| 00:00.14 | husimon | http://en.wikipedia.org/wiki/Image:Leeroy_Jenkins_Jeopardy_clue.jpg |
| 00:00.16 | husimon | lol |
| 00:00.17 | Qwell | that video was clearly faked |
| 00:00.33 | drmessano | Yes |
| 00:00.38 | husimon | Qwell, obviously but its amazing it got so much coverage |
| 00:00.45 | Qwell | husimon: it was pretty funny |
| 00:00.47 | husimon | it would be sweet to have been the person controlling the char |
| 00:00.59 | angryuser | even i hear of lerooy ;) |
| 00:01.02 | Qwell | at least I've got chicken.. |
| 00:01.05 | drmessano | lol |
| 00:01.11 | husimon | Qwell, yeah especially after having played that instance many times |
| 00:01.12 | angryuser | heard* |
| 00:01.15 | Qwell | there's a Leeroy soundboard that somebody made in flash |
| 00:01.31 | drmessano | stick to the plan |
| 00:01.34 | *** join/#asterisk warewolf (i=warewolf@warewolf.org) |
| 00:01.38 | husimon | wow |
| 00:01.43 | husimon | there huge cultural references to it |
| 00:01.46 | husimon | scrubs used it |
| 00:01.50 | husimon | a toyota commercial |
| 00:01.57 | angryuser | http://fr.youtube.com/watch?v=LkCNJRfSZBU here it is lerooooooooy |
| 00:02.00 | warewolf | anyone here ever try to use Asterisk like a voice-chat service similar to rogerwilco/ventrilo/teamspeak? |
| 00:02.20 | husimon | warewolf a meetme extension? |
| 00:02.21 | husimon | hehe |
| 00:02.30 | Qwell | warewolf: no, but I thought it would be a cool idea to have a vent channel driver for asterisk |
| 00:02.38 | warewolf | husimon: I dunno what meetme is |
| 00:02.41 | Qwell | vent is *really* popular ... |
| 00:02.45 | husimon | warewolf, conference call |
| 00:03.31 | Qwell | husimon: see msg :p |
| 00:03.31 | husimon | Qwell, so you could call a number and be added to a vent channel? |
| 00:03.31 | warewolf | Qwell: vent is popular, but I like teamspeak better. Vent is BW heavy, and TS does a lot of AGC stuff that vent doesn't. |
| 00:03.31 | Qwell | husimon: yeah, something like that |
| 00:03.31 | husimon | Qwell, that would be pretty neat if it was a 1800 number |
| 00:03.31 | warewolf | Qwell: well I think teamspeak's protocol is fairly well documented, ventrilo I have no clue. |
| 00:03.35 | tzanger | tzafrir: around? |
| 00:03.40 | Qwell | warewolf: oh? |
| 00:03.45 | warewolf | Qwell: basically I want to stop paying for ventrilo and use something free and open source :) |
| 00:03.50 | husimon | it doesn't seem like it would be too hard to do that |
| 00:03.56 | drmessano | app_leroy <--- randomly places a call to a $29 a minute 900 number |
| 00:04.17 | husimon | you could just make a sip -> ventrillo client |
| 00:04.20 | husimon | that sat in the middle |
| 00:04.21 | warewolf | Qwell: there are win/mac/linux clients for teamspeak |
| 00:04.26 | husimon | don't even modify asterisk |
| 00:04.28 | warewolf | husimon: yeah, but then where'd you do the muxing? |
| 00:04.41 | warewolf | husimon: that'd also lose the concept of separate users |
| 00:04.42 | husimon | warewolf, it would have to be a thick client I guess |
| 00:04.46 | warewolf | husimon: nod |
| 00:04.51 | warewolf | husimon: or some kind of gateway |
| 00:04.55 | husimon | warewolf, why ? each person would be a different sip user |
| 00:05.00 | husimon | which would be a different ventrilo user |
| 00:05.01 | warewolf | husimon: that'd work |
| 00:05.43 | warewolf | I don't fully grok asterisk yet, I just know it's cool :) I don't really (personally) have a need for a PBX at home, but the geek in me really wants to play with one :) |
| 00:05.59 | plik | so play... it's awesome |
| 00:06.33 | husimon | warewolf, yeah the thing stopping me is that I don't really need much beyond my cell phone |
| 00:06.36 | warewolf | I'm taking a wild stab in the dark that for conference call bridges ..etc.. the asterisk server muxes all the audio streams together, instead of sending all the audio streams back to each client |
| 00:06.42 | warewolf | husimon: exactly my situation. |
| 00:06.44 | husimon | paying for a voip did or another physical phone lines seems silly |
| 00:07.11 | husimon | now the question is: is there anyway to get my mobile phone to act as an fx0 for asterisk |
| 00:07.13 | drmessano | Watching this video again.. I realize they could have gotten out of there alive..... |
| 00:07.20 | drmessano | ... if they cast Magic Missile |
| 00:07.33 | warewolf | drmessano: sleep! sleep! magic missile! resist! |
| 00:07.43 | angryuser | <husimon> chan_mobile |
| 00:07.50 | husimon | angryuser, does it work? |
| 00:07.50 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 00:07.58 | angryuser | <husimon> never tryed |
| 00:08.04 | Qwell | husimon: it works alright |
| 00:08.08 | drmessano | Summonger Geeks is another awesome one |
| 00:08.18 | husimon | Qwell, do only certain phones work? |
| 00:08.37 | Qwell | nah, anything that supports the handsfree profile should work. |
| 00:08.47 | husimon | hmm |
| 00:08.49 | *** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 00:08.55 | husimon | i have a blackjack |
| 00:09.03 | Qwell | newish moto? |
| 00:09.05 | Qwell | it'll work |
| 00:09.09 | husimon | it's samsung |
| 00:09.15 | Qwell | newish? |
| 00:09.19 | husimon | yeah |
| 00:09.21 | Qwell | it'll work |
| 00:09.33 | husimon | vewy interesting |
| 00:09.37 | plik | Qwell: any good docs on chan_mobile? |
| 00:09.40 | husimon | you obviously need a server with bluetooth right |
| 00:09.45 | Qwell | plik: chan_mobile.txt |
| 00:09.46 | husimon | i guess i could run it on my laptop |
| 00:09.49 | Qwell | husimon: obviously |
| 00:09.52 | Qwell | and it's Linux only |
| 00:09.57 | husimon | hmm |
| 00:09.59 | husimon | doh |
| 00:10.06 | husimon | i wonder |
| 00:10.11 | husimon | if I could run it in a linux vm |
| 00:10.16 | husimon | in os x |
| 00:10.28 | Qwell | you'd have to get osx to share it with vmware somehow...unlikely |
| 00:10.36 | Qwell | unless it acts as a usb device |
| 00:10.37 | angryuser | what about bt headsets, no quality issues? |
| 00:11.05 | eric2 | the only issues you'll experience is with the voip line itself |
| 00:11.05 | husimon | you mean bt -> sip phone -> asterisK? |
| 00:11.23 | eric2 | bluetooth is designed to carry audio traffic |
| 00:11.27 | husimon | i guess that would be bt headset -> sip softphone -> asterisk |
| 00:11.41 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:11.41 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:11.50 | angryuser | <eric2> a lot of thing designet to do something but the dont do it ;) |
| 00:11.56 | angryuser | *designed |
| 00:12.05 | eric2 | ya, you have a point |
| 00:12.21 | husimon | how do I tell what channels asterisk has been compiled with? |
| 00:12.38 | husimon | sorry I mean modules |
| 00:12.46 | angryuser | modules show |
| 00:12.49 | angryuser | in cli |
| 00:12.55 | husimon | kk yeah just saw that |
| 00:13.01 | angryuser | or core show modules |
| 00:13.07 | *** part/#asterisk warewolf (i=warewolf@warewolf.org) |
| 00:13.09 | angryuser | dont remember |
| 00:13.32 | husimon | or show modules |
| 00:15.22 | craigk | i have noticed that when a call is not answered, two CDRs are created, one with disposition NO_ANSWER and one with disposition ANSWERED ... is this normal ? |
| 00:16.29 | lmadsen | craigk: depends how your dialplan is setup. If the call is coming in from somewhere else, you might have an ANSWERED call from the first channel, and a NO_ANSWER on the 2nd channel |
| 00:16.57 | craigk | lmadsen: soory i was not cleare. I am making an outgoing call. |
| 00:17.01 | lmadsen | ITSP --channel 1--> asterisk --channel 2--> SIP phone |
| 00:17.11 | craigk | also sorry i appear unable to type correclty this monring - need mroe coffee :) |
| 00:17.17 | lmadsen | SIP phone --channel 1--> asterisk --channel 2--> ITSP |
| 00:18.47 | lmadsen | eeesh.... JACK requires a kernel rebuild... |
| 00:18.47 | craigk | ah, i see (i think), so my voip provider is actually 'answering' the call as far as asterisk is concerned |
| 00:19.00 | Mavvie | file: With regarding to bug #11917, is there anything else you want to have from me? |
| 00:20.05 | lmadsen | craigk: hard to say... the call actually goes through? |
| 00:20.50 | craigk | lmadsen: yes. so the path is SIPPhone->asterisk->VoipProvider->externalPhone. the external phone is ringing, and i ahng up the SIPPhone |
| 00:21.13 | lmadsen | still might depend on your dialplan... I haven't done much with CDRs (thank god) |
| 00:21.21 | lmadsen | I'd think you'd have 2 answered legs |
| 00:21.33 | lmadsen | unless you're not answering the other phone that is ringing |
| 00:21.41 | *** part/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 00:21.43 | lmadsen | in which case that might be the cause of the NO_ANSWER |
| 00:22.22 | lmadsen | hrmmmmmmmmmmmm........ |
| 00:22.36 | lmadsen | any idea how to build cdr_sqlite? I seem to be able to build cdr_sqlite3_custom no problem |
| 00:23.20 | craigk | lmadsen: cdr_sqlite uses sqlite 2 .... I am still using asterisk 1.4.17 and had to patch it to work with sqlite3 :/ |
| 00:23.43 | *** join/#asterisk patrickv0x (n=patrick@67.131.93.17) |
| 00:23.44 | lmadsen | hrmmm... wonder what package I need to install then on centos5 |
| 00:23.49 | craigk | lmadsen: I assume you are using 1.6 beta - but that is a guess as i have not started using it yet |
| 00:23.58 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 00:24.05 | lmadsen | craigk: using trunk actually -- about to do a 2nd round of CLI audits |
| 00:24.12 | lmadsen | so I'm just trying to build as many modules as possible |
| 00:24.19 | patrickv0x | I got a bunch of phones with extension 8XX, what kind of 'lines' i need to add in my extensions.conf to allow me to dial from 801 to 802 ? |
| 00:24.30 | lmadsen | ~book |
| 00:24.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:24.34 | lmadsen | patrickv0x: read the above |
| 00:27.52 | kyron | mvanbaak, are-you the one that recommended I buy "The Pragmatic Programmer..." ? |
| 00:28.55 | craigk | lmadsen: you were right about my 2 CDRs ... my dialplan said Answer, Wait(1), Dial(...) - i think i read somewhere to do that. I changed it just Dial(...) and now i get 2 CDRs where both say NO_ANSWER |
| 00:29.00 | lmadsen | kyron: he went to bed |
| 00:29.18 | lmadsen | kyron: I have that book though... read part of it :) |
| 00:29.20 | craigk | strangely, i i do pick up the call, i just get one CDR that says ANSWERED .. hmmm |
| 00:29.21 | drmessano | I'm going to write a book.. "Veeoheyepee for Dummies" |
| 00:30.27 | kyron | lmadsen, I wanted to thank him for recommending it ;) |
| 00:32.27 | kyron | drmessano, wtf mate? |
| 00:32.32 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) [NETSPLIT VICTIM] |
| 00:32.32 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) [NETSPLIT VICTIM] |
| 00:32.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 00:32.32 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 00:32.34 | *** join/#asterisk JonMcN (n=Jon@cpc4-sout2-0-0-cust715.sotn.cable.ntl.com) |
| 00:32.34 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) [NETSPLIT VICTIM] |
| 00:32.34 | *** join/#asterisk andrewn (n=andrew@76.191.151.229) [NETSPLIT VICTIM] |
| 00:32.34 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) [NETSPLIT VICTIM] |
| 00:32.43 | *** join/#asterisk AndyGraybeal_ (n=andy@node227.34.251.72.1dial.com) |
| 00:32.44 | drmessano | wtf? |
| 00:32.44 | kyron | lmadsen, so...part of it...got boored |
| 00:33.14 | lmadsen | kyron: I don't do a lot of coding, so I put it on the shelf |
| 00:33.21 | kyron | "Veeoheyepee for Dummies" define Veeoheyepee plz |
| 00:33.27 | lmadsen | was doing some at the time, but that came to an end, and I am much happier for it |
| 00:33.29 | drmessano | V O I P |
| 00:33.37 | drmessano | Vee Oh Eye Pee |
| 00:33.43 | kyron | lmadsen, ahhhh...ehhh...uhhm...oh...you're an integrator! |
| 00:33.50 | kyron | LOOOOOOOOOOOLLLLLLLLLLLLL |
| 00:33.54 | lmadsen | kyron: well, I do lots of dialplan work :) |
| 00:34.11 | lmadsen | I wrote an E911 portal that integrated with SOAP... wanted to kill myself |
| 00:34.14 | kyron | lmadsen, hehehe, you da king (pff...clustering) |
| 00:34.22 | kyron | lmadsen, heard you got a tatoo ;) |
| 00:34.30 | lmadsen | bloody lies! :) |
| 00:34.32 | drmessano | Esemteepee for Dummies too |
| 00:34.32 | tzanger | lmadsen: not just wash? |
| 00:34.52 | kyron | lmadsen, I need more reading to appreciate your suicidal tendencies |
| 00:36.17 | lmadsen | kyron: http://farm2.static.flickr.com/1386/1429670303_9823c7261e.jpg?v=0 |
| 00:36.33 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:36.43 | Qwell | lmadsen: geek |
| 00:36.45 | kyron | lmadsen, it's true! |
| 00:36.49 | lmadsen | Qwell: heck ya |
| 00:37.00 | Qwell | lmadsen: I met one of the authors of that book once |
| 00:37.03 | Qwell | cool guy |
| 00:37.09 | lmadsen | Qwell: I heard they were cool |
| 00:37.18 | Qwell | just Jared. The other two are nubs. |
| 00:37.26 | lmadsen | Qwell: but I heard that Leif guy is a jerk |
| 00:37.39 | putnopvut | Yeah, that Leaf guy is a total nub. |
| 00:37.47 | kyron | oh god...would love to hang around and laugh some more but have some people around... |
| 00:37.53 | husimon | someone seriously got that as a tatoo |
| 00:37.54 | husimon | lol |
| 00:37.59 | lmadsen | kyron: ya... go hang out with real people, not irc people |
| 00:38.01 | Qwell | husimon: lmadsen ... |
| 00:38.04 | lmadsen | husimon: yes, I did :) |
| 00:38.04 | kyron | husimon, the _author_ maybe... |
| 00:38.06 | kyron | pfffffff |
| 00:38.08 | kyron | hehehe |
| 00:38.19 | drmessano | I got rid of all my real friends.. they didn't have cool quit messages |
| 00:38.23 | kyron | lmadsen, yeah...you all virtual...in my head..llalalalalala |
| 00:38.25 | husimon | drmessano, AHAHAHHAHA |
| 00:38.32 | husimon | lmao |
| 00:38.40 | husimon | oh <deity> |
| 00:38.42 | kyron | drmessano, LOOL |
| 00:39.00 | husimon | i'm still laughing |
| 00:39.04 | husimon | hope no one walks past my office |
| 00:39.29 | drmessano | People would call me up "Hey man, wanna go bowling" and i'm like "What site?" |
| 00:39.32 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:39.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:40.05 | husimon | drmessano, or your friend sitting there telling you a story and you exlaim "PIX OR IT DIDN't HAPPEN"! |
| 00:40.12 | drmessano | HAH! |
| 00:40.43 | drmessano | Breaking up with a girl sucks if you can't tell her "G T F O" |
| 00:40.50 | husimon | woke up, fell out of bed, dragged a comb across my head... |
| 00:40.58 | drmessano | irc > real life |
| 00:41.24 | lmadsen | drmessano: rule number 1 -- don't let the girl move in, and especially don't move in with her |
| 00:41.36 | [hC] | wtf is gtfo? |
| 00:41.41 | lunaphyte | giyf |
| 00:41.42 | husimon | get the fuck out |
| 00:41.52 | drmessano | Rule #2 don't marry them |
| 00:42.02 | lmadsen | drmessano: I thought that rule was implied |
| 00:42.27 | lmadsen | never lived with a girl, and never let a girl move in with me -- I've been smart |
| 00:42.31 | putnopvut | lmadsen: no, it's totally cool to get married....as long as they don't move in and it's cool to see other girls. |
| 00:42.32 | drmessano | No, you can be stupid and skip to 2 |
| 00:42.50 | lmadsen | putnopvut: lol |
| 00:42.54 | putnopvut | ...it can happen. |
| 00:43.04 | lmadsen | ya... I'm still looking for that I guess |
| 00:43.09 | tzanger | lmadsen: way smarter than me |
| 00:43.16 | lmadsen | ideally my 'wife' would do all the work and find the girls too |
| 00:43.29 | lmadsen | tzanger: ya well, we can't all be as smart as me |
| 00:43.29 | tzanger | getting married's great... but to the right woman |
| 00:43.35 | putnopvut | So you'd just come home and see who she found for tonight's >2 way? |
| 00:43.35 | tzanger | and it's so fucking hard to do that |
| 00:43.43 | lmadsen | pfffffft... that's what they say, but I never hear anyone really *mean* it |
| 00:43.52 | lmadsen | putnopvut: exactly |
| 00:43.56 | Qwell | >2? |
| 00:44.05 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 00:44.08 | lmadsen | greater than 2 way (3-way+) |
| 00:44.12 | tzanger | 19:45 <@lmadsen> ideally my 'wife' would do all the work and find the girls too |
| 00:44.17 | tzanger | Qwell: ^^ |
| 00:44.38 | tzanger | I dunno though, Corydon's been talking about Leif again lately.../ |
| 00:44.44 | putnopvut | lol |
| 00:44.52 | lmadsen | ya... he got separated/divorced |
| 00:45.00 | putnopvut | Oh, really? :( |
| 00:45.28 | drmessano | Been there, done that.. bought the T-shirt, she took it with her :( |
| 00:45.35 | tzanger | drmessano: amen |
| 00:45.41 | drmessano | I miss my shirt... not her |
| 00:45.46 | drmessano | I WANT MY SHIRT BACK |
| 00:45.47 | tzanger | married again though... this one is tough to live with but it's because she's as stubborn as me |
| 00:45.53 | drmessano | lol |
| 00:45.55 | [hC] | bkruse: so i've just given the vlan thing another go, and still no luck. telnetting to say port 22 connects, but no data passes. its very odd. |
| 00:45.56 | drmessano | Same here.. |
| 00:46.33 | drmessano | This one is pretty good... even if she's a technodunce |
| 00:47.22 | lmadsen | you poor bastards and your wives |
| 00:47.40 | drmessano | She whips up something unique for dinner, and I tell her "Oh, get that from SVN?" she has no clue |
| 00:47.48 | tzanger | dont' care if she's a technodunce... she is crazy about me , she's smart and she doesn't back down, which is both a blessing and a curse |
| 00:49.01 | drmessano | If they get too clingy, there's always a Tazer |
| 00:49.24 | drmessano | Nothing wrong setting it on low and giving a little love tap when you need your space |
| 00:49.30 | lmadsen | pfft... I don't date girls that know stuff about computers -- conversations get boring |
| 00:49.31 | Qwell | low? |
