| 00:00.09 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:00.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:00.24 | Greek-Boy | seanbright: yes |
| 00:00.44 | seanbright | Greek-Boy: ok... so you are able to select it? or it has an XXX in front of it? |
| 00:00.55 | Greek-Boy | i am able to select it |
| 00:00.59 | Greek-Boy | but it doesn't compile |
| 00:01.39 | _ShrikE | Borgon: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 00:01.55 | seanbright | Greek-Boy: it fails or it doesn't even try to compile? |
| 00:02.10 | Greek-Boy | it fails |
| 00:02.14 | seanbright | ohhhhhhhhhhhhhhhhh |
| 00:02.16 | seanbright | heh |
| 00:02.16 | Greek-Boy | should I paste output? |
| 00:02.18 | seanbright | yes |
| 00:02.23 | seanbright | that changes *everything* |
| 00:02.32 | seanbright | pastebin please |
| 00:03.24 | Greek-Boy | k |
| 00:03.27 | Greek-Boy | http://pastebin.ca/1227354 |
| 00:03.34 | *** join/#asterisk Segnale007 (n=Segnale0@host202-254-dynamic.18-79-r.retail.telecomitalia.it) |
| 00:04.12 | seanbright | Greek-Boy: doh! |
| 00:04.19 | seanbright | Greek-Boy: you need an updated app_rpt.c :) |
| 00:04.27 | seanbright | Greek-Boy: i fixed that in SVN last week |
| 00:04.41 | Greek-Boy | i searched google |
| 00:04.44 | Greek-Boy | didn't find anything |
| 00:05.13 | seanbright | hold |
| 00:07.06 | seanbright | Greek-Boy: cd /path/to/asterisk/source |
| 00:07.16 | seanbright | Greek-Boy: wget -O apps/app_rpt.c "http://svn.digium.com/view/asterisk/branches/1.4/apps/app_rpt.c?revision=146244&content-type=text%2Fplain" |
| 00:07.17 | Greek-Boy | k |
| 00:07.26 | seanbright | Greek-Boy: make |
| 00:07.27 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 00:08.26 | Greek-Boy | thanks seanbright |
| 00:08.34 | Greek-Boy | seanbright: i am glad you are finding the bugs |
| 00:08.47 | seanbright | Greek-Boy: that one is fixed in SVN. will be in the next release. |
| 00:09.01 | seanbright | just fell through the cracks since none of the core devs actually compile that module :) |
| 00:10.07 | cvnet | what does : failed (bearercapability notauth) <-- mean ? |
| 00:10.16 | De_Mon | ~question |
| 00:10.17 | jbot | [question] If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 00:10.31 | seanbright | cvnet: it means that bearercapability notauth failed. |
| 00:10.32 | seanbright | :) |
| 00:10.33 | Greek-Boy | seanbright: yeah I can understand that |
| 00:10.44 | Greek-Boy | seanbright: i decided to make use of it :P |
| 00:10.49 | seanbright | De_Mon: he *asked* a question |
| 00:11.23 | seanbright | cvnet: paste the full line |
| 00:11.24 | De_Mon | its for Ritzerisk |
| 00:11.32 | seanbright | De_Mon: ahh, sorry. my bad. |
| 00:11.41 | De_Mon | I didn't realize i was still in scroll back buffer :) |
| 00:11.41 | seanbright | we left though. |
| 00:12.01 | Greek-Boy | seanbright: thanks for your help |
| 00:12.06 | seanbright | Greek-Boy: no sweat |
| 00:12.14 | seanbright | i accept donations via paypal |
| 00:12.20 | seanbright | :) |
| 00:12.31 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:12.35 | De_Mon | oh well, he wasn't too bright |
| 00:12.53 | seanbright | hmmm |
| 00:12.57 | seanbright | now there is an idea |
| 00:13.06 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:13.06 | seanbright | paid asterisk support via irc |
| 00:13.56 | seanbright | cvnet: wake up |
| 00:14.13 | cvnet | trying to past it |
| 00:14.15 | cvnet | one sec plz |
| 00:15.13 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
| 00:15.16 | cvnet | Oct 14 20:13:38] NOTICE[2987] chan_sip.c: -- Registration for '1398784101@sipgw.voicenetwork.ca' timed out, trying again (Attempt #176) |
| 00:15.16 | cvnet | [Oct 14 20:13:38] NOTICE[2987] chan_sip.c: Failed to authenticate on REGISTER to '1398784101@sipgw.voicenetwork.ca' (Tries 3) |
| 00:15.19 | seanbright | running trunk, huh? |
| 00:15.39 | cvnet | but on teh system it shows its registered |
| 00:15.44 | seanbright | neither of those lines contains "failed (bearercapability notauth)" |
| 00:16.01 | cvnet | i get that error from softphone |
| 00:16.16 | cvnet | 20:13:30 Line 1 : outgoing call to '14168298210' |
| 00:16.16 | cvnet | 20:13:31 Line 1 : failed (bearercapability notauth) |
| 00:16.23 | cvnet | thats from softphone |
| 00:17.20 | seanbright | hmmm |
| 00:17.25 | seanbright | not sure. |
| 00:17.29 | seanbright | i thought that was coming from asterisk |
| 00:17.29 | cvnet | i paid for DID which also allows me to make local calls in 416 my whole goal is to make a call from my softphone to my cell phone using asterisk server |
| 00:18.05 | seanbright | and you are running trunk for some reason? |
| 00:18.31 | seanbright | nm |
| 00:18.48 | cvnet | yes, i ceated a trunk for the voip provider (did) |
| 00:19.14 | seanbright | no, i mean the version of asterisk you are running |
| 00:19.19 | cvnet | Registeredvoicenetworksip1398759035sipgw.voicenetwork.ca |
| 00:20.04 | seanbright | i need to see a SIP debug of the failing registers |
| 00:20.32 | seanbright | run 'sip set debug' at the asterisk CLI |
| 00:20.46 | cvnet | ok |
| 00:21.06 | seanbright | shut down your softphone, bring it back up, and pastebin everything that shows up on the console |
| 00:21.09 | seanbright | ~pb |
| 00:21.10 | jbot | [~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 |
| 00:21.13 | seanbright | cvnet: ^^^ |
| 00:21.15 | *** join/#asterisk outtolunc (n=me@c-24-130-75-122.hsd1.ca.comcast.net) |
| 00:22.41 | seanbright | and while you're at it, pastebin your sip.conf making SURE to mask your passwords (but only mask your passwords, nothing else) |
| 00:22.57 | seanbright | in the meantime i am going to do some smoking |
| 00:22.59 | seanbright | back in a moment |
| 00:23.58 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 00:28.29 | seanbright | back. |
| 00:30.55 | cvnet | I can not copy paste from the box, im logged via RDP the the server running windows 2003 there I have a virtual machine installed which is running asteriskNow |
| 00:31.53 | seanbright | http://pastebin.ca/tools.php |
| 00:31.59 | seanbright | download paste2pastebin.pl |
| 00:32.07 | seanbright | chmod 755 paste2pastebin.pl |
| 00:32.37 | *** join/#asterisk grithe (n=chandler@c-71-206-155-255.hsd1.va.comcast.net) |
| 00:32.48 | seanbright | cat copy_of_sip.conf.that.you.have.masked.passwords.in | ./paste2pastebin.pl |
| 00:33.54 | cvnet | i can copy sip.conf and other config files, just not the sip debug from terminal |
| 00:33.54 | seanbright | so you are remote desktoped into a win2k3 server which has virtual server installed on it? |
| 00:34.10 | seanbright | why can't you just SSH into the virtual machine? |
| 00:34.16 | seanbright | using putty or securecrt? |
| 00:34.18 | cvnet | yes it has virtual server installed and on the virtual server im running asterisk now |
| 00:34.27 | cvnet | i tried connecting via telnet it didnt allow me |
| 00:34.28 | Nugget | telnet is eeeeeeevil! |
| 00:34.30 | drmessano^ | Which version? |
| 00:34.47 | seanbright | cvnet: telnet is probably disabled. you should use ssh. |
| 00:35.03 | seanbright | i like to telnet |
| 00:35.14 | seanbright | hmm. Nugget = bot? |
| 00:35.15 | drmessano^ | Which version of Asterisknow? |
| 00:35.29 | cvnet | im in now |
| 00:35.30 | seanbright | Nugget: you're ugly |
| 00:35.31 | cvnet | one sec |
| 00:35.33 | cvnet | lastest |
| 00:36.43 | edibrac | do i have to tell asterisk which DIDs it should handle? or basically that's what the telco provider does? |
| 00:37.10 | drmessano^ | 1.02? |
| 00:37.14 | edibrac | i ordered a new block of DIDs which have a different prefix - will that be a problem? |
| 00:37.24 | seanbright | edibrac: telco does that. but if you want asterisk to behave differently based on inbound DID, you have to do extensions.conf magic |
| 00:37.27 | Nugget | eyes seanbright |
| 00:37.29 | seanbright | edibrac: no |
| 00:37.34 | cvnet | seanbright should i paste it here ? or in private? |
| 00:37.41 | seanbright | cvnet: to a pastebin |
| 00:37.43 | seanbright | ~pb |
| 00:37.43 | jbot | [~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 |
| 00:37.47 | drmessano^ | ARE YOU USING 1.0X OR THE 1.5 BETA |
| 00:37.58 | seanbright | Nugget: do you have some kind of auto-responder for "telnet?" |
| 00:38.01 | edibrac | seanbright: the odd part is, i can dial the new DID and it works from an internal phone, but from my cell phone I get: "we're sorry you call cannot be completed as dialed. Please check the number and dial again. 000 000" |
| 00:38.04 | Nugget | do I? |
| 00:38.10 | seanbright | Nugget: yes. |
| 00:38.29 | edibrac | i guess it's XO's fault |
| 00:38.43 | cvnet | http://pastebin.ca/1227386 |
| 00:38.49 | seanbright | edibrac: have you tried dialing a 1 before the number? |
| 00:39.28 | edibrac | seanbright: yeah, you mean from an internal phone? I dial 9 and then 1 |
| 00:39.50 | seanbright | edibrac: no, from your cell |
| 00:40.05 | edibrac | i did and i see no messages when i'm in *CLI |
| 00:40.27 | seanbright | edibrac: that's bizarre. |
| 00:40.46 | seanbright | edibrac: and you don't have any loopback rules in asterisk that match that DID? |
| 00:40.56 | seanbright | cvnet: looks like the register is succeeding |
| 00:41.03 | cvnet | http://pastebin.ca/1227387 <-=- my sip.conf file |
| 00:42.25 | seanbright | cvnet: but i notice you don't have a register => line in sip.conf |
| 00:42.34 | edibrac | seanbright: hmm wait i think i'm the guilty one. not XO |
| 00:42.45 | seanbright | edibrac: figures... :P |
| 00:42.54 | edibrac | unless they fixed it while putting me on hold |
| 00:43.04 | seanbright | cvnet: oh right, this is asterisknow |
| 00:43.24 | cvnet | yes |
| 00:43.31 | seanbright | edibrac: telco tends to do that. |
| 00:43.50 | seanbright | edibrac: "everything looks fine on our end... yep... looks good." |
| 00:43.54 | seanbright | and then it magically works. |
| 00:44.18 | seanbright | cvnet: but the register appears to work, when are you seeing that 'bearercapability notauth' in your softphone? when you try to place an outbound call? |
| 00:44.55 | cvnet | i think i know what hte problem is |
| 00:45.01 | seanbright | cvnet: what's that? |
| 00:45.55 | cvnet | i had two users registered on my softphone, i think its using the old registeration info,Allow-Events: presence |
| 00:45.55 | cvnet | Content-Length: 0 |
| 00:45.55 | cvnet | <-------------> |
| 00:45.55 | cvnet | --- (13 headers 0 lines) --- |
| 00:45.55 | cvnet | Using latest REGISTER request as basis request |
| 00:46.08 | seanbright | sigh... |
| 00:46.45 | seanbright | da hitler? |
| 00:46.54 | seanbright | wha-wha-what? |
| 00:48.11 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 00:48.32 | seanbright | edibrac: on all of my installs, i have patterns in asterisk for internal devices that match all of our DIDs |
| 00:48.48 | seanbright | edibrac: that way, if someone from inside dials one of them, i just re-route it back into dialplan instead of going out over the wire. |
| 00:50.08 | edibrac | that's a good idea |
| 00:50.09 | seanbright | saves me a couple channels |
| 00:50.56 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:50.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:51.12 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 00:51.19 | *** join/#asterisk StephenF[W] (n=StephenF@198.144.197.28) |
| 00:51.20 | *** join/#asterisk cvnet (n=dahitler@74.210.108.245) |
| 00:51.24 | cvnet | sorry guys |
| 00:52.55 | seanbright | cvnet: if you have to paste more than 2 lines, use a pastebin |
| 00:52.59 | seanbright | cvnet: always always always |
| 00:53.09 | seanbright | cvnet: if you paste more than 2 lines to the channel, a puppy dies |
| 00:53.13 | drmessano^ | Yeah, stop being a douchebag, noob |
| 00:53.24 | seanbright | way to contribute |
| 00:53.29 | seanbright | gives drmessano^ a gold star |
| 00:53.56 | drmessano^ | We dont want to see your fucking dialplan pasted in here amongst the skype requests and join/parts |
| 00:54.15 | seanbright | heh |
| 00:54.27 | drmessano^ | So keep it in your pants, Adolf |
| 00:54.48 | drmessano^ | Sorry, "DA HITLER" |
| 00:54.49 | seanbright | cvnet: so were you able to resolve your issue? |
| 00:55.04 | drmessano^ | ZOMG, A DICTITATOR? EVERYBODY PALIN |
| 00:56.03 | cvnet | seanbright this is what i get when i try to make an outbound calls --> http://pastebin.ca/1227398 |
| 00:56.21 | *** join/#asterisk chendy (n=chatzill@58.60.218.174) |
| 00:57.19 | *** join/#asterisk jameswf-home (n=james@ip72-200-94-120.tc.ph.cox.net) |
| 00:57.27 | seanbright | cvnet: your dial statement needs auth information in it, doesn't it? i have to admit i've never done SIP out to a provider before. |
| 00:58.43 | cvnet | not really |
| 00:58.46 | seanbright | cvnet: i think your dial has to be -> Dial(SIP/voicenetwork-out/<INSERT DID HERE>) |
| 00:58.59 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 00:58.59 | cvnet | ; To use VoiceNetwork.ca to termination your calls |
| 00:59.00 | seanbright | or is that a big fat lie? |
| 00:59.10 | cvnet | ; add the following line to your extensions.conf file |
| 00:59.22 | cvnet | exten => _X.,1,Dial(SIP/voicenetwork_out/${EXTEN}) <-- thats the provider is telling me |
| 00:59.29 | cvnet | but i have no idea where to put that statement |
| 00:59.52 | seanbright | asterisknow must have a setting for it |
| 01:00.10 | seanbright | bkruse: you around? |
| 01:00.17 | cvnet | i dont see, it,however i do have access to all the files, but i have no idea where to put that statement |
| 01:00.30 | seanbright | don't manually put it in |
| 01:00.36 | seanbright | the interface has a way |
| 01:00.44 | seanbright | i assume |
| 01:01.05 | seanbright | asterisknow uses... freepbx? |
| 01:01.20 | cvnet | i put the other info in trunk as a voip provider and it resistered it ok |
| 01:01.28 | cvnet | no it uses linux |
| 01:01.31 | cvnet | redhat |
| 01:01.52 | edibrac | i'd like to match all 7 digits of a specific phone number so is this wrong: exten => _4130662,1,Macro(voicemailonly,${PHONE_Test_DID},${EXT_Test_DID}) |
| 01:02.02 | seanbright | edibrac: lose the _ |
| 01:02.17 | seanbright | edibrac: but it should work with or without it |
| 01:02.25 | seanbright | cvnet: no, i mean the GUI that asterisknow uses |
| 01:02.53 | cvnet | no idea, you can check it here --> http://174.133.158.163 |
| 01:03.00 | cvnet | thats my box |
| 01:03.22 | Borgon | does sip trunk termination mean that i can change the caller id in asterisk conf? |
| 01:04.13 | edibrac | seanbright: i'm telling zapata.conf to go to [pri-inbound] first, then I have my exten at the top of that block .. but * seems to skip over it: http://pastebin.com/m6e2ba1b1 |
| 01:05.35 | edibrac | if i change exten => 4130662,1,Macro(voicemailonly,${PHONE_Test_DID},${EXT_Test_DID}) to: exten => _0662,1,Macro(voicemailonly,${PHONE_Test_DID},${EXT_Test_DID}) that works |
| 01:05.45 | edibrac | so i guess i'm not sure why it's limiting it to last 4 digits |
| 01:06.47 | edibrac | are you limited to 4 digits for inbound matching? i have some contexts setup for outbound calling where I can match across the entire number |
| 01:07.24 | StephenF[W] | seanbright, pretty sure AsteriskNOW uses AsteriskGUI doesnt it? |
| 01:07.31 | seanbright | StephenF[W]: i think you're right, yes. |
| 01:07.40 | seanbright | edibrac: no, but your provider might only send you 4 |
| 01:07.48 | StephenF[W] | be weird if they didnt use their own GUI... |
| 01:07.54 | StephenF[W] | thats how you know your products suck |
| 01:08.25 | edibrac | does hotmail still run on bsd or linux? |
| 01:09.03 | edibrac | seanbright: how can i tell how many digits they send..without asking? |
| 01:09.17 | edibrac | ..and i guess even if i ask i'm not sure if I can trust them :) |
| 01:09.38 | seanbright | edibrac: _.,1,NoOp(${EXTEN}) |
| 01:10.17 | edibrac | aww i see 4 |
| 01:10.31 | edibrac | that means if we order more DIDs we have to be careful not to overlap |
| 01:10.46 | seanbright | edibrac: indeed. or you can tell them to send you more. |
| 01:10.48 | edibrac | but i guess that is a standard thing to worry about when dealing iwth dids? |
| 01:10.53 | seanbright | edibrac: we get all 10. |
| 01:10.56 | seanbright | edibrac: yes, it is. |
| 01:11.01 | seanbright | edibrac: we have no overlap. |
| 01:11.14 | edibrac | what carrier is that? |
| 01:11.20 | seanbright | global crossing |
| 01:11.25 | seanbright | but we requested 10 |
| 01:11.31 | edibrac | or maybe it's depending per-carrier, per-region, per-building, per-manager |
| 01:11.32 | seanbright | we have hundres of DIDs |
| 01:11.43 | seanbright | edibrac: it's per-what-you-want-or-need |
| 01:12.15 | edibrac | can i request that they send more digits? or is that set in stone? |
| 01:12.28 | edibrac | or maybe that too, depends on what equipment XO uses |
| 01:12.42 | seanbright | edibrac: you can request that they send more digits |
| 01:12.53 | seanbright | edibrac: what provider? |
| 01:13.03 | edibrac | we use XO |
| 01:13.28 | seanbright | they resellers for someone? |
| 01:13.42 | edibrac | i'm not sure |
| 01:14.43 | edibrac | are you allowed to be a reseller of resellers? I see that you can be a reseller of XO |
| 01:16.00 | seanbright | i don't really know |
| 01:16.44 | edibrac | oh well i can survive without matching more digits..for now |
| 01:17.16 | *** join/#asterisk salzh (n=Administ@58.247.194.241) |
| 01:19.25 | drmessano^ | "du Hast" |
| 01:20.05 | WimpMan | Who has what? |
| 01:22.01 | drmessano^ | It's "You Hate" |
| 01:22.15 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 01:22.19 | WimpMan | no |
| 01:23.33 | jameswf-home | the song really loses something in translation... sort of like David Hasselhoff |
| 01:25.01 | Katty | hi james. |
| 01:26.09 | jaytee | jameswf-home, don't be dissing my VP pick. I think the Shatner/Hasselhoff 08 ticket is the only logical choice left to America. |
| 01:26.17 | Katty | hugs jaytee |
| 01:26.32 | jaytee | hugs Katty |
| 01:26.48 | [TK]D-Fender | jaytee: No, that'd be the Nimoy/Travolta ticket |
| 01:26.52 | jameswf-home | i really hoped Nader would have picked Ron Paul |
| 01:27.11 | jaytee | I'd rather have Ron Paul in the 1st seat |
| 01:27.18 | Katty | horay for Ron Paul |
| 01:27.21 | Katty | horay for fender! |
| 01:27.22 | jameswf-home | nader/paul that would be cool, wouldnt last but it would be cool |
| 01:27.24 | Katty | did you eat pie? |
| 01:27.25 | Katty | hugs [TK]D-Fender |
| 01:27.29 | jaytee | the man knew all this crap was gonna happen over 8 years ago |
| 01:27.32 | [TK]D-Fender | Katty: Pumpkin even! |
| 01:27.38 | Nugget | http://www.ar15.com/forums/topic.html?b=1&f=5&t=771045 <-- awesome |
| 01:27.42 | jaytee | [TK]D-Fender!!!! OMG!!!! you're back!!! |
| 01:27.46 | drmessano^ | Losers.. Still loving ron paul, even after the sex tape |
| 01:28.00 | jameswf-home | I am voting for nader F the black guy and the old white dude |
| 01:28.01 | jaytee | dude, I was starting to worry. You've been AWOL for like 4 days |
| 01:28.13 | jameswf-home | ron paul sex tape? ewwwwww |
| 01:28.23 | Katty | Nugget: ohai |
| 01:28.25 | Katty | hugs Nugget |
| 01:28.31 | Katty | [TK]D-Fender: mmm, pumpkin |
| 01:28.31 | Nugget | I like ron paul except for his goofy christian views. |
| 01:28.43 | Katty | and his slightly obsessive behavior? |
| 01:28.47 | drmessano^ | Me? |
| 01:28.50 | [TK]D-Fender | jaytee: Yup, went to Toronto (more or less) fri through today. |
| 01:28.51 | jaytee | Nugget, yeah, that's the downside of his platform |
| 01:28.57 | drmessano^ | Oh no |
| 01:28.58 | Katty | hugs drmessano^ |
| 01:28.59 | jameswf-home | an ocd president how awesome would that be |
| 01:29.00 | drmessano^ | Fender |
| 01:29.21 | Nugget | I'm CDO. That's like being OCD except it's in alphabetical order like things are supposed to be. |
| 01:29.38 | [TK]D-Fender | Nugget: we would be largely protected from his religious views but his uber-Constitutionalism |
| 01:29.40 | Katty | Nugget: this is why i love you. |
| 01:29.44 | voxter | Nugget: Hahaha |
| 01:29.45 | drmessano | I figured Fender got overly frustrated at IRC and went off and killed a hobo in an alley or something to relieve the stress |
| 01:29.47 | Nugget | huggles Katty |
| 01:30.10 | [TK]D-Fender | by* |
| 01:30.11 | jaytee | dealing with OCD is a real bitch. If I go into someone's house and they have paintings or pictures hanging that aren't straight I have to really push hard to supress the urge to straighten them. |
| 01:30.37 | Katty | i have this habit of clearing unused time off microwaves. |
| 01:30.44 | jaytee | haha |
| 01:30.47 | Katty | it annoys the crap out of me |
| 01:31.09 | drmessano | "By" would imply IRC frustrated you, "at" implies your misdirected anger on that hobo ;) |
| 01:31.28 | jaytee | or god forbid if someone sets a table and puts the knife and spoon on the left and the fork or forks on the right. |
| 01:32.30 | Nugget | the big one for me is the little screws on light switch plates. gotta be straight up and down. or, in a pinch, horizontal if straight up and down threatens to crack the plastic by being too tight |
| 01:34.03 | Borgon | q/quit |
| 01:34.08 | Katty | Nugget: you are odd. |
| 01:34.16 | Nugget | guilty |
| 01:34.23 | Katty | Nugget: and here i thought straightening up the coffee table everytime i walked by was odd... |
| 01:34.48 | drmessano | I get really OCD about peoples ears.. If they're crooked, I like to cut them off |
| 01:35.30 | drmessano | Son of a bitch |
| 01:35.54 | drmessano | I just installed the AsteriskNOW 1.5 beta on a box, and it formatted my Vista install |
| 01:36.02 | jaytee | cool! |
| 01:36.09 | Nugget | I love my microwave because it lets me turn off the clock, so when it's not running the display is just blank. |
| 01:36.14 | drmessano | DAMN YOU!!! (shakes fist anime style) |
| 01:36.29 | jaytee | KHAAAAAAAANNNN!!!!!!! |
| 01:36.38 | Nugget | clocks in appliances are annoying because they're impossible to get all in sync |
| 01:37.15 | jaytee | it's the techotardy people that have VCRs that always blink 12:00 that amuse me because they can't figure out how to set them. |
| 01:37.17 | Nugget | I don't want or need my oven, microwave, coffeemaker, and ricemaker to all helpfully tell me a different time of day. |
| 01:37.38 | Katty | guess who's comming to dinner |
| 01:37.41 | Nugget | unless they're going to run ntpd I don't want it. :) |
| 01:37.51 | drmessano | I bought a Sony alarm clock that had some feature that implied it never needed to be set |
| 01:37.52 | Katty | Nugget: Geek. |
| 01:38.03 | StephenF[W] | drmessano, Atomic Clock |
| 01:38.06 | drmessano | So I am thinking "15 bucks, built in WWVB?" |
| 01:38.17 | drmessano | Shut the fuck up and let me finish |
| 01:38.33 | drmessano | So I start reading the book |
| 01:38.45 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:38.52 | drmessano | It has a 5 yr lithium battery, and is "factory set" |
| 01:38.55 | drmessano | lol |
| 01:39.02 | drmessano | So, "never needs to be set" |
| 01:39.13 | drmessano | Of all the gimmicks |
| 01:39.55 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:40.55 | jameswf-home | 5Years != never |
| 01:41.10 | drmessano | Dude |
| 01:41.21 | drmessano | $15 consumer alarm clock bought in 2008? |
| 01:41.28 | drmessano | 5 years is FOREVER |
| 01:41.39 | jaytee | 5 years is longer than the life of most chinese manufactured appliances anyways |
| 01:41.39 | drmessano | 5 years is OPTIMISM |
| 01:41.56 | jameswf-home | good call i have been through 3 blackberrys in the last year |
| 01:42.19 | drmessano | This last 8830 I got has been solid |
| 01:42.57 | jaytee | try and find a Dell Dimension or Optiplex made after 2003 that'll last more than 4 years before the motherboard dies. |
| 01:43.01 | drmessano | Still cant believe the total lack of a SIP phone for BB |
| 01:44.15 | jaytee | http://i251.photobucket.com/albums/gg282/rizzy811/iphonevsstone.jpg |
| 01:44.59 | drmessano | ROFL |
| 01:45.09 | drmessano | I saw one of those for the thinbook |
| 01:45.46 | Katty | roflmao |
| 01:45.50 | Katty | jbot: roflmao? |
| 01:45.51 | jbot | from memory, roflmao is rolling on the floor laughing my arse off, or painful, or http://www.youtube.com/watch?v=iEWgs6YQR9A |
| 01:45.51 | jameswf-home | I see no price comparison... how much for the rock ? |
| 01:46.15 | drmessano | http://regmedia.co.uk/2008/01/19/macbookcommodorecompare.jpg |
| 01:46.17 | drmessano | There you go |
| 01:46.19 | drmessano | VERY WIN |
| 01:47.34 | jaytee | hahahaha |
| 01:49.44 | drmessano | HA |
| 01:49.54 | drmessano | FF3 B1 is supposed to be faster |
| 01:50.02 | drmessano | Faster than 3, or 2, or 1? |
| 01:50.08 | drmessano | Because.. theres a difference |
| 01:52.50 | Katty | jaytee: approved. |
| 01:52.57 | jaytee | yay!!! |
| 01:53.07 | drmessano | Oh lord |
| 01:53.08 | drmessano | dont do that |
| 01:53.18 | jaytee | don't do what? |
| 01:53.34 | Katty | have another bacardi |
| 01:53.37 | Katty | then you won't care! |
| 01:53.41 | drmessano | Katty's facebook is full of useless friend updates.. like "my toe just cramped" |
| 01:53.51 | drmessano | SHE KILLED MY PHONE BATTERY EATING ICE CREAM |
| 01:53.59 | jaytee | hahahaha |
| 01:54.16 | Katty | drmessano: and pictures of my 10 week old. |
| 01:54.24 | Katty | drmessano: i'm sure that's useless. |
| 01:54.27 | jaytee | but he's sooooo cute!!! |
| 01:54.59 | drmessano | I dont need to know how every bite of Cherry Garcia tastes.. I get it. You like it. Really |
| 01:55.34 | drmessano | For god sakes, if you want to spam me with one lines, follow me on twitter.. |
| 01:55.38 | drmessano | liners* |
| 01:55.45 | drmessano | Now that is SPAMOCITY |
| 01:56.02 | drmessano | Im just messing with you katty |
| 01:56.25 | Katty | considers herself, messed with? |
