| 00:01.45 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:08.09 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 00:11.42 | *** join/#asterisk varys_ (n=varys@74.255.38.2) |
| 00:18.14 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 00:18.52 | iratik | Anyone have a recomendation on a php ami api/interface ... trying various combinations of "php ami asterisk" on google... and suprisingly... i'm getting a release loose assortment of half assed projects |
| 00:19.20 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 00:21.39 | iratik | nevermind... i just found this "AJAM" stuff... this is great/awesome/amazing... it will work fine |
| 00:22.04 | jsgoecke | iratik Or go with Ruby and use Adhearsion... ;) |
| 00:23.39 | iratik | i knew i recognized your nickname |
| 00:23.55 | iratik | I'm ___plenty____ acquainted with adhearsion |
| 00:24.37 | iratik | I know adhearsion better than i want to know it... and had i been a better humanitarian and had more time...... and had learned github earlier... i probably would be a main contributer |
| 00:27.12 | seanbright | Qwell: hmmm. do you "smoke" it for as long as you smoke a regular cigarette? or is it like a one-hitter? |
| 00:28.58 | Qwell | about as long, not quite |
| 00:30.26 | iratik | <PROTECTED> |
| 00:30.58 | Qwell | iratik: my electronic cigarette |
| 00:31.28 | iratik | ahh..... |
| 00:31.58 | seanbright | had you tried anything before this? |
| 00:32.02 | seanbright | gum, patch, whatever? |
| 00:32.30 | Qwell | nope |
| 00:32.45 | Qwell | the intent isn't to "quit" |
| 00:33.13 | seanbright | it's to...? |
| 00:33.41 | rob0 | Keep smoking without the carbon monoxide? |
| 00:34.34 | Qwell | rob0: basically |
| 00:34.34 | seanbright | don't assume |
| 00:34.36 | seanbright | damn. |
| 00:34.38 | seanbright | :) |
| 00:34.48 | seanbright | how long have you smoked? |
| 00:34.58 | Qwell | ~13 years |
| 00:35.00 | seanbright | wow |
| 00:35.03 | seanbright | same as me |
| 00:35.04 | seanbright | weird. |
| 00:35.10 | rob0 | I didn't assume, see, the assumption ended with one of these thingies ---> ? :) |
| 00:35.21 | seanbright | rob0: touche |
| 00:37.20 | *** join/#asterisk otomoto (n=aoun@gateway3.aleks.com) |
| 00:37.53 | otomoto | hello |
| 00:38.09 | otomoto | can we connect 2 TE122 cards together via a cable? |
| 00:39.57 | iratik | sulks |
| 00:42.08 | Micc | anyone use any China Roby products? |
| 00:43.55 | Qwell | otomoto: sure |
| 00:45.22 | *** join/#asterisk gramulhaozin (n=user@office.xtremeon.com) |
| 00:45.24 | gramulhaozin | Hey guys |
| 00:45.32 | gramulhaozin | anyone ever had trouble with VONAGE giving DOUBLE DTMF ? |
| 00:47.27 | MaliutaLap | gramulhaozin: is that 2 for the price of one? ;) |
| 00:48.48 | KyleK | why would vonage give you dtmf tones |
| 00:48.49 | gramulhaozin | gramulhaozin yep |
| 00:49.01 | gramulhaozin | gramulhaozin One I press 1 I receive 11 |
| 00:49.08 | gramulhaozin | but that only happens with one number |
| 00:49.13 | gramulhaozin | one did |
| 00:49.22 | gramulhaozin | I mean, 1 DID has problem with Double DTMF |
| 00:49.39 | KyleK | I'd move to see your keyboard |
| 00:51.42 | pdmmm | are u using dtmf inband? |
| 00:52.48 | KyleK | gramulhaozin: problem like that sounds like it'd be caused by sending dialed numbers in and out of band at the same time |
| 00:54.33 | Micc | http://www.chinaroby.com/ |
| 00:54.37 | Micc | $19 ata? |
| 00:58.18 | f0ner00t | Qwell : I am so confused right now.. |
| 00:58.28 | f0ner00t | Anybody here with setting up sip using IPKALL? |
| 00:59.14 | rob0 | Micc, where do you see prices? |
| 00:59.29 | rob0 | Sure, I have several ipkall numbers. |
| 00:59.38 | f0ner00t | robo PM please? |
| 00:59.50 | rob0 | Why not in channel? |
| 01:00.06 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-127d5416eb1268ad) |
| 01:00.21 | rob0 | You can PM, but I bet your question belongs in the channel. |
| 01:00.49 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 01:02.13 | f0ner00t | rob0: http://pastebin.com/d2f996811 What is wrong with this sip setup? |
| 01:02.26 | f0ner00t | It does your right. |
| 01:03.31 | f0ner00t | Hmmp |
| 01:04.07 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 01:04.07 | *** join/#asterisk ScriptFanix (i=vincent@cl-54.mrs-01.fr.sixxs.net) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 01:04.07 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fec0:3dc0) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk crunge (n=Crunge@dsl093-034-021.snd1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk styelz (n=yoohoo@jbd.broker.freenet6.net) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk inckie (i=kimse1@please.press.control-alt-delete.eu) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk DGTL_Magician (n=boerg@siona.servers.nosco-ict.nl) |
| 01:04.07 | *** join/#asterisk Kayin (i=mverstee@shell2.skyberate.net) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) [NETSPLIT VICTIM] |
| 01:04.07 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 01:04.07 | *** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) [NETSPLIT VICTIM] |
| 01:04.07 | *** mode/#asterisk [+o denon] by irc.freenode.net |
| 01:04.41 | rob0 | I think the register belongs in in the global section. |
| 01:05.39 | rob0 | s/global/general/ |
| 01:06.02 | rob0 | Oh, you don't register for ipkall |
| 01:06.15 | f0ner00t | I thought you needed to register. |
| 01:06.19 | rob0 | did you give them a hostname that resolves to your IP? |
| 01:07.44 | f0ner00t | arbelosservices.com |
| 01:08.14 | rob0 | and no firewall issues, that SIP gets to your * ? |
| 01:08.55 | f0ner00t | Hmmp Yea that might be it too .. But it was working before I changed stuff. I'm getting Sip/2.0 404 Not FOund. |
| 01:08.56 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 01:09.18 | rob0 | the SIP comes in and your dialplan rejects it? |
| 01:09.41 | f0ner00t | I'll give you an whole sip captur. |
| 01:09.43 | f0ner00t | Hold on. |
| 01:10.37 | f0ner00t | http://pastebin.com/d5786bf89 |
| 01:14.16 | rob0 | I think "2537531745" should be the name of your section in sip.conf, but I am no expert. |
| 01:14.47 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:15.07 | rob0 | I know I gave them (ipkall) a 4-digit extension@my.hostname |
| 01:15.26 | rob0 | and that 4-digit extension is the name of my section in sip.conf |
| 01:15.31 | f0ner00t | SIP/2.0 403 Forbidden |
| 01:16.43 | otomoto | Qwell: reagrding connection 2 TE122 cards with a cable.... have you done it? |
| 01:16.57 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:18.55 | f0ner00t | Hmmp Sip not found. |
| 01:18.59 | *** join/#asterisk lwizardl (n=James@c-67-177-142-128.hsd1.mi.comcast.net) |
| 01:19.01 | lwizardl | hi |
| 01:19.35 | f0ner00t | would it has to be 1745@arbelosservices.com ossibly? |
| 01:20.03 | f0ner00t | sip:1745@arbelosservices.com |
| 01:21.10 | lwizardl | I was wondering if I could use a asterisk pbx to work for a traditional lanline phone and playback hold music, and a phone menu? |
| 01:21.34 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:24.26 | rob0 | "To: <sip:2537531745@arbelosservices.com>", that's from your SIP header |
| 01:29.07 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 01:30.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:30.50 | f0ner00t | Yea.. I know.. LOL.. Just was playing with it. |
| 01:30.58 | f0ner00t | I think I might have a LAN issue. |
| 01:31.09 | Aiatek | how many question do i have to answer correct to past the write exam of the dcaap cert? |
| 01:34.44 | kc8pxy | Aiatek: enough??:) |
| 01:35.24 | Aiatek | waooo, very explicity |
| 01:35.48 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 01:35.51 | kc8pxy | is ALSO not Dcap certified (yet) |
| 01:39.31 | rob0 | Looking for 2537531745 in incoming (domain arbelosservices.com) <--- I think you might need an extension 2537531745 in [incoming] context ( extensions.conf ) |
| 01:40.07 | rob0 | I hope one of the numerous experts around here will look in and confirm or correct that. |
| 01:41.15 | gramulhaozin | hey guys |
| 01:41.19 | gramulhaozin | anyone using vonage there ? |
| 01:42.43 | kc8pxy | is adhersion a pretty good standard for a gui admin tool? |
| 01:43.08 | drmessano | Anyone have a good TDMoE modem driver for windows? Heh |
| 01:45.37 | f0ner00t | drmessano: Welcome back! |
| 01:46.46 | f0ner00t | gramulhaozin: You signup for their softphone ? |
| 01:47.06 | f0ner00t | drmessano: Sip/2.0 404 Not Found.. Please help |
| 01:47.23 | f0ner00t | rob0 I put an context=incoming in my extensions.conf |
| 01:49.06 | f0ner00t | r0b0: I'm getting the SIP/2.0 not found as soon as I reload dialplan. |
| 01:49.54 | rob0 | Does the [incoming] context have an extension, 2537531745? |
| 01:50.00 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 01:50.01 | rob0 | or s, maybe |
| 01:54.58 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 02:00.11 | f0ner00t | Same thing. |
| 02:02.22 | f0ner00t | Ohwell beback tommorrow Have a good night! |
| 02:07.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:08.15 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:11.33 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
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| 02:29.32 | *** join/#asterisk astassistant (n=support@h-72-244-204-143.sttnwaho.dynamic.covad.net) |
| 02:31.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:44.07 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:28.18 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:31.31 | drmessano | throws REO Speedwagon LPs at jaytee |
| 03:31.50 | jaytee | scoops them up |
| 03:32.43 | jaytee | realizes he hasn't had a working turntable since 1993. |
| 03:33.35 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:36.19 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 03:41.40 | *** join/#asterisk blkry (n=blkry@24-241-112-018.dhcp.gnvl.sc.charter.com) |
| 03:43.12 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:43.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:47.30 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 03:51.28 | drmessano | lol |
| 03:51.37 | drmessano | throws some Laserdisc's at jaytee |
| 03:52.27 | jaytee | my buddy Vic actually has a working laserdisc player and a large collection of laserdiscs |
| 03:52.45 | jaytee | and why are you throwing things? is this a tantrum? |
| 03:54.03 | drmessano | Cleaning out the closet.. Need room for all my POTS phones |
| 03:54.10 | drmessano | Out with the old, in with the old |
| 03:54.21 | eppigy | hello |
| 03:54.24 | eppigy | i am dave |
| 03:54.48 | drmessano | This tendinitis is gonna piss me off |
| 03:54.56 | drmessano | I think its this goddamn mouse |
| 03:56.14 | drmessano | I used to think an MS Mouse was the best out there |
| 03:56.26 | drmessano | Now they're as useless as umm... something useless |
| 04:13.20 | leifmadsen | dances |
| 04:15.09 | drmessano | dances "pain pill time" |
| 04:15.40 | leifmadsen | that reminds me... I could use a drink... |
| 04:15.44 | leifmadsen | although it is already 12:15am |
| 04:16.20 | drmessano | Tweetdeck is freaking me out |
| 04:16.29 | leifmadsen | oh? |
| 04:16.39 | leifmadsen | I just tried it for the first time today; I currently use twitterfox |
| 04:16.43 | drmessano | These noises it makes |
| 04:16.55 | drmessano | IM BEING INVADED BY BIRDS |
| 04:17.28 | drmessano | Ive been using a Pidgin plugin for Tweeting |
| 04:17.39 | drmessano | But Tweetdeck is growing on me |
| 04:19.05 | leifmadsen | my kitchen is finally painted! |
| 04:19.11 | leifmadsen | 2nd coat is now drying |
| 04:19.23 | leifmadsen | I can finally put most things back into the cupboards |
| 04:19.27 | drmessano | heh |
| 04:20.09 | drmessano | At least it was for painting and not bug bombing |
| 04:20.23 | leifmadsen | ya, not too many bugs on the 34th floor |
| 04:20.36 | drmessano | lol |
| 04:20.48 | drmessano | I had some nasty neighbors |
| 04:21.02 | drmessano | I heard that walls had to be ripped out when they left |
| 04:21.07 | drmessano | :/ |
| 04:21.49 | leifmadsen | ouch |
| 04:24.22 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 04:27.42 | *** join/#asterisk MIR100 (n=mir100@24-148-89-37.frg-bsr1.chi-frg.il.cable.rcn.com) |
| 04:28.29 | ricko73 | drmessano: I had some tennants like that |
| 04:28.45 | ricko73 | they snuck in a few dogs |
| 04:29.25 | ricko73 | I ended up tearing out all the carpet and even some of the baseboards |
| 04:29.44 | drmessano | Nice |
| 04:29.56 | ricko73 | so who's gonna give that guy a cheap cheap ata? lol |
| 04:30.10 | ricko73 | ata by Campbells and yarn? |
| 04:34.22 | drmessano | lol |
| 04:35.44 | leifmadsen | nice |
| 04:38.58 | drmessano | rocks out to some Warlock |
| 04:44.04 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-ec7361cf0098a98c) |
| 04:46.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 05:08.50 | kc8pxy | YAY! i have my first REAL sip to sip call :) |
| 05:10.19 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 05:13.01 | *** join/#asterisk edoceo (n=edoceo@c-98-247-254-241.hsd1.wa.comcast.net) |
| 05:19.48 | leifmadsen | kc8pxy: congrats :) |
| 05:19.53 | leifmadsen | it's pretty exciting huh? :) |
| 05:20.48 | kc8pxy | leifmadsen: this lappy is foreign to me, so half the battle was with my softphone and headset.. but it's nice. |
| 05:21.39 | kc8pxy | leifmadsen: now i just need to work on getting outside clients in to the asterisk.. |
| 05:21.48 | drmessano | The first time I set up Asterisk, I was using an old Linksys ATA |
| 05:21.50 | kc8pxy | leifmadsen: what issues shoudl i expect with that? |
| 05:21.59 | drmessano | and my friend had a softphone |
| 05:22.06 | drmessano | We made ONE call |
| 05:22.19 | drmessano | and I was farting around.. and somehow he couldnt call me after that |
| 05:22.32 | drmessano | Reinstalled, twice |
| 05:22.41 | rob0 | And now he's not your friend anymore? |
| 05:23.03 | rob0 | Friends don't let friends use softphones. |