| 00:50.03 | drmessano | Yeah one notch above "stun the cat" and just below "pee their pants" |
| 00:50.06 | drmessano | Thats the ideal setting |
| 00:50.08 | patrickv0x | I got a bunch of phones with extension 8XX, what kind of 'lines' i need to add in my extensions.conf to allow me to dial from 801 to 802 ? |
| 00:50.29 | lmadsen | patrickv0x: guess you didn't read the documentation I pointed you to |
| 00:50.49 | drmessano | At least he can cut/paste, lmadsen |
| 00:50.54 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.96.253) |
| 00:50.55 | patrickv0x | xten => 8XX,1,Answer |
| 00:50.55 | lmadsen | drmessano: or hit the up key :) |
| 00:50.59 | drmessano | heh |
| 00:51.01 | lmadsen | _8XX |
| 00:51.03 | lmadsen | not 8XX |
| 00:51.09 | lmadsen | when you understand the difference, come back |
| 00:51.10 | patrickv0x | ahh |
| 00:51.12 | patrickv0x | :-) |
| 00:51.14 | patrickv0x | thanks |
| 00:51.15 | patrickv0x | let me try |
| 00:51.22 | lmadsen | documentation is a glorious thing |
| 00:51.25 | drmessano | now he has to work at the UP |
| 00:51.40 | drmessano | Get them to talk.. CCCCCCCOMBO BREAKER |
| 00:51.48 | lunaphyte | how can i avoid having a litany of extensions listed in extensions.conf to match all of the various prefixes that should go out a particular channel? |
| 00:52.45 | husimon | lunaphyte, _X. => |
| 00:52.46 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 00:52.55 | husimon | err exten => _X. |
| 00:53.07 | [hC] | Qwell: are there parts of the aa50 fs that get wiped on every reboot? (for example /usr/lib/asterisk/modules)? If i overwrite something in there, and reboot, does it go back to a factory image, or will it keep my modification? |
| 00:53.27 | Qwell | [hC]: anything except what's listed in save_config, or on the CF |
| 00:53.50 | patrickv0x | can someone tell me what's wrong with this: exten => _8XX,1,,Dial(SIP/8XX,60,r) |
| 00:53.53 | [hC] | Qwell: everything gets wiped at boot... okay.. I was also gonna ask why / wasnt set read only... but i guess it kinda is. |
| 00:54.02 | [hC] | well until the next reboot anyways. |
| 00:54.16 | putnopvut | patrickv0x: too many commas |
| 00:54.33 | patrickv0x | ahh |
| 00:54.35 | patrickv0x | good catch |
| 00:54.35 | patrickv0x | thanks |
| 00:54.37 | patrickv0x | let me fix and reload |
| 00:54.42 | husimon | patrickv0x, it should be exten => _8XX,1,Dial(SIP/8XX,60,r) |
| 00:54.44 | putnopvut | And SIP/8XX probably isn't a valid channel. |
| 00:54.56 | putnopvut | s/channel/interface |
| 00:54.58 | husimon | yeah you need to replace XX with digits |
| 00:54.58 | putnopvut | s/channel/interface/ |
| 00:55.08 | putnopvut | ...I can't type. |
| 00:55.16 | lunaphyte | husimon: if i do that, how do i handle overlap? i have other extensions that are in there as well. |
| 00:55.30 | lmadsen | don't use the 'r' flag |
| 00:55.36 | lmadsen | it's almost never necessary |
| 00:55.36 | husimon | lunaphyte, just do SIP/${EXTEN} |
| 00:55.44 | husimon | then it gets what ever extension you dialed that matched the pattern |
| 00:56.03 | lmadsen | I heard there's a free book you can get that explains all of this |
| 00:56.11 | husimon | so exten => _8xx,1,Dial(SIP/${EXTEN},60) |
| 00:56.14 | husimon | lmadsen, me too |
| 00:56.17 | patrickv0x | thanks |
| 00:56.18 | patrickv0x | let me try |
| 00:56.39 | husimon | lunaphyte, sorry sort of mixed you and patrickv0x up. |
| 00:56.53 | drmessano | Asterisk + Wakeup.php > PAP2 > 60V DIAC > 120V Relay > Toaster |
| 00:56.58 | [hC] | Qwell: was there any developer documentation included in the aadk? I think i just got a regular s800i when i ordered my aadk... i could probably save a lot of time asking questions if there was a dev manual somewhere explaining a bunch of this stuff |
| 00:57.34 | husimon | lunaphyte, you'll have to explain your problem a little better for me to understand it. |
| 00:57.39 | *** part/#asterisk patrickv0x (n=patrick@67.131.93.17) |
| 00:57.51 | lunaphyte | husimon: no worries. |
| 00:58.16 | husimon | lunaphyte, my guess would use patterns that match more prefixes |
| 00:59.32 | lunaphyte | i have a fairly long list of prefixes that i'd like to go out a particular sip channel, but they don't really follow any pattern, so i'm wondering i can have this list of prefixes, but not a repetitive collection of extension entries. |
| 00:59.46 | lunaphyte | err, wondering how i can have... |
| 00:59.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:03.01 | putnopvut | lunaphyte: if they don't follow a pattern, but they all do the same thing, you can at least minimize the amount of extension listing by having them all call a common macro. |
| 01:03.13 | putnopvut | You'll still have to list each one at least once so that it calls that macro. |
| 01:03.24 | husimon | putnopvut, yeah i dunno i'd probably just leave it outside a macro |
| 01:03.30 | husimon | putnopvut, for simplicities sake |
| 01:03.42 | husimon | i guess unless you have hundreds |
| 01:04.06 | putnopvut | I'm out. Have a good night. |
| 01:04.21 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 01:04.22 | putnopvut | husimon: yeah, I guess it's a matter of preference, really. |
| 01:04.33 | husimon | lunaphyte, you could also include your outgoing patterns in a separate file |
| 01:04.36 | husimon | to make it neater |
| 01:05.25 | tzanger | bah, i can't seem to make 'indent' use tabs for the initial indent |
| 01:05.27 | tzanger | stupid |
| 01:05.40 | husimon | tzanger in what, vi? |
| 01:05.46 | tzanger | husimon: no, the indent utility |
| 01:06.09 | tzanger | I like -kr for the most part, but I can't seem to get it to do what I want for initial indentation of code |
| 01:06.31 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 01:07.41 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:08.41 | florz | Is there any reason why not all of digium's hardware is listed in the pci.ids file? |
| 01:20.27 | Robba | hi guys |
| 01:20.56 | Robba | any ideas why s,1,SetCallerID(numbergoeshere) isn't working? |
| 01:21.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:29.29 | _ShrikE | Robba: thats been deprecated for some time |
| 01:29.45 | jblack | robba: Yeah. Look at CallerID(num)=number |
| 01:30.12 | jblack | Pardon, Set(Callerid(num)=number) |
| 01:31.49 | *** join/#asterisk tuxfoo (n=tmmarini@pool-72-65-149-149.chrlwv.east.verizon.net) |
| 01:33.53 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:41.08 | *** join/#asterisk Daejeo (n=chatzill@211.177.189.80) |
| 01:41.39 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 01:42.01 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 01:42.02 | Daejeo | hello guys! any good voip for usa/canada? |
| 01:43.04 | pigpen | Hi all. When I use the page app more than a few times, I get "asterisk[29459] general protection rip:716b9ddf244 rsp:440e62d0 error:0" |
| 01:43.07 | pigpen | and asterisk pukes. |
| 01:43.14 | lunaphyte | husimon: here's a practical example from my dial plan : http://rafb.net/p/vwl3KT67.html |
| 01:43.24 | pigpen | running asterisk 1.4.17 |
| 01:44.04 | lunaphyte | how can i list patterns in a separate file? |
| 01:44.23 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:44.28 | pigpen | 53 phones, split between 3 groups, called as a page group (kinda like a ring group) |
| 01:44.31 | mosty | lunaphyte, #include |
| 01:44.46 | lunaphyte | oh, c style? |
| 01:45.17 | mosty | #include "filename" |
| 01:45.21 | pigpen | any help would be great, as if I don't get this settled, I'll get my ass ripped. |
| 01:45.42 | mosty | works in every asterisk config file i think |
| 01:47.50 | lunaphyte | that command simply substitutes the contents of the referenced file in place of that line, right? |
| 01:48.02 | lunaphyte | so it would need to contain proper syntax? |
| 01:50.00 | lunaphyte | ideally, it would be great if i could have a simple text file called local_prefixes.txt, for example, and in that file simply list any number of 3 digit prefixes, one per line. |
| 01:50.14 | husimon | lunaphyte, what I was saying |
| 01:50.20 | husimon | was that you could put all those in a separate file |
| 01:50.26 | husimon | but they would still look like that |
| 01:50.47 | lunaphyte | still look like standard asterisk syntax, you're saying? |
| 01:50.59 | husimon | ya |
| 01:51.02 | husimon | so you would create a context |
| 01:51.07 | husimon | put it in a file by it self |
| 01:51.12 | husimon | then do #include filename |
| 01:52.26 | husimon | i think you can also programatically define your dial plan with age ? |
| 01:52.39 | husimon | sorry not age |
| 01:52.41 | lunaphyte | regarding my imaginary file above (local_prefixes..) is it possible to then, in extensions.conf, have something along the lines of exten => _${magical_pattern}XXX,1,Dial,SIP/ata1.1/${EXTEN} |
| 01:52.46 | husimon | agi |
| 01:53.00 | lunaphyte | hmm |
| 01:53.03 | husimon | lunaphyte, yeah not that I know of |
| 01:53.07 | husimon | but i'm pretty new |
| 01:54.08 | mosty | lunaphyte, you could do that with AGI |
| 01:54.33 | husimon | mosty does that mean you need to use agi for your whole dialplan? |
| 01:54.41 | lunaphyte | a little bit of shell programming and some variable substitution, maybe, huh? |
| 01:54.46 | mosty | husimon, no |
| 01:55.14 | mosty | you could write an AGI script that parses a config file for patterns, then returns a pattern that matches all of them in some channel variable |
| 01:55.33 | lunaphyte | that sounds promising. |
| 01:56.05 | lunaphyte | probably overkill for my particular application, but worth learning for the sake of it, probably. |
| 01:56.07 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:56.22 | husimon | mosty i don't see how you would use that channel variable then |
| 01:56.35 | husimon | say you have numbers 1-100 |
| 01:56.44 | husimon | you have 50 randomly that need to goto one trunk |
| 01:56.49 | husimon | and the other 50 randomly goto the other trunk |
| 01:57.05 | husimon | you can't create a "pattern" that matches 50, you'd need to write 50 lines per each trunk |
| 01:57.28 | Robba | ok new issue |
| 01:57.54 | mosty | husimon, isn't there an OR operator for asterisk patterns? |
| 01:57.57 | Robba | on a 10 line PRI i can't dial out on another phone when someone is using another line |
| 01:58.01 | husimon | mosty i didn't think so |
| 01:58.04 | husimon | i can check |
| 01:59.35 | Mavvie | When I write to stderr in a AGI script, where does it send the output to? |
| 01:59.38 | lunaphyte | hmm, so i would have to call the agi script with agi() from within the dialplan? that seems like it might defeat the purpose. |
| 02:00.43 | husimon | Mavvie, i'd guess /var/log/asterisk/messages |
| 02:01.15 | husimon | hmm looks like it maybe goes to console |
| 02:01.42 | Mavvie | yes, I see it on the console but I don't see it in the logfiles. |
| 02:01.58 | Mavvie | maybe I should put debug in logger.conf |
| 02:02.16 | lunaphyte | funny, i just read that section in the book. |
| 02:02.19 | husimon | Mavvie you could try opening stderr to something |
| 02:02.20 | husimon | liek open STDERR, "| /usr/bin/logger -p local0.notice -t AGI"; |
| 02:02.22 | lunaphyte | page 157 ;) |
| 02:02.25 | mosty | lunaphyte, or you could just write a script to generate the dialplan from a file containing patterns, then regenerate and reload your dialplan when it changes |
| 02:03.24 | lunaphyte | that's true. |
| 02:06.15 | Mavvie | husimon: that seems to be the solution. |
| 02:06.17 | *** join/#asterisk HeXeD (n=hex@87-194-8-43.bethere.co.uk) |
| 02:08.49 | *** join/#asterisk saftsack (n=saftsack@p4FC74A6C.dip.t-dialin.net) |
| 02:08.58 | husimon | mavvie at least that's what I saw with a quick google search |
| 02:11.22 | Mavvie | hmm... seems like I can't give either channel variables nor command line arguments to AGI scripts initiated by call-files. |
| 02:12.41 | mosty | Mavvie, there are no channel variables to give, are there? |
| 02:13.16 | Mavvie | oh damned, maybe I've been looking at the API Action Originate instead of the call-file section. |
| 02:14.09 | Mavvie | aha, the syntax in the call-file is "Set: " |
| 02:14.16 | *** join/#asterisk NoRemorse (n=fred@203.217.93.153) |
| 02:14.20 | NoRemorse | hi all |
| 02:14.41 | NoRemorse | can anyone suggest a good rating and billing app for trixbox please? |
| 02:14.55 | Mavvie | woohoo! that works better. |
| 02:15.20 | mosty | NoRemorse, #trixbox |
| 02:15.23 | Mavvie | always tricky, two identical twins. |
| 02:15.52 | NoRemorse | ty |
| 02:17.18 | NoRemorse | pfft as expected, no answer in #trixbox |
| 02:19.45 | *** join/#asterisk gregg21 (n=Fender21@75-1-212-76.lightspeed.snantx.sbcglobal.net) |
| 02:20.58 | gregg21 | I was hoping for a little help with a question. How can I set extensions.conf to route all incoming calls straight to voicemail? |
| 02:21.11 | *** join/#asterisk zobia (n=laurashr@222.212.72.130) |
| 02:22.45 | mosty | gregg21, use an extension pattern that matches all extensions, and use the voicemail app |
| 02:23.16 | pigpen | mosty, I seem to remember that you are a bit "seasoned" with asterisk right? |
| 02:23.44 | gregg21 | I think that's what I'm getting to but just can't make it all the way..this is what I have in my extensions.conf |
| 02:23.45 | gregg21 | exten => s,1,VoiceMail(${EXTEN}) |
| 02:23.47 | mosty | pigpen, i've been using it for a few years, yes |
| 02:24.00 | NoRemorse | exten => _X.,2,Voicemail(${exten}@default,u) |
| 02:24.04 | NoRemorse | same same |
| 02:24.19 | pigpen | yeah, it's been awhile...deal much with the page app and 50+ sip phones? |
| 02:24.26 | gregg21 | what does the _X. do? |
| 02:24.26 | mosty | gregg21, that only matches the s extension, ie calls that don't have a destination extension |
| 02:24.53 | gregg21 | ah, thanks Mosty..that would explain the busy signals |
| 02:24.55 | mosty | pigpen, which page app? there's the bristuff one and the asterisk 1.4 one |
| 02:25.10 | pigpen | the asterisk 1.4 one. |
| 02:25.14 | mosty | gregg21, look up asterisk patterns on the wiki, or in the book |
| 02:26.22 | pigpen | anyway, I have a deployment with only about 65 polycom phones. page app was running fine (for the most part), moved to 1.4.17 and now it seems to be killing asterisk after about 10 pages. |
| 02:27.03 | pigpen | I ran into a similar issue with page app where the total number of phones I was paging exceeded the max command string length, causing a segfault. |
| 02:27.14 | pigpen | however this was with paging 180 phones. |
| 02:28.03 | *** join/#asterisk Faithful (n=Faithful@202-136-108-110.static.adam.com.au) |
| 02:28.08 | mosty | pigpen, asterisk 1.4 is buggy |
| 02:28.39 | pigpen | yeah, for the most part it has been good, but the dam page app has been an issue. |
| 02:28.45 | mosty | pigpen, i recommend that you revert to whatever you were using before when it was working, and submit a bug report in the meantime |
| 02:29.03 | pigpen | yeah....kind what I was thinking. |
| 02:29.16 | pigpen | any benefit of the bristuff "stuff'? |
| 02:31.12 | mosty | i try to avoid bristuff if i can, i only use it if it gives me a feature i need on asterisk 1.2 when 1.2 doesn't have the feature itself |
| 02:31.23 | pigpen | ah.... |
| 02:31.38 | pigpen | we moved to 1.4.x pretty early as it had realtime postgresql support. |
| 02:32.03 | pigpen | Let me tell you, that makes me a leaper in the realm of getting help... |
| 02:32.27 | mosty | i still have showstopper bugs with 1.4 |
| 02:32.39 | pigpen | like what? |
| 02:32.57 | pigpen | other than paging? :) |
| 02:33.52 | zobia | hello everyone |
| 02:34.20 | pigpen | everyone says "hello zobia" |
| 02:34.34 | zobia | pigpen :) |
| 02:34.48 | pigpen | well, someone had to do it. |
| 02:34.54 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:34.54 | zobia | finally i install 1.4 on a new machine to try the skinny stuff |
| 02:34.59 | zobia | pigpen, thank you |
| 02:35.02 | mosty | pigpen, i have problems with thread deadlocks |
| 02:35.43 | pigpen | I know that with iax there is an entry something like maxtreads= |
| 02:35.57 | pigpen | maybe something like that for skinny....but I haven't done much with it. |
| 02:36.19 | zobia | i find that i can not see registering infomation when the phone connect to the box. i use set verbose 10 and skinny set debug , still can not see anything. how can i see if a phone is trying to register? |
| 02:37.22 | pigpen | tcpdump ? |
| 02:37.36 | pigpen | if asterisk won't show it, tcpdump will. |
| 02:37.50 | pigpen | or not. |
| 02:37.51 | pigpen | :) |
| 02:38.17 | zobia | pigpen: let me ask a question not related to asterisk |
| 02:38.25 | zobia | pigpen: let me ask a question not related to skinny |
| 02:38.27 | pigpen | sorry, I am married. |
| 02:38.28 | zobia | [Feb 6 02:34:01] WARNING[15131]: config.c:1316 find_engine: Realtime mapping for 'realtime_ext' found to engine 'mysql', but the engine is not available |
| 02:38.32 | gregg21 | heh |
| 02:38.41 | zobia | i got this error after i move from 1.2 to 1.4 |
| 02:38.50 | zobia | any idea? |
| 02:39.05 | pigpen | looks like realtime is enabled....turn it off in extconfig.cfg |
| 02:39.35 | pigpen | oops, extconfig.conf |
| 02:40.01 | zobia | yes. i need realtime enable |
| 02:40.12 | pigpen | well, in that case... |
| 02:40.21 | zobia | pigpen: sorry i need realtime enable. but don't know why it don't work anymore |
| 02:40.37 | zobia | pigpen:i use the same extconfig while the 1.2 use |
| 02:40.39 | lunaphyte | is there a difference between exten => _624XXXX,2,Dial,SIP/ata1.1/${EXTEN} and exten => _624XXXX,2,Dial(SIP/ata1.1/${EXTEN}) ? |
| 02:40.41 | pigpen | possibly it is having a login issue. |
| 02:40.54 | mosty | zobia, are you missing the asterisk-addons package? |
| 02:40.55 | pigpen | I think a few things changed for the auth between 1.2 and 1.4 |