| 01:56.35 | Katty | that doesn't sound appropriate. hrm. |
| 01:57.00 | drmessano | Consider yourself <something that wont hold up in court" |
| 01:57.07 | drmessano | Im not going back to prison |
| 01:57.41 | jaytee | god, he really loves that pink cow. awwww |
| 01:57.48 | Katty | jaytee: i know :> |
| 01:58.55 | drmessano | jaytee actually takes some of those dumbass quizzes that sound remotely interesting.. |
| 01:59.12 | drmessano | As long he doesnt throw up a "What cheese are you?" |
| 01:59.18 | jameswf-home | WOW walmmart sells cars now |
| 01:59.23 | jaytee | my neice keeps sending me requests for them. some of them are fun |
| 01:59.54 | Katty | facebook is a fairly good way to kill time |
| 02:00.04 | Katty | and if you browse at it, you always have something to talk about with your friends |
| 02:00.07 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 02:00.12 | drmessano | jaytee: Its all fun and games until you find out you're "Australian Beaver Cheese" |
| 02:00.18 | jaytee | hahahaha |
| 02:00.20 | Katty | especially us people with extreme social skills. |
| 02:00.29 | drmessano | lol |
| 02:01.09 | drmessano | I love using applications designed for networking with your friends as a substitute for having friends |
| 02:01.13 | drmessano | Its the only way to go |
| 02:01.48 | Katty | well there's always another good topic |
| 02:01.56 | Katty | whatever cute pictures were on cuteoverload |
| 02:02.18 | drmessano | My wife is all like "Why dont you ever ask me about your day", and I am all like "Why dont you update your damn twitter and fucking tell me" |
| 02:02.20 | drmessano | Du |
| 02:02.22 | drmessano | Duh |
| 02:02.34 | Katty | horrible. |
| 02:02.42 | drmessano | heh ;) |
| 02:03.17 | jaytee | why would your wife ask you why you don't ask her about YOUR day? |
| 02:03.18 | drmessano | "you didnt even notice I changed my hair color!!!" "How the hell was I supposed to know, you have the same pic on your profile!!!" |
| 02:03.30 | drmessano | my* |
| 02:03.33 | cvnet | seanbright first of all thank you very much for all your work and you have no idea how much this has helped me, secondly i got few questions to ask |
| 02:03.40 | jaytee | ah, that makes much more sense |
| 02:03.50 | drmessano | Although, I have to admit |
| 02:03.53 | drmessano | In all seriousness |
| 02:03.56 | cvnet | seanbright if i want to run a calling card biz would AsteriskNow would be a good choice? |
| 02:04.00 | drmessano | I am that way about VoIP vs Telco |
| 02:04.15 | seanbright | cvnet: i don't personally think so. |
| 02:04.16 | drmessano | "I tried calling your cell 3 times, WTF?" "Ok, is that VOIP?" |
| 02:04.20 | jaytee | so one of these days you'll get a twitter message saying, "Hey, I want a divorce" |
| 02:04.40 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:04.41 | drmessano | "I didnt hear my PAP2 ring!!" |
| 02:04.56 | seanbright | cvnet: you're better off with a custom install, since there is (relatively) complex logic that would have to go into it. |
| 02:05.00 | cvnet | seanbright: what other system would you suggest? |
| 02:05.12 | seanbright | cvnet: i would just do vanilla asterisk |
| 02:05.22 | seanbright | cvnet: like installing asterisk from source or a tarball |
| 02:05.27 | drmessano | I prefer neopolean myself |
| 02:05.48 | cvnet | seanbright sorry for my ignorance, but whats the difference? |
| 02:06.26 | seanbright | cvnet: asterisknow is more for your cookie cutter install. small business. no in-depth technical knowledge of asterisk required. |
| 02:06.39 | jaytee | cvnet, "vanilla" asterisk is highly flexible. With AsteriskNOW or (insert that T word thingy here) it's pretty much, "We'll give you your dialplan and you'll like it, bitch!" |
| 02:06.58 | seanbright | cvnet: when you want to start building applications with asterisk, you want more control over dialplan logic, external data sources, etc. |
| 02:07.55 | cvnet | hum |
| 02:08.37 | seanbright | cvnet: asterisknow = OS + asterisk + GUI |
| 02:09.23 | seanbright | cvnet: the GUI is the "limiting" factor when you want to build anything advanced like... well... a calling card application |
| 02:09.46 | cvnet | I c what you mean now |
| 02:10.02 | seanbright | cvnet: even though a calling card application is probably on the order of 30-40 lines of dialplans and 50 or so lines of Perl/PHP/<insert your favorite language here> |
| 02:10.22 | seanbright | cvnet: have you bought "the book" yet? |
| 02:10.24 | seanbright | ~tfot |
| 02:10.25 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 02:10.31 | seanbright | cvnet: that ^^^ |
| 02:10.44 | cvnet | I'll get that |
| 02:10.46 | seanbright | if not, you should pick it up or check out the online PDF |
| 02:10.53 | jameswf-home | cvnet: Alternitive pay someone $$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$$444 |
| 02:10.54 | seanbright | i prefer dead-tree documentation myself |
| 02:10.57 | cvnet | in past 5 days been doing alot of reading... |
| 02:11.06 | seanbright | cvnet: yes, or pay me money |
| 02:11.07 | seanbright | heh |
| 02:11.40 | jameswf-home | i will do it for $400,005.92 |
| 02:12.24 | jaytee | I love dead-tree documentation because I can read it on my "porcelain lounge chair" and because it really pisses off the druids. |
| 02:12.29 | cvnet | secondly when i call into my system it repeads the digits, when try to enter an extions (6000) it disconnects me, while 6000 is a valid extension |
| 02:12.53 | seanbright | cvnet: looking... |
| 02:13.17 | drmessano | Sounds like you hosed it from abusing the config files |
| 02:13.26 | drmessano | Mission accomplished! |
| 02:15.27 | seanbright | cvnet: you don't have inbound dialing rules |
| 02:15.53 | cvnet | hum |
| 02:15.58 | seanbright | cvnet: in the GUI, take a look at "Incoming call rules" on the left hand menu |
| 02:16.53 | cvnet | ya im there right now |
| 02:17.10 | seanbright | you need to define an inbound rule |
| 02:17.39 | cvnet | _X. <-- means any number ? |
| 02:17.43 | seanbright | correct |
| 02:18.47 | seanbright | runs off for nicotine |
| 02:19.22 | Nugget | I quit smoking 11 years, 5 months, 1 week, 3 days, 7 hours, 19 minutes, and 22 seconds ago. During that time, I would have smoked 91,902 cigarettes. (That's like smoking a 4.35 mile-long cigarette) By quitting, I've saved $16,082.85! (That's 8 Apple 30" Cinema Displays and change) I've avoided inhaling 2.39 kg of tar, 147 grams of nicotine, and 1.47 kg of carbon monoxide. |
| 02:19.32 | *** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net) |
| 02:20.45 | jaytee | Nugget, awesome! congrats. |
| 02:21.19 | jaytee | I'm still hooked on "goverment subsidized crack" |
| 02:24.00 | drmessano | HA.. you've also missed out on a ton of bartrash smokersluts.. too bad |
| 02:24.20 | Katty | jaytee: new pictures!! |
| 02:24.30 | drmessano | Sure, you're lungs are pinker.. but is it worth being less cool? doubtful |
| 02:24.36 | drmessano | Hang on, need to clean my stoma |
| 02:24.46 | drmessano | your* |
| 02:24.49 | *** join/#asterisk StephenF[W] (n=StephenF@198.144.197.28) |
| 02:25.52 | seanbright | good times. |
| 02:26.33 | drmessano | I used to enjoy putting out my smokes on the foreheads of dweebs in high school |
| 02:26.37 | drmessano | Oh wait, nevermind |
| 02:26.44 | drmessano | hides his forehead |
| 02:26.47 | drmessano | :( |
| 02:26.58 | Katty | jaytee: http://www.facebook.com/photo.php?pid=34154315&l=45fdc&id=37617946 |
| 02:27.13 | drmessano | OK STOP |
| 02:27.17 | jameswf-home | Apologies - our website is struggling to cope with the unprecedented |
| 02:27.17 | jameswf-home | demand for the new release 3.0 of OpenOffice.org |
| 02:27.25 | drmessano | APP_CUTENESS is NOT in 1.6 |
| 02:27.56 | seanbright | jameswf-home: FAIL |
| 02:28.32 | seanbright | well, she's a joss whedon fan, she's ok in my book |
| 02:28.43 | Katty | drmessano: http://www.facebook.com/photo.php?pid=34117767&l=44e95&id=37617946 |
| 02:28.51 | jameswf-home | but only 87 peers on torrent hmmmm |
| 02:29.03 | jameswf-home | 200k down i guess thats ok |
| 02:29.28 | jaytee | who's a joss whedon fan? |
| 02:29.40 | seanbright | Katty |
| 02:29.41 | Katty | drmessano: you can't handle my puppeh |
| 02:29.54 | seanbright | or someone in her home is |
| 02:30.03 | seanbright | based on the angel DVDs in that pic |
| 02:30.10 | drmessano | You're sick, Katty.. really, really, sick |
| 02:30.22 | jaytee | so am I. favorite all time show ever was Firefly. |
| 02:30.22 | Katty | seanbright: that would be both of us, actually |
| 02:30.40 | seanbright | well then both of you are ok in my book |
| 02:30.44 | seanbright | points to his book |
| 02:30.56 | drmessano | Next you'll post some pic where he innocently ripped something up chewing on it, and you happened to catch him in a pose where he looked so sorry and sad.. and make us all look at it. |
| 02:31.02 | drmessano | ./ban |
| 02:31.23 | seanbright | Katty: but 28 days later? |
| 02:31.27 | seanbright | revises his book |
| 02:31.43 | Katty | i have that novel |
| 02:31.43 | jaytee | drmessano, see, this is a sign of your tragic space dementia. All paranoid and crotchety, it breaks the heart |
| 02:32.11 | Katty | the movie wasn't too shabby either |
| 02:32.17 | *** join/#asterisk cvnet (n=dahitler@74.210.108.245) |
| 02:32.29 | seanbright | Katty: "wasn't too shabby" is far from a rave review. |
| 02:32.44 | jaytee | Katty!! yay! Bablylon 5. |
| 02:32.44 | *** join/#asterisk Cooltalk (n=io@125.16.91.210) |
| 02:32.45 | Katty | seanbright: it was entertaining! |
| 02:32.47 | Katty | jaytee: YES |
| 02:32.48 | cvnet | general questoins, G729 uses 16kbs per chanel, how does ulaw uses? |
| 02:32.52 | Katty | jaytee: huge b5 fan. |
| 02:32.57 | seanbright | HAH |
| 02:32.57 | drmessano | Ok, heres the question |
| 02:33.00 | seanbright | Willow! |
| 02:33.01 | cvnet | general questoins, G729 uses 16kbs per chanel, how does MUCH bandwith ulaw uses? |
| 02:33.01 | Katty | jaytee: particularly mister garabaldi |
| 02:33.03 | jaytee | "no one listens to Zathras. Zathras not mind |
| 02:33.03 | seanbright | good times |
| 02:33.06 | drmessano | Millenium Falcon or USS Enterprise? |
| 02:33.11 | Katty | poor zathras |
| 02:33.12 | jaytee | Enterprise |
| 02:33.17 | Katty | no one ever listen to zathras |
| 02:33.17 | drmessano | Oh you bastard |
| 02:33.28 | Katty | i don't watch the new crap |
| 02:33.43 | jaytee | although the Millenium Falcon is "the fastest hunk o junk in the galaxy" |
| 02:33.45 | Katty | i actually have a full picture of that case somewhere |
| 02:33.58 | drmessano | USS Enterprise is too over engineered.. and why do they keep having to go all dramatic and seperate the saucer in every 4th Star Trek film? |
| 02:34.05 | drmessano | I'll tell you why.. Design Flaw |
| 02:34.08 | jaytee | "she made the Kessel run in under two parsecs." (whatever the hell that's supposed to mean. |
| 02:34.22 | seanbright | parsecs is a measure of distance, not time. |
| 02:34.25 | Katty | http://webcon.net/~izaah/gallery/v/startrek/ <- back at the old apt. |
| 02:34.26 | seanbright | are* |
| 02:34.41 | Katty | the case has been rearranged somewhat. |
| 02:34.42 | seanbright | cvnet: not sure. |
| 02:35.02 | drmessano | Thats disturbing |
| 02:35.06 | jaytee | seanbright, yes you are correct which is the second reason I want to beat the crap out of George Lucas. The first being Jar Jar Binks |
| 02:35.20 | drmessano | I hope Emperor Palpatine shows no mercy on your soul |
| 02:35.34 | jaytee | seanbright, but that's a line from the first movie |
| 02:35.40 | seanbright | jaytee: yes i know |
| 02:35.56 | seanbright | jaytee: humorously parodied by kevin spacey on SNL |
| 02:36.25 | seanbright | kevin spacey impersonating christopher walken impersonating han solo |
| 02:36.29 | seanbright | truly classic. |
| 02:37.01 | jaytee | if you want to really torture yourself with inaccurate science fiction try the old 50's black and white file Missile to the Moon with Sonny Tufts. Only sci fi film I've ever seen an astronaut take a pack of cigarettes out of his spacesuit pocket to watch it burst into flame in a total vacuum. |
| 02:37.20 | drmessano | William Shatner impersonating william Shatner in Airplane II was my fav |
| 02:37.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:37.27 | jaytee | kevin spacey is awesome but kevin pollack does a better job of shatner and walken |
| 02:37.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:37.51 | Katty | lmadsen: YOU |
| 02:37.53 | Katty | lmadsen: GETO UT |
| 02:38.00 | drmessano | I'm not sure even Shatner knows now how good of a Parody he did of himself |
| 02:38.01 | seanbright | the only funny part of that movie was the "shhh" to open the doors |
| 02:38.04 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:38.07 | drmessano | I loved all of it |
| 02:38.18 | drmessano | and its not "Shhh" |
| 02:38.25 | drmessano | Its "Shoook" "Shoook" |
| 02:38.32 | drmessano | Shook Shook doors |
| 02:38.37 | seanbright | i hear no 'k' |
| 02:38.45 | drmessano | Fail |
| 02:38.51 | Katty | Shook Shook! |
| 02:39.01 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:39.08 | jaytee | "clap on, clap off......" |
| 02:39.16 | Katty | hellllooo computer |
| 02:39.22 | seanbright | heh |
| 02:39.26 | seanbright | fucking whales |
| 02:39.30 | jaytee | "just use the keyboard" |
| 02:39.30 | seanbright | HORRIBLE MOVIE |
| 02:39.59 | seanbright | "how... quaint" |
| 02:40.07 | drmessano | Oh god, his whole rant at the end of the movie |
| 02:40.10 | jaytee | "he did too much LDS in the 60's" |
| 02:40.11 | drmessano | When Striker is landing |
| 02:40.28 | drmessano | That whole bit is classic |
| 02:40.46 | jaytee | Airplane 2 was good |
| 02:40.48 | seanbright | caught airplane on hbo last night |
| 02:40.53 | drmessano | Hes guiding him in and being the most encouraging, abusive, dramatic SOB ever |
| 02:41.00 | seanbright | never get tired of watching that one |
| 02:41.02 | jaytee | "I don't think I'll ever be over Macho Grande" |
| 02:41.36 | seanbright | "a bah--?" |
| 02:41.40 | seanbright | "no, a bomb." |
| 02:41.44 | lmadsen | Katty: in many aspects you are the exact opposite of file, yet in many ways you are the same person |
| 02:41.58 | lmadsen | I don't believe there are many aspects which fall in between those two states |
| 02:42.06 | seanbright | well "Katty" is an anagram for "file" |
| 02:42.11 | jaytee | how would we know since file rarely ever talks |
| 02:42.24 | lmadsen | he does if you know what places to look in |
| 02:42.28 | voxter | lmadsen: hello fellow society member. :) |
| 02:42.42 | drmessano | "We have no tower sir" |
| 02:42.46 | drmessano | "No tower" |
| 02:42.52 | file | moo |
| 02:42.53 | drmessano | "Just a bridge, sir" |
| 02:42.55 | lmadsen | asterisk is like the stonecutters society where lower numbers outrank lower numbers |
| 02:42.55 | voxter | lmadsen: i cant believe those krispy kreme doughnutburger pictures you posted dude.. insane. |
| 02:43.02 | drmessano | "Why the hell arent I notified of these things" |
| 02:43.04 | seanbright | file has a highlight filter for "audihooks bridge muffins" |
| 02:43.04 | lmadsen | voxter: dayton? |
| 02:43.10 | voxter | lmadsen: you bet man. |
| 02:43.25 | voxter | lmadsen: i changed my nick awhile back, i figured it would cause confusion |
| 02:43.26 | lmadsen | ~muffin |
| 02:43.27 | jbot | Filtering proxy server for the World Wide Web written entirely in Java. URL: http://muffin.doit.org/ |
| 02:43.34 | seanbright | fail |
| 02:43.36 | lmadsen | ~muffin is http://leifmadsen.com/images/muffin.jpg |
| 02:43.37 | jbot | ...but muffin is already something else... |
| 02:43.39 | jaytee | Katty, if you ever come to Indy for any reason I'll get you in the zoo free. |
| 02:43.55 | lmadsen | jbot: no, muffin is http://leifmadsen.com/images/muffin.jpg |
| 02:43.56 | jbot | okay, lmadsen |
| 02:43.56 | voxter | lmadsen: whats the deal with the muffin? |
| 02:44.01 | Katty | jaytee: :> |
| 02:44.03 | Katty | file: mew. |
| 02:44.07 | drmessano | I'm not allowed in Indiana |
| 02:44.08 | Katty | lmadsen: i knows. |
| 02:44.12 | Katty | lmadsen: <3 |
| 02:44.12 | file | hugs on Katty |
| 02:44.14 | lmadsen | voxter: but yet I somehow still knew it was you :) |
| 02:44.15 | Katty | hugs lmadsen |
| 02:44.17 | Katty | huggles file |
| 02:44.20 | lmadsen | and I didn't even cheat with a whois |
| 02:44.34 | *** join/#asterisk AndyML (n=AndyML@pool-96-227-91-204.phlapa.fios.verizon.net) |
| 02:44.34 | seanbright | jbot: muffin is also a filtering proxy server for the World Wide Web written entirely in Java. URL: http://muffin.doit.org/ |
| 02:44.35 | jbot | seanbright: okay |
| 02:44.40 | voxter | lmadsen: nice! company name recognition! :) |
| 02:44.46 | file | sleeeepy |
| 02:44.52 | file | blames lmadsen |
| 02:44.54 | jaytee | drmessano, are we allowed to know why you aren't allowed in the Hoosier state? |
| 02:45.04 | Katty | file: go nap. |
| 02:45.17 | lmadsen | seanbright: thanks |
| 02:45.21 | drmessano | jaytee: Not unless you want to be called to testify |
| 02:45.29 | seanbright | lmadsen: my pleasure. |
| 02:45.31 | jaytee | I'll pass then |
| 02:45.34 | seanbright | wanders off to bed |
| 02:45.42 | jaytee | nite seanbright |
| 02:46.11 | Katty | ninite mister lgihtbright |
| 02:46.42 | seanbright | wanders back to beat up Katty |
| 02:46.43 | drmessano | I do have safe refuge during the months of May and October in 2/3rds of the states containing federally recognized indian reservations |
| 02:46.45 | seanbright | wanders off again |
| 02:47.03 | Katty | :P |
| 02:47.04 | drmessano | I cant really go into details |
| 02:47.11 | file | drifts off to bed |
| 02:47.16 | Katty | file: bai |
| 02:48.47 | drmessano | HAW |
| 02:49.02 | drmessano | They have a YouTube vid of all the shatner scenes in Airplane II |
| 02:49.23 | Katty | want nap. |
| 02:50.54 | drmessano | Blinking and beeping and flashing, and blinking, and BEEPING.. AND FLASHING |
| 02:51.22 | drmessano | We all have our buttons, knobs, and switches to deal with |
| 02:52.46 | jaytee | http://www.youtube.com/watch?v=L1IrUAmq4bE |
| 02:54.11 | Katty | nitenite |
| 02:54.17 | jaytee | nite Katty |
| 02:55.03 | *** join/#asterisk RB2 (n=RB2@pool-71-255-92-53.nwrknj.east.verizon.net) |
| 02:55.32 | *** join/#asterisk mikealeonetti (n=mike@ool-18b9d1e4.dyn.optonline.net) |
| 02:55.34 | drmessano | How we survived, I dont know |
| 02:55.46 | drmessano | Howie survived? |
| 02:55.53 | drmessano | No i'm afraid we lost Howie |
| 02:55.56 | mikealeonetti | what is a situation that regcontext might be used? |
| 02:56.07 | mikealeonetti | I'm not sure how it would be useful. |
| 02:56.18 | drmessano | Then you probably dont need it |
| 02:56.32 | drmessano | That'll be $37.50 |
| 02:56.37 | mikealeonetti | heh |
| 02:56.42 | drmessano | PM for Paypal details |
| 02:57.51 | mikealeonetti | well, is it supposed to make it easier to create extensions so you don't have to duplicate lines in your extensions.conf? |
| 02:58.06 | mikealeonetti | or is it really supposed to be for temporary registered devices only? |
| 03:01.06 | voxter | lmadsen: were you a networks guy before you got into asterisk? |
| 03:01.37 | lmadsen | voxter: ummm... I went to school for routing analysis |
| 03:02.13 | voxter | lmadsen: neat stuff. I was just reading some of the papers on your website. the BGP analysis document would have helped me a lot when i was learning about it years ago :) great stuff youve got there |
| 03:02.21 | lmadsen | from physical layer to spanning tree, pim sparse and dense mode, rip, ospf, bgp, etc... |
| 03:02.23 | drmessano | BGP = Best Guess Protocol |
| 03:03.06 | drmessano | RIP = Routes Incomplete Packets |
| 03:03.12 | voxter | lmadsen: i am/was a network engineer prior to my voip foray, and a sysadmin before that. I still do both on a much lesser scale |
| 03:03.36 | drmessano | Oh I love a good network administrator |
| 03:03.38 | jaytee | here?.no..here???..no...aw, screw it! HERE! ...... ERROR: Destination target not found. |
| 03:03.42 | lmadsen | voxter: before asterisk I worked in computer stores building and doing sales |
| 03:03.42 | voxter | lmadsen: we were just acquired by a wireless ISP which i built originally, so ill probably be pulled back into some router and network design work in the near future, i imagine. |
| 03:03.45 | drmessano | That crunchy sound when you bite down |
| 03:04.02 | lmadsen | voxter: also did a bunch of help desk stuff before school |
| 03:04.04 | lmadsen | and during |
| 03:04.35 | De_Mon | you built a wireless ISP changed companies and now that company was aquired by your old company??? |
| 03:04.43 | drmessano | Before Asterisk, I sold aluminum siding to homeless people |
| 03:04.47 | drmessano | It didnt work out |
| 03:04.59 | voxter | I did some of that too, during high school. internship job placement type stuff. i thought it was insanely awesome that at the end of the month we could buy stuff from inventory that hadnt been purchased by clients in (x) amount of time for some really cheap price. |
| 03:04.59 | jaytee | haha |
| 03:05.36 | drmessano | Apparently the housing market crash affected homeless people too |
| 03:05.39 | voxter | De_Mon: i played a hand in building the wisp from a technology perspective yes |
| 03:06.35 | jaytee | Sears used to call me for vinyl siding since they were too friggin stupid to check that I owned a house first. I got tired of them calling so I strung one of their telemarketers along for about 15 minutes asking serious questions then I asked for a quote on how much it would cost to have siding installed on my Nissan Sport Truck. |
| 03:07.45 | De_Mon | "please take me off your list" has worked pretty good for me |
| 03:08.01 | lmadsen | I have an extension I transfer them to |
| 03:08.02 | De_Mon | heads to sleepy town |
| 03:08.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:10.27 | drmessano | mmm bacon wrapped asterisk |
| 03:11.08 | drmessano | I installed ESXi on my shitty test box |
| 03:11.20 | drmessano | Tried to install the 64-bit AsteriskNOW beta |
| 03:11.28 | drmessano | It told me I didnt have a 64-bit processor |
| 03:11.32 | drmessano | Im all like "LOOK UP" |
| 03:11.33 | *** part/#asterisk mikealeonetti (n=mike@ool-18b9d1e4.dyn.optonline.net) |
| 03:12.15 | drmessano | Apparenty VMWARE ESXi is free, so long as you dont want to do anything cool with it |
| 03:12.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:14.51 | drmessano | http://www.chow.com/recipes/11130 <-- Reminds me of my dear, old pappy :( |
| 03:16.25 | jaytee | chocolate dipped bacon wrapped bacon sprinkled with bacon bits |
| 03:17.03 | jaytee | I've always suspected that turducken was invented by someone who couldn't make up their damned mind what poultry to eat. |
| 03:17.17 | drmessano | Actually |
| 03:19.36 | drmessano | I think the turducken was invented when a farmer in louisiana cranked up his truck, it backfired, and a chicken freaked out, ran head first up the business end of a duck, which then, horrified, ran up the business end of a turkey, and it was so horrified it died of a heart attack. The farmer jumped out, saw the shenanigans and said "Lets cook it" |
| 03:20.03 | jaytee | hahaha, excellent!!!!! |
| 03:21.05 | drmessano | Had the farmer had some ostriches around, he would have needed a bigger table |
| 03:21.37 | drmessano | osturducken sounds pretty tasty |
| 03:21.54 | drmessano | Was also the name of my 4th grade science teacher |
| 03:25.46 | slingr | anyone here have an spa3102 |
| 03:26.21 | drmessano | Yah |
| 03:26.25 | drmessano | Still cant get it working? |
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| 03:32.03 | slingr | nah :/ |
| 03:32.09 | slingr | everything is proper |
| 03:33.08 | drmessano | ok |
| 03:33.34 | drmessano | Not sure what to tell ya |
| 03:33.40 | drmessano | But the config works |
| 03:33.47 | drmessano | So theres something not correct |
| 03:35.37 | slingr | how do i set the unit back to factory defaults |
| 03:35.55 | drmessano | **** 73738# |
| 03:36.11 | slingr | tks |
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| 03:48.52 | *** join/#asterisk metfan2007 (n=jc@189.180.216.169) |
| 03:48.57 | metfan2007 | Hi all! |
| 03:49.12 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 03:49.21 | metfan2007 | do you know a linux iax client that supports pulseaudio? |
| 03:55.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:57.44 | slingr | http://dev.savoirfairelinux.net/sflphone/ is on its way |
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| 04:06.09 | metfan2007 | slingr: only for Ubuntu/Debian? |
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| 04:09.53 | slingr | what is SIP/2.0 489 Bad event |
| 04:10.09 | slingr | metfan it'll probably work with other distro's |
| 04:10.11 | slingr | but i use debian |
| 04:21.50 | pputman | Does anyone know if you can dowload the extra sounds in french or spanish? The ftp site only has core sounds for them and extra sounds in english. doing a quick google I see references to a french and spanish extra sounds though, anyone know if they were taken down? |
| 04:23.09 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:39.15 | drmessano | pputman |
| 04:39.18 | drmessano | Are you kidding? |
| 04:39.45 | pputman | why would I be kidding? |
| 04:39.48 | drmessano | http://ftp.digium.com/pub/telephony/sounds/ <---- I see nothing missing |
| 04:40.26 | pputman | drmessano, That list has the core sounds in all languages, but the extra sounds are only in english |
| 04:40.50 | pputman | the question is were they ever there in another language? |
| 04:47.42 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-99cea512e23f4a11) |
| 04:47.57 | Qwell | extra is only english |