| 05:23.07 | drmessano | Found out after much cursing the feature codes in the ATA were overridding *, and callblock was set up.. for his extension |
| 05:23.28 | drmessano | Good times |
| 05:23.48 | EmleyMoor | My first experiments with * were all with softphones. The box was too slow but I had a plan. I bought a TDM400P card and converted one of my less useful machines into my * box. I have never looked back |
| 05:23.50 | drmessano | kc8pxy: ~sipnat |
| 05:23.51 | kc8pxy | drmessano: heheh |
| 05:24.06 | drmessano | ~sipnat |
| 05:24.07 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:24.18 | drmessano | That will help you with the outside config |
| 05:27.35 | leifmadsen | kc8pxy: not too many, except perhaps NAT issues |
| 05:27.49 | leifmadsen | heh, I didn't read the scrollback, but looks like drmessano got ya :) |
| 05:28.05 | leifmadsen | TFoT v3 will deal extensively with NAT scenarios |
| 05:28.49 | drmessano | Thank <diety> |
| 05:29.07 | drmessano | err |
| 05:29.12 | drmessano | Thank <deity> |
| 05:29.13 | drmessano | !!! |
| 05:29.25 | drmessano | NAT is the #1 problem I see |
| 05:29.25 | leifmadsen | heh |
| 05:29.32 | leifmadsen | yep, and it's not really that hard to get around |
| 05:29.45 | leifmadsen | you just need to know how to configure asterisk and your NAT device (where appropriate) |
| 05:29.49 | EmleyMoor | leifmadsen: When is TFoT v3 due? |
| 05:29.57 | drmessano | No, once you understand whats being asked for, its cake |
| 05:29.58 | leifmadsen | long time from now |
| 05:30.03 | leifmadsen | I'd say at least 8-12 months |
| 05:30.09 | leifmadsen | we have barely even started |
| 05:30.26 | leifmadsen | however, I'm working on getting time allocated to allow me to write documentation about 5-7 hours a week |
| 05:30.36 | leifmadsen | which will significantly speed up the process |
| 05:30.52 | drmessano | TFOT in 3 weeks then |
| 05:30.54 | drmessano | Cool |
| 05:30.57 | leifmadsen | right |
| 05:30.58 | leifmadsen | :) |
| 05:31.05 | drmessano | BRB, preorder |
| 05:31.30 | leifmadsen | at least 3 months of writing, followed by at least that amount of time doing the editing and layout at O'Reilly, followed by 2 months to get it printed and distributed |
| 05:31.43 | leifmadsen | drmessano: pre-order directly from me |
| 05:32.23 | drmessano | Should I just type my credit card number in channel? |
| 05:32.34 | leifmadsen | sure! |
| 05:32.37 | drmessano | Sorry, new to this |
| 05:32.39 | leifmadsen | then we can all benefit |
| 05:32.47 | denon | don't forget the CVV2 |
| 05:32.56 | kc8pxy | disallow and allow, why do they narrow me into ulaw only on the first link? |
| 05:33.25 | drmessano | Thats a GUIDE |
| 05:33.32 | drmessano | You need to interpret and use |
| 05:33.41 | kc8pxy | i kno |
| 05:34.00 | leifmadsen | eh? |
| 05:34.03 | rob0 | 4589 1750 8622 2881 cvv 173 exp 03/11 |
| 05:34.09 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 05:34.23 | leifmadsen | hawt |
| 05:34.34 | leifmadsen | the funny thing is that might be a real CC, but no one will act on it :) |
| 05:34.50 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:34.55 | leifmadsen | oej: go back to bed! |
| 05:34.55 | jplank | i'd like to send in a preorder too (just in case) |
| 05:34.55 | jplank | 4589 1750 8622 2881 cvv 173 exp 03/11 |
| 05:35.02 | kc8pxy | I know i could narrow in on some other codec. is that generally a good idea? |
| 05:35.17 | leifmadsen | kc8pxy: use whatever codec you want, and which your phone supports |
| 05:35.29 | drmessano | Me too |
| 05:35.29 | leifmadsen | gsm is popular if bandwidth is an issue, but ulaw sounds better |
| 05:35.30 | drmessano | 4589 1750 8622 2881 cvv 173 exp 03/11 |
| 05:35.47 | oej | leifmadsen: Go directly to bed! |
| 05:35.47 | drmessano | leifmadsen: I think he wants to know why ULAW is specified in that NAT guide |
| 05:35.51 | drmessano | There is not reason |
| 05:35.56 | drmessano | Its a sample config |
| 05:36.02 | drmessano | no* |
| 05:36.08 | leifmadsen | because ulaw is the most popular, that's the only reason |
| 05:36.11 | jplank | isn't it because ULAW is just totally awesome |
| 05:36.11 | kc8pxy | what's the bw use on gsm? |
| 05:36.16 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 05:36.22 | leifmadsen | oej: I still have vodka to drink, and a living room to finish cleaning |
| 05:36.22 | EmleyMoor | Will 1.6 support "Dial Gosubs"? |
| 05:36.24 | drmessano | If [TK]D-Fender put allow=g729 in there, people would bitch he broke their NAT |
| 05:36.26 | drmessano | ETC |
| 05:36.30 | florz | alaw certainly is more popular ;-) |
| 05:36.34 | leifmadsen | EmleyMoor: yes, GoSub's are the new Macro :) |
| 05:36.37 | oej | leifmadsen: Done with painting then? |
| 05:36.41 | leifmadsen | florz: europeans suck! :) |
| 05:36.54 | leifmadsen | oej: I just finished about an hour ago! Just putting some things away |
| 05:37.02 | florz | =:-) |
| 05:37.03 | leifmadsen | Kristian Kielhofner visits tomorrow for the weekend |
| 05:37.41 | drmessano | alaw is unamerican! |
| 05:37.47 | drmessano | I mean, it is |
| 05:37.47 | leifmadsen | EmleyMoor: elaborate? I use GoSub's in place of Macro, and you can trigger a GoSub from Dial() using the... G() flag I believe (just like M() is for macro) |
| 05:37.57 | leifmadsen | american law |
| 05:37.59 | oej | Oh, yes. Kristian is known to be picky. Did you double-check the bathroom too? You don't want to have him upset. |
| 05:38.11 | EmleyMoor | leifmadsen: Have already converted all my no parameters non-dial macros to Gosubs... but have some that take (either implicit or explicit) parameters and some that are Dial-called |
| 05:38.14 | leifmadsen | oej: he'll take what he gets and he'll love it :) |
| 05:38.31 | oej | leifmadsen: ...and that's an order! |
| 05:38.33 | leifmadsen | EmleyMoor: not sure I understand what you mean, but I've had a couple vodkas |
| 05:38.38 | leifmadsen | and it's 1:38am |
| 05:38.51 | EmleyMoor | leifmadsen: In 1.4 I don't think Gosub parameters or Dial Gosubs are supported |
| 05:38.51 | leifmadsen | oej: oic... hmmm... well I cleaned the shower, but I guess that means I should clean the rest :) |
| 05:39.14 | leifmadsen | EmleyMoor: no, parameters to goSub are only 1.6, or 1.4 with the backport from svncommunity |
| 05:39.20 | oej | leifmadsen: Have another drink and stop bothering... :-) |
| 05:39.24 | kc8pxy | wishes he knew why his Voicemail() doesn't work :-( |
| 05:39.25 | leifmadsen | oej: you're the one :) |
| 05:39.41 | leifmadsen | kc8pxy: use pastebin to show the configuration you're using along with the error you're seeing on the console |
| 05:39.42 | EmleyMoor | G is for "Goto" according to what I read |
| 05:39.55 | leifmadsen | EmleyMoor: quite possible, I can't remember all the Dial() flags :) |
| 05:40.44 | kc8pxy | leifmadsen: no error. just slid through and went answer(),voicemail(),hangup(), and i never heard anything. |
| 05:41.00 | EmleyMoor | Macros may well be deprecated in 1.6 but at least they will still work while I rewrite. |
| 05:41.08 | leifmadsen | kc8pxy: probably means you haven't configured voicemail.conf, or have not passed the correct values to it |
| 05:41.40 | leifmadsen | EmleyMoor: yes, it's not so much they are deprecated, but that they use so much stack space that you can only nest them 7 levels deep, and that presumes you're not using ODBC |
| 05:41.47 | leifmadsen | func_odbc specifically |
| 05:41.47 | kc8pxy | leifmadsen: email and such are not REQUIRED, are they? |
| 05:41.57 | leifmadsen | I'm pretty much the reason GoSub() exists |
| 05:42.21 | leifmadsen | kc8pxy: no, you should be able to just do Voicemail() and get asked for the extension you want to login to the default VM context |
| 05:42.24 | leifmadsen | VM == voicemail |
| 05:42.48 | EmleyMoor | I don't think I ever go deeper that 2 anyway |
| 05:42.56 | leifmadsen | EmleyMoor: or example, if you have: Macro(Macro(func_odbc)) (not literal), then asterisk will crash because it runs out of stack space |
| 05:43.06 | leifmadsen | so you can't have a nested macro call func_odbc |
| 05:43.12 | leifmadsen | with GoSub(), it's a non-issue |
| 05:44.15 | kc8pxy | leifmadsen: so gosub is a code-cleaning of the old macro coding? |
| 05:45.13 | leifmadsen | GoSub() is another approach to subroutines |
| 05:45.29 | leifmadsen | it uses less memory because its essentially a Goto() that remembers where it came from |
| 05:45.42 | leifmadsen | so you have to explicitly call Return() to go back |
| 05:45.49 | EmleyMoor | Familiar concept to those of us who used BASIC |
| 05:46.04 | drmessano | 10 Print "Hello World" doesnt work |
| 05:46.14 | drmessano | I want C-64 Basic in my dialplan |
| 05:46.19 | kc8pxy | leifmadsen: so asterisk dialplans are edging towards "real" coding?? |
| 05:46.26 | drmessano | edging? |
| 05:46.29 | leifmadsen | kc8pxy: not at all :) |
| 05:46.30 | drmessano | They're there |
| 05:46.35 | drmessano | heh |
| 05:46.35 | kc8pxy | ok |
| 05:46.38 | kc8pxy | hehe |
| 05:46.51 | drmessano | Writing asterisk dialplan is a language in itself |
| 05:46.55 | leifmadsen | dialplan does work a lot like basic because you have goto's and it's linear |
| 05:46.58 | kc8pxy | is a hobbyist x86 asm coder.. :) |
| 05:46.59 | EmleyMoor | I was thinking of trying to rewrite my dialplan in AEL |
| 05:47.16 | leifmadsen | I gave up on AEL after finding issues about 10 mins into it |
| 05:47.20 | leifmadsen | although I'm sure it has matured a lot since then |
| 05:47.27 | denon | EmleyMoor: just so asterisk can rewruite it back? |
| 05:47.34 | leifmadsen | but I'm so good at dialplan now, that I haven't bothered to try anything else |
| 05:47.43 | kc8pxy | AEL == ?? |
| 05:47.55 | leifmadsen | ~ael |
| 05:47.56 | infobot | hmm... ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 05:48.11 | leifmadsen | kc8pxy: AEL is just a way to write dialplan in a different format |
| 05:48.15 | drmessano | I see/saw AEL as an attempt to come up with higher level dialplan language for Asterisk when really, Asterisk rocks because of the diaplan language |
| 05:48.18 | leifmadsen | but it "compiles" back into dialplan |
| 05:48.21 | EmleyMoor | denon: Partly - but partly to save me from tangled extens like I found earlier in the week. |
| 05:48.36 | drmessano | Asterisk doesnt need ruby, perl, or lolcat dialplans |
| 05:48.38 | leifmadsen | drmessano: AEL is useful because it can save your wrists from a lot of typing |
| 05:48.43 | denon | EmleyMoor: you do understand that AEL gets converted back to extensions.conf format though, right? |
| 05:48.52 | EmleyMoor | Yes |
| 05:48.53 | leifmadsen | internaly |
| 05:48.58 | drmessano | leifmadsen: Wrists are shot anyway.. bring me the dialplan |
| 05:49.04 | leifmadsen | drmessano: amen |
| 05:49.18 | denon | *shrugs* extensions.conf is easy |
| 05:49.23 | denon | why bother |
| 05:49.23 | leifmadsen | I currently have a 1600+ dialplan for a customer |
| 05:49.30 | leifmadsen | and none of it is generated automatically |
| 05:49.43 | leifmadsen | denon: that's why I use extensions.conf |
| 05:49.54 | leifmadsen | plus I've gotten real good with dialplan functions |
| 05:50.02 | leifmadsen | especially nesting them with IF() |
| 05:50.39 | leifmadsen | exten => _X.,n,Voicemail(${MY_EXTENSION}@${MY_CONTEXT},${IF($[${DIALSTATUS} = BUSY]?b:u)}) |
| 05:51.00 | denon | now that's just ugly :) |
| 05:51.06 | leifmadsen | most people would use a Goto() there and have 3 lines of dialplan that could be put into 1 |
| 05:51.19 | leifmadsen | denon: that's nothing :) to me that's perfectly readable |
| 05:51.43 | leifmadsen | I have lines that end up being about 400 chars long with nested functions.... |
| 05:51.47 | denon | dunno, I dont mind something being a bit more verbose, especially if it lives in a macro anyway |
| 05:52.19 | leifmadsen | the only reason that happens though is because you can't use a Goto() in the 'h' exten without modifying your CDRs |
| 05:52.38 | drmessano | File a bug report |
| 05:52.44 | leifmadsen | it's not a bug |
| 05:52.52 | leifmadsen | it's just how it works |
| 05:52.52 | drmessano | I know.. Was .. nevermind |
| 05:53.07 | denon | submit a patch!!! |
| 05:53.08 | kc8pxy | woudl like a syntax filter for dialplan in bluefish :) |
| 05:53.10 | denon | snickers |
| 05:53.34 | denon | I'd like it to be in LOGO |
| 05:53.51 | drmessano | Im too busy writing patches for FreePBX.. most of them being dialplan functions I refuse to just override |
| 05:54.27 | denon | ah well, I'm out |
| 05:54.29 | denon | g'nite gents |
| 05:54.45 | leifmadsen | night |
| 05:54.54 | leifmadsen | would love dialplan via logo |
| 05:55.22 | drmessano | HA! |
| 05:55.25 | drmessano | Logo FTW |
| 05:56.22 | drmessano | FORWARD 90 |
| 05:56.25 | drmessano | RIGHT 90 |
| 05:56.32 | kc8pxy | dialplan in flowgraph :) THAT woudl be cool :) |
| 05:57.02 | drmessano | REPEAT 10 [ FORWARD 50 RIGHT 50 ] |
| 05:57.29 | kc8pxy | decagon? |
| 05:57.57 | kc8pxy | anyone here good with flowgraph? |
| 05:57.59 | drmessano | No clue, I remember the syntax.. |
| 05:58.10 | drmessano | 10 Print "Asterisk 1.6" |
| 05:58.14 | drmessano | 20 Goto 10 |
| 05:58.31 | drmessano | Porting complete! |
| 05:58.41 | kc8pxy | hehe |
| 05:59.18 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:02.25 | jblack | drmessano: You suck! |
| 06:03.16 | drmessano | I do? |
| 06:03.18 | jblack | gets a mischievous look |
| 06:03.46 | jblack | Yeah. If only for the reason that grass is green. |
| 06:04.13 | drmessano | throws barbie sized chairs at jblack |
| 06:04.37 | jblack | throws pokeman dolls at drmessano |
| 06:05.09 | drmessano | My BarbieLinux crashed earlier |
| 06:05.14 | oej | leifmadsen: http://www.thedailyshow.com/video/index.jhtml?videoId=225113&title=the-stockholm-syndrome |
| 06:05.18 | drmessano | I had to boot into Ken and fix the kernel |
| 06:05.21 | jblack | forgot to change the D batteries? |
| 06:06.03 | jblack | I think the chantix is starting to work. I've been up for 30 min, had two smokes, and I feel ike I still need one. |
