mypbx:/download/asterisk_tmp/asterisk# ldconfig -p |grep *Xext* |
| 18:40.51 | ngb | root@mypbx:/download/asterisk_tmp/asterisk# |
| 18:40.56 | Zeeek | no wars other than distro wars are allowed here :) |
| 18:40.57 | ngb | gooD! very good! |
| 18:41.00 | markit | anthm: and "going back" to the caller is a basic feature of assisted transfer |
| 18:41.06 | outtolunc | ok i am green and from mars <G> |
| 18:41.07 | *** join/#asterisk pelotas (~alek@host-ip130-216.crowley.pl) |
| 18:41.20 | anthm | umm i mean make an extension that calls answer() then Dial to the secretary device |
| 18:41.31 | anthm | then you can cancel it |
| 18:41.34 | abbas_ | anthm http://pastebin.ca/5405 |
| 18:41.42 | abbas_ | pls have a look on it |
| 18:41.46 | mrempire | Guys can you help: capiinfo says:capi not installed, i have compiled capi_chan |
| 18:41.55 | abbas_ | my device is is not getting reg on * |
| 18:41.56 | anthm | if you want i can remove the patch from cvs since it's not good enough for yo u |
| 18:42.07 | Zeeek | multrix there is no address on that site |
| 18:42.19 | markit | anthm: ? why remove? the fact that could be improved does not mean that is not good? |
| 18:42.35 | mrempire | How can I get chan_capi to work on fedoracore 3 |
| 18:42.58 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 18:43.56 | markit | anthm: I was simply asking you if there are technical reason not to have it work as expected, or just your lack of time (of course, I can't pretend you to make it perfect under my criteria) |
| 18:43.57 | Slainte | fixed my mp3 playback |
| 18:44.25 | Slainte | cant get Festival to work. no log file is being created so I dont know why it is returning ER |
| 18:44.52 | multrix | http://www.iaxtalk.com/ |
| 18:44.58 | *** join/#asterisk jsolares (~jsolares@mail.epa.com.gt) |
| 18:45.09 | Zeeek | yes but where is this company? No address on site - shipping VERY expensive |
| 18:45.35 | Zeeek | an IP Phone, the shipping fee will be $30 for US, $40 for Spain, $50 for Trinidad. |
| 18:45.49 | mikegrb | one would hope that is overnight |
| 18:45.52 | Zeeek | too bad if you're in Trinidad, huh? |
| 18:46.13 | netsurfer | mikegrb - with shipping prices like that i'd want it there within 30 seconds |
| 18:46.17 | Zeeek | it's DHL but there is no choice of shipping |
| 18:46.38 | mikegrb | ahh |
| 18:46.41 | mikegrb | is in china |
| 18:46.43 | mikegrb | bejing |
| 18:46.49 | anthm | I just told you how to get around it, to get it the way you want it would require a lot of work/restructuring atm if you perhaps are interesting in paying me to do this work along with enduring the political fustration it takes to get patches into CVS that is another story otherwise I suggested removing it cos it pisses me off that you think I have nothing better to do that hand craft asterisk for your needs. |
| 18:46.50 | Zeeek | how you findee? |
| 18:47.02 | Zeeek | I see no address anywhere? |
| 18:47.03 | Luke-Jr | Can anyone suggest any cheap way to get to one or two FXS ports from my server? (PCI, USB, Ethernet, etc) |
| 18:47.23 | Zeeek | somone already told you: ATA |
| 18:47.25 | JunK-Y | Luke-Jr: TDM400P |
| 18:47.27 | Nugget | Luke-Jr: what is "cheap" in your world |
| 18:47.33 | Zeeek | not TDM |
| 18:47.39 | JunK-Y | encourage digium. |
| 18:47.45 | mikegrb | Zeeek: |
| 18:47.46 | Zeeek | I do, I buy from them |
| 18:47.48 | mikegrb | <PROTECTED> |
| 18:47.48 | mikegrb | <PROTECTED> |
| 18:47.48 | mikegrb | <PROTECTED> |
| 18:47.48 | mikegrb | <PROTECTED> |
| 18:47.48 | mikegrb | <PROTECTED> |
| 18:47.50 | mikegrb | <PROTECTED> |
| 18:47.54 | markit | anthm: unpolite answer, I was just asking |
| 18:47.55 | Zeeek | nice |
| 18:48.01 | Nugget | I'd be more inclined to buy two IAXy boxes than a TDM400P. |
| 18:48.11 | Zeeek | well that explains the $60 phone and $30 shipping |
| 18:48.16 | anthm | unpolite is looking a gift horse in the mouth |
| 18:48.18 | Nugget | but if $200 isn't "cheap" to Luke, there's no suggestion he'll like. |
| 18:48.28 | JunK-Y | 200$ is nothing. |
| 18:48.34 | Nugget | *to Luke*. |
| 18:48.37 | Beirdo | the TDM400P isn't "cheap" in any sensible person's view :) |
| 18:48.40 | Zeeek | isn't there a $100 SIP box with two FXS? |
| 18:48.44 | Luke-Jr | Nugget: Preferably < $50 |
| 18:48.45 | Nugget | "cheap" is different for everyone |
| 18:48.49 | Nugget | see? :) |
| 18:48.52 | Beirdo | it's just not as expensive as other solutions :) |
| 18:48.55 | Luke-Jr | Zeeek: ATA isn't a generic term? |
| 18:48.56 | Zeeek | Luke-Jr muhahahahha $50 |
| 18:48.57 | JunK-Y | Nugget: moyuhjahahah ya no shit. |
| 18:49.01 | Nugget | Luke-Jr: then no, there's no solution. |
| 18:49.11 | Zeeek | eBay |
| 18:49.29 | abbas_ | Linksys, Grandstream are below 60 USD\ |
| 18:49.31 | markit | anthm: ok, waiting here or in mantis you to asking something or requiring something to someone else... |
| 18:49.34 | mikegrb | Luke-Jr: ATA is a generic term |
| 18:49.37 | Zeeek | not for 2 channels |
| 18:49.58 | Zeeek | anyway, Luke-Jr there is a lot of info about this on the wiki |
| 18:50.00 | Zeeek | http://www.voip-info.org/tiki-index.php |
| 18:50.07 | Zeeek | devices |
| 18:50.08 | Luke-Jr | Zeeek: already looked there |
| 18:50.10 | multrix | mikegrb: what's this adress you wrote ? a provider ? |
| 18:50.12 | *** join/#asterisk dolson (~dana@Sudbury-HSE-ppp3979883.sympatico.ca) |
| 18:50.15 | *** join/#asterisk Connor- (~billy@198-144-174-5.knx.tn.nxs.net) |
| 18:50.16 | Luke-Jr | It doesn't have recommendations, really |
| 18:50.17 | ngb | lXext <---- where i can download it ? |
| 18:50.17 | anthm | don't hold your breath |
| 18:50.26 | markit | LOL |
| 18:50.30 | Zeeek | no just a long list of stuff |
| 18:50.41 | Connor- | I've been having a issue with one of my carriers... For whatever reason, asterisk stops registereing.. |
| 18:50.54 | *** join/#asterisk Uajal (~icechat5@ool-182e86f3.dyn.optonline.net) |
| 18:50.55 | Zeeek | you guys saw the snom software phone free download? |
| 18:50.59 | Connor- | When I issue a sip reload, it re-registers... any ideas? |
| 18:51.00 | Zeeek | (oh, it's windows) |
| 18:51.04 | mikegrb | multrix: domain registrant for iaxtalk.com |
| 18:51.05 | techie | getting a Grandstream phone is like buying a piece of plastic with buttons |
| 18:51.17 | Zeeek | but they work great |
| 18:51.22 | techie | sure |
| 18:51.32 | Zeeek | once you find the firmware that works with yours |
| 18:51.40 | Zeeek | if... |
| 18:51.47 | anthm | i'll say 1 more time , if you simply call app_answer before you dail the device the hangup will work that is the best you can do until some more development is done on asterisk core. |
| 18:51.49 | Beirdo | for music on hold, does it only support MP3? |
| 18:51.52 | *** part/#asterisk penguin (~stratis@dyn164.kif8.nas.panafonet.gr) |
| 18:52.11 | multrix | mikegrb: ok kewl |
| 18:52.25 | multrix | mikegrb: I think I will try, it's the first IAX phone I found |
| 18:52.36 | shido6 | Zeeek that is so right |
| 18:52.39 | Zeeek | mikegrb - I guess the final point is, buy by credit card into china with no address and no recourse? I dunno :) |
| 18:52.59 | Zeeek | bad enough in EU |
| 18:53.00 | ngb | asterisk have this dependence ---------> lXext <---- where i can download it ? |
| 18:53.30 | Zeeek | besides that though, my BT work very well indeed... if you like SIP |
| 18:53.40 | Zeeek | waiting for a farfon to give it a spin |
| 18:54.01 | Zeeek | heh |
| 18:54.19 | Uajal | I am new to Asterisk. Which hardware should I have (which computer requirements, which card for T1, I will need to use several T1, possibly 4)? |
| 18:54.33 | JunK-Y | Uajal: take a T405P |
| 18:54.44 | djin | Or a TE410P |
| 18:54.55 | JunK-Y | same shit, just different voltage. |
| 18:55.01 | Uajal | of Digium? |
| 18:55.03 | Zeeek | you guys know that India has the lowest rate of prostrate cancer in the world, right? |
| 18:55.09 | harryvv | zeek, Im looking though the pop mail notifiers dont see one that would actually read heders and notify on that one from address only. |
| 18:55.26 | Zeeek | haryvv I already downloaded one that does that and more |
| 18:55.27 | *** join/#asterisk mbranca_home (~matteo@adsl-84-222-11-154.tiscali.it) |
| 18:55.35 | harryvv | which one is that |
| 18:55.37 | techie | you gotta love that business-class speakerphone |
| 18:55.38 | Zeeek | 'poptray' |
| 18:55.40 | mikegrb | Zeeek: you could use paypal virtual master card thing, generates single use credit card number for online purchase |
| 18:55.42 | harryvv | k |
| 18:55.59 | Zeeek | mikegrb I think I'll pass for the moment |
| 18:56.13 | Uajal | Should I buy TE410P or T405P from Digium or other provider? |
| 18:56.14 | Zeeek | but the phone does actually *look* decent |
| 18:56.19 | *** join/#asterisk Tornad (~Tornad@81.56.183.143) |
| 18:56.25 | JunK-Y | from digium yes. |
| 18:56.31 | Tornad | hi |
| 18:56.33 | Zeeek | techie if you mean BT it works great for us |
| 18:56.41 | Moc | Uajal, you might get the lastest revision if you get it from digium ;) |
| 18:56.43 | Zeeek | but we're not on it all day |
| 18:56.53 | slePP | for those of you that have asked about it, http://pastebin.ca/auction.php |
| 18:56.59 | slePP | so you can all battle it out :P |
| 18:57.08 | Tornad | is there a way to remove all old links and librairies of asterisk, zaptel and libpri |
| 18:57.29 | Tornad | i had upgrade to 1.0.5, but, now, no more zap stands up |
| 18:58.18 | Tornad | my 4 span are in RED alarm... no more links. I think there's an old libpri file which make all comes down |
| 18:58.26 | shido6 | are you using a PC or a Dual Xeon box |
| 18:58.26 | shido6 | ? |
| 18:58.32 | shido6 | Vajal? |
| 18:58.37 | shido6 | Uajal, rather |
| 18:58.57 | Uajal | Mok, what do you mean "latest revision"? Is it last version of Asterisk? |
| 18:59.06 | shido6 | no |
| 18:59.09 | shido6 | revision of hardware |
| 18:59.19 | shido6 | Asterisk can be downloaded from cvs |
| 18:59.27 | shido6 | FREE |
| 19:01.06 | harryvv | zeek, poptray download page is timing out. |
| 19:01.20 | harryvv | poptray works just a bad or dead link to poptray. |
| 19:02.13 | Delvar | nity night all |
| 19:02.27 | Moc | Uajal, not of the board itself |
| 19:02.37 | Moc | Uajal, no, of the board itself |
| 19:03.26 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 19:04.07 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 19:04.58 | Zeeek | harryvv works fine here: http://www.snapfiles.com/download/dlpoptray.html |
| 19:05.29 | Slainte | any festival users here. I get a app_festival.c:444 festival_exec: Festival returned ER |
| 19:05.44 | Slainte | when I try a Executing Festival("SIP/110-1ba2", "Hello asterisk user, how are you today?") in new stack |
| 19:06.08 | Slainte | but there is no logfile |
| 19:09.04 | *** join/#asterisk ghoti (paul@haggis.it.ca) |
| 19:09.12 | outtolunc | isn't poptray the one that as listed on a malware site <G> |
| 19:10.31 | Uajal | How much T1 lines can be inserted into TE410P or T405P? |
| 19:10.56 | mogorman | 4 |
| 19:12.02 | djin | http://www.digium.com/index.php?menu=hardware_products |
| 19:12.28 | Uajal | I sow on picture that there are 4 jacks in the card. Each one can be T1 or E1? Will board determine whether it is T1 or E1 itself? |
| 19:13.07 | *** join/#asterisk keith778 (~kobrien@ip-207-145-80-2.nyc.megapath.net) |
| 19:13.20 | vaewyn | Uajal: is part of the software setup |
| 19:13.32 | djin | or jumpers. |
| 19:13.38 | Tornad | Uajal, There is jumper on the board to force E1 or T1 |
| 19:13.46 | vaewyn | that also :} |
| 19:14.05 | vaewyn | (sorry... forgot software is only signalling) |
| 19:14.52 | vaewyn | Wish out meridian would let me mix Ts and Es... cause then I would use Es for connecting with * and get the extra channels |
| 19:15.03 | Uajal | Will asterisk work with all 4 T1 lines (will it be able generate such phone traffic if it is used as autodialer)? |
| 19:15.23 | mogorman | yes |
| 19:15.26 | vaewyn | Uajal: with an appropriately sized machine yes... |
| 19:16.56 | Uajal | I found http://www.digium.com/index.php?menu=config_packages. There are some hardware requirements. You mean such requirements or I should ask Digium about them? |
| 19:18.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 19:19.08 | *** join/#asterisk modulus_ (modulus@rm-f.net) |
| 19:19.09 | modulus_ | bleh |
| 19:19.20 | shido6 | u have to know what you want to do first |
| 19:19.37 | vaewyn | Uajal: depends if you are going to translating codecs... passthrough.... etc.. |
| 19:20.08 | shido6 | how many simultaneous users you want to push through your box |
| 19:20.15 | shido6 | how many ppl will be accessing vmail |
| 19:20.22 | shido6 | will u have calling card customers? |
| 19:20.26 | shido6 | do you need billing? |
| 19:20.31 | shido6 | etc |
| 19:20.32 | vaewyn | straight IAX <-> T1/E1 from u/a law you can almost drive 2 quad boards with a dual xeon |
| 19:20.52 | vaewyn | add g729 and good luck getting 1 T/E card fulkl with a dual xeon |
| 19:21.20 | shido6 | err |
| 19:21.21 | shido6 | it works |
| 19:21.28 | Uajal | Can I try asterisk on regular LAN connection to Internet 100 Mbps without buying Digium cards for T1/E1? |
| 19:21.28 | shido6 | for the past 3 yrs :) |
| 19:21.34 | shido6 | yes you can |
| 19:21.57 | shido6 | you can have us48toll free inbound numbers, and outbound dialing through us , canada and place international calls |
| 19:22.22 | Uajal | Thans for answers |
| 19:22.42 | Slainte | I ahve a mismatch in my dialplans between my IP600 polycom phone and my * extensions.conf |
| 19:22.54 | shido6 | whats the CLI say? |
| 19:22.58 | *** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 19:23.13 | *** join/#asterisk jsolares (~jsolares@mail.epa.com.gt) |
| 19:23.47 | Slainte | if I hit the new Call button on the phone and try to dial a number the phone trunks it at 7 digits. |
| 19:24.08 | Slainte | if I type all digits and then press send, the phone sends the whol group to * server |
| 19:24.33 | Slainte | I think it is because of the dialplan built into the phone. |
| 19:24.40 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 19:24.54 | shido6 | u need to look at the config of the phone AND your dialplan |
| 19:24.57 | *** join/#asterisk keith778 (kobrien@216.sub-70-213-14.myvzw.com) |
| 19:25.42 | shido6 | pastebin.ca both :) |
| 19:25.45 | Uajal | Can I try Asterisk in Windows as AstWind to try it as Autodialer through regular LAN? Or it is recommended to install Linux and try asterisk itself? |
| 19:26.09 | shido6 | the latter |
| 19:28.02 | netsurfer | what the fxxk is it with these damn unresolved symbols in zaptel install |
| 19:29.04 | Slainte | shido, do you know much about the polycom IP600's? |
| 19:30.05 | *** join/#asterisk egrat (1005@pcp03275560pcs.pthurn01.mi.comcast.net) |
| 19:30.48 | egrat | I am new to asterisk and was wondering if anyone could answer some questions for me... |
| 19:30.58 | ChulJin | we could certainly try. :) |
| 19:31.29 | egrat | Well, I am the network admin for an ISP and we are looking at doing VoIP. |
| 19:31.41 | egrat | Would Asterisk be a good solution for that kind of market? |
| 19:31.53 | egrat | Or would we want to find some other commercial system? |
| 19:31.57 | djin | depending on requirements, yes. |
| 19:32.05 | denon | has anyone seens grandstreams repeatedly and frequently lose their SIP registration with current *? |
| 19:32.06 | redder86 | egrat: you're looking for an unbiased opinion on #asterisk ? |
| 19:32.10 | egrat | We are doing broadband wireless. |
| 19:32.16 | denon | its happening occasionally with a 7960 here too |