| 04:48.00 | Qwell | extras* |
| 04:49.42 | Qwell | all the stuff in core sounds is used in Asterisk, so it was worth it to have all of those recorded in the supported languages |
| 04:50.06 | Qwell | it would be fairly expensive to have the stuff in extra sounds re-recorded |
| 04:51.41 | Qwell | bed |
| 04:55.48 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
| 05:08.20 | slingr | drmessano> what is your PSTN Line SIP port set to? |
| 05:08.41 | drmessano | slingr |
| 05:08.58 | drmessano | You are way overengineering this |
| 05:09.01 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 05:09.07 | drmessano | Let me make it easy |
| 05:09.12 | drmessano | I take a SPA-3102 |
| 05:09.15 | drmessano | Factory Reset |
| 05:09.23 | drmessano | Change ONLY the settings I listed |
| 05:09.26 | drmessano | PERIOD |
| 05:09.32 | drmessano | and it WORKS |
| 05:10.08 | drmessano | No, I am not going to go down the insane slippery slope of comparing settings, tweaking, etc |
| 05:10.15 | drmessano | There is no need, and the work has been done |
| 05:10.25 | slingr | yeah its working |
| 05:10.44 | slingr | i was just wondering what the port you have listed there for default is |
| 05:10.59 | drmessano | I change nothing |
| 05:12.21 | *** join/#asterisk dug (n=chatzill@adsl-71-131-8-80.dsl.sntc01.pacbell.net) |
| 05:12.38 | slingr | you still aren't answering me :P |
| 05:14.36 | drmessano | Im not sure the significance? |
| 05:14.54 | slingr | sigh |
| 05:14.55 | slingr | nevermind |
| 05:15.08 | drmessano | I have whatever the factory default is |
| 05:17.35 | *** join/#asterisk hi365_m (n=hi365@bzq-79-183-238-15.red.bezeqint.net) |
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| 05:42.03 | drmessano | Hmm |
| 05:42.07 | drmessano | How the hell do I remove Zaptel |
| 05:42.15 | voxter | rm -rf / |
| 05:42.24 | drmessano | thats helpful |
| 05:42.39 | voxter | well, it WILL remove zaptel |
| 05:42.43 | voxter | ;) |
| 05:48.18 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 05:50.08 | dug | I just compiled asterisk 1.4.22 and after compiling the new zaptel and running ztcfg -vv and showing all the channels rasterisk still doesnt show any zap commands |
| 05:52.17 | *** join/#asterisk hi365_m (n=hi365@bzq-79-183-238-15.red.bezeqint.net) |
| 06:01.41 | *** join/#asterisk Cutlass (n=chatzill@c-67-176-208-15.hsd1.il.comcast.net) |
| 06:09.31 | dug | how do I make asterisk 1.4.22 compile with zaptel support? I have compiled 1.4.16 without this problem? should I go back a version/ |
| 06:09.33 | dug | ? |
| 06:11.27 | drmessano | dug, you need to read |
| 06:11.36 | drmessano | They're DAHDI commands now |
| 06:15.29 | dug | right ... gotcha... thought zap was still supported (commands and all) according to the docs it was only dropped in 1.6 ... I misunderstood |
| 06:17.31 | dug | thanks |
| 06:20.06 | *** join/#asterisk tengulre11 (n=tengulre@125.71.208.16) |
| 06:23.38 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-81d5d80fd32d7500) |
| 06:25.30 | Cutlass | what are the dependencies for installing app_meetme?...when I do "make menuselect", the option for app_meetme is not checkable (i.e. it has "XXX" and can not be selected) |
| 06:26.00 | Cutlass | I guess I need to install the driver for my FXO card, is that right? |
| 06:27.29 | pputman | Cutlass, it should tell you what dependency is missing, does it say anything about zaptel? |
| 06:27.42 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 06:27.50 | Cutlass | in the menuselect app? |
| 06:28.03 | Cutlass | where would I see that message? |
| 06:28.42 | *** join/#asterisk dug (n=chatzill@adsl-71-131-8-80.dsl.sntc01.pacbell.net) |
| 06:28.54 | *** join/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 06:28.55 | pputman | at the bottom left it should say depnds on dahdi or zaptel, you need to compile one of those first. |
| 06:29.59 | Cutlass | aah! |
| 06:30.10 | Cutlass | it says depends on dahdi (E) |
| 06:30.30 | Cutlass | how do I install that? |
| 06:31.30 | pputman | http://www.asterisk.org/downloads has the source, with documentation on how to install. |
| 06:32.49 | Cutlass | nice!...thanks |
| 06:33.30 | Cutlass | I appreciate the help pputman! |
| 06:33.40 | pputman | yw |
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| 06:44.13 | dug | I got zaptel working with 1.4.22 but I cannot seem to get 1.4.22 to work with the dahdi kernel modules... dahdi_cfg shows all the channels but it wont build with dahdi commands |
| 06:47.46 | *** join/#asterisk mihinomenest (i=Vgah404u@66.255.220.17) |
| 06:48.48 | dug | I have rebooted ... run dahdi_genconf and tested with dahdi_cfg and did a make clean and reinstall of asterisk 1.4.22 and still no dahdi cmds |
| 06:49.01 | dug | in rasterisk |
| 06:49.43 | *** join/#asterisk synchris (n=synchris@athedsl-89958.home.otenet.gr) |
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| 06:51.30 | *** mode/#asterisk [+o file] by ChanServ |
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| 07:12.10 | *** join/#asterisk terlouw (n=naquada@81.175.94.140) |
| 07:12.36 | terlouw | hi there! i have a quick question. When an anonymous caller calls me, asterisk tells me the caller is "Anonymous". Where can i change this? I want it to say "Onbekend" but i really would not know where to start looking for this.... any hints ? :) |
| 07:14.44 | *** join/#asterisk mazpe (n=mazpe@adsl-065-006-163-191.sip.mia.bellsouth.net) |
| 07:15.40 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:17.56 | Greek-Boy | so what is the device name for Dahdi? |
| 07:18.02 | Greek-Boy | it changed from /dev/zap to what? |
| 07:18.18 | justdave | terlouw: probably in your dialplan for the incoming call path |
| 07:18.19 | voxter | Qwell: gone? |
| 07:18.31 | justdave | terlouw: check the callerid to see if it's Anonymous, and change it to what you want if it is |
| 07:21.42 | sivadnz | APP_SMS 1.6 users: having troble getting SMS to answer call, well it won't but it should, I've uploaded patch that moves the intended answer if ont already to the correct place. refer http://bugs.digium.com/view.php?id=13675 report back to bugtracker results please |
| 07:22.18 | terlouw | justdave: i was talking about incoming calls, should have mentioned that :) |
| 07:25.56 | justdave | terlouw: so was I |
| 07:26.14 | justdave | you should have a context that handles incoming calls |
| 07:26.30 | justdave | with either 's' or an "extension number" that matches your DID phone number |
| 07:26.44 | *** join/#asterisk cobex4 (n=cobex4@unaffiliated/cobex4) |
| 07:26.50 | cobex4 | bonjour |
| 07:27.34 | justdave | Greek-Boy: I remember reading it in the release notes for dahdi, but I don't remember offhand. release notes is a good place to look :) |
| 07:28.04 | Greek-Boy | justdave thanks |
| 07:28.25 | justdave | seems like the basics was anything that used to be "zt*" is now "dahdi_*" |
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| 07:36.35 | cobex4 | Hello, speak french ? |
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| 07:54.35 | Cutlass | I'm trying to install app_meetme and under the menuselect, it shows a dependency for "dahdi (E)"...I went to the * website and downloaded/installed "DAHDI Linux 2.0.0" and "DAHDI Tools 2.0.0" but still no luck...does anyone have any suggestions? Did I overlook something? |
| 07:55.42 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 07:56.34 | joobie | hey guys.. is there a name for the technology that pushes configuration to handsets? |
| 07:56.42 | joobie | trying to read morea bout it.. not sure what to google |
| 07:57.40 | sivadnz | joobie: provisioning? |
| 07:58.07 | Cutlass | I thought that was accomplished via tftp (trivial FTP)...for example to configure cisco desk sets and config file can be loaded on the TFTP server and all the desk sets can pull it down at boot up |
| 07:58.17 | joobie | ahh yea |
| 07:58.22 | tzafrir_laptop | Cutlass, what version of Asterisk is it? |
| 07:58.27 | Cutlass | 1.6 |
| 07:58.44 | joobie | but is there a method that can be used for remote management off phones? say when the sip server is offsite and there is no local dhcp server? |
| 07:59.08 | joobie | can say the phones be partially configured and then have their configs remotely upgraded / amended |
| 07:59.09 | tzafrir_laptop | egrep -i 'tonezone|dahdi' build_tools/menuselect-deps |
| 07:59.25 | tzafrir_laptop | Cutlass, ==^ |
| 07:59.55 | Cutlass | ok thanks...I'll try that...stand by... |
| 08:00.23 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 08:00.36 | Cutlass | elect-deps |
| 08:00.38 | Cutlass | DAHDI=0 |
| 08:00.40 | Cutlass | TONEZONE=0 |
| 08:00.49 | Cutlass | not sure what that means :( |
| 08:01.24 | tzafrir_laptop | looks like you have not installed either dahdi-linux (DAHDI=0) and dahdi-tools (TONEZONE=0) |
| 08:01.36 | Cutlass | humm |
| 08:01.54 | Cutlass | is there a particular prefix I should use on "make install"? |
| 08:01.58 | tzafrir_laptop | let's start with dahdi-linux |
| 08:02.09 | Cutlass | ok |
| 08:02.12 | tzafrir_laptop | ls /usr/include/dahdi |
| 08:02.32 | Cutlass | I see the 5 header files |
| 08:03.30 | tzafrir_laptop | could you pasteibn config.log anywhere? e.g. http://paste.debian.net |
| 08:03.57 | *** join/#asterisk pcrack (n=pcrack@116.50.213.210) |
| 08:04.39 | pcrack | hi im going to setup asterisk vicidial, about 60 agents |
| 08:05.11 | Cutlass | one sec... |
| 08:05.45 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:06.23 | pcrack | im thinking to do 3 server |
| 08:06.38 | pcrack | 1 for asterisk, 1 for vicidial and 1 for mysql |
| 08:06.55 | pcrack | what hardware can you recommend? |
| 08:07.07 | Cutlass | http://paste.debian.net/19246/ |
| 08:11.16 | mort_gib | Hi, I'm having an issue with Zaptel, using a Sangoma A200 card. Incoming callerid is not picked up and * then waits for it?? How come it's not coming in?? |
| 08:12.11 | *** join/#asterisk easycrypt (n=savek@ip-174.emscb.ruhr-uni-bochum.de) |
| 08:14.29 | tzafrir_laptop | Cutlass, you don't see to have /usr/include/dahdi/user.h . Or something is badly wrong with your compiler |
| 08:14.47 | tzafrir_laptop | conftest.c:161:24: error: dahdi/user.h: No such file or directory |
| 08:15.13 | tzafrir_laptop | (due to the line: #include <dahdi/user.h>) |
| 08:16.00 | Cutlass | I see... |
| 08:16.09 | Cutlass | there are many other errors in that file also |
| 08:16.32 | Cutlass | is this the output from when I built asterisk? |
| 08:16.34 | tzafrir_laptop | did you run 'make install' from dahdi-linux? did it complete successfully? |
| 08:16.57 | Cutlass | yes....but I did not recompile asterisk...is that the next step? |
| 08:17.33 | Cutlass | everything was fine for dahdi-linux and dahdi-tools |
| 08:18.53 | tzafrir_laptop | Cutlass, what is the output of: ls /usr/include/dahdi |
| 08:19.22 | Cutlass | fasthdlc.h kernel.h tonezone.h user.h wctdm_user.h |
| 08:20.18 | tzafrir_laptop | Cutlass, try re-running ./configure if you installed the dahdi packages after running ./configure |
| 08:20.49 | Cutlass | the configure script for asterisk? |
| 08:20.54 | Cutlass | ...and then recompile? |
| 08:22.00 | tzafrir_laptop | yes, the configure script of asterisk |
| 08:22.35 | Cutlass | I thnk that worked |
| 08:23.04 | Cutlass | I did the configure and now "app_meetme" is checked in the menuselect app |
| 08:23.46 | Cutlass | I guess it's obvious now....once the dahdi stuff is installed you have to rebuild * :) |
| 08:23.56 | Cutlass | thanks a bunch!! |
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| 09:03.19 | ana_micho | hi |
| 09:03.23 | ana_micho | all |
| 09:03.50 | ana_micho | I am getting these every second -- Remote UNIX connection |
| 09:03.50 | ana_micho | <PROTECTED> |
| 09:04.00 | ana_micho | how can I I know which module is connecting ?. |
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| 09:05.55 | synthetiq | im trying to set up a tdm400p fxo card with dadhi using old zaptel style configs (upgraded from zaptel) and having no luck starting asterisk, it tells me there is an errors with dadhi but doesnt not tell me the error, where can i look for errors |
| 09:10.37 | synthetiq | well unloaded the drivers |
| 09:11.12 | synthetiq | and modprobed back drivers and asterisk now loads, but unable to opeen chans 1-4 |
| 09:15.01 | dandre | hello, |
| 09:15.24 | tzafrir_laptop | synthetiq, what error do you get? |
| 09:16.07 | tzafrir_laptop | what do you see on /proc/dahdi/* ? |
| 09:25.50 | tzafrir_laptop | synthetiq, ==^ |
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| 09:31.16 | synthetiq | proc/dahdi = 1 2 3 |
| 09:31.22 | dandre | is it possible to have full control on a communication thru the AGI interface. For instance is it possible to place the party on hold using an agi command? |
| 09:32.11 | synthetiq | starting *, i get.. Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: |
| 09:32.16 | synthetiq | hmm maybe i have a bad chan |
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| 09:33.57 | Greek-Boy | what happened to ztdummy? looks like there is no dummy in dhadi |
| 09:33.57 | Greek-Boy | not even dhadi_dummy |
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| 09:39.17 | pputman | anyone happen to know anything about the smtp settings missing in the gui 2.0 now? Was it moved to a different tab and I just can't find it or taken out completely? |
| 09:47.50 | tzafrir_laptop | Greek-Boy, dahdi_dummy |
| 09:48.21 | tzafrir_laptop | pputman, maybe it is something specific to the asterisk-now branch? |
| 09:49.02 | pputman | tzafrir_laptop, I don't think so, I downloaded the svn 2.0 and tried it as well |
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| 09:49.58 | Tricky_Dicky | Good morning |
| 09:51.47 | Tricky_Dicky | I'm looking for some technical help on ASterisk |
| 09:52.06 | Tricky_Dicky | such as how much bandwidth does a call take up? |
| 09:52.32 | viraptor | Tricky_Dicky: depends on a codec |
| 09:52.49 | Tricky_Dicky | ok |
| 09:53.09 | Tricky_Dicky | which codec uses the lowest amount but still provides a decent call quality? |
| 09:53.50 | viraptor | Tricky_Dicky: gsm probably, or if you're prepared to pay, then g729 |
| 09:54.49 | Tricky_Dicky | and how much would each call use up at a time, I'm on a very small uplink |
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| 09:56.05 | viraptor | Tricky_Dicky: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 09:56.49 | viraptor | all: what can be a reason for asterisk to ignore packets like "OPTIONS" from localhost? there are some threads free to handle calls, cpu is < 25%, sent on loopback |
| 09:57.46 | viraptor | I mean - it responds ok, but sometimes it just starts ignoring them for 2-3 minutes and goes back to normal |
| 09:58.09 | synthetiq | hmm is there anything that can check what is wrong with your dahdi configs, drivers? |
| 09:59.26 | Tricky_Dicky | viraptor: thanks for that |
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| 10:01.16 | flohack | Is there a way to avoid the "SELECT * FROM extension_table WHERE exten LIKE '\\_%' AND..." database lookups when using realtime? I.e. can I disable pattern matching for realtime extensions? |
| 10:01.56 | flohack | It really hits my DB server hard |
| 10:02.28 | petererer | Do you have an index on exten? |
| 10:02.35 | flohack | yes |
| 10:03.06 | flohack | but with nested dialplan contexts it means one like for each nested context |
| 10:04.22 | petererer | :o |
| 10:05.16 | flohack | Definitely not good if you have say, 20 calls in a queue and the queue looks up the agent extension in 'default' (which is necessary). |
| 10:06.46 | flohack | Without the index I had the effect of a 30 SECONDS lag for calling certain extensions and AMI messages.... :-) |
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| 10:09.30 | flohack | Ok, I'll write a patch then |
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| 10:21.24 | flohack | I have used google, but found no propert documentation of the 'options' one can pass to a switch in the dialplan. Can someone please shed some light on it? |
| 10:22.12 | MaliutaLap | ~book |
| 10:22.13 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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| 10:23.41 | flohack | Maliuta: I have tried the book, but it only mentions switch very briefly |
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| 10:28.36 | Specialist1 | hi everyone |
| 10:28.43 | Specialist1 | any folks using a2billing here ? |
| 10:28.47 | flohack | I had another look at the book, but it certainly does not contain any detailed information on switch |
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| 10:54.46 | mark_csi | hi all - could anyone shed some light on a dialplan issue I'm having? |
| 10:55.49 | joat | flohack - there aren't that many options... just the config file name and the corresponding db |
| 10:56.10 | joat | and, concur, the available documentation is sparse |
| 10:56.26 | joat | did manage to get meetme and extensions into realtime tho |
| 10:59.50 | synthetiq | s there anything that can check what is wrong with your dahdi configs, drivers? |
| 11:00.14 | synthetiq | asterisk says my dahdi isnt configured properly but doesnt give the reason |
| 11:03.22 | tzafrir_laptop | what's that output of lsdahdi ? |
| 11:05.13 | synthetiq | sec |
| 11:05.24 | synthetiq | rebooted my machien to try new config |
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| 11:12.49 | synthetiq | ## Span 3: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" |
| 11:12.49 | synthetiq | Use of uninitialized value in string eq at /usr/local/share/perl/5.8.8/Dahdi/Chans.pm line 220. |
| 11:12.49 | synthetiq | <PROTECTED> |
| 11:13.47 | synthetiq | /etc/dahdi/system.conf is simply: fxoks=49-52 |
| 11:16.07 | synthetiq | any ideas tzafrir_laptop ? |
| 11:16.46 | synthetiq | dahdi_scan reports it unconfigured |
| 11:16.55 | synthetiq | must i indicate the span in system.conf? |
| 11:17.08 | synthetiq | since the card is /proc/dahdi/3 |
| 11:17.33 | tzafrir_laptop | synthetiq, cat /proc/dahdi/* |
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| 11:18.11 | synthetiq | Span 3: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" |
| 11:18.11 | synthetiq | <PROTECTED> |
| 11:18.11 | synthetiq | <PROTECTED> |
| 11:18.11 | synthetiq | <PROTECTED> |
| 11:18.12 | synthetiq | <PROTECTED> |
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| 11:26.44 | middleman | Hello folks, I'm having a problem with asterisk where after an incoming call succesfully rings on my 4 SNOM SIP phones, The call can is answered but there is no audio. This is a intermittent problem, sometimes calls work fine and then minutes later there is no audio again. I'm looking for advice as the best place to start looking for an answer as to what is going on. |
| 11:27.28 | UnixDawg | ok is the pbx and the phones on the same network ? |
| 11:27.33 | middleman | yes |
| 11:27.44 | UnixDawg | is this a pstn call or sip trunk call |
| 11:28.57 | middleman | call is coming from BT landline to voiptalk number, this is sent to our first asterisk server in a datacenter which has an AIX2 trunk to a second asterisk server in our office. |
| 11:30.12 | UnixDawg | what ver of asterisk |
| 11:30.42 | middleman | Asterisk 1.2.13 in office |
| 11:30.55 | middleman | and same in datacenter |
| 11:31.01 | UnixDawg | ok 1.2.13 is old |
| 11:31.18 | UnixDawg | I really suggest you look at updating your boxes to 1.4 |
| 11:31.30 | UnixDawg | asn it is the current stable release |
| 11:31.36 | UnixDawg | 1.2 is eol |
| 11:31.40 | middleman | Has been running phones here for about 3/4 years so yes it is an old setup |
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| 11:32.20 | middleman | I agree updating to 1.4 would be nice, don't know how much of an option it is at the moment. |
| 11:35.51 | UnixDawg | it could be a codec issues it could be a firewall issue it could be a network routing issue |
| 11:36.19 | UnixDawg | it could be a iax trunk timing issue also |
| 11:36.44 | UnixDawg | check them 1 by one |
| 11:38.49 | middleman | okay, thanks |
| 11:39.29 | synthetiq | ok i got it to work |
| 11:40.27 | middleman | One last thing, I have noticed that when call is answered and there is no audio, if the calling phone (as in the BT phone I'm calling from) is tapped on the mic you can here a sort of short burst of static on the SIP phone....do you think that this would point more to a codec issue or could the other that you mention cause a problem there. |
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| 11:55.23 | cobex4 | hello, j' have a problem of configuration in sip.conf, I do not manage to parameterize 3 accounts peer and to manage independently |
| 11:56.09 | cobex4 | do you have an example of configuration of several accounts sip peer? thank you for your assistance |
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| 12:31.07 | cobex4 | [13:55] <cobex4> hello, j' have a problem of configuration in sip.conf, I do not manage to parameterize 3 accounts peer and to manage independently |
| 12:31.07 | cobex4 | [13:56] <cobex4> do you have an example of configuration of several accounts sip peer? thank you for your assistance |
| 12:32.43 | yang | cobex4: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
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| 12:33.35 | [TK]D-Fender | wow, clearly run directly through a french-english converter.... and not a particularly bright one... |
| 12:34.58 | tzanger | hahaha |
| 12:35.01 | tzanger | morning [TK]D-Fender |
| 12:35.06 | cobex4 | [TK]D-Fender: french-english \o/ |
| 12:35.08 | gsiener | is there a script to publish an html page of user extensions? |
| 12:35.42 | [TK]D-Fender | gsiener: vi,vim,pico,nano,emacs,gedit,kwrite....... (add 500 names here) |
| 12:36.04 | gsiener | was hoping for a _little_ bit of automation... |
| 12:36.23 | [TK]D-Fender | gsiener: translation = who would care for such a hideous and customization required kind of thing? |
| 12:36.45 | russellb | [TK]D-Fender: ... |
| 12:36.53 | [TK]D-Fender | gsiener: If your CID names are consistent I suppose it would be fairly simple to parse. |
| 12:37.18 | gsiener | yeah |
| 12:37.59 | [TK]D-Fender | russellb: make a phone list out of "extensions"? Lets even assume that its all in sip.conf or users.conf, consider that names might very well be truncated, etc. Not a good basis. |
| 12:38.58 | russellb | yes, it is not a trivial problem. I know that. |
| 12:39.21 | russellb | However, it's still a question that deserves a respectful response, or none at all. |
| 12:39.47 | russellb | carry on :) |
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| 12:45.14 | flohack | In case anyone is interested in disabling extension pattern lookups for realtime dialplans: Here is a patch I just wrote: http://bugs.digium.com/view.php?id=13698 |
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| 12:52.25 | buzzyd | Does anyone have a simple way for removing number such as +441234 to become 00441234 |
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| 12:52.52 | gr0mit | buzzyd, define 'simple' |
| 12:53.23 | gr0mit | in extensions.conf, where? |
| 12:53.41 | buzzyd | gr0mit, easy one two line approach I can add to my existing dialplan |
| 12:53.48 | buzzyd | or anyway of doing it :) |
| 12:54.55 | buzzyd | if I create exten => +_ then strip + is that possible? |
| 12:56.33 | UnixDawg | I want to know why everyone puts down bsd as a os for running linux on? |
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| 12:57.02 | [TK]D-Fender | buzzyd: Yes, thats how. |
| 12:57.05 | UnixDawg | not linux |
| 12:57.11 | UnixDawg | linux/asterisk |
| 12:57.22 | UnixDawg | sorry having a piss me off daay |
| 12:57.33 | UnixDawg | I got hate mail about asterisk on bsd . |
| 12:57.48 | CGMChris | I hate: mail, asterisk, and BSD. |
| 12:57.50 | [TK]D-Fender | UnixDawg: Key reason was getting Zaptel/dAHDI working on it. Aside from that it doesn't matter |
| 12:57.59 | UnixDawg | the fact we are not fast enough keeping the ports uptodate |
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| 12:58.17 | UnixDawg | yeah dahdi has been a iss |
| 12:58.20 | UnixDawg | ue |
| 12:58.25 | [TK]D-Fender | UnixDawg: And of course this reliance on 1 guy maintaining packages for it |
| 12:58.36 | UnixDawg | yeah |
| 12:58.40 | UnixDawg | I agree on that |
| 12:58.52 | [TK]D-Fender | UnixDawg: Packages tend to get made with all sorts of little bugs, don't they? |
| 12:59.02 | UnixDawg | execpt I do alot of the porting but the guy who updates the port in the tree is a pain |
| 12:59.10 | [TK]D-Fender | UnixDawg: Apache? 100 people working on hat, Asterisk? / 100 :) |
| 12:59.25 | UnixDawg | no I have worked out 99% of all the pkg biugs for asterisk |
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| 12:59.45 | [TK]D-Fender | UnixDawg: Sounds like you already had your answer before you asked your questio. Just know that we agree with all of them :) |
| 12:59.50 | UnixDawg | and zaptel 1.4.11 is about rady to be in the ports tree |
| 13:00.14 | [TK]D-Fender | UnixDawg: Perhaps you should think about why YOU had to work them out in the first place. |
| 13:00.51 | UnixDawg | hey I am even updating the asterisk-gui port more today |
| 13:01.48 | UnixDawg | because linux coding slacks |
| 13:01.51 | UnixDawg | lol |
| 13:01.56 | UnixDawg | and lay out |
| 13:02.08 | UnixDawg | we call it usr land lay out |
| 13:03.43 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:03.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:03.52 | seanbright | good morning, ladies. |
| 13:05.10 | UnixDawg | morning seandimm |
| 13:05.19 | Katty | mew. |
| 13:06.41 | Katty | hugs seanbright |
| 13:06.47 | seanbright | hello dear. |