| 06:06.32 | *** join/#asterisk Gopher_77 (n=Jim@cpe-174-100-98-181.neo.res.rr.com) |
| 06:06.45 | jblack | I was gonna distract myself with a fight.. but barbie? How lame! |
| 06:06.46 | drmessano | Weird.. Isnt chantix made my RJR Tobacco, inc? |
| 06:06.57 | drmessano | by* |
| 06:07.02 | jblack | Phizer, but all those large companies own each other, right? |
| 06:07.19 | leifmadsen | oej: nice, I'm in Canada, so I can't view it |
| 06:07.38 | jblack | In some sort of unholy kentuckian inbred stock swap way. |
| 06:07.54 | Gopher_77 | I'm having trouble getting * to "playback". I can put in the full path or exclude the path, and - the .gsm extension, and I get a "ast_streamfile failed" error. Is there something I'm missing? |
| 06:07.55 | oej | leifmadsen: How can I view it and not you? |
| 06:07.59 | drmessano | jblack: Take viagra.. It will make you forget about the smoking |
| 06:08.16 | jblack | man, that's brilliant! |
| 06:08.22 | leifmadsen | oej: because we have comedy central here who is the distributor I guess |
| 06:08.37 | [TK]D-Fender | o>O |
| 06:08.43 | leifmadsen | Gopher_77: do not use the extension |
| 06:08.53 | Gopher_77 | leifmadsen: I'm not using an extension |
| 06:09.26 | EmleyMoor | What's .gsm then? |
| 06:09.38 | Gopher_77 | EmleyMoor: it's what I'm not using |
| 06:09.39 | leifmadsen | he meant minus the .gsm |
| 06:09.44 | EmleyMoor | Ah |
| 06:09.58 | leifmadsen | hates when people don't just type the f'n word |
| 06:10.24 | EmleyMoor | I don't mind as long as I can understand it |
| 06:10.35 | leifmadsen | ok, bed time! :) |
| 06:10.50 | Gopher_77 | just one more! |
| 06:10.54 | Gopher_77 | lol |
| 06:11.51 | Gopher_77 | here's the command from CLI: originate dahdi/4 application PlayBack (/var/lib/asterisk/sounds/vm-youhave) |
| 06:12.51 | Gopher_77 | phone rings, pick up and it gives the error and a busy signal |
| 06:13.59 | Gopher_77 | doh, I just tried taking the () off... it seems to work |
| 06:14.34 | EmleyMoor | Gopher_77: Have you tried without the space? |
| 06:15.20 | Gopher_77 | EmleyMoor: ah, seems to work fine without the () |
| 06:15.35 | [TK]D-Fender | calling that from CLI = no braces. Sorta like the instructsions say |
| 06:15.37 | Gopher_77 | EmleyMoor: I thought wherever I've seen tutorials it has () around the filename |
| 06:15.53 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 06:15.56 | EmleyMoor | I have lots of Playback, with () but without a space in it |
| 06:15.57 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-341e5095fdd1422b) |
| 06:16.10 | Gopher_77 | [TK]D-Fender: so they're required for extensions.conf but not cli? |
| 06:16.13 | EmleyMoor | Ah, I see |
| 06:16.28 | Gopher_77 | [TK]D-Fender: thanks |
| 06:16.34 | EmleyMoor | (CLI - never done it from there) |
| 06:17.04 | [TK]D-Fender | and FFS stop trying to use CLI to test calls, etc. Get a softphone and set up a proper dialplan |
| 06:17.27 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 06:19.44 | drmessano | jblack: |
| 06:19.45 | drmessano | http://2.bp.blogspot.com/_2eI4lUPyBjM/SEAjxGWDmCI/AAAAAAAAAHo/VvQzylKy7q8/s400/Thanks+Chantix.jpg |
| 06:20.13 | jblack | heh |
| 06:21.13 | [TK]D-Fender | And on that note.. its checkout time. later all |
| 06:21.18 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
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| 06:56.41 | arekm | hi, does asterisk have soem escapeshellcharacters() type of function? So MixMonitor() will be able to save to files even with "/" in name? |
| 07:00.39 | jblack | is not a valid filename character in *nix |
| 07:00.47 | jblack | its a path seperator. |
| 07:10.34 | arekm | oh, URIENCODE() will be good for my needs |
| 07:17.31 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 07:19.27 | arekm | but some plain string replace function would be better |
| 07:28.28 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
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| 07:31.55 | jeffspeff | i've just compiled and installed asterisk on centos... it's been a while since i've used it, and when i try to run 'asterisk -r' i get the error 'Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)' What am i forgetting to do? |
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| 07:37.34 | baliktad | have you actually started asterisk again? |
| 07:39.46 | creativx | asterisk before asterisk -r hehe |
| 07:43.40 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
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| 08:06.12 | EmleyMoor | Someone, knowing I use festival with asterisk, asked me "How is festerisk now?" - I was mildly amused |
| 08:08.31 | gego | when I start * by init it's running according to ps but I still can't log in with asterisk -r |
| 08:09.10 | gego | it's set up with AST_USER and AST_GROUP = root and nothings touched in /var/run |
| 08:10.20 | gego | what's the difference to manually starting /etc/init.d/asterisk start ? |
| 08:12.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-1ce30b2ad7c6446c) |
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| 08:16.12 | MrParity | hi ho :-) |
| 08:17.29 | drmessano | 8,n,1 |
| 08:18.46 | MrParity | i have a problem with asterisk. i spend a long time with google, but i didn't find a solution. my first question is: waht is a overlap voice call? |
| 08:26.21 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 08:29.46 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
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| 08:53.21 | jeffspeff | i've got my new asterisk box accepting calls now, but I can't find where to disable the sample IVR |
| 08:53.52 | hi365 | jeffspeff: /etc/asterisk/extensions.conf |
| 08:54.45 | jeffspeff | what's the command for reloading extensions.conf ? |
| 08:55.00 | hi365 | its a text file |
| 08:55.43 | jeffspeff | yes, but like when you edit sip.conf you have to do a 'sip reload' for changes to take effect... same for extensions.conf ? |
| 08:55.56 | hi365 | dialplan reload |
| 08:56.12 | jeffspeff | ty |
| 09:03.22 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
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| 09:28.52 | proxium | Hi all, I didn't find [macro-dialout-trunk] context in my extensions.cons but I found [macro-dialout-trunk-predial-hook] are they similiar? |
| 09:29.10 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 09:32.14 | jeffspeff | refresh me, what is _X for again? |
| 09:32.57 | JT | matches 1 digital in the range of 0 to 9 |
| 09:33.05 | JT | s/digital/digit/ |
| 09:34.26 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 09:36.43 | jeffspeff | i copied my extensions.conf and sip.conf to my setup using 1.6.x and i'm getting an error on inbound calls about 'can't find extension 's'' i think i've traced it... can you start a context with _X ? |
| 09:37.26 | jeffspeff | my old system used 1.4.x |
| 09:42.19 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:43.51 | freh | jeffspeff, I believe you can |
| 09:44.30 | jeffspeff | i can't figure out what the hell is wrong... would you mind taking a look? |
| 09:44.40 | freh | but that would be for an extension with only one digit |
| 09:45.00 | freh | for any extension use _X. |
| 09:46.08 | freh | but do be carefull with how you use . |
| 09:47.35 | jeffspeff | extensions.conf http://pastebin.ca/1401719 sip.conf http://pastebin.ca/1401722 |
| 09:52.37 | freh | In which context do you inbound calls end up? |
| 09:54.11 | jeffspeff | don't hate me, but i don't remember... i used to mess with * a lot. then i slacked off for about a year or more and have forgotten more than i realized |
| 09:54.28 | jeffspeff | too many bottles i guess... :( |
| 09:54.57 | freh | where do inbound calls come from? |
| 09:55.32 | jeffspeff | sip.conf teliax i believe |
| 09:57.24 | jeffspeff | ok, i've found my error |
| 09:57.59 | jeffspeff | in SIP/teliax i had my host set incorrectly... i knew it was something stupid i was missing |
| 09:58.21 | freh | =) |
| 10:06.05 | jeffspeff | freh, has the Goto command changed? when using 'exten => s,1,Goto(voicemenu-custom-5|s|1)' in the 'default' context of extensions.conf i get an error on inbound call saying 'No such label 'voicemenu-custom-5....'' |
| 10:09.13 | jeffspeff | nvm... i'll worry about it later... i gotta get to bed. it's 5am here. |
| 10:09.20 | jeffspeff | freh, thanks for your help |
| 10:10.04 | *** join/#asterisk WildyLion (n=simba@ve-nat.pleer.ru) |
| 10:19.21 | *** join/#asterisk MCCob (n=MCCob@bre91-1-82-238-224-20.fbx.proxad.net) |
| 10:19.43 | MCCob | helllo |
| 10:21.04 | MCCob | I have some problem loading wcb4xxp dahdi driver witch Junghanns QuadBRI ISDN card, aynone can help me ? dahdi_hardware show wcb4xxp- |
| 10:29.00 | *** join/#asterisk roflz (n=sunny@61.5.134.34) |
| 10:41.15 | freh | MCCob, is it uncommented in /etc/dahdi/modules ? |
| 10:41.16 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 10:42.31 | proxium | Where to fix the CallerID for sip phone? |
| 10:43.06 | MCCob | yes freh |
| 10:44.14 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:44.18 | freh | what is the output of dahdi_hardware? |
| 10:44.27 | roflz | how can i remove prefix (91) and add 0 example 913212138888. |
| 10:44.59 | MCCob | pci:0000:0d:04.0 wcb4xxp- 1397:08b4 Junghanns QuadBRI ISDN card |
| 10:44.59 | MCCob | <PROTECTED> |
| 10:45.24 | freh | roflz, use ${exten:2} to strip the first 2 numbers |
| 10:45.40 | freh | ${EXTEN:2} |
| 10:46.21 | roflz | thanx freh and how can i add 0 |
| 10:47.08 | freh | Don't exactly know how to prepend a 0 except for set(EXTEN=0${EXTEN}) |
| 10:47.52 | freh | to append a 0 use set(EXTEN=${EXTEN}0) |
| 10:48.33 | freh | MCCob, sorry, can't help you. Have no experience on the matter |
| 10:49.29 | freh | MCCob, does it work with dahdi_genconf? |
| 10:49.48 | MCCob | thx, no :( |
| 10:50.28 | roflz | exten => _X.,1,Dial(SIP/${EXTEN:2}0@6131812) plz check this do mean like this |
| 10:50.53 | roflz | exten => _X.,1,Dial(SIP/${EXTEN:2}0@6131812) freh "plz check this do mean like this" |
| 10:54.47 | roflz | it not working |
| 10:55.23 | roflz | it's not working |
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| 11:48.26 | Vec | If I do a Goto(context,555,1) and extension 555, does not exist in the context but I have an extension _X., will it match the _X. or will it go to extention 'i' ? |
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| 12:03.45 | EmleyMoor | Vec: It will follow _X. if there is no better match |
| 12:05.03 | freh | i seem to have asterisk dropping some calls |
| 12:05.29 | freh | I see this message on the console: |
| 12:05.30 | freh | Channel 0/1, span 2 got hangup request, cause 16 |
| 12:06.02 | freh | I'm using dahdi |
| 12:08.12 | freh | Calls come in on the dahdi channel. I see them going through the dialplan. going through a playback(). Then they are queued. Sometimes, when an agent answers the call, it's dropped with that message |
| 12:10.27 | freh | http://pastebin.com/d3d11d3fe |
| 12:10.40 | Vec | EmleyMoor : thanks |
| 12:11.02 | freh | This does not happen all the time. I did notice that the last call had no callerid |
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| 12:39.52 | harryv | http://www.voip-info.org/wiki/view/ITU+G.711 - "t takes more bandwidth then other codecs, up to 84 Kbps including all TCP/IP overhead." |
| 12:40.06 | harryv | is that one way? or is it total |
| 12:40.59 | desdesdesdes | i have a dual port pri connected diguim, one span connected to telco the other one connected to Siemens system, if i have a incoming call to pri from telko i route it to the second pri connected to the siemens, to extension 100 if i want to transfer a call from 100 to 101 on the siemens system it works fine, if i tansfer the call to an external call back out over the telkom, it rings the number and 100 can connect to the external call but it does no |
| 12:41.34 | [TK]D-Fender | harryv: per direction |
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| 12:42.57 | desdesdesdes | does that make sence |
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| 12:45.18 | [TK]D-Fender | desdesdesdes: right up to the point where your question got cut in half. |
| 12:45.22 | harryv | thanks |
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| 13:06.42 | Zeeek | beer? |
| 13:07.27 | creativx | meep |
| 13:07.43 | _ShrikE | yes please |
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| 13:10.36 | ricko73 | mmmmm, beeeer |
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| 13:21.31 | Kobaz | i have a little digium iaxy... and i tried to provision it with a static gateway (rather than getting it from dhcp)... now i can't provision it with anything... it's not getting dhcp either... how do you do a full reset? |
| 13:22.07 | freh | I'm having trouble getting call parking to work. my features.conf is at pastebin http://pastebin.com/d53097012 |
| 13:22.32 | freh | when I dial 700 during a call, nothing happens |
| 13:23.00 | EmleyMoor | freh: How are you dialling 700 during a call? |
| 13:23.27 | freh | I am including the parkedcalls context in the dialplan an using the tTkK options in the dial application |
| 13:23.39 | EmleyMoor | ... and... ? |
| 13:24.13 | EmleyMoor | How are you initiating the transfer? |
| 13:24.24 | freh | EmleyMoor, In fact, when I transfer the call to extension 700, it is indeed parked, but the position isn't read to me |
| 13:25.03 | EmleyMoor | Oh - so you can see in in the CLI but not hear it? |
| 13:26.14 | freh | No, It also isn't showing up in the cmdline. Just a sec, doing another pastebin |
| 13:27.09 | EmleyMoor | Once agai, how are you initiating the transfer? |
| 13:27.13 | EmleyMoor | again... |
| 13:27.25 | freh | with ## |
| 13:28.46 | [TK]D-Fender | freh: What phone are you trying to park with? |
| 13:28.49 | EmleyMoor | freh: With verbose 3, what shows in CLI weh you try? |