| 19:32.30 | ChulJin | there are some successful ITSPs running wholly on asterisk...nufone comes to mind first...I believe voicepulse too |
| 19:32.32 | egrat | Well, not necessarily unbiased, just informed about the product. :) |
| 19:32.46 | redder86 | egrat: you can do VoIP with Asterisk, yet |
| 19:32.51 | Nivex | do the cisco 7940's come with SIP firmware by default? |
| 19:32.51 | redder86 | s/yet/yes/ |
| 19:33.14 | egrat | How would we go about using Asterisk to provide dial-out to POTS? |
| 19:33.21 | Nugget | Nivex: no, you have to download it from cisco.com |
| 19:33.25 | Slainte | Nivex no |
| 19:33.28 | Uajal | At http://www.voip-info.org/tiki-pagehistory.php?page=AstWind&diff=2 I found that AstWind "Linux installation isn't actually installed. So etc/networking/interfaces and etc/resolv.conf and etc/hosts can't be edited, because they simply don't exist. When you log in, the once-off upgrade script TRIES to upgrade, but finding nothing there it fails. And so it all has to be uninstalled and reinstalled to try again." Is it true? |
| 19:33.30 | redder86 | egrat: get a PSTN interface card like a TE405P |
| 19:33.39 | egrat | I assume we would need some PRIs plugged into the linux box? |
| 19:33.57 | ChulJin | egrat: digium, the biggest sponsor of *, sells various cards for just that purpose |
| 19:34.37 | djin | egrat, depends. You can also link to a VOIP Wholesale. |
| 19:34.47 | ChulJin | egrat: exactly...and to facilitate that, the interface card that redder just mentioned |
| 19:34.55 | djin | And just use SER for SIP. |
| 19:35.01 | Nivex | Nugget, Slainte: thanks. |
| 19:35.13 | egrat | Ok, so the PRIs plug into the TE405P and we need to have a dial-out service on the PRI? |
| 19:35.27 | Nivex | I was hoping to whip up an Asterisk box here at $WORK to have as a testbed before their cisco callmanager arrived |
| 19:35.31 | egrat | We have 24 PRIs coming in for dial-up users right now. |
| 19:35.53 | egrat | I have no idea if we can dial out on these or not... |
| 19:35.54 | Slainte | egrat are they comming in via DS3 ? |
| 19:36.04 | Uajal | Did somebody try to upgrade AstWind on Windows? |
| 19:36.04 | egrat | Yes, on a MUX. |
| 19:36.53 | Slainte | so you have teh Ds3, pluggedinot a AS5300/5400 or some sort of NAS like a TNT? |
| 19:37.00 | Slainte | plugged into |
| 19:37.08 | *** join/#asterisk abbas (nidobas@203.81.218.2) |
| 19:37.34 | shido6 | Nivex, DO IT |
| 19:37.45 | egrat | We have a DS3 that goes to a WideBank DS3 Multiplexer which then has individual T1s that plug into a AS5300. |
| 19:37.54 | ChulJin | hmmm...are there any other implementations of IAX2 as a server besides *? |
| 19:37.58 | shido6 | smartjack ports? |
| 19:38.16 | shido6 | ChulJin, why would you use anything other than asterisk for iax2? |
| 19:38.27 | shido6 | Inter Asterisk eXchange Protocol :) |
| 19:38.38 | ChulJin | shido: but they did release it to the world. |
| 19:38.55 | ChulJin | I ask only because...I was listing a few ITSPs that use * before... |
| 19:39.03 | Uajal | Did somebody work at all with AstWind? |
| 19:39.06 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 19:39.22 | ChulJin | and I just wanted to make sure that my assumption that 'if they support IAX2, they must be using asterisk' was a good one. |
| 19:39.24 | Slainte | egrat, there are a number of ways to do it. |
| 19:39.37 | *** part/#asterisk hans (fugalh@falcon.fugal.net) |
| 19:39.52 | egrat | So, I guess what I really need to know is, what kind of PRI services would/should I get for doing dial-out long distance? |
| 19:40.01 | egrat | And what kind of port density could I expect? |
| 19:40.14 | Slainte | I dont think those are the first questions to ask |
| 19:40.22 | egrat | If I have 100 customers, would I need just 1 or 2 or 3 PRIs? |
| 19:40.23 | Slainte | but they are some valid questions. |
| 19:40.38 | ctooley | has anyone written a handy utility for listing the current channels and what those channels are doing? |
| 19:40.41 | Slainte | all depends on if they are home users, business users, or a mix of both |
| 19:40.41 | wankel | depends on your customers |
| 19:40.49 | ctooley | kind of like a ps for asterisk channels |
| 19:40.52 | ChulJin | egrat: rather than how many customers, consider 'how many simultaneous calls' |
| 19:40.56 | egrat | And how fast of a machine should I get for running Asterisk? |
| 19:41.08 | ChulJin | ctooley: asterisk -rx 'show channels'? |
| 19:41.11 | Nugget | ctooley: you mean other than "show channels"? |
| 19:41.13 | harryvv | depending on type of bussiness and what kind of call volume it will generate egrat |
| 19:41.13 | wankel | good luck getting any scaling data. the only people who have it guard it jealously. |
| 19:41.22 | egrat | So, if one customer has this VoIP service, do they get to make more than one call at a time? |
| 19:41.29 | Nivex | shido6: will it be able to handle the Skinny? I've heard that *'s skinny impl isn't that good |
| 19:41.32 | ChulJin | egrat: that's your choice. |
| 19:41.35 | ctooley | ChulJin, Nugget yeah I was thinking something a little prettier |
| 19:41.37 | shido6 | yes |
| 19:41.38 | wankel | there's some conflicting suggestions on the wiki, but i don't place much faith in them. |
| 19:41.53 | egrat | You mean it is possible for them to do it if we set it up to work on our end? |
| 19:42.02 | shido6 | you can oversubsribe your T1's yes |
| 19:42.15 | *** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl) |
| 19:42.22 | shido6 | egrat they can make as many calls as their bandwidth will allow, this is how we do it |
| 19:42.23 | ChulJin | yup, it is either possible or impossible to have simultaneous calls, as you yourself choose. |
| 19:42.30 | shido6 | and if bandwidth is a problem we turn on trunking |
| 19:42.35 | shido6 | on both ends |
| 19:42.44 | *** join/#asterisk sskyles (~sskyles@155.205.205.68.cfl.rr.com) |
| 19:42.49 | egrat | what do you mean turn on trunking? |
| 19:42.52 | shido6 | ChulJin you can place as many and receive as many as your bandwidth will allow |
| 19:43.02 | shido6 | trunking slims the packets down consideradbly |
| 19:43.31 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 19:43.32 | wankel | iax doesn't do that by default? |
| 19:43.45 | egrat | but what do you mean by trunking? |
| 19:43.56 | shido6 | no |
| 19:43.58 | egrat | is it multiplexing or what? |
| 19:43.59 | shido6 | u have to turn it on |
| 19:44.05 | shido6 | trunk=yes in the peer or user |
| 19:44.12 | shido6 | wow, ok |
| 19:44.17 | wankel | that's silly. |
| 19:44.17 | shido6 | forget the old Telecom world |
| 19:44.32 | dan2 | oej: ping |
| 19:44.35 | jsolares | what does trunking do? |
| 19:44.48 | ChulJin | shido: sorry. I thought he was asking from the PoV of what kind of services he can/should offer his customers. |
| 19:44.54 | ChulJin | I apologise. |
| 19:44.55 | shido6 | minimizes the amount of bandwidth used for simultaneous calls to a single endpoint |
| 19:45.11 | egrat | hold on a sec... have to take a call... |
| 19:45.15 | dan2 | what happended to bkw? |
| 19:46.03 | Slainte | egrat, if you are looking to turn your ISP into a voice provider there are a tonne of issues to look at. You need to be organized about it otherwise you are going to waste alot of time |
| 19:46.16 | shido6 | and money :) |
| 19:46.31 | greg_work | iax2 + trunking: http://voip-info.org/wiki-Asterisk+bandwidth+iax2 |
| 19:46.38 | eipi | newbie question: runnning asterisk i receive "res_odbc: Error Data source name not found...:", where i can read about this? |
| 19:46.41 | shido6 | yeah |
| 19:46.45 | shido6 | but he only used 2 calls |
| 19:47.08 | shido6 | which isnt the best route to take if ur using trunking you want to slim a few dozen calls or so and the benefits improve |
| 19:47.14 | shido6 | the more calls u make |
| 19:47.15 | eipi | (asterisk is looking for asterisk dsn, but i never configured) |
| 19:47.17 | shido6 | simultaneously |
| 19:47.37 | mrempire | more then 4 call is the turning point |
| 19:47.44 | shido6 | for example in our studies u can push about 90+ on a 1.5 meg connection |
| 19:48.06 | abbas | d |
| 19:48.15 | abbas | ~LucyAli |
| 19:48.25 | abbas | ~ LucyAli |
| 19:48.35 | abbas | ~help |
| 19:48.47 | wankel | shido: what codec? |
| 19:49.25 | wankel | eipi: it's configured to use an ODBC data source for the config files. the wiki has info on configuring the static and realtime data source stuff. |
| 19:49.44 | shido6 | g729 or speex use next no proc resources and our systems bang out the calls effortlessly |
| 19:49.51 | shido6 | but Jeremy has tweaked a lot of things |
| 19:50.01 | shido6 | a LOT of things |
| 19:51.02 | wankel | that sounds a little low for 1.5mbps for G729, but i'm not sure how efficient iax2 trunking is compared to cRTP |
| 19:51.44 | shido6 | avg numbers |
| 19:52.20 | cbachman | shido6, is that with trunking enabled on that 90+ ? |
| 19:52.23 | *** join/#asterisk Tough_Nuts (~Tough_Nut@204.110.228.254) |
| 19:52.28 | shido6 | yes |
| 19:52.36 | *** join/#asterisk tafazzi (~ddrig@83.224.64.56) |
| 19:52.39 | egrat | ok, i'm back. |
| 19:52.52 | egrat | and yes, we do want to turn our ISP into doing VoIP as well. |
| 19:53.04 | cbachman | thanks, just a useful data point |
| 19:53.07 | shido6 | what services do you want to offer? |
| 19:53.08 | egrat | we are willing to consider anything that will work and has a decent profit margin on it. |
| 19:53.25 | egrat | we basically want it to be as simple as using a normal telephone. |
| 19:53.39 | egrat | our customers aren't too bright and we do not want this to be over their heads. :) |
| 19:53.53 | wankel | shouldn't be hard for the customers regardless of the solution you go with |
| 19:53.55 | shido6 | do u want to be a Broadband VoIP service? |
| 19:54.03 | shido6 | or are you going ahead a step further |
| 19:54.13 | ChulJin | er, well...start again |
| 19:54.22 | shido6 | are you a CLEC? |
| 19:54.27 | ChulJin | what exactly do you mean by 'doing VoIP'? |
| 19:54.36 | egrat | so, we are really looking at doing something like plug a VoIP telephone into their router that goes over our broadband connection to our VoIP server and then to wherever they are calling. |
| 19:54.37 | ChulJin | what features do you want to offer to your customers? |
| 19:54.46 | ChulJin | ah, gotcha. |
| 19:54.48 | shido6 | ok |
| 19:54.51 | ChulJin | eminently doable with * |
| 19:55.13 | egrat | we want E911, Call Waiting/Forwarding, Voicemail, caller ID, mainly. |
| 19:55.29 | modulus_ | so... |
| 19:55.29 | wankel | e911 is hard. the res is easy. |
| 19:55.35 | modulus_ | what's with all the VOIP hype these days? |
| 19:55.35 | wankel | s/res/rest/ |
| 19:55.47 | *** join/#asterisk Inv_arp (junya@adsl-8-230-175.mia.bellsouth.net) |
| 19:55.53 | wankel | modulus: i dunno. voip's already flopped twice. you'd think people would be over it :) |
| 19:56.02 | egrat | we would settle without having e911, but would like it if its possible. |
| 19:56.12 | shido6 | it isnt hype anymore as it works well with Asterisk |
| 19:56.13 | *** join/#asterisk ngb (~joshua@200.49.156.89) |
| 19:56.23 | ChulJin | modulus: it's the 5678's 'Whoo-hoo' in the Vonage commercials. Now that VoIP has a catchy theme song, everybody wants a piece. :P |
| 19:56.27 | ngb | why asterisk 1.0.5 need x11 ? |
| 19:56.35 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543956pcs.mainf01.in.comcast.net) |
| 19:56.39 | ngb | i'nt wanna rut it on x11 |
| 19:56.46 | egrat | we don't mind "hard" as we have done many things already that people have told us were "hard" or even "impossible". |
| 19:56.50 | wankel | it worked well before asterisk. sip is the biggest improvement lately. it STILL sucks with NAT. |
| 19:56.52 | ngb | i'nt wanna run it on x11 |
| 19:57.05 | ngb | /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lX11 |
| 19:57.25 | ngb | how i can compile it without x11? |
| 19:57.32 | ChulJin | i'nt=i don't? |
| 19:57.33 | egrat | we are mainly researching our options so far, but we are collocated with a CLEC that has awesome pricing on PRI services ($300 a line). |
| 19:57.33 | modulus_ | "hasidic reggae beatbox superstar" |
| 19:57.35 | modulus_ | wtf? |
| 19:57.45 | ngb | ChulJin yes |
| 19:57.47 | egrat | and we want to leverage that pricing. |
| 19:57.57 | shido6 | thats why we use IAX for NAT |
| 19:57.58 | ChulJin | modulus: is that about kram or jerjer? |
| 19:58.05 | shido6 | and IAXys |
| 19:58.06 | modulus_ | chuljin, nope |
| 19:58.06 | ngb | i don't run it on x11 |
| 19:58.09 | modulus_ | lol |
| 19:58.21 | wankel | shido6: that'd be great except that then you need something that supports IAX on the other end. |
| 19:58.30 | ngb | how i can make asterisk without x11 ? |
| 19:58.45 | ChulJin | OK re: kram's 'my script saw my nick in the channel, I will now NOTICE you': WTF is a BishopChicken? |
| 19:58.48 | ngb | i don't like GUI |
| 19:58.50 | wankel | ngb: no one has even had a chance to answer the previous three times you asked. |
| 19:59.03 | djin | Asterisk doesn't have a X11 gui. . |
| 19:59.05 | ChulJin | ngb: if you don't like GUI's, you will LOVE asterisk. :P |
| 19:59.23 | shido6 | err |
| 19:59.27 | shido6 | well |
| 19:59.31 | shido6 | ok i'll shut up |
| 19:59.43 | shido6 | gastman isnt a real interface |
| 19:59.57 | ngb | ChulJin yes! but in version 1.0.5 it require x11 :( |
| 20:00.07 | djin | nope. |
| 20:00.07 | bjohnson | ngb: no |
| 20:00.10 | ngb | /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lX11 |
| 20:00.44 | ngb | bjohnson when i run make, it stop printme |
| 20:00.45 | ngb | /usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lX11 |
| 20:00.45 | ngb | collect2: ld returned 1 exit status |
| 20:00.45 | ngb | make[1]: *** [pbx_gtkconsole.so] Error 1 |
| 20:00.57 | modulus_ | * 1.0.5 requires x11? |
| 20:01.13 | bjohnson | ngb: how did you get the source? |
| 20:01.18 | wankel | if you try to compile something called gtkconsole, probably. |
| 20:01.23 | ngb | bjohnson cvs |
| 20:01.29 | wankel | i suspect if you don't compile that part, you won't need x11 :) |
| 20:01.36 | *** join/#asterisk CMike (~a_mike@c-304171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 20:01.48 | eipi | wankel: im newbie, i installed asterisk latest version from CVS, and when it starts looks for odbc dsn... but i cant found how to configure odbc or what tables asterisk needs on mysql |
| 20:02.18 | wankel | eipi: your config files point to ODBC, then. use the default ones that come in the package to start with. |
| 20:02.28 | ngb | wankel where i can disable the gtk console... i don't wanna it |
| 20:02.33 | ngb | x11 sucks |
| 20:02.45 | wankel | yes, you've successfully communicated that you don't like GUIs |
| 20:02.48 | wankel | several times |
| 20:02.55 | ngb | :/ |
| 20:03.00 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 20:03.02 | bjohnson | a clean cvs co shouldn't require X at all |
| 20:03.05 | shido6 | eipi |
| 20:03.08 | shido6 | if u dont have it |
| 20:03.09 | shido6 | turn it off |
| 20:03.13 | bjohnson | make sure you get stable cvs version |
| 20:03.16 | shido6 | in "/etc/asterisk/modules.conf" |
| 20:03.30 | bjohnson | follow make instructions on wiiki .. make clean, make, make install |
| 20:03.40 | wankel | i think it does build gtkconsole. |
| 20:03.41 | shido6 | make clean install works |
| 20:03.43 | mrempire | wget ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.5.tar.gz |
| 20:03.48 | wankel | i've got the x libs on my box, though, so it would work here. |
| 20:03.57 | wankel | i don't think i have a box without x libs to test on. |