| 13:07.00 | UnixDawg | I have looked at ast linux and its more complicated then it needs to be |
| 13:07.32 | UnixDawg | askozia is nice but it looks like they are even porting to linux |
| 13:07.39 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-115.moldtelecom.md) |
| 13:07.46 | kotique | hi. is there any method to adjust gain for SIP channel ? |
| 13:07.48 | UnixDawg | so maybe its time to find a minilinux distro to play with |
| 13:07.52 | kotique | outgoing audio |
| 13:08.02 | Katty | txgain |
| 13:08.11 | Katty | or set volume, if you're running 1.6 |
| 13:08.32 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 13:09.13 | UnixDawg | I wish digium would make a .img of the new asterisknow with the digium gui |
| 13:10.00 | Katty | i wish this sonic breakfast thingy had more bacon. |
| 13:10.16 | seanbright | i wish jesus would come again |
| 13:10.33 | Katty | too early for religious debate. |
| 13:10.39 | seanbright | heh |
| 13:11.01 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb121-6-19-164.singnet.com.sg) |
| 13:11.03 | Katty | too early for anything but diet dr. pepper and bacon |
| 13:11.12 | UnixDawg | the issue is I have yet to get linux to install on the micro drive I have . it always has issues with acpi and the drive times out |
| 13:11.12 | seanbright | diet dr. pepper... bah! |
| 13:11.28 | Nugget | I didn't know that jesus had a problem with premature ejaculation. |
| 13:11.45 | Katty | probably heraditary |
| 13:11.49 | Katty | his mom was a virgin |
| 13:11.53 | Nugget | heh, true |
| 13:11.54 | UnixDawg | so is the asterisk now 1.5 beta based on asterisk 1.4 or 1.6 ? |
| 13:12.04 | russellb | 1.4 |
| 13:12.08 | russellb | but 1.6 is available as well |
| 13:12.14 | Katty | and jesus is based on russellb |
| 13:12.18 | russellb | or will be at least |
| 13:12.22 | UnixDawg | when |
| 13:12.29 | russellb | I don't know. |
| 13:12.53 | Katty | after russellb comes again. |
| 13:12.57 | UnixDawg | thats not good enough |
| 13:12.59 | UnixDawg | lol |
| 13:13.01 | Katty | on a cloud. |
| 13:13.25 | Katty | telling someone that they're just not good enough is unacceptable. |
| 13:13.44 | russellb | Katty: o.O |
| 13:13.44 | UnixDawg | so I guess I install and update to 1.6 for now |
| 13:13.49 | Katty | it's just mean. |
| 13:13.51 | seanbright | "when it's done(tm)" |
| 13:13.53 | Katty | russellb: <3 |
| 13:13.58 | Katty | russellb: good morning sunshine |
| 13:13.59 | russellb | :-D |
| 13:14.03 | russellb | g'morning |
| 13:14.07 | lmadsen | Katty: good morning moonshine! |
| 13:14.17 | Katty | lmadsen: ewwo lovey |
| 13:14.24 | d3wayne | woot! a full sun today |
| 13:14.32 | Katty | lmadsen: i think it was actually full moon, or close too, last night |
| 13:14.47 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 13:15.09 | russellb | Katty: yeah, it looked cool on my way to work this morning |
| 13:15.50 | UnixDawg | hmmm |
| 13:15.51 | Katty | russellb: did you have some clouds for it to peek through as well? |
| 13:16.09 | russellb | it was in clear sky, actually |
| 13:16.13 | russellb | but it was nice and bright. |
| 13:16.15 | Katty | oh ah |
| 13:16.19 | russellb | and full (or fullish) |
| 13:16.51 | Katty | that calls for a picnic |
| 13:16.53 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 13:17.26 | russellb | are you asking me out? :-p |
| 13:17.34 | russellb | sorry, I'm married. |
| 13:17.56 | Katty | and i'm engaged |
| 13:18.00 | Katty | bring your mrses! |
| 13:18.01 | russellb | ooh |
| 13:18.03 | Katty | we'll have a lovely time |
| 13:18.05 | *** join/#asterisk DarkRift (n=dark@65.92.169.146) |
| 13:18.06 | easycrypt | hi - i'm having a bit of a problem getting chan capi to work. when i call my asterisk box, all i get is: |
| 13:18.06 | easycrypt | <PROTECTED> |
| 13:18.06 | easycrypt | *CLI> |
| 13:18.06 | easycrypt | <PROTECTED> |
| 13:18.06 | easycrypt | <PROTECTED> |
| 13:18.18 | russellb | blinks |
| 13:18.24 | easycrypt | but then nothing happens - the call isn't picked up in the dialplan or anything |
| 13:18.52 | easycrypt | with about 4 or 5 seconds between the two isdn messages |
| 13:19.08 | easycrypt | (asterisk 1.4.21) |
| 13:19.33 | easycrypt | any ideas? |
| 13:19.41 | seanbright | idea 1: don't use CAPI |
| 13:19.45 | seanbright | other than that... |
| 13:19.47 | seanbright | shrugs |
| 13:19.49 | Katty | idea 2: picnic |
| 13:19.57 | seanbright | ohhh |
| 13:19.59 | yidiyuehan | hi, guys, anybody knows when I dial an extension number, how to invoke the System() call and pass an integer to an external C program? like exten => 201,1, System(/usr/local/src/testingC 1)?? |
| 13:20.02 | easycrypt | just got back from lunch ;) |
| 13:20.12 | seanbright | i'll bring the army ants |
| 13:20.16 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 13:20.22 | Katty | yidiyuehan: i know how to pass it to mutt and uhh |
| 13:20.30 | Katty | yidiyuehan: that one thing that makes popups onw indows machines... |
| 13:20.33 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 13:20.35 | Katty | yidiyuehan: can't think of it at the moment |
| 13:20.37 | feeds | waves goodbye |
| 13:20.41 | Katty | yidiyuehan: would you like some pastebinnery? |
| 13:20.46 | Katty | hugs Zeeek |
| 13:20.58 | Zeeek | {{{{{Katty}}}}} |
| 13:21.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:21.16 | Zeeek | shakes all the guys hands |
| 13:21.16 | easycrypt | dialing out on the same interface isn't a problem though |
| 13:21.23 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:21.31 | yidiyuehan | Katty, yes that will be great |
| 13:21.33 | Katty | i'm pretty sure [intra]lanman would be down with a man-hug. |
| 13:21.35 | Katty | yidiyuehan: ksec |
| 13:21.47 | seanbright | yidiyuehan: what happens when you run that line? |
| 13:22.14 | [intra]lanman | Katty: a man-hug? not so sure about that |
| 13:22.28 | yidiyuehan | i did exten => 201, system(/usr/local/src/testingC 1) and call 201, nothing happens, and it didn't pass an integer to testingC program |
| 13:22.37 | Katty | why are men always so squeamish. |
| 13:22.37 | Zeeek | I don't give them out that easily |
| 13:22.41 | yidiyuehan | with CLI it just said executing this line |
| 13:22.44 | seanbright | yidiyuehan: pastebin directory from your extensions.conf |
| 13:22.46 | seanbright | err |
| 13:23.00 | seanbright | yidiyuehan: pastebin your extensions.conf (the part with the System call in it) |
| 13:23.03 | seanbright | ~pb |
| 13:23.03 | jbot | [~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 |
| 13:23.05 | seanbright | yidiyuehan: ^^^ |
| 13:23.21 | Zeeek | and I don't paste my extensions around town, either |
| 13:23.36 | seanbright | i'm a .conf slut, however. |
| 13:23.52 | *** join/#asterisk feeds (n=feeds@85-135-228-100.adsl.slovanet.sk) |
| 13:24.11 | Zeeek | I have my priorities |
| 13:24.12 | yidiyuehan | hi seanbright, a second |
| 13:24.21 | Katty | yidiyuehan: http://pastebin.ca/1227802 <- 3 examples. |
| 13:24.28 | Katty | hugs [intra]lanman |
| 13:24.45 | Zeeek | wants to be pampled |
| 13:24.50 | [intra]lanman | hugs Katty back.... wait... you're not really a man are you? ;-) |
| 13:25.03 | feeds | xD |
| 13:25.07 | seanbright | not anymore |
| 13:25.14 | yidiyuehan | hi seanbright, FYI: http://pastebin.com/m641ac1ea |
| 13:25.18 | Katty | hehe |
| 13:25.26 | Zeeek | "on the internet, no one knows you're a cat" |
| 13:25.33 | Katty | no i've been a whiny female all my life ;) |
| 13:25.38 | yidiyuehan | hi,Katty, let me look at it first. |
| 13:25.44 | Katty | yidiyuehan: kay |
| 13:25.50 | [intra]lanman | Katty: whiny? you don't strike me as the type |
| 13:25.51 | Zeeek | carob is the only milk she drinks (tm) |
| 13:26.02 | seanbright | yidiyuehan: now turn on verbose (core set verbose 10) and pastebin the output from the CLI when you call 201 |
| 13:26.02 | Katty | [intra]lanman: i have my moments ;) |
| 13:26.04 | Zeeek | (or was that soy?) |
| 13:26.07 | [intra]lanman | oic |
| 13:26.46 | Katty | don't drink much meelks. |
| 13:26.52 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 13:26.55 | Zeeek | soy milk isn't. |
| 13:27.05 | Katty | vanilla soy milk is pretty good. |
| 13:27.09 | [intra]lanman | soy is better for you than milk.... |
| 13:27.16 | seanbright | Katty: if you moved some of that stuff to a bash script or something that would be a lot cleaner. |
| 13:27.19 | [intra]lanman | causes less mucasy build-ups that can cause cancer |
| 13:27.20 | cobex4 | Sans dec personne parle français ici ? |
| 13:27.21 | Zeeek | carob milk is the best |
| 13:27.22 | De_Mon | I started asterisk as myself, using the command 'sudo asterisk -C /etc/asterisk.conf' (changes ownership to asterisk/asterisk |
| 13:27.24 | seanbright | (and yes, i knowo it "works") |
| 13:27.31 | yidiyuehan | hi Katty, can you use system call and pass some integer to the external C program? |
| 13:27.33 | Katty | seanbright: yeah probably. |
| 13:27.41 | Zeeek | sans dec, si de nombreuses personnes parle la langue de Molière |
| 13:27.43 | Katty | yidiyuehan: yes--see pastebin. |
| 13:27.47 | Katty | yidiyuehan: i gave you 3 examples. |
| 13:27.58 | De_Mon | the pid's environ confirms my TERM is linux, yet asterisk -c doesn't give me any colors, what gives? |
| 13:28.14 | Zeeek | cobex4: quel est la nature du problème ? |
| 13:28.15 | cobex4 | Zeeek: haaaa super, alors je peux poser mon problème en FR ? |
| 13:28.23 | Zeeek | By the wau, anyone seen TK Def? |
| 13:28.29 | Katty | he was around last night |
| 13:28.32 | Katty | talking about pie. |
| 13:28.38 | Zeeek | was he just out for Thanksgiving? |
| 13:28.41 | Katty | aye |
| 13:28.48 | Zeeek | pei? as in pumpkin? |
| 13:28.49 | Katty | [TK]D-Fender: weren't you? |
| 13:28.51 | Katty | yes. |
| 13:28.52 | Katty | pie. |
| 13:28.54 | UnixDawg | tuning in to the punk channel |
| 13:28.58 | UnixDawg | coding time |
| 13:29.07 | Katty | have fun. |
| 13:29.08 | Zeeek | UnixDawg: dogs are frowned upon |
| 13:29.15 | [TK]D-Fender | Cam back yesterday afternoon |
| 13:29.18 | [TK]D-Fender | came* |
| 13:29.23 | cobex4 | Zeeek: en fait j'ai 3 comptes sip chez freeconet, je suis bien identifier et je peux recevoir des appelles sans problème des 3 lignes, sauf que je voudrais que chaque ligne fasso sonner certin poste |
| 13:29.35 | Katty | that looks like french. |
| 13:29.49 | Zeeek | cobex4: c'est ce qu'asterisk fait par excellence |
| 13:29.50 | cobex4 | Zeeek: mais dé que je tente de séparrer, c'est toujours le premier compte qui sonne |
| 13:30.01 | seanbright | yidiyuehan: now turn on verbose (core set verbose 10) and pastebin the output from the CLI when you call 201 |
| 13:30.05 | Katty | Zeeek: mister suave. |
| 13:30.23 | Zeeek | cobex4: [TK]D-Fender parle un excellent français, il va se mattre àquatre pour te répondre :) |
| 13:30.37 | cobex4 | lol |
| 13:30.47 | [TK]D-Fender | Zeeek: Va t'ens! |
| 13:30.58 | Zeeek | cobex4: mais sérieusement, il faut paster ton dialplan dans le pastebin |
| 13:31.11 | *** join/#asterisk easycrypt (n=savek@ip-186.emscb.ruhr-uni-bochum.de) |
| 13:31.15 | cobex4 | ok |
| 13:31.20 | Zeeek | [TK]D-Fender: les canadiens parle mieux le français que nous autres français |
| 13:31.24 | yidiyuehan | Katty, yes I have looked the pastebin, I am sorry it's a little bit complex, if I just want to paste an integer 1 to C program testingC, is it right like exten => 201, system(/usr/local/src/testingC 1)? |
| 13:31.39 | seanbright | yidiyuehan: now turn on verbose (core set verbose 10) and pastebin the output from the CLI when you call 201 |
| 13:31.45 | seanbright | yidiyuehan: stop making me repeat myself. |
| 13:31.52 | [TK]D-Fender | yidiyuehan: "paste"? thats a parameter like any other program. |
| 13:31.53 | seanbright | everytime i do, a puppy dies. |
| 13:32.01 | Katty | yidiyuehan: ehoe -e "integer 1" | /usr/local/src/testingc)? |
| 13:32.09 | seanbright | ugh |
| 13:32.10 | seanbright | no |
| 13:32.12 | yidiyuehan | hi seanbring, sorry abou that, I will do that |
| 13:32.19 | Katty | seanbright: don't you touch my puppy |
| 13:32.33 | lmadsen | shouldn't that be plural? |
| 13:33.02 | Zeeek | lmadsen: what? parlent or dogs? |
| 13:33.08 | lmadsen | yes |
| 13:33.14 | Zeeek | because yeah, I erred |
| 13:34.17 | Zeeek | but most people here (incl fr speakers) can't spell anyway |
| 13:34.17 | cobex4 | Zeeek: http://pastebin.com/m4222d7fb |
| 13:34.17 | yidiyuehan | D-Fender, 'paste' means just to invoke C program and input an integer like 1 or 0 |
| 13:34.17 | seanbright | sigh... |
| 13:34.19 | Zeeek | cobex4: what is X-Fid ? |
| 13:34.25 | seanbright | Katty: typo in your email text. |
| 13:34.27 | [TK]D-Fender | yidiyuehan: Either way you have not shown us the problem yet. |
| 13:34.29 | Zeeek | oh, nevermind |
| 13:34.30 | Katty | seanbright: no sigh for YOU! |
| 13:34.38 | Katty | seanbright: that was probably on purpose |
| 13:34.48 | cobex4 | oui Zeeek, j'ai suivi la conf fu fourniseur https://www.freeconet.pl/forum/viewtopic.php?t=2430 |
| 13:34.49 | file | tickles Katty |
| 13:34.50 | yidiyuehan | seanbright, brother I am doing it, as I am off of my server so I need to login in remotel |
| 13:34.55 | seanbright | Katty: "Model" is duped in the second to last sentence. |
| 13:35.04 | Katty | seanbright: oh, thanks. |
| 13:35.09 | Katty | snuzzles file |
| 13:35.17 | seanbright | yidiyuehan: super. |
| 13:35.37 | Zeeek | cobex4: où est-ce la définition de freeconet1,2..3 ? |
| 13:36.01 | cobex4 | dabs le context de chaque compte |
| 13:36.05 | cobex4 | dans* |
| 13:36.18 | Zeeek | support génial en français àhttp://Asterisk-France.net |
| 13:36.21 | Katty | disappears for awhile |
| 13:36.34 | cobex4 | Zeeek: sur le tuto du fourniseur, j'ai dans mon sip.conf répliqué 3 fois ce qui est donné |
| 13:36.45 | cobex4 | avec les paramètres pour chaque compte |
| 13:37.21 | Zeeek | cobex4: montre moi une ligne ailleurs qui contient "freeconet2" |
| 13:37.33 | yidiyuehan | D-fender, my problem is: I want to call an extension like 201, and invoke an external C program, and pass a certain integer like 0 or 1 to the C program, but cannot make it |
| 13:37.49 | cobex4 | Zeeek: ailleurs ? |
| 13:37.51 | gr0mit | buzzyd, you can also just exten _+X.,1,Dial(Zap/g1/00{EXTEN:-1}) iirc |
| 13:38.01 | Zeeek | tu peux mettre la ligne ici |
| 13:38.09 | cobex4 | ok |
| 13:38.09 | seanbright | yidiyuehan: PASTEBIN THE OUTPUT FROM THE CLI |
| 13:38.28 | seanbright | it's too early for a fist fight... |
| 13:38.33 | Zeeek | ailleurs que dans le fichier que tu a posté je voulais dire |
| 13:38.47 | seanbright | freedom fries! |
| 13:38.48 | seanbright | heh |
| 13:38.52 | [TK]D-Fender | yidiyuehan: Yes, I keep hearing the problem, but not seeing it. |
| 13:38.57 | seanbright | ok, i give up |
| 13:39.02 | Zeeek | too much punkin pie? |
| 13:39.14 | lmadsen | I don't like pumpkin pie |
| 13:39.26 | [TK]D-Fender | <3 pumpkin pie |
| 13:39.28 | Zeeek | too bad, pass over your slice |
| 13:39.29 | lmadsen | maybe I will now that I'm older and mature :) |
| 13:39.39 | jaytee | I love pumpkin pie and squash pie |
| 13:39.42 | Zeeek | punkin pie is great! |
| 13:39.45 | lmadsen | apple pie and a slice of chedder cheese though... mmmmmmmmm |
| 13:39.52 | Zeeek | and sweet potato pie isn't bad, either |
| 13:39.52 | cobex4 | Zeeek: register => myuser:mypass@sip.freeconet.pl/freeconet2 et aussi context = freeconet2 |
| 13:39.56 | [TK]D-Fender | lmadsen: Nah... a la mode! |
| 13:40.06 | lmadsen | [TK]D-Fender: a la mode + cheese! |
| 13:40.07 | jaytee | but my very bestest fav-o-rite is razzleberry with 'nilla ice cream |
| 13:40.19 | lmadsen | when did we get so hip in here? |
| 13:40.29 | Zeeek | cobex4: il est possible que le provider ne fait pas passer l'info correctement |
| 13:40.30 | jaytee | we've always bee hip |
| 13:40.34 | jaytee | been |
| 13:40.45 | gr0mit | hands jaytee a spare letter 'u' |
| 13:40.49 | cobex4 | Zeeek: si non ma conf peux fonctionner ? |
| 13:41.08 | Zeeek | cobex4: dans quel cas utilise un s extension et fait un NoOP avec la valeur de l'extension appelé |
| 13:41.10 | lmadsen | I need breakfast |
| 13:41.29 | Zeeek | cobex4: si eux t'appellent correctement, ça devait marcher comme ça |
| 13:41.38 | cobex4 | Zeeek: oula désolé, j'ai pas trop compris |
| 13:41.39 | jaytee | gr0mit, u ? |
| 13:41.45 | Zeeek | buys lmadsen a double punkin pie breakfast |
| 13:41.50 | gr0mit | favourite |
| 13:42.24 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:42.37 | Zeeek | cobex4: le context ne doit pas avoir un 2! |
| 13:42.45 | jaytee | I'm an American. We spell favorite just like that. We use color instead of colour and flavor instead of flavour. |
| 13:42.48 | Zeeek | context=freeconet |
| 13:43.06 | gr0mit | knows - he was just stirring the pot a little ;-) |
| 13:43.12 | cobex4 | juste me suis trompé c'est context = freeconet |
| 13:43.22 | Zeeek | cobex4: ah... |
| 13:43.45 | jaytee | and we pronounced schedule as sked-jewel, not shed-yule. |
| 13:44.03 | *** join/#asterisk jplank (n=gbove@reports.nyigc.net) |
| 13:44.12 | Zeeek | cobex4: rajout une ligne en tête du context freeconet: s,1,NoOp(${EXTEN} ) |
| 13:44.22 | gr0mit | jaytee, I know - i work for a Very Big (but Shrinking ) American company |
| 13:44.27 | jplank | anyone play with PIKA's WARP box? |
| 13:44.40 | cobex4 | Zeeek: elle fait quoi cette ligne ? |
| 13:44.43 | Zeeek | jplank: please, this is a PG13 channel! |
| 13:44.52 | jplank | lol |
| 13:45.03 | Zeeek | cobex4: si tu ne sais pas, il faut que tu trouve la réponse en lisant des docs |
| 13:45.19 | jaytee | wow, I never realized how retarded Gnomebaker was for copying. |
| 13:45.23 | cobex4 | ben justement, 1 semaine que je ne fait que ça |
| 13:45.37 | Katty | i leave for a few minutes and i miss the pumpkin pie and liberty fries! )= |
| 13:45.44 | Zeeek | làcommence le dialogue classic: "j'ai déjàlu..." |
| 13:45.55 | cobex4 | lol |
| 13:45.57 | [TK]D-Fender | Zeeek>cobex4: rajout une ligne en tête du context freeconet: s,1,NoOp(${EXTEN} ) <--- you've got to be kidding me... |
| 13:46.06 | seanbright | yidiyuehan: any luck? |
| 13:46.14 | Zeeek | cobex4: donc, tu vas aller voir ce fait "NoOp" et l'extension 's' |
| 13:46.23 | cobex4 | oki |
| 13:46.26 | Katty | NoOp( [TK]D-Fender ) |
| 13:46.41 | Zeeek | [TK]D-Fender: did I forget something? |
| 13:46.49 | jaytee | I love my country and all but all that bullshit about freedom fries or liberty fries is just right wing redneck hysteria. I'd rather drive to Montreal for a heap big serving of poutine. |
| 13:46.51 | [TK]D-Fender | 8(Katty) --- nomnomnomnomnomnomnom |
| 13:47.07 | [TK]D-Fender | Zeeek: what is ${EXTEN} in that case? |
| 13:47.08 | UnixDawg | cool |
| 13:47.11 | gr0mit | poutine?! |
| 13:47.17 | Zeeek | errrrrrrrr |
| 13:47.18 | seanbright | freedom fries! |
| 13:47.22 | UnixDawg | asterisknow 1.5 iso is installing on my micro drive |
| 13:47.28 | jaytee | fries with gravy and cheese curds |
| 13:47.28 | Zeeek | isn't that the EXTENsion? It's been soooo long |
| 13:47.31 | Katty | libery cabbage! |
| 13:47.43 | *** join/#asterisk jmacz (n=c81a9f2a@acuario.unicauca.edu.co) |
| 13:47.45 | seanbright | liberty too! |
| 13:47.47 | [TK]D-Fender | Zeeek: You need to think about what the value might be... |
| 13:47.48 | gr0mit | shudders |
| 13:47.49 | Zeeek | `do I have to go look for my old config files? |
| 13:47.52 | UnixDawg | it might just work on the aklix board |
| 13:47.56 | UnixDawg | we will see |
| 13:48.06 | gr0mit | my daughter used to get books from the libery |
| 13:48.06 | Zeeek | oh, you mean it will be 's' ? |
| 13:48.08 | Katty | dial(zap/g1/${EXTEN}) |
| 13:48.29 | Zeeek | I dunno, how can he tell what extension was called? |
| 13:49.04 | Zeeek | maybe exten => _ something? |
| 13:49.25 | jmacz | Hi everyone, any idea how to make asterisk take an old astdb file (from a previous instalation) without CLI access (-r is not working) and without restarting Asterisk? |
| 13:49.36 | Zeeek | <PROTECTED> |
| 13:49.42 | jmacz | maybe a dialplan func? |
| 13:51.07 | Katty | dialplan disfunction |
| 13:51.15 | Katty | that should be a song. |
| 13:52.40 | UnixDawg | ok this sucks |
| 13:52.42 | UnixDawg | dood |
| 13:52.57 | UnixDawg | I just did a asterisknow 1.5 install its over a gig |
| 13:53.02 | UnixDawg | why so much crap |
| 13:53.07 | UnixDawg | jeesh |
| 13:53.11 | Katty | it's the liberty cabbage. |
| 13:53.13 | nikko | anyone using redfone T1-TDMoE Bridges? |
| 13:53.20 | Katty | possible the freedom fries. |
| 13:53.24 | UnixDawg | digium needs to roll a smaller linux |
| 13:53.49 | De_Mon | its called asteriskLater |
| 13:53.52 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 13:54.03 | Katty | hehe |
| 13:55.23 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:55.43 | fskrotzki | f05j19s63 |
| 13:56.23 | yidiyuehan | hi Seanbright, finally I got the paste bin: http://pastebin.com/m2cdd9d22 |
| 13:56.29 | Katty | hugs fskrotzki |
| 13:56.59 | seanbright | yidiyuehan: sweet. do it again with 'core set verbose 10' and 'core set debug 10' |
| 13:57.07 | fskrotzki | hugs Katty back.. :-) |
| 13:57.35 | yidiyuehan | ok, seanbright, FYI /src/asterisk is an executable C program |
| 13:57.57 | seanbright | you mean /usr/local/src/parallel/asterisk |
| 13:57.57 | seanbright | ? |
| 13:58.08 | yidiyuehan | yes seanbright, |
| 13:58.18 | seanbright | yidiyuehan: who is asterisk running as? asterisk? root? |
| 13:58.21 | yidiyuehan | I just happen to name it as asterisk although it's an external C program |
| 13:58.28 | *** join/#asterisk johnakabean (n=none@pool-72-82-113-223.nrflva.east.verizon.net) |
| 13:58.32 | yidiyuehan | Seanbright, it's running under root |
| 13:58.46 | Katty | someone pass the pain killers. |
| 13:58.48 | *** join/#asterisk axisys (n=axisys@117.18.229.185) |
| 13:58.53 | seanbright | yidiyuehan: ok. and if you run it from the command line, what happens? |
| 13:59.17 | johnakabean | yidi, make sure all files that are being used by asterisk are owned by root |
| 13:59.25 | johnakabean | since its running as root |
| 13:59.36 | seanbright | uh |
| 13:59.43 | Katty | seanbright: no uhs please. |
| 13:59.47 | Katty | seanbright: be nice. |
| 14:00.02 | [TK]D-Fender | yidiyuehan: go run it by hand and see what happens |
| 14:00.22 | yidiyuehan | Hi, seanbright, I set the debug and CLI seems to be the same |
| 14:00.24 | seanbright | debugging by committee |
| 14:00.25 | seanbright | super. |
| 14:00.47 | Katty | i think a hugging committee would be more fun |
| 14:00.48 | seanbright | yidiyuehan: run '/usr/local/src/parallel/asterisk 1' from the linux command line |
| 14:00.54 | yidiyuehan | seanbright, if I run it and nothing happens and it seems not passing any integer to my C program |
| 14:01.13 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-085-014.pools.arcor-ip.net) |
| 14:01.17 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 14:01.51 | seanbright | yidiyuehan: how do you know it's not getting the 1 passed to it? |
| 14:01.56 | yidiyuehan | hi seanbright, If I run the command in another separate C program, it works well |
| 14:02.03 | [TK]D-Fender | yidiyuehan: So it doesn't work when called by hand from *NIX CLI? |
| 14:02.11 | yidiyuehan | seanbright, because there is no output printed out. |
| 14:02.27 | seanbright | yidiyuehan: the output isn't piped back to asterisk |
| 14:02.50 | Blackvel | can you help me with English IVR prompts? what's the correct use of English? one, two, three... or first, 2nd, 3rd, 4th? (I just say "please choose from the following options between ...") |
| 14:03.01 | yidiyuehan | D-Fender, it works if running alone, here is the output> http://pastebin.com/m3502e13a |
| 14:03.13 | defswork | Blackvel: 1, 2 3..... |
| 14:03.19 | yidiyuehan | Seanbright, if running alone it works,: http://pastebin.com/m3502e13a |
| 14:03.20 | seanbright | yidiyuehan: an app run by the asterisk System() application does not print output to asterisk |
| 14:03.32 | seanbright | yidiyuehan: you aren't listening |
| 14:03.39 | yidiyuehan | seanbright, you are right |
| 14:03.50 | [TK]D-Fender | yidiyuehan: And what is that expected to accomplish when executed from *? |
| 14:03.52 | yidiyuehan | that's what I am requesting for help |
| 14:04.21 | yidiyuehan | D-Fender, I expect it passes an integer 1 to this external C program when executed from * |
| 14:04.21 | [TK]D-Fender | yidiyuehan: Make it an AGI and VERBOSE it |
| 14:04.33 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-2938b9beec89f8b1) |
| 14:04.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.27 | yidiyuehan | D-fender, first of all is it correct to do something like exten => 88888, system(/usr/local/src/parallel/testing 1) under *? |
| 14:05.54 | seanbright | in a way, i'm kinda glad [TK]D-Fender bogarted this one |
| 14:06.55 | [TK]D-Fender | yidiyuehan: Yes |
| 14:07.20 | De_Mon | bogarted... |
| 14:07.23 | [TK]D-Fender | yidiyuehan: Forgiving your constant dialplan errors |
| 14:07.46 | seanbright | De_Mon: inserted-himself-into-though-it-was-already-covered |
| 14:08.15 | De_Mon | seanbright where did that term come from? Humphrey Bogart do this frequently? |
| 14:08.31 | seanbright | i think the slang means 'to steal' |
| 14:08.38 | seanbright | not sure where i heard it or where it came from |
| 14:08.40 | Blackvel | defswork: thanks |
| 14:08.54 | Zeeek | to keep it to one's self as in don't borgart that jojnt, my friend |
| 14:08.58 | seanbright | ahhhh |
| 14:08.58 | De_Mon | yeah -- the urban dictionary told me what it meant, but not why |
| 14:09.00 | seanbright | yeah, i was off. |
| 14:09.02 | yidiyuehan | D-Fender, I am sorry about the errors as it's only a testing server, but how come it doesn't work under * although it works alone |
| 14:09.07 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 14:09.12 | seanbright | heh |