| 02:41.01 | *** join/#asterisk matthew_i (n=matthew@pdpc/supporter/sustaining/MasterYoda) |
| 02:41.10 | matthew_i | what time is it in nashville? |
| 02:41.15 | zobia | mosty: i installed asterisk-addons package. |
| 02:41.19 | pigpen | 9:41 |
| 02:41.20 | pigpen | pm |
| 02:41.27 | matthew_i | thanks |
| 02:41.41 | pigpen | nashville is eastern right? |
| 02:41.54 | pigpen | if so, 9:42p |
| 02:42.16 | lunaphyte | i believe so. |
| 02:42.24 | zobia | mosty: i installed asterisk-addons-1.4.5 , is it a problem? |
| 02:43.12 | mosty | i don't understand what you're asking exactly. i know you need asterisk-addons for mysql support |
| 02:43.44 | zobia | mosty: yes i do installed that. but still get engine is not avaliable error |
| 02:43.49 | *** join/#asterisk RoyK (n=roy@ip-2-29-149-91.dialup.ice.no) |
| 02:43.50 | *** join/#asterisk matthew_i (n=matthew@pdpc/supporter/sustaining/MasterYoda) |
| 02:44.04 | pigpen | it could be several things, but probably an auth issue. |
| 02:44.16 | matthew_i | pigpen: it's 8:44 in nashville |
| 02:44.16 | mosty | zobia, check your mysql and asterisk logs |
| 02:44.16 | pigpen | do a debug on mysql and see if it is connecting. |
| 02:44.36 | pigpen | matthew_i, shit, then why is my dell rep always leaving an hour early....bastard. |
| 02:44.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 02:44.49 | matthew_i | pigpen: thanks anyway :) |
| 02:46.56 | lunaphyte | i'm looking at an example on a itsp's support page for use with asterisk : exten => _1NXXNXXXXXX,2,Dial,IAX2/1234@itsp/${EXTEN} - is that syntax valid? |
| 02:47.20 | lunaphyte | should it not be Dial(.......) ? |
| 02:47.25 | plik | lunaphyte: Don't think so. |
| 02:47.28 | plik | exactly |
| 02:47.30 | plik | AFAIK |
| 02:47.38 | lunaphyte | it's weird - i think it works. |
| 02:47.40 | plik | DIAl(parameters,,) |
| 02:47.55 | plik | odd |
| 02:48.00 | lunaphyte | i had just copied and pasted ages ago without paying attention, and have been using it for some time. |
| 02:48.12 | plik | nearly as odd as that matthew |
| 02:48.53 | zobia | mosty:ok |
| 02:49.03 | plik | maybe its a format thats deprecated, but still works |
| 02:49.27 | lunaphyte | that's what i was just thinking. |
| 02:49.35 | *** part/#asterisk RoyK (n=roy@ip-2-29-149-91.dialup.ice.no) |
| 02:50.01 | VitoCorleon | hey guys i have installed a TDM400P and do not see any calls comming in |
| 02:50.18 | zobia | pigpen: it works. the mysql connection failed. thank you |
| 02:50.28 | VitoCorleon | and when i do zap show channels all i get is "Chan Extension Context Language MOH Interpret" |
| 02:51.19 | zobia | any one have experience of skinny? |
| 02:51.49 | mosty | zobia, very few people use it, compared to sip and iax |
| 02:51.50 | zobia | when my phone try to dial , it unregistered the phone |
| 02:52.13 | zobia | mosty: yes i know that. thank you. |
| 02:52.27 | lunaphyte | i use sccp. |
| 02:52.36 | x86 | zobia: put the sip firmware on the phone |
| 02:52.37 | lunaphyte | on one lonely phone, for s and g. |
| 02:52.43 | VitoCorleon | please guys i need support, im at my clients place right now |
| 02:52.57 | zobia | x86 my phone could not change to sip. this is a problem |
| 02:53.19 | zobia | lunaphyte: do you know skinny? |
| 02:53.31 | x86 | zobia: ah, i see |
| 02:53.46 | lunaphyte | probably not enough to be of any practical value. |
| 02:53.52 | x86 | VitoCorleon: what's my cut if I help you? |
| 02:54.03 | VitoCorleon | lol whats the charge? |
| 02:54.06 | zobia | lunaphyte: i use chan_sccp before on 1.4 but my phone keep registered and locked so someone here suggest me to use chan_skinny |
| 02:54.09 | VitoCorleon | im getting 120 to set it up |
| 02:54.14 | x86 | $55/hr |
| 02:54.22 | x86 | friend price |
| 02:54.23 | VitoCorleon | sure |
| 02:54.27 | x86 | sure? |
| 02:54.33 | VitoCorleon | half lol i guess better then not finishing |
| 02:54.41 | lunaphyte | zobia: i wedged some add on into asterisk ages ago, fiddled around until it functioned and then never went back to it. |
| 02:54.51 | x86 | if it takes more than an hour, that's more than half ;) |
| 02:54.53 | lunaphyte | zobia: yeah, i think it's chan_sccp |
| 02:54.56 | VitoCorleon | lol |
| 02:58.28 | zobia | lunaphyte: did you have problem that it could not register phone or could not release the connection with the phone? |
| 02:59.13 | Inssomniak | what is "zaptel" exactly? google seems to pull up a calling card? |
| 02:59.45 | lunaphyte | zobia: to be honest, it was so long ago, i don't quite recall specifically. i think i did have trouble getting it to register, at least initially, but i think that there were a handful of issues. |
| 03:00.27 | mosty | Inssomniak, it's a driver for digium, sangoma etc telephony cards |
| 03:00.37 | lunaphyte | Inssomniak: software |
| 03:00.43 | x86 | Inssomniak: it's a telephony driver |
| 03:00.50 | Inssomniak | thx! |
| 03:01.06 | x86 | Inssomniak: it runs telephony cards such as POTS, BRI, and T1/E1/J1 interfaces |
| 03:01.34 | zobia | lunaphyte: thank you . i think i can not use sccp_chan. it's so difficult to make it run. so i am trying skinny |
| 03:01.58 | lunaphyte | zobia: what phone? |
| 03:02.06 | Inssomniak | if I was to spend the money on a digium card, with FXO and FXS, is it really that much better than say a SPA 3102 ata? |
| 03:02.56 | mosty | Inssomniak, you're better off with an ATA for FXS |
| 03:03.14 | mosty | and the sangoma cards are better than digium's |
| 03:04.46 | Inssomniak | mosty, so far the ATAs are working better than I imagined for FXS, but for FXO (my pots line), I get some weird echos and overall quality of sound is not that good |
| 03:04.57 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 03:05.17 | Inssomniak | sorry, the othe way around I think |
| 03:05.35 | mosty | Inssomniak, sangoma a200d is my preferred analogue telephony card |
| 03:06.34 | zobia | lunaphyte: i use 7910 and 7960 |
| 03:06.36 | *** join/#asterisk tuxfoo (n=tmmarini@pool-72-65-149-149.chrlwv.east.verizon.net) |
| 03:08.00 | Robba | can someone tell me where i have gone wrong, when i dial out on our 10 Channel ISDN no one else can seem to make calls |
| 03:09.28 | mosty | Robba, pastebin your dialplan |
| 03:11.11 | zobia | any one know why my redial and transfer button not working with skinny? or skinny does not support this? |
| 03:11.16 | husimon | so i have a problem |
| 03:11.25 | husimon | for some reason all of a sudden I can't hear anything from the voicemail menus |
| 03:11.35 | husimon | i had my system on heartbeat, swapped to secondary, came back to primary |
| 03:11.45 | husimon | and now the voicemailmain doesn't even work |
| 03:11.50 | husimon | no errors on cli |
| 03:12.27 | husimon | hmm something is wrong with my zaptel chan |
| 03:12.59 | Mavvie | very inconsistent behaviour: for zap channels you do need agi->answer, for SIP channels you don't need it. |
| 03:14.13 | *** join/#asterisk johndbritton (n=john@cpe-72-226-79-202.nycap.res.rr.com) |
| 03:14.45 | johndbritton | I've checked my authentication multiple times, anyone know of a way I can troubleshoot this error "[Feb 6 03:11:00] WARNING[9181]: chan_sip.c:8272 check_auth: username mismatch, have <102>, digest has <1 02>" |
| 03:19.17 | lunaphyte | zobia: oh, i just have a really old 12sp+ |
| 03:22.27 | Robba | http://rafb.net/p/h8dj9A26.html |
| 03:22.29 | zobia | lunaphyte: do you know how to call a skinny channel in the dialplan? |
| 03:22.47 | zobia | lunaphyte: is it like dial(skinny/phonenumber|30|r)? |
| 03:22.55 | lunaphyte | i call mine like this: exten => 1151,1,Dial(SCCP/1151,20,tr) ; cisco 12sp |
| 03:23.28 | zobia | still use sccp? oh sorry i forget you are not using chan_skinny. |
| 03:23.37 | lunaphyte | right. |
| 03:23.54 | zobia | i use both sccp and skinny to dialthat phone. bot said could not create channel type sccp or skinny. |
| 03:23.54 | Robba | mosty: http://rafb.net/p/h8dj9A26.html |
| 03:24.07 | zobia | lunaphyte: thank you anyway for help |
| 03:24.13 | lunaphyte | aside from 800, 888, 877, and 866, are there other toll-free area codes? |
| 03:24.35 | Frogzoo | mosty: in what way sangoma's better than the digium analogue cards? |
| 03:24.42 | lunaphyte | i saw a reference to 844, 844, 833 and 822 - but it sounded like those may not yet be in use? |
| 03:24.56 | lunaphyte | zobia: sure, sorry i couldn't be more help |
| 03:25.08 | mosty | Frogzoo, better sound quality, better debugging utils, better driver |
| 03:26.07 | Frogzoo | better driver than digium? that's a big call |
| 03:26.38 | zobia | lunaphyte: no problem. hope i canfind that developer who suggest me to use chan_skinny |
| 03:27.12 | Corydon76-dig | Better driver? That's weird, since Sangoma's driver is essentially a bastardized Digium driver |
| 03:27.24 | zobia | @Qwell: hello, do you know "could not create skinny channel" if i already load the module for skinny.so ? |
| 03:27.47 | mosty | wanpipe just hooks into zaptel |
| 03:28.18 | Corydon76-dig | According to some of the module authors, it does so in a potentially dangerous way |
| 03:28.37 | mosty | recent versions of wanpipe don't even patch zaptel |
| 03:28.58 | mosty | i've never had irq issues with wanpipe |
| 03:29.05 | Corydon76-dig | So they finally fixed their stuff? That's good. Digium has also fixed their sound quality issues. |
| 03:29.11 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-112-81.knology.net) |
| 03:29.53 | mosty | i never had problems with wanpipe+zaptel anyway |
| 03:30.46 | *** join/#asterisk PepOSX (n=angeldav@190.72.146.204) |
| 03:34.55 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-124-120.tor.primus.ca) |
| 03:36.39 | pigpen | this is great...I do 15 pages in a row, to 60 phones....works great. |
| 03:36.50 | pigpen | take a piss, come back....first one bombs the server. |
| 03:36.58 | pigpen | well, bombs asterisk anyway. |
| 03:37.48 | Robba | Dial Plan http://rafb.net/p/h8dj9A26.html |
| 03:38.01 | Robba | can someone take a look and tell me whats wrong? |
| 03:38.40 | eric2 | anyone ever use asterisk fax? or get faxing going using g.711? |
| 03:39.02 | mosty | eric2, it's unreliable, even on a LAN |
| 03:39.41 | pigpen | I use asterisk/iaxmodem/hylafax with great success. |
| 03:39.50 | pigpen | (you see, it doesn't include paging, so I am ok...) |
| 03:40.24 | Mavvie | I have given up on faxing with asterisk after I talked to the person who made libdspan and ap_[rt]xfax. |
| 03:40.37 | eric2 | so in general, should I just forget about the faxing setup? I'd like fax to email... but sending fax's is the issue at hand |
| 03:40.50 | Mavvie | I'm now back to a DSP card with enough memory to do everything in hardware. |
| 03:40.51 | eric2 | hmm, not good to hear Mavvie :( |
| 03:41.34 | Mavvie | eric2: it might work fine on the USA phonesystem, but outside the USA its euhm.... three times nothing. |
| 03:41.37 | pigpen | once again, I have several good size deployments using asterisk/iaxmodem/hylafax. |
| 03:41.50 | pigpen | delivery to email...works great. |
| 03:41.55 | eric2 | I'm within north america |
| 03:42.07 | mosty | eric2, hylafax works well, but i would never recommend using sip or iax anywhere on the fax call path |
| 03:42.37 | pigpen | eric2, yeah, I am in Texa |
| 03:42.39 | eric2 | pigpen, are you within north america? |
| 03:42.43 | Mavvie | mgetty+sendfax + hylafax work like a charm here. |
| 03:42.44 | pigpen | s/texa/Texas |
| 03:42.46 | eric2 | ah |
| 03:42.53 | pigpen | iaxmodem is local, so no issues. |
| 03:43.03 | eric2 | iaxmodem, is that software? |
| 03:43.06 | pigpen | ie: never touches the ethernet segment. |
| 03:43.09 | pigpen | yup. |
| 03:43.19 | eric2 | hylafax I"ve seen before somewhere |
| 03:43.29 | eric2 | so your setup is working over g.711? |
| 03:43.35 | pigpen | no. |
| 03:43.51 | eric2 | oh ya, iax? |
| 03:43.59 | eric2 | I'll have to read up on what you're using |
| 03:44.00 | pigpen | iaxmodem is a iax client to asterisk. |
| 03:44.17 | pigpen | hylafax uses iaxmodem, well, as a modem. |
| 03:44.25 | mosty | pigpen, how do you send faxes with iaxmodem? ie how do your end users use it? |
| 03:44.46 | pigpen | you can split iaxmodem and hylafax, but not iaxmodem and asterisk. |
| 03:45.00 | pigpen | I use a hylafax client....iaxmodem is behind the scenes. |
| 03:45.39 | mosty | pigpen, ahh, i use a dedicated machine for hylafax, so i have no need to asterisk in that setup |
| 03:45.39 | eric2 | so do you have a normal gateway like a linksys 2102 with the legacy fax connected to it? |
| 03:45.49 | pigpen | no. |
| 03:46.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:46.19 | eric2 | what does the physical setup look like? |
| 03:46.20 | pigpen | if I have any legacy fax's, I connect them directly to an fxs off the asterisk box. |
| 03:46.27 | eric2 | yikes |
| 03:46.50 | eric2 | I'm all digital |
| 03:47.24 | pigpen | define digital....like you have lcd's where your eyes are? |
| 03:47.37 | pigpen | Johnny Neumonic? |
| 03:48.05 | eric2 | ha! |
| 03:48.34 | eric2 | all my client phones connect to asterisk remotely |
| 03:48.39 | eric2 | asterisk is not running on site |
| 03:48.58 | eric2 | typical hosted setup |
| 03:49.05 | drmessano | Getting the fax TO asterisk over SIP is iffy at best |
| 03:49.10 | pigpen | then use a client software. |
| 03:49.46 | mosty | eric2, fax over voice over ip is never going to work well |
| 03:50.06 | pigpen | well, put a small box locally, running hylafax, have it use the modems (iaxmodem) remotely.... |
| 03:50.25 | pigpen | well...what...got it backwards. |
| 03:51.00 | pigpen | that would be having the customer manage their own fax server. |
| 03:51.54 | eric2 | ok, so let me repeat my muffled understanding.. have a computer running locally with hylafax running on it |
| 03:51.59 | pigpen | I have had pretty good success using a remote asterisk box via an iax trunk, with on the remote side, having a 24port fxs, connected to a fax... |
| 03:52.02 | pigpen | good results. |
| 03:52.22 | pigpen | but the link is a DS3 |
| 03:52.38 | eric2 | I wanted to stay away from any additional hardware.. now I see its all just a pipe dream |
| 03:52.48 | drmessano | Hmm |
| 03:54.04 | pigpen | may be easier just to have them pick up a few flat lines, probably need them for alarm systems anyway. |
| 03:54.21 | Robba | Ok can someone help with my extensions.conf file |
| 03:54.33 | eric2 | ask away Robba |
| 03:54.47 | Robba | http://rafb.net/p/h8dj9A26.html |
| 03:54.53 | Robba | thats the dial plan |
| 03:55.17 | Robba | but we have a 10 Channel ISDN circuit and only one person can dial out at once |
| 03:55.56 | Robba | example is i'm on the phone no one else can dial out |
| 03:56.02 | mosty | robba: you are only dialing over a single zap channel |
| 03:56.21 | Robba | how do i fix this? |
| 03:56.31 | pigpen | use a group. |
| 03:56.36 | pigpen | g0 |
| 03:56.37 | pigpen | or G0 |
| 03:56.41 | pigpen | depending on glare. |
| 03:56.42 | mosty | dial(ZAP/g1/...etc) not dial(ZAP/1/...etc) |
| 03:56.52 | Robba | ahhhh |
| 03:56.55 | pigpen | mosty, beat you...but yours was prettier. |
| 03:57.04 | Robba | cause i had g0 in originally |
| 03:57.06 | scooby2 | Whats the proper way to set "default gateway" for zaptel in 1.4? I was using ZAP/g2 in 1.2 |
| 03:57.09 | Robba | but it didn't work |
| 03:57.16 | *** join/#asterisk RoyK (n=roy@ip-2-29-149-91.dialup.ice.no) |
| 03:57.28 | mosty | Robba, you need to define groups in zapata.conf |
| 03:57.57 | Robba | JT from this channel re wrote my Zapata.conf |
| 03:58.13 | pigpen | sure...blame it on JT.... |
| 03:58.14 | pigpen | :) |
| 03:58.16 | Robba | want me to pastebin zapata? |
| 03:58.28 | Robba | i'm not laying blame |
| 03:58.43 | Robba | after he fixed that my ISDN actually connected |
| 03:58.51 | Robba | lol |
| 03:58.53 | scooby2 | thats it the zapata group |
| 03:59.24 | *** part/#asterisk RoyK (n=roy@ip-2-29-149-91.dialup.ice.no) |
| 03:59.29 | Robba | pastebin? |
| 04:00.22 | mosty | yes |
| 04:01.07 | Robba | http://rafb.net/p/Hbivd586.html |
| 04:01.53 | pigpen | mosty, man, I have done like 50 pages in a row, like 15 seconds apart...no issue. |
| 04:01.56 | pigpen | weird eh? |
| 04:02.08 | pigpen | Seems like if the system has been idle, it happens then. |
| 04:02.38 | *** join/#asterisk joez212 (n=jhart@CPE001c101b40b5-CM0018c0d91624.cpe.net.cable.rogers.com) |
| 04:02.41 | joez212 | i got it working! |
| 04:02.46 | joez212 | thanks for everyone who helped me |
| 04:02.48 | joez212 | :) |
| 04:03.01 | JT | Robba: was it a different version of zap/ast that fixed it? |
| 04:03.42 | Robba | nah when you changed my zaptel and zapata, it just seemed to work |
| 04:03.44 | mosty | pigpen, submit a bug report |
| 04:03.48 | *** join/#asterisk weazahl_ (n=jeremy@adsl-66-143-53-16.dsl.ksc2mo.swbell.net) |
| 04:04.05 | JT | Robba: oh ok, i don't remember you telling me this :P |
| 04:04.07 | pigpen | well, I would like to have more rhyme and reason...but yeah, I plan to. |
| 04:04.20 | drmessano | At least I have chicken |
| 04:04.21 | Robba | well i also reinstalled as well |
| 04:04.35 | JT | ah |
| 04:04.48 | Robba | LEEEEEEEEEEROYYYYYYYY JJJJENNNNKKKIIINNNNSSSS |
| 04:05.14 | joez212 | so ya |
| 04:05.46 | joez212 | centos sucks |
| 04:05.46 | Robba | so when i reinstalled and put in your zapata.conf and zaptel.conf it all seemed to come together |
| 04:05.46 | joez212 | debian is currently my favourite * os |
| 04:05.46 | joez212 | :) |
| 04:05.46 | JT | :) |
| 04:05.46 | scooby2 | This crack asterisk dCAp certified guru made us this IVR and it looks like he copied our 1.2 ivr and added waitexten. When you call it it hangs up after the |
| 04:05.50 | scooby2 | welcome |
| 04:05.51 | scooby2 | http://rafb.net/p/DVpCPl44.html |
| 04:05.59 | scooby2 | should the WaitExten be moved down? |
| 04:10.16 | Robba | ok after changing to g0 it still doesn't seem to work |