| 13:28.55 | freh | then I hear "transfer" and then dialtone. I dial 700, and it gets parked. Then it's hung up |
| 13:29.04 | [TK]D-Fender | freand you need to do an ATTENDED transfer to parking, not BLIND |
| 13:29.17 | [TK]D-Fender | freh: You are doing the wrong kind of transfer |
| 13:29.18 | *** join/#asterisk porche (n=kursad@81.215.117.205) |
| 13:29.23 | porche | hi all |
| 13:29.28 | freh | [TK]D-Fender, ok thanks |
| 13:29.39 | [TK]D-Fender | freh: Again, who kind of phone are you using? |
| 13:29.47 | freh | Zoiper softphone |
| 13:29.53 | porche | does anyone knows a service, who is providing direct PSTN connection? |
| 13:30.07 | [TK]D-Fender | freh: Stop using DTMF transfers, Zoiper support proper OOB transfers. |
| 13:30.26 | [TK]D-Fender | porche: Define "direct" |
| 13:30.31 | ricko73 | [TK]D-Fender: is there a way to do a 'one step' park, but still hear the parking location? |
| 13:30.48 | porche | need an asterisk box with zap channels |
| 13:30.52 | freh | Yes but I need it to work on other phone's that don't support it too |
| 13:30.53 | porche | in US or Can |
| 13:31.13 | Zeeek | who needs a car? no car, no parking |
| 13:31.17 | EmleyMoor | porche: That's a DIY job |
| 13:31.29 | [TK]D-Fender | ricko73: with DTMF transfers and a phone capable of doing better timed inband-speed-dials, yes |
| 13:31.31 | porche | :) yeah, if I was in US/Can |
| 13:32.00 | [TK]D-Fender | porche: Again, what do you mean by "direct"? |
| 13:32.05 | EmleyMoor | porche: Why do you need it? What do you want to do that a VoIP-to-PSTN service can't do? |
| 13:32.23 | porche | there is an ANI2 variable available in zap channels |
| 13:32.37 | porche | which defines the phone number type, |
| 13:33.17 | porche | this one: http://www.voip-info.org/wiki/view/ANI2 |
| 13:33.18 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173) |
| 13:33.41 | Kobaz | porche: build a server with a t1 card, put it in a colocation facility, and get a t1 line (around $250-350 a month) |
| 13:34.12 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:34.36 | porche | how much does it cost with t1? |
| 13:34.47 | Kobaz | porche: plus the cost of colocation, and bandwidth |
| 13:35.10 | porche | no known provider then |
| 13:35.12 | porche | got it |
| 13:35.18 | Kobaz | huh? |
| 13:35.35 | porche | i mean, there is no one providing this kind of service |
| 13:35.37 | Kobaz | search google for colocation, there's 5 zillion of them |
| 13:35.37 | *** join/#asterisk intralanman (n=lanman@67.76.163.226) |
| 13:35.49 | freh | EmleyMoor, This is what shows on the CLI with verbosity set to 3 and with an attended transfer: http://pastebin.com/d4a7b200c |
| 13:35.53 | porche | no, I mean the complete solution, |
| 13:35.54 | Kobaz | porche: you can get sip channels from any itsp |
| 13:36.01 | Kobaz | ~itsp-list |
| 13:36.02 | infobot | hmm... itsp-list is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 13:36.22 | porche | but the issue is, ANI is not available with SIP, isnt it? |
| 13:36.42 | Kobaz | depends on your provider, they may add an additiona header with ani |
| 13:37.07 | Kobaz | you want to set any, or recieve ani? |
| 13:37.14 | Kobaz | s/any/ani |
| 13:37.34 | porche | i want to detect, the called number type/ani |
| 13:37.48 | Kobaz | talk to itsp's to see if they give you ani |
| 13:37.50 | porche | so receive |
| 13:38.15 | Kobaz | either that, or buy a box with a t1 card, and host it somewhere |
| 13:38.32 | EmleyMoor | freh: Did it not announce the number? |
| 13:38.43 | freh | EmleyMoor, no it did not |
| 13:39.10 | EmleyMoor | Hmmm... no idea why not... |
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| 13:39.30 | porche | thank you kobaz |
| 13:39.43 | porche | thank you fender |
| 13:40.33 | freh | If I dial 700 directly, It does get read to me. But then I'm not in a call... |
| 13:40.52 | freh | I mean like when I dial 700 not during a call |
| 13:41.15 | ricko73 | freh: you need to TRANSFER the call to 700 |
| 13:41.30 | ricko73 | but do an attended transfer, not a blind transfer |
| 13:41.52 | freh | ricko73, Yes that's what I do now. But the parked extension is not being read to me by asterisk |
| 13:42.07 | ricko73 | no, you're doing a a BLIND transfer |
| 13:42.14 | ricko73 | if it was attended, you'd still be on the line |
| 13:42.23 | [TK]D-Fender | freh: you don't jsut DIAL. As you've been told repeatedly, you do an ATTENDED TRANSFER to an exten with the Park application, which you neev to be able to reach |
| 13:42.51 | freh | I am doing an attended transfer and it's still not being read to me |
| 13:43.04 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 13:43.22 | ricko73 | trust us, you're NOT doing attended. You may think you are, but that doesn't make it so |
| 13:43.35 | Aiatek | what sentence can i use to allow an specific ANI? |
| 13:43.54 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:44.40 | freh | In features.conf I defined ## as blind transfer, and ** as attended transfer. I am using ** to transfer. Does that not make it an attended transfer? |
| 13:45.32 | ricko73 | do a pastebin of what's being seen in the Asterisk CLI |
| 13:45.54 | *** part/#asterisk porche (n=kursad@81.215.117.205) |
| 13:46.04 | [TK]D-Fender | <PROTECTED> |
| 13:46.35 | [TK]D-Fender | How the hell is priority 3 getting executed on an answered call that has no "continue" option? |
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| 13:47.10 | [TK]D-Fender | And if you transfered to 700, we should see the dialplan executed for it |
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| 13:53.50 | proxium | I run agi dumphtml <filename> in astersikcli, where will I find the file? |
| 13:54.19 | proxium | I have this message AGI HTML Commands Dumped to: <test.txt> |
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| 14:02.03 | freh | This doesn't make any sense. The call does get parked It's just asterisk not reading the position where it's parked. |
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| 14:04.59 | plundra | I've got a simple queue with an iax-member, but when I start asterisk it's marked as "Invalid", until I use it some how, call out or something similar. And after that, it's marked as Not In Use, and can recieve calls. |
| 14:05.26 | plundra | Am I missing something or maybe missunderstod how you "should" do? |
| 14:06.12 | plundra | I tried setting qualify=yes on the peer aswell, and it says it's online. But the queue still says Invalid. |
| 14:06.27 | [TK]D-Fender | plundra: This sounds like what happens when you don't have a NAT keep-alive on a remote agent |
| 14:06.50 | [TK]D-Fender | plundra: While the queue says "invalid" do normal dials TO it work? |
| 14:07.05 | *** join/#asterisk af_ (n=getsmart@88-149-240-185.dynamic.ngi.it) |
| 14:07.46 | plundra | [TK]D-Fender: I havn't tried that, I'll do a Dial() directly to it instead of the Queue. (And yes, the client (Voixer) is behind nat) |
| 14:09.45 | plundra | [TK]D-Fender: Dialing to it works fine, and the queue is updated with the right status. |
| 14:10.04 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 14:10.06 | [TK]D-Fender | plundra: Wait for it to drop out and repeat |
| 14:10.28 | plundra | [TK]D-Fender: Wait for it to be Invalid again you mean? |
| 14:10.35 | [TK]D-Fender | plundra: Yup |
| 14:10.40 | plundra | I have tried that, maybe just not long enough I guess :) |
| 14:10.43 | [TK]D-Fender | plundra: Also, what ver of *? |
| 14:12.08 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 14:12.13 | plundra | 1.6.0.9 built from source, but it was the same with the 1.4.0.something thar came with Ubuntu 8.04, 1.0.2 of Voix Phone, but it's the same with zoiper. |
| 14:13.35 | plundra | Hmm, MIGHT be wrong about zoiper though, so ignore that for now :-) |
| 14:20.55 | kc8pxy | plundra: ekiga comes standard with hardy(/me install hardy ALL the time) setup an account for yoru asterisk with it, and try it. as a control. |
| 14:21.41 | *** join/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com) |
| 14:22.00 | plundra | kc8pxy: And it works fine with the nat-thingy [TK]D-Fender talked about? |
| 14:22.01 | *** join/#asterisk codelemur (n=codelemu@24.96.140.159) |
| 14:22.11 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 14:22.31 | plundra | I did try zoiper on this machine and it get Not In Use directly after registering, so it's fine. |
| 14:23.23 | kc8pxy | plundra: i've not tried it through a nat yet, but i plan to do that today:) |
| 14:23.35 | codelemur | Have a quick PRI question. If I turn PRI debugging on for a span and the telco attempts to send a call to a non-provisioned channel in that span, will I see any indication of it? I have D-chan 24, B-chan 1-10. Say they try to send a call down channel 11 for whatever reason. |
| 14:25.30 | [TK]D-Fender | codelemur: I believe you get an explicit warning for that and it should show up in debug as while its requesting B-chan 11, the signalling still arrives on the D, and * should refuse the call. |
| 14:25.36 | [TK]D-Fender | codelemur: Visibly of course |
| 14:26.26 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:29.29 | kc8pxy | i'm not sure I've heard an answer to this question i have. how many concurent calls can be crammed into "all" 23 channels of a pri? |
| 14:30.29 | eppigy | hi |
| 14:31.42 | coppice | I guess 23 channels will handle 23 calls rather well, and 24 somewhat poorly |
| 14:33.41 | [TK]D-Fender | There are 3 kinds of people in this world, those that understand math, and those that don't... |
| 14:34.12 | coppice | there are 10 kinds. those who understand binary and those who don't |
| 14:41.09 | eppigy | WOW |
| 14:41.46 | russellb | lol ... |
| 14:44.13 | anonymouz666 | LOL |
| 14:45.12 | mort_gib | Uhm... Friday :-) |
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| 14:56.40 | Zeeek | Friday is http://tr.im/voip day! |
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| 15:06.38 | Zeeek | who's going to AMOOOOOOOOOOOCON in 10 days? |
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| 15:12.34 | dni | what is the best way to ring numerous phones at once,. for exmaple if i wish to ringall, sip extension, 1001, 1002 , 1003 ,.. do i need to have the agents login or can i do it with some sort of group ? |
| 15:13.36 | dni | can i do something like member => SIP/1001 in the queus.conf? |
| 15:14.20 | Zeeek | dni SIP/100@SIP/200@SIP/300 |
| 15:14.25 | Zeeek | typo |
| 15:14.43 | Zeeek | SIP/100&SIP200&SIP/300 |
| 15:14.52 | Zeeek | concat them with the ampersand |
| 15:15.02 | dni | got you |
| 15:15.33 | dni | can i also create a group ? |
| 15:15.48 | dni | and then just have it Dial/Group1 |
| 15:15.49 | dni | ? |
| 15:15.51 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 15:15.54 | dni | i mean |
| 15:16.00 | dni | Dial(SIP/group1) |
| 15:18.08 | DavidR2008 | is features.conf and the applicationmap the only way for a caller/callee to execute macros during a call? It's not working the way I would expect it to, and I don't know if something's wrong or if I just don't get it <--- much more likely ;-) |
| 15:20.21 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 15:20.51 | watchy | whenever i try to dialout I get [Apr 24 10:20:37] WARNING[17399]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 15:21.19 | [TK]D-Fender | dni: No |
| 15:21.52 | [TK]D-Fender | DavidR2008: Yes, its the only way, and show us |
| 15:22.16 | proxium | Hello, I got this message is AsteriskCli: [Apr 24 16:11:52] NOTICE[2766]: chan_local.c:617 local_alloc: No such extension/context 55558600051@default creating local channel |
| 15:22.23 | proxium | Any idea? |
| 15:22.23 | watchy | it looks like incoming calls are fine, but outgoing arent |
| 15:22.57 | [TK]D-Fender | proxium: Means exactly what it says. Its looking for a match for that exten in that context and it can't find one |
| 15:23.00 | file | proxium: the extension 55558600051 does not exist in the context default |
| 15:23.56 | [TK]D-Fender | watchy: ... you know what to do... |
| 15:24.08 | watchy | punch myself in the face for being a retard |
| 15:24.11 | watchy | i know that much |
| 15:24.13 | plundra | kc8pxy: A late comment (I've been disctracted with outher stuff); Can't really use ekiga, I want a IAX2-client, since the users will be mostly behind different nat. |
| 15:24.29 | watchy | tk: pastebin my extensions.conf? |
| 15:24.48 | plundra | And so far, the only two usable clients I've found are closed source, voix phone and zoiper. What's the deal with that? :-] |
| 15:24.51 | DavidR2008 | http://pastebin.com/d6b5beb49 |
| 15:24.55 | proxium | I use astguiclient to manually call 003399995555 so why his message appears with 55558600051 and I have 8600051 as the conf ID in vicidial_conference? |
| 15:25.09 | [TK]D-Fender | watchy: show the line generating the error, confirm dahdi's status at CLI..... |
| 15:25.28 | watchy | i have |
| 15:25.32 | DavidR2008 | the record feature works as expected, the ring & r123 features don't do anything and the touchtones get sent to the callee |
| 15:25.33 | watchy | its perfectly fine looking |
| 15:25.59 | [TK]D-Fender | watchy: I don't know that. |
| 15:26.26 | watchy | i'm pastebining it now |
| 15:27.00 | watchy | http://pastebin.com/m35e586a9 |
| 15:27.13 | [TK]D-Fender | DavidR2008: I don't see you enabling those features in your dialplan. You don't just get access to them globally you know... Dial options required, vars to set... |
| 15:27.19 | watchy | im sure im doing something wrong, but i'm not sure what |
| 15:27.24 | [TK]D-Fender | DavidR2008: Go read the doc's on this again |
| 15:27.59 | [TK]D-Fender | watchy: I asked for CLI output, and what card are you using? |
| 15:28.00 | DavidR2008 | [TK]D-Fender, sorry to bring stupid Qs to you, that's the problem, I enable it for the record feature that already works |
| 15:28.08 | watchy | Sangoma A200e i think |
| 15:28.42 | watchy | http://pastebin.com/m718dcee9 |
| 15:28.46 | watchy | i updated it with the full dial |