| 14:09.32 | Zeeek | Song by Fraternity of Man |
| 14:10.03 | [TK]D-Fender | yidiyuehan: It does work, its jsut that * doesn't care that your program outputs to STDOUT. It is not *'s job to pipe that program's output into its own CLI output |
| 14:10.12 | yidiyuehan | seanbright, I just saw our message, are you saying I couldn't output an integer to external program by running an application under * using system()? |
| 14:10.20 | [TK]D-Fender | yidiyuehan: You want your app to print to * CLI? AGI + Verbose <- |
| 14:10.35 | [TK]D-Fender | yidiyuehan: Is there some apart of this that is not entirely clear at this point? |
| 14:10.37 | seanbright | yidiyuehan: [TK]D-Fender has you covered. |
| 14:11.03 | De_Mon | yidiyuehan exten => s,n,TrySystem(/home/asterisk/notify-callqueues SupportMenu \'${CALLERID(name)}\' \'${CALLERID(num)}\' \'${PGSQL_GET_AREACODE(${CALLERID(num):0:3})}\' \'${CDR(start)}\') |
| 14:11.17 | seanbright | haha |
| 14:11.18 | De_Mon | yidiyuehan that's how I pass parameters to a script |
| 14:11.20 | seanbright | weeeeeeeeeeeeeeeeeeeee! |
| 14:11.25 | seanbright | wanders off. |
| 14:11.43 | yidiyuehan | D-Fener, as far as the the program is concerned, I just want to output an integer to the external C program once executed by *, that's it, no need to print to * CLI |
| 14:12.27 | [TK]D-Fender | yidiyuehan: You are already passing the parameter. What leads you to believe that it isn't processing properly? |
| 14:12.34 | yidiyuehan | De_Mon, that maybe works, I will give a try, thanks |
| 14:12.50 | [TK]D-Fender | yidiyuehan: Your syntax in the dialplan you pasted was jsut fine |
| 14:13.19 | yidiyuehan | D-Fender, because if I run separately, the LED connected will be on if I pass an integer 1. however nothing happens I run execute under * |
| 14:13.27 | *** join/#asterisk virtexPro (n=virtex5@213.150.163.105) |
| 14:14.14 | De_Mon | yidiyuehan I'd wager that it has something to do with the user asterisk is running under vs the user your testing with |
| 14:14.23 | seanbright | asterisk is running as root |
| 14:14.40 | De_Mon | wanders off hoping nobody will notice |
| 14:14.53 | yidiyuehan | De_Mon, all the programs are running under root as it's a testing server and I login in as root |
| 14:14.58 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:15.05 | [TK]D-Fender | yidiyuehan: I share De_Mon's guess as it appears you are running FreePBX |
| 14:15.21 | [TK]D-Fender | yidiyuehan: It matters what user ASTERISK is running as. |
| 14:15.46 | De_Mon | Emo! |
| 14:15.53 | yidiyuehan | D-Fender, yes I am using freepbx, or mabe I need to set the c program as asterisk user? |
| 14:16.48 | [TK]D-Fender | yidiyuehan: I don't think I even want to validate that.... |
| 14:17.23 | [TK]D-Fender | yidiyuehan: Just go looka t what you're actually doing. |
| 14:18.11 | yidiyuehan | D-Fender, I am really sorry as I don't know where the problem is |
| 14:19.11 | [TK]D-Fender | yidiyuehan: Every FreePBX install I've ever seen was as user "asterisk". Go pay attention to what you're doing. |
| 14:19.39 | [TK]D-Fender | yidiyuehan: Not knowing that is by itself a really bad sign about your own setup. |
| 14:21.13 | yidiyuehan | D-Fender, yes the user will be asterisk. and the user of the C program is root, or I need to give up on the freepbx in order to run * as root? |
| 14:22.09 | yidiyuehan | does that will solve my problem? |
| 14:22.17 | [TK]D-Fender | yidiyuehan: this is UNIX 101. If the user can't run your program then I guess it jsut isn't going to work out so well for you will it? |
| 14:22.35 | [TK]D-Fender | yidiyuehan: make the program executable by the user you expect to call it. |
| 14:22.53 | Zeeek | a great looking icon will be important |
| 14:23.39 | yidiyuehan | D-Fender, ok I will make asterisk as the user of the C program and test it again. thanks so much brother. |
| 14:23.50 | yidiyuehan | thanks to seanbright and De_Mon as well man. |
| 14:23.52 | Zeeek | Burghburg |
| 14:25.27 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 14:27.31 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-d5c146213172cf4a) |
| 14:27.31 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:27.36 | *** join/#asterisk hi365_m (n=hi365@bzq-79-176-238-74.red.bezeqint.net) |
| 14:29.11 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 14:33.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:33.47 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 14:36.43 | Assid | hrmm |
| 14:37.53 | Katty | hugs anthm |
| 14:38.07 | anthm | hi |
| 14:38.40 | Katty | hewwo. |
| 14:39.20 | Assid | err.. who here runs flowroute? |
| 14:40.35 | UnixDawg | ok the asterisknow does not work on alix boards |
| 14:40.37 | UnixDawg | lol |
| 14:41.06 | russellb | Assid: implicit |
| 14:41.43 | Assid | wasnt it shrike? |
| 14:41.59 | russellb | it might be more than one person, but I know that he is involved. |
| 14:42.11 | russellb | and if you knew, why would you be asking? |
| 14:42.11 | russellb | :-p |
| 14:42.21 | *** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 14:42.33 | Assid | was jyust confirming |
| 14:43.11 | Assid | implicit: _ShrikE you guys around? |
| 14:44.15 | _ShrikE | I dont work for flowroute |
| 14:45.03 | Assid | was it some other co? |
| 14:45.08 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:45.12 | Assid | bah |
| 14:45.43 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.199.65) |
| 14:46.07 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 14:49.00 | [TK]D-Fender | Only reason people talk about FlowRoute is because of that retard video with Mitnick. And because of it we're flooded with altogether too many retard kiddies who want to be a "l33t hacker". |
| 14:49.19 | [TK]D-Fender | Assid: their rates were nothing to write home about. |
| 14:49.43 | *** join/#asterisk mog (n=mog@nat/digium/x-59b99dc4c3d98e41) |
| 14:49.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:50.06 | Assid | [TK]D-Fender: you know any other decent ITSP by chance with similar rates ? |
| 14:50.10 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 14:50.13 | Assid | rapidvox has been acting strange as of late |
| 14:50.22 | seanbright | hopefully without a high 'retard' count |
| 14:50.48 | Assid | hehe |
| 14:51.39 | Katty | hugs anonymouz666 |
| 14:51.41 | Katty | hugs mog |
| 14:51.46 | Assid | ~itsp |
| 14:51.47 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 14:51.52 | Assid | !itsp |
| 14:51.53 | mog | hey katty |
| 14:51.55 | anonymouz666 | hi Katty |
| 14:52.05 | Assid | ~itsp-us |
| 14:52.17 | seanbright | you can PM jobt |
| 14:52.19 | seanbright | jbot |
| 14:52.29 | seanbright | <PROTECTED> |
| 14:52.40 | seanbright | thanks for playing. |
| 14:52.55 | Assid | hehe |
| 14:53.04 | Assid | which ones would you guys suggest |
| 14:53.27 | *** join/#asterisk n3hxs (n=HAMming@216.64.72.226) |
| 14:53.29 | Assid | at around similar rates to rapidvox - flowroute. voipjet |
| 14:54.33 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 14:56.42 | [TK]D-Fender | Assid: Go look for yourself. We aren't here to compare prices for you |
| 14:57.24 | Assid | but you personallly dont think froute is worth it? |
| 14:59.37 | [TK]D-Fender | Assid: Go do the math |
| 15:00.04 | Assid | 1.3c/min right? |
| 15:00.13 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.36 | [TK]D-Fender | Assid: I'm not looking this up for you. |
| 15:00.37 | *** join/#asterisk johnakabean (n=none@pool-72-82-113-223.nrflva.east.verizon.net) |
| 15:00.54 | Assid | no no.. thats what it is.. 1.35c/min |
| 15:01.06 | johnakabean | Hey guys. How can I build asterisk 1.6 and keep 1.4 running for production while having 1.6 on say ports 5070 and such |
| 15:01.40 | jaytee | "Ten percent of nuthin' is...let me do the math here...nuthin' into nuthin'... carry the nuthin'..." |
| 15:02.17 | johnakabean | assid, why would you pay by the minute for outgoing calls |
| 15:02.36 | johnakabean | other than internation |
| 15:03.00 | Assid | johnakabean: need to US48 .. and decent quality |
| 15:03.05 | Assid | also multiple channels |
| 15:03.38 | Assid | would appreciate any help here |
| 15:03.38 | johnakabean | i pay 29.95 for perfect ulaw qualtity and as many channels I want to put online |
| 15:03.57 | Assid | johnakabean: where? |
| 15:04.06 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:04.12 | johnakabean | broadvoice but I have a dialplan hack |
| 15:04.19 | johnakabean | I can also have 3 didÂ’s per account |
| 15:04.37 | jaytee | anyone using Polycom 330's experiencing lcd flickering after upgrading from sip 2.1.2 to 2.2.0? |
| 15:04.50 | johnakabean | using distinctive ring from SIP header, I translate them into separate DIDÂ’s |
| 15:05.01 | johnakabean | and broadvoice allows unlimited incoming lines |
| 15:05.15 | johnakabean | but i made it where I can have unlimited outgoing channels |
| 15:05.16 | Assid | last time i checked.. call quality on broadvoice isnt something people liked too much |
| 15:05.27 | johnakabean | one thing I have never had a problem with |
| 15:05.39 | johnakabean | I switched from voip.com with terrible phone quality to them |
| 15:05.57 | Assid | johnakabean: pm ok? |
| 15:05.57 | johnakabean | I have never had bad quality but I have a 98% quality of service rating for my ISP |
| 15:06.01 | johnakabean | yeah |
| 15:08.42 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 15:10.39 | *** part/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 15:11.54 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 15:11.56 | casix | hello |
| 15:12.13 | casix | how can I know if an extension is calling or is free? |
| 15:12.13 | *** join/#asterisk shaw22dog (n=shaw@pacman.oaklandcorp.com) |
| 15:13.18 | shaw22dog | Hello, do I need to worry about overruns and frames on my Sangoma card? |
| 15:13.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:14.12 | buzzyd | [TK]D-Fender, thanks works a treat |
| 15:14.58 | [TK]D-Fender | casix: First "extensions" don't call, they are numbers in your dialplan. For "devices" -> "core show application chanisavail" |
| 15:15.31 | cesar_CR | casix, what about core show channels |
| 15:16.14 | shaw22dog | When I do an ifconfig my w2g1 devices has a large amount of RX overruns and frames, is this something I should worry about? |
| 15:16.57 | [TK]D-Fender | shaw22dog: I would. |
| 15:17.08 | [TK]D-Fender | shaw22dog: Check your CPU & IRQ load |
| 15:17.28 | casix | [TK]D-Fender: thank you :) |
| 15:18.02 | casix | cesar_CR: yes but with core show channels I have to parse the result |
| 15:18.47 | [TK]D-Fender | casix: You never said HOW you wanted to check or what kind of action you might want to take based on it. therefor either approach could be valide. |
| 15:20.02 | shaw22dog | [TK]D-Fender: Brand new server, cpu and irq load look okay. I was reading on Sangoma's website and they mention checking my hard drive I/O. Does that make sense? |
| 15:20.04 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:20.38 | *** join/#asterisk xloafx (n=Sean@rrcs-72-45-234-5.nys.biz.rr.com) |
| 15:20.43 | pif | hi, using 1.4.20 I see a different callerid on my phone and on the cdr's 'src' field, what gives? |
| 15:20.47 | [TK]D-Fender | shaw22dog: If they tell you to, then do it |
| 15:20.50 | shaw22dog | [TK]D-Fender: Specifically my dma setting. |
| 15:21.03 | shaw22dog | [TK]D-Fender: Okay. |
| 15:22.40 | Assid | johnakabean: broadvoice claims business plans are limited to a single channel only |
| 15:22.48 | Assid | and if i need more .. i gotta buy more accounts |
| 15:23.05 | shaw22dog | [TK]D-Fender: Hmmm... that tested fine. They recommond 40 MB/sec, and I'm getting 103 MB/sec |
| 15:23.30 | Assid | man i cant find a half decent call provider |
| 15:27.33 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:28.14 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-169.lsanca.dsl-w.verizon.net) |
| 15:28.20 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 15:28.31 | Assid | is confused |
| 15:29.39 | *** join/#asterisk crevetor (n=crevetor@IP-208-88-110-90.mtl.fibrenoire.ca) |
| 15:29.46 | crevetor | hi |
| 15:30.58 | crevetor | Quick question : in asterisk 1.6 I want to receive a fax using t.38. Unfortunately Asterisk doesn't seem to detect that what it receives is a fax and doesn't reinvite the caller for t.38 communication |
| 15:31.45 | crevetor | Am I getting something wrong ? If not how would I get asterisk to detect a fax on a SIP call |
| 15:33.02 | Assid | brb |
| 15:33.48 | jstocks | If I need to get FXS ports, what would be the best device to use? |
| 15:34.04 | [TK]D-Fender | jstocks: Depends how many |
| 15:35.23 | nikko | jstocks: and your budget |
| 15:35.30 | jstocks | [TK]D-Fender: would be mostly just line here or there, mainly at remote locations. |
| 15:35.54 | [TK]D-Fender | jstocks: "line"? What is it you're looking to plug in here? |
| 15:37.17 | nikko | jstocks: like an ATA? |
| 15:37.34 | jstocks | [TK]D-Fender: My setup would be a central PBX system at our office, but have an extention/extra line at home or something that I can plug in any where and be on the office line. |
| 15:37.36 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 15:37.59 | jstocks | so yeah just a standard phone that you would plug in to a pstn network |
| 15:38.27 | [TK]D-Fender | jstocks: Ok, your second description was very broken, but to be clear, you want a device you plug a PHONE into, correct? |
| 15:38.40 | jstocks | yes |
| 15:38.49 | [TK]D-Fender | jstocks: Linksys PAP2T-NA ro SPA-2102 then. |
| 15:38.51 | [TK]D-Fender | or* |
| 15:39.49 | jstocks | ok, I saw the PAP2T-NA, I was just wantting to ask before I did any thing. |
| 15:39.53 | nikko | jstocks: a step up for business would be an adtran TA (Total Access) device |
| 15:39.58 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 15:39.59 | Assid | hey [TK]D-Fender, what was wrong about flowroute? i dont really care about the video about the hidden callerid thing. |
| 15:40.29 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 15:41.04 | [TK]D-Fender | Assid: You seem obsessed with them. Just go ahead and they them, I doubt it'd matter what anyone has to say at this point. |
| 15:41.20 | *** join/#asterisk purple_v45 (n=rmarc@71-91-227-115.static.stls.mo.charter.com) |
| 15:42.36 | Assid | [TK]D-Fender: nah.. im not.. honestly.. im trying to find alternate itsp's .. broadvoice claims 1 channel on their unlimited business.. and their setup is kind of high.. rapidvox i am using, but recently has been going down. voipjet doesnt do call centers EVEN if the job of the call center in 70-75% incoming |
| 15:42.43 | Assid | i honestly am open to suggestions |
| 15:42.51 | purple_v45 | I've got a iax setup that works for most things, but I can't call a particular NPANXX. The call terminates and there's an "s" in the destination field of the CDR. Wondered if anyone knew what that meant so I can try to find a solution. |
| 15:42.53 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:44.11 | *** join/#asterisk cirosou (n=ciro@201.20.206.172.corp.ajato.com.br) |
| 15:44.23 | cirosou | hello everyone |
| 15:44.31 | Assid | [TK]D-Fender: another thing is the prices must be around the same as rapidvox/voipjet .. or even the unlimited ones |
| 15:44.32 | Blackvel | anyone with the problem of electronic noise on a snom 370 mic? |
| 15:44.58 | rwaite | i am trying to tune my rxgain and txgain in chan_dahdi.conf, but im following this document and it says to try to get the rx (in this case) as close to 14844 as possible but dahdi_monitor has no numbers only a graph? |
| 15:45.04 | rwaite | am i missing something here? |
| 15:45.16 | cirosou | i ned some help with and zaptel wcfxo... asterisk keeps telling me that could not create channel of type Zap, but dahdi is correclty configured |
| 15:47.02 | *** part/#asterisk jplank (n=gbove@reports.nyigc.net) |
| 15:47.37 | Assid | [TK]D-Fender: so if you do have any suggestions.. i am open to it |
| 15:48.17 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 15:48.22 | gaetronik | Hi there |
| 15:48.27 | gaetronik | v a question about queues |
| 15:48.32 | tzafrir_laptop | cirosou, please pastebin: cat /proc/zaptel/* ; /etc/asterisk/zapata.conf |
| 15:48.46 | gaetronik | can i disable press '*' to hangup |
| 15:48.54 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:48.55 | gaetronik | i changed features.conf |
| 15:49.11 | gaetronik | and made an module reload res_features.co |
| 15:49.15 | gaetronik | but it still work |
| 15:49.28 | gaetronik | does the agent need to logoff and login |
| 15:49.38 | cirosou | just a moment gonna fetch these files... the asterisk server has been shutdown... |
| 15:49.49 | gaetronik | in order to take care of change |
| 15:50.16 | cirosou | but /proc/zaptel/1 has Wildcard X100P etc... correct |
| 15:50.34 | tzafrir_laptop | cirosou, I want to see the exact line |
| 15:51.21 | tzafrir_laptop | specifically: the exact line of the channel (the one that begins with " 1" |
| 15:51.23 | tzafrir_laptop | ) |
| 15:52.15 | cirosou | ok i'm powering the server again... will have the information on a few moments |
| 15:52.57 | cirosou | but the mais problem is that the x100p seem to take the line off hook all the time... i dont know if this is hardware problem |
| 15:53.40 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:54.42 | Kobaz | how would i get a queue to keep ringing a phone while playing a track to the caller |
| 15:54.52 | Kobaz | <PROTECTED> |
| 15:54.58 | Kobaz | and the phones stop ringing |
| 15:55.07 | Kobaz | and then after the track, the phones start ringing again |
| 15:55.07 | gaetronik | no one for this queue issue? |
| 15:57.08 | *** join/#asterisk ManxPower (n=manxpowe@43.sub-75-248-111.myvzw.com) |
| 15:59.25 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 15:59.31 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 16:00.21 | Kobaz | gaetronik: which one? |
| 16:00.27 | Kobaz | and how about my queue issue? :P |
| 16:00.45 | gaetronik | yours i've dont have any idea |
| 16:00.56 | gaetronik | i need to prevent agents to press * to hangup call |
| 16:01.12 | gaetronik | i believed that it was in features.conf but not |
| 16:01.18 | *** join/#asterisk Firass-z0r (n=asadf@juicebox.vikcomm.wwu.edu) |
| 16:01.27 | ManxPower | gaetronik: look at the Dial options |
| 16:01.37 | gaetronik | i don't have h |
| 16:02.03 | ManxPower | you're going to make me go look up the option, aren't you? |
| 16:02.18 | gaetronik | no |
| 16:02.22 | gaetronik | i swear |
| 16:02.29 | Kobaz | gaetronik: that's an option in the queues.conf |
| 16:02.35 | *** join/#asterisk dmhardison (n=derek@204-181-49-161.skybest.com) |
| 16:02.48 | Kobaz | you can enable/disable it, by default it's disabled i believe |
| 16:02.58 | ManxPower | You have neither H nor h as a dial option? |
| 16:02.59 | gaetronik | Kobaz, it seems it's enabled |
| 16:03.00 | Assid | anyone here tried grnvoip before? |
| 16:03.05 | Kobaz | so anyways |
| 16:03.09 | gaetronik | ManxPower, in the Queue no |
| 16:03.25 | ManxPower | gaetronik: and you are not using any chan_local to get to the queue? |
| 16:03.26 | Kobaz | is there a way to keep the phones ringing while playing a track to a user in a queue |
| 16:03.29 | dmhardison | my old panasonic system allowed me to press flash on my phone and then make a phone call to an extension to tell them they had a call on line x, can i not do that easily with asterisk, i have been snooping around and it doesn't appear so? |
| 16:03.52 | gaetronik | ManxPower, i have a goto then a Queue |
| 16:03.53 | ManxPower | gaetronik: pastebin the CLI output of a call coming in from the outside and going into a queue that you can exit of by pressing * |
| 16:04.04 | gaetronik | ok |
| 16:04.35 | ManxPower | dmhardison: that is all handled by the IP phone you are using. |
| 16:04.46 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:05.23 | gaetronik | http://rafb.net/p/ud066036.html |
| 16:05.30 | gaetronik | ManxPower, |
| 16:05.34 | ManxPower | Now asterisk doesn't really support the idea of "lines" that is a Key System feature., but you can put calls on hold and call other people all day. |
| 16:06.03 | russellb | unless you use the SLA applications ... |
| 16:06.10 | *** join/#asterisk xuser (i=jaood@unaffiliated/xuser) |
| 16:06.25 | Assid | okay i pulled out 4 strands of hair.. and its all your fault |
| 16:06.34 | nikko | Hey Manx, your in B'Ham? |
| 16:06.40 | nikko | you're |
| 16:06.45 | *** join/#asterisk bmg505 (n=leon@196-209-8-66-ndn-esr-2.dynamic.isadsl.co.za) |
| 16:06.58 | ManxPower | gaetronik: it must be a Queue issue then. I don't see chan_local in there |
| 16:07.14 | ManxPower | russellb: thank you for volunteering to help dmhardison |
| 16:07.33 | ManxPower | dmhardison: look into "parking" |
| 16:07.41 | Assid | okay so i found another company called grnvoip.. however, minimum purchase there is $100 |
| 16:07.45 | Kobaz | ManxPower, dmhardison: you can simulate "lines" by putting each caller into a meetme and yell out, pick up on meetme #1! |
| 16:08.08 | sp00k3y | anyone here from AZ? |
| 16:08.10 | russellb | ManxPower: you're welcome. |
| 16:08.24 | ManxPower | nikko: I am 60 miles from Birmingham |
| 16:08.43 | nikko | Cool. Mobile reprasentin' |
| 16:09.00 | ManxPower | nikko: I have no idea what you just said. |
| 16:09.05 | nikko | LA as it's referred to |
| 16:09.22 | nikko | You travel for consulting? |
| 16:09.27 | Assid | russellb: you free to help me choose an itsp? |
| 16:09.45 | russellb | nope |
| 16:09.54 | ManxPower | nikko: I am willing to travel. I'm not willing to move. |
| 16:09.56 | Assid | :( |
| 16:10.08 | Kobaz | Assid: there's like, 3843928478932749234 of them |
| 16:10.13 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 16:10.21 | Assid | Kobaz: yeah thatrs why i want a suggested one :P |
| 16:10.23 | Kobaz | Assid: personally i use voicepulse |
| 16:10.27 | ManxPower | ~itsp |
| 16:10.28 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:10.30 | Assid | and something within a certain price range |
| 16:10.39 | gaetronik | Wtf with the queue |
| 16:10.45 | nikko | ManxPower: I'm in Mobile, that was some street lingo that went over like a lead balloon in IRC |
| 16:10.51 | gaetronik | i'm the only one qho wants to disable the * hangup |
| 16:11.11 | ManxPower | We don't get many gangstas here. |
| 16:11.23 | nikko | Lucky you |
| 16:11.26 | ManxPower | gaetronik: I think you are the only one using Queues in that way. |
| 16:12.03 | nikko | that ~itsp list looks a little rusty :) |
| 16:12.29 | gaetronik | ManxPower, what's wrong with my way |
| 16:13.05 | nikko | Assid, "suggest an ITSP" is like asking "How long is a string?" |
| 16:13.07 | ManxPower | gaetronik: I have no idea. Queues are far too complicated for the limited use we do, so I emulated the features we wanted in the dialplan. Never looked back |
| 16:13.25 | gaetronik | to use exten => s,1,Queue(qUrgencia||||180) |
| 16:13.45 | gaetronik | it did not seem strange |
| 16:13.48 | Assid | nikko: well.. something used personally whihc you think is good.. and must be around 1.2-1.5c/min |
| 16:13.49 | nikko | You might see if you can find a VOIP Master agent tat will do your work for you |
| 16:14.12 | ManxPower | Assid: if you do the isplist for US or CA you might get some suggestions. |
| 16:14.15 | Assid | like if you know about any.. just mention it |
| 16:14.27 | ManxPower | ~itsp-us |
| 16:14.46 | mikealeonetti | if I use a SIP provider, is there a specific configuration I need for the SIP phones in the network to communicate with the outside world? I am able to dial out, but I can't hear anything on either phone, but I can hear hold music if I put myself on hold. |
| 16:14.54 | Assid | ManxPower: i did check a few there.. either they are 2c/min or have certain quirks |
| 16:15.04 | ManxPower | Assid: what part of " Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs" did you not understand. |
| 16:15.07 | Assid | like setup charges and such |
| 16:15.17 | ManxPower | 2 cents/min is not expensive. |
| 16:15.25 | mocker | mikealeonetti: Sounds like a NAT problem |
| 16:15.28 | mocker | ~nat |
| 16:15.29 | jbot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:15.37 | ManxPower | ~sipnat |
| 16:15.37 | jbot | [~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 |
| 16:15.43 | mocker | ManxPower: Better. :) |
| 16:16.10 | ManxPower | Last time I saw the Wiki NAT page it was pretty useless. |
| 16:16.15 | cirosou | i'm back |
| 16:16.19 | mikealeonetti | mocker: lemme modify my config accordingly then |
| 16:16.28 | mikealeonetti | the phones aren't behind a nat though |
| 16:16.32 | cirosou | results of cat /proc/zaptel/1 |
| 16:16.33 | cirosou | Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) RED |
| 16:16.34 | cirosou | <PROTECTED> |
| 16:16.46 | mocker | mikealeonetti: And no firewall? |
| 16:16.50 | ManxPower | cirosou: On that card RED means "line not connected" |
| 16:17.14 | cirosou | yes.. there is no line connected. |
| 16:17.27 | cirosou | the card keeps taking the line off hook |
| 16:17.33 | ManxPower | cirosou: then you are seeing what is expected. |
| 16:17.39 | cirosou | results of cat /etc/asterisk/zapata.conf |
| 16:17.40 | mikealeonetti | mocker: they are on the same internal network as the asterisk server, and no firewalls are implemented currently |
| 16:17.40 | cirosou | [general] |
| 16:17.40 | cirosou | signalling=fxs_ks |
| 16:17.40 | cirosou | callerid=asreceived |
| 16:17.41 | cirosou | group=0 |
| 16:17.41 | cirosou | context=from-pstn |
| 16:17.42 | cirosou | channel => 1 |
| 16:17.44 | cirosou | context=default |
| 16:17.53 | bpgoldsb | Has anyone tried using Asterisk in some kind of High-Availability setup? I'd like to setup a cluster of 2 machines running asterisk, with a bunch of sip clients connecting and a T1 for outgoing. |