| 20:04.05 | ngb | bjohnson i run: cvs co -r v1-0-5 asterisk |
| 20:04.16 | bjohnson | wankel: I don't have X on my * server and people would scream if the default make required X |
| 20:04.28 | shido6 | then dont run .5 |
| 20:04.30 | shido6 | yeesh |
| 20:04.53 | shido6 | if u want the easy way out run another version |
| 20:05.01 | wankel | or just edit the makefile |
| 20:05.01 | netsurfer | has ANYONE figured out why a standard install of debian causes all those damn unresolved symbols messages when u install zaptel ? |
| 20:05.20 | shido6 | got the kernel sources installed, netsurfer? |
| 20:05.21 | bjohnson | editing the makefile would be the way to tell for sure |
| 20:05.25 | netsurfer | shido6 - yes |
| 20:05.35 | netsurfer | and soft link to /usr/src/linux |
| 20:06.35 | jsolares | netsurfer: have you tried getting the zaptel sources from cvs? |
| 20:06.50 | netsurfer | jsolares - yes.. thats what im using |
| 20:06.52 | jsolares | i had a .tar.gz that was being a bitch, i got the one from cvs and it worked the first time |
| 20:07.03 | jsolares | hrm, then no idea and gluck |
| 20:07.04 | wankel | are you sure they're the right kernel sources? i had a hell of a time getting the right sources for the kernel that was on the debian box i took over. the ones in the package with the same name were not correct. |
| 20:07.29 | egrat | ok, another question I have about this VoIP stuff... how does anyone make money on it when offering unlimited local and long-distance calling? |
| 20:07.32 | netsurfer | wankel - that could be the problem.. |
| 20:07.43 | Qwell | egrat: "unlimited" isn't "unlimited" |
| 20:07.55 | egrat | well, then how does that work? |
| 20:08.05 | jsolares | one does not spend 24/7 on the phone talking |
| 20:08.20 | Slainte | egrat, it is complex, but you need to look at the whole thing in a methodical manner |
| 20:08.24 | egrat | yes, but what if you are spending 10 hours a day on the phone? |
| 20:08.36 | Qwell | Then an "unlimited" plan won't work for you. |
| 20:08.39 | jsolares | it's a bet that the avg used wont be more expensive than what you collect from montly fees |
| 20:08.47 | wankel | then you may lose money on that customer. but you'll make money on other customers. |
| 20:09.06 | wankel | you're rolling the dice and betting that most customers won't get their moeny's worth. same thing all-you-can-eat restaurants do. |
| 20:09.27 | egrat | i understand the laws of averages... it just that the rates I see for calling are about $0.025 per minute. and that could add up very quickly. |
| 20:09.48 | wankel | your rates or what you plan to charge? |
| 20:09.53 | Qwell | many places have a hard set number for what "unlimited" is, don't they? |
| 20:10.08 | Qwell | ie; if you go over 10,000 minutes a month, we're gonna start charging $.02 per minute |
| 20:10.19 | egrat | well, i was figuring $29.95/mo for unlimited, but thats not much more than 1000 minutes |
| 20:10.28 | netsurfer | wankel - which headers file did you end up apt-get 'ing ? |
| 20:10.35 | shido6 | hrmm |
| 20:10.55 | shido6 | when you see the word "unlimited" you have to locate the fine print |
| 20:11.03 | Qwell | precisely |
| 20:11.12 | wankel | netsurfer: hmm. can't remember. in the end i had to hack a few of the headers by hand. there were some macros that were wrong in the ancient kernel on the debian stable box. |
| 20:11.16 | shido6 | when companies use the word Unlimited it is a psychological tactic to suck customers in |
| 20:11.19 | wankel | i finally just upgraded |
| 20:11.32 | shido6 | then they cover themselves by giving a totally different definition of the word in their fine print |
| 20:11.36 | ngb | bjohnson: i download ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.5.tar.gz it |
| 20:11.51 | ngb | bjohnson and i get the last error :( |
| 20:11.57 | ngb | de version is correctly |
| 20:12.01 | netsurfer | wankel - I think this is why I ended up upgrading the production box to 2.4.29 kernel |
| 20:12.04 | ctooley | This look like the only places that the default install of asterisk puts things? /etc/asterisk /usr/lib/asterisk /var/lib/asterisk /var/run/asterisk /var/spool/asterisk /var/log/asterisk |
| 20:12.07 | egrat | well, right now i pay $29.95/mo for unlimited local and long distance from a CLEC and i use way more than 2000 minutes a month. it has to be costing them. most of the people I know who have switched to these plans use similar amounts of minutes in a month. |
| 20:12.21 | wankel | netsurfer: i finally just upgraded the box to red hat and now it's happy :) |
| 20:12.23 | ctooley | Note that I changed the /var/run stuff to /var/run/asteisk |
| 20:12.46 | netsurfer | wankel - as much as im pissed with debian for making this faux pas i am still die hard debian at heart |
| 20:13.03 | wankel | yeah, whatever you're used to is best |
| 20:13.17 | netsurfer | true |
| 20:13.17 | shido6 | egrat |
| 20:13.27 | wankel | i've been doing red hat since i started migrating off solaris, so it's easiest for me to work with. |
| 20:13.30 | lohelle | how many here is running bristuff? Next question is: how many is running zaphfc with NT-mode? |
| 20:13.53 | shido6 | for those users that DONT spend hours on the phone ...... if u can get enough customers who DONT "abuse" the unlimited plan you can turn a profit |
| 20:13.58 | bjohnson | Qwell: few specify the hard number .. most say something to the effect of .. unlimited until we decide you're using too much |
| 20:14.08 | shido6 | you are going to get users who place a call on Monday Jan 1st |
| 20:14.15 | shido6 | and try to keep it up |
| 20:14.19 | shido6 | until December 30th |
| 20:14.27 | Qwell | bjohnson: I was just giving an example. Surely most places have the number, whether they publish that number though, is a different story |
| 20:14.30 | shido6 | I used to be that user :) |
| 20:14.36 | bjohnson | Qwell: worst one I read actually stipulated that they would charge at their highest advertised rate in the last 6 months if they decided you were abusing the priveliege |
| 20:14.46 | Qwell | nice |
| 20:16.10 | egrat | well, also keep in mind that creating lengthy legal documents explaining the definitions and precise intentions of "unlimited" service is kind of a slap in the face to the consumer. We want to breed customer satisfaction and that can't happen if they don't trust us because we have all these legal dislaimers of which they only understand a fraction. |
| 20:16.21 | bjohnson | egrat: residential or commercial? They don't offer unlimited commercial LD in my area |
| 20:16.26 | egrat | both |
| 20:16.51 | bjohnson | $30 must be for residential |
| 20:17.01 | egrat | my company has 4 lines on unlimted local and long distance. we used to pay over $1,000 a month in telephone bills. now we pay about $150/mo.. |
| 20:17.04 | bjohnson | 2000 minutes a month is nuts |
| 20:17.20 | florz | "We sell you unlimited supply. But if you ask us to deliver too much, we will fine/sue/whatever you." ... that sounds logical :-) |
| 20:17.23 | bjohnson | we pay that just for local usage |
| 20:17.34 | bjohnson | florz: exactly |
| 20:17.58 | shido6 | wow |
| 20:18.11 | bjohnson | similar to unlimited internet connection (dsl and cable) .. we just want to limit how much you actually USE it |
| 20:18.25 | vaewyn | unlimited "i don't think that word means what you think it means" |
| 20:18.27 | vaewyn | :P |
| 20:18.28 | bjohnson | unlimited voip .. just don't use it :) |
| 20:19.03 | vaewyn | unlimited... within limts |
| 20:19.06 | vaewyn | limits even |
| 20:19.09 | florz | After all, it's pretty much the same as "We sell you $ARTICLE for n $CURRENCY. But if you really wanna have $ARTICLE, you'll have to pay extra!" |
| 20:19.26 | bjohnson | handling |
| 20:19.28 | egrat | now, on these PRI services, how do you get long distance calling? do you pay some long distance provider for it, or do you order the services with it bundled in? |
| 20:19.34 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 20:19.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:19.45 | wankel | depends on who you buy the PRI from |
| 20:19.47 | vaewyn | "unlimited VoIP" == "Punch the monkey for a free ipod" |
| 20:19.54 | bjohnson | hehe |
| 20:19.55 | Nugget | haha |
| 20:20.05 | Nugget | vaewyn is wise |
| 20:20.10 | vaewyn | ass |
| 20:20.11 | vaewyn | :} |
| 20:20.13 | vaewyn | + |
| 20:20.16 | Nugget | heh |
| 20:21.03 | wankel | egreat: generally you pay a local loop (possibly just a cross-connect) and a port charge. then you have all sorts of ways to pay for usage. sometimes incoming is free, sometimes you pay. sometimes it's all domestic at the same rate, sometimes local is cheaper |
| 20:22.10 | egrat | and is it possible to do some sort of area code matching in Asterisk so that, say, you could have a PRI with numbers for city A and another with numbers for city b and they each have their own local calling area and when someone makes a call to that area, the server puts them on the PRI that has that area as a local calling area? |
| 20:22.14 | wankel | if you're getting a really good deal, it's usually got a very complex set of pricing assocated with various lata and bands. you may be able to get a good deal on a single blended rate, though. |
| 20:22.35 | Qwell | egrat: sure |
| 20:22.41 | wankel | yes, you can route calls based on the number. |
| 20:23.00 | shido6 | u can do all of that in the dialplan |
| 20:23.22 | egrat | if we can do that, i can see how we can make money on it. |
| 20:23.28 | Slainte | how do I change teh volume of the on hold? |
| 20:23.40 | Qwell | Slainte: look at musiconhold.conf |
| 20:23.41 | tzanger | shido6: does nufone have trunk=yes in their type=user entries in iax.conf? |
| 20:23.46 | Qwell | The default is "quiet", I believe. |
| 20:23.50 | shido6 | if you ask us to turn it on |
| 20:23.51 | shido6 | we will |
| 20:23.55 | tzanger | shido6: ahh |
| 20:23.58 | shido6 | I can turn it on now for your account if you like |
| 20:24.00 | tzanger | not yet :-) |
| 20:24.08 | shido6 | if its turned on on one end |
| 20:24.11 | tzanger | oh wait you don't have any jitter buffer on anyway |
| 20:24.12 | shido6 | u need to turn it on on the other |
| 20:24.24 | tzanger | shido6: actually the discussion on -dev says otherwise |
| 20:24.35 | harryvv | anyone know any voip to pstn carriers for the vancouver bc area? |
| 20:24.38 | tzanger | I can send you trunked frames and you'll acepet them even with trunk=no but you won't send back trunked frames |
| 20:24.38 | shido6 | not at 18gb/sec and the way we're peered on our hardware we're at 1% of our capactiy |
| 20:24.52 | shido6 | well |
| 20:24.56 | shido6 | our boxes will spit out errors |
| 20:25.03 | tzanger | so if I have two concurrnet calls to you you get 1 packet from me but I get 2 back from you |
| 20:25.10 | Slainte | Qwell, users still say it is too loud |
| 20:25.18 | shido6 | but the call should go through unless u have hacked your box to use an unreliable timing source |
| 20:25.22 | tzanger | yeah |
| 20:25.26 | tzanger | calls go through just fine |
| 20:25.48 | tzanger | actually yes can you turn on trunking for our calls please? I'll /msg you the user acct |
| 20:27.06 | *** join/#asterisk Liquide (liquide@liquide.user) |
| 20:28.02 | ngb | some one have problems for compile asteriks 1.0.5 |
| 20:28.03 | ngb | ? |
| 20:28.20 | wankel | just you |
| 20:28.37 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 20:29.14 | ngb | wankel are u running asterisk 1.0.5 ? |
| 20:29.28 | Nivex | the wiki makes references to chan_sccp, but the stock asterisk load has chan_skinny. Does chan_skinny work? |
| 20:29.37 | *** join/#asterisk kingtaco|laptop (~kingtaco@kingtaco.developer.gentoo) |
| 20:30.46 | harryvv | yes its for cisco |
| 20:30.50 | harryvv | need to disable it |
| 20:31.11 | Nukemizer | Are zaptel FXO cards prone to providing chopy calls when using a softphone to make a call ? |
| 20:31.13 | ngb | some one are running asterisk 1.0.5 ? |
| 20:31.16 | harryvv | if not using cisoc phones |
| 20:31.50 | *** join/#asterisk zpn (~xpn@dhcp-166.digium.com) |
| 20:31.54 | wankel | ngb: i'm running the latest cvs most of the time. i have run 1.0.5, but i have X11 libs installed to build against. |
| 20:32.40 | wankel | if you get a pre-build version of it, it should work if you don't try to run the GTK console |
| 20:32.58 | egrat | so, if i have a dual proc p4 3.06 GHz with 4GB RAM, how many calls do you think that would hold using asterisk? |
| 20:33.10 | ngb | fucking x11!!! i heat it! |
| 20:33.22 | modulus_ | i cool it. |
| 20:33.51 | wankel | i'm not very fond of X11. however, i learned a long time ago that allowing my technical bias to get in the way of getting work done is sorta dumb. |
| 20:33.59 | Nugget | X11 is pretty awful. |
| 20:34.20 | eKo1 | What's wrong with X11? |
| 20:34.22 | wankel | it was pretty cool in the late 80s :) |
| 20:34.23 | modulus_ | xorg is good |
| 20:34.31 | *** join/#asterisk psywar (psywar@rasterburn.org) |
| 20:34.45 | modulus_ | xorg worked suprprisingly well on first install |
| 20:34.57 | vaewyn | X11 does what it was meant to do VERY well... and does what people have shoehorned it to do horribly :} |
| 20:35.13 | Nugget | X11 is awful independent of any particular implementation. |
| 20:35.17 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 20:35.32 | Nugget | although some implementations are awfuller than others. |
| 20:35.44 | *** join/#asterisk riksta (~rick@81-178-168-162.dsl.pipex.com) |
| 20:35.46 | wankel | eko1: well, anything that only has basic 2D primitives and not a full set of common widgets will tend to encourage people to build a lot of inconsistent, buggy user interfaces. |
| 20:36.05 | Slainte | yeah even the quietmp3 is too loud for some people. any ideas how I can manually lower the volume? |
| 20:36.06 | Nugget | it's inadequate to support modern user interface design. |
| 20:36.13 | wankel | it's wonderful that you can create your own UI. it's terrible that everyone else can ALSO create their own ui. |
| 20:36.19 | wankel | because everyone else sucks at it |
| 20:36.42 | vaewyn | Nugget: that's overexagerated |
| 20:36.51 | Nukemizer | does anyone here have experience usning Digium FXO Cards and the call quality they have ? or is call quality an issue for Softphones not the Analog CO ? |
| 20:36.57 | Nugget | look at all the hilariously bad ways people have faked transparent windows. |
| 20:37.30 | AgiNamu | egrat: depends on what you're doing |
| 20:37.32 | vaewyn | that's in the newer X11 implementations |
| 20:37.33 | wankel | are transparent windows necessary for modern UI design? i thought they were a stupid fad that made the screen difficult to read. |
| 20:37.45 | AgiNamu | egrat: If you're transcoding G729 to a PRI, maybe um... 90? |
| 20:37.47 | vaewyn | wankel: agreed |
| 20:37.53 | *** join/#asterisk KalD|Work (~KalD@proxy.corp.telesym.com) |
| 20:38.00 | Nugget | some transparency is useful. useful and necessary aren't the same thing. |
| 20:38.05 | AgiNamu | if you're just handling registration aond stuff... a ton |
| 20:38.16 | Nugget | transparency is overused by the screenshot brigade, though. I'll grant you that. |
| 20:38.27 | AgiNamu | Have you guys seen Avalon? |
| 20:38.33 | wankel | agi: what about running SIP to SIP? i haven't seen any good numbers on that yet. |
| 20:38.56 | AgiNamu | if the clients redirect themselves? i cant see why you couldnt hold a ton |
| 20:39.02 | wankel | or just SIP to IAX2. i have external hardware for doing all the PSTN interfacing and transcoding. |