| 04:10.53 | JT | g1... |
| 04:10.55 | scooby2 | g1 |
| 04:10.59 | Robba | ahhh |
| 04:11.03 | Robba | i see |
| 04:11.06 | Robba | *nods* |
| 04:11.06 | JT | G1 or g1 |
| 04:11.07 | scooby2 | you have group=1 in your zapata.conf |
| 04:11.14 | JT | they just dial in different orders |
| 04:11.36 | JT | Robba: change channel to 1-10 |
| 04:11.43 | JT | since you only have 10 |
| 04:11.54 | JT | you were probably trying to dial from the highest number down |
| 04:11.59 | JT | and since it's fractional pri |
| 04:12.02 | JT | it was failing |
| 04:12.14 | JT | only modify zapata.conf |
| 04:12.18 | JT | leave zaptel as is |
| 04:13.08 | pigpen | just in case anyone is wondering: |
| 04:13.21 | pigpen | g1 dial out 1,2,3,4..... |
| 04:13.30 | pigpen | G1 dial out 10,9,8,7..... |
| 04:13.42 | pigpen | this is for glare consideration... |
| 04:14.02 | pigpen | ie: your telco probably sends in calls on channels 1,2,3,4.... |
| 04:14.16 | pigpen | so you would want to dial out 10,9,8,7,.... |
| 04:18.35 | JT | it probably makes little difference on a pri though |
| 04:18.44 | pigpen | heavy use it does. |
| 04:18.54 | pigpen | and I have 4 |
| 04:19.55 | JT | in the case of asterisk dialling out at the exact same instance as an incoming call? |
| 04:21.00 | joez212 | my extensions.conf has this |
| 04:21.18 | joez212 | exten => 9250,1,dial)sip/9250,20) |
| 04:21.24 | joez212 | next line is hangup |
| 04:21.35 | joez212 | i thought I could dial 1000 and hear the voicemail services |
| 04:21.59 | joez212 | or should i use the default extensions.conf? |
| 04:24.21 | *** join/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
| 04:27.16 | pigpen | JT, it was more of a bitch fest with a telco. |
| 04:27.26 | pigpen | odd things were happening, glare came up. |
| 04:27.44 | pigpen | telco's like to bitch if they see this when things are happening and they don't know what to do. |
| 04:27.57 | pigpen | But I have seen oddities with the glare incorrect...yes. |
| 04:27.57 | JT | heh |
| 04:28.18 | *** join/#asterisk Swabby (n=e741533@12.46.189.1) |
| 04:28.32 | Swabby | Hey. Just a quick question. What do folks typically use for the DHCP server on an Asterisk System? |
| 04:28.34 | dudes | Teleco's seem to be going into the toilet the last few years |
| 04:28.50 | phix | Swabby: I use bind9 |
| 04:28.56 | pigpen | Yeah, but ATT will come around.... :) |
| 04:29.06 | pigpen | after they split. |
| 04:29.10 | scooby2 | ma bell is back |
| 04:29.10 | Swabby | phix: I was thinking about buying a switch that did this. Do you recommend running bind on the asterisk server instead? |
| 04:29.15 | phix | I have a spanking new TDM400p card with 3 FXS modules |
| 04:29.39 | phix | I can only get two working |
| 04:30.08 | joez212 | does the default extensions.conf allow you dial 1000 from a sip extension and check voicemail, etc |
| 04:30.10 | phix | I tried shuffeling the modules around, and they all work, it is just the third and forth doesn't. the first and second do |
| 04:30.23 | dudes | perhaps a config issue |
| 04:30.29 | phix | I am thinking configu issue |
| 04:31.00 | phix | but I have fxo_ks=1-3 in /etc/zapatel.conf and I define them in /etc/asterisk/zapata.conf |
| 04:31.03 | joez212 | * is tricky to setup |
| 04:31.08 | phix | any ideas? |
| 04:31.15 | dudes | it's strange how the configs work in the en |
| 04:31.25 | phix | Any one experienced a similar prog and resolved ot? |
| 04:31.35 | phix | prog = prob |
| 04:31.50 | dudes | perhaps try setting 1-2 and 3-4 sep |
| 04:32.06 | phix | I tried setting them all seperate |
| 04:32.31 | dudes | I've never done one of those cards but if the config is anything like having multiple t410p's |
| 04:32.35 | Mavvie | my boss wants me to install trixbox to overcome this problem: http://bugs.digium.com/view.php?id=11917 |
| 04:32.41 | Mavvie | then he wants me to do it with SER. |
| 04:33.04 | dudes | trixbox --- OpenSER |
| 04:33.05 | Daejeo | Mavvie: who is ur boss? |
| 04:33.26 | Swabby | We're using VoiP phones, Has anyone hooked it up where it's SEPERATE from the other "network" inhouse and had a switch with dhcp in which the phones connect? |
| 04:33.32 | phix | dudes: ok, hmmmm so what are my choices? |
| 04:33.45 | scooby2 | Swabby: yes |
| 04:33.47 | Mavvie | and finally he found a new codebase call Yate which I now need to install and make it in. |
| 04:33.51 | joez212 | interesting set of problems on here |
| 04:33.51 | dudes | I'm thinking try to pair 1-2 and then run 3 |
| 04:33.56 | phix | dudes: I have signed up to the mailing list, should I ask there? |
| 04:34.01 | Mavvie | Daejeo: www.barnet.com.au |
| 04:34.13 | phix | dudes: done :) doesnt work |
| 04:34.19 | dudes | hmm |
| 04:34.21 | dudes | strange |
| 04:34.36 | dudes | are all the modules fxo's? |
| 04:34.44 | phix | dudes: It makes the same sound as if zaptel isn't running (I hear whatever goes into my mic) |
| 04:34.47 | phix | fxs |
| 04:34.54 | phix | with fxo signalling |
| 04:35.10 | dudes | I know, they are fxo |
| 04:35.18 | dudes | it's all ass backwards hehe |
| 04:35.30 | phix | the actual module on the card are FXS |
| 04:35.41 | *** join/#asterisk pc600 (n=fewa@69.92.253.90) |
| 04:35.57 | phix | They provide 3 more lines to an existing PBX system |
| 04:35.59 | dudes | I thought your config was fxx |
| 04:36.08 | dudes | err fxs, but I"m drunk so |
| 04:36.09 | Swabby | scooby: do you have a paticular model you recommend from a hardware switch perspective? |
| 04:36.17 | phix | dudes: :) |
| 04:36.25 | phix | dudes: what you drinking? :) |
| 04:36.33 | drmessano | dudes: I gave up FXS and FXO.. I use FXP now |
| 04:36.37 | phix | or jsut a figure of speach? |
| 04:36.40 | drmessano | FXP pwns |
| 04:36.43 | pc600 | Scenario problem: I have an office in another state (USA) with 5 people. I would like them to use our private WAN going to this location for VoIP. How do I get a DID to them? I have a PRI in another state, but I Can't get out-of-state (or lata) DIDs from the telco. |
| 04:36.44 | phix | FXP? |
| 04:36.47 | dudes | some Michelob Draft |
| 04:37.06 | drmessano | Yep |
| 04:37.08 | scooby2 | Swabby: you can always run dhcp on your asterisk box. just make sure it has two interfaces. One for the phone network and one for the normal network. |
| 04:37.09 | dudes | not my prime choice but beer left over from the weekend |
| 04:37.11 | phix | who? |
| 04:37.24 | pc600 | scooby2 - Or trunk one port :) |
| 04:37.30 | scooby2 | or that |
| 04:37.31 | phix | dudes: ok, not a big beer fan my self. Bourbon++; for me |
| 04:37.55 | dudes | phix - I prefer whiskey myself, but it's rare to find anyone willing to go shot for shot |
| 04:38.02 | drmessano | Warm Brandy and a snuff pipe FTW, old chap |
| 04:38.18 | Swabby | scooby: i wouldn't need an extra interface if i was using a card for Analog lines though right? |
| 04:38.21 | pc600 | Anyone have any advice for my DID problem? |
| 04:38.25 | drmessano | Johnny Walker Red Label is the best for figuring out problems |
| 04:38.41 | dudes | I like Michael Collins Blend |
| 04:38.42 | drmessano | Johnnie rather |
| 04:38.44 | drmessano | bah |
| 04:39.40 | drmessano | If Black Label will get just a LITTLE cheaper, i'll start investing in it more.. Much better stuff |
| 04:40.22 | Swabby | that's funny. My name is John Walker..people mention red all the time |
| 04:41.03 | drmessano | lol |
| 04:41.18 | drmessano | Swabby, you and I have met on many occasions |
| 04:41.20 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:41.24 | dudes | <drmessano> - give a bottle of Mick's a chance --- not much cheaper but man, freeze it up and put it on the rocks |
| 04:41.32 | dudes | it's good |
| 04:42.05 | drmessano | I can tell you what to AVOID.. Dewars |
| 04:42.27 | drmessano | Dewars is blended from the worlds finest goat piss |
| 04:42.35 | dudes | haha |
| 04:42.42 | dudes | never tried let alone heard of it |
| 04:43.23 | drmessano | When I was a tee.... when I reached legal drinking age, I used to drink that stuff all the time.. I spent more time hungover than drunk.. and I would be sick for days afterwards |
| 04:43.44 | dudes | that's like Phillips then |
| 04:43.53 | drmessano | Getting even slightly better Scotch made all the difference |
| 04:44.03 | Swabby | nice hehehe |
| 04:44.07 | dudes | I off times wonder about that |
| 04:44.37 | dudes | at the end of the day it's booze, but certain booze seems to be better for the system yes |
| 04:45.20 | Frogzoo | what's the pathname to the dialplan? |
| 04:45.25 | dudes | I say die Phillips, except root 100 if managed right |
| 04:45.26 | drmessano | I've learned as I have gotten older that cheap alcohol is just that.. cheap.. in every way.. Better stuff is worth it, not just for taste, but for the lack of rat poison and goat piss they use in the cheap stuff |
| 04:45.45 | dudes | I agree |
| 04:46.10 | dudes | I'd rather drop $40 on a decent bottle than be overly sick the next morning --- vs days |
| 04:46.11 | drmessano | You can also drink LESS and get MORE from it |
| 04:46.28 | dudes | agreed, cause it taste better, at least in my case |
| 04:46.48 | drmessano | Cheap alcohol is good for chugging a bunch and passing out |
| 04:46.51 | drmessano | For $10, sure |
| 04:46.58 | dudes | Although I kind of like Cherry Coke and Black Velvet |
| 04:47.15 | dudes | but it's not my choice, but I just like it sometimes |
| 04:47.31 | dudes | but I steer away from it cause I'm boycotting Canadians |
| 04:47.40 | drmessano | lol |
| 04:47.52 | drmessano | So no Elsinore beer for you? |
| 04:48.01 | dudes | no |
| 04:48.04 | dudes | hehe |
| 04:48.19 | *** part/#asterisk joez212 (n=jhart@CPE001c101b40b5-CM0018c0d91624.cpe.net.cable.rogers.com) |
| 04:48.23 | dudes | Canadians are pissing right and proper lately |
| 04:48.43 | dudes | Americans are lazy but they take the cake |
| 04:48.49 | drmessano | I'm not going to boycott Molson |
| 04:48.52 | drmessano | Not for anyone |
| 04:49.09 | dudes | never tried Molson but I liked commercials |
| 04:49.15 | Frogzoo | is the dialplan a single file - or an aggregation done by asterisk from everything in /etc/asterisk/ ? |
| 04:49.18 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 04:49.33 | dudes | Frogzoo - it's like one time with includes |
| 04:49.37 | drmessano | ~book |
| 04:49.38 | 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 |
| 04:49.48 | drmessano | jameswf-home knows what I am talking about |
| 04:49.55 | Frogzoo | dudes: thanks - what's the default file? |
| 04:49.58 | drmessano | Molson Canadian is good stuff |
| 04:50.04 | dudes | extensions.conf |
| 04:50.18 | dudes | I prefer Killians myself |
| 04:50.28 | drmessano | Killians is good |
| 04:50.37 | Frogzoo | dudes: thanks - calling it dialplan.conf might be better, but all clear, thanks |
| 04:50.42 | dudes | I love red beer provided it's brewed right |
| 04:51.04 | dudes | I don't develop it so talk to the devs hehe |
| 04:51.13 | *** join/#asterisk [DS]LynxW (n=jzawacki@pool-71-191-163-40.washdc.fios.verizon.net) |
| 04:51.57 | drmessano | Someone laughed and argued with me a few weeks ago, but 2008 is clearly the year of the "Call Me" button |
| 04:52.04 | [DS]LynxW | Hello, I need help getting rid of a squelch issue.. |
| 04:52.15 | dudes | Call me? |
| 04:52.17 | drmessano | Turn the RF Gain off |
| 04:52.22 | drmessano | Web "Call Me" buttons |
| 04:52.28 | [DS]LynxW | TE220 with PRI between Telco and Nortel MICS |
| 04:52.32 | dudes | oh that bullshit |
| 04:52.39 | dudes | err, pardon me french |
| 04:52.42 | drmessano | lol |
| 04:52.53 | [DS]LynxW | tx and rxgain = 0 |
| 04:53.04 | dudes | I've made about 2k the last year from the web call stuff |
| 04:53.05 | [DS]LynxW | and it appears to be CPU or NIC related.. not quite sure yet. |
| 04:53.09 | dudes | so whatever you knw |
| 04:53.19 | Frogzoo | drmessano: 2009? |
| 04:53.30 | drmessano | My prediction for 2009? |
| 04:53.40 | [DS]LynxW | but I was getting messages about "Losing some ticks... checking if CPU frequency changed." |
| 04:53.54 | [DS]LynxW | but google tells me it is an SMP problem. .and I'm not running an SMP kernel. |
| 04:53.57 | Frogzoo | drmessano: no, I think it will take til then |
| 04:54.04 | drmessano | Nope |
| 04:54.08 | drmessano | It's already taking |
| 04:54.11 | drmessano | Look around |
| 04:54.12 | JT | [DS]LynxW: what's zttest like? |
| 04:54.18 | [DS]LynxW | but, the test to reproduce the issue is 'ls -R /' and see what happens.. |
| 04:54.35 | dudes | have you checked for a interrupt issue? |
| 04:54.38 | [DS]LynxW | 99.994240% |
| 04:54.40 | dudes | or are you running x? |
| 04:54.42 | [DS]LynxW | ish |
| 04:54.48 | [DS]LynxW | Nope. |
| 04:55.15 | [DS]LynxW | not that I'm proud.. but it's on a celeron 3.0Ghz.. 512MB RAM.. but 313MB is free.. |
| 04:55.16 | dudes | you are getting "squel" in your phones |
| 04:55.24 | [DS]LynxW | Not really.. |
| 04:55.30 | dudes | and you ask questions hehe |
| 04:55.36 | [DS]LynxW | it's kinda like a digital squelch.. |
| 04:55.40 | dudes | sorry I couldn't help myself |
| 04:55.50 | dudes | are you transcoding? |
| 04:56.07 | dudes | if you are from ulaw to g729 .... |
| 04:56.08 | [DS]LynxW | Not that I know of, but how can I tell to make sure? |
| 04:56.26 | dudes | show "your tech " channels |
| 04:56.29 | [DS]LynxW | Well, right now I have my cell phone dialed into a meetme.. and it's sitting on hold. |
| 04:56.41 | [DS]LynxW | every once in a while I hear it. |
| 04:56.43 | jameswf-home | drmessano: you going to SC |
| 04:56.54 | dudes | so you're thinking interference |
| 04:57.00 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 04:57.04 | dudes | cat /proc/interrupts |
| 04:57.18 | drmessano | jameswf-home: I wish |
| 04:57.26 | dudes | if you have a NIC shared with another device or a hardware borad shared |
| 04:57.35 | dudes | change it up |
| 04:57.51 | [DS]LynxW | Dang.. what's the nopaste place again? |
| 04:57.55 | drmessano | My current employer can't spell VoIP, and I can't out-of-pocket it right now |
| 04:58.11 | jameswf-home | cant sell voip? |
| 04:58.17 | drmessano | spell |
| 04:58.29 | jameswf-home | oh lol |
| 04:58.39 | [DS]LynxW | http://pastebin.com/d7203cbc4 |
| 04:58.55 | [DS]LynxW | Oops. |
| 04:59.13 | jameswf-home | have you played with the joe roper thing |
| 04:59.16 | [DS]LynxW | http://pastebin.com/d7103cbc4 |
| 04:59.20 | drmessano | Another reason i've been searching for a new job lol |
| 04:59.31 | dudes | what do you do now |
| 04:59.46 | jameswf-home | were hiring for tech support :) |
| 05:00.02 | drmessano | Broadcast (Radio) IT and Engineering |
| 05:00.06 | drmessano | 7 stations |
| 05:00.14 | drmessano | 7x the headaches |
| 05:00.18 | jameswf-home | clearchannel? |
| 05:00.25 | drmessano | yeah lol |
| 05:00.35 | *** join/#asterisk LakeSolon (n=blake@12-202-198-20.client.mchsi.com) |
| 05:00.41 | [DS]LynxW | dudes: I really think it could be NIC related.. |
| 05:00.46 | jameswf-home | clearchannel was a 4 letter word in washington |
| 05:00.51 | [DS]LynxW | I'll have to check to see what slots are available. |
| 05:00.55 | dudes | is there a issue with your NIC |
| 05:01.04 | dudes | if it is, fix the issue, or replace it |
| 05:01.05 | [DS]LynxW | Not that I know of really. |
| 05:01.17 | [DS]LynxW | and it's 100 FD |
| 05:01.18 | dudes | did you cat /proc/interrupts ? |
| 05:01.23 | [DS]LynxW | yes. |
| 05:01.25 | *** join/#asterisk AJayMN (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 05:01.29 | drmessano | jameswf-home: I promise not to make you say things to get you in trouble with your boss, if you agree to do the same |
| 05:01.29 | drmessano | lol |
| 05:01.35 | [DS]LynxW | (11:59:15 PM) [DS]LynxW: http://pastebin.com/d7103cbc4 |
| 05:01.37 | dudes | and is there a shared resource |
| 05:02.09 | Robba | you there JT? |
| 05:02.12 | [DS]LynxW | Hell, at this point.. I'll pull the NIC out just for testing. |
| 05:02.58 | dudes | the question is --- is it sharing |
| 05:03.04 | scooby2 | anyone willing to share a working 1.4 ivr example w/ Directory |
| 05:03.05 | scooby2 | ? |
| 05:03.10 | jameswf-home | drmessano: your like right next door to sc... |
| 05:03.13 | drmessano | I've had a lot of good experience, but i've far outgrown what I am doing all day.. I may as well be working at McDonalds when it comes to what I know vs. what I get to apply all day |
| 05:03.25 | dudes | it looks like it is with your USB maybe |
| 05:03.32 | drmessano | 5 hour drive to CHS |
| 05:03.36 | [DS]LynxW | 10: Â Â Â 17222 Â Â Â Â Â XT-PIC Â uhci_hcd, uhci_hcd, eth0 |
| 05:03.44 | dudes | try disabling that in your bios |
| 05:03.57 | Robba | mosty you there? |
| 05:04.00 | LakeSolon | Is there a way to make Asterisk use one interface for one trunk, and the other interface for another? |
| 05:04.04 | [DS]LynxW | Can do.. I'll be back in a little bit.. thanks.. |
| 05:04.12 | jameswf-home | I should encourage my goss to pony up for a car and gas money to tour the region |
| 05:04.21 | mosty | Robba, for about the next 2 minutes |
| 05:04.26 | Robba | lol ok |
| 05:04.33 | Robba | quick question for you |
| 05:04.36 | jameswf-home | *boss |
| 05:04.51 | Robba | exten => 101,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) |
| 05:04.53 | dudes | <drmessano> - you do any coding? |
| 05:05.09 | Robba | responds back with extension not found |
| 05:05.13 | drmessano | I dabble.. Nothing to write home about |
| 05:05.34 | dudes | no specialties in general then |
| 05:05.54 | drmessano | Not coding-wise |
| 05:06.11 | Robba | any ideas? |
| 05:06.29 | jameswf-home | i wrote home and drmessanoyour famous |
| 05:06.34 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 05:06.34 | drmessano | lol |
| 05:06.38 | dudes | what are you making how? |
| 05:06.40 | dudes | err now? |