| 16:17.57 | ManxPower | cirosou: either put multi line pastes on someplace like pastebin or get out. |
| 16:18.02 | ManxPower | ~pb |
| 16:18.03 | jbot | [~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 |
| 16:18.13 | mocker | mikealeonetti: Is your Asterisk server connecting to an external ITSP though? |
| 16:18.16 | cirosou | what is paste bin? |
| 16:18.18 | cirosou | sorry |
| 16:18.25 | ManxPower | cirosou: set the options THEN have the channel => line. |
| 16:18.30 | mikealeonetti | mocker: another interface directly connected to the internet |
| 16:18.30 | bpgoldsb | Ideally. if machine A or even Asterisk on Machine A dies, all the calls would seemlessly migrate to Machine B without dropping |
| 16:18.31 | mocker | Or are you just trying to call from one phone in your house to another. |
| 16:18.32 | sp00k3y | http://pastebin.com/ |
| 16:18.40 | sp00k3y | oops |
| 16:19.02 | mikealeonetti | mocker: if I call the other phones in the network they can talk just fine |
| 16:19.08 | Kobaz | so is there no way to keep the phones ringing while playing a track to a caller in a queue? |
| 16:19.23 | ManxPower | mikealeonetti: less talk, more reading |
| 16:19.27 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:19.34 | mocker | lol |
| 16:19.39 | ManxPower | Kobaz: that is the default |
| 16:19.47 | Kobaz | bpgoldsb: that's the gold question of voip, let me know when you set up asterisk to do that |
| 16:19.50 | Kobaz | ManxPower: oh, hmm |
| 16:20.00 | cirosou | ok, i will try to explain the problem from the beginning |
| 16:20.11 | mikealeonetti | ManxPower: what am I missing that I'm not reading? |
| 16:20.12 | ManxPower | Kobaz: of course you did not say which phone you want ringing or what options you are using to keep it from ringing. |
| 16:20.29 | Kobaz | ManxPower: the current phone that's ringing, stops ringing when playing a track to a user |
| 16:20.29 | ManxPower | mikealeonetti: Your Asterisk box is behind NAT, right? |
| 16:20.39 | Kobaz | ManxPower: and then after playing the track, the queue goes back to ringing that phone |
| 16:20.57 | cirosou | ManxPower: asterisk, zaptel, libpri and asteriskgui have been downloaded from svn /branches/1.4 |
| 16:20.57 | ManxPower | Kobaz: then report it as a bug. The desk phone should not stop ringing when queue times are announced. |
| 16:21.03 | mikealeonetti | ManxPower: no, it has its own dedicated static ip |
| 16:21.16 | cirosou | compiled like this, libpri, zaptel, asterisk, asteriskgui |
| 16:21.22 | ManxPower | mikealeonetti: ah, then the only thing would be firewall issues. |
| 16:21.34 | mikealeonetti | ManxPower: firewall is not enabled |
| 16:21.52 | Kobaz | ManxPower: lets say my timeout is 30 seconds, an announce of 15, and i have a 3 second track, the phone will ring for 15 seconds, play a 3 second track to the caller, and then ring for 15 seconds |
| 16:21.57 | Kobaz | ManxPower: hmm, okay |
| 16:22.09 | cirosou | everything works perfectly all my sip phones can log in and place calls to another extens or through sip trunks, but when it comes to POTS asterisk gives me this message |
| 16:22.11 | ajohnson | So, I'm missing something. I upgraded from 1.4.21.2 to 1.4.22 and installed DAHDI, but I'm getting an error message: |
| 16:22.21 | ajohnson | [Oct 15 09:19:48] WARNING[11973]: app_meetme.c:2475 find_conf: No DAHDI channel available for conference, conference recording disabled |
| 16:22.24 | ManxPower | mikealeonetti: canreinvite=no in Asterisk |
| 16:22.37 | ajohnson | dahdi show status shows: DAHDI_DUMMY/1 (source: RTC) 1 UNCONFIGUR 0 0 0 |
| 16:23.04 | mikealeonetti | ManxPower: in the global config? |
| 16:23.16 | ManxPower | mikealeonetti: actually canreinvite=no in all the sip.conf entries for phones that do not have a public IP |
| 16:23.32 | mikealeonetti | ok |
| 16:25.03 | cirosou | Unable to create channel of type 'ZAP' |
| 16:25.42 | mikealeonetti | ManxPower: that fixed it... thanks! |
| 16:26.36 | mikealeonetti | ManxPower: not exactly sure why that caused a problem. |
| 16:26.57 | cirosou | i've tried no narrow down the problem and i do not understand why asterisk keeps the line off hokk all the time |
| 16:27.34 | ManxPower | cirosou: exactly how do you know it is off hook |
| 16:29.28 | mikealeonetti | I guess this applies "This is necessary if the client and the Asterisk server is on opposite sides of a NAT gateway or firewall." |
| 16:30.25 | cirosou | because the card has one pass thru plug, when i put a phone on the pass thru and say, call my own mobile even if i put the phone on hook again the call persists... |
| 16:33.18 | ManxPower | mikealeonetti: the phones by default will try to bypass asterisk and send the audio directly to the far end -- doesn't work if there's NAT involved |
| 16:33.33 | ManxPower | cirosou: what country are you in? |
| 16:33.40 | cirosou | brazil... |
| 16:34.42 | ManxPower | cirosou: You know that the X100P cards were only designed for the telephone system in USA/Canada, right? |
| 16:35.10 | ManxPower | cirosou: you should contact the vendor you purchased your card from. Maybe they know of some option you can try. |
| 16:36.06 | dmhardison | Well, is there a way I can place a call on hold, and say have asterisk not block other users from dialing the extension that is on hold to pick it up? |
| 16:36.53 | ManxPower | dmhardison: I already told you. It is called "parking" |
| 16:37.04 | ManxPower | go read up on it in The Book. |
| 16:37.06 | ManxPower | ~book |
| 16:37.07 | jbot | book is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 16:37.07 | tzafrir_laptop | cirosou, what is the output of: asterisk -rx 'zap show channels' |
| 16:37.30 | ManxPower | they will actually dial the parking lot number, rather then the extension that parked the call. |
| 16:37.51 | dmhardison | Ah, but thats not quite the same, user has to know the park extension, I want to use the same extension that is assigned to the phone on hold. |
| 16:38.11 | ManxPower | dmhardison: then you are on your own. |
| 16:38.18 | purple_v45 | I've got a iax setup that works for most things, but I can't call a particular NPANXX. The call terminates and there's an "s" in the destination field of the CDR. Wondered if anyone knew what that meant so I can try to find a solution. |
| 16:38.20 | dmhardison | That is a very nice feature of the panasonic pbx phones. |
| 16:38.36 | ManxPower | dmhardison: not really. It's just a feature you already know. |
| 16:38.42 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:39.04 | *** part/#asterisk bpgoldsb (n=bpgoldsb@gleim-gw.atlantic.net) |
| 16:39.11 | ManxPower | For any but the smallest systems the Key System way (that the Panasonic uses) does not scale. |
| 16:39.24 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.29) |
| 16:39.30 | cirosou | ManxPower: i know that, but i kave seen dozens of them working here with no problem, |
| 16:40.24 | cirosou | tzafir: http://pastebin.com/d2290205b |
| 16:42.07 | ManxPower | dmhardison: Asterisk is not well positioned for the very small sites (less then 8 or 10 lines). Of course Key systems are not well positioned for more than 8 or 10 lines. Do you really want everyone to have a phone with 20 lines on it? |
| 16:44.07 | *** join/#asterisk xJoeMx (n=JoeM@p4FD81C5E.dip0.t-ipconnect.de) |
| 16:44.16 | tzafrir_laptop | ajohnson, chan_dahdi is not needed for app_meetme (dahdi is, but not chan_dahdi) |
| 16:44.30 | *** part/#asterisk xJoeMx (n=JoeM@p4FD81C5E.dip0.t-ipconnect.de) |
| 16:44.36 | ajohnson | right, just found out. |
| 16:46.41 | dmhardison | Well, I am actually using this with atas, and my previous pansonic system I used with cordless phone which I am not terribly familiar with the panasonic setup, but I dialed everyting on the cordless phones, flash for hold, dial the extension to transfer to, tell them whats going on, and then I hit # and it connected the two, then I hung up. |
| 16:47.30 | dmhardison | Or if I wanted to, I could place the call on hold, and then one another phone dial the extension (say line 1 would be 8801) and answer it that way. |
| 16:49.23 | [TK]D-Fender | dmhardison: With other parking apps its possible to specify a lot #. then again, what happens when you get ANOTHER call and what to park it? |
| 16:51.21 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 16:51.45 | ManxPower | dmhardison: why not just do a supervised transfer? |
| 16:52.36 | [TK]D-Fender | ManxPower: Absolutely no reason he can't just park calls like the rest of us. |
| 16:52.37 | Alan_Hicks | Howdy. Can anyone point me to some documentation on "ringback"? basicaly, I've been asked if it's possible to setup a system so that if an employee is on the phone, asterisk will wait for that employee to become available, then ring her phone, as well as the phone of the person trying to contact her. |
| 16:52.46 | [TK]D-Fender | ManxPower: Its just a quwstion of adapting |
| 16:53.07 | pif | what's the diff betw callerid and ANI ? |
| 16:53.39 | Alan_Hicks | Example: user at exten 101 wants to call user at exten 100. If 100 is currently on the phone, 101 can dial a certain extension and hang up. As soon as the person at 100 hangs up, both 100 and 101 ring. |
| 16:53.46 | *** part/#asterisk marlow (n=marlow@loke.tuxbox.ie) |
| 16:53.47 | [TK]D-Fender | Alan_Hicks: Make your exten check the status, if not available, wait a little, try again X times. |
| 16:53.54 | [TK]D-Fender | Alan_Hicks: Its all just dialplan. |
| 16:53.57 | ManxPower | [TK]D-Fender: Supervised transfer, call parking, and SLA are all options for this guy. |
| 16:54.10 | Alan_Hicks | [TK]D-Fender: I know I can do that, but that's not exactly what the user is wanting. |
| 16:54.16 | ManxPower | Alan_Hicks: you mean like app_retrydial ? |
| 16:54.18 | [TK]D-Fender | ManxPower: I have yet to see any time where I would want to ever touch *'s SLA |
| 16:54.36 | ManxPower | [TK]D-Fender: me neither. Only girly men use those sorts of things. |
| 16:54.36 | Kobaz | why would calling MusicOnHold work for iax, and not for sip |
| 16:55.04 | ManxPower | Alan_Hicks: try doing a "core show applications" next time. |
| 16:55.24 | [TK]D-Fender | Kobaz: load chan_psychic.so |
| 16:55.26 | Alan_Hicks | retrydial isn't exactly what I want. |
| 16:55.33 | Kobaz | it's this one box, it's so wierd.... MusicOnHold() will execute but not play music for sip... with asterisk 1.4.21.2... but with asterisk 1.4.1... it works fine |
| 16:55.36 | Kobaz | oh |
| 16:55.39 | Kobaz | uhh |
| 16:55.40 | [TK]D-Fender | Alan_Hicks: and I just described what you would do. |
| 16:55.55 | Alan_Hicks | The caller wants to be able to hang up her phone and go about her business, then have asterisk call her back once the callee is available. |
| 16:56.09 | ManxPower | Alan_Hicks: then you could use .call files. |
| 16:56.11 | Alan_Hicks | From what I understand, retrydial can't do this. |
| 16:56.22 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 16:56.24 | Alan_Hicks | .call files? Hmm... first I've heard of these. |
| 16:56.26 | ManxPower | any time you think "automated call" also think ".call files" |
| 16:56.28 | c4t3l | hello world |
| 16:56.29 | Alan_Hicks | goes off to read documentation. |
| 16:56.51 | c4t3l | .call file are pretty sweet! |
| 16:57.10 | Nugget | it's tricky to rock a rhyme - to rock a rhyme that's right on time - it's tricky |
| 16:58.14 | [TK]D-Fender | ManxPower: Yup... thats what I'd do for it. |
| 16:59.30 | *** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com) |
| 17:00.07 | Alan_Hicks | So bear with me. I just want to make sure I'm understanding this correctly first. |
| 17:00.22 | Alan_Hicks | To set this up I'd have to do the following. |
| 17:00.45 | Alan_Hicks | 1- Create a new extension that asks the caller to enter the extension of the person they want to talk to. |
| 17:01.03 | Alan_Hicks | 2- retrydial() that extension n times or until the person picks up. |
| 17:01.16 | viraptor | do you know any phones that send ;expires=xxx in the contact header by default? (and not the 'Expires: XXX header') |
| 17:01.31 | Alan_Hicks | 3- Generate a .call file that gets run when that person picks up the phone and dials back the original caller. |
| 17:01.34 | Alan_Hicks | Correct? |
| 17:02.01 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 17:03.34 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 17:05.35 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:06.56 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 17:07.31 | shriven | hello, I have a sort of general question.... Is there a "good" way to incorporate named extensions into the dialplan? |
| 17:08.00 | shriven | what is the general opinion on doing this? Bad idea? more work than it's worth? |
| 17:08.13 | [TK]D-Fender | shriven: What do you mean by that? |
| 17:08.25 | shriven | ok |
| 17:08.45 | shriven | so I want to be able to call the person named john at extension 7115 as well as by dialing "john" |
| 17:09.02 | shriven | which obviously means using an extension named john somewhere in teh dialplan |
| 17:09.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.47 | [TK]D-Fender | shriven: There you have it. |
| 17:09.49 | shriven | but there doesn't seem to be an "easy" way to correlate them... So far I've used goto to send ext #s to their named extension |
| 17:09.57 | [TK]D-Fender | shriven: there is nothing else. |
| 17:10.01 | shriven | well |
| 17:10.20 | shriven | right but I don't want to write out every extension for every user... especially for both their ext# and name |
| 17:10.39 | cirosou | there is another message saying something about busy line |
| 17:10.41 | cirosou | http://pastebin.com/d5c1ceb4c |
| 17:10.44 | [TK]D-Fender | shriven: Since you can't do it on 1 line,t he best you can do is 1 extra each |
| 17:10.58 | cirosou | this is part of the output of asterisk -vvvvr |
| 17:11.09 | shriven | Fender: ok, thanks. I was just wondering if I was missing an easy way to do this. |
| 17:11.32 | [TK]D-Fender | shriven: Since you can't specify 2 patterns on 1 line... no, there isn't anything more to say on it |
| 17:11.35 | *** part/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 17:11.38 | shriven | I'm thinking I'll use some variables and route # extensions to their named extension |
| 17:12.14 | *** join/#asterisk hi365_m (n=hi365@bzq-79-176-238-74.red.bezeqint.net) |
| 17:13.41 | shriven | I was also just hoping someone would volunteer some best practice methods for me. |
| 17:13.57 | shriven | I'm new at writing the dialplan, so I just want to do it the easiest/most flexible way |
| 17:14.06 | [TK]D-Fender | shriven: Nothing is "variable" there, and "routing" is a BS term. |
| 17:14.21 | shriven | ummm yes I was using the terms loosely |
| 17:14.24 | [TK]D-Fender | shriven: exten => fred,1,Goto(1000,1) |
| 17:14.25 | ManxPower | shriven: on MY systems sip accounts are based on the MAC of the device with -a, -b, -c appended for the individual line appearances. |
| 17:14.37 | [TK]D-Fender | ManxPower: We aren't talkiong peer neames here..\ |
| 17:14.39 | ManxPower | then each extension has an entry in sip.conf |
| 17:14.53 | ManxPower | ,,er,, |
| 17:15.04 | ManxPower | then each extension has an entry in extensions.conf |
| 17:15.19 | ManxPower | trying to wildcard routing to phones won't scale |
| 17:15.28 | shriven | Fender: that is what I am going to do, but the other way around. |
| 17:15.34 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 17:16.47 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 17:16.53 | [TK]D-Fender | shriven: I would make the # primary.. PSTN users can't type out the alphabet |
| 17:17.08 | ManxPower | shriven: this is how I do it. http://www.fnords.org/~eric/macro-std-exten-v2.inc Just look at the doc info and example at the beginning of the script, don't worry about the actual code |
| 17:17.20 | shriven | that's ok, they'll dial the #extension then use goto to send them to the named extension |
| 17:18.31 | [TK]D-Fender | shriven: Its a question of which is important by comparison. You don't make 10% the rule, and 90% the exception. Thats a bad management practice. |
| 17:19.13 | shriven | Fender: I don't really understand what you're telling me is bad. Will it cause issues if a user that cannot dial a named extension ends up at one because I Goto()d him there? |
| 17:19.15 | ManxPower | and that "10% the rule, and 90% the exception" is what you'll end up with anyway. Why not just make the dialplan flexible enough to handle different user's needs? |
| 17:19.39 | ManxPower | shriven: why do you even have named extensions? |
| 17:19.45 | shriven | I want to. |
| 17:20.10 | shriven | why NOT have them if it's not terribly difficult? |
| 17:20.21 | shriven | which is part of what I asked. |
| 17:20.22 | ManxPower | shriven: and I want to kill my boss and throw him into the river. That doesn't mean it's a good idea. |
| 17:20.28 | shriven | is there some huge reason I don't see that makes this hard? |
| 17:20.53 | ManxPower | shriven: The thing is you don't have enough real world experience with Asterisk to understand why it's not a good idea. |
| 17:21.04 | WimpMan | thinks that names are THE one big pro of voip. |
| 17:21.07 | shriven | manxpower: that is why i'm here asking |
| 17:21.17 | shriven | give me a good reason, don't say, it's a bad idea and leave it there. |
| 17:21.27 | ManxPower | shriven: But I can tell you that doing it your way will make the dialplan much more complicated, less easy to maintain, and more confusing |
| 17:21.48 | shriven | hmmmm |
| 17:21.55 | boolean12 | Seriously. |
| 17:22.04 | WimpMan | Only one address to remember for both mail and phone, and that's usually the easier one. |
| 17:22.09 | jaytee | named extensions? you mean like "bob" instead of 101? |
| 17:22.15 | shriven | not instead of |
| 17:22.18 | shriven | in addition to |
| 17:22.22 | ManxPower | shriven: Back when I was a newbie I thought the same way you did. Then I started having real world experience and ended up having rewrite my dialplan because it did not scale |
| 17:22.26 | [TK]D-Fender | shriven: Only thing you can dial them on is soft-phones really and they suck. Spending effort on a solution where you should be able to click on a name rather than type it anyways makes the entire framework a silly idea. |
| 17:22.44 | *** join/#asterisk tmjb (n=tane@212.200.239.230) |
| 17:22.48 | shriven | 90% of my users will be on softphones |
| 17:23.20 | ManxPower | In six months you'll be telling noobs the same thing we are telling you now. |
| 17:23.37 | *** join/#asterisk angryuser (n=Miranda@lns-bzn-29-82-248-213-214.adsl.proxad.net) |
| 17:23.41 | WimpMan | Hardware voip phones should allow you to do that as well, but some indeed only do via their phone book. |
| 17:24.09 | boolean12 | Just, don't do one thing.. Don't use names instead of numbers for your voicemail extensions. You will hate your life if you do. |
| 17:24.20 | shriven | Ok, well this answered my questions I guess... Their is nothing innately wrong with it, it's just a pain in the ass to manage. |
| 17:24.23 | WimpMan | But hopefully some time, the voip people will leave the seventies as well... |
| 17:24.54 | shriven | Now if only we could use a regex as an extension...... |
| 17:24.59 | jaytee | I'm still something of a noob or semi-noob and I wouldn't consider using named extensions. It would severely limit the flexibility of coding advanced call handling I would suspect. |
| 17:25.04 | mort_gib | shriven: You asked this a few days ago, same answer :-) |
| 17:25.18 | shriven | yeah, still wishing I could. |
| 17:25.28 | ManxPower | shriven: did you see the regex discussion on the mailing list? |
| 17:25.39 | mort_gib | But why?? Number so so nice, almost magical ;-) |
| 17:25.40 | shriven | hmmm nope. I'll search my recent archives |
| 17:25.49 | shriven | well |
| 17:25.54 | mort_gib | And damn easy to key in on normal phones |
| 17:25.54 | shriven | because people do not think with numbers |
| 17:25.58 | shriven | that's why we have dns |
| 17:26.04 | shriven | if I want to call john, I think |
| 17:26.15 | shriven | I want to call john..... oh what was his number? |
| 17:26.19 | mort_gib | That, I'm afraid is THEIR problem! |
| 17:26.19 | *** join/#asterisk feeds (n=feeds@85-135-228-100.adsl.slovanet.sk) |
| 17:26.20 | shriven | so nice if I could just dial john |
| 17:26.22 | ManxPower | shriven: People have been remembering phone numbers since the late 1960s |
| 17:26.42 | shriven | that is NOT a good reason to leave things as they are. |
| 17:26.42 | ManxPower | Even my users, who have the technical abilities of a turnip can remember extensions |
| 17:26.44 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:26.44 | cirosou | i really cannot understand what is going on with this x100p. |
| 17:26.45 | WimpMan | I really don't understand that attitude here. There is nothing wrong with having extra features and I don't see how they should interfere withe anything else or make anything harder or more complex. |
| 17:26.53 | jaytee | what about the 10% of users that don't have softphones? are they going to press 2 twice to get the b in bob jones and then all the other key combos? Cripes, I'd give up and email the guy |
| 17:26.59 | shriven | just cause that's the way it IS or has been doesn't 'make it a good reason to not try and do it a better way |
| 17:27.02 | WimpMan | It surely didn't for me, but maybe I'm doing something wring. |
| 17:27.09 | angryuser | ManxPower : it's not a reall reason to keep it |
| 17:27.15 | mort_gib | shriven: You need to look at Cisco and M$, where you ask for John, based on an AD lookup |
| 17:27.19 | shriven | jaytee: I will use extension #s as well |
| 17:27.30 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 17:27.32 | shriven | mort_gib or ldap ;) |
| 17:27.39 | ManxPower | jaytee: no, he wants to double his dialplan size by aliasing all the numeric extensions with goto the named extension. |
| 17:27.42 | mort_gib | Or, the AGI script that someone mentioned that did the very same in * |
| 17:27.45 | mort_gib | Quite |
| 17:27.47 | _ShrikE | shriven: why not just go to the directory on your phone and select john? |
| 17:27.55 | shriven | Softphones. |
| 17:27.59 | shriven | they don't do that. |
| 17:28.01 | shriven | bleh |
| 17:28.06 | mort_gib | shriven: They do! |
| 17:28.08 | ManxPower | shriven: maybe YOUR softphones don't do that, but some do. |
| 17:28.11 | shriven | I haven't found a good one yet. |
| 17:28.20 | mort_gib | But I admit that provisioning is harder! |
| 17:28.35 | [TK]D-Fender | shriven: they should be able to jsut click a name on their PC to dial them and forego any kind of entry period. |
| 17:28.37 | ManxPower | I think people that use softphones are crazy -- there's really no point in me helping them. |
| 17:28.51 | shriven | manxpower: heh. |
| 17:29.00 | ManxPower | so, best of luck shriven. |
| 17:29.01 | shriven | fender: true. |
| 17:29.05 | mort_gib | Ah, so while you haven't found a decent Softphone, but intend on using them :-) Curious |
| 17:29.15 | jaytee | shriven, maybe you ought to look at Trixbox with HUD instead of Asterisk? |
| 17:29.16 | shriven | mort_gib: that part isn't really my decision |
| 17:29.27 | shriven | jaytee: why? I have no issue with asterisk. |
| 17:29.44 | tmjb | hello i trying to connect openvox b400 using misdn to my netmod http://img71.imageshack.us/img71/5659/netmode3bz.jpg s0 interface i got my card working but it blinks red even when i connect to the netmod s0. I also configure the card to NT mode. tnx |
| 17:29.49 | mort_gib | shriven: I use a nice integration with MS Outlook, that does exactly what fender suggested |
| 17:30.03 | shriven | we don't use MS |
| 17:30.05 | jaytee | mort_gib, what is that? |
| 17:30.05 | mort_gib | I do that as part of a "standard install" |
| 17:30.12 | mort_gib | Outlook?? |
| 17:30.20 | mort_gib | or Standard install?? |
| 17:30.22 | *** part/#asterisk feeds (n=feeds@85-135-228-100.adsl.slovanet.sk) |
| 17:30.22 | jaytee | mort_gib, I know what Outlook is |
| 17:30.27 | mort_gib | :-) |
| 17:30.29 | *** part/#asterisk cirosou (n=ciro@201.20.206.172.corp.ajato.com.br) |
| 17:30.30 | jaytee | what are you using with it for Asterisk |
| 17:30.36 | jaytee | siptapi? |
| 17:30.48 | shriven | All my users will be on OS X. |
| 17:31.05 | mort_gib | I use a small plugin that allows my users to dial from their Outlook contacts |
| 17:31.14 | mort_gib | Click on a user and click dial |
| 17:31.20 | jaytee | mort_gib, that's what I'm using siptapi for |
| 17:31.25 | mort_gib | Magic, like numbers :-) |
| 17:31.27 | ManxPower | Why not just run Asterisk on OSX? |
| 17:31.30 | shriven | That is nice, so far all the softphones that integrate with the address book in os x seem to suck. |
| 17:31.41 | shriven | manxpower: because debian is better |
| 17:31.50 | mort_gib | siptapi?? Sounds complicated! |
| 17:31.54 | jaytee | is it the softphone or the Mac OSX address book that sucks |
| 17:32.09 | mort_gib | Mac OSX of course! |
| 17:32.11 | jaytee | mort_gib, nope. piece of cake |
| 17:32.18 | ManxPower | All softphones suck |
| 17:32.36 | shriven | afaik os x address book is fine. I think there just aren't a lot of people writing softphones for os x |
| 17:32.37 | mort_gib | well, actually the little plugin works good too and is free and all |
| 17:32.53 | jaytee | ManxPower, Eyebeam is the shiznit!!! Everybody knows that! :-) |
| 17:32.55 | Blackvel | I like phonesuite.de very much |
| 17:32.59 | mort_gib | jaytee: do you have a link to a howto?? |
| 17:33.12 | jaytee | mort_gib, uno momento por favor senor |
| 17:33.18 | *** join/#asterisk Valmon (n=m_dorset@viliar.dialup.corbina.ru) |