| 20:39.03 | AgiNamu | but i despise sip and dont use it |
| 20:39.23 | AgiNamu | i dont know... |
| 20:39.28 | wankel | well, i get SIP from the external hardware. then i either talk ot phones or other PBXs via IAX2 or SIP. |
| 20:39.35 | AgiNamu | NT 6 has a powerful GUI. full desktop compositing engine |
| 20:39.37 | zpn | i'm having problems compiling zaptel. get this error :: configure: error: termcap support not found |
| 20:39.38 | zpn | make: *** [editline/libedit.a] Error 1 |
| 20:40.11 | wankel | agi: yeah, i have no idea either. once asterisk crashes or the audio starts to drop out, i guess i'll know what the box can handle :) |
| 20:41.00 | zpn | actually, its when i compile asterisk |
| 20:41.01 | zpn | sorry |
| 20:41.03 | AgiNamu | these are numbers digium should be putting up |
| 20:41.08 | AgiNamu | astertest.com shows a lot of stuff too |
| 20:41.18 | AgiNamu | like 300+ calls on a $300 machne |
| 20:43.04 | wankel | so far all i see there is announcements that they have accomplished something or other. no real data or configs. |
| 20:43.21 | bjohnson | AgiNamu: 790 |
| 20:43.24 | *** join/#asterisk keith778 (kobrien@216.sub-70-213-14.myvzw.com) |
| 20:43.28 | djin | well zpn, install termcap-devel |
| 20:43.41 | AgiNamu | in other words... a lot for one PC |
| 20:43.45 | AgiNamu | not a lot for 1U of space... |
| 20:43.48 | bjohnson | AgiNamu: although I guess that was one way |
| 20:44.30 | bjohnson | AgiNamu: I think for most small deployments, internet connection will be biggest limitation on # of concurrent calls |
| 20:44.38 | AgiNamu | hell ya |
| 20:44.39 | ariel_ | AgiNamu, if the system $300.00 system using ulaw no transcode would still be a something to see. |
| 20:44.55 | AgiNamu | yea, that's awesome |
| 20:45.16 | AgiNamu | Even so.... Asterisk has to get up a lot more |
| 20:45.20 | wankel | it SHOULD handle a metric assload of calls if it's just acting as a media gateway and SIP registry. i've just seen really low estimates of how many calls a system can handle on the wiki and mailing lists. for now i'm just assuming that those people are idiots. |
| 20:45.31 | AgiNamu | I'd think that 1000ports/U is a minium |
| 20:46.13 | AgiNamu | i think asterisk is going to be limited until there's bigger hardware |
| 20:46.15 | ariel_ | The most we put into a system was about 90 calls of ulaw via a 10mg link. It worked on a P4 running sip and only ulaw. But I do not think we could put more into that one box. |
| 20:46.26 | AgiNamu | going around at trade shows, everyone bitchs about the low densiuty of asterisk |
| 20:46.53 | AgiNamu | ariel_ that's because you're maxing out the ethernet |
| 20:47.09 | wankel | yeah, easily. |
| 20:47.27 | AgiNamu | 90ulaw calls is almost 8mbps |
| 20:47.30 | AgiNamu | *full duplex* |
| 20:47.31 | vaewyn | Compared to our meridian * servers with channel banks to support them are WAY smaller still :P |
| 20:48.04 | AgiNamu | vaewyn... channel banks? :P |
| 20:48.10 | ariel_ | vaewyn, with a channel bank there is no problem in having 8 t1 in an asterisk box. We did that for some time last year. |
| 20:48.28 | AgiNamu | what about getting a DS3 into an asterisk box |
| 20:48.37 | vaewyn | AgiNamu: I'm saying that the available port density is relative to the application |
| 20:48.49 | AgiNamu | yea... im talking more higher end |
| 20:49.12 | ariel_ | Higher end I would put a Lucent TNT in between serveral asterisk boxes. |
| 20:49.13 | wankel | put the DS3 into a TNT and run SIP into the * box |
| 20:49.24 | vaewyn | Well.. replacing a Meridian Option11e is pretty high end in my book :P |
| 20:49.30 | AgiNamu | yea, everyone seems to be doing that... or at least recommending it |
| 20:49.45 | wankel | doing DSP on a PC is just silly |
| 20:49.49 | wankel | that's what hardware is for |
| 20:49.54 | vaewyn | wankel: agreed |
| 20:49.55 | ariel_ | I have replaced 3 Nortel's and one Merlin so far. No problems. |
| 20:50.09 | AgiNamu | someone mentioned a DSP+Quadspan PRI |
| 20:50.12 | vaewyn | ariel_: anything 11c/11e sized? |
| 20:50.14 | AgiNamu | and 8-span PRI |
| 20:50.30 | ariel_ | no not that models. |
| 20:51.03 | ariel_ | Anyone going to the Telephony conference & Expo this year? |
| 20:51.08 | AgiNamu | i know our clec spent $1million on some telica thing to put in in front of asterisk |
| 20:51.34 | vaewyn | ariel_: which conference? |
| 20:51.44 | vaewyn | ariel_: (that title fits many :} ) |
| 20:51.45 | wwalker_ | anyone know an aggregator that can turn on DID's dynamically? |
| 20:51.45 | AgiNamu | ariel_ is that IPTEL? |
| 20:51.53 | *** join/#asterisk christo (~chris@212.18.226.160) |
| 20:52.00 | wankel | if you buy a telica i dunno why you'd bother with asterisk |
| 20:52.08 | wwalker_ | We're getting stories like 15 to 30 days for a DID |
| 20:52.10 | wankel | a softswitch should be able to do everything |
| 20:52.12 | ariel_ | <PROTECTED> |
| 20:52.18 | vaewyn | ahh |
| 20:52.18 | AgiNamu | yea i dunno. im happy, cause i want IAX |
| 20:52.33 | vaewyn | not I... but I will be at VON in San Jose :} weeeeee! |
| 20:53.01 | AgiNamu | Maybe Mark will say "So, I'd like to announce our partnership with <someone with tons of money> and we're now hiring 20 SDEs and 40SDE/Ts as well as experiences project managers" |
| 20:53.13 | ariel_ | sorry actual name is Internet Telephony Conference & Expo. |
| 20:53.26 | AgiNamu | I was at that last year |
| 20:53.26 | ngb | yes people!!! asteriks 1.0.5 require libx11 |
| 20:53.40 | AgiNamu | but i'll be in VON in march |
| 20:53.57 | ariel_ | I can't afford going out of the area. |
| 20:54.10 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 20:54.18 | dontmsgme | Anyone have problems with VP? |
| 20:54.18 | blitzrage | ngb: huh? |
| 20:54.31 | dontmsgme | "NOTICE (2/7/2005 2PM EST): One of our underlying providers for outgoing calls is currently experiencing a problem." |
| 20:54.37 | vaewyn | I'm dragging the 2 "old-sk00l" telco maniacs with me to VON and I'm taking a shotgun to keep them away from the Nortel booth |
| 20:54.46 | wankel | blitzrage: he's been bitching and moaning for hours about how he can't build 1.0.5 on his slackware box that has no x11 libs. |
| 20:54.57 | blitzrage | wankel: thats stupid. Asterisk doesn't need X11 |
| 20:54.58 | *** join/#asterisk keith778 (kobrien@216.sub-70-213-14.myvzw.com) |
| 20:55.09 | wankel | blitzrage: the gtk console links to it |
| 20:55.20 | blitzrage | wankel: oh well then yes, if you are using the GTK console, yes :) |
| 20:55.32 | *** join/#asterisk keith778 (kobrien@216.sub-70-213-14.myvzw.com) |
| 20:55.32 | blitzrage | wankel: but you don't *need* X11 at ALL |
| 20:55.38 | wankel | apparently he can't figure out how to edit the makefile so it doesn't build the gtk console |
| 20:55.55 | AgiNamu | why doesnt he just drop libx11 on it? |
| 20:55.56 | wankel | i don't have a box without X11 libs to figure it out and pastebin a diff. |
| 20:56.04 | wankel | agi: he HATES x11 |
| 20:56.13 | wankel | hates hates hates. guis are evil! |
| 20:56.16 | AgiNamu | so fucking what... having a file on his disk isnt going to destroy the world. |
| 20:56.18 | wankel | or something like that. |
| 20:56.24 | wankel | yeah, that's about what i figure. |
| 20:56.39 | AgiNamu | YES PEOPLE!!!! asterisks require libwin32 to build |
| 20:56.47 | blitzrage | CentOS-server install minimal. yum install gcc, yum install bison, yum install openssl-devel, yum install libnewt-devel |
| 20:56.56 | blitzrage | that is all you *need* |
| 20:57.09 | Nugget | asterisk has troubles building if you don't have the x11 libs but you do have some gtk libs. |
| 20:57.14 | Nugget | I ran into that with slackware. |
| 20:57.18 | AgiNamu | libnewt? sounds like gecko which has a G like GUI... |
| 20:57.27 | Nugget | even if you don't install x11 it still pollutes the machine with some wonky gtk library |
| 20:57.31 | blitzrage | Libnewt is only if you need zttool |
| 20:57.51 | wankel | it would be nice if the makefile didn't build the gtk console by default. |
| 20:57.55 | Nugget | the asterisk makefile wrongly uses the presence of that gtk library to detect if x11 were on there. |
| 20:57.57 | blitzrage | Nugget: I've never seen Asterisk complain about X11 stuff... but I've never isntalled it on slack either |
| 20:58.05 | AgiNamu | whats so cool about the GTK console anyways? |
| 20:58.13 | AgiNamu | sorta just looked like a... console. |
| 20:58.17 | wankel | nugget: ah, is that the problem? |
| 20:58.20 | Nugget | yeah |
| 20:58.34 | mgeorge | Is there a way from the CLI to hangup an IAX channel? |
| 20:58.36 | wankel | i'd go fix that, but i don't have a slackware box with no X11 libs to test on. |
| 20:58.44 | eKo1 | mgeorge: soft hangup |
| 20:58.45 | Nugget | although there's some sanity -- no sane system would have that gtk library without having x11. it's useless otherwise. |
| 20:58.55 | Nugget | I blame slackware, not asterisk. |
| 20:59.00 | wankel | yeah, why does it have gtk without x? |
| 20:59.06 | Nugget | *shrug* |
| 20:59.29 | *** join/#asterisk ennuyeux7 (~ennuyeux7@83.146.53.34) |
| 20:59.35 | wankel | some people actually use the c++ toolkit outside of X, i suppose. ick. |
| 20:59.36 | mgeorge | How do I identify the channel? I gave it the IAX2/id@sys and it said no chanel... |
| 20:59.46 | Nugget | you can either install the x11 libs or comment the gtkconsole stuff out of the makefile. |
| 20:59.49 | eKo1 | mgeorge: show channels |
| 20:59.52 | *** join/#asterisk buleeahn (~asanders@66-141-61-2.ded.swbell.net) |
| 21:00.07 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 21:00.09 | wankel | nugget: i told him to edit the makefile but apparently he'd rather keep repeating himself and cursing X11. |
| 21:00.24 | Nugget | heh |
| 21:00.33 | AgiNamu | yea, its easier |
| 21:00.41 | AgiNamu | doesnt get anything working. but it's easier. |
| 21:00.54 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.116) |
| 21:01.14 | mgeorge | eKol: that did it. Thanks! |
| 21:01.39 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.rr.com) |
| 21:02.00 | jgaviria | hi, i need to connect a E1 card, but the local telco doesnt support RDSI, then what kind of signaling can i use? |
| 21:02.08 | |Vulture| | anyone have an IP-500 in front of them? I am trying to write a howto for one of my offices, and I just need to write the part on Transfers/BlindXfers |
| 21:02.10 | ngb | <PROTECTED> |
| 21:02.38 | |Vulture| | I think it is, place call on hold, hit transfer, type in extension, talk to person, hit transfer again for regular Transfer |
| 21:02.57 | |Vulture| | and Blind Xfer is, place on hold, hit Blind Xfer, dial extension |
| 21:02.59 | shido6 | ccs,hdb3,crc4 |
| 21:03.22 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 21:03.33 | dontmsgme | Is anyone having trouble with VoiceP or just me? |
| 21:03.54 | shido6 | euroisdn |
| 21:03.54 | ariel_ | dontmsgme, I just made a 800 call via vpc no problem. |
| 21:04.03 | Nugget | everyone should take a moment out of their otherwise lazy and boring monday and add e164.org lookups to their dialplan. |
| 21:04.10 | |Vulture| | dontmsgme: what kinda problem? |
| 21:04.21 | ariel_ | Nugget, why? |
| 21:04.23 | Nugget | it's good karma. |
| 21:04.31 | Nugget | and it will save you money |
| 21:04.32 | ariel_ | krama? |
| 21:04.34 | dontmsgme | When I login in first of allit says there is a problem in their service which might affect outbound calls |
| 21:04.42 | dontmsgme | And when I dial it I just get asterisk telling me Nine Six Nine Sx |
| 21:04.42 | |Vulture| | dontmsgme: working fine for me for 1800 calls |
| 21:04.52 | hardwire | anybody have any experience sending to POCSAG transmitters |
| 21:05.00 | hardwire | add a ? to the end of that |
| 21:05.40 | *** join/#asterisk Beave (~beave@vistech.org) |
| 21:06.14 | *** join/#asterisk Raj_ (raj@linux3.tennistown.com) |
| 21:06.38 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 21:06.56 | bjohnson | Nugget: that's for PRI systems only right? |
| 21:07.05 | *** join/#asterisk Tough_Nuts (~Tough_Nut@204.110.228.254) |
| 21:07.10 | Raj_ | Hey. Got a question regarding the VoicePulse service... I want to setup a VoicePulse Connect account so that we can use SIP phones, I know that incoming calls are free, but how can I get outgoing calls for free too? |
| 21:07.18 | Nugget | bjohnson: no, it's for anyone. |
| 21:07.38 | Nugget | Raj_: you can't. |
| 21:07.43 | bjohnson | any advantage for small users? I don't understand the purpose behind it |
| 21:07.58 | bjohnson | Raj_: tap into your neighbour's line |
| 21:08.06 | Raj_ | Nugget: so I would have to use an analog phone if i wanted free outgoing calls & the Business Unlimited account? |
| 21:08.15 | Raj_ | bjohnson: hahahah yah, i was thinkin of that |
| 21:08.18 | Nugget | the purpose of e164.org is to allow your asterisk server to automagically know when a number you've dialed can instead be reached directly, for free, via SIP or IAX2. |
| 21:08.33 | Luke-Jr | Anyone here use MyPhoneCompany? |
| 21:08.41 | Nugget | so if you call one of the 100,000+ numbers in the database, you'll pay no toll charges and you'll receive better, more direct quality. |
| 21:08.43 | bjohnson | ahh .. I see |
| 21:08.57 | Nugget | and, similarly, when people call you they can experience the same benefit |
| 21:09.03 | |Vulture| | Raj_: wow what kinda question is that? |
| 21:09.23 | bjohnson | I suppose I should look into that .. might not mean much to me now .. but if done, I don't have to think about it |
| 21:09.31 | Nugget | exactly. :) |
| 21:09.41 | Nugget | it really only takes a few minutes to get going. |
| 21:09.49 | Nugget | and the more people that use it, the more useful it becomes. |
| 21:10.11 | Raj_ | |Vulture|: well i was under the impression that I could make outgoing calls for free with my SIP phones, but apparently not |
| 21:10.16 | ngb | bjohnson: asterisk 1.0.5 require libx11 |
| 21:10.23 | bjohnson | Raj_: use FWD |
| 21:10.26 | Nugget | Raj_: if you call someone over the internet, sure, it's free. |
| 21:10.28 | bjohnson | wow |
| 21:10.28 | |Vulture| | Raj_: you can to other IP phones |
| 21:10.33 | Nugget | but if you're calling a phone number, you'll pay. |
| 21:10.45 | Nugget | (unless they use e164.org :) |
| 21:10.49 | Raj_ | ohhhhhhh i see |
| 21:10.50 | bjohnson | ha |
| 21:10.54 | ngb | bjohnson: the developers of asterisk not include this information in the README file |
| 21:11.05 | ngb | :/ |
| 21:11.08 | bjohnson | no .. we should shave theor heads |
| 21:11.12 | bjohnson | their |
| 21:11.15 | |Vulture| | someone has to have an IP500 in front of them... |
| 21:11.33 | bjohnson | if so, package it up and send it to me |
| 21:11.38 | bjohnson | haha |
| 21:11.53 | Raj_ | Polycomm? i was looking at purchasing those or possibly some WiFi phones |
| 21:12.05 | |Vulture| | yea I am trying to write an office manual for them |
| 21:12.06 | *** join/#asterisk ACiDV (~joel@69.156.197.246) |
| 21:12.07 | Nugget | do NOT buy the zyxel/pulver wifi phones. |
| 21:12.09 | vaewyn | Polycom are nice phones |
| 21:12.10 | Nugget | they suck. |
| 21:12.16 | bjohnson | whaa? The free phoner wants to shell out for wifi phones? |
| 21:12.19 | |Vulture| | we have 7960s and IP500s in the offices |
| 21:12.20 | Nugget | heh |
| 21:12.28 | |Vulture| | I got a 7960 in front of me but no 500s |
| 21:12.33 | ACiDV | WIP5000 are better wifi phone that Zyxel |
| 21:12.33 | Raj_ | Nugget: thanks for the heads up, i saw the zyxel phone |
| 21:12.37 | Nugget | it's awful. |
| 21:12.40 | Nugget | truly, truly awful. |