| 15:28.54 | [TK]D-Fender | DavidR2008: Record is basic and requires a single dial opt. Dynamic needs a VAR set which is well documented, and opts to enable * to even CARE to look for DTMF |
| 15:29.17 | [TK]D-Fender | watchy: [Apr 24 10:28:23] WARNING[17407]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 15:29.34 | [TK]D-Fender | watchy: Stop doing broken CLI dials and use a friggen proper separate device |
| 15:29.41 | watchy | i have |
| 15:29.48 | watchy | i get the same issue with the dahdi stuff |
| 15:29.52 | [TK]D-Fender | watchy: And now I'd like to see your DAHDI configs |
| 15:29.58 | watchy | i'm just using console since i'm not on site |
| 15:30.15 | DavidR2008 | [TK]D-Fender: I'm not using the built-in dial opt, I'm using the DYNAMIC_FEATURES var to enable recording so I can use MixMonitor instead of just Monitor |
| 15:30.41 | [TK]D-Fender | DavidR2008: Feel free to show something complete for us to examine |
| 15:31.02 | [TK]D-Fender | watchy: You know VoIP works when you're not on the site as well... |
| 15:31.25 | Zeeek | IRC #voip-users-conference today in 30 minutes Dan Berninger who helped evaluate VoIP at Bell Labs before you were born... |
| 15:31.59 | Zeeek | http://sessions.voipUsersConference.org is giving you a free cold beer if you attend |
| 15:31.59 | DavidR2008 | well it was just the fact that I wasn't enabling the two new features I added using the VAR. just be me being dumb, thanks for helping me see it. |
| 15:32.24 | [TK]D-Fender | DavidR2008: You're welcome |
| 15:33.05 | watchy | hmm |
| 15:33.20 | watchy | i owe you a steak tk. add it to my bill |
| 15:33.25 | watchy | im an idiot |
| 15:34.05 | watchy | i was trying to dialout using the wrong group |
| 15:34.32 | watchy | i guess the sangoma set everything on group 0, i was trying to use group 1 |
| 15:36.07 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
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| 15:36.22 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 15:36.59 | DavidR2008 | ok, now I just have a question, is there a way to hangup the callee and dial another extension from a macro without hanging up on the caller? |
| 15:39.25 | [TK]D-Fender | DavidR2008: Not sure I follw whre everyone is going. |
| 15:39.34 | [TK]D-Fender | watchy: That's what I was betting on :) |
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| 15:40.55 | DavidR2008 | I can explain why if it would be helpful, but this is what I want to do: user1 calls user2, user1 executes macro, user2 gets hungup on, user3 starts ringing, user3 answers, user1 talks to user3 |
| 15:41.40 | Qwell | seanbright: I left the real ones at home today... let's see how this goes! |
| 15:42.48 | [TK]D-Fender | DavidR2008: make a macro that makes an AMI redirect call for the activator. |
| 15:43.32 | [TK]D-Fender | DavidR2008: Of course the only reason i can see to do this is if you're talking about some guy calling in over Zap and he needs to have call-flow control and can't just transfer himself |
| 15:45.55 | DavidR2008 | no, actually it's to duplicate the functionality of an existing pbx. Right now users can call each other and the phone's autoanswer. However if someone's not in their office and the caller want's to leave a voicemail the caller can press 1 and the phone starting ringing after the ring timeout it gets sent to voicemail. In this case, user2 and user3 are the same device but user3 is a call to... |
| 15:45.57 | DavidR2008 | ...the device without the auto-answer sip header |
| 15:46.59 | mort_gib | Hi I get some of these, but can't find UPGRADE.txt app_queue.c: The device state of this queue member, SIP/XXX, is still 'Not in Use' when it probably should not be! |
| 15:47.30 | Vec | Is there any way to distinguish a SIP phone from a SIP trunk, or they essentially the same thing ? |
| 15:47.49 | *** join/#asterisk grx0 (n=root@70.94.220.114) |
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| 15:51.34 | Zeeek | http://sessions.voipUsersConference.org |
| 15:51.38 | Zeeek | on the air |
| 15:52.13 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
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| 16:02.47 | Zeeek | FWD: http://www.siptosip.net/ |
| 16:03.24 | Zeeek | If you're too lazy to read aout DB: |
| 16:03.25 | Zeeek | Daniel Berninger - Washington, DC based independent technolgy analyst. Expert in technical and regulatory aspects of Internet enabled disruptive communications. Active in VoIP since 1995. Daniel worked on the original assessment of VoIP at Bell Laboratories and led early gateway deployments at Verizon , HP, and NASA after joining VocalTec Communications. He won the 1999 VON Pioneer Award as co-founder of the VON Coalition and worked on the foun |
| 16:03.44 | Zeeek | http://sessions.voipUsersConference.org |
| 16:05.53 | coppice | "He won the 1999 VON Pioneer Award" - how embarrassing :-) |
| 16:06.50 | Zeeek | what can people sign up for on that sip2sip site? |
| 16:08.21 | Zeeek | I am hearing a little noise on Dan, is anyone else? Or is it on my end? |
| 16:09.10 | Qwell | Zeeek: no noise here |
| 16:09.11 | Zeeek | I seem to be in the wrong channel |
| 16:09.22 | Qwell | maybe I should call into the conf before saying that though |
| 16:09.34 | Zeeek | come see up at #voip-users-conference *enog* |
| 16:09.38 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:11.10 | seanbright | Qwell: good luck, amigo. |
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| 16:13.00 | Great_Anta_Baka | yoyo hi all |
| 16:14.18 | dni | does anyone here integrate soma fm + music on hold by any chance ? |
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| 16:26.31 | Qwell | dni: that would be a rebroadcast. you very likely don't want to do that |
| 16:26.58 | dni | how come? bandwidth ? i just figured out how to do it, |
| 16:27.06 | DavidR2008 | [TK]D-Fender: how can I send AMI message from a dialplan (or specifically a macro) do I need to write an external program that I execute using the system application or is there a better way? |
| 16:27.19 | Qwell | dni: the legality is questionable at best |
| 16:28.16 | dni | ah |
| 16:28.35 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:30.06 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:30.26 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:30.41 | bijit | is it possible to pick up any rining extension? |
| 16:31.00 | juanIMP | yep bijit |
| 16:31.31 | juanIMP | the extensions must have the same pickupgroup |
| 16:32.09 | bijit | juanIMP: which out passing the extension... like lets say 4* and that picks up the call |
| 16:34.55 | juanIMP | features.conf bijit |
| 16:35.27 | *** part/#asterisk friartuck (n=friartuc@24.174.124.21) |
| 16:35.48 | bijit | juanIMP: I was just thinking on Pickup...I see now pickupgroup |
| 16:37.14 | juanIMP | with pickupexten = *0 or whatever you want you define the digits to piucp the call bijit |
| 16:37.59 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 16:38.05 | bijit | juanIMP: Thank you very much |
| 16:38.37 | juanIMP | and in sip.conf define pickupgroup and callgroup, =1 or 0 or ..... for all sips bijit |
| 16:43.19 | sulex | hello, i have some problem within and AGI. Perl based, using Asterisk::AGI. At a certain point I use $AGI->wait_for_digit($sometimeout). Well, the dtmf is received but the value is not set as what pushed on the dialpad... I have a value of "49" if "1" is pressed, "50" for "2" and so on. Just wondering if this is normal or a I'm messing something somwhere. Thanks... 1.4 btw here |
| 16:45.00 | ltd | sulex: perl -e 'print ord("1")."\n"' |
| 16:46.24 | sulex | ltd: what a fool am I? spare the comment... going now to selfcrucify myself. Thanks a lot :) |
| 16:47.06 | ltd | sulex: have a beer instead. it works for me. |
| 16:47.26 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 16:48.14 | sulex | ltd: think I'll get a full six pack to forget I'm a fool ;) |
| 16:48.29 | Urthwhyte | This isn't really the right channel, but has anyone played with SIP over HSDPA/HSUPA? |
| 16:49.25 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:49.58 | sulex | bye and thanks again, have a nice weekend guys |
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| 16:52.18 | dni | Hello, could someone give me some advice regarding MusicOnHold and streams ? This is what I have in the .conf [default] |
| 16:52.18 | dni | mode=custom |
| 16:52.18 | dni | application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 - (at) |
| 16:52.18 | dni | /etc/asterisk/stream.playlist and this is the error i see in the console. [Apr 24 12:54:47] NOTICE[23880]: res_musiconhold.c:542 monmp3thread: Request to schedule in the past?!?! |
| 16:52.58 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 17:00.18 | Katty | eppigy: DAVE |
| 17:00.33 | Corydon76-dig | dni: are you running on a virtual host? |
| 17:00.40 | eppigy | Katty: HI |
| 17:00.50 | Katty | eppigy: omg |
| 17:00.51 | dni | Corydon76-dig, yea a vmware image if thats what you mean |
| 17:00.52 | Katty | eppigy: like |
| 17:00.54 | Katty | eppigy: hi and stuff |
| 17:00.59 | eppigy | HERRO |
| 17:01.13 | Corydon76-dig | dni: the reason you're getting that message is that the host image isn't getting enough CPU time |
| 17:01.27 | Katty | eppigy: dinner was absolutely awesome |
| 17:01.36 | eppigy | oh nice what was on the neu? |
| 17:01.39 | eppigy | menu? |
| 17:01.41 | Katty | eppigy: http://42ndgeekstreet.blogspot.com/2009/04/pork-chop-apple-pie-and-stuffing.html |
| 17:01.42 | dni | Corydon76-dig, I see,. thanks for the feedback,. let me try and tweak the esx server |
| 17:01.57 | Corydon76-dig | dni: so requests are getting scheduled "in the past", as the error message says |
| 17:02.27 | eppigy | holy crapple |
| 17:02.31 | Corydon76-dig | (That is, by the time the CPU has time to process a schedule request, the time it's scheduled for is already in the past) |
| 17:02.41 | eppigy | Katty: that looks delish |
| 17:03.15 | dni | Got ya,. |
| 17:03.27 | dni | guess im just going to stay with the default class for now |
| 17:03.46 | eppigy | DONDE ESTA |
| 17:03.48 | Katty | eppigy: the cinnamon was PERFECT |
| 17:03.53 | eppigy | :D |
| 17:03.57 | Katty | eppigy: it was like... summer, all wrapped up in a yummy dinner |
| 17:04.09 | eppigy | hahaha awesome |
| 17:04.15 | eppigy | man it is sucj a beautiful day |
| 17:04.21 | Katty | it's /literally/ apple pie on pork chops |
| 17:04.23 | eppigy | I almost didnt come back from lunch |
| 17:04.26 | Katty | because that's how i make apple pie. |
| 17:04.38 | Katty | well, except for putting it in a crust. i guess that's different. |
| 17:04.42 | eppigy | a nice tender pork chop with apples |
| 17:04.48 | eppigy | sounds really grood |
| 17:04.50 | Katty | and stuffing |
| 17:04.53 | Katty | you can't forget the stuffing |
| 17:05.08 | Katty | you must try it! |
| 17:05.10 | Katty | MUST! |
| 17:05.22 | Katty | the hardest part is chopping apples :P |
| 17:06.39 | Katty | eppigy: dinner tomorrow: http://www.recipezaar.com/Refried-Beans-10614 |
| 17:06.49 | Katty | eppigy: i'm using michelob ultra amber in it |
| 17:07.04 | Katty | eppigy: you must come over for lunch! |
| 17:07.17 | Katty | eppigy: bring a bib! |
| 17:07.57 | *** join/#asterisk friartuck (n=friartuc@24.174.124.21) |
| 17:08.50 | plundra | Hmm, I've just started using 1.6 instead of 1.4, and one small thing is, I get no ringing when waiting in the queue. Any ideas? I've started disabling some modules I do not use, and only keep the configurations I need. Maybe I'm missing something to be able to "ring"? :) |
| 17:10.06 | plundra | Ah! :-D indications.conf My bad. |
| 17:10.29 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173) |
| 17:10.49 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 17:10.55 | Katty | hugs anonymouz666 |
| 17:11.07 | Katty | Qwell: the thorium hump in JCing sucks. |
| 17:11.16 | Katty | Qwell: can you believe i opted for JCing and BSing?! |
| 17:11.18 | [TK]D-Fender | is all sushi-d out |
| 17:11.23 | Katty | Qwell: i'm such a retarded katty. |
| 17:12.04 | Katty | [TK]D-Fender: what kind of sushi? sushimi? |
| 17:12.40 | eppigy | I will need a bib for certain |
| 17:12.46 | Katty | i like the one that isn't raw fishies. |
| 17:13.05 | eppigy | i love raw fishies |
| 17:13.06 | eppigy | :D |
| 17:13.09 | eppigy | TUNA |
| 17:13.26 | Katty | i prefer that cooked, and flaked, with vinegar and lemon juice |
| 17:13.35 | Katty | with melty gooey cheese on swirly rye bread |
| 17:13.36 | eppigy | but i also love eel tempora |
| 17:13.44 | jaytee | we may be out of tuna by 2026 according to current estimates |
| 17:13.46 | Katty | is it shocking? |
| 17:13.52 | Katty | jaytee: :< |
| 17:13.52 | eppigy | jaytee: :[ |
| 17:14.04 | Katty | that makes me sad. |
| 17:14.09 | Katty | i'm going to start a tuna pond in my back yard. |
| 17:14.10 | eppigy | i will no longer be able to eat their uncooked family members |
| 17:14.20 | Katty | i will save the fishies! |
| 17:14.21 | coppice | jaytee: we're out of smoked salmon right now |
| 17:14.30 | jaytee | don't think tuna care much for ponds |
| 17:14.30 | eppigy | LIES |
| 17:14.39 | Katty | well. |
| 17:14.48 | Katty | i could make a back yard ocean pond |
| 17:14.59 | jaytee | coppice, plenty in the pacific northwest and some of it is smoked already. |
| 17:15.10 | Katty | what do tuna eat? |
| 17:15.11 | Katty | krill? |
| 17:15.20 | jaytee | smaller fish |
| 17:15.23 | coppice | jaytee: there's none in our fridge |
| 17:15.26 | Katty | krill is smaller. |
| 17:15.30 | Katty | much... much.. smaller |
| 17:15.43 | eppigy | a little to agile for th tuna |
| 17:15.46 | eppigy | too |
| 17:15.50 | Katty | probably |
| 17:15.58 | eppigy | THEY HUNT IN PACKS |
| 17:16.01 | jaytee | coppice, that's a shopping problem, not an environmental concern although how clean the inside of the fridge might be. |
| 17:16.07 | eppigy | TUNA CAN PACKS |
| 17:16.19 | Katty | wikipedia says: squid, shellfish, plankton |