| 17:33.20 | mort_gib | Claro |
| 17:33.35 | Valmon | Hello All! |
| 17:33.45 | jaytee | mort_gib, http://www.enum.at/SIP-TAPI.479.0.html |
| 17:33.49 | shriven | I can't seem to find eyebeam |
| 17:33.50 | Blackvel | do you use Record (e.g on snom 370) to record asterisk ivr prompts? or do you use some other professional solution directly on PC? |
| 17:33.58 | shriven | I find the counterpath site, but no eyebeam |
| 17:34.01 | jaytee | shriven, Eyebeam is X-Lite Pro |
| 17:35.00 | shriven | ah hmmm |
| 17:35.00 | shriven | ty |
| 17:35.00 | Blackvel | this snom 370 mic sucks soo much...it has electronic noise on it (brummen) |
| 17:35.00 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 17:35.23 | *** join/#asterisk stooj_ (n=stooj@stooj.plus.com) |
| 17:35.30 | mort_gib | jaytee: Have a look here http://www.voip.com.sg/asterisk/voip_asterisk_outlook_dialer.html |
| 17:35.42 | jaytee | shriven, http://www.counterpath.net/eyebeam.html |
| 17:35.52 | shriven | ah ty |
| 17:37.30 | Blackvel | mort_gib: but it doesn't display incoming number? |
| 17:37.38 | Valmon | I have one q about asterisk feauters. Is it possible in dial plan play some message not to sip caller (yeah it's easy and well documented) but to whom, who recieve call? |
| 17:37.41 | mort_gib | Blackvel: True |
| 17:37.45 | jaytee | mort_gib, looks interesting but the reason they give for using theirs doesn't apply. SIP-TAPI doesn't use the Asterisk AMI. |
| 17:38.02 | Valmon | before connects both sides |
| 17:38.17 | mort_gib | Users got REALLY pissed of with Avaya popups |
| 17:38.28 | WimpMan | Valmon: Look at the Options to Dial() |
| 17:38.41 | jaytee | mort_gib, it does a refer so if you click on a name to dial it dials your phone and when you pick up it dials the number you wanted. |
| 17:38.54 | mort_gib | JayTee: I use it because it works really well, it's really easy to install and manage... |
| 17:39.01 | shriven | Does anyone use openfire/spark with asterisk? |
| 17:39.07 | jaytee | so is SIP-TAPI pretty much. |
| 17:39.25 | *** join/#asterisk tmjb (n=tane@212.200.239.230) |
| 17:39.39 | Valmon | WimpMan: thx, I'll check it |
| 17:39.39 | jaytee | mort_gib, we're using Exchange 2007 Unified Messaging with Asterisk instead of Comedian Mail. It totally rocks. |
| 17:40.08 | tmjb | can some help with netmod and misnd maybe i am using wrong cable or something ? |
| 17:40.19 | mort_gib | I will have a closer look at the siptapi stuff, that look interesting too :-) |
| 17:40.28 | Blackvel | have a nice evening..cu |
| 17:41.05 | mort_gib | I have loads of users that uses Blackberrys, so they have ALL contacts in (countrycode) number format |
| 17:41.38 | mort_gib | Yeah?? |
| 17:41.48 | mort_gib | I would like to hear more about that! |
| 17:42.24 | WimpMan | tmjb: Maybe you should tell us, what it is? Looks like a TA to me. Are you trying to use it as such or what? |
| 17:43.01 | mort_gib | So you have to use Communications server too huh?? |
| 17:44.28 | jaytee | mort_gib, nope, just Exchange 2007 with UM enabled and I'm running * 1.4 so I have to use sipX as a proxy as Exchange only speaks SIP/TCP and 1.4 only does UDP. I'm running sipX in a VM. |
| 17:44.35 | jaytee | here's a link: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html |
| 17:45.22 | mort_gib | which means that you are not using much of *'s features?? |
| 17:45.23 | tmjb | WimpMan, ok the netmod is ta i have to analog lines on it and two s0 ports as i read the manual i should setup my card it NT mode tell im i am wrong |
| 17:45.46 | tmjb | WimpMan, and netmod TA is BRI ISDN |
| 17:46.16 | jaytee | mort, I use MeetMe and Page and store CDR in mysql but all the voicemail stuff is done on Exchange. |
| 17:46.21 | WimpMan | tmjb: So you want to use it to connect analogue phones to you * via a BRI card? |
| 17:46.36 | tmjb | WimpMan, no no |
| 17:47.09 | WimpMan | Sounds interesting, then. |
| 17:47.25 | tmjb | WimpMan, it is like this ptt company --> netmod isdn ---- openvox b400 card --- asterisk --- iphones |
| 17:47.56 | tmjb | WimpMan, i got this modems from my ptt company |
| 17:48.21 | synchris | aah |
| 17:48.43 | WimpMan | Does not sound like 1. you want NT mode and 2. is that netmod thing any use for you? |
| 17:49.08 | *** join/#asterisk jplank (n=gbove@reports.nyigc.net) |
| 17:49.11 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 17:49.32 | *** join/#asterisk _supermario (n=mario@S0106000f66827ad1.gv.shawcable.net) |
| 17:49.33 | tmjb | WimpMan, netmod support one isdn phone or 2 analog phones to the s0 port you can connect isdn phone |
| 17:49.41 | jplank | where can I start troubleshooting recording quality issues? |
| 17:49.44 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 17:50.38 | WimpMan | tmjb: Do you have a seperate NT or does that netmod have one integrated and thus a U(something) Port we don't see on that picture? |
| 17:51.29 | WimpMan | Any way: If you want to connect your BRI card to a line, it has to be in TE mode. |
| 17:51.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:51.59 | tmjb | WimpMan, te mode |
| 17:52.00 | WimpMan | ... and connected using a normal(straight) cable. |
| 17:52.18 | tmjb | WimpMan, that could be the problem |
| 17:53.22 | WimpMan | So unless you net that netmod thing for analogue phones, leave it in the box and connect the card to the NT. |
| 17:53.22 | tmjb | WimpMan, one more thing i manual they say S0 bus is for ISDN terminal devices |
| 17:53.28 | *** join/#asterisk C4colo (n=DJpyro@66.185.107.193) |
| 17:53.49 | C4colo | what is the google-searchable name for pressing * to exit voicemail? |
| 17:54.09 | tmjb | WimpMan, so probaly i should setup it for TE |
| 17:54.31 | Katty | core show application voicemail? |
| 17:54.43 | WimpMan | TE = Terminal Equipment |
| 17:54.48 | [TK]D-Fender | C4colo: Go read your list of Asterisk Standard Extensions again |
| 17:55.08 | *** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com) |
| 17:55.22 | Katty | [TK]D-Fender: i haz a pumpkin pie blizzard |
| 17:55.53 | aliver | Google is not being helpful or maybe I'm just being a bonehead, but I can't seem to find anywhere that gives me a list of the fields and their order for the Master.csv data. Anyone got a link to that? |
| 17:56.36 | tmjb | WimpMan, could be i got confused with this one http://www.asteriskguru.com/tutorials/bri.html 3.2 NT "This is the interface between an ISDN user and the ISDN provider. It is a small hardware box to which the user has to connect his ISDN devices via a so called S0 interface. In most European countries, the ISDN provider supplies the NT' |
| 17:56.46 | [TK]D-Fender | Katty: Yumz |
| 17:57.02 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 17:57.13 | [TK]D-Fender | aliver: Go look in your source tarball doc folder |
| 17:57.34 | Katty | [TK]D-Fender: quite. |
| 17:57.34 | aliver | k |
| 17:57.43 | WimpMan | tmjb: Well, that tell you, the other end is an NT. |
| 17:59.40 | Katty | [TK]D-Fender: so good, that my blood sugar levels are making me ill. |
| 17:59.51 | tmjb | WimpMan, so then i configured correct in NT if use little box for telco |
| 17:59.58 | WimpMan | tmjb: Not a good tutorial, it seems after looking into it briefly. |
| 18:00.36 | WimpMan | tmjb: No. You always connect NT <--> TE. |
| 18:00.40 | [TK]D-Fender | Katty: I'm currently crashing from the caloric withdrawl from this past weekend.... |
| 18:01.00 | Katty | [TK]D-Fender: that's gonna take you a good week or two to get over. |
| 18:01.10 | Katty | [TK]D-Fender: and the skin to firm up again >.< |
| 18:01.41 | tmjb | WimpMan, ok i will try to change jumpers on the card to TE mod then will see back in a few minutes. Thankyou very much very much :))) |
| 18:04.31 | jplank | muffled audio, doesn't *seem* like a QOS issue - Only happens on audio routed through the asterisk - audio that gets reinvited away from the asterisk sounds perfect. Ideas? |
| 18:04.53 | Katty | chocolate fixes everything. |
| 18:05.15 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 18:06.10 | jplank | i tried that, but I couldn't fit the candy bar into the PCI slot or the CDROM |
| 18:06.28 | Katty | foiled again :< |
| 18:06.34 | jplank | I also tried sticking it into the end users ears, but that just made the audio sound worse |
| 18:06.41 | Katty | hehe |
| 18:06.58 | jplank | (but delicious) |
| 18:07.44 | CGMChris | When I dial to an outside IVR from my VoIP system, DTMF is not recognized. I have dtmfmode=rfc2833 in sip.conf. Thoughts? |
| 18:07.47 | jplank | Not a CPU problem, never broke past 0.02% cpu usage, and doesn't seem like a RAM problem |
| 18:08.05 | jplank | CGMChris: what version of asterisk? |
| 18:08.09 | Katty | what sort of connection does it have? |
| 18:08.17 | *** join/#asterisk Shido6 (n=shido6@209.114.208.111) |
| 18:08.43 | CGMChris | jplank: asterisknow running 1.4.18.1 |
| 18:09.13 | jplank | there goes my idea - answer katty's question |
| 18:09.25 | CGMChris | Katty: SIP/net2phone |
| 18:09.26 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:09.33 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:09.33 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:09.48 | lmadsen | anyone seen this error before? WARNING[12169]: app_voicemail.c:2273 inboxcount: Failed to obtain database object for 'Asterisk'! |
| 18:09.55 | jplank | does net2phone support rfc2833? |
| 18:10.30 | CGMChris | jplank: their configuration parameters docs say to set dtmfmode=rfc2833... but then again, they also say to enable g729 and g723, which dont work. |
| 18:10.30 | Katty | CGMChris: sorry, that question was meant for jplank |
| 18:10.35 | DarKnesS_WolF | lmadsen: mmmm normal voicemail setup ? no IMAP or database stuff ? |
| 18:10.46 | lmadsen | DarKnesS_WolF: yes database stuff |
| 18:10.56 | lmadsen | 'failed to obtain database object' :) |
| 18:11.08 | lmadsen | I've never seen that warning before |
| 18:11.11 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 18:11.23 | jplank | google seemed to have seen it before: http://www.google.com/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-GB%3Aofficial&hs=y0&q=app_voicemail.c%3A+Failed+to+obtain+database+object+for+%27Asterisk%27&btnG=Search |
| 18:11.30 | DarKnesS_WolF | lmadsen: ah sorry i just came back from work 12 hours and so tired :-) didn't noticed , i never used database with voicemail :-s |
| 18:11.38 | lmadsen | DarKnesS_WolF: heh |
| 18:12.30 | DarKnesS_WolF | lmadsen: try to do the quary manual ? |
| 18:13.16 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 18:14.08 | CGMChris | Katty and jplan: interestingly enough, DTMF *works* when I dial through Gizmo5. Maybe I will ring net2phone and see if they know how to fix it. |
| 18:15.18 | Katty | oh ah |
| 18:16.07 | *** join/#asterisk tuxfoo21 (n=tmmarini@64.127.17.94) |
| 18:16.23 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 18:17.13 | tmjb | WimpMan, thank you Port 4 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0 Debug:0 |
| 18:17.16 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 18:18.42 | *** join/#asterisk hi365_m (n=hi365@bzq-79-176-238-74.red.bezeqint.net) |
| 18:29.43 | *** join/#asterisk xloafx (n=Sean@rrcs-72-45-234-5.nys.biz.rr.com) |
| 18:33.19 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 18:36.15 | CGMChris | K, Net2phone has requested a SIP Trace from wireshark or ethereal |
| 18:36.19 | CGMChris | terrific ! |
| 18:37.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:37.14 | *** join/#asterisk cirosou (n=chatzill@201.20.206.172.corp.ajato.com.br) |
| 18:38.37 | *** join/#asterisk zippytech (n=ron@75.149.24.162) |
| 18:39.08 | zippytech | what port needs opened for an outside firewall sip user |
| 18:39.35 | sp00k3y | 5060, 10000-20000 |
| 18:39.50 | sp00k3y | u aslo need to configure sip_nat.conf or somehting i think |
| 18:40.00 | zippytech | all udp |
| 18:40.03 | sp00k3y | yeah |
| 18:41.02 | sp00k3y | are u using straight asterisk? |
| 18:41.18 | Qwell | as opposed to...what? |
| 18:41.55 | cirosou | to distros like trixbox, meucci(digivoice) or other suff like this |
| 18:42.12 | sp00k3y | yes |
| 18:42.44 | sp00k3y | cuz im pretty sure that sip_nat.conf comes by default is most distros like freepbx but in straight asterisk u have to make the file urself and configure it |
| 18:43.14 | sp00k3y | someone tell me if im wrong or nto lol |
| 18:43.17 | [TK]D-Fender | sp00k3y: No, that file is created exclusively by FreePBX and does not have anything to do with * in and of itself |
| 18:43.27 | sp00k3y | ah ok |
| 18:43.50 | cirosou | i've tried some of this distros... they're pretty limited... no c compiler, no kernel source..., not speaking about trixbox... never used |
| 18:44.09 | [TK]D-Fender | zippytech: go read : |
| 18:44.11 | [TK]D-Fender | ~sipnat |
| 18:44.11 | jbot | [~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 |
| 18:44.17 | sp00k3y | so for straight asterisk you only need to forward the ports on the firewall? |
| 18:45.15 | [TK]D-Fender | sp00k3y: Next tidbit for you : this is not a matter of "straight asterisk" or not. |
| 18:45.29 | sp00k3y | IM LEARNING! |
| 18:45.38 | [TK]D-Fender | sp00k3y: Running a GUI to config the rest doesn't change what needs to be done, onlyt hat some of it is generated for you in an easier way |
| 18:45.43 | cirosou | sp00ky: i'm no asterisk guru, but afaik no difference between a local or remote sip user if the port as correctly forwarded |
| 18:46.19 | zippytech | thanks again |
| 18:46.22 | sp00k3y | ok well then im not sure what needs ot be done then, gonna go read that link :P |
| 18:46.40 | [TK]D-Fender | and I saw "forwarding" and firewall" thrown around without a proper description of which sides were firewalled, and how they were routed. |
| 18:46.57 | [TK]D-Fender | Half of a half-baked description. |
| 18:47.27 | cirosou | fender is right, but i'm assuming the only logical description that is a local asterisk server, a local firewall and a remote sip extension |
| 18:47.55 | Kobaz | yawns |
| 18:48.37 | Kate|afk | Need sleep? |
| 18:49.38 | Kobaz | i need an office hammock |
| 18:49.43 | sp00k3y | i need to go read my book again lol |
| 18:50.02 | AndyML | Kobaz: where do you work? |
| 18:50.03 | jplank | whats the going rate for a sip trunk? |
| 18:50.07 | Kate-o | I've found sleeping on or under my desk works nicely |
| 18:50.12 | Kobaz | AndyML: on the RPI campus |
| 18:50.34 | AndyML | if you were at digium i'd be able to point you towards one... sorry - I don't know of any at RPI :) |
| 18:50.44 | Kobaz | heh |
| 18:51.03 | [TK]D-Fender | ~siptrunk |
| 18:51.04 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 18:51.07 | [TK]D-Fender | ^^^ |
| 18:51.23 | CGMChris | I have another question about DTMF. Asterisk (over net2phone SIP) doesnt properly Rx or Tx DTMF. It works over Gizmo5, but not net2phone. Thoughts on where the problem might be? |
| 18:51.31 | Kobaz | i think i've reached my limit of salty plum soda |
| 18:51.33 | Qwell | CGMChris: net2phone |
| 18:51.50 | [TK]D-Fender | CGMChris: your configs |
| 18:51.50 | Kate-o | What in the world is salty plum soda? |
| 18:52.06 | [TK]D-Fender | Kate-o: Something that doesn't exist |
| 18:52.07 | CGMChris | D-Fender: dtmfmode=rfc2833 in sip.conf [global] |
| 18:52.16 | Kate-o | lol alrighty |
| 18:52.34 | [TK]D-Fender | CGMChris: I'm not going to comment on single little lines and no debug. |
| 18:52.45 | Kobaz | [TK]D-Fender: got that right |
| 18:52.53 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:53.14 | CGMChris | How do I get a debug log of DTMF not working? I have core verbosity at 10 and nothing outputs. |
| 18:53.30 | [TK]D-Fender | CGMChris: SIP debug of the complete call & configs |
| 18:54.04 | CGMChris | so, what do I do? Type sip debug at CLI ? |
| 18:54.27 | cj | okay... how much does a PRI line run? the distance is negligible and can be covered easily with only copper (cat5). |
| 18:54.34 | tzafrir_laptop | CGMChris, you also need to log debug? |
| 18:54.39 | Kobaz | cj: ask your local telco |
| 18:54.39 | cj | I want to start with 2 channels and eventually move up to 24 |
| 18:54.51 | cj | Kobaz: is it that variable? |
| 18:55.00 | Kobaz | cj: you won't get a 2 channel pri from your telco |
| 18:55.20 | CGMChris | in logger.conf I already have debug => debug |
| 18:55.23 | cj | Kobaz: I've been quoted $75 for a BRI... is that not the same as a 2-channel PRI? |
| 18:55.30 | [TK]D-Fender | CGMChris: CLI not logs |
| 18:55.34 | tzafrir_laptop | Kobaz, 24? why not 30? |
| 18:55.43 | Kobaz | cj: i get a pri for 200 a month, but i have servers sitting in the next office over from One Communications |
| 18:55.48 | cj | tzafrir_laptop: T1 only does 24, no? |
| 18:55.57 | [TK]D-Fender | cj: I've never heard of a telco that would bother offering PRI for less than 4 channels |
| 18:55.58 | Kobaz | cj: BRI != PRI |
| 18:56.10 | CGMChris | I am completely confused on what to do. |
| 18:56.14 | cj | Kobaz: ah, okay. thanks :) |
| 18:56.33 | CGMChris | D-Fender: what do I do at the CLI ? |
| 18:56.41 | [TK]D-Fender | CGMChris: Go to * CLI, max out verbose & sip debug, pastebin a COMPLETE call. |
| 18:56.42 | cj | Kobaz: same here. I'm one floor away from at least one telco. |
| 18:56.55 | cj | Kobaz: is that $200 all 24 channels? |
| 18:57.06 | Kobaz | cj: 23B and one D |
| 18:57.28 | cj | okay. thank you! It's probably not a lot less for fewer channels, if they even offer it, eh? |
| 18:57.44 | Kobaz | the wireing cost is the same, the hardware cost is the same |
| 18:57.50 | [TK]D-Fender | cj: Depends on the telco |
| 18:57.52 | Kobaz | you still need a t1 dsu on both ends |
| 18:58.04 | Kobaz | but yeah, it depends on the telco |
| 18:58.23 | cj | okay. Thanks again for the info. I'm still new to this and trying to formulate a business plan :) |
| 18:59.46 | Kobaz | er |
| 18:59.54 | Kobaz | 1.4.22 isn't building |
| 19:00.03 | Kobaz | maybe i'll disable dahdi |
| 19:00.08 | Kobaz | chan_dahdi.c: In function 'get_alarms': |
| 19:00.08 | Kobaz | chan_dahdi.c:3693: error: 'struct zt_params' has no member named 'chan_alarms' |
| 19:00.38 | Kobaz | oh wait, i thought only dahdi was in 1.4 |
| 19:00.41 | Kobaz | 1.6 rather |
| 19:01.08 | *** join/#asterisk riccyb (n=rnbrady@193.82.139.119) |
| 19:01.16 | riccyb | hi folks |
| 19:01.36 | riccyb | anyone around who would be familiar with asterisk/sip/music on hold behaviour |
| 19:01.37 | riccyb | ? |
| 19:01.55 | Kate-o | no but I'd like to be |
| 19:02.11 | tzafrir_laptop | Kobaz, you get this error in 1.6?? |
| 19:02.18 | riccyb | lol |
| 19:02.19 | Kobaz | yeah |
| 19:02.20 | Kobaz | er |
| 19:02.20 | Kobaz | no |
| 19:02.22 | Kobaz | 1.4.22 |
| 19:03.10 | tzafrir_laptop | Kobaz, looking into that |
| 19:03.11 | *** join/#asterisk stephank (n=urk@82-197-207-120.dsl.cambrium.nl) |
| 19:03.18 | tzafrir_laptop | you're building it vs. zaptel, right? |
| 19:04.33 | Kobaz | well i've been having problems with stuff in 1.4.21.2, so i figured might as well see if i have the same results in 1.4.22 |
| 19:04.44 | Kobaz | but i need zaptel (unless sangoma supports dahdi now) |
| 19:04.56 | stephank | Hello! Our asterisk installation seems to be chewing on pound keys. We've got two SIP devices registered with it, and I can clearly see the device send the pound key event in the rtp data, but asterisk doesn't replicate it to the other device. It's usually only the first pound. res_features is unloaded. What's doing this in asterisk? |
| 19:05.01 | tzafrir_laptop | what version of zaptel do you use? |
| 19:05.09 | Kobaz | mmm |
| 19:05.18 | Kobaz | version: 1.4.5.1 |
| 19:05.30 | Qwell | upgrade.. |
| 19:05.37 | tzafrir_laptop | yeah. figures |
| 19:05.38 | Kobaz | i've had problems with later releases and sangoma drivers, so i haven't migrated yet |
| 19:06.23 | Kobaz | asterisk ./configure, should check for apropriate zaptel/dahdi versions |
| 19:06.29 | Qwell | ...it does |
| 19:06.33 | Kobaz | oh |
| 19:06.49 | CGMChris | D-Fender: http://pastebin.com/dbefd801 |
| 19:07.22 | tzafrir_laptop | hmmm... but that part is protected by #if defined(HAVE_DAHDI) || defined(HAVE_ZAPTEL_CHANALARMS) |
| 19:07.55 | Kobaz | configure doesn't check for specific versions it doesn't seem (on a quick glance) |
| 19:07.56 | Qwell | zt_params doesn't even exist in 1.4.22 |
| 19:08.01 | Kobaz | it just checks for various h files |
| 19:08.21 | Qwell | Kobaz: what, exactly, did you download? |
| 19:08.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:08.34 | Kobaz | lftpget http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.22.tar.gz |
| 19:08.59 | [TK]D-Fender | CGMChris: I can see you didn't even configure your system properly to work behind NAT : Contact: <sip:18124254246@10.0.0.8> |
| 19:09.19 | CGMChris | My firewall has pass thru |
| 19:09.25 | [TK]D-Fender | CGMChris: And apparently a call rejection : SIP/2.0 407 Unauthorized |
| 19:09.30 | Qwell | Kobaz: zt_params does not exist in that version. |
| 19:09.40 | Qwell | Kobaz: What did you change? |
| 19:09.58 | Kobaz | Qwell: heh, i didn't change anything |
| 19:10.03 | Qwell | Then you aren't using 1.4.22 |
| 19:10.05 | tzafrir_laptop | zt_params is from dahdi_compat.h, I guess |
| 19:10.55 | Kobaz | <PROTECTED> |
| 19:11.16 | [TK]D-Fender | CGMChris: And i don't see dialplan executing for the inbound call. |
| 19:11.26 | CGMChris | D-Fender: This is an outbound call |
| 19:11.35 | CGMChris | D-Fender: Outbound DTMF does not work |
| 19:11.37 | [TK]D-Fender | CGMChris: Would be nice if you proved your phone was fully functional between * as well |
| 19:11.49 | [TK]D-Fender | CGMChris: there are *2* legs to this call |
| 19:12.11 | CGMChris | D-Fender: Internally, everything works great. Inbound: DTMF works great. Outbound, remote IVR does not recognize DTMF from my system. |
| 19:12.37 | [TK]D-Fender | CGMChris: Prove it. |
| 19:12.44 | CGMChris | D-Fender: How? |
| 19:12.44 | Qwell | >.< |
| 19:12.51 | Qwell | #define HAVE_DAHDI HAVE_ZAPTEL |
| 19:12.56 | Qwell | that ain't right |
| 19:13.03 | cj | that looks wrong to me... |
| 19:13.06 | [TK]D-Fender | CGMChris: Go place another call with that phone to verify that DTMF is working fine with it. |
| 19:13.15 | cj | it's a nice idea, though... perhaps we should float it to the language designers? |
| 19:13.21 | CGMChris | [TK]D-Fender: and pastebin again? |
| 19:13.55 | [TK]D-Fender | CGMChris: Natually |
| 19:14.42 | CGMChris | [TK]D-Fender: The last pastebin was from my VoIP phone to an external number. Where do you want me to call this time? Internally? |
| 19:15.18 | [TK]D-Fender | CGMChris: Test the phone direct to * |
| 19:15.25 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:15.27 | [TK]D-Fender | CGMChris: No 3rd party |
| 19:15.44 | CGMChris | http://pastebin.com/dbefd801 <- 3rd party, please hold for internal. |
| 19:16.14 | [TK]D-Fender | CGMChris: And skip SIP debug on all of these. |
| 19:16.25 | [TK]D-Fender | CGMChris: And provide the rest of what I asked for twice since the start |
| 19:16.42 | CGMChris | D-Fender: What did you ask for twice since the start? |
| 19:18.24 | CGMChris | sip set debug <- this just turns debugging on, how do I disable it? |
| 19:18.39 | [TK]D-Fender | CGMChris: Go read, I'm not feeling inclined to spell it out again. |
| 19:19.11 | CGMChris | Read this conversation? |
| 19:20.44 | jplank | fender I missed your ~siptrunk - I don't get the meaning of it not existing? or a I missing the joke? |
| 19:21.09 | *** join/#asterisk cvnet (n=dahitler@74.210.108.245) |
| 19:21.15 | CGMChris | Can anyone tell me how to turn off "sip set debug" ? |
| 19:21.23 | cvnet | hi all |
| 19:22.20 | cvnet | waht is the difference between asterisk 1.4 and 1.6 ? |
| 19:22.39 | cj | 0.2 |
| 19:22.42 | cvnet | CGMChris its sip set debug disable |
| 19:22.43 | cj | ducks |
| 19:23.26 | Kobaz | [TK]D-Fender: okay, so i have some more info on my MusicOnHold() problem |
| 19:23.37 | cj | cvnet: svn diff http://svn.digium.com/svn/asterisk/branches/1.4 http://svn.digium.com/svn/asterisk/branches/1.6.0 | less |
| 19:23.43 | Kobaz | [TK]D-Fender: it affects both sip and iax.... |
| 19:23.49 | Kobaz | [TK]D-Fender: pasting... :) |
| 19:24.16 | CGMChris | [TK]D-Fender: http://pastebin.com/dd59b0f6 <- pastebin of internal call to voicemail, with debugging off |
| 19:24.41 | cirosou | cj: would be nice to see someone trying this one |
| 19:24.56 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gleim-gw.atlantic.net) |
| 19:25.27 | cj | cirosou: what, the svn diff? |
| 19:25.34 | cirosou | yep... |
| 19:25.37 | cirosou | funny |
| 19:25.39 | cj | yeah... :) |
| 19:25.49 | cj | and reading it all :) |
| 19:26.05 | cirosou | hehehehe.. |
| 19:26.12 | Qwell | Kobaz: can you show me the HAVE_ZAPTEL_CHANALARMS line from include/asterisk/autoconfig.h ? |
| 19:26.12 | bpgoldsb | I'm using AEL and Asterisk-1.6. Just to confirm, anytime I want to go run a sub-set of dialplan code, I should be using a macro, correct? Even if there is no arguement to the macro. i.e. '&foo()' |
| 19:26.33 | CGMChris | QWell: Any ideas what to try next to fix my DTMF problem? |
| 19:26.39 | cj | looks like there's generated code in the repository. wah. |
| 19:27.40 | [TK]D-Fender | CGMChris: And now the configs I asked for at the start... |
| 19:27.50 | CGMChris | k, just sip.conf or extensions.conf also? |
| 19:27.59 | [TK]D-Fender | CGMChris: just sip |
| 19:28.42 | Kate-o | Anyone know what could've happened for me to get a congestion error? It only happens when people try to call my phone, but I can call anyone else's |
| 19:29.48 | CGMChris | [TK]D-Fender: http://pastebin.com/d2292ab37 |
| 19:31.01 | [TK]D-Fender | CGMChris: You should set the actual mode in your peer you know... |
| 19:31.13 | [TK]D-Fender | CGMChris: And I take it thats users.conf... |
| 19:31.28 | CGMChris | [TK]D-Fender: I had that originally in the peer section. Users.conf is empty. echo "" > users.conf |