| 21:12.51 | Nugget | I love my 7960, but I dunno if they're worth what they cost. |
| 21:13.03 | vaewyn | Nugget: I have a Hitachi Cable IP-5000 on order... we will see how well those wifi ones work :} (they are different hardware... unlink zyxel/wisip) |
| 21:13.12 | |Vulture| | Nugget: I want to try a IP600 |
| 21:13.15 | shido6 | about $350 each |
| 21:13.15 | Nugget | oh, cool. I've been waiting to hear from someone who bought one. |
| 21:13.17 | shido6 | or less |
| 21:13.29 | Nugget | there's that other wifi phone out now. a senoa or something like that. |
| 21:13.47 | Raj_ | ACiDV: Yeah i like the look of the WIP5000s |
| 21:13.47 | vaewyn | yeah.. senao... i want to try one of those... also go under "Engenius Tech" |
| 21:14.10 | hardwire | snoms new phone looks perty |
| 21:14.11 | ACiDV | Raj_ and they work great, I currently have about 10 of theses in my office (in Canada) |
| 21:14.35 | vaewyn | heck... IP300s are 130$... you just can't beat the $$/feature :P |
| 21:14.35 | Raj_ | really? awesome |
| 21:14.48 | Raj_ | what kind of WiFi router are you using? |
| 21:15.28 | Raj_ | the UTStarcom F1000 looks amazing. its the size of a cell phone |
| 21:15.33 | ACiDV | Raj_ I've try with Netgear and Dlink router... but it's currently only for testing |
| 21:15.50 | vaewyn | remember though... with wifi phones... size==battery==talk time |
| 21:16.08 | Raj_ | vaewyn: true... the spec sheet says talk time of approx 3hrs |
| 21:16.22 | harryvv | who knows of a cheap voipjet like service here in bc canada? want to create a voip to pstn account for my self. |
| 21:21.37 | bjohnson | harryvv: use voipjet or aleph-com.net |
| 21:22.07 | *** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com) |
| 21:22.37 | Raj_ | Is there any VoIP provider that gives us a flat rate when making domestic calls and supports an Asterisk backend? |
| 21:22.44 | harryvv | voipjet said thay dont want to provice inter phone connections within the country because of tax reasons. only called between usa and canada. |
| 21:23.05 | bjohnson | Nugget: that e164 conf examples look bad |
| 21:23.15 | Nugget | yeah, I hacked mine up a bit from the sample. |
| 21:23.18 | *** join/#asterisk empty- (empty@beetle.ispnet.ca) |
| 21:23.20 | Nugget | it's really just two lines |
| 21:23.40 | bjohnson | exten => _.,1,Goto(mainmenu,s,1)? |
| 21:23.47 | Nugget | it's not even close to being complicated enough to make me happy. |
| 21:23.49 | Nugget | :) |
| 21:23.56 | empty- | has anyone got a sip chanel to work with primus (their lingo service) |
| 21:24.05 | empty- | my head is going to explode. |
| 21:24.13 | Nugget | http://slacker.com/~nugget/stuff/extensions.conf <-- [macro-enumdial] is what I'm using |
| 21:24.13 | AgiNamu | isnt primus evil? |
| 21:24.19 | empty- | yes. |
| 21:24.21 | empty- | primus is evil. |
| 21:24.49 | bjohnson | harryvv: good for you for being so ethical |
| 21:25.02 | bjohnson | harryvv: check the aleph one .. they are in AB |
| 21:25.11 | bjohnson | err Manitoba I think |
| 21:25.24 | dca[laptop] | from what part of the sip packet does asterisk get it's user/peer from? i.e. the via, the from, the record-route.... |
| 21:25.24 | empty- | I am setup as a wholesaler. I talked to someone there, they said it has been done. |
| 21:25.32 | AgiNamu | _NX.,1,Goto(1${EXTEN}) <-- that'd work to add a "1" to incoming calls? (my provider doesnt prefix country code) |
| 21:25.41 | AgiNamu | so my miami number comes in as 305-xxx-xxxx |
| 21:25.41 | Nugget | yes. |
| 21:25.41 | bjohnson | the hardware they provide I thought was MGCP |
| 21:25.52 | empty- | primus? I know primus canada is. |
| 21:25.59 | empty- | this is their lingo service, which is sip. |
| 21:26.06 | bjohnson | oh |
| 21:26.09 | Slainte | anyone know how to get the call park to work for a POlycom IP600. * is not acting on the # press in the missle of a call |
| 21:26.18 | Slainte | middle |
| 21:26.30 | AgiNamu | is goto gonna drop anything? |
| 21:26.33 | AgiNamu | or change anything? |
| 21:27.02 | AgiNamu | [provider-incoming] -- _NX.,1,Goto(1{$EXTEN}) -- include => my_extensions |
| 21:27.10 | shido6 | what in the hell is that |
| 21:27.12 | mgeorge | I have an * box and it doesn't seem to take down calls right when they come in on iax2 |
| 21:27.23 | *** join/#asterisk guest3993939393 (~jan@bender.linugen.com) |
| 21:27.32 | Nugget | "take down"? |
| 21:27.33 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 21:27.38 | guest3993939393 | hi |
| 21:27.42 | shido6 | mgeorge, what do you have there ? an inbound IAX number? |
| 21:27.46 | mgeorge | It will bridge them fine, but when the incoming call hangs up the channel stays there. |
| 21:27.54 | Nugget | do you just mean it's not logging them in the console? |
| 21:28.11 | guest3993939393 | I'm having troubz with receiving FAX'es, anyone care to help this poor spa? |
| 21:28.18 | mgeorge | Yes, we ahve a VOIP provider who'se sending us iax2 calls, but when an incoming call hangs up , our box doesn't seem to handle it |
| 21:28.21 | guest3993939393 | sap |
| 21:28.32 | mgeorge | some times it does, sometimes it doesn't. |
| 21:28.52 | *** join/#asterisk philz (~philzama@borg.zamigo.net) |
| 21:28.55 | decentnick | much better |
| 21:29.01 | bjohnson | AgiNamu: goto needs the prioirity I think |
| 21:29.16 | bjohnson | _NX.,1,Goto(1{$EXTEN},1) |
| 21:29.19 | AgiNamu | yea |
| 21:29.19 | psywar | okay so I've got * sitting in-line with my apartment wiring. I am still using analog to dial out. Where do I get started with VoIP that leaves my LAN? Are there test numbers I can call free? |
| 21:29.24 | bjohnson | haven't tried it that way |
| 21:29.45 | bjohnson | psywar: FWD |
| 21:29.53 | psywar | bjohnson: ok ty |
| 21:29.59 | Raj_ | bjohnson: what is FWD? |
| 21:30.02 | bjohnson | psywar: is limited to other FWD users |
| 21:30.05 | psywar | free world dialup |
| 21:30.08 | Raj_ | ohh |
| 21:30.14 | mgeorge | shido6: is there something I need to set so that the iax disconect is handled better? |
| 21:30.21 | Nugget | Raj_: it's not what you're hoping it is. :) |
| 21:30.29 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 21:30.31 | bjohnson | psywar: they have some special extensions to read the time, etc so you can test your connection |
| 21:30.32 | decentnick | I'm using asterisk 1.0.5, zaptel 1.0.4 and spandsp 0.0.2 pre 10 and libtiff 3.5.7-7 -> my FAX'es render good (as in readable) but only 5% shows up :( any assistance please? |
| 21:30.33 | AgiNamu | any general hints on large .confg files? |
| 21:30.44 | Raj_ | Nugget: hehehehe |
| 21:30.50 | psywar | what protocol does FWD speak? |
| 21:30.56 | Luke-Jr | psywar: SIP |
| 21:30.58 | AgiNamu | im wondering what sizes you've people got .conf files up to |
| 21:31.02 | AgiNamu | and IAX no? |
| 21:31.03 | philz | SIP or IAX |
| 21:31.04 | Luke-Jr | When was Asterisk 1.0.5 released? |
| 21:31.06 | bjohnson | psywar: they started another service where you and others share your telco connections |
| 21:31.12 | bjohnson | psywar: sip or iax |
| 21:31.12 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-1056.nb.aliant.net) |
| 21:31.19 | Nugget | my extensions.conf is only 14KB |
| 21:31.26 | AgiNamu | first they called it "bellster". and then got sued by bellsouth |
| 21:31.32 | psywar | haha |
| 21:31.33 | AgiNamu | so they called it 'fwdout' and are waiting for Skype to say something |
| 21:31.34 | psywar | lame |
| 21:31.45 | dca[laptop] | file!!!!!!!!!!!!!!! |
| 21:31.58 | AgiNamu | psywar, it'd be lame if bellster was anything to care about. being seeing as its silly, gimmick, marketing.... |
| 21:32.04 | decentnick | I'm using asterisk 1.0.5, zaptel 1.0.4 and spandsp 0.0.2 pre 10 and libtiff 3.5.7-7 -> my FAX'es render good (as in readable) but only 5% tops shows up :( any assistance please? thank you very much ! |
| 21:32.09 | psywar | what's skype |
| 21:32.16 | AgiNamu | www.skype.com |
| 21:32.24 | denon | skype's a wannabe |
| 21:32.26 | AgiNamu | and their PSTN service is called "SkypeOut" |
| 21:32.37 | AgiNamu | they still dont offer DIDs AFAIK\ |
| 21:32.37 | denon | designed to get dialup and cable lusers interested in voip :) |
| 21:32.38 | *** join/#asterisk zoa (~zoa@213.219.141.7.adslpower.by.edpnet.be) |
| 21:32.44 | psywar | I think bellsouth should have called itself "Southern Bel" |
| 21:32.44 | AgiNamu | nor does DTMF work with them lol |
| 21:32.50 | psywar | *Bell |
| 21:32.59 | philz | LOL |
| 21:33.04 | labo | true, skype is not for business. |
| 21:33.08 | AgiNamu | im pissed at bellsouth. They let Telefonica buy them (in guatemala) |
| 21:33.15 | Nugget | skype offers a decent solution, in many ways superior to the current asterisk offerings for end users, but it's ultimately a technological dead end. |
| 21:33.19 | JunK-Y | any idea why ast_indicate(chan, AST_CONTROL_BUSY); gimme a fast busy? |
| 21:33.28 | decentnick | Feb 7 23:09:14 DEBUG[2794]: Bad rows 77 |
| 21:33.29 | AgiNamu | nugget... huh? |
| 21:33.36 | AgiNamu | if skype got DIDs and some nice hardware |
| 21:33.36 | decentnick | Feb 7 23:09:14 DEBUG[2794]: Longest bad row run 20 |
| 21:33.42 | AgiNamu | they'd seriously kick ass |
| 21:33.47 | decentnick | Feb 7 23:09:14 DEBUG[2794]: Image size (bytes) 0~ |
| 21:33.49 | Nugget | skype's softphone is way nicer than any SIP or IAX client software I've used. |
| 21:34.02 | AgiNamu | firefly seems nice |
| 21:34.06 | AgiNamu | I got FireFly USB phones in |
| 21:34.09 | decentnick | no FAX experts in the house? need real help |
| 21:34.10 | Nugget | and their protocol can fall back on tcp for nat traversal, which is also nifty |
| 21:34.12 | Luke-Jr | When was Asterisk 1.0.5 released? |
| 21:34.15 | philz | sjphone isntbad either |
| 21:34.16 | AgiNamu | I'm selling them, $59 each, min. order 10 pcs |
| 21:34.22 | AgiNamu | Virbiage wants $100 for em |
| 21:34.25 | decentnick | Luke-Jr: maybe 1 or 2 weeks ago |
| 21:34.58 | Luke-Jr | So it should support AUTH w/ MyPhoneCompany, right? |
| 21:35.12 | harryvv | What other recomended domain registeration site is recomended other then networksolutions? |
| 21:35.12 | AgiNamu | what does that mean? |
| 21:35.20 | AgiNamu | harryvv. ... godaddy? |
| 21:35.22 | decentnick | I'm using asterisk 1.0.5, zaptel 1.0.4 and spandsp 0.0.2 pre 10 and libtiff 3.5.7-7 -> my FAX'es render good (as in readable) but only 5% tops shows up :( any assistance please? thank you very much ! |
| 21:35.24 | AgiNamu | netsol/verisign BLOW |
| 21:35.49 | greg_work | harryvv: ANYWHERE but netsol |
| 21:35.50 | Beirdo | harryvv: I use easydns.com |
| 21:35.56 | AgiNamu | godaddy is $8 |
| 21:35.59 | greg_work | i also recommend godaddy |
| 21:36.00 | AgiNamu | and offers SSL for $29 or so |
| 21:36.10 | greg_work | AgiNamu: oh, i didnt know that |
| 21:36.14 | AgiNamu | without any verification |
| 21:36.30 | harryvv | Beirdo how many domain names have you registered with them and what do thay charge? |
| 21:36.31 | DaddySr | Can someone help with my FAX problem please? |
| 21:36.31 | greg_work | you can get geotrust SSL certs from ev1servers.net (referal program) pretty cheap |
| 21:36.35 | greg_work | AgiNamu: thats a BAD thing |
| 21:36.41 | AgiNamu | greg, not really |
| 21:36.52 | AgiNamu | the current ssl / Pki infraestructure is fucked up |
| 21:36.53 | greg_work | is godaddy's CA in any browsers? |
| 21:36.54 | Beirdo | about 4 or 5, and the cost depends on the type of domain |
| 21:36.57 | AgiNamu | it's broken |
| 21:36.58 | AgiNamu | greg, yep |
| 21:37.00 | AgiNamu | it's not godaddy |
| 21:37.03 | AgiNamu | it's ... Starfire |
| 21:37.05 | AgiNamu | or something like that. |
| 21:37.17 | AgiNamu | So, you'd be an idiot to pay $1000 for a verisign cert |
| 21:37.28 | AgiNamu | since a $29 cert from any reseller is just as good |
| 21:37.48 | AgiNamu | AND, if any one of the "trusted roots" is slightly untrsted, it breaks the whole chain |
| 21:38.03 | AgiNamu | so seeing as each browser has a shitload of certs installed... it's pretty damn easy to fake things. |
| 21:38.13 | DaddySr | anyone use spandsp? |
| 21:38.18 | psywar | is FAX supposed to work with voip? I did FAX->SIP->Zap/analog-PSTN-FAX and it bombed out on the first non-text page |
| 21:38.27 | AgiNamu | im pretty sure for under $10,000, you could get a microsoft.com cert issued to you |
| 21:38.47 | AgiNamu | even verisign fucked up and issued fake ms certs |
| 21:38.59 | AgiNamu | psywar, not really |
| 21:39.01 | DaddySr | psywar: FAX work with voip is either FOIP (asterisk does not support FOIP) or |
| 21:39.04 | AgiNamu | you need T.38 for it to work right |
| 21:39.14 | AgiNamu | otherwise, it'll only work with really good conditions |
| 21:39.39 | DaddySr | Can someone help with my FAX problem please? |
| 21:39.39 | psywar | I have to deal with backwards people who actually want FAX instead of email |
| 21:40.00 | AgiNamu | yea, run it via PSTN most likely |
| 21:40.06 | AgiNamu | as asterisk does not have a solid way to do it |
| 21:40.26 | DaddySr | AgiNamu: did you already do it the 'via PSTN way'? |
| 21:40.33 | *** join/#asterisk kant (~bernd@63.245.57.70) |
| 21:40.41 | AgiNamu | I've sent faxes over the PSTN.......... |
| 21:40.51 | DaddySr | received em? |
| 21:40.55 | AgiNamu | that too |
| 21:40.57 | AgiNamu | not with asterisk |
| 21:41.05 | DaddySr | ah, with what? |
| 21:41.06 | ctooley | Does anyone have any experience with LiveVoip? |
| 21:41.09 | AgiNamu | a fax machine... :P |
| 21:41.12 | DaddySr | hehehe |
| 21:41.17 | DaddySr | doih |
| 21:41.21 | AgiNamu | yea, doh :) |
| 21:41.22 | DaddySr | ;p |
| 21:41.30 | AgiNamu | i hear of peopel doing asterisk pstn faxing ok |
| 21:41.32 | AgiNamu | but not sip |
| 21:41.40 | AgiNamu | or ulaw, unless it's awesome conditions |
| 21:41.58 | DaddySr | AgiNamu: I am looking for doing that... asterisk pstn faxing |
| 21:42.17 | DaddySr | but my faxes don't turn up too good |
| 21:42.17 | AgiNamu | maybe in RedHat Asterisk |
| 21:42.33 | AgiNamu | or wait, that's trademarked |
| 21:42.36 | AgiNamu | "RedHat Wildcard" |
| 21:42.38 | AgiNamu | or something |
| 21:42.49 | DaddySr | ? seperate product |
| 21:42.53 | AgiNamu | it's a joke |
| 21:42.58 | DaddySr | ic ;p |
| 21:43.05 | AgiNamu | its me fantasizing about a company taking asterisk and making it into a product |
| 21:43.20 | DaddySr | AgiNamu; look at www.packetbox.net |
| 21:43.35 | AgiNamu | whats that |
| 21:43.42 | DaddySr | what you just said |
| 21:43.44 | AgiNamu | no |
| 21:43.47 | AgiNamu | that's asterisk inside a product |
| 21:43.52 | DaddySr | ic |
| 21:43.53 | AgiNamu | I mean Asterisk AS the product. |
| 21:43.55 | DaddySr | ic |
| 21:43.56 | DaddySr | anyhoo |
| 21:43.59 | *** join/#asterisk JuzzM (~jr@196.41.210.125) |
| 21:44.01 | AgiNamu | i.e., as RedHat took linux |
| 21:44.11 | AgiNamu | tested the shit out of it, has engineers working on it, support contracts, etc. |
| 21:44.21 | DaddySr | you don't know anything that would help me more with asterisk as a pstn fax? |
| 21:44.29 | AgiNamu | probably not |
| 21:44.42 | AgiNamu | you have what? 2 Zap interfaces? |
| 21:44.43 | DaddySr | ok |
| 21:44.47 | DaddySr | yup |
| 21:44.56 | AgiNamu | and it;s doing native slin? |
| 21:44.58 | DaddySr | installed spandsp 0.0.2pre10 |
| 21:45.03 | DaddySr | native slin? |
| 21:45.06 | AgiNamu | slinear |
| 21:45.20 | DaddySr | how can I make sure? |
| 21:45.20 | AgiNamu | i mean, you've got the fax on the FXS and then recieving via FXO? |
| 21:45.28 | AgiNamu | asterisk -vvvvv |
| 21:45.31 | AgiNamu | and check what it says |
| 21:45.33 | AgiNamu | for the codecs |