| 17:16.22 | coppice | jaytee: this is my environment, and I'm concerned |
| 17:16.23 | eppigy | oh wow |
| 17:16.42 | Katty | bioluminecent squid are the shit |
| 17:16.49 | eppigy | i agree |
| 17:16.51 | Katty | you ever seen a video of them?! |
| 17:16.53 | jaytee | by the time the last tuna is gone the cloned tuna meat vats will be up and running in full production |
| 17:16.54 | eppigy | i want some |
| 17:17.05 | Katty | the pressure of the water has to be incredibly high |
| 17:17.08 | jaytee | "Soylent Yellow is dolphin!!!!!" |
| 17:17.25 | Katty | shouldn't that be... soylent... blue? |
| 17:17.28 | eppigy | i wish they would have had bioluminescent squid in clubs during the rave era |
| 17:17.34 | [TK]D-Fender | Katty: EVERYTHING. Maki, Sushu, sashimi, yaki-tori, the works. |
| 17:17.35 | Katty | dolphins are greyish blue, right? |
| 17:17.39 | [TK]D-Fender | Katty: AYCE! |
| 17:17.41 | Katty | [TK]D-Fender: dear lord. |
| 17:17.49 | Katty | [TK]D-Fender: you're going to asplode. |
| 17:17.51 | [TK]D-Fender | Katty: dumplings, etc |
| 17:18.05 | Katty | [TK]D-Fender: where did you put it all?! |
| 17:18.12 | [TK]D-Fender | Katty: Nothing like AYCE sushi to make you feel like a beached whale :p |
| 17:18.13 | jaytee | more grey than blue, at least for tursiops truncatus, the common bottlenose dolphin |
| 17:18.18 | [TK]D-Fender | Katty: GET IN ME BELLy! |
| 17:18.33 | coppice | around here the dolphins are pink |
| 17:18.44 | Katty | cause they eat so much shrimp? |
| 17:18.46 | jaytee | those aren't bottlenose |
| 17:18.47 | [TK]D-Fender | jaytee: Yeah, I remember you in that pic with the dolphin..... |
| 17:19.05 | jaytee | her name's China and she's a sweetie |
| 17:19.10 | Katty | i wonder how much a dolphin costs. |
| 17:19.26 | eppigy | haha |
| 17:19.44 | eppigy | riddick and a dolphin |
| 17:19.48 | coppice | if her name's China she should be pink :-) |
| 17:19.52 | [TK]D-Fender | My sister just came back from swimming with the dolphins in the Carribean |
| 17:20.57 | jaytee | we had a dolphin dog named Gene that would play catch with the dolphins during shows but despite the fact that he was smart and trained well he needed too much attention between shows and it became a problem keeping him so he went to live at one of the keeper's homes. |
| 17:21.30 | jaytee | their skin is amazingly soft and smooth |
| 17:21.55 | jaytee | and they are way more intelligent than any dog I've met. |
| 17:22.36 | coppice | more playful, too |
| 17:22.41 | jaytee | yep |
| 17:23.25 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 17:23.54 | jaytee | I'd say they are probably almost equal if not actually equal to a human in intelligence, just without the tools using capabilites. Who knew opposable thumbs would be such a problem down the road? |
| 17:25.24 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:25.40 | coppice | Its wonderful to follow them chasing the fishing boats. they are after their dinner, but they can't stop being playful while doing so |
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| 17:31.36 | Urthwhyte | Dolphins? |
| 17:31.45 | [TK]D-Fender | jaytee: http://www.cracked.com/article_15853_6-cutest-animals-that-can-still-destroy-you.html |
| 17:31.59 | [TK]D-Fender | jaytee: Bottlenose = OMG RUN |
| 17:33.03 | Urthwhyte | Platypodes really are cute |
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| 17:36.56 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 17:39.15 | f0ner00t | Hello |
| 17:41.23 | grx0 | lo |
| 17:42.15 | Urthwhyte | Sup? |
| 17:42.54 | *** join/#asterisk twanny796 (n=twanny@195.158.83.121) |
| 17:44.31 | twanny796 | trying to set ekiga to send dtmf in a call but it cannot send and I do not hear the tone, any ideas on what I should configure? |
| 17:45.42 | *** join/#asterisk skirmisha (n=viki@90.154.223.91) |
| 17:45.48 | skirmisha | hello guys |
| 17:45.58 | skirmisha | i have one question about dundi |
| 17:46.19 | skirmisha | is it normal when server A do dundi request and server B answer it |
| 17:46.33 | skirmisha | then server A to dial server B |
| 17:46.47 | skirmisha | so call is passing thru server A to reach server B |
| 17:47.26 | grx0 | yeah, otherwise whats the point of making a request...? it checks if the extension is registered elsewhere and then if so connects sip/iax to that server. |
| 17:47.37 | grx0 | unless its cached |
| 17:47.49 | grx0 | but in that case its checked before so ...thats kinda the same |
| 17:48.38 | skirmisha | hmmm |
| 17:48.58 | skirmisha | but then for some reasong if server A went down then call will be dropped |
| 17:49.03 | skirmisha | am i right |
| 17:50.03 | grx0 | yeah it doesnt connect the endpoint to server b directly...the traffic tunnels...eg endpointa -> server a -> server b -> endpoint b...not endpoint a -> server b -> endpoint b. |
| 17:50.16 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-e9ab2ca90cb17cbf) |
| 17:50.34 | skirmisha | i see |
| 17:50.53 | skirmisha | any ideas i can avoid this, even using diff protocol |
| 17:51.52 | twanny796 | any ideas why I cannot send dtmf signals on ekiga |
| 17:51.56 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 17:52.34 | grx0 | you can always use dundi to setup redundancy...its a little costly, but it helps in the if server a dies situation.... |
| 17:53.53 | grx0 | but aside from that you can use straight sip but you're still in a situation if one of the servers dies your call is probably gonna fail, eg endpoint a -> server a -> server b -> endpoint b....if there's two servers involved doing trunking theres not much you can do unless all the endpoints are registered to the same server... as far as I know at least, i'm probably wrong lol |
| 17:54.06 | *** join/#asterisk bpgoldsb (n=bpgoldsb@spatialdata2-gru-gw.customer.gru.net) |
| 17:54.48 | skirmisha | i understand |
| 17:55.19 | grx0 | dundi works great though for trunking and dealing with more than one office, we have 6 servers setup all using dundi with a live backup server for redundancy (the phones are registered to both the primary and the backup in each office... |
| 17:55.23 | skirmisha | or the other way is to send call to all asterisk and which one answer he will take the call |
| 17:56.22 | grx0 | well unfortunatley the way dundi works is it will pick the path of least resistance (eg lowest ping time) or the one with the most priority (which is a setting in the dundi.conf).... so you can't ring them all at once I dont think. |
| 17:57.38 | skirmisha | i am thinking also that for example if call flow is gw -> server A -> server B -> endpoint |
| 17:57.55 | skirmisha | and i have load balance of registranion to both servers |
| 17:58.09 | skirmisha | then on server A i will have double the calls compared to server B |
| 17:58.50 | grx0 | dundi isn't really great for load balancing since there is no facilities really designed to do that, you might try looking into openser (or kamino) to do something like that. |
| 17:59.11 | skirmisha | i know openser |
| 17:59.18 | grx0 | and use openser for your registration...theres documentation on voip-info.org somewhere on doing that im pretty sure |
| 17:59.37 | skirmisha | that's why i said instead to use dundi i can use openser for mass call to all servers |
| 18:00.04 | skirmisha | what is kamino |
| 18:00.09 | skirmisha | another project? |
| 18:00.40 | coppice | its a badly spelt sports car |
| 18:00.42 | grx0 | sorr kamailio...way off..brain fart....its opensers 'new name' |
| 18:00.55 | skirmisha | i see |
| 18:01.00 | grx0 | and apparently opensips is a fork of that as well... |
| 18:01.04 | skirmisha | is it much easier for configuration |
| 18:02.09 | grx0 | I don't know, I use dundi frequently and openser on one small project...dundi is easier to me but thats just because I have more experience with it... openser(or whatever its name is today) seems more suited for what you're talking about. |
| 18:02.36 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 18:02.40 | skirmisha | actualy i have load balancer based on foundry |
| 18:02.53 | skirmisha | that is doing sip calls and sip registration load balancing |
| 18:03.19 | skirmisha | but sometimes thing could go wrong |
| 18:03.19 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 18:03.28 | skirmisha | then i use dundi internally |
| 18:03.45 | *** join/#asterisk lasko_ (n=lasko@70.102.15.210) |
| 18:03.48 | skirmisha | as foundry is making logs of registered users to what server they belong |
| 18:05.07 | skirmisha | anyway thanks for dundi info |
| 18:05.17 | grx0 | np |
| 18:09.48 | *** join/#asterisk orn (n=orn@85-220-9-3.dsl.dynamic.simnet.is) |
| 18:10.53 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 18:12.07 | [TK]D-Fender | SER isn't like other OSS projects. When they fork you get an entire cutlery set for *12* |
| 18:12.37 | jaytee | yes, but are the knives sharpened or dull? |
| 18:13.05 | grx0 | who says they're not spoons....? |
| 18:13.39 | [TK]D-Fender | more like rusty sporks |
| 18:13.58 | *** join/#asterisk rOfLz (n=unit90@119.160.7.167) |
| 18:14.30 | grx0 | indeed.... |
| 18:15.02 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 18:15.51 | KyleK | anyone here got a SPA series FXO (like a SPA3102)? I'm wondering if it immediately picks up the line when a call comes in or if it waits for the other end to answer (asterisk) |
| 18:15.53 | orn | I can't seem to be able to figure out how to have asterisk Register with a provider |
| 18:16.16 | grx0 | orn: what happens when you try? |
| 18:16.20 | orn | I am able to make a call through the provider, but I don't think my asterisk is sending REGISTER messages |
| 18:16.23 | orn | http://pastebin.com/m751a6d15 |
| 18:16.41 | orn | (i have sip debugging enabled in the asterisk CLI output, but no messages for the REGISTRATION attempts) |
| 18:17.04 | orn | i've tried various register => strings |
| 18:17.31 | grx0 | firewall ports open? |
| 18:17.58 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 18:18.55 | *** join/#asterisk Hazuki (n=Hazuki@cpe-72-229-137-30.nyc.res.rr.com) |
| 18:18.56 | [TK]D-Fender | orn: And that isn't going to work because your register statments have to come at the end of everything under [general] and before your first peer entry. Nowhere else |
| 18:19.37 | [TK]D-Fender | orn: It is being ignored |
| 18:19.46 | [TK]D-Fender | orn: and its normally : register => user:pass@host/extentodialomfdon'tforgettofillthisin |
| 18:19.56 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 18:20.06 | [TK]D-Fender | KyleK: Whichever way you configure it to |
| 18:21.48 | rOfLz | I am trying to make dialplan for outbound calls and I need to remove prefix and add 0 can any 1 help me out |
| 18:22.54 | grx0 | give us an example. |
| 18:24.14 | *** join/#asterisk [netman] (n=netman@35.Red-88-25-136.staticIP.rima-tde.net) |
| 18:24.15 | rOfLz | _91XXXXXXXXXXX,1,Dial(SIP/$(SIP/0${EXTEN:2}@1234) |
| 18:24.41 | rOfLz | I can remove 91 but it is not adding 0 |
| 18:25.14 | orn | [TK]D-Fender: Thanks a lot :) Didn't have a clue about the required location of register :D |
| 18:25.19 | grx0 | make it Dial(SIP/0${EXTEN:2}@1234 |
| 18:25.21 | grx0 | ) |
| 18:26.24 | grx0 | no reason to have SIP/ in there twice and beyond that you have an extra ( so thats bad syntax anyway... |
| 18:26.51 | grx0 | gotta have a closing brace/paren for every open one. |
| 18:27.14 | grx0 | () {}...or it'll not play nice. |
| 18:27.15 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 18:27.15 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:27.27 | *** join/#asterisk twanny796 (n=twanny@195.158.83.121) |
| 18:29.40 | rOfLz | it is not making any diff to number |
| 18:29.55 | grx0 | ? |
| 18:29.59 | rOfLz | not even removing 91 |
| 18:30.22 | [TK]D-Fender | rOfLz: show us the latest version please |
| 18:30.46 | *** join/#asterisk bbryant (n=ssddsasd@m4c5a36d0.tmodns.net) |
| 18:31.19 | rOfLz | _91XXXXXXXXXXX,1,Dial(SIP/$(SIP/${EXTEN:2}@1234) |
| 18:31.40 | grx0 | sooo what did you change? because I see nothing diff than before |
| 18:31.45 | rOfLz | it is wrking but I need to add 0 |
| 18:32.00 | grx0 | <PROTECTED> |
| 18:32.30 | rOfLz | yes but when I change it to Dial(SIP/0${EXTEN:2}@1234) |
| 18:32.43 | rOfLz | it does not remove even 91 |
| 18:32.55 | [TK]D-Fender | rOfLz: yes, it DOES |
| 18:33.07 | grx0 | :) |
| 18:33.33 | grx0 | then hell....try Dial(SIP/0${EXTEN:-10}@1234).... either way same results. |
| 18:33.44 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 18:33.55 | grx0 | or how many ever digits that is |
| 18:33.57 | grx0 | I can't count :) |
| 18:34.21 | Corydon76-dig | Ask him if he reloaded after changing it |
| 18:34.55 | *** join/#asterisk [netman] (n=netman@231.Red-88-27-59.staticIP.rima-tde.net) |
| 18:34.57 | [TK]D-Fender | Corydon76-dig: Oh now why would he go and do something silly like actually putting his changes into effect? You'd think he wants it to work or something :p |
| 18:35.23 | rOfLz | I did |
| 18:35.27 | Corydon76-dig | thought I'd try the obvious solution |
| 18:36.33 | Corydon76-dig | rOfLz: core show 9122233344455@whatever-context |
| 18:36.37 | grx0 | lol.....sooo whats the console output |
| 18:37.10 | Corydon76-dig | or dialplan show 9122233344455@whatever-context |
| 18:38.21 | Corydon76-dig | btw, I suspect you have an extra X in there, but whatever |
| 18:39.13 | [TK]D-Fender | rOfLz: show us the failed call in a pastebin along with "dialplan show |
| 18:39.15 | [TK]D-Fender | ~pb |
| 18:39.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 18:40.02 | grx0 | is sitting on the edge of his seat awaiting r0flz reply. (Its friday, I don't have any 'work' to do) lol |
| 18:40.50 | rOfLz | sorry m8 trying again with ur proposed plan |
| 18:41.08 | f0ner00t | I'm getting 404 SIP Not found I plugged my asterisk box right into my cable box last night any ideas? I'm using IPKALL for incomming sip. |
| 18:42.08 | [TK]D-Fender | f0ner00t: 404 from what? And loko at the SIP DEBUG to see what its looking for. |