| 05:06.54 | pc600 | Scenario problem: I have an office in another state (USA) with 5 people. I would like them to use our private WAN going to this location for VoIP. How do I get a DID to them? I have a PRI in another state, but I Can't get out-of-state (or lata) DIDs from the telco. |
| 05:07.11 | dudes | you do any marketing, sales, type work? |
| 05:08.06 | VitoCorleon | I have a Cisco 7960 setup to a Asterisk box. Incoming works great but when i dial out i get "Reorder", any help please? |
| 05:08.14 | pc600 | jameswf-home - There's probably more money in it :) |
| 05:08.27 | dudes | perhaps pc600 |
| 05:08.28 | drmessano | I could probably handle sales.. |
| 05:08.57 | jameswf-home | I work for a manufacturer as long as there are companies out there using asterisk I should have a job :) |
| 05:09.17 | drmessano | I've spent years learning to win arguments with salespeople |
| 05:09.19 | dudes | what are they making |
| 05:09.25 | drmessano | I could probably outsell some of them too |
| 05:09.32 | drmessano | lol |
| 05:09.46 | jameswf-home | Dude I get to go out and talk about open source software and computers to people its not realy sales its a hobby I get money to do |
| 05:09.49 | dudes | I was good as a teen telemarketing |
| 05:10.07 | dudes | but I'm f'n awesome selling my shizat |
| 05:10.21 | VitoCorleon | anyone here taht can help me? |
| 05:10.28 | jameswf-home | ~ask |
| 05:10.29 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 05:10.29 | drmessano | In our case, I know more about the product than they do.. I know the signals, I get waaay to involved in the programming side, so I know what we have on the air.. Most of them are clueless |
| 05:11.06 | VitoCorleon | jameswf-home, my question was clear :) |
| 05:11.16 | [TK]D-Fender | VitoCorleon, Pastebin the CLI output with SIP debug enabled for you call attempt. |
| 05:11.18 | [TK]D-Fender | ~pb |
| 05:11.19 | 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 |
| 05:11.24 | [DS]LynxW | back.. |
| 05:11.24 | [TK]D-Fender | your* |
| 05:11.26 | dudes | <drmessano> - I'm just thinking, but a friend in Ireland and I are trying to get things booking with what we do ... |
| 05:11.34 | Frogzoo | VitoCorleon: apparently not |
| 05:11.41 | [DS]LynxW | :/ |
| 05:11.51 | dudes | and we will have a open spot --- that's why I'm wondering what you're making now. |
| 05:11.58 | [DS]LynxW | eth0 is no longer shared.. but I'm still getting it. |
| 05:12.01 | jameswf-home | no VitoCorleon no one can help you the room only has 1 doctor which is drmessano but he is an OB/GYN so |
| 05:12.25 | drmessano | No offense, but I shy away from random job offers from startups on IRC.. lol |
| 05:12.34 | dudes | [DS]LynxW - kill process running that aren't required |
| 05:12.45 | drmessano | Yes, is there crowning? Can you see the feet? |
| 05:12.48 | dudes | run 'top' too see your CPU killer |
| 05:12.58 | drmessano | How dialated is she? |
| 05:13.08 | dudes | <drmessano> - no problem there |
| 05:13.10 | drmessano | Have you any sheets or towels |
| 05:13.19 | [DS]LynxW | I have top running.. |
| 05:13.24 | dudes | but if you're interested in the least you can reach in #gnudialer |
| 05:13.27 | [DS]LynxW | nothing is really a CPU killer.. that's what gets me. |
| 05:13.32 | Robba | Ok in my extensions.conf file i have exten => 101,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) in the [services] context. [services] are included, when dialling 101 i can't get to voice mail it just responds with extension 101 not found, does anyone have any clue as to what could be causing this? |
| 05:13.42 | [DS]LynxW | average: 0.02, 0.13, 0.08 |
| 05:13.50 | [DS]LynxW | and the .13 was starting back up. |
| 05:14.15 | dudes | I'm always on and always busy --- but even if not a full time gig you can make some extra cash |
| 05:14.18 | [DS]LynxW | I see kjournald pop to the top once in a while.. |
| 05:14.51 | [DS]LynxW | yeah.. |
| 05:16.07 | jameswf-home | if we did a fork called xobxirt do you think they would get steamed |
| 05:16.24 | drmessano | lol |
| 05:16.28 | drmessano | thats too good |
| 05:16.41 | drmessano | Gotta have a slogan to get EVERYONE on board |
| 05:16.42 | drmessano | Like |
| 05:16.51 | drmessano | xobxirt: "I'd hit it" |
| 05:16.51 | dudes | <drmessano> - just an offer, don't take it serious, but I often times could use help with support and I pay well |
| 05:16.55 | jameswf-home | pronounced zobzirt |
| 05:17.14 | drmessano | dudes, thanks.. something to consider :) |
| 05:17.31 | [DS]LynxW | jameswf-home: if the repositories worked.. it'd be golden. :) |
| 05:17.36 | drmessano | jameswf-home, we can make viral videos of geeks proclaiming "I'd hit it" |
| 05:17.58 | drmessano | and then have them wearing xobxirt t-shirts |
| 05:18.17 | drmessano | It's just dumb enough.. |
| 05:18.27 | Robba | Ok in my extensions.conf file i have exten => 101,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) in the [services] context. [services] are included, when dialling 101 i can't get to voice mail it just responds with extension 101 not found, does anyone have any clue as to what could be causing this? |
| 05:18.28 | [TK]D-Fender | Robba, pastebin is your friend. |
| 05:18.35 | jameswf-home | we should make xobxirt shirts anyway and sell em on cafe press |
| 05:18.41 | [TK]D-Fender | Robba, And stop spamming the same question over and over again. |
| 05:18.48 | [DS]LynxW | Well... damn.. This is a test system anyway.. I guess it's time to purchase the production system and install from scratch. |
| 05:18.59 | Robba | well the bot said to ask the same question |
| 05:19.07 | Robba | i made it as concise as possible |
| 05:19.13 | [TK]D-Fender | Robba, we heard it the first time, so if we knew the answer, or felt like answering your question based on the way you asked it you'd have heard from us. |
| 05:19.48 | [DS]LynxW | Robba: I'm hear for help as well, but what is in your voicemail.conf? |
| 05:19.58 | jameswf-home | I am going to ask tony's graphic designer to make a logo |
| 05:20.03 | drmessano | xobxirt CE "clean edition" and we can do one with 70% of the GUI full of adsense and call it SE "Sellout Edition" |
| 05:20.11 | [TK]D-Fender | [DS]LynxW : wrong approach.... |
| 05:20.21 | [DS]LynxW | [TK]D-Fender: my bad.. |
| 05:20.27 | [DS]LynxW | I'll stick to my problem then. |
| 05:21.11 | [TK]D-Fender | [DS]LynxW, s'ok.... start at the point of origin of the problem which is to say "look at what is happening", not "look at some config thinging the problem is there without examining the evidence as it happens. |
| 05:21.38 | drmessano | [TK]D-Fender: Fine, insult how the rest of us troubleshoot ;) |
| 05:21.45 | [DS]LynxW | ah.. |
| 05:21.54 | [DS]LynxW | so maybe tail /var/log/asterisk/full first? |
| 05:22.16 | [TK]D-Fender | [DS]LynxW, CLI + live debug. Debug log files = waste of time. |
| 05:22.19 | [DS]LynxW | or set verbose 3 to 'see' what asterisk is doing? |
| 05:22.30 | [DS]LynxW | well, you have tail -f as well. |
| 05:22.31 | [TK]D-Fender | [DS]LynxW, max out CLI. |
| 05:22.38 | [DS]LynxW | and I think the logs have more info, don't they? |
| 05:22.47 | [DS]LynxW | ah.. higher verbose? |
| 05:23.00 | [TK]D-Fender | [DS]LynxW, I like 10 personally. It feels substantial. |
| 05:23.03 | dudes | the best debug is some bloody printf's in asterisk code |
| 05:23.24 | drmessano | Set verbose at 10 and watch your problem fix itself in front of your eyes |
| 05:23.27 | [TK]D-Fender | dudes, valuable if you're trying to track a problem at the source level. |
| 05:23.38 | dudes | isn't the trend eh |
| 05:23.50 | Robba | http://rafb.net/p/xnr7b686.html |
| 05:23.55 | dudes | in my case anyway |
| 05:24.04 | [TK]D-Fender | drmessano, or perhaps at least announce itself blatantly which is the case in the vast majority of cases. |
| 05:24.09 | jameswf-home | core set verbose 999999999999999999999 |
| 05:24.20 | drmessano | Well, thats sorta what I meant :) |
| 05:24.22 | [DS]LynxW | Wouldn't it be 'set verbose 666'? |
| 05:24.27 | dudes | that doesn't do crap except waste time |
| 05:24.33 | drmessano | The feeling of "Oh shit.. there it is" |
| 05:24.38 | [TK]D-Fender | Robba, 45 ......Polo |
| 05:24.56 | dudes | does it really matter once you set debut over 100 ? |
| 05:25.14 | [TK]D-Fender | Robba, 61, *ICK* Deprecated |
| 05:25.16 | [DS]LynxW | Heh.. |
| 05:25.28 | Robba | lol whoops... |
| 05:25.44 | [TK]D-Fender | Robba, 61-63 : _s is not a pattern. That should simply be "s" |
| 05:26.00 | [DS]LynxW | Wow.. so.. since I set verbose to 10.. I haven't heard the squelch.. good fix.. ;) |
| 05:26.07 | dudes | f'n bison heh |
| 05:26.08 | [TK]D-Fender | Robba, 70 again deprecated. |
| 05:26.57 | [TK]D-Fender | Robba, And the "mystery factor" for 70 is "where are all of these variables, some of which dangerously named actually being set?" |
| 05:28.15 | dudes | curious, [TK]D-Fender, what do you do? |
| 05:28.42 | dudes | I've seen you around for years, I'm simply curious |
| 05:28.55 | jameswf-home | dudes: mostly women I imagine |
| 05:29.13 | dudes | perhaps --- but women do get on IRC so |
| 05:29.31 | *** part/#asterisk Swabby (n=e741533@12.46.189.1) |
| 05:29.42 | jameswf-home | ~[TK]D-Fender |
| 05:29.42 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 05:30.06 | [TK]D-Fender | dudes, full-time head of IT for non tech company, freelance */general IT consultant |
| 05:30.25 | drmessano | I'm sure there's a lot of women on IRC using male nicks |
| 05:30.28 | dudes | according to jbot |
| 05:30.32 | [TK]D-Fender | dudes, as far as tech goes. personal stuff is a huge list |
| 05:30.32 | dudes | hehe |
| 05:30.41 | [TK]D-Fender | dudes, .... |
| 05:30.43 | [TK]D-Fender | ~jbot |
| 05:30.43 | jbot | rumour has it, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 05:30.45 | [TK]D-Fender | ^^ |
| 05:30.53 | jameswf-home | ~dudes |
| 05:30.55 | jameswf-home | ~dude |
| 05:30.56 | jbot | Be most excellent to each other! |
| 05:30.59 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 05:31.00 | drmessano | ~drmessano |
| 05:31.01 | jbot | i guess drmessano is the leading cause of censorship in #asterisk |
| 05:31.02 | [TK]D-Fender | WOAH! |
| 05:31.07 | dudes | hehe |
| 05:31.17 | dudes | that's funny |
| 05:31.18 | [TK]D-Fender | Party tiem! |
| 05:31.34 | [DS]LynxW | Thanks for all your help.. time to go home. I'll see if installing the TE220 on a production box AKA, not celeron with only 512MB RAM helps my situation. if Not.. I'm sure I'll be back. |
| 05:31.36 | [DS]LynxW | Thanks again. |
| 05:31.52 | dudes | why 512 |
| 05:31.55 | dudes | ouch |
| 05:32.05 | [DS]LynxW | Well, it was a test system. |
| 05:32.08 | drmessano | As much as I hated Bill and Ted 2, best scene: Grim Reaper "You sank my battleship" |
| 05:32.19 | dudes | but a gig is cheap |
| 05:32.20 | [DS]LynxW | that I kept growing.. and then this started happening.. |
| 05:32.28 | [DS]LynxW | It was what was laying around at the time. |
| 05:32.36 | [DS]LynxW | and it has 300MB free.. |
| 05:32.43 | dudes | makes sense then --- bugt |
| 05:32.56 | [DS]LynxW | But, switching to a gigabit NIC and real server will probably help. |
| 05:33.03 | dudes | but ram isn't your issue |
| 05:33.07 | [DS]LynxW | No.. |
| 05:33.11 | jameswf-home | jbot: What's mine say |
| 05:33.12 | jbot | dude! ... What's Mine Say? |
| 05:33.15 | [DS]LynxW | I think it's processing.. but I'm still not sure. |
| 05:33.17 | jameswf-home | sweet |
| 05:33.27 | dudes | a gigabit isn't going to fix your issue |
| 05:33.33 | dudes | it's more a chipset issue |
| 05:33.42 | drmessano | ~leeroy jenkins |
| 05:33.43 | jbot | extra, extra, read all about it, leeroy jenkins is a hero to us all. |
| 05:33.48 | drmessano | HA!! |
| 05:33.49 | [DS]LynxW | Yeah. the box I'm using isn't ideal. |
| 05:34.04 | dudes | as it seems |
| 05:34.07 | [DS]LynxW | the onboard NIC isn't even supported under Linux yet. |
| 05:34.25 | Robba | thanks [TK]D-Fender |
| 05:34.28 | [DS]LynxW | so I had to toss in a different one. again.. from parts laying around. |
| 05:34.35 | [DS]LynxW | I'm going to order a nice Dell server for it.. |
| 05:34.41 | Robba | seems to be ok now |
| 05:34.53 | [DS]LynxW | now that the project has been proven.. and this problem has just crept up.. as we kept adding SIP phones.. |
| 05:35.00 | scooby2 | good ole centos 5.1/asterisk 1.4.17 crashing about something smp on a single cpu server |
| 05:35.01 | drmessano | Realtek 8139 FTW |
| 05:35.14 | [TK]D-Fender | Robba, Now see I had to decode that from your dialplan code itself. I could have pinpointed a specific issueif I had seen the actual CALL like I asked. |
| 05:35.38 | [TK]D-Fender | Robba, So I spouted of the top 10 things I saw wrong and hopefully caught your issue in the process. |
| 05:35.53 | *** part/#asterisk UnixDog (n=unixdog@adsl-69-230-170-165.dsl.irvnca.pacbell.net) |
| 05:36.04 | [DS]LynxW | Goodnight guys. |
| 05:36.07 | drmessano | #1 rule of asking for help is knowing how to ask for help |
| 05:36.33 | dudes | or a decent question that appears well informed |
| 05:36.39 | [TK]D-Fender | #2 rule of asking for help is showing the right stuff for us to be able to help you |
| 05:36.44 | drmessano | Yes |
| 05:36.51 | Robba | on verbose 13 it jsut said the extension could not be found |
| 05:37.10 | dudes | which would probably entitle the obvious |
| 05:37.16 | [TK]D-Fender | #3 rule of asking for help, is being proactive enough not to make us beat #1 & 2 out of you over a drawn out process |
| 05:37.17 | drmessano | I was gonna say "If you don't know what the problem is, you wouldnt be asking for help, so you can't effectively tell me/us what we do or do not need to see" |
| 05:37.44 | jameswf-home | ~1 |
| 05:37.45 | jbot | 1 is a number, silly |
| 05:37.47 | [TK]D-Fender | Robba, You are concentrating on the ERROR message when I'm sure the line its trying to EXECUTE would have made it obvious <- |
| 05:37.52 | jameswf-home | ~#1 |
| 05:37.53 | jbot | #1 is probably more like it |
| 05:37.55 | drmessano | "My car is broke, I don't know what the problem is, but it's not the fuel injectors" |
| 05:38.01 | drmessano | FAIL |
| 05:38.13 | Robba | ok sorry guys |
| 05:38.14 | jameswf-home | now what |
| 05:38.14 | dudes | perhaps your fuel rail is jacked |
| 05:38.29 | dudes | or your PCM is jacked up |
| 05:38.37 | drmessano | or you're out of gas? |
| 05:38.51 | dudes | then your fuel pump is sucking air |
| 05:38.55 | jameswf-home | My tires are flat could it be cause I just changed my oil |
| 05:39.02 | drmessano | HA |
| 05:39.05 | dudes | yes for sure hehe |
| 05:39.18 | drmessano | I had a user tell me her monitor was going out on her |
| 05:39.27 | Mavvie | heh... I can't use AGI because it will hangup when the call is dropped, and I can't use DeadAGI because the call isn't dropped yet. |
| 05:39.39 | drmessano | I asked her if she was SURE it wasnt moved and to make sure the power cord was in the back firmly |
| 05:39.40 | dudes | I had a guy ask why his keyboard wasn't working --- after he moved his computer |
| 05:39.49 | dudes | perhaps --- it wasn't unplugged? |
| 05:39.52 | Mavvie | And when I hangup the call it is going to the h extensions which I don't want to use for this purpose. |
| 05:39.54 | dudes | I don't know... |
| 05:40.13 | drmessano | I get over there today.. and her monitor was 7 or 8 inches to the left of center, and she tells me "I figured it out.. it was the power cord" |
| 05:40.42 | dudes | haha |
| 05:40.48 | drmessano | She moved it over, pulled the cord out just enough.. duh |
| 05:40.57 | drmessano | Like I haven't dealt with morons for 10 years |
| 05:41.06 | dudes | Shock and awe --- that name annoyed me during the beginning Iraq campaign |
| 05:41.11 | drmessano | I know how you're going to fuck up before you even do it |
| 05:41.25 | dudes | at least we could have killed Saddam given the name --- and that's what it was for |
| 05:42.03 | drmessano | Obama will save us all, don't worry |
| 05:42.10 | jameswf-home | I need to change the hologen fluid in my car... and switch the winter air to summer air |
| 05:42.15 | drmessano | What was it a heard today.. |
| 05:43.13 | drmessano | Oh, someone told me "You know how to get Georgians to not vote for Obama.. Throw a G in front of his name, so it's "GoBama".. they hate Alabama" |
| 05:43.31 | drmessano | Corny but good |
| 05:44.15 | dudes | I kind of like Ron Paul |
| 05:44.23 | drmessano | Jesus |
| 05:44.38 | dudes | because, Ron Paul's got 99 problems but a bitch ain't one |
| 05:44.42 | drmessano | HA |
| 05:44.50 | drmessano | Damnit, what was my line |
| 05:45.07 | dudes | I actually don't care for any of them but |
| 05:45.09 | jameswf-home | put a pull an r add a g and e in ron paul is gone paul |
| 05:45.17 | drmessano | PAUL IS DEAD |
| 05:45.21 | dudes | But I'd probably vote for McCain if I do |
| 05:45.29 | scooby2 | paul might run libertarian |
| 05:45.40 | dudes | good for him |
| 05:45.48 | drmessano | McCain is cool.. If you're into 90 yr old men |
| 05:45.58 | [TK]D-Fender | RP is not spending the money he's collected to try to win. |
| 05:46.00 | dudes | he's 76 I think |
| 05:46.08 | jameswf-home | ~ron paul |
| 05:46.09 | jbot | ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
| 05:46.10 | scooby2 | someday we may have a real third party |
| 05:46.13 | drmessano | HA |
| 05:46.16 | drmessano | There it is! |
| 05:46.28 | drmessano | RONPAULAPPLEUBUNTU FTW |
| 05:46.35 | dudes | everyone cried when Bob ran against Clinton |
| 05:46.38 | dudes | he's alive |
| 05:46.44 | dudes | and he's a damn good person |
| 05:46.45 | [TK]D-Fender | scooby2, GWB already talks in the thrid-person, what are you talking about?!?! |
| 05:47.21 | drmessano | If Hillary doesn't win the nomination, she can run as a Lesbitarian |
| 05:47.44 | dudes | I wish Dole wouldn |