| 21:46.43 | blitzrage | hey, someone said asterisk-users was too easy to get help on! |
| 21:46.52 | AgiNamu | lol |
| 21:46.56 | AgiNamu | wtf does that mean |
| 21:47.05 | DaddySr | hehe |
| 21:47.08 | blitzrage | means someone got help to a problem too easily |
| 21:47.09 | AgiNamu | that we're supposed to respond to dumbass questions with "google"? |
| 21:47.27 | blitzrage | everyone drop their newbie answer ratio down a couple notches |
| 21:48.02 | DaddySr | not seeing anything |
| 21:48.03 | bjohnson | psywar: some voip providers will handle fax->email for you |
| 21:48.06 | AgiNamu | what's so hard about: Download. build. make config. read. |
| 21:48.09 | DaddySr | but what does a codec have to do with it |
| 21:48.14 | DaddySr | I have alaw installed |
| 21:48.17 | DaddySr | only alaw |
| 21:48.19 | bjohnson | psywar: you could look at using a fax/data/modem switch |
| 21:48.21 | AgiNamu | seems like reading the config files is gonna answer a shitload of stuff with it |
| 21:48.22 | epoch | anyone here have any experience using headsets with polycom phones? |
| 21:48.35 | AgiNamu | the native format for ZAP afaik is SLIN |
| 21:48.43 | AgiNamu | from poking at it with my own code |
| 21:48.51 | AgiNamu | course, id ont understand chan_zap at all |
| 21:48.52 | bjohnson | psywar: or .. you could stick a fax machine on a fxs |
| 21:48.54 | AgiNamu | so i could be utterly wrong |
| 21:49.21 | AgiNamu | well i must go |
| 21:49.25 | AgiNamu | cya |
| 21:49.48 | bjohnson | my logs needed a rest anyway |
| 21:50.17 | bjohnson | e164 intergration is complete |
| 21:50.20 | bjohnson | now to test |
| 21:50.36 | Nugget | hooray |
| 21:50.38 | bjohnson | any victims .. I mean volunteers? |
| 21:50.49 | Nugget | +1 512 538-0508 |
| 21:51.27 | philz | hmm austin tx? |
| 21:51.31 | Nugget | yes |
| 21:51.42 | kingtaco|laptop | guys, I'm running into issues with my TDM400P card |
| 21:51.53 | philz | cool. Victoria Tx here |
| 21:52.12 | kingtaco|laptop | 4/5 calls are fine, the fifth one allways dials some other same number |
| 21:52.24 | kingtaco|laptop | it's a different number for any number I try |
| 21:52.33 | Qwell | it dials a random number? |
| 21:53.02 | kingtaco|laptop | allways the same number, but yes |
| 21:53.31 | kingtaco|laptop | so if I dial xxx1234 it allways goes to a fax machine, but if I dial xxx4321 it goes to someones cell phone |
| 21:53.36 | Qwell | sounds like dialplan funkyness (I'm a newb though, so don't listen to me) |
| 21:53.41 | kingtaco|laptop | it's not |
| 21:53.46 | kingtaco|laptop | it really isnt |
| 21:53.56 | kingtaco|laptop | if I use the 100P card, don't have issues |
| 21:54.03 | kingtaco|laptop | only with the 400 card |
| 21:54.12 | philz | what does your log say when the connection is made? |
| 21:54.14 | bjohnson | Nugget: btw .. Ontario, Canada |
| 21:54.25 | Nugget | brr! |
| 21:54.28 | kingtaco|laptop | it says it's calling the correct number |
| 21:54.53 | philz | hmm. |
| 21:54.57 | *** join/#asterisk fizbar (~jason@dsl-232-31.speedsite.com) |
| 21:55.04 | Qwell | Have you asked the person what number you just called? heh |
| 21:55.18 | Slainte | I sometimes wonder if I was dropped when I was a kid. Today I am certain I was bounced |
| 21:55.21 | philz | put a phone in-line with the fxo and have a listen to the digits |
| 21:55.26 | kingtaco|laptop | I wonder, is there any way to put a delay in asteric, so that it takes the line off hook, then waits(.25 sec) then dials? |
| 21:56.39 | Nugget | kingtaco|laptop: yes. |
| 21:56.41 | jetscreamer | pots can, so hopefully |
| 21:56.43 | bjohnson | not sure .. my SPA 2000 will do that |
| 21:56.57 | philz | I have my own CID question. |
| 21:58.03 | philz | how do you get the initial ring into a fxo to display on a TAPI enabled machine |
| 21:58.45 | Raj_ | gotta go. thanks for the help fellas |
| 21:59.35 | jsolares | who has tried to record calls in windows with usb handsets? |
| 21:59.44 | lohelle | I was thinking about placing an asterisk server between an Siemens Hicom pbx and provider to add functionality.. there are 3 ISDN's from provider.. will 3 ISDN cards in TE-mode (provider=>asterisk) and 3 in NT-mode (asterisk => hicom) do? |
| 22:00.39 | lohelle | and is bristuff stable enough? (zaphfc). Any alternatives? |
| 22:01.16 | bjohnson | got to go |
| 22:03.35 | *** join/#asterisk CoaxD (coax@shell1.cornernet.com) |
| 22:04.07 | CoaxD | Okay. if anyone knows what the hell about a LinkSys WRT54G would break 'qualify = yes', please, by all means, enlighten me |
| 22:04.29 | CoaxD | Feb 7 15:41:57 NOTICE[1119030192]: chan_sip.c:5950 handle_response: Peer 'nicole' is now REACHABLE! |
| 22:04.29 | CoaxD | Feb 7 15:44:59 NOTICE[1119030192]: chan_sip.c:5955 handle_response: Peer 'nicole' is now TOO LAGGED! |
| 22:04.29 | CoaxD | Feb 7 15:45:13 NOTICE[1119030192]: chan_sip.c:7033 sip_poke_noanswer: Peer 'nicole' is now UNREACHABLE! |
| 22:04.35 | CoaxD | (Sorry about the paste.) |
| 22:05.24 | |Vulture| | sounds like lag to me |
| 22:05.38 | |Vulture| | CoaxD: you got QoS turned on? |
| 22:05.40 | labo | its lag. you should put a min or max in the qualify parameter |
| 22:05.52 | |Vulture| | qualify=200 |
| 22:06.05 | CoaxD | I have 'qualify=90' set |
| 22:06.07 | |Vulture| | CoaxD: is this local? |
| 22:06.09 | labo | too low. |
| 22:06.10 | CoaxD | Vulture: Yes |
| 22:06.13 | |Vulture| | damn |
| 22:06.16 | CoaxD | Why on earth is it too low? |
| 22:06.21 | |Vulture| | try 200 |
| 22:06.38 | labo | i put much more, since most of my peers are satellite tho |
| 22:06.39 | |Vulture| | 90 is fast as hell Ive had problems setting qualify too low |
| 22:07.00 | |Vulture| | CoaxD: I use WRT54Gs with Sveasoft firmware |
| 22:07.13 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 22:07.19 | kietlak | lohelle: it's good solution, but with multi hfc |
| 22:07.20 | *** part/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 22:07.22 | CoaxD | Vulture: I've thought of moving to Sveasoft, but.. Why? If it works with the linksys firmware, there aint no reason to dick with it |
| 22:07.45 | CoaxD | nicole/nicole 207.195.212.14 D N 255.255.255.255 5056 OK (15 ms) |
| 22:07.58 | lohelle | kietlak: multi hfc? |
| 22:07.59 | |Vulture| | CoaxD: I use their QoS and their SSH |
| 22:08.09 | CoaxD | Feb 7 16:07:53 NOTICE[1119030192]: chan_sip.c:5955 handle_response: Peer 'nicole' is now TOO LAGGED! |
| 22:08.18 | CoaxD | nicole/nicole 207.195.212.14 D N 255.255.255.255 5056 LAGGED (2014 ms) |
| 22:08.19 | |Vulture| | still with 200? |
| 22:08.22 | kietlak | lohelle: quad/octo bri |
| 22:08.30 | lohelle | ok |
| 22:08.37 | CoaxD | Vulture: Ahhh |
| 22:08.40 | labo | well nicole is not local .. |
| 22:09.02 | kietlak | it is stable |
| 22:09.04 | labo | 2014 looks like a cheap conn |
| 22:09.04 | CoaxD | labo: nicole is indeed local. there's a firewall in the middle, but its all in the same building |
| 22:09.42 | CoaxD | labo: There's a linksys WRT54G (for the intranet) in the middle. otherwise, its ethernet to ethernet |
| 22:09.46 | |Vulture| | CoaxD: but that firewall has been there all along.. this is an issue with the WRT right? |
| 22:09.54 | CoaxD | and you'll note: right now.. |
| 22:09.56 | CoaxD | nicole/nicole 207.195.212.14 D N 255.255.255.255 5056 OK (12 ms) |
| 22:10.00 | CoaxD | You got it. |
| 22:10.02 | |Vulture| | damn 12ms |
| 22:10.04 | labo | oh |
| 22:10.05 | |Vulture| | thats crazy |
| 22:10.12 | CoaxD | With an etherfast 802.11b firewall it doesnt happen |
| 22:10.19 | CoaxD | vulture: Crazy? |
| 22:10.23 | |Vulture| | mine always say like ~60ms I duno why either |
| 22:10.26 | |Vulture| | IP500s |
| 22:10.27 | CoaxD | ohhhh |
| 22:11.02 | |Vulture| | there isn't any QoS on default firware right? |
| 22:11.09 | CoaxD | Vulture: Not that i know of |
| 22:11.16 | CoaxD | This thing has been configured right out of the box |
| 22:11.27 | |Vulture| | I know if you set it wrong then it can gimp certian ports |
| 22:11.32 | |Vulture| | oky |
| 22:11.43 | |Vulture| | try running a ping from the server to the device |
| 22:11.49 | |Vulture| | see if it fluctuates |
| 22:11.58 | CoaxD | Cant |
| 22:12.01 | |Vulture| | or if it is just * and the WRT not |
| 22:12.03 | CoaxD | device is on fake ip |
| 22:12.04 | |Vulture| | ah oky |
| 22:12.25 | CoaxD | (So in truth, i dont know how it is even qualifying it) |
| 22:12.27 | Juggie | vul |
| 22:12.33 | |Vulture| | sup? |
| 22:12.35 | CoaxD | probably is qualifying straight to the firewall's ip |
| 22:12.45 | CoaxD | but i cant ping that |
| 22:12.48 | Juggie | i fucking broke my collar bone :( |
| 22:12.51 | |Vulture| | CoaxD: yea thats and interesting setup.. |
| 22:12.55 | |Vulture| | Juggie: how the hell? |
| 22:12.57 | CoaxD | Vulture: Well, its gonna be pretty standard |
| 22:13.06 | CoaxD | Vulture: Sip clients behind the firewall, asterisk ahead of the firewall |
| 22:13.22 | |Vulture| | CoaxD: you have NAT on under the sip.conf and stuff right? |
| 22:13.29 | CoaxD | Vulture: of course |
| 22:13.37 | |Vulture| | CoaxD: I just figured Id check |
| 22:13.39 | Juggie | skiiing, kid cut me off, i tried to avoid, we hooked skis (my right leg between his two skis |
| 22:13.46 | CoaxD | nat = yes |
| 22:13.46 | CoaxD | canreinvite=no |
| 22:13.46 | CoaxD | qualify = 200 |
| 22:13.55 | |Vulture| | CoaxD: cisco phones? |
| 22:13.58 | Juggie | i flew on an angle, landed shoulder first |
| 22:14.01 | CoaxD | Vulture: Newp. Its a Sipura 2000 |
| 22:14.04 | labo | maybe congestion on the link ? |
| 22:14.07 | |Vulture| | Juggie: damn thats nasty |
| 22:14.07 | Mother_ | Juggie: sorry to hear that :( |
| 22:14.12 | Juggie | yes |
| 22:14.19 | Juggie | sling for 4 weeks |
| 22:14.20 | CoaxD | Vulture: It also doesn't always happen. It only happens when calls come in |
| 22:14.20 | |Vulture| | Juggie: how long you down for? |
| 22:14.23 | CoaxD | Vulture: (or go out) |
| 22:14.26 | Juggie | 4 weeks |
| 22:14.27 | CoaxD | Vulture: Then it eventually settles down |
| 22:14.40 | |Vulture| | yea it sounds like a QoS issue but I duno how it could be |
| 22:14.46 | CoaxD | Vulture: Me either ;/ |
| 22:14.59 | Juggie | sucks, one handed tying |
| 22:15.10 | CoaxD | Vulture: That and the fact that the firewall would have to be the only QoS itnerpreter; because there's nothing else on this network that even knows to speak QoS |
| 22:15.14 | Luke-Jr | externip = 207.192.221.172 |
| 22:15.14 | Luke-Jr | register => 9136744395:xxxx:9136744395@sip.sipmedia.com |
| 22:15.17 | Juggie | i will have to work for 3 weeks one handed |
| 22:15.24 | CoaxD | course, this firmware might be old |
| 22:15.25 | Luke-Jr | Any idea why Asterisk isn't using the externip? |
| 22:15.28 | CoaxD | came out august 3rd, 2004 |
| 22:16.20 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 22:16.28 | |Vulture| | CoaxD: you try turning off the firewall ability on the WRT to see if it works? |
| 22:16.40 | CoaxD | Vulture: That would, um, bust everything, no? |
| 22:16.49 | CoaxD | Vulture: By definition, NAT *IS* firewall? |
| 22:17.13 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 22:17.24 | CoaxD | Vulture: I had actually noticed the option, but wondered if it'd do bad things |
| 22:17.30 | Derkommissar | Music on hold is choppy, what can i do to actually fix this ? |
| 22:17.36 | jskcr | Hy all. |
| 22:18.32 | *** join/#asterisk kant (~bernd@63.245.57.70) |
| 22:18.51 | CoaxD | Vulture: there is firmware that is quite a bit newer; let me research that |
| 22:19.05 | terrapen | is NuFone not accepting PayPal payments any longer? |
| 22:19.36 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 22:19.39 | jerlique | whats the difference between AgentLoginCallback and AddQueueMember, dont they achieve the same task? |
| 22:19.39 | jskcr | Coax are you running ztdummy or digium device? |
| 22:19.45 | |Vulture| | CoaxD: I think the WRT has strict rules for blocking packets and that might be causing it |
| 22:20.00 | |Vulture| | CoaxD: by turning it off you should retain the NAT but remove the IDS functions |
| 22:20.20 | |Vulture| | CoaxD: which might be a problem of dropping packets... worth a try |
| 22:20.26 | CoaxD | Vulture: What the fuck is IDS functions?!#$ |
| 22:20.31 | terrapen | fuck! |
| 22:20.34 | CoaxD | Vulture: I'm a sysadmin and have never heard of that! *lol* |
| 22:20.38 | terrapen | i set up my nufone account w/ paypal |
| 22:20.44 | terrapen | but it doesn't let you add money |
| 22:21.02 | CoaxD | terrapen: They have a CC# thing on their website now |
| 22:21.12 | shido6 | aroo? |
| 22:21.21 | CoaxD | Vulture: So basically it stops the ability to forward ports and shit |
| 22:21.25 | CoaxD | Vulture: Will try |
| 22:21.32 | terrapen | i dont want CC# :( |
| 22:21.35 | terrapen | i need paypal |
| 22:21.49 | CoaxD | ah |
| 22:21.50 | CoaxD | lame |
| 22:21.56 | CoaxD | i'm sure jerjer would take it |
| 22:22.14 | CoaxD | Vulture: nicole/nicole 207.195.212.14 D N 255.255.255.255 5056 OK (9 ms) |
| 22:22.21 | Luke-Jr | How can I get * to act as a SIP client only on a specific IP/interface? |
| 22:22.22 | CoaxD | Vulture: It may just work. let me test |
| 22:22.31 | |Vulture| | CoaxD: Intruder Detection System |
| 22:22.37 | CoaxD | ohhh |
| 22:22.38 | |Vulture| | CoaxD: oky lemme know |
| 22:22.45 | CoaxD | Vulture: How does it detect an intruder, exactly? :) |
| 22:23.12 | CoaxD | VUlture: nicole/nicole 207.195.212.14 D N 255.255.255.255 5056 LAGGED (1016 ms) |
| 22:23.15 | CoaxD | Newp. didnt fix |
| 22:23.18 | |Vulture| | CoaxD: not sure.. but the general concept is look for a large number of packets to a single port and if it is.. |
| 22:23.19 | |Vulture| | damn |
| 22:23.22 | Luke-Jr | (I have network interfaces for both LAN (w/ inet via NAT) and WAN; SIP really only works directly via WAN) |
| 22:23.42 | terrapen | how lame |
| 22:23.59 | terrapen | you can set up the damn account w/ paypal |
| 22:24.00 | |Vulture| | CoaxD: might want to try a different WRT firmware see if that is it |
| 22:24.01 | CoaxD | Vulture: How do i know, from looking at the web interface, what version my WRT54G is? |
| 22:24.09 | |Vulture| | its on the bottom of the WRT |
| 22:24.12 | terrapen | but there just isn't a way to add funds with it |
| 22:24.14 | CoaxD | Vulture: bought it yesterday from a CompUSA |
| 22:24.16 | Derkommissar | How can i fix the choppy music on hold ? |
| 22:24.21 | |Vulture| | its either a 2.2 or 2.0 |
| 22:24.32 | |Vulture| | I got one 2 weeks ago from compusa and it was a 2.2 |
| 22:24.33 | CoaxD | Vulture: well apparently they have a 3.0 unit |
| 22:24.43 | CoaxD | Oh. Does that prevent me from doing any sort of cool shit that a 3.0 woudl? |
| 22:24.46 | CoaxD | :) |
| 22:24.48 | |Vulture| | oh damn... I duno if sveasoft even has 3.0 |
| 22:24.55 | |Vulture| | lemme check 1s |
| 22:25.21 | outtolunc | last i read 3.0 wasn't in the list |
| 22:25.31 | mikegrb | terrapen: send paypal funds to sales@nufone.net with the username in the comment |
| 22:25.44 | nestAr | sweet.. if all agents logout of a queue.. callers waiting in the queue just sit there.. forever.. |
| 22:25.46 | |Vulture| | yea it still isnt |
| 22:25.52 | |Vulture| | looking to see if it is in the forum |
| 22:26.01 | Luke-Jr | How can I get * to act as a SIP client only on a specific IP/interface? (I have network interfaces for both LAN (w/ inet via NAT) and WAN; SIP really only works directly via WAN) |
| 22:26.27 | mikegrb | nestAr: there is an alternative solution, sends them to an extention so you can play a mesages then send them to voice mail or hang up or something |
| 22:27.13 | |Vulture| | nope cant find it |