| 18:44.20 | rOfLz | grx0 : I have tryied this 1 again after reload configs ;) Dial(SIP/0${EXTEN:2}@6131731,60) and it is same |
| 18:44.38 | [TK]D-Fender | rOfLz: provider what I have requested |
| 18:44.42 | f0ner00t | [TK]D-Fender: Thank you I will get myself a fresh cup of debug and beback |
| 18:44.46 | [TK]D-Fender | -r |
| 18:44.47 | grx0 | what [TK said |
| 18:46.34 | grx0 | you can tell is nots working all ya want but that is the 'correct' way to do what you want so if we cant see any console output we don't have nothin for ya :) |
| 18:46.56 | *** join/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com) |
| 18:47.00 | grx0 | dialplan show 9122233344455@whatever-context |
| 18:48.07 | [TK]D-Fender | not even. Show the entire damn mess |
| 18:48.17 | [TK]D-Fender | trust = 0 |
| 18:48.53 | grx0 | lol... yeah I am too trusting....its a fact. |
| 18:52.58 | rOfLz | sorry guys cannot use irc on that machine so have to work on 2 diff location |
| 18:53.00 | rOfLz | :( |
| 18:53.49 | [TK]D-Fender | rOfLz: SSH doesn't care <- |
| 18:54.44 | rOfLz | I need to dial 11 digits with area code just want to remove 91 which is workin with this context but I have to include 0 by my self_91XXXXXXXXXXX,1,Dial(SIP/0${EXTEN:2}@1234) |
| 18:55.01 | [TK]D-Fender | rOfLz: that line is fine |
| 18:55.20 | [TK]D-Fender | (if you actually put "exten => " in front of course) |
| 18:55.44 | grx0 | shhhh don't give away the farm [tk] thats an insider secret |
| 18:55.47 | [TK]D-Fender | and you have a sip peer named [1234] |
| 18:56.27 | [TK]D-Fender | grx0: well... he's saying it didn't strip off, not that the entire line didn't work. I'd like to think that is what it sound like and his pasting is laze |
| 18:56.30 | [TK]D-Fender | lazy* |
| 18:57.17 | [TK]D-Fender | grx0: then again I'm wondering if due to his claims of how he's actually maintaining his server on the machine itslf and IRCing from another that his hand-typed line he's showing us is NOT what was actually done on the server |
| 18:57.32 | grx0 | yeah why arent you just ssh'ing to your asterisk box and cut and paste that way... I mean if you're physically in the same place you're bound to be on the same network |
| 18:57.55 | grx0 | granted my * boxes and my laptops are on 2 different nats, I can still get to them all lol... |
| 18:58.00 | grx0 | ssh = friend |
| 18:59.12 | tzafrir_laptop | Yeah. This IRC server is not in my LAN and I still manage to chat here... |
| 18:59.54 | rOfLz | sorry i made a mistake last line I typed it has SIP/0$ it is bit changed |
| 18:59.59 | rOfLz | exten => _91XXXXXXXXXX,1,Dial(SIP/${EXTEN:2}@1234) |
| 19:00.12 | grx0 | uhm... |
| 19:00.21 | grx0 | exten => _91XXXXXXXXXX,1,Dial(SIP/${EXTEN:2}@1234) |
| 19:00.33 | grx0 | errr _91XXXXXXXXXXX,1,Dial(SIP/0${EXTEN:2}@1234) |
| 19:00.46 | grx0 | damnit...stupid cut and paste ...exten => _91XXXXXXXXXXX,1,Dial(SIP/0${EXTEN:2}@1234) |
| 19:00.54 | grx0 | or you're doing it wrong lol.... period. |
| 19:00.58 | rOfLz | this is actuall line and I have to add 0 in dialing numberlike 9102224455666@1234 |
| 19:01.31 | grx0 | like [tk] said...check if you're even registered to the peer you have there if its simply not dialing and failing... |
| 19:01.43 | grx0 | 6131731 |
| 19:02.05 | SuPrSluG | i'm working with level 3 and on emum dialing internation . they want the 011 stripped, so i have _011.,1,Dial(SIP/+${EXTEN:3}@t38) , but when dialing the 001 is there. Dial("SIP/111-2e0b", "SIP/+011442079778073@t38) why is that? |
| 19:02.11 | [TK]D-Fender | rOfLz: FFS SSH into your server and pastebin what I requested. |
| 19:02.30 | [TK]D-Fender | roI am not going to trust any more of this hand-entered stuff. |
| 19:03.17 | [TK]D-Fender | SuPrSluG: et tu brute? |
| 19:03.32 | grx0 | lollol |
| 19:04.03 | SuPrSluG | yon D-Fender has that lean and hungry look.... |
| 19:04.23 | [TK]D-Fender | was born on the Ides of March |
| 19:04.36 | SuPrSluG | my brother was too! |
| 19:04.50 | [TK]D-Fender | is also very skiled with bladed weapons |
| 19:06.47 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 19:08.57 | SuPrSluG | "There 's daggers in men's smiles" |
| 19:09.15 | f0ner00t | Suprslug. Are you set up as International numbering plan. |
| 19:09.23 | f0ner00t | or national numbering lan? |
| 19:10.07 | SuPrSluG | we're working with them now. I can call international , but they don't want the 011 format |
| 19:10.54 | f0ner00t | Suprslug What are you using sip or trunks? |
| 19:10.59 | f0ner00t | co trunks |
| 19:11.08 | SuPrSluG | sip |
| 19:11.12 | jeffspeff | is this the proper setup for sip.conf? http://pastebin.ca/1402176 |
| 19:12.22 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 19:12.23 | SuPrSluG | jeffspeff:looks ok |
| 19:12.37 | jeffspeff | ty |
| 19:13.12 | SuPrSluG | contexts will have to match in extensions.conf |
| 19:14.05 | [TK]D-Fender | SuPrSluG: Ferengi Rules of Acquisition # 48. The bigger the smile, the sharper the knife. |
| 19:14.07 | SuPrSluG | also, no username for 1000? |
| 19:14.51 | [TK]D-Fender | jeffspeff: No, that config isn't good. REGISTER's have to come after everything else under [general]. Everything following them gets IGNORED |
| 19:14.53 | SuPrSluG | who's says star trek is low brow? obvious allusion |
| 19:15.13 | grx0 | thats terrible.....just...terrible |
| 19:15.51 | [TK]D-Fender | jeffspeff: Also for being behind NAT you seem to have forgotten to specify your WAN IP. |
| 19:16.48 | jeffspeff | [TK] the sample IVR setup worked fine last night... i think my error is in extensions.conf somewhere |
| 19:17.32 | guax | sorry about the nick changes |
| 19:18.32 | grx0 | thinks why goto the doctor if you *know* whats wrong? lol |
| 19:18.38 | [TK]D-Fender | jeffspeff: You just showed me a broken sip.conf. fix this. Your "IVR" issues are another matter. Please pick a problem you are actually going to focus on CORRECTING please. |
| 19:19.42 | [TK]D-Fender | grx0: Because knowing you have a cancerous growth on your pancreas doesn't mean you want to stab yourself with a scalpel trying to remove it. some things are best left to professionals :p |
| 19:19.58 | *** join/#asterisk orn (n=orn@85-220-9-3.dsl.dynamic.simnet.is) |
| 19:20.27 | grx0 | [TK]D-Fender: thats half the fun... you get way more respect when you show up and say "I tried to fix it myself doc"... ok maybe not so much |
| 19:21.31 | grx0 | I mean who really needs blood anyway and sterilization anyway? a good kitchen knife and some gloves will work fine. |
| 19:22.22 | grx0 | has to go put our sales office in time out...bbl! |
| 19:23.42 | *** part/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:23.48 | orn | Hmmm, do incoming calls when you are registered always come in as "peer" rather than "user"? |
| 19:25.58 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 19:28.00 | SuPrSluG | orn: up to the provider |
| 19:28.08 | [TK]D-Fender | orn: The auth method (IP vs name, etc) is what differentiates them for what little difference remains. Generally "user" isn't required for much |
| 19:30.38 | orn | My problem is that when I put fromuser, defaultuser & password in the provider's section in sip.conf (when he is type=friend), my call doesn't go into the dialplan correctly and I can't figure out why. Only when I remove fromuser, defaultuser & secret does it work. So I tried to set up seperate instances in sip.conf for incoming and outgoing, but the incoming seems to be coming to the type=peer entry rather than the type=user entry |
| 19:31.20 | orn | when I have those fields set, asterisk sends my provider a 401 unauthorized repeatedly |
| 19:31.33 | *** join/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 19:31.34 | orn | probably because it is expecting the provider to send me that user authentication |
| 19:32.06 | ThatKidKel | anyone know of a good sip origination/termination wholesaler--preferably connected to Level3? |
| 19:33.44 | orn | so is there any way for me to force the incoming messages from the provider to come in as "user" and not "peer"? |
| 19:34.57 | orn | that, or another method of saying that the defaultuser and secret authentication are only supposed to be used in outgoing traffic? |
| 19:37.02 | [TK]D-Fender | orn: many providers send you calls WITHOUT auth at all. |
| 19:37.13 | [TK]D-Fender | orn: So the more you try to enforce the worse you fail |
| 19:38.22 | *** join/#asterisk nayro_ii (n=dshfp943@189.27.24.52.dynamic.adsl.gvt.net.br) |
| 19:38.32 | orn | [TK]D-Fender: I don't want to enforce their auth to me... but they require auth from me, and I am not sure how I can set them up in sip.conf so that I send them auth but don't expect auth back |
| 19:38.51 | orn | [TK]D-Fender: Because they come in as "peer" when they send me calls, instead of "user" |
| 19:39.59 | [TK]D-Fender | orn: "insecure=port,invite" |
| 19:40.22 | [TK]D-Fender | orn: and forget the "defaultuser" , etc |
| 19:41.08 | nayro_ii | hello, i'm having problems with (asterisk 1.4.11, zaptel, TDM400). When i call to one FXO, the receiver hear a noise and nothing happens, who's making the calling, still hear the rings.... ne1 already saw this kind of thing? |
| 19:41.38 | nayro_ii | it started on Monday.... :( |
| 19:43.23 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 19:44.03 | nayro_ii | when I call to the other FXO, the call is 100% |
| 19:44.37 | *** join/#asterisk telecos (n=sergio@119.166.219.87.dynamic.jazztel.es) |
| 19:44.43 | [TK]D-Fender | nayro_ii: Maybe your module 1 is dead |
| 19:48.18 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 19:48.30 | nayro_ii | [TK]D-Fender, yeah, it's possible.... |
| 19:48.50 | [TK]D-Fender | nayro_ii: Swap them and test |
| 19:48.54 | *** join/#asterisk Josmar9999 (n=martin@187.131.154.71) |
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| 19:49.50 | unixdawg | anyone here looking for a embedded system to play with |
| 19:50.09 | unixdawg | I have a unit I want to unload |
| 19:50.34 | ayeso | Im having an issue with the meetme app. If someone enters incorrect numbers to join a conference, it askes them to try again, but rather than giving them the opportunity to reenter the digits, it hangs up on them. |
| 19:50.49 | orn | [TK]D-Fender: Thanks. That did the trick. I needed to set the caller-id in the dialplan as I couldn't use fromuser anymore, but after that it works great :-) |
| 19:50.56 | Josmar9999 | sorry i went down... |
| 19:51.45 | [TK]D-Fender | orn: You can ususally use "fromuser" if you also set "sendrpid=yes" and "trustrpid"=yes |
| 19:52.32 | orn | [TK]D-Fender: ah ok! Great. Thanks again |
| 19:52.48 | [TK]D-Fender | orn: You're welcome |
| 19:56.21 | orn | [TK]D-Fender: That also worked :-) |
| 19:56.32 | orn | thanks |
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| 20:20.45 | eppigy | yeah |
| 20:20.48 | eppigy | YEAH |
| 20:22.05 | DavidR2008 | my redirect to the same phone isn't working, could someone take a look and suggest anything I might be doing wrong. Basically as soon as the phone gets the second call it hangs up. |
| 20:22.07 | DavidR2008 | http://pastebin.com/d64f2ff4a I tried to highlight asterisk verbose lines in the sip debug |
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| 20:28.25 | *** part/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 20:29.09 | *** part/#asterisk pwebguy (n=pwebguy@200.110.240.130) |
| 20:33.50 | DavidR2008 | ok, I think what is happening is that even though the call is redirected to some else as soon as the original callee hangs up the whole call is dropped |
| 20:34.32 | DavidR2008 | I put a wait statement in the dialplan after the redirect but before the second dial and it hangs up while in the wait |
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| 20:59.02 | jeffspeff | has DigitTimeout been removed? I'm getting an error that there is no application by that name when i try to use it |
| 20:59.21 | jeffspeff | exten => s,2,DigitTimeout,5 |
| 21:00.19 | Qwell | jeffspeff: Set(TIMEOUT(digit)=5) |
| 21:00.55 | jeffspeff | ty |
| 21:01.33 | jeffspeff | is that the same for responsetimeout ? |
| 21:01.40 | Qwell | TIMEOUT(response) |
| 21:01.55 | jeffspeff | oh, duh... ty |
| 21:02.33 | DavidR2008 | is there a way to remove a header that was added with sipaddheader? |
| 21:03.40 | jeffspeff | how badly broken is this example? I'm now having issues with the custom dir it wants to use.... http://www.voip-info.org/wiki/view/Asterisk+tips+phrase+recording+menu |
| 21:03.59 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 21:09.46 | *** join/#asterisk sisto (n=silvio@ic64-250.intercanal.com.uy) |
| 21:10.09 | sisto | hi |
| 21:12.04 | sisto | I have been sending inward calls to a specific extension which runs a phpagi script |
| 21:13.20 | sisto | What I want to do now is to run the agi script while there is a conversation between two people (a connection between two sip phones) |
| 21:13.26 | sisto | and capture the key presses |
| 21:13.30 | sisto | is this possible? |
| 21:15.14 | sisto | I don't know whether it's possible to run an agi script over a point to point connection |
| 21:15.49 | sisto | I was only able to make it work for an incoming call connected to asterisk but not to another phone |
| 21:18.19 | sisto | I don't know where the control goes when it passes through this command: exten => 509,1,Dial(SIP/Prueba-509) |
| 21:18.28 | sisto | (for example) |
| 21:22.51 | sisto | like once the connection is established between the two parties |
| 21:23.03 | sisto | where would the execution go? |
| 21:23.21 | sisto | can I so stuff while people are talking at all> |
| 21:23.22 | sisto | ?? |
| 21:25.04 | sisto | anyone? |
| 21:27.05 | sisto | maybe I should direct the incoming call to the agi script and then capture the sound output from asterisk somehow |
| 21:35.55 | *** part/#asterisk lasko_ (n=lasko@70.102.15.210) |