| 05:47.48 | jameswf-home | ~hillary |
| 05:47.49 | jbot | it has been said that hillary is see pole hugger |
| 05:47.53 | dudes | would've won in 96' |
| 05:48.03 | jameswf-home | obama |
| 05:48.26 | dudes | I can't believe they are pushing a 3.1 trillion dollar budget |
| 05:48.34 | dudes | that pisses me off |
| 05:48.46 | jameswf-home | ~obama is <reply> yeah OBama is black like Michael Jackson is White |
| 05:48.46 | jbot | ...but obama is already something else... |
| 05:48.53 | [TK]D-Fender | dudes, That shouldn't be the problem... its the DEFICIT it comes with thats the problem. |
| 05:49.02 | jameswf-home | ~obama |
| 05:49.03 | jbot | hmm... obama is a pimp |
| 05:49.04 | lunaphyte | pole hugger? |
| 05:49.04 | dudes | that's the end result isn't it |
| 05:49.15 | jameswf-home | ~no obama is <reply> yeah OBama is black like Michael Jackson is White |
| 05:49.15 | jbot | okay, jameswf-home |
| 05:49.17 | drmessano | I actually think Hillary will be a good president.. Obama isn't ready to handle the tabloid media, and Hillary has been in the spotlight for years.. |
| 05:49.26 | [TK]D-Fender | dudes, in this case yes, but say the part you really care about. |
| 05:49.31 | lunaphyte | oh please god no. |
| 05:49.40 | dudes | I care about where this Country is |
| 05:49.41 | [TK]D-Fender | dudes, See people complain about stuff in the wrong way. |
| 05:49.50 | drmessano | Obama will spend his first 3 years answering for dead bodies and other assorted skeletons.. |
| 05:50.00 | dudes | and it's the "I approve" all the time for this budget |
| 05:50.01 | [TK]D-Fender | dudes, like for instance people say "I pay too much in taxes". |
| 05:50.06 | dudes | and it's been like that sicne 2002 |
| 05:50.14 | [TK]D-Fender | dudes, See My problem is that I don't pay ENOUGH taxes. |
| 05:50.24 | dudes | what do you pay? |
| 05:50.40 | [TK]D-Fender | dudes, I need to owe $100,000K / month in taxes, but I don't. |
| 05:50.50 | dudes | why is that |
| 05:51.06 | [TK]D-Fender | dudes, Because what you have to ask yourself is how much you have to EARN to owe that much :) |
| 05:51.14 | drmessano | It's not HOW MUCH we spend, it's how we spend it.. I'd trust the democrats with 3.1 trillion dollars before a republican |
| 05:51.15 | [TK]D-Fender | dudes, You must learn PERSPECTIVE child! |
| 05:51.42 | dudes | if you had to pay 100k/mth you're making a lot |
| 05:51.50 | [TK]D-Fender | dudes, See my problem? :) |
| 05:52.04 | dudes | no |
| 05:52.08 | drmessano | I want to pay 100k a month in taxes too |
| 05:52.17 | [TK]D-Fender | dudes, I'm not making millions... that's my problem. |
| 05:52.36 | dudes | if you're paying 100k/mth you're not makign millions a month |
| 05:52.45 | drmessano | LOL |
| 05:52.46 | drmessano | yes, you are |
| 05:52.52 | dudes | no you are not |
| 05:52.58 | drmessano | You fail at fractions |
| 05:52.59 | scooby2 | no one pays $100k a month in taxes |
| 05:53.12 | [TK]D-Fender | dudes, I also never attached a unit of measure in terms of currency or time, try not to assume one for me :) |
| 05:53.14 | [TK]D-Fender | ~assume |
| 05:53.15 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 05:53.15 | dudes | I do not fail at fractions |
| 05:53.21 | scooby2 | little things called tax shelters |
| 05:53.28 | drmessano | If I make $1000 a month and pay $100 in taxes |
| 05:53.31 | dudes | then you're being a ass |
| 05:53.43 | dudes | you're paying a tid bit |
| 05:53.44 | drmessano | I would pay $100,000 in taxes on $1 million a month |
| 05:53.52 | drmessano | I want to pay $100,000 in taxes |
| 05:54.00 | dudes | if you paid 10% |
| 05:54.08 | dudes | but who pays 10% |
| 05:54.12 | drmessano | You're still missing it |
| 05:54.25 | dudes | let's stick with what is real |
| 05:54.35 | jameswf-home | if you make 1,000,000 a month you have shelers so you pay like $6 bucks in taxes |
| 05:54.37 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:54.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 05:54.50 | jameswf-home | *shelters |
| 05:54.54 | scooby2 | jameswf-home: bingo |
| 05:54.54 | drmessano | [TK]D-Fender was trying to say he would like to be able to make enough money that he would be paying $100,000 in taxes, scaled up proportionally |
| 05:54.59 | dudes | money bags |
| 05:55.01 | drmessano | Do you get it now? |
| 05:55.23 | jameswf-home | If you re paying 100,000 in taes fire your accountant |
| 05:55.31 | drmessano | lol |
| 05:55.33 | jameswf-home | damn i cant type |
| 05:55.46 | dudes | after you go over 120k you get some breaks |
| 05:55.56 | Robba | Thanks everyone for your help today. |
| 05:56.07 | dudes | so I hear anyway |
| 05:56.27 | dudes | personally, taxes should be fair vs what we have now. |
| 05:56.33 | jameswf-home | I have to tell my boss to pay me over 120k o i can see a tax break |
| 05:56.50 | dudes | cause SS is such a drain |
| 05:57.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:57.12 | scooby2 | Warren Buffett keeps complaining that he pays less in tax then his administrative assistant does due to tax breaks and tax shelters |
| 05:57.32 | dudes | cause the tax system is bull |
| 05:57.33 | jameswf-home | ssi should be eliminated as far as a retirement income source |
| 05:57.37 | dudes | I wont argue |
| 05:57.56 | dudes | family should take care of their elders |
| 05:58.04 | scooby2 | let us poor saps paying in put some part in our own fund |
| 05:58.07 | dudes | period, come on, take some responsibility |
| 05:58.36 | dudes | they should axe the fed tax and leave it up to the state community as the consititution says |
| 05:58.54 | dudes | the fed is there to protect the foreign interest of the union period |
| 05:58.56 | drmessano | So you wouldn't mind paying out $35,000 a year to support your elderly parents? |
| 05:59.17 | dudes | this bullshit that's been since FDR crippled ass is crap |
| 05:59.30 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 05:59.36 | dudes | if it's not out of tax dollars and that's what it took |
| 05:59.45 | dudes | I'd get a second job |
| 05:59.48 | Robba | Sorry guys i have another one |
| 06:00.30 | dudes | the point people need to take responsibility and quit thinking the government is going to fix it |
| 06:00.43 | dudes | socialism is communism and both are bad |
| 06:01.39 | [TK]D-Fender | no, socialism and communism are not bad. They are alternative system that have different inherent weaknesses |
| 06:01.44 | Robba | http://rafb.net/p/7tcbnG94.html i hope this is what you require |
| 06:02.21 | [TK]D-Fender | Robba, And I told you that you were using something DEPRECATED on that line. 1 guess what it was... |
| 06:02.46 | Robba | voicemail? |
| 06:02.49 | Robba | i have no idea |
| 06:02.54 | Robba | i am a n00b |
| 06:02.57 | Robba | i admit that |
| 06:03.03 | [TK]D-Fender | Robba, look at the line, whats MISSING? |
| 06:03.08 | jameswf-home | I depricated on the side of the road once after clearing a 5th of vodka |
| 06:03.12 | dudes | they aren't bad if people were not selfish |
| 06:03.16 | jameswf-home | ~newb |
| 06:03.17 | jbot | Don't bother telling us you're a "newb" or a "n00b". We can tell. |
| 06:03.22 | [TK]D-Fender | Robba, I don't care about newb. "newb" does not mean "blind as a bat" |
| 06:03.26 | dudes | but people are so insentive is key |
| 06:03.37 | dudes | but I would't expect a tool to understand |
| 06:03.51 | Robba | the username? |
| 06:04.12 | [TK]D-Fender | Robba, Stare at the output of your dialplan as its executed, and then stare at your dialplan code in extensions.conf. What are you THINKGING you should see that you clearly are not getting? |
| 06:04.25 | dudes | err, incentive |
| 06:04.26 | Robba | i am new to this, i am not sure whats supposed to be there |
| 06:04.42 | [TK]D-Fender | Robba, what is your line in extensions.conf trying to do? |
| 06:05.02 | Robba | connect to voice mail? |
| 06:05.38 | dudes | [TK]D-Fender - I don't know what world you live in, but the one I live in, people dont deserve shit |
| 06:05.39 | [TK]D-Fender | Robba, keep going... you are referencing a bunch of stuff there to a very specific end... |
| 06:06.04 | dudes | aside a swift kick in the ass cause they are cunts |
| 06:06.13 | Daviey | O_o |
| 06:06.28 | xcompass | hi, you guys know fwdOut? is it dead? |
| 06:06.35 | [TK]D-Fender | dudes, What your world lacks is a social support system that actually gets the PEOPLE to support it. |
| 06:06.52 | Daviey | communism ftw |
| 06:07.02 | Daviey | or soclialism, meh |
| 06:07.10 | dudes | I have a well rounded support system thank you |
| 06:07.23 | dudes | I have a wealth of friends and friend I enjoy the company of |
| 06:07.29 | Daviey | dudes: your sports bra? |
| 06:07.36 | [TK]D-Fender | lol |
| 06:07.42 | Robba | ok sorry i just don't get it |
| 06:07.43 | [TK]D-Fender | Daviey++ |
| 06:07.51 | dudes | I wear a sports bra cause I'm a lazy fat fuck |
| 06:07.54 | dudes | year |
| 06:07.55 | Daviey | oh dear, i'll get told off for that :( |
| 06:08.43 | [TK]D-Fender | Robba, please describe in detail exactly how and why that dialplan is being called, and why you are trying to call it the way you are. |
| 06:08.49 | Daviey | Robba: http://www.asteriskguru.com/tutorials/voicemailmain.html or voip-wiki |
| 06:08.56 | Daviey | Can you see what you are missing? |
| 06:08.57 | *** join/#asterisk neonerz (i=18bb0206@gateway/web/ajax/mibbit.com/x-3e7fc19ccf0452d4) |
| 06:08.58 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 06:09.04 | [TK]D-Fender | Daviey, shhh |
| 06:09.18 | Daviey | sorry |
| 06:09.18 | [TK]D-Fender | Daviey, Let him look at what he's doing. |
| 06:09.21 | dudes | <Daviey> - perhaps you people helping fucks should tell him eh! |
| 06:09.23 | dudes | cunts |
| 06:09.33 | Daviey | dudes: why are you here? |
| 06:09.40 | dudes | to be a dick |
| 06:10.02 | dudes | you limey fuck |
| 06:10.08 | [TK]D-Fender | comic relief? Warning to others? Accident of birth? "random string theory remark here"? |
| 06:10.21 | [TK]D-Fender | Calm down people.... |
| 06:10.28 | jameswf-home | ~random |
| 06:10.29 | dudes | yea --- accident of birth eh |
| 06:10.42 | dudes | I wasn't an accident unlike your self from your whore mum |
| 06:10.50 | Daviey | oh geez |
| 06:10.59 | [TK]D-Fender | dudes, you are a step away from a boot in the ass... |
| 06:11.09 | dudes | oh scared |
| 06:11.17 | jameswf-home | ~random |
| 06:11.19 | drmessano | 0 to "your mom" in 30 seconds.. welcome to IRC |
| 06:11.28 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 06:11.34 | [TK]D-Fender | .... |
| 06:11.34 | jameswf-home | uh oh |
| 06:11.36 | Daviey | ttfn |
| 06:12.11 | [TK]D-Fender | dudes, please just stop now already. |
| 06:12.19 | drmessano | I've not seen anyone push to TK to the point that he OPs up... BRB, going to make popcorn |
| 06:12.22 | *** join/#asterisk asteriskUser5443 (n=fgfdhgdf@ool-43527288.dyn.optonline.net) |
| 06:12.34 | dudes | you started it with the accident comment |
| 06:12.39 | dudes | you quit I shall too |
| 06:12.55 | [TK]D-Fender | dudes, at the end of an obvious joke. |
| 06:13.07 | asteriskUser5443 | anyone awake? |
| 06:13.22 | Daviey | Hmm, can we wrap this up already - i want to see how it finishes, but i'll be late for work |
| 06:13.23 | [TK]D-Fender | dudes, since it was a random pile of "I dunno, could be anythings". |
| 06:13.32 | dudes | I must have misplaced that memo =| |
| 06:14.08 | Robba | ok from what i can see, Voicemailmain tells it to connect to voicemail |
| 06:14.12 | [TK]D-Fender | dudes, Check for Jimmy Hoffa while you're at it... who knows you might get lucky. |
| 06:14.29 | Robba | CALLERIDNUM tells it to use the extension of the phone |
| 06:14.34 | dudes | perhaps |
| 06:14.37 | craigk | sorry to side track people, but i have a question :). I am using the Dial application to ring several phones at once: Dial(SIP/1&SIP/2). If I pick the call up on phone 1, when the call ends the CDR says it was picked up on phone 2. It seems that it is always being recorded as being taken by the last entry in the Dial ... anybody else seen this behaviour ? |
| 06:14.41 | dudes | if you say so |
| 06:15.03 | SwK | anyone remember what the service is where you can get CNAM over HTTP? |
| 06:15.31 | [TK]D-Fender | Robba, more like it uses the caller ID number. But then again, this is the part that is deprecated. That variable does not exist in 1.4 |
| 06:15.59 | [TK]D-Fender | Robba, a fact you should notice because it immediately shows you the "@" without the number. |
| 06:16.03 | asteriskUser5443 | Polycom question. They recently implemented a feature in their phones that when you press DND on the phone it notifies the sever about it. However despite the fact that it's in SIP specs current version if asterisk doesn't support it. Does the newest beta version have it? |
| 06:16.09 | [TK]D-Fender | Robba, "core show function CALLERID" |
| 06:16.31 | [TK]D-Fender | asteriskUser5443, Go check Mantis..... maybe it does. |
| 06:16.46 | neonerz | craigk: just takeing a guess, but do you have a hangup priority? |
| 06:16.50 | Daviey | asteriskUser5443: then come back and let us know either way :) |
| 06:17.16 | craigk | neonerz: not that i am aware of |
| 06:17.31 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 06:18.07 | neonerz | oh wait I misunderstood the question |
| 06:18.17 | neonerz | but you should have a hangup priority anyway :) |
| 06:18.20 | drmessano | Recall the bombers, and take us back to defcon 5 please |
| 06:18.21 | Robba | ok so what your saying is i should use CALLERID instead of CALLERIDNUM? |
| 06:18.22 | dudes | <asteriskUser5443> - Almost anything SIP in asterisk is a pain |
| 06:18.27 | dudes | DND for example |
| 06:18.36 | dudes | I wish it worked =) |
| 06:18.38 | craigk | I have added debug statements to the ast_cdr_setdestchan function and can see that the correct destChannel is set ... but during the hangup processing the incorrect one is set :( |
| 06:18.50 | [TK]D-Fender | Robba, You should be using the CALLERID function, not the deprecated variables. |
| 06:19.24 | dudes | I would think the phone would take care of that -- since it should be on that end -- at least I think it should any how |
| 06:19.52 | neonerz | craigk: try throwing in a exten => h,1,Hangup at the end of the context |
| 06:20.16 | neonerz | I can't say it will solve your problem, but it's better then not having it |
| 06:20.27 | asteriskUser5443 | So far the DND is the biggest issue. We're going to have 90 phones and it's either we use DND on the phone and the receptionist doesn't know the status or we dial in DND and the users have no way of seeing their status |
| 06:20.34 | [TK]D-Fender | neonerz, Sorry, that really won't do anything of value. |
| 06:20.56 | [TK]D-Fender | neonerz, thats saying "Don't quite.... I wanna quit first" |
| 06:20.59 | [TK]D-Fender | quit* |
| 06:21.10 | [TK]D-Fender | neonerz, it'll only confuse your logs |
| 06:21.12 | Robba | sorry i didn't realise that value was no longer used |
| 06:21.14 | dudes | asteriskUser5443 - you ever considered using a FastAGI backend to manage that? |
| 06:21.15 | neonerz | I thought he was doing dial(sip/1) dial(sip/2) and it was calling the second extension |
| 06:21.18 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:21.30 | [TK]D-Fender | asteriskUser5443, there are a few ways. |
| 06:21.30 | Daviey | [TK]D-Fender: "No you hangup" "no you!" |
| 06:21.40 | neonerz | its not good to throw a h in there? |
| 06:21.57 | neonerz | I had the problem of calls going to the next priority after hangup without it? |
| 06:22.05 | [TK]D-Fender | asteriskUser5443, you can use a link key w/ presence to indicate the server based DND. Or you can advertise it via the idle microbrowser |
| 06:22.08 | craigk | it did not help anyhow ... and i am dial all extensions at the same time, not in order |
| 06:22.18 | Daviey | on a detected hangup, then hangup |
| 06:22.21 | Daviey | :/ |
| 06:22.22 | [TK]D-Fender | neonerz, "h" for no good reason is a complete waste |
| 06:22.36 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 06:22.51 | asteriskUser5443 | never heard of fastagi - will check it out. |
| 06:23.09 | [TK]D-Fender | asteriskUser5443, AGI not required. |
| 06:23.15 | neonerz | The reason I added it was if the farend dropped the call, the dialplan would goto the next priority without it |
| 06:23.52 | neonerz | so I had a catch all at the end of my context that grabbed any calls that couldn't be dialed and would return a playback |
| 06:23.58 | [TK]D-Fender | neonerz, then don't put another priority. |
| 06:24.13 | [TK]D-Fender | neonerz, and let your dial be the end of it. |
| 06:24.14 | asteriskUser5443 | Fender, know of any how-to on how to implement what you wrote? |
| 06:24.23 | *** join/#asterisk MaliutaWrk (n=nikolai@203.201.152.211) |
| 06:24.50 | [TK]D-Fender | asteriskUser5443, lookup "asterisk custom deviceState patch" in Google. You'll find it soon enough. |
| 06:25.06 | [TK]D-Fender | asteriskUser5443, and the Microbrowser bit? Thats a "gimme" |
| 06:26.02 | asteriskUser5443 | Cool! Thanks man. I should be able to figure it out from here. |
| 06:27.30 | [TK]D-Fender | asteriskUser5443, Its just a question of which approach works best for your scenario and in your bigger picture. |
| 06:27.50 | neonerz | I'm an idiot, I was using _. instead of _X. for my catch all |
| 06:27.58 | neonerz | thats why it was playing on hangup |
| 06:28.00 | [TK]D-Fender | asteriskUser5443, I use the MB for my call center agents I monitro 4 agents & 2 queues in high detail on 10s frequency. |
| 06:28.30 | [TK]D-Fender | neonerz, indeed "_." is considered a capitol offense. |
| 06:29.20 | [TK]D-Fender | neonerz, if you are ever forced to use it the first thing you should do is call a macro passing the exten as an arg and get yourself to a safe exten. |