| 22:27.15 | |Vulture| | I got to get to class |
| 22:27.17 | nestAr | mikegrb: as an alternative to using call queues? |
| 22:27.23 | |Vulture| | CoaxD: try looking at the bottom it should say it |
| 22:27.30 | nestAr | i need call queues.. we're an ISP |
| 22:27.35 | netsurfer | nestAr - check out ICD on the wiki - it might provide more features |
| 22:27.40 | mikegrb | nestAr: no, alternitive setting for call queue |
| 22:27.44 | CoaxD | Vulture: i HAVE A VERSION 2 |
| 22:27.49 | mikegrb | nestAr: ICD hasn't been updated in a long time |
| 22:27.53 | CoaxD | er. caps |
| 22:28.29 | outtolunc | nestAr: show application queue |
| 22:28.30 | nestAr | nvm |
| 22:28.32 | nestAr | i fixed it |
| 22:28.38 | nestAr | i had leavewhenempty = yes in [general] |
| 22:28.44 | nestAr | i added it to all queues |
| 22:29.08 | CoaxD | Holy SHIT the firmware is big. 3.3mb! hahaha |
| 22:29.40 | CoaxD | Firmware 3.01.3 |
| 22:29.40 | CoaxD | - Updated wireless driver |
| 22:29.40 | CoaxD | - Supports hardware version 2.2 (cannot downgrade to previous versions) |
| 22:29.40 | CoaxD | - Resolves issue with VoIP adapters |
| 22:29.40 | CoaxD | - Resolves issue with long domain names |
| 22:29.42 | CoaxD | Hahahahahahaha |
| 22:29.47 | outtolunc | coaxd, there is a 5meg one some one did up in there <G> |
| 22:30.32 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 22:30.32 | nestAr | wheee |
| 22:31.45 | |Vulture| | CoaxD: haha nice |
| 22:32.03 | CoaxD | Vulture: For god damn sure |
| 22:32.14 | CoaxD | Vulture: Its in the process of upgrading. Lets hope my computer doesn't power off |
| 22:32.22 | CoaxD | Vulture: (is it actually possible to brick these things?) |
| 22:32.27 | |Vulture| | VERY HARD |
| 22:32.39 | CoaxD | Vulture: Thats what I thought. i thought that perhaps it'd have a minimal firmware in place |
| 22:32.40 | |Vulture| | they have TFTP on another firmware chip |
| 22:32.41 | *** join/#asterisk zoa (~zoa@213.219.141.4.adslpower.by.edpnet.be) |
| 22:32.55 | CoaxD | Vulture: See, THATS what these companies should ALL DO |
| 22:32.59 | |Vulture| | yea you can tftp flash |
| 22:33.08 | CoaxD | Upgrade is successful. |
| 22:33.08 | CoaxD | <PROTECTED> |
| 22:33.08 | Godsey | nestAr: how do you manage queues? |
| 22:33.19 | Godsey | agents? |
| 22:33.24 | |Vulture| | yea the WRT with sveasoft firmware is like my fav. router and Ive used cisco routers :P |
| 22:33.37 | CoaxD | Vulture: not sure if you noticed, but this is a rev2.0 |
| 22:33.50 | CoaxD | Vulture: Honestly, the ciscos are just as hard to brick |
| 22:33.53 | sivana | what's a good mfg of echo cancellation equip for T1/PRI? |
| 22:33.53 | niZon | WRT's own :P |
| 22:34.19 | CoaxD | Vulture: Even if you brick both bootstrap and main boot os, and hell, even the bios, there's still the debugger |
| 22:34.40 | wankel | heh. yeah, you have to be pretty talented to kill a cisco. |
| 22:34.48 | CoaxD | vulture: within the debugger it is possible to copy over an image via xmodem, and jmp and execute |
| 22:34.49 | wankel | you can always reload it over the serial with y-modem |
| 22:34.50 | outtolunc | make sure to set the 'wait boot' |
| 22:34.58 | CoaxD | wankel: Oh, is it y-modem now? *g* |
| 22:35.05 | wankel | yeah, crc32! |
| 22:35.07 | CoaxD | wow! 1024 bytes at a time nowadays! *g* |
| 22:35.16 | Nugget | ymodem-g! |
| 22:35.17 | CoaxD | wankel: Did you ever use ZedZap? |
| 22:35.31 | |Vulture| | CoaxD: no but someone made a jtag solution for it |
| 22:35.32 | wankel | nope |
| 22:35.40 | CoaxD | wankel: it was a download protocol very similar to z-modem except it supported...*GASP*...8k chunks! |
| 22:35.50 | wankel | heh |
| 22:35.54 | CoaxD | (zmodem was 1k) |
| 22:35.54 | wankel | i used bimodem! |
| 22:36.01 | wankel | and lots and lots of zmodem |
| 22:36.10 | zoa | smodem was the best |
| 22:36.12 | Mavvie | zmodem went from 64 bytes to 8192 bytes. |
| 22:36.13 | wankel | and tried a few other bizarre things |
| 22:36.16 | shido6 | zmodem xfers |
| 22:36.25 | zoa | smodem was up and down at the same time |
| 22:36.27 | zoa | + chat |
| 22:36.27 | CoaxD | Mavvie: Ya |
| 22:36.31 | Nugget | bah. I just F'Req'd the file and went to bed. |
| 22:36.33 | *** join/#asterisk florz (nobody@odnb-d9baa555.pool.mediaWays.net) |
| 22:36.34 | wankel | zoa: yeah, bimodem, too. |
| 22:36.44 | nestAr | Godsey: i'm using AddQueueMember at the moment.. |
| 22:37.01 | wankel | some of the bi-directional stuff didn't work right with HST modems, though, since they were asymmetric. |
| 22:37.09 | nestAr | my boss seems mildly interested in doing the off-hook thing though.. |
| 22:37.15 | |Vulture| | if there was one class I didn't have to go to, it would be this one, Modern Hebrew Culture ack |
| 22:37.28 | Nugget | wankel: http://slacker.com/photos/computers/IMG_0864 |
| 22:37.31 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfkh2.dialup.mindspring.com) |
| 22:37.56 | Nugget | not quite a round-led hst 9600, but still nice and old. |
| 22:38.07 | wankel | it's doing reverse dns on me, i think |
| 22:38.18 | Nugget | are you using linux 2.6.8? |
| 22:38.29 | wankel | no |
| 22:38.38 | |Vulture| | CoaxD: good luck with that new firmware |
| 22:38.47 | CoaxD | Vulture: Thanks. Its not working anyway. |
| 22:38.49 | *** join/#asterisk implicit (~implicit@lgb-cust-66.18.140.106.mpowercom.net) |
| 22:38.49 | Nugget | linux 2.6.8 has bugs that make it not work through my firewall. same symptoms. :) |
| 22:38.54 | Nugget | linux is poo. |
| 22:38.54 | CoaxD | Vulture: I mean the firmware is, but the problem still remains |
| 22:38.54 | wankel | okay, it timed out finally. |
| 22:38.57 | CoaxD | Vulture: I'll find the issue |
| 22:39.01 | CoaxD | Vulture: No worries :) |
| 22:39.17 | wankel | our reverse dns is broken, so sites that try to resolve before serving content (bad idea) hang for like 3 minutes) |
| 22:39.21 | Nugget | *nod* |
| 22:39.24 | CoaxD | Vulture: Thank you very much for the help you gave |
| 22:39.45 | wankel | yay. it's an courier. they look like aircraft carriers to me. |
| 22:39.47 | nestAr | it's amazing the number of ISP's who don't have working reverse dns |
| 22:39.55 | wankel | i had one of those and a multitech standard 14.4 |
| 22:40.13 | wankel | multitech was a nice metal case. had a little callback shell with multiple accounts in the bios. |
| 22:40.22 | Nugget | that one was my second courier. my first was the round-led 9600bps hst. |
| 22:40.22 | wankel | way cool modem. |
| 22:40.35 | Nugget | with the "CLICK" relay when you made a 9600 connect. |
| 22:40.42 | wankel | heh |
| 22:40.49 | Nugget | so I could tell from across the room when someone cool called the board. |
| 22:40.49 | CoaxD | I still have a couple courier v.everythings |
| 22:41.00 | Luke-Jr | externip in sip.conf doesn't work with Call-ID, apparently :/ |
| 22:41.23 | nestAr | i had a courier HST too |
| 22:41.28 | nestAr | as big as my laptop |
| 22:41.40 | nestAr | i think i still have it somewhere in my basement |
| 22:42.02 | Mother_ | them days where you could see PCB components with the naked eye... |
| 22:42.03 | file[laptop] | Luke-Jr: Call-ID doesn't matter... it just identifies the call |
| 22:44.34 | Nukemizer | Asterisk for Windows ? not there is a recipe ! |
| 22:44.41 | Luke-Jr | file[laptop]: Hrm; can't seem to register w/ POTS provider |
| 22:45.03 | file[laptop] | then it's another problem |
| 22:46.15 | Luke-Jr | Would you be able to identify a problem from the packet? |
| 22:46.37 | file[laptop] | a sip debug probably |
| 22:46.47 | Luke-Jr | What's that? |
| 22:46.48 | file[laptop] | put it on pastebin.ca |
| 22:46.51 | *** join/#asterisk SuperAlex (~SuperAlex@adsl-19-19-207.asm.bellsouth.net) |
| 22:46.54 | file[laptop] | you type sip debug |
| 22:46.57 | file[laptop] | then sip reload |
| 22:47.03 | file[laptop] | and when it trys to register, it'll show up on the screen |
| 22:47.08 | file[laptop] | all the SIP packets that go back and forth! |
| 22:47.47 | SuperAlex | hello |
| 22:47.49 | Luke-Jr | ok |
| 22:47.56 | dca[laptop] | hi file! |
| 22:47.58 | file[laptop] | hi |
| 22:48.01 | dca[laptop] | hehe |
| 22:48.30 | SuperAlex | can i ask a simple asterisk related question ? |
| 22:49.15 | Nugget | You just did. |
| 22:49.15 | dca[laptop] | file: 'nother question...when the caller (from PSTN) calls my ip phone and then hangs up, the sip phone doesnt' hangup, wouldn't asterisk understand the BYE |
| 22:49.25 | file[laptop] | dca[laptop]: yes it does |
| 22:49.34 | SuperAlex | i've got a budgetone 100 phone that i'm trying to get to access asterisk voicemail, but when it asks for the password and i enter the password it doesn't come through on the other end ... i.e. password is '' |
| 22:49.38 | dca[laptop] | then why doesnt the ip pohen hangup? |
| 22:49.40 | file[laptop] | dca[laptop]: problem is elsewhere |
| 22:49.47 | dca[laptop] | when it is just asterisk it works |
| 22:49.58 | dca[laptop] | but throw ser into the equation and it poops |
| 22:50.07 | file[laptop] | then your ser.cfg isn't right |
| 22:50.13 | dca[laptop] | hmmm |
| 22:50.13 | nestAr | SuperAlex: check the DTMF setting in the Budgetone's web interface |
| 22:50.21 | dca[laptop] | can i im ya? |
| 22:50.35 | SuperAlex | nestAr: it is set to In-Audio, should i set it to something else? |
| 22:50.41 | netsurfer | yup.. i had that prob too.. wrong dtmf |
| 22:51.07 | nestAr | SuperAlex: I used SIP INFO |
| 22:51.08 | SuperAlex | nestAr: my only concern is that if i change it, then i may not be able to use other tone related systems outside my network |
| 22:51.09 | netsurfer | SuperAlex - to make sure dtmf is working, punch the digits in fast then press # u should get the password prompt straight away no delay |
| 22:51.14 | Luke-Jr | http://pastebin.ca/5424 |
| 22:51.24 | file[laptop] | dca[laptop]: I'm working, on work. |
| 22:51.25 | nestAr | SuperAlex: i don't recall having any other problems. |
| 22:51.25 | SuperAlex | ah k thnx |
| 22:51.32 | file[laptop] | so unless you want to donate money to me... |
| 22:51.41 | dca[laptop] | file, k |
| 22:51.41 | nestAr | but i've since ditched the budgetone's |
| 22:52.03 | dca[laptop] | file: np, i'll keep at it, more educational this way i suppost |
| 22:52.06 | dca[laptop] | er suppose |
| 22:52.09 | SuperAlex | nestAr: ? ditched? why? |
| 22:52.16 | Luke-Jr | file[laptop]: http://pastebin.ca/5424 |
| 22:52.44 | nestAr | SuperAlex: just had a lot of problems with them.. and they're not really "office quality" |
| 22:52.47 | file[laptop] | haha they're using SER |
| 22:52.56 | nestAr | they'd be ok for my house, but not for my office |
| 22:52.58 | Luke-Jr | what's SER? |
| 22:52.59 | SuperAlex | nestAr: ah, thnx, i'll take that into consideration |
| 22:53.11 | file[laptop] | Luke-Jr: it's some SIP software |
| 22:53.11 | nestAr | :) |
| 22:53.22 | Luke-Jr | file[laptop]: Incompatible w/ Asterisk? |
| 22:53.30 | shido6 | budgetones work |
| 22:53.35 | file[laptop] | Luke-Jr: oh no it's compatible, lemme read this |
| 22:54.19 | file[laptop] | Luke-Jr: either asterisk's sip.conf is not quite right, it's behaving wrong, or else your other side is simply rejecting it |
| 22:54.35 | Luke-Jr | file[laptop]: What part of sip.conf should I pastebin? |
| 22:55.07 | file[laptop] | gimme time :p I'm working on actual work |
| 22:56.21 | file[laptop] | Luke-Jr: it might be picky... it might need all your stuff to be going through sip1.xchangetele.com |
| 22:56.25 | file[laptop] | not just the realm... |
| 22:57.49 | file[laptop] | experiment |
| 22:59.03 | Luke-Jr | Feb 7 22:59:00 [asterisk] WARNING[3981]: chan_sip.c:1398 in create_addr: No such host: sip1.xchangetele.com_ |
| 22:59.04 | SuperAlex | does anyone know if it's okay to plug in an analog phone line (not ethernet cat-5) into a TDM400P card? |
| 22:59.39 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:00.20 | jgaviria | SuperAlex: of course |
| 23:00.36 | Derkommissar | :-/ |
| 23:00.36 | SuperAlex | jgaviria: thnx! |
| 23:00.58 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 23:01.04 | blitzrage | file[laptop]: you should stop hanging out in here |
| 23:01.24 | Derkommissar | :-/ |
| 23:01.31 | Derkommissar | im still puzzled. |
| 23:01.38 | Derkommissar | why is music on hold choppy ? |
| 23:01.41 | file[laptop] | yes, very unproductive |
| 23:01.58 | *** part/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 23:02.18 | nestAr | Derkommissar: overloaded box? |
| 23:02.23 | netsurfer | Derkommissar - u got a timer source running ? |
| 23:02.38 | modulus_ | jbot babelfish de en du bist eine kleine dummer schweinehund |
| 23:02.45 | netsurfer | LMFAO |
| 23:02.45 | nestAr | lol |
| 23:03.19 | *** join/#asterisk nickweb (Nick@host81-157-112-117.range81-157.btcentralplus.com) |
| 23:03.23 | nickweb | hey guys, |
| 23:03.28 | nickweb | very quick question.. |
| 23:03.35 | nickweb | (hardware related..) |
| 23:03.55 | nickweb | does anyne have any experience with the Wildcard X100P? |
| 23:04.02 | nestAr | i've got one at home |
| 23:04.24 | outtolunc | i 'experienced' viewing it on the digium website <G> |
| 23:04.28 | nickweb | if so, if i just have that one card in my box, is that enough to run asterisk, and how many internal lines can i have from that one card? |
| 23:04.44 | netsurfer | iv got experience.. at not receiving one I bought on ebay :o\ |
| 23:04.58 | nestAr | X100P gives you the ability to have one incoming POTS line into asterisk |
| 23:04.59 | Derkommissar | netsurfer, no |
| 23:05.00 | postel | nickweb: the X100 is a single FXO card |
| 23:05.06 | Derkommissar | i dont have a timer |
| 23:05.16 | Derkommissar | why would a timer affect the MOH |
| 23:05.25 | netsurfer | Derkommissar - make zaptel drivers up and modprobe "ztdummy" |
| 23:05.32 | nickweb | postel, does that mean i need one for each extension? or have i completley read it wrong? lol |
| 23:06.01 | netsurfer | Derkommissar - its the way it is.. check on the wiki for more info |
| 23:06.02 | postel | nickweb: you got it all wrong, it just means what it says on the tin, you can get ONE POTS line in |
| 23:06.06 | outtolunc | http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer |
| 23:06.08 | nestAr | nickweb: you need one for each pots line |
| 23:06.12 | nestAr | incoming phone line |
| 23:06.19 | nestAr | it doesn't do anything for phones.. |
| 23:06.27 | nickweb | ah.. oh |
| 23:06.36 | nickweb | man. i need to research more. lol |
| 23:06.37 | nestAr | you need VOIP phones or a TDM400 + analog phones |
| 23:06.42 | netsurfer | x100p is just a glorified modem |
| 23:06.52 | nestAr | netsurfer: true |
| 23:06.55 | denon | anyone know what the deal is with this stuff? : Dec 13 14:52:03 NOTICE[25886]: app_dial.c:927 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 23:07.02 | nestAr | in fact.. mine is a modem... :) |
| 23:07.12 | nestAr | that's what the auction said |
| 23:07.13 | nickweb | so i would need another card to run internal phones then.. any cheap ones kicking about? (what can i say.. im economical! lol) |
| 23:07.26 | Derkommissar | netsurfer, thanks doing so rigth now |
| 23:07.27 | nestAr | nickweb: softphone is free. |
| 23:07.43 | netsurfer | u need FXS for phones |
| 23:07.46 | nestAr | nickweb: budgetone ip phones are $60-something |
| 23:07.52 | nickweb | geez |
| 23:07.57 | nickweb | i better think this through again. lol |
| 23:08.02 | netsurfer | or use the voip part ;) |
| 23:08.14 | Derkommissar | netsurfer, i get this afther i install ztdummy |
| 23:08.15 | Derkommissar | Notice: Configuration file is /etc/zaptel.conf |