| 19:31.53 | CGMChris | DTMF works 100% with Gizmo5... just not net2phone. |
| 19:33.23 | cirosou | i have a question... have anyone here worked with an avaya phone, they have a nice feature that allows the user to "log in" the phone.. something like this, the phone is configured as extension 6000 and if i dial 6000 it rings, when a user sits to use the fone it ypes for example 78877(userid) and then this phone (besides keep ringing when someone calls 6000) starts to ring when someone... |
| 19:33.25 | cirosou | ...calls the user id, or the user id is associated with a calling group for example. this allows the user to be anywhere within the company building and keep his extension with him. I kown it is possible with VOip phones, but it it extremely troublesome to enter the phone menu just to change the sip user name, is ther no other way to make it as simple as in avaya? |
| 19:33.46 | jaytee | FollowMe? |
| 19:34.27 | rwaite | ven conmigo! |
| 19:34.45 | cirosou | vem comigo! |
| 19:35.32 | cvnet | waht is the difference between asterisk 1.4 and 1.6 ? |
| 19:36.41 | jaytee | cvnet, the version number and lots of changes. |
| 19:37.41 | jaytee | for instance, zaptel is gone in 1.6 replaced by DAHDI |
| 19:37.53 | jaytee | and SIP TCP and TLS is now supported |
| 19:38.43 | cvnet | if you are a newbie which one would you install? someone was suggesting to install vanilla asterisk, but I can't find it there (maybe this sound stupid sorry) |
| 19:38.43 | jaytee | and reading the UPGRADE.txt file in the 1.6 tarball would tell you alot |
| 19:39.04 | codefreeze-lap | bpgoldsb: anyone answer you? If not, the answer from me is "yes". |
| 19:39.15 | jaytee | 1.4 or 1.6 are both "vanilla". It's just slang for plain asterisk without a GUI |
| 19:39.40 | bpgoldsb | codefreeze-lap: No, but thanks to your previous guidance I was pretty sure I was right. |
| 19:39.43 | bpgoldsb | Thanks for confirming :) |
| 19:40.09 | cvnet | a ha, thanks |
| 19:40.15 | cirosou | cvnet: you can have an idea reading thish ttp://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.1 |
| 19:40.24 | cvnet | let me do some reading |
| 19:40.37 | cvnet | cirosou thanks, lets start with that |
| 19:40.47 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 19:42.22 | cirosou | it's not a easy readingm though |
| 19:42.24 | [TK]D-Fender | CGMChris: It is aparent that you just tried dropping users.conf parameters right over. |
| 19:42.44 | CGMChris | [TK]D-Fender: Some were copied, but I later found out they do nothing. |
| 19:46.49 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 19:47.00 | devilsoulblack | hi |
| 19:48.07 | devilsoulblack | i have DISCONNECT from isdn pri but the telco tell me the asteisk pbx send the DISCONNECT i have read the pri debug can any one read too and tell me if asterisk send the DISCONNECT |
| 19:48.13 | devilsoulblack | http://pastebin.ca/1228130 |
| 19:49.14 | CGMChris | [TK]D-Fender: I just read the SIP trace... from what I can see, I need to just use ulaw with net2phone. Is that correct? |
| 19:51.45 | *** join/#asterisk swampwork (n=rew@64.238.252.218) |
| 19:52.05 | devilsoulblack | any one ? |
| 19:53.25 | nikko | Anyone use a supermicro dual chassis for an asterisk implementation? These are 2 dual processor servers in a single 1U chassis, sharing a PSU |
| 19:54.09 | nikko | Have these specd for DNS and chache boxes in a colo I'm building, but am thining they might make good pbxs as well, unless theer are any issues with them |
| 19:54.19 | nikko | chache=cache |
| 19:54.33 | nikko | i tipe guud |
| 19:54.47 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-127-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:56.43 | bpgoldsb | What replaces ValetParking in Asterisk 1.6? |
| 19:59.55 | *** join/#asterisk virtexPro (n=virtex5@41.224.178.72) |
| 20:03.29 | jameswf | app_parkyourowncar |
| 20:03.42 | jameswf | the economy is down |
| 20:09.03 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:09.15 | mog | heh |
| 20:10.55 | *** part/#asterisk Valmon (n=m_dorset@viliar.dialup.corbina.ru) |
| 20:12.11 | bpgoldsb | I'm calling Record(foo/1.gsm); in my dialplan. When I get to this step, it immediately completes and moves on to the next priority. Any idea why it doesn't record? |
| 20:13.04 | Kobaz | stupid scammers |
| 20:13.13 | Kobaz | i keep getting calls to lower your interest rates |
| 20:13.17 | Kobaz | they use asterisk too |
| 20:13.28 | Kobaz | they haven't even changed the default music on hold |
| 20:14.24 | *** join/#asterisk blepsoaf (n=pbaker@nnat-gw.adeptra.com) |
| 20:14.43 | blepsoaf | is asterisk mostly written in objective-c? |
| 20:14.53 | [TK]D-Fender | I know *I'm* becoming less interested by the second... |
| 20:14.58 | Qwell | blepsoaf: just c |
| 20:15.15 | blepsoaf | Qwell: thanks much |
| 20:15.37 | Kobaz | [TK]D-Fender: me? |
| 20:15.53 | [TK]D-Fender | Kobaz: Sarcasm |
| 20:16.11 | Kobaz | yeah i know |
| 20:17.02 | Kate-o | haha I just changed our msuic to 311 |
| 20:18.06 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:18.23 | Kate-o | s%/music/music/ |
| 20:18.50 | Kate-o | ha |
| 20:20.09 | *** join/#asterisk vonkleist (n=vonkleis@201.155.129.19) |
| 20:20.51 | rwaite | hmm. if i call my own zap interface thru my sip provider it sounds like crap. but if i call it from any other phone, cell, my home phone, it sounds great |
| 20:21.10 | Katty | Wocka. |
| 20:21.14 | *** join/#asterisk CrazyTux (n=brandon@rrcs-67-52-124-226.west.biz.rr.com) |
| 20:24.48 | *** join/#asterisk Shotygun (n=thorn@82.166.246.204) |
| 20:26.53 | *** part/#asterisk rabbit7 (i=rabbit7@stat.siff.org) |
| 20:27.44 | [TK]D-Fender | heading home. Later all |
| 20:29.11 | *** part/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 20:31.03 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:10) |
| 20:45.28 | mikealeonetti | my ITSP says they get 404 errors when trying to dialin, what should I be looking for to debug? |
| 20:45.33 | mikealeonetti | Asterisk gives me no output |
| 20:45.40 | mikealeonetti | there is no firewallon |
| 20:47.47 | *** join/#asterisk hi365_m (n=hi365@bzq-79-176-238-74.red.bezeqint.net) |
| 20:47.52 | lmadsen | mikealeonetti: it'll give you lots of output if you turn sip debugging on at the CLI |
| 20:48.21 | lmadsen | mikealeonetti: 404 Not Found typically means the other end is request an extension that it can't find in the context it is setup for |
| 20:48.39 | *** join/#asterisk proute (n=none@ARouen-153-1-42-5.w90-17.abo.wanadoo.fr) |
| 20:48.46 | proute | hello all |
| 20:49.58 | proute | I use asterisk 1.4.21.2. On my log I have this message :sched.c: Request to schedule in the past?!?! . I use ntpd services, my system is not loaded, I have only one call when this message appeared. And when i have this message, I lost my call. Any idea? Thanks |
| 20:50.10 | mikealeonetti | it's definitely giving me a lot more information now |
| 20:51.59 | devilsoulblack | i have DISCONNECT from isdn pri but the telco tell me the asteisk pbx send the DISCONNECT i have read the pri debug can any one read too and tell me if asterisk send the DISCONNECT |
| 20:52.04 | devilsoulblack | http://pastebin.ca/1228130 |
| 20:52.26 | *** join/#asterisk eliel (n=eliel@200.61.172.61) |
| 20:52.43 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:53.14 | *** join/#asterisk deeperror (n=deeperro@76.226.185.124) |
| 20:54.03 | deeperror | Any suggestions on SIP termination with decent UK rates? |
| 20:54.21 | mikealeonetti | I see |
| 20:56.57 | WimpMan | devilsoulblack: I'd need an intense debug to be sure of the meaning. |
| 20:57.52 | lesouvage | I'm using automon with feature code # tos top and start recording during phonecall. The current name pattern of the recordings is auto-1224099722-4755606698-s-out/in.raw and the current directory /var/spool/asterisk/monitor. Is there a way to set the pattern and the location on per call basis? |
| 20:58.47 | deeperror | Need to make a lot of UK calls! Anyone know of a good provider with decent rates? |
| 20:59.39 | proute | deeperror, I know good provider, but in france |
| 21:01.27 | edwin_quijada | how can I access the asterisk database? |
| 21:01.29 | deeperror | in detroit here would probably have quality issues |
| 21:01.46 | lesouvage | <PROTECTED> |
| 21:02.25 | *** join/#asterisk fberretta_ (n=fberrett@190.190.141.136) |
| 21:02.35 | mikealeonetti | When it says "Looking for +1....in default", where is it looking for it? in sip.conf ? |
| 21:02.42 | jplank | errr I can't figure out this audio problem, its driving me nuts |
| 21:03.19 | jplank | if I reinvite the * out of the media path, audio sounds perfect |
| 21:03.29 | jplank | once I put * back in, it sounds muffled |
| 21:03.37 | lesouvage | edwin_quijada: install phpmyadmin . It is not that hard, I managed ;-) |
| 21:04.26 | jplank | very little CPU usage, and about 90% of the memory is cached |
| 21:04.35 | edwin_quijada | lesouvage: If I remember this if for mysql |
| 21:04.37 | jplank | SIP to SIP |
| 21:04.53 | edwin_quijada | I think it is asteriskdb , BerkeleyDB |
| 21:08.39 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:11.00 | deeperror | jplank, could it be network issues |
| 21:16.57 | hardwire | anbody ever used a T1 "splitter" against multiple tdm cards? |
| 21:17.05 | hardwire | heh. |
| 21:17.09 | hardwire | I know that sounds silly |
| 21:17.27 | hardwire | but if they are slave timed, and one is programmed not to respond to anything, then it's all good right? |
| 21:17.41 | hardwire | it could join in if the other TDM device failed |
| 21:20.27 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 21:20.35 | fberretta_ | Hi, When an Asterisk IP PBX is thisconnected from Internet, and if it has an active registration with an external voip service provider through a fqdn name ej: register => asterisk:password@iptel.org/jan, after a couple of seconds the IP PBX starts to work very sloooowly and makes it unusable with a very poor performance between internal calls. This problem remains until Internet connection is back or register line is removed then... all starts |
| 21:20.35 | fberretta_ | to work perfectly. Is this a known problem ? is this a bug related with register command and DNS resolution or only related with linux DNS resolution ? |
| 21:23.36 | *** join/#asterisk edageneR (n=d@65.126.237.4) |
| 21:23.40 | edageneR | Anyone around? |
| 21:25.31 | thedonvaughn | hrm, will the dms100 switchtype in asterisk work with a dms250 switch? |
| 21:25.44 | hardwire | is the dms250 dms100 compatible? |
| 21:25.58 | hardwire | also.. cool.. what are you working on? |
| 21:26.43 | thedonvaughn | a ds3 turn up :) |
| 21:26.56 | thedonvaughn | just found out qwest wants us on their dms250 switch |
| 21:27.06 | thedonvaughn | trying to see if it'll work with asterisk |
| 21:27.08 | _ShrikE | Ha.. our professional liability insurance provider actually asks if we have any "microsoft based" internet facing servers as a determination of exposure. |
| 21:27.31 | mikealeonetti | It keeps saying "Looking for +1.... in default (domain [ip address])" and then "SIP/2.0 404 Not Found" |
| 21:27.46 | mikealeonetti | where is it looking for it? the context should handle it correctly... |
| 21:28.27 | CGMChris | bkrukse: still alive? |
| 21:28.29 | hardwire | cute |
| 21:28.31 | CGMChris | bkruse: test. |
| 21:28.34 | edageneR | Anyone know a good addon or program to monitor queues on our Asterisk setup? |
| 21:28.40 | hardwire | the last company I was with went all Microsoft for everything |
| 21:28.42 | hardwire | they go down often. |
| 21:29.19 | bkruse | CGMChris: yes, answered response in #asterisknow |
| 21:31.15 | lesouvage | edwin_quijada: you are right, it is for mysql and not for asteriskdb. My fault. |
| 21:40.15 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 21:41.14 | maxxim | i'v registered to a sip server using a username & password. How can i make a call via this sip connection using Dial(SIP/808101@XXXXX via extensions? |
| 21:41.24 | maxxim | how should be the correct Dial command for it? |
| 21:42.15 | C4colo | anyone have an example of *67 or CID blocking? |
| 21:43.42 | ManxPower | Dial(SIP/808101@thesipconfsectionforthisaccount) |
| 21:45.07 | maxxim | ManxPower> i've registered to remote sip server using "register =>" command globaly, so there is no section :( |
| 21:45.28 | maxxim | or i should add this "register=> xx" command into one created section for a specific user? |
| 21:46.16 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 21:47.17 | *** join/#asterisk devhen_ (n=devhen@216.194.118.110) |
| 21:47.58 | *** join/#asterisk outtolunc (n=me@c-24-130-75-122.hsd1.ca.comcast.net) |
| 21:48.55 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:50.05 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:50.49 | fberretta_ | maxxim have you tried using proxy ipaddress or fqdn in Dial cmd ? |
| 21:51.18 | mikealeonetti | this SIP 2.0 / 404 error is driving me crazy |
| 21:51.45 | Lorax | anyone have a speach-to-text thing going on voicemail? |
| 21:51.55 | maxxim | fberretta_> can you give me an example, please? |
| 21:52.01 | Qwell | Lorax: not for less than about $500k |
| 21:52.46 | maxxim | fberretta_> when i start asterisk, it shows me that it was able to register to the remote sip server (sip show registery). But i don't know how to make a call using Dial command via this sip peer.. thanks |
| 21:53.36 | fberretta_ | Dial(SIP/ipaddress/number2dial) |
| 21:53.38 | Lorax | Qwell: that would seem to be a bizzare amount of money to pay for technology commercially available for $200. |
| 21:53.55 | Qwell | Lorax: Show me anything that can be bought for $200 without training it first. |
| 21:53.59 | mikealeonetti | I get this error, http://pastebin.com/d1b4d07ea , but the context should handle it no problem... |
| 21:54.15 | Lorax | Qwell: Dragon Naturally Speaking, $199. |
| 21:54.20 | Qwell | Requires training. |
| 21:54.24 | Lorax | nope. |
| 21:54.33 | Lorax | not for english |
| 21:54.54 | Lorax | it hasn't needed training for many years |
| 21:55.35 | maxxim | fberretta_> it doesn;'t work. * tryies to register again to that remote sip server using some dummy username like 'sip' :( |
| 21:55.36 | ManxPower | mikealeonetti: You have a match for +15162084679 in your extensions.conf? |
| 21:55.44 | maxxim | fberretta_> any other method? |
| 21:56.50 | mikealeonetti | ManxPower: I have exten => +15162084679,1,Answer in the context |
| 21:57.29 | Qwell | Lorax: I can assure you that it most definitely does. You will not get anywhere near any "acceptable" level of accuracy otherwise. |
| 21:58.38 | maxxim | anybody , help me please :( |
| 21:58.49 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:59.41 | mikealeonetti | ManxPower: would that not answer the call? |
| 22:00.17 | fberretta_ | is your user registered ? |
| 22:01.50 | maxxim | fberretta_> yes, it says, state: Registered |
| 22:02.13 | maxxim | fberretta_> 1 SIP registrations. |
| 22:02.16 | mikealeonetti | changing it to exten => _+15162084679,1,Answer produces the same results |
| 22:02.34 | fberretta_ | if it doesn't works you'll need to create a peer like |
| 22:02.35 | fberretta_ | [youruser] |
| 22:02.36 | fberretta_ | type=peer |
| 22:02.36 | fberretta_ | secret= |
| 22:02.36 | fberretta_ | username= |
| 22:02.36 | fberretta_ | host= |
| 22:02.37 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:02.37 | fberretta_ | fromuser= |
| 22:02.41 | fberretta_ | fromdomain= |
| 22:02.43 | fberretta_ | nat= |
| 22:02.46 | fberretta_ | insecure=very |
| 22:02.47 | fberretta_ | canreinvite= |
| 22:02.49 | fberretta_ | qualify= |
| 22:02.51 | fberretta_ | dtmfmode=rfc2833 |
| 22:03.05 | maxxim | k, let me try |
| 22:05.00 | [TK]D-Fender | fberretta_: use a pastebin next time instead of spamming the channel |
| 22:05.02 | [TK]D-Fender | ~pb |
| 22:05.03 | jbot | [~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 |
| 22:05.04 | [TK]D-Fender | ^^^^^^^^^ |
| 22:05.16 | mikealeonetti | ~pb |
| 22:05.17 | jbot | [~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 |
| 22:05.19 | mikealeonetti | dude! |
| 22:05.31 | mikealeonetti | ~datingwebsites |
| 22:05.38 | mikealeonetti | didn't work |
| 22:06.02 | CGMChris | [TK]D-Fender: Solution to earlier issue: net2phones documentation inaccurately states to use dtmfmode=rfc2833, however they actually use inband. |
| 22:06.50 | [TK]D-Fender | CGMChris: Good quality provider... |
| 22:06.57 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:07.18 | maxxim | fberretta_> i'm getting a "Forbidden" |
| 22:07.22 | CGMChris | [TK]D-Fender: I havent actually had the SIP session drop, not once. But the docs suck. |
| 22:08.04 | maxxim | fberretta_> but registration goes fine... i think he tries to place a direct call bypassing the registration, this is why it may fail |
| 22:08.19 | *** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com) |
| 22:09.01 | *** join/#asterisk devhen_ (n=devhen@216.194.118.110) |
| 22:09.56 | fberretta_ | have you filled the parameters ? are you using ths peer in Dial cmd ? |
| 22:11.57 | maxxim | fberretta_> oh, cool it worked... i've just added use the full regsitered command like: register => 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID |
| 22:12.12 | maxxim | added the last bit "/1777MYCCID" to match the username |
| 22:12.44 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 22:13.23 | maxxim | fberretta_> thanks a lot!!! |
| 22:15.12 | fberretta_ | u're welcome |
| 22:17.38 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:24.28 | outtolunc | pulls another munched headset from my backpack.. sheesh |
| 22:25.55 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 22:33.15 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:37.19 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:37.37 | jameswf | I also use supersecret as my password... I should probably change it |
| 22:38.34 | Iamnacho | that's my password too! |
| 22:38.36 | Iamnacho | :( |
| 22:38.50 | CGMChris | Is there a reason that macros exit immediately if any key is pressed, with complete disregard to any extensions defined within said macro (particularly when macro is called from the dial application) ? |
| 22:39.10 | jaytee | my password is ooeeahahahbingbangramalamadingdong |
| 22:39.37 | outtolunc | missing a w? |
| 22:41.20 | CGMChris | outtolunc: w param for the dial command? |
| 22:44.33 | *** part/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 22:50.28 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 22:51.29 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 22:55.27 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 22:55.30 | Dovid | evening all |
| 22:55.57 | Dovid | if I have agi(my_agi.agi,123456) what is the variable name for 123456 in the AGI ? |
| 22:56.13 | [TK]D-Fender | CGMChris: Because you should never ever try to make an IVR out of a macro. |
| 22:57.07 | [TK]D-Fender | Dovid: There is non, its a param like passed to any other program |
| 22:57.35 | Dovid | TK: how do i get that param ? |
| 22:58.01 | Dovid | i did it a while back with php agi it was somevariable[0] for the first somevariable[1] for the second etc. |
| 23:00.50 | *** join/#asterisk voxter (n=voxter@mail.metrobridge.com) |
| 23:04.25 | *** join/#asterisk gr00t (n=the_html@dyn-62-56-98-143.dslaccess.co.uk) |
| 23:05.01 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 23:06.20 | gr00t | guys - just building up a asterisk box without any zap/wanpipe/etc hardware - is ztdummy still required with 1.6.xx for use with sip/iax only? |
| 23:10.00 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 23:11.24 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 23:12.49 | *** join/#asterisk StooJ (n=stooj@stooj.plus.com) |
| 23:14.36 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:17.14 | Dovid | TK: any idea ? |
| 23:18.40 | [TK]D-Fender | Dovid: Go learn the language you are using for your app. |
| 23:19.08 | [TK]D-Fender | gr00t: only if you want MeetMe, Page, or IAX2 Trunk Mode |
| 23:19.35 | Dovid | TK: So its something specifc to PHP as opposed to the AGI ? |
| 23:20.51 | *** join/#asterisk LiNeTuX_Home (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 23:22.54 | gr00t | D-Fender - appreciated. may end up using iax in trunk, so will install - get to play with the new name :p |
| 23:23.13 | jaytee | hahahhaa, http://www.collegehumor.com/video:1884025/ |
| 23:27.10 | *** join/#asterisk purple_v45 (n=rmarc@71-91-227-115.static.stls.mo.charter.com) |
| 23:27.15 | *** join/#asterisk kiteOregon (n=kiteoreg@70.89.181.157) |
| 23:27.49 | kiteOregon | I need to implement Follow Me functionality on a Digium device running asterisk 2.0.3 business |
| 23:28.09 | purple_v45 | I'm having a problem connecting to a particular NPANXX and was trying to figure out a why to determine the actual cause |
| 23:28.17 | kiteOregon | i believe i need to add some script to the extensions.conf, but that is about it |
| 23:28.39 | purple_v45 | I've got an iax connection, in the detail record the destination shows up as "s" |
| 23:29.33 | kiteOregon | just want to forward cell from extension to cell number and back to VM if no answer |
| 23:32.26 | kiteOregon | no help out there with in regards to the digium appliance? |
| 23:39.39 | jaytee | kiteOregon, if your Asterisk system is an appliance and you purchased it from Digium you can call them for support. |
| 23:40.30 | kiteOregon | jaytee, yes i have support, however since their GUI does not support follow me they won't support other then say it's possible and i need to edit the extensions.conf file |
| 23:41.34 | jaytee | kiteOregon, I haven't implemented FollowMe and wouldn't know how with the GUI version. You might try this as a reference material though. |
| 23:41.36 | jaytee | ~book |
| 23:41.37 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:42.26 | kiteOregon | jaytee, thanks, i will check it out |
| 23:42.50 | jaytee | and read the followme.conf.sample file |
| 23:43.09 | jaytee | kiteOregon, are you in Oregon? |
| 23:43.24 | kiteOregon | jaytee, yep |
| 23:43.43 | jaytee | I used to live in Medford. I miss the state. |
| 23:44.10 | kiteOregon | medford is nice, i am in the pdx area |
| 23:44.35 | [TK]D-Fender | jaytee: So do I.... thats what you get for buying aftermarket ballistic guidance systems.... ;) |
| 23:44.37 | jaytee | spent time across the river in Vancouver, WA alot too |
| 23:45.16 | jaytee | [TK]D-Fender :-) |
| 23:45.45 | [TK]D-Fender | replaces the avionics with a Garmin GPS, paperclip, wad of chewing gum, and screws it back together with his handy-dandy Swiss Army Knife |
| 23:46.13 | [TK]D-Fender | downloads some updated mapgs |
| 23:47.29 | edageneR | Incase anyone else has any input, I am looking for a program for Asterisk that I can display call queues, specifically either ones with clients waiting or which I designate. |
| 23:47.35 | purple_v45 | anyone know what would case the dst field in the CDR to show up as "s" and what that means? |
| 23:47.38 | jaytee | Anyone going to watch the third debate? |
| 23:47.43 | edageneR | Not me |
| 23:47.45 | Dovid | TK: took some brains. i had a bat setting in php.ini |
| 23:47.51 | Dovid | who woulda thought ;) |
| 23:48.11 | jaytee | purple_v45, read about Asterisk Standard Extensions in the WIKI |
| 23:48.20 | purple_v45 | danke |
| 23:48.43 | jaytee | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 23:49.06 | [TK]D-Fender | jaytee: I'm off in 5 mins to look at a Minolta HTSI 25mm Camera. Good deal for that + 28-80mm lens, polarized filter, and a nice bag all in really good shape. Will give me a film body compatible with all of my FF lenses & flash. |
| 23:49.32 | [TK]D-Fender | purple_v45: Because thats the exten the call entered on. |
| 23:49.53 | jaytee | must be nice to be you, living in a country where the economy isn't circling in the whirlpool of water like a turd being flushed. |
| 23:50.39 | thehar | flushes |
| 23:51.05 | [TK]D-Fender | jaytee: Keep in mind I have very few expenses... this accoutns for 2 weeks of "fun" expenditures. |
| 23:51.09 | purple_v45 | Not sure why I'd get that on dialing a particular NPA-NXX |
| 23:51.42 | purple_v45 | Just get a busy for no reason I can come up with (I have the destination phone in hand). |
| 23:52.03 | purple_v45 | the only odd thing I noticed was the "s" in the destination field |
| 23:52.13 | *** join/#asterisk sjobeck (n=sjobeck@c-24-20-130-45.hsd1.or.comcast.net) |
| 23:52.13 | jaytee | [TK]D-Fender, both candidates here keep promising to save us all but I'm a skeptic. I'm a member of the Whig Party and we haven't had a President in the WH since Millard Fillmore. |
| 23:52.51 | [TK]D-Fender | RP2008!!!! |
| 23:53.27 | jaytee | [TK]D-Fender, his economic views are dead on, don't like his fuzzy wuzzy Christian value crap though |
| 23:54.11 | [TK]D-Fender | jaytee: as I've said before, his hyper-Constitutionalism would prevent him from actually attempting to bridge church & state. |
| 23:54.24 | [TK]D-Fender | jaytee: The ultimate safety check |
| 23:54.54 | [TK]D-Fender | jaytee: He can believe one thing, but what he'd enact is a known constant |
| 23:55.27 | edageneR | Must be nice to be who? |
| 23:55.46 | edageneR | Oh, the guy buying the camera |
| 23:55.58 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:56.09 | [TK]D-Fender | BBAIB |
| 23:56.15 | jaytee | [TK]D-Fender, yes, a very salient point |
| 23:56.25 | jaytee | pity the media sabotaged his campaign |
| 23:56.53 | edageneR | They're all idiots IMO |
| 23:57.13 | edageneR | I think all politicians in every branch should work for free, so we would get people who actually WANT to do the job. |
| 23:57.19 | edageneR | But that wouldnt work either. |