| 06:29.34 | asteriskUser5443 | Interesting, I'll research both ways. Thank you for pointing me in the right direction. |
| 06:30.02 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 06:30.35 | [TK]D-Fender | asteriskUser5443, You're welcome. It WOULD be nice if a lot of the funky stuff Polycom supports got accepted into chan_sip. There's talk about finalizing Park & Agent Login stuff for 1.6 last I heard. |
| 06:31.00 | neonerz | thanks, I'll make a note of that |
| 06:31.01 | [TK]D-Fender | asteriskUser5443, Server based DBD announce in SIP 3.0 is another nifty option. |
| 06:31.03 | *** part/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
| 06:31.24 | neonerz | is there anyway to get asterisk to mark the RTP traffic with dscp tags? |
| 06:31.48 | [TK]D-Fender | asteriskUser5443, Would be relatively trivial to make * react to that SIP message by initiating a Local channel in which you could do whatever you wanted with it. |
| 06:32.00 | asteriskUser5443 | I thought that what polycom did is standard SIP |
| 06:32.04 | [TK]D-Fender | neonerz, Yup... you have the source.... get coding :) |
| 06:32.12 | neonerz | lol |
| 06:32.20 | [TK]D-Fender | neonerz, (translation : No, nothing *easy*) |
| 06:32.24 | neonerz | I was hoping for a dscp=46 |
| 06:32.52 | neonerz | would be nice |
| 06:32.53 | [TK]D-Fender | asteriskUser5443, All of these funky new things are not part of the standard spec. |
| 06:33.17 | [TK]D-Fender | asteriskUser5443, but easy enough to accomodate. My last idea for the Local channel would be quite powerful. |
| 06:33.58 | [TK]D-Fender | asteriskUser5443, And at minimal effort. I somehow think even being a complete newb to C I might be able to do it :) |
| 06:34.19 | [TK]D-Fender | asteriskUser5443, But otehr would care too much about HOW I do it... *sigh* |
| 06:34.36 | [TK]D-Fender | anyways, keep it cool people... bed beckons. |
| 06:34.47 | neonerz | same here |
| 06:34.48 | neonerz | night |
| 06:34.49 | drmessano | chow TK |
| 06:34.53 | *** part/#asterisk neonerz (i=18bb0206@gateway/web/ajax/mibbit.com/x-3e7fc19ccf0452d4) |
| 06:34.59 | [TK]D-Fender | later |
| 06:37.46 | *** part/#asterisk Robba (n=rob@203.56.181.15) |
| 06:38.46 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:48.32 | *** join/#asterisk Frogzoo (n=Frogzoo@CPE-121-217-198-130.nsw.bigpond.net.au) |
| 06:49.08 | Frogzoo | any recommendations for voip/sip/iax handsets? |
| 06:49.51 | Frogzoo | cisco are nice but too exxie, otherwise the grandstream phones look good |
| 06:50.11 | drmessano | Grandstream is cheap shit |
| 06:50.36 | Frogzoo | so cisco or nothing? |
| 06:50.49 | Frogzoo | any alternatives? |
| 06:50.51 | drmessano | Who the heck said that |
| 06:50.56 | drmessano | Google is your friend |
| 06:51.15 | drmessano | Linksys, Polycom, Aastra, Snom |
| 06:51.32 | drmessano | Varying degrees of success with any of those |
| 06:52.44 | Frogzoo | really? grandstream is cheap vis a vis linksys? |
| 06:53.27 | drmessano | Yeah |
| 06:54.17 | Frogzoo | hard to tell from a web photo, but these look good: http://grandstream.com/gxp2020.html |
| 06:54.36 | drmessano | ok |
| 07:01.57 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:02.26 | drmessano | ~ron paul |
| 07:02.27 | jbot | ZOMG RONPAULAPPLEUBUNTU FOR PRESIDENT |
| 07:02.48 | *** join/#asterisk ahbritto (n=guest@adsl-69-104-3-183.dsl.pltn13.pacbell.net) |
| 07:06.14 | *** join/#asterisk MaliutaWrk (i=nikolai@119.11.98.42) |
| 07:14.05 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 07:20.27 | mvanbaak | kyron: yes, I was the one reccommending that book |
| 07:24.40 | *** join/#asterisk littleball (n=littleba@bb220-255-71-66.singnet.com.sg) |
| 07:24.41 | littleball | hello |
| 07:25.06 | littleball | i have installed 1.4.7, from CLI, i cannot disable the debug info by set verbose 0 |
| 07:25.17 | littleball | i can see lots of output |
| 07:25.29 | littleball | but i want to disable all these output from cli |
| 07:28.55 | styelz | littleball: edit logger.conf .. console => ... |
| 07:29.25 | littleball | thanks |
| 07:30.19 | mvanbaak | and you can set debug seperately from verbose |
| 07:30.24 | mvanbaak | set debug off |
| 07:31.25 | littleball | then reload, right? |
| 07:31.46 | mvanbaak | I'm off |
| 07:32.18 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 07:33.06 | *** join/#asterisk awk (n=awk@65.111.177.74) |
| 07:36.11 | awk | good day, does somebody have a way to archieve all sent faxes through hylafax... so I can view the archive of all faxes at a later stage? |
| 07:42.54 | *** join/#asterisk af_ (n=getsmart@88-149-240-211.dynamic.ngi.it) |
| 07:43.28 | *** join/#asterisk Frogzoo (n=Frogzoo@121.217.198.130) |
| 07:53.53 | *** join/#asterisk rabelais (n=blank@hpolaris.Stanford.EDU) |
| 07:56.37 | rabelais | I have a linksys spa3102 that's pulling in an PTSN line to my asterisk server, I can dial out and receive calls just fine, I need help with a dialplan entry, to unmask call blocking if I were directly connected to the PTSN line, I would have to put in *82 wait 1 sec for it to register, then dial my number...I want to emulate this functionality from my asterisk system, but am having trouble |
| 07:58.02 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:58.15 | rabelais | I have tried setting a dialplan entry in my spa3102 of: *82 P1 91xxxxxxxxxx and a corresponding dial entry in extensions.conf of SIP/spa3102/*82ww91${EXTEN}... |
| 07:58.40 | rabelais | but the two don't seem to match properly |
| 08:00.11 | drmessano | Try getting rid of the *82ww91 in Asterisk |
| 08:00.58 | rabelais | getting rid of the ww's? |
| 08:01.06 | drmessano | Try getting rid of the *82ww91 in Asterisk |
| 08:01.06 | rabelais | or everything? |
| 08:01.46 | *** join/#asterisk sergey (n=sergey@213.24.100.5) |
| 08:02.16 | rabelais | sorry, I don't quite follow, do you mean manually type the *8291.... stuff from my phone and let asterisk pass all of it over as the extension? |
| 08:02.47 | rabelais | or just trying to make a local call? |
| 08:03.20 | drmessano | You dial the xxxxxxxxx part |
| 08:03.25 | rabelais | that will work |
| 08:03.44 | rabelais | I have that part working, I can make calls out...only they will show up with CID that is private |
| 08:04.11 | drmessano | ok |
| 08:04.15 | *** join/#asterisk AJayMN (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 08:04.57 | drmessano | Sounds like the *82 isn't working |
| 08:05.04 | rabelais | it's just this pause business that is messing me up, I don't quite know how it works |
| 08:05.47 | rabelais | I've put in stuff that has *xx's and things before, and that works just fine...it's just a timing issue, I don't know how to get asterisk and the linksys machine to communicate that there needs to be a wait after the *82 |
| 08:06.38 | *** join/#asterisk nebojsajsimic (n=nebojsaj@cable-89-216-16-106.static.sbb.co.yu) |
| 08:06.45 | rabelais | from what I understand, a w in an extensions.conf dial string will give it a 0.5 sec pause wherever inserted |
| 08:06.47 | AJayMN | I tried to upgrade 1.2.23 to 1.2.26.3 and now im getting errors when asterisk tries to load format_mp3.so NO Such file or Direcorty.. yet its there |
| 08:07.47 | rabelais | AJayMN, try explicitly specifying the path to the file, it sounds like the asterisk default directory got moved during the upgrade |
| 08:07.54 | drmessano | Does EVERY call thru the SPA3102 need the *82? |
| 08:08.12 | rabelais | drmessano, no, not at all, only the ones that I want to unmask callerid blocking on |
| 08:08.18 | drmessano | oh |
| 08:08.39 | drmessano | So I think that you need is |
| 08:08.40 | AJayMN | rabelais /usr/lib/asterisk/modules/format_mp3.so: cannot open shared object |
| 08:08.45 | AJayMN | what would i do? |
| 08:09.51 | rabelais | AJayMN, the other thing I would suggest is to check ownership/permissions to the file, find out what user the asterisk process is running as, and then see if that user can access that file |
| 08:10.31 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:11.06 | drmessano | (*82 P1 x.|x.) maybe.. |
| 08:11.23 | drmessano | Hmm |
| 08:11.23 | drmessano | no |
| 08:12.30 | rabelais | drmessano, you don't need to worry about getting the non *82 calls working, I only need the part of the dial plan that will work with the *82 |
| 08:13.02 | rabelais | I have a long dial plan on the spa3102 that does other stuff, it's just this *82 part that I can't seem to get to work, but that's because it's the only one that has a pause |
| 08:13.21 | drmessano | (<*82x.>*82 P1 x.|x.) |
| 08:13.23 | drmessano | Try that |
| 08:14.12 | rabelais | ooh, a substition! I didn't think of that |
| 08:15.12 | *** join/#asterisk Frogzoo (n=Frogzoo@121.217.198.130) |
| 08:17.01 | rabelais | hehe! |
| 08:17.06 | rabelais | that was the trick I needed |
| 08:17.10 | drmessano | Cool |
| 08:17.27 | rabelais | ended up needing a P3, but it worked! |
| 08:17.32 | rabelais | thanks drmessano |
| 08:17.41 | drmessano | I was thinking that earlier.. too.. P3 is better |
| 08:17.45 | drmessano | Cool |
| 08:18.10 | rabelais | I had no idea of what the timing was like, but I knew the voice error that I'd get if the pause wasn't long enough |
| 08:18.11 | rabelais | hehe |
| 08:18.25 | drmessano | You dial *82 and you get a bum bum bum bummmm here before the tone stabilizes again.. thats abou t3 seconds |
| 08:18.37 | drmessano | about 3* |
| 08:18.56 | rabelais | hehe, ya....it's exactly that weird bum bum thing |
| 08:18.57 | rabelais | hehe |
| 08:19.58 | rabelais | thanks again drmessano, the substitution did the trick |
| 08:20.04 | drmessano | Youre welcome |
| 08:20.06 | drmessano | Have fun! |
| 08:20.12 | drmessano | I am outta here... night all |
| 08:20.42 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:26.26 | *** join/#asterisk oej (n=olle@gw-sthlm01.rebtel.com) |
| 08:28.00 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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| 08:39.13 | *** join/#asterisk AJayMN (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 08:39.37 | AJayMN | Someone I have multiple instances of Asterisk on my box.. 2 different versions.. how can i uninstall the old ver? |
| 08:40.40 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 08:41.13 | *** join/#asterisk tuxfoo (n=tmmarini@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 08:41.28 | mort_gib | AJayMN rm -fr /usr/src/asterisk-version-you-want-to-delete |
| 08:41.50 | mort_gib | AJayMN cd /usr/src/asterisk-you-want-to-use |
| 08:42.01 | mort_gib | make clean |
| 08:42.25 | mort_gib | ./configure and menuselect if applicable ... |
| 08:42.29 | mort_gib | make |
| 08:42.32 | mort_gib | make install |
| 08:42.36 | mort_gib | make config |
| 08:42.39 | mort_gib | make samples |
| 08:43.07 | AJayMN | mm ok.. ill try that. |
| 08:44.23 | AJayMN | mort_gib stupid question is /usr/src/asterisk-version... have the asterisk system program stored there? |
| 08:44.30 | *** join/#asterisk _gm (n=mustafa@221.132.115.174) |
| 08:44.37 | _gm | hi guys |
| 08:44.48 | uwe | is there any softphone that supports g729 ... i need it to test, my asterisk is crashing since i installed g729 codec from digium! and i dont want to wait for calls to crash :) |
| 08:44.55 | _gm | i m having a strange problem with pri disconnection |
| 08:45.18 | _gm | if called party hangs up the line. zap channel hangs up after 15 seconds |
| 08:45.25 | AJayMN | uwe older versions of X-Lite did |
| 08:45.54 | mort_gib | AJayMN you have to download the source, then untar it, that becomes your /usr/src/asterisk-version |
| 08:46.22 | uwe | AJayMN, like, how old ? before which version ... do you have any idea ? |
| 08:46.25 | mort_gib | If you make clean and make make install you should have a clean install of /usr/src/asterisk-version |
| 08:46.25 | AJayMN | ya already did that .. thats probibly why evertime id reboot and start asterisk it would be a different version ;) |
| 08:47.43 | _gm | anyone out here? |
| 08:47.45 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 08:47.59 | mort_gib | the "make config" copies the startup scripts |
| 08:48.53 | *** join/#asterisk sx|lappy (n=sxpert@lgit-1225.obs.ujf-grenoble.fr) |
| 08:50.23 | rabelais | _gm, there isn't a free softphone with g729, but i believe you can buy a copy of eyebeam that has g729 included |
| 08:51.10 | AJayMN | mort_gib when it finished make install i get a message saying there are some files in the modules dir that this version did not install... but when i run asterisk nothing happens i check the log and it says cant find format_mp3.so |
| 08:51.23 | AJayMN | yet that file is in the modules dir.. (it is one file it complained didnt install) |
| 08:51.30 | rabelais | unless you don't care about the licensing, then I'd go with bol sipphone at www.bol2000.com |
| 08:51.44 | Alexandre_fr | hello |
| 08:51.50 | _gm | rabelais, that's not my question ;) |
| 08:52.16 | mort_gib | Uhm, remove those files first, or at least unload them! |
| 08:52.31 | rabelais | _gw sorry, got mixed up with the names, apologies, it's late here |
| 08:52.56 | uwe | how can debug why asterisk crashed on g729 ? it says nothing in dmesg, and logs show nothing either i have verbosity and debug on :( |
| 08:53.02 | AJayMN | mort_gib stupid question.. where are they listed to be loaded? |
| 08:54.13 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 08:54.35 | mort_gib | AJayMN I'm guessing that it's zaptel modules that in the way, but have a look at /etc/asterisk/modules.conf |
| 08:54.35 | Alexandre_fr | I have a question about users.conf , what are the advantages and inconveniences of using this file instead of sip.conf etc ... ? |
| 08:55.10 | AJayMN | ok |
| 08:55.16 | synthetiq | how can i have asterisk agi run a perl script returning a varaible to asterisk to use in the dial plan? |
| 08:55.23 | mort_gib | uwe you would want to look in /var/log/asterisk/messages (trail -f /var/log/asterisk/messages) |
| 08:55.39 | synthetiq | tail -f =] |
| 08:55.59 | mort_gib | :-) Sorry |
| 08:56.49 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 08:57.50 | uwe | mort_gib, i have messages, errors ..etc all writing to /var/log/asterisk/full ... still nothing useful there ... asterisk dies silently :( |
| 08:58.41 | mort_gib | uwe -Sorry then I can't help... |
| 08:59.07 | mort_gib | uwe install of g729 went fine??, reinstall?? |
| 08:59.42 | uwe | reinstall g729 ? its just copying the .so file and registering :S |
| 09:00.17 | mort_gib | Yes, so file rights?? |
| 09:00.52 | mort_gib | does the .so file load correctly?? |
| 09:11.15 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 09:11.34 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 09:12.27 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:17.15 | uwe | WELL, YES I TDOES |
| 09:17.49 | agx | morning, anyone from germany that can point me out to the regional settings for a Linksys PAP2(T) for that country? |
| 09:17.59 | *** join/#asterisk jblack (n=jblack@pool-71-173-53-239.sctnpa.east.verizon.net) |
| 09:19.11 | uwe | sorry, didnt mean to scream :) |
| 09:20.57 | AJayMN | mort_gib any other idea? im getting Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when i try to start asterisk, then do asterisk -rv |
| 09:21.52 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-cc20dd35ca640345) |
| 09:26.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:27.22 | *** join/#asterisk WorgiL (n=usta@78.162.92.112) |
| 09:27.52 | jblack | AJayMN: It means asterisk isn't running. Perhaps it's not starting. In the case of debian and ubuntu, look in /etc/default/asterisk to see if startup is disabled. |
| 09:28.33 | WorgiL | hi everyone how can buy g729a codec ? |
| 09:28.55 | J4k3 | www.digium.com |
| 09:28.59 | J4k3 | they sell them in the shop, afair |
| 09:29.35 | WorgiL | J4k3, can i register from teher ? |
| 09:30.35 | J4k3 | register? |
| 09:33.45 | *** join/#asterisk XnOSX (n=d491af1a@gateway/web/ajax/mibbit.com/x-f129527e290c4b81) |
| 09:37.14 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:40.37 | uwe | AJayMN, what happens when you start using asterisk -cvvvvvvvvvv |
| 09:45.12 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.252) |
| 09:50.33 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 10:03.00 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:07.31 | *** join/#asterisk cjk (n=ldidelot@195.26.5.254) |
| 10:07.35 | cjk | hi, is there a variable that tells me if the channel is in t.38 or not ? |
| 10:08.23 | synthetiq | how can i have asterisk agi run a perl script returning a varaible to asterisk to use in the dial plan? |
| 10:08.54 | Alexandre_fr | <PROTECTED> |
| 10:13.26 | jblack | syntetiq: Use the Set application. ;) |
| 10:13.39 | jblack | just as you would in the dialplan |
| 10:14.03 | jblack | Alexandre_fr: It's simpler to add new users, but more limited in the configurability. |
| 10:16.44 | J4k3 | is there a free/reasonably-priced voice recognition package for asterisk? |
| 10:16.51 | J4k3 | something for numbers |
| 10:17.00 | J4k3 | what I'm thinking is voice dialing bluetooth headsets |
| 10:18.05 | Alexandre_fr | jblack: have you examples of what is limited ? |
| 10:18.31 | jblack | I don't use users.conf |
| 10:39.22 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
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| 10:45.40 | *** join/#asterisk orn (n=orn@85.197.193.24) |
| 10:46.01 | orn | I'm getting a stupendous amount of "Really destroying SIP dialog" messages all of a sudden, and SIP debugging is disabled. Any ideas? |
| 10:46.19 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 10:46.20 | orn | This is for Register, Notify, Bye and Invite messages |
| 10:47.17 | JT | orn: high load? |
| 10:47.30 | Azam | Hello, i want to know how can i do sip redirection in asterisk |