| 23:08.15 | Derkommissar | line 0: Unable to open master device '/dev/zap/ctl' |
| 23:08.15 | Derkommissar | 1 error(s) detected |
| 23:08.15 | Derkommissar | FATAL: Error running install command for ztdummy |
| 23:08.33 | netsurfer | su root |
| 23:08.37 | netsurfer | oops |
| 23:09.01 | Derkommissar | im as root |
| 23:09.04 | netsurfer | Derkommissar - edit Makefile uncomment ztdummy |
| 23:09.11 | Derkommissar | i did |
| 23:09.34 | Derkommissar | dam i hate redhat and the different root levels |
| 23:09.39 | Derkommissar | su - did it |
| 23:10.01 | Derkommissar | ztdummy 7748 0 |
| 23:10.01 | Derkommissar | zaptel 190980 1 ztdummy |
| 23:10.07 | netsurfer | im the wrong person to ask.. been fighting with zaptel for 3 days now.. only way I can get it to install is with a nice clean bloated 2.4.29 kernel install |
| 23:10.10 | Derkommissar | so now MOH should work better |
| 23:10.22 | outtolunc | devon: it can't find the peer |
| 23:10.29 | netsurfer | great Derkommissar |
| 23:10.51 | nickweb | thanks for the help guys! :P |
| 23:11.03 | Derkommissar | do i need to edit zapata.conf or zaptel.conf or anything like that ? |
| 23:11.09 | netsurfer | no |
| 23:11.26 | netsurfer | once u modprobe ztdummy ur good to go |
| 23:13.09 | Derkommissar | MOH still choppy |
| 23:13.15 | eipi | derkomissar i had the same problem: read README.udev |
| 23:13.16 | nestAr | whee.. the timeout option for the Queue command actually works |
| 23:13.20 | nestAr | HUZZAH! |
| 23:14.00 | netsurfer | Derkommissar - maybe an overloaded box? |
| 23:14.00 | *** join/#asterisk jerlique (jerlique@lnk2.adl.adsl.esc.net.au) |
| 23:14.07 | Derkommissar | nope |
| 23:14.28 | Derkommissar | Cpu(s): 0.0% us, 0.0% sy, 0.0% ni, 100.0% id, 0.0% wa, 0.0% hi, 0.0% si |
| 23:14.28 | Derkommissar | Mem: 1033840k total, 997648k used, 36192k free, 101244k buffers |
| 23:14.37 | Derkommissar | i made chan_alsa not load |
| 23:14.38 | eipi | derkomissar: Linux Kernel 2.6 changes the way that manages Devices and you have to read README.udev to make work ztdummy |
| 23:14.40 | Derkommissar | is that why ? |
| 23:14.42 | netsurfer | Derkommissar - are u using an mp3 that came with * ? |
| 23:14.55 | Derkommissar | yes |
| 23:15.02 | Derkommissar | uff i do have kernel2.6 |
| 23:15.11 | netsurfer | :o\ |
| 23:15.17 | eipi | derkomisssar: i loose 1 week with with that problem |
| 23:15.29 | Derkommissar | what do i have to do ? |
| 23:15.46 | eipi | read README.udev under asterisk directory |
| 23:16.58 | fizbar | does anyone know how to put in a bit of delay between when asterisk "picks up" a zaptel line and when it starts sending DTMF digits for dialing? |
| 23:17.02 | jerlique | whats the difference between AgentLoginCallback and AddQueueMember, dont they achieve the same task? |
| 23:17.45 | postel | <PROTECTED> |
| 23:17.45 | postel | [Synopsis]: |
| 23:17.45 | postel | Waits for some time |
| 23:17.48 | postel | fizbar: |
| 23:18.17 | outtolunc | wait(x) or dial(tech/dev/wwwXXXXXX) |
| 23:18.21 | fizbar | postel: yes, this all takes place within the Dial application, so any waits would be before or after the dial, i need a wait within the dial |
| 23:18.44 | modulus_ | jbot babelfish de en du werdest eine krankenschwester brauchen |
| 23:19.13 | *** part/#asterisk zpn (~xpn@dhcp-166.digium.com) |
| 23:19.40 | florz | jbot babelfish de en du werdest einen Deutschlehrer brauchen =:-) |
| 23:20.11 | *** join/#asterisk nicolasg (~chatzilla@ip-189.houseware.com.ar) |
| 23:20.21 | florz | oh, IC, =:-) is =: -) in English ... |
| 23:20.38 | eipi | now i have this problem: when i do dial under CLI> the sounds goes choppy and receives BROKEN PIPE error messages |
| 23:20.41 | eipi | any idea???? |
| 23:21.25 | Qwell | heh, my CLI stopped playing sound altogether. It worked fine until last night, then it started going REALLY slow(and not playing sound). Now it just hangs on the dial commands |
| 23:21.32 | jgaviria | somebody running asterisk with R2 signalling? |
| 23:22.06 | implicit | hi jgaviria |
| 23:22.31 | modulus_ | jbot babelfish en de you will need a nurse |
| 23:22.33 | eipi | qwell... :) |
| 23:22.39 | modulus_ | eww |
| 23:23.23 | florz | jbot babelfish de en Sie benötigen eine Krankenschwester |
| 23:23.36 | eipi | a sexy nurse please |
| 23:23.39 | modulus_ | babelfish is borked |
| 23:23.46 | florz | jbot babelfish en de they need a nurse |
| 23:23.59 | florz | hmmm |
| 23:24.33 | modulus_ | jbot babelfish de en est is zeit fur racht |
| 23:24.51 | modulus_ | jbot babelfish en de it is time for revenge |
| 23:25.01 | modulus_ | jbot babelfish de en es is zeit fur rache |
| 23:25.08 | Derkommissar | ok ztdummy is working |
| 23:25.12 | Derkommissar | and i still get this |
| 23:25.13 | Derkommissar | Feb 7 12:20:23 WARNING[18078]: res_musiconhold.c:788 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 23:25.50 | eipi | derkommisar, you have to uncommet ztdummy from makefile |
| 23:25.52 | eipi | wait me |
| 23:25.55 | eipi | let me check |
| 23:25.59 | Derkommissar | yea i did |
| 23:26.36 | *** join/#asterisk Druken (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 23:27.03 | *** part/#asterisk kingtaco|laptop (~kingtaco@kingtaco.developer.gentoo) |
| 23:27.54 | eipi | on zaptel Makefile i have: |
| 23:27.56 | netsurfer | Derkommissar - does it appear when u lsmod ? |
| 23:28.08 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 23:28.09 | *** join/#asterisk liquidno2 (~foo@cs68201147-69.sw.rr.com) |
| 23:28.10 | eipi | MODULES=zaptel tor2 torisa wcusb wcfxo wctdm ztdummy |
| 23:28.34 | Qwell | Doesn't uhci_usb have to be loaded for ztdummy to work in 2.4.x? |
| 23:28.52 | eipi | derkomissar: other... for zaptel do: make linux26 |
| 23:28.58 | netsurfer | uhmm u could be right, Qwell |
| 23:29.07 | liquidno2 | Is it necessary to run ztcfg after running modprobe? |
| 23:29.11 | Qwell | I'm only assuming, of course, that he's running 2.4 |
| 23:30.43 | redder86 | I'm using CVS stable from maybe a month ago and cannot transfer calls picked up from the Queue. |
| 23:30.46 | eipi | dermomissar i made another change more but im searching for it |
| 23:30.47 | redder86 | Anybody else seeing that? |
| 23:31.12 | liquidno2 | redder86: you know that they will suggest you update from the most recent cvs |
| 23:31.29 | liquidno2 | it almost goes without saying |
| 23:31.33 | Derkommissar | yes |
| 23:31.33 | liquidno2 | _almost_ |
| 23:31.34 | Derkommissar | Feb 7 12:23:35 WARNING[18110]: res_musiconhold.c:788 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. |
| 23:31.42 | Derkommissar | ztdummy 7748 0 |
| 23:31.42 | Derkommissar | zaptel 190980 1 ztdummy |
| 23:31.45 | redder86 | liquidno2: yeah, but I want to know if someone knows that this problem was fixed within the last month |
| 23:31.50 | Derkommissar | lsmod shows it |
| 23:31.58 | redder86 | liquidno2: *someone* knows what goes on in CVS |
| 23:32.04 | Qwell | Derkommissar: You aren't running 2.4.x, are you? |
| 23:32.11 | eipi | derkomissar: do you have DIGIUM hardware? |
| 23:32.17 | liquidno2 | redder86: I think it persoanlly maintained by a bunch of cave gnomes |
| 23:32.33 | Derkommissar | no |
| 23:32.36 | Derkommissar | its 2.6 |
| 23:32.41 | eipi | ok |
| 23:32.42 | liquidno2 | 1) Steal the CVS. 2) ????? 3) Profit. |
| 23:32.43 | redder86 | liquidno2: and if those committing changes are going to be as careless as to not have a changelog that gives useful information, then I just have to ask first |
| 23:32.48 | Derkommissar | but i alredy did everything that the README.udev says |
| 23:32.55 | redder86 | liquidno2: in hopes that someone gives me something useful. |
| 23:33.03 | liquidno2 | redder86: I am sure there is a changelog |
| 23:33.14 | eipi | ok comment lines in /etc/sysconfig/zaptel --- leaving only uncommented the ztdummy line |
| 23:33.15 | *** part/#asterisk Corvin (~zbysio@chello084010031149.chello.pl) |
| 23:33.15 | redder86 | liquidno2: there is, but it sucked last time I looked |
| 23:33.30 | redder86 | liquidno2: "lots of bug fixes" or something to that nature |
| 23:33.38 | liquidno2 | man |
| 23:33.46 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.221.172) |
| 23:33.50 | liquidno2 | the CVS change log is way out of date |
| 23:33.56 | redder86 | precicely my point |
| 23:34.02 | redder86 | cave gnomes indeed |
| 23:34.02 | liquidno2 | I know there is a way to check it in the CVS itself |
| 23:34.08 | redder86 | cvs history |
| 23:34.21 | liquidno2 | but it requires me remembering the cvs command and also good accounting from the commiters |
| 23:34.22 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 23:34.23 | redder86 | but have you *ever* actually tried to make sense of the stuff they write in those tags? |
| 23:34.46 | Druken | evening ass tricks world :) |
| 23:34.56 | eipi | darkomissar: do rmmod ztdummy and rmmod zaptel, compile zaptel: make clean; make linux26; make install; then start zaptel serivce |
| 23:35.03 | redder86 | Druken: it's a disasterisk tonight |
| 23:35.20 | Druken | redder86: oh? |
| 23:35.30 | Druken | who screwed what up ? |
| 23:35.41 | redder86 | yeah, can't transfer calls placed to the queue |
| 23:35.55 | Druken | i never could... hehehe |
| 23:35.59 | liquidno2 | odd |
| 23:35.59 | Druken | so that's nothing new |
| 23:36.08 | Derkommissar | :-/ |
| 23:36.10 | eipi | i have this problem: when i do dial under CLI> the sounds goes choppy and receives BROKEN PIPE error messages |
| 23:36.11 | liquidno2 | off hook and hangups are showing twice in the console |
| 23:36.12 | redder86 | Druken: do you know if it's been fixed recently? |
| 23:36.25 | brc_ | ~seen agi_namu |
| 23:36.27 | jbot | brc_: i haven't seen 'agi_namu' |
| 23:36.50 | eipi | darkomissar: any news? |
| 23:37.00 | Druken | redder86: no idea... do you have the transfer flag on the queue? |
| 23:37.00 | Derkommissar | nope |
| 23:37.19 | Derkommissar | MOH sucks |
| 23:37.28 | redder86 | Druken: I don't know about any transfer flag on the queue. In the Dial command? |
| 23:37.28 | brc_ | eh? |
| 23:37.29 | Derkommissar | and ztdummy has been properly installed |
| 23:37.34 | redder86 | Druken: I do SIP transfers |
| 23:37.36 | eipi | derkommissar: did you add the lines described in README.udev? |
| 23:37.41 | Derkommissar | yes |
| 23:37.55 | brc_ | Derkommissar, sounds like your timing is not working correctly |
| 23:38.14 | Derkommissar | what can i do to fix it ? |
| 23:38.19 | eipi | derkommissar: you have to follow the steps described in /usr/src/asterisk/README.udev |
| 23:38.30 | eipi | letme transcribe here |
| 23:39.29 | eipi | oops |
| 23:39.29 | Druken | redder86: http://www.voip-info.org/wiki-Asterisk+cmd+Queue |
| 23:39.29 | eipi | derkommissar: you have to follow the steps described in /usr/src/zaptel/README.udev |
| 23:39.31 | eipi | was on zaptel directory |
| 23:39.51 | redder86 | Druken: is the "transfer" mentioned in 'show application queue' a "#" transfer or a SIP transfer? |
| 23:40.22 | Derkommissar | yes i did |
| 23:40.29 | Derkommissar | what does RFC3389: 1 bytes, level 256... |
| 23:40.38 | Derkommissar | it keeps showing that message |
| 23:40.41 | eipi | did you add lines on: /etc/udev/rules.d/50-udev.rules? |
| 23:40.52 | Derkommissar | yes |
| 23:41.11 | Druken | redder86: no idea.... |
| 23:41.13 | redder86 | Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call. |
| 23:41.21 | eipi | did you add lines on: /etc/udev/permissions.d/50-udev.permission? |
| 23:41.29 | Derkommissar | yes |
| 23:41.34 | eipi | did you restart? |
| 23:41.42 | eipi | ed |
| 23:41.57 | eipi | or restart udev service |
| 23:41.57 | redder86 | "SIP transfers result in the Agent...." is incorrect |
| 23:42.15 | redder86 | SIP transfers result in the caller getting hung up on. |
| 23:42.16 | Derkommissar | i restarted udev |
| 23:42.33 | eipi | and you continue receiving the same? |
| 23:42.44 | Derkommissar | yes |
| 23:42.47 | Druken | redder86: sure it's not a dialplan error? |
| 23:43.08 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 23:43.22 | Derkommissar | what is this suposed to mean |
| 23:43.27 | eipi | i followed that steps, and now is working |
| 23:43.33 | Derkommissar | everytime the music cuts off it shows this. |
| 23:43.36 | Derkommissar | RFC3389: 1 bytes, level 256... |
| 23:43.49 | Derkommissar | let me reboot the computer to see what happens |
| 23:44.01 | eipi | dekommissar: i have that problem too, but first you have to resolve ztdummy and zaptel problem |
| 23:44.36 | eipi | i receive a BROKEN PIPE error message every time that sounds cut |
| 23:44.44 | redder86 | Druken: hrmmm... maybe. Do SIP transfers only happen within the same context? |
| 23:44.51 | redder86 | Druken: probably so. |
| 23:45.24 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 23:45.26 | Derkommissar | :-/ |
| 23:45.28 | Druken | redder86: yes, the extension has to be in the same context |
| 23:45.39 | redder86 | Druken: just checked, the extensions are within the same context. |
| 23:46.09 | redder86 | Druken: I've watched the CLI during this "transfer" and there is no indication of what is happening. Just a disconnection. |
| 23:46.13 | Druken | what context is the queue in? |
| 23:46.24 | redder86 | "operator" |
| 23:46.57 | Druken | hmm... |
| 23:47.15 | modulus_ | ES IS ZEITE FUR RACHE! |
| 23:47.46 | redder86 | Druken: and the extensions to which the call is being transferred is included (with an include) into the operator context |
| 23:48.19 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 23:48.34 | Druken | redder86: hmm.. strange... everything is on the local system right? no server inbetween ? |
| 23:48.58 | buddah | anyone know how i set up phones to dial other sip phones/transfer to other sip phones, via 4 digit extensions? |
| 23:49.59 | eipi | derkomissar any news? |
| 23:50.56 | redder86 | Druken: the call comes in on a PRI to an Asterisk box (TE405P) the call is routed to a different Asterisk box via IAX2 where it is then dispatched to the queue. |
| 23:51.07 | *** part/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 23:52.38 | *** join/#asterisk Mneumonic (~Mnemonic@206.231.230.230) |
| 23:53.24 | redder86 | Druken: http://bugs.digium.com/bug_view_page.php?bug_id=0003518 I can't make sense of Mark's comments here. |
| 23:53.30 | jerlique | How can I get agents to manually loggout? The voip-info site appears incorrect |
| 23:53.45 | redder86 | Druken: not that I ever do understand why Mark doesn't "get" the error report. |
| 23:54.08 | redder86 | jerlique: AgentCallBackLogin has a logout feature. Just press "#" for the new extension. |
| 23:54.51 | Mneumonic | Hey... I just have a few questions about asterisk... I am considering it for my small business. I have 2 business lines via vonage for the incoming lines, what is the most inexpensive hardware to connect them? and then what card do i need for connecting the phones? Plus are VoIP phones the only type of phones I can use with asterisk? Or can i pick up digital PBX phones? |
| 23:54.59 | liquidno2 | hmm |
| 23:55.25 | jerlique | if i enter # I get login incorrect? |
| 23:55.40 | liquidno2 | why am I getting two lines in the console for off hook and on hook notification |
| 23:57.16 | Druken | redder86: he said it's a config issue, you have # for starting a record, not for transfer |
| 23:58.22 | redder86 | Druken: I don't have # configured for anything except to send DTMF # |
| 23:58.42 | redder86 | Druken: I don't believe in # transfers, and I don't do recordings. |
| 23:58.51 | Druken | i'm just telling ya what mark said in his comments |