| 21:36.33 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 21:36.43 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 21:38.14 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 21:41.01 | war9407 | Hello all! Had a quick question, is it possible (upon a call) for asterisk to send me an e-mail when the caller-id information arrives "Person ${CALLERID(name)} is calling from ${CALLERID(num)}" ? |
| 21:41.17 | war9407 | as soon as its passed, I would like this information, is it possible? |
| 21:42.43 | rob0 | core show application system |
| 21:44.20 | war9407 | rob0: hm nice |
| 21:44.28 | war9407 | rob0: after .Ringing()/.Answer() |
| 21:45.09 | war9407 | rob0: exten => XXX,1,System("mail -s "$callerid" email) ? |
| 21:45.13 | war9407 | rob0: exten => XXX,1,System("mail -s "$callerid" email") ? |
| 21:47.16 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.213) |
| 21:47.18 | *** part/#asterisk takashi_85 (n=ahmed@41.196.80.213) |
| 21:53.36 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:56.13 | war9407 | doh |
| 21:56.14 | war9407 | it didnt work |
| 21:56.34 | war9407 | exten => 123,2,System(echo "CID: ${CAlLERID(all)}" | mail -s "CID: ${CAlLERID(all)}" me@email.com) |
| 21:56.35 | war9407 | hmm |
| 21:57.39 | war9407 | worked |
| 21:57.43 | war9407 | got the email but notthe caller ID info |
| 21:57.56 | war9407 | how do I encapsulate the caller Information into the system command? |
| 21:58.45 | war9407 | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System |
| 21:58.47 | war9407 | will try this |
| 22:00.32 | war9407 | Incoming Call From: |
| 22:00.33 | war9407 | :( |
| 22:00.41 | war9407 | hrmm |
| 22:01.56 | watchy | hmm maybe there is no cid? |
| 22:02.55 | war9407 | when I leave a voice mail it says 'you have a voice mail from Cell Phone XX (XXXXXXXXXX), ', will keep digging |
| 22:04.38 | war9407 | closer, :) Incoming Call From: Cell |
| 22:05.01 | [TK]D-Fender | war9407: Maybe you should try getting all of the letters in your function name capitalized. |
| 22:05.32 | war9407 | [TK]D-Fender: in the conf. they are |
| 22:05.46 | [TK]D-Fender | war9407: not in what you just showed us. |
| 22:05.51 | sisto | is it possible to run extension commands after the dial command while two people are talking? |
| 22:06.09 | [TK]D-Fender | sisto: like? |
| 22:06.41 | sisto | [TK]D-Fender: I want to run an agi script while two people are talking and capture the keyboard tones |
| 22:07.27 | sisto | they press keys in the phone and they control an external php script through phpagi |
| 22:07.31 | sisto | is this possible? |
| 22:07.51 | sisto | they key here is that I want to do it WHILE they are talking |
| 22:07.55 | [TK]D-Fender | sisto: no "normal" way for this, but you could Originate a new local channel using M() that will bridge to it |
| 22:08.08 | [TK]D-Fender | sisto: this requires 1.6+ |
| 22:08.44 | sisto | [TK]D-Fender: ok! thanks |
| 22:09.06 | sisto | [TK]D-Fender: so I guess I have to start a new channel connected to both channels |
| 22:09.15 | sisto | using the M() command? |
| 22:09.38 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:09.39 | EmleyMoor | M() option to Dial - calls a Macro |
| 22:10.09 | [TK]D-Fender | sisto: use M() as the trigger so that you can pass the channel to latch onto |
| 22:10.20 | war9407 | in the body it works but in the subject it does not, possibly due to a bad character? |
| 22:10.30 | war9407 | subject: Subject: CID: Cell |
| 22:10.33 | sisto | ok thank you both.. I will look into it |
| 22:10.40 | war9407 | body: CID: Cell Phone XX <234XXXXXX> |
| 22:11.10 | war9407 | (when using callerid{all} |
| 22:11.16 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.229.38) |
| 22:11.16 | war9407 | ${CALLERID(all)} rather |
| 22:11.29 | f0ner00t | [TK]D-Fender: See sip message please http://pastebin.ca/1402321. |
| 22:12.16 | [TK]D-Fender | f0ner00t: No user 'Unknown' in SIP users list <- doesn't match against a peer |
| 22:12.37 | [TK]D-Fender | f0ner00t: Looking for 1745 in incoming (domain arbelosservices.com) SIP/2.0 404 Not Found |
| 22:12.39 | andrebarbosa | After upgrading from latest zaptel 1.2.* to dahdi 2.0.0, I'm having lots of echo on my FXO connected to a verizon line |
| 22:12.48 | andrebarbosa | anyone notice something like that? |
| 22:12.53 | [TK]D-Fender | f0ner00t: tells you what it is looking for and where and that it doesn't find it. Go fix |
| 22:13.01 | Qwell | andrebarbosa: Did you setup an echo can? |
| 22:13.47 | [TK]D-Fender | andrebarbosa: How the heck do upgrade just Zaptel 1.2 to DAHDI? DAHDI requires 1.4.20+ |
| 22:14.12 | andrebarbosa | yes |
| 22:14.19 | andrebarbosa | i've tried 4 algos |
| 22:14.24 | andrebarbosa | mg2 and kb1 |
| 22:14.34 | andrebarbosa | and oslec and even i registered an hpec license |
| 22:14.36 | andrebarbosa | no luck |
| 22:14.36 | andrebarbosa | :( |
| 22:14.50 | f0ner00t | [TK]D-Fender: Is it possible the domain is incorrect? |
| 22:14.51 | andrebarbosa | [TK]D-Fender, I also upgrade |
| 22:15.00 | andrebarbosa | asterisk from 1.2.27 to 1.4.24 |
| 22:15.02 | andrebarbosa | ;) |
| 22:15.24 | f0ner00t | [TK]D-Fender: The domain is setup is to my asterisk box is this incorrect? |
| 22:15.33 | denon | andrebarbosa: 1.6 :) |
| 22:16.13 | andrebarbosa | eheh |
| 22:16.24 | andrebarbosa | lets get 1.4 stable and then I think to jump again |
| 22:16.25 | andrebarbosa | :d |
| 22:16.37 | denon | 1.4's stable for me.. :) |
| 22:16.44 | drmessano | Microsoft posted its first loss ever!! LINUX IS WINNING |
| 22:16.48 | andrebarbosa | yes |
| 22:16.49 | andrebarbosa | is fine |
| 22:17.04 | war9407 | [TK]D-Fender: do you notify (email) when someone calls, or smb, etc? |
| 22:17.06 | drmessano | thinks 1.6 is more stable than 1.4 |
| 22:17.14 | tzafrir_laptop | drmessano, linux? LNUX? |
| 22:17.24 | andrebarbosa | altought some cdr bugs are anoying :s |
| 22:17.25 | drmessano | Im kidding |
| 22:17.33 | drmessano | Linux isnt winning anything |
| 22:17.51 | drmessano | Just shocked I havent seen more spin |
| 22:18.07 | drmessano | Like when the giant sneezes and the ant goes "See.. *I* did that" |
| 22:18.16 | denon | they all think Novell is winning :) |
| 22:19.09 | f0ner00t | drmessano: Got Linux! |
| 22:19.09 | drmessano | Id love to see Asterisk on netware |
| 22:19.15 | [TK]D-Fender | f0ner00t: ... no EXTENSION to match in that CONTEXT. Fix. Your. Dialplan. |
| 22:19.55 | f0ner00t | [TK]D-Fender: Thank you! |
| 22:19.55 | denon | drmessano: that's kinda sick |
| 22:20.04 | drmessano | denon: Muhahahaha |
| 22:20.06 | tessier_ | drmessano: First loss? |
| 22:20.08 | tessier_ | drmessano: er...no |
| 22:20.08 | denon | I wouldnt mind asterisk running on netware, but it's ip stack..... |
| 22:20.33 | tessier_ | drmessano: They did not lose money. Their revenue was down. They ONLY made 13.7 BILLION instead of the 14.5 BILLION they were expecting. |
| 22:20.50 | tessier_ | drmessano: Hopefully the beginning of a trend but hardly cause for rejoicing or declaring victory. |
| 22:21.12 | denon | a trend of .. poor projections? |
| 22:21.44 | drmessano | You take things way too seriously |
| 22:21.58 | drmessano | It was meant to be HUMOROUS |
| 22:22.02 | drmessano | ..... |
| 22:22.42 | tessier_ | drmessano: Get a better sense of humor. |
| 22:22.59 | drmessano | tessier_: Get one, period |
| 22:23.05 | denon | oh boy, here we go |
| 22:23.52 | tessier_ | http://www.i18nguy.com/humor/unicode-humor.html Now that's funny. :) |
| 22:26.19 | drmessano | Those are as lame as every windows joke that metaphorically references windows in a house, car, or office building, etc |
| 22:26.28 | drmessano | Lets party like its 1991 |
| 22:27.41 | andrebarbosa | Do you that upgrading for the lastest dahdi will be usefull to fix this problem? |
| 22:34.06 | drmessano | How did I not know Google owns Jaiku |
| 22:35.44 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 22:37.14 | *** join/#asterisk Aiatek (n=munoz@190.6.143.194) |
| 22:41.33 | *** join/#asterisk voxter (n=voxter@76.77.91.250) |
| 22:42.24 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 22:44.08 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 22:46.51 | tzafrir_laptop | andrebarbosa, do you use oslec? |
| 22:47.00 | tzafrir_laptop | also: what FXO device is it, exactly? |
| 22:49.28 | andrebarbosa | TDM410p |
| 22:49.37 | andrebarbosa | i've tried with oslec also |
| 22:49.39 | *** part/#asterisk Schreiber1337 (n=SCHREIBE@216.169.165.178) |
| 22:49.39 | andrebarbosa | no luck |
| 22:49.59 | andrebarbosa | I did calibrate with fxotune |
| 22:50.06 | andrebarbosa | setup de gains with dahdi_monitor |
| 22:50.17 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 22:50.20 | andrebarbosa | my sip to TDM410p calls still with lots of echo |
| 22:50.21 | andrebarbosa | :s |
| 22:50.54 | andrebarbosa | I also tried with jbenable=yes on sip.conf and chan_dahdi.conf |
| 22:50.55 | andrebarbosa | no luck |
| 22:50.59 | andrebarbosa | same thing |
| 22:51.31 | tzafrir_laptop | for starters, is the echo canceller enabled? |
| 22:51.36 | andrebarbosa | yes |
| 22:51.38 | tzafrir_laptop | cat /proc/dahdi/* |
| 22:51.54 | tzafrir_laptop | do you see a EC: for each channel? |
| 22:51.56 | andrebarbosa | [newjersey-lan]> cat /proc/dahdi/1 |
| 22:51.57 | andrebarbosa | Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) |
| 22:52.02 | andrebarbosa | 1 WCTDM/0/0 FXOKS (In use) (EC: HPEC) |
| 22:52.03 | andrebarbosa | yes |
| 22:52.07 | andrebarbosa | for the 4 channels |
| 22:52.25 | jameswf | ~pb |
| 22:52.26 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 22:52.26 | andrebarbosa | 2 WCTDM/0/1 FXSKS (In use) (EC: HPEC) |
| 22:52.30 | andrebarbosa | I'm using fxo line 2 |
| 22:52.36 | tzafrir_laptop | and hpec is enabled and functional? there's some registration utility and such |
| 22:52.43 | tzafrir_laptop | (I'm not familiar with it) |
| 22:52.48 | andrebarbosa | on a channel |
| 22:52.49 | andrebarbosa | Echo Cancellation: 128 taps, currently ON |
| 22:53.03 | andrebarbosa | that is output of dahdi sho channel 2 |
| 22:53.05 | andrebarbosa | when on a call |
| 22:53.21 | andrebarbosa | yes tzafrir_laptop is activated, but i got same results with mg2, and kb2 and oslec |
| 22:53.45 | tzafrir_laptop | what version of asterisk do you use? |
| 22:54.43 | tzafrir_laptop | for oslec, I would have suggested to send samples to David Rowe, but the current dahdi_echocan_oslec does not have his sampling code |
| 22:54.56 | tzafrir_laptop | I wonder what it would take to get him useful samples |
| 22:55.36 | andrebarbosa | 1.4.24 |
| 22:57.05 | sisto | why is programming asterisk extensions so complicated? |
| 22:57.19 | sisto | it's frustrating |
| 22:57.57 | seanbright | sisto: what are you trying to do? |
| 22:58.43 | sisto | seanbright: nevermind me I am just discharging my frustration :( |
| 22:58.48 | sisto | seanbright: thanks anyway |
| 23:00.17 | seanbright | good talk. |
| 23:01.03 | andrebarbosa | tzafrir_laptop, can this be problem with dahdi? because I've upgrade recently to asterisk 1.4 and zaptel 1.2.*, and with the same line, same card (tdm410p) i didn't have echo |
| 23:01.22 | tzafrir_laptop | what version of dahdi do you have? |
| 23:01.30 | andrebarbosa | s/to asterisk 1.4 and zaptel 1.2.*/asterisk 1.4 and dahdi 2.0.0 |
| 23:01.34 | andrebarbosa | dahdi 2.0.0 |
| 23:01.41 | andrebarbosa | I will test it with lastest dahdi |
| 23:01.44 | jameswf | This RPM robot has to be the coolest thing I have ever written. should probably break it to ensure job security... |
| 23:01.53 | tzafrir_laptop | for starters, 2.1.0.x would be a better idea |
| 23:01.57 | andrebarbosa | ok |
| 23:02.10 | andrebarbosa | I'll do it in minutes |
| 23:02.10 | andrebarbosa | ;) |
| 23:02.15 | andrebarbosa | working on that |
| 23:02.59 | jameswf | oh and synergy <3 |
| 23:13.58 | *** join/#asterisk laggo (n=user@80.175.142.209) |
| 23:14.03 | laggo | hi everyone |
| 23:14.49 | laggo | if i want my inbound call to be able to enter an extension followed by the hash key, do i need to pattern match _XXX# for example? |
| 23:15.22 | laggo | because i would like to be able to use ${EXTEN} which might be complicated by the trailing hash character |
| 23:25.37 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1279278104.dsl.bell.ca) |
| 23:25.52 | dlynes | Does asterisk still have a maximum # of sip registrations or sip peers? |
| 23:32.07 | sisto | seanbright [TK]D-Fender I was able to do what I was trying to do |
| 23:32.21 | sisto | I used the features.conf file |
| 23:32.33 | sisto | and enabled a new feature |
| 23:32.35 | sisto | thanks |
| 23:32.41 | sisto | gotta go bye |
| 23:36.12 | KyleK | dlynes: its got a limit of around 10000 for something |
| 23:36.18 | KyleK | not sure if thats sip registrations tho |
| 23:39.34 | f0ner00t | Why did I get this as soon as I dialplan reload http://pastebin.ca/1402398 |
| 23:39.51 | f0ner00t | Sorry sip reload |
| 23:39.54 | f0ner00t | Not dialplan |
| 23:41.05 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 23:41.58 | f0ner00t | [TK]D-Fender: ? |
| 23:45.49 | mmlj4 | dlynes: as many as your resources permit |
| 23:46.57 | mmlj4 | dlynes: but if you have a metric load of SIP devices, at some point you'll want a SIP proxy, such as SER |
| 23:51.19 | *** join/#asterisk autobus (n=autobus@213.22.170.168) |
| 23:51.32 | autobus | hello |
| 23:52.10 | autobus | one question, how i make auth per ip but for one range ip/netmask |
| 23:52.11 | *** join/#asterisk tfrew (n=tfrew@c-68-57-89-103.hsd1.va.comcast.net) |
| 23:52.32 | tfrew | hello, I was wondering if anyone here is in the buisness of running a voip provider service |
| 23:52.34 | tfrew | I'm looking to pick somones brain for a little while, I would be willing to offer a payment through paypal |
| 23:54.01 | *** part/#asterisk unixdawg (n=unixdawg@pool-96-236-182-100.pitbpa.east.verizon.net) |
| 23:54.02 | autobus | i test configuration: permit=ip/netmask and hostname=dynamic |
| 23:54.14 | autobus | this configuration is correctly for ip auth? |
| 23:57.23 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 23:59.46 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