| 10:48.45 | jblack | I think it's automagically done when possible (i.e. there's three legs and nobody has disabled redirect) |
| 10:49.24 | Azam | i want to send a call to a softswitch so that asterisk is not a part of the call anymore. The softswitch should handle the call itseld |
| 10:49.38 | orn | JT: Shouldn't be... I was seeing this in off-office hours as well. |
| 10:51.47 | nebojsajsimic | Hi does anybody can tell me can i on some way catch Hungup event in asterisk |
| 10:52.41 | nebojsajsimic | i need to program my Agents on that way to know the state of agent who is on a cell phone |
| 10:53.14 | nebojsajsimic | i make login and logout |
| 10:53.25 | nebojsajsimic | which i need |
| 10:53.41 | jblack | nebojsajsimic: Sure. There's an option for Dial() |
| 10:54.04 | jblack | It's g, which you can look up with asterisk -r "show application dial" |
| 10:55.05 | jblack | Once you fall through the dial, check DIALSTATUS for the type of call you're interested in (for example, you're probably interested in ANSWER, but not BUSY) |
| 10:55.29 | nebojsajsimic | no i need dial hungup |
| 10:55.52 | jblack | ?? |
| 10:56.10 | nebojsajsimic | when dial occure i get agent on cellphone and i must mark when he hungup |
| 10:56.26 | jblack | Then use what I said. |
| 10:57.00 | nebojsajsimic | ok i will go to look all Dial parameters Thanks jb!!! |
| 10:57.12 | jblack | Dial with the g option. Check that DIALSTATUS was ANSWER (which means there was an actual call), and then do what you want after the call. |
| 10:58.25 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
| 10:58.44 | J4k3 | woo, excitement at my house |
| 10:58.52 | J4k3 | appears a lot of these skype phones have xlite support |
| 10:59.05 | J4k3 | USB dect cordless here I come |
| 10:59.30 | jblack | I don't understand how those USB phones work when the machine is off. |
| 10:59.31 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 10:59.37 | J4k3 | they don't |
| 10:59.46 | J4k3 | I have an XP box that runs pretty much 24x7 here |
| 10:59.58 | jblack | Heh. That's $20 a month down the toilet. |
| 11:00.07 | J4k3 | nah, about $12 |
| 11:00.13 | J4k3 | cool n quiet to the rescue |
| 11:00.49 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:00.53 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 11:01.00 | cpina | ehllo |
| 11:01.07 | jblack | Ahh. I guess your windows box isn't you game box w/ your typical 3-4 gigs of ram w/ an nvidia 8800 and a pair of drives etc etc? |
| 11:01.14 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 11:01.16 | J4k3 | well |
| 11:01.30 | cpina | i'm trying to do some transcoding to send the calls outside |
| 11:01.47 | J4k3 | its a opteron 1210 with 2gb ddr2-667, 4 HDs (2xRAID1, two 160s, two 500s), nv6150... |
| 11:02.03 | cpina | and I get this error message: Changing codec to 'g729' for this call because of ${SIP_CODEC} variable . I know that SIP softphone doesn't have g729, but Asterisk has the codecs (we bought it). How can i force Asterisk to do the transcoding? |
| 11:02.07 | J4k3 | igp saves a ton of power |
| 11:02.28 | mosty | cpina, disallow=g729 on the sip client |
| 11:02.39 | J4k3 | and a certified 80%+ eff psu |
| 11:02.40 | jblack | Quick math here shows a 400 watt machine runs about 288 kilowatt hours a month. |
| 11:02.50 | cpina | mmm... mosty: i will check it thanks... :-) |
| 11:03.02 | cpina | (i guess in the SIP client section in sip.conf) |
| 11:03.02 | sysadmin-lb22 | Hi All has anyone any good thoughts about an SDK to create a VOIP client ? |
| 11:03.13 | J4k3 | you're going to need a couple 8800s and a high end core2quad and a stack of drives to near 400W sustained consumption |
| 11:03.16 | tzafrir | iaxclient? |
| 11:03.19 | mosty | cpina, yes |
| 11:03.31 | tzafrir | There are several SIP libraries |
| 11:03.35 | J4k3 | typical gamer PC draws about 120W idleish |
| 11:03.43 | J4k3 | 200-250 under load |
| 11:03.55 | J4k3 | sli adds like 50W to that |
| 11:04.02 | jblack | I don't know why you think 200-250 |
| 11:04.19 | jblack | anyways, it's your money |
| 11:04.34 | J4k3 | its needed for business anyways, its my network monitoring client. |
| 11:04.44 | J4k3 | but running XP its less reliable than the network, usually ;) |
| 11:04.45 | cpina | it seems that the same mosty: Ignoring ${SIP_CODEC} variable because it is not shared by both ends. |
| 11:04.59 | mosty | i'm trying to decipher some asterisk logs, what are these "Scheduling destruction of call 'foo@host' in 1500ms" messages? |
| 11:05.00 | cpina | i'm forcing the codec when calling using: |
| 11:05.15 | cpina | exten => s,2,Set(SIP_CODEC=g729) |
| 11:05.49 | sysadmin-lb22 | tzafrir talking to me ? |
| 11:05.58 | jblack | mosty: connections aren't immediately deleted from memory. |
| 11:06.00 | tzafrir | sysadmin-lb22, yes |
| 11:06.17 | mosty | jblack, but the call was still going at this point |
| 11:06.17 | sysadmin-lb22 | tzafrir thanks..I checked it out |
| 11:06.25 | tzafrir | sysadmin-lb22, in fact, why not take an existing client and modify it to your needs? |
| 11:06.26 | sysadmin-lb22 | tzafrir actually still at it |
| 11:06.34 | jblack | you got a scheduled destruction in the middle of a call/ |
| 11:06.58 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:07.01 | mosty | cpina, what are you trying to do? |
| 11:07.08 | mosty | jblack, i believe so |
| 11:07.14 | cpina | phones only supports g711 (ulaw) |
| 11:07.19 | cpina | i want to send the calls using g729 |
| 11:07.28 | cpina | so asterisk has to transcode |
| 11:07.28 | sysadmin-lb22 | tzafrir..well for long term purposes it would be bretter if we build it from scratch..and of course royality issues |
| 11:07.47 | Azam | Can anyone please help me with sip redirection? |
| 11:07.50 | jblack | mosty: You've got me then. |
| 11:07.59 | mosty | cpina, where are you sending the calls to? disalllow=all and allow=g729 there |
| 11:08.18 | jblack | azam: Just remove any references to disabling redirect, and it'll happen automatically if possible. |
| 11:08.48 | jblack | Make sure you don't give any dial options that require monitoring (such as blind transfer, autopark, etc) |
| 11:08.52 | cpina | i'm sending the calls to other system (not asterisk, but using sip), i'm not registered to this gateway so i cannot add disallow=all and allow=g729 |
| 11:09.01 | tzafrir | sysadmin-lb22, why not use one that is free software? |
| 11:09.08 | mosty | jblack, i get this message several times during a call that lasted 25 minutes |
| 11:09.17 | tzafrir | Why pay extra to develop on your own? |
| 11:09.22 | jblack | mosty: I still don't know. |
| 11:09.28 | jblack | tzafrir: To keep copyright? |
| 11:09.31 | mosty | cpina, you don't have to register to set codec preference on a sip peer |
| 11:09.41 | tzafrir | jblack, why? |
| 11:10.05 | tzafrir | What does it help you that you have copyrights for an inferior product? |
| 11:10.05 | sysadmin-lb22 | tzafrir..actually as jblack said copyright issues ...and of course secrecy etc |
| 11:10.08 | jblack | Because code takes resources to develop, thus having value? |
| 11:10.22 | sysadmin-lb22 | and SDKs are well tested |
| 11:10.42 | tzafrir | I wouldn't want to use something that depends on secrecy |
| 11:11.06 | sysadmin-lb22 | well it is not all technical there are business rules you have to abide to |
| 11:11.10 | Azam | jblack: in a normal senario a call will come to asterisk, asterisk will send and outbound call to softswitch and bridge the two calls, if i do show channels on my asterisk i will see tht bridged call. What i want to do is, i want my asterisk to get rid of the bridged call. the softswitch should handle the bridged call. |
| 11:11.20 | tzafrir | sysadmin-lb22, again, why? |
| 11:11.25 | jblack | azam: Then do what I said. :) |
| 11:11.53 | tzafrir | you have to spend time anyway on development. So why add extra artificial licensing costs to the process? |
| 11:11.57 | BBHoss | whats the deal with this new voicebus card from digium? |
| 11:11.58 | jblack | sysadmin-lb22: I don't know if secrecy is the right route. Less eyes means less code review. |
| 11:12.00 | Azam | jblack: sir can u please advise how to do it? :) |
| 11:12.32 | jblack | azam: I already told you how. To do more, I'd have to do it for you. |
| 11:12.43 | tzafrir | (licensing costs are not just money. They include all sorts of other resources wasted) |
| 11:12.46 | cpina | mosty: sip peer? where can i setup to force on G729 for outboudn calls? |
| 11:13.15 | Azam | jblack: i m a bit confused anyways i will try Thanks Alot for your help |
| 11:13.50 | mosty | cpina, in your sip peer definition, disallow=all and allow=g729 |
| 11:14.06 | mosty | cpina, then asterisk will only be able to use g729 when sending calls to that peer |
| 11:14.22 | J4k3 | am I completely out of my mind for considering a low-end phenom on a nv6100 board for my asterisk box |
| 11:14.26 | J4k3 | ;) |
| 11:14.32 | jblack | sysadmin-lb22: By the way, there are forks of asterisk that don't require copyright assignment. |
| 11:14.43 | coppice | BBHoss: voicebus is the result of some serious engineering the marketing department :-) |
| 11:14.54 | Frogzoo | BBHoss: link? |
| 11:15.23 | jblack | sysadmin-lb22: But even asterisk itself only requires copyright assignment to have your patches accepted into the codebase. |
| 11:15.31 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 11:15.36 | atis_work | anyone knows the policy of argument separation - pipe vs comma? |
| 11:15.55 | BBHoss | Frogzoo, blogs.digium.com |
| 11:16.06 | jblack | sysadmin-lb22: Generally, it's a good idea to avoid forking, because by providing your patches back to the system you're using, the code is maintained and improved by everyone, not just by you. |
| 11:16.13 | cpina | mosty: because i'm not registering where i'm sending the calls, this peer doesn't appear on sip.conf |
| 11:16.17 | Frogzoo | BBHoss: thanks |
| 11:16.24 | cpina | what i did is to allow g729, g711 and disallow in the individual sip peers |
| 11:16.30 | orn | ? |
| 11:16.32 | cpina | but still not working |
| 11:16.43 | atis_work | i just tried my dialplan in 1.6 - and GoSub doesn't accept pipes in realtime, however i remember that i was pointed out some time ago that realtime arguments should be separated by pipe |
| 11:16.44 | jblack | cpina: Did you buy licenses for g729? |
| 11:16.47 | cpina | yes |
| 11:17.03 | cpina | 0/0 encoders/decoders of 10 licensed channels are currently in use |
| 11:17.07 | *** join/#asterisk Grash (n=grash@cdbil1-a1-2-23.ipcom.comunitel.net) |
| 11:17.10 | mosty | cpina, it doesn't matter if you register or not, asterisk will use the allow/disallow settings in the sip peer when it sends calls to that peer |
| 11:17.14 | Grash | Hi people ! |
| 11:17.22 | jblack | Ok, well, as mosty said, disallow all, then allow g729. |
| 11:17.25 | cpina | mosty: then i will check... |
| 11:18.32 | Grash | Does anybody know how to avoid the message "chan_iax2.c:6027 update_registry: Restricting registration for peer 'X' to 60 seconds (requested 300)" ? |
| 11:18.49 | atis_work | Corydon76-dig: Corydon76-vcch: ping |
| 11:19.06 | BBHoss | damn tornadoes woke me up this morning |
| 11:19.18 | jblack | You say it as if tornados were roosters. |
| 11:19.24 | J4k3 | hmm, I'll get a cheapo 4000+ x2 for now, then upgrade to a phenom when load increases. |
| 11:19.38 | BBHoss | jblack, might as well be with all the sirens and such |
| 11:19.44 | jblack | Roosters don't throw around mooing cows.... and houses with dorothy's. |
| 11:19.44 | uwe | confirmed ... g729 crashes my asterisk :( ... and asterisk dies silently !! |
| 11:20.14 | mosty | uwe, which version of the g729 module, and which version of asterisk? |
| 11:20.26 | BBHoss | damn, i was hoping the democratic primary would be clear cut :( |
| 11:20.27 | jblack | What's with all these people this morning? |
| 11:20.49 | uwe | mosty, Asterisk 1.4.15, and g729 v33 i686 |
| 11:21.26 | jblack | BBHoss: Heh. Oh well. (/me hopes clinton doesn't win because of the "get health insurance or we'll tax health insurance out of you" thing) |
| 11:21.28 | mosty | uwe, i notice that there are fixes for g729 in the asterisk 1.4.18 release candidates, try reading the changelog on those |
| 11:21.49 | uwe | the crash happens when i try to i try to answer a call that uses g729 somewhere in its path |
| 11:22.10 | BBHoss | jblack, yeah that concerned me to, my biggest fear is that she allows the corporations to control it, just mandating prices, which will of course be changed in a few years |
| 11:22.38 | BBHoss | she did win california and florida :( |
| 11:22.44 | J4k3 | jblack: clinton is talking republican-speak |
| 11:23.05 | uwe | oh my god !!! [TK] is not here !!! :D this is really strange !! |
| 11:23.07 | jblack | I think that might be the plan (which I'm ok with). Force insurance, then hospitals/insurance companies bilk, then she has to "fix" it by socializing it (which I'm actually for) |
| 11:23.14 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
| 11:23.28 | uwe | mosty, ill look into that |
| 11:23.35 | jblack | uwe: He does have one of those "job" things. Probably has a nifty 'girlfriend' model too. |
| 11:23.43 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 11:24.59 | uwe | jblack, hehee ... sure he does ... but its like ... he has been always there ever since i started visiting this room :) |
| 11:25.09 | uwe | err, channel |
| 11:25.18 | jblack | You've been here for how long? |
| 11:26.00 | J4k3 | ([tk], not me) |
| 11:26.09 | jblack | sure. But it's 6:30 am where he is. |
| 11:26.45 | BBHoss | hell its 5:30 here :) |
| 11:26.59 | jblack | where is it 5:30 am? Brazil? |
| 11:27.05 | BBHoss | Alabama |
| 11:27.27 | jblack | Oh, silly me. |
| 11:27.55 | jblack | I'm getting tired. I spun the intenal time-zone clock in the wrong direction |
| 11:29.15 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:30.47 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:35.15 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 11:37.27 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 11:39.04 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 11:39.25 | sergee | jblack: 14:39 (02:39 pm :) ) |
| 11:39.28 | sergee | here |
| 11:41.25 | jblack | That's pretty far east. |
| 11:41.51 | coppice | 14:39 sounds very west to me |
| 11:42.52 | tzafrir | That's only slightly west |
| 11:43.02 | tzafrir | But very far east |
| 11:43.53 | jblack | two wrongs don't make a right, but three lefts do. If you can reach it by going west, you can reach it by going east. |
| 11:45.10 | *** join/#asterisk qdk (n=qdk@193.164.155.7) |
| 11:45.11 | Frogzoo | but not left three times |
| 11:45.23 | jblack | Though there are two spots where you can't go go east at all.... |
| 11:49.35 | coppice | If I go more than a few hundred metres east from here I might drown |
| 11:51.02 | J4k3 | wear a life preserver |
| 11:54.40 | coppice | its better to use a pier to pier service |
| 11:55.04 | sergee | jblack: it's more like far east europe :) |
| 11:55.46 | coppice | more like far west asia |
| 11:56.45 | cpina | hello again :-) |
| 11:56.49 | cpina | mosty: thanks for your help |
| 11:56.56 | sergee | coppice: asia ends in Urals :) |
| 11:57.04 | cpina | system worked fine removing the Set(CODEC=g729) |
| 11:57.11 | cpina | and adding in sip.conf |
| 11:57.43 | mosty | cpina, no problem |
| 11:58.17 | cpina | the peer :-) |
| 11:58.22 | cpina | sorry i was busy changing it |
| 11:58.35 | cpina | now is working fine, incorrect approach (Set(CODEC)) to fix this problem |
| 11:58.38 | cpina | so thanks again |
| 11:59.09 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 11:59.19 | cpina | (here i am again) |
| 12:03.28 | uwe | are older versions of g729 codecs available, and would it be considered acceptable to try to use an older one ? |
| 12:03.46 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:19.19 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 12:28.07 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 12:29.12 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:32.25 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 12:41.14 | orn | I'm getting a stupendous amount of "Really destroying SIP dialog" messages all of a sudden, and SIP debugging is disabled. This is related to REGISTER, INVITE, NOTIFY and BYE. Any idas? |
| 12:46.21 | *** join/#asterisk alrs (i=non-knav@pozug.com) |
| 12:54.50 | BBHoss | orn, i would ignore them unless you're having trouble. I am pretty sure they are just debug messages |
| 12:56.09 | BBHoss | codename-pineapple.org |
| 12:56.11 | BBHoss | ? |
| 12:56.30 | cjk | hi, is there a variable that tells me if the channel is in t.38 or not ? |
| 12:59.22 | yang | I am having a hard time hearing a call. I am calling via VOIP phone<->Router<->ASTER<->internet<->E1 . Here are debug lines from ASTER http://openpaste.org/en/4995/ and E1 http://openpaste.org/en/4994/ ... I cannot hear the back signal in my voip phone, but i can hear the voice when calling my mobile phone. |
| 13:00.15 | yang | Aster has IP 10.105.2.3 and also public IP 212.13.242.122 |
| 13:01.51 | yang | I have tried all nat=yes nat=no options |
| 13:02.30 | BBHoss | yang, you must forward port 5060 and ports 10000-20000(UDP for both) to your asterisk box |
| 13:02.55 | yang | Yeah |
| 13:03.07 | yang | From the router? |
| 13:03.15 | BBHoss | yeah |
| 13:03.20 | BBHoss | make sure its UDP |
| 13:06.24 | yang | but as i know its all opened |
| 13:06.35 | yang | becouse ASTER |
| 13:06.44 | yang | is mirrored |
| 13:06.46 | BBHoss | do you have externip and localnet defined in sip.conf? |
| 13:06.48 | yang | to a public ip |
| 13:06.55 | yang | no I don't have - shit |
| 13:06.59 | BBHoss | heh |
| 13:07.03 | yang | :/ |
| 13:07.18 | yang | externip would be 212.13.242.122 |
| 13:07.18 | BBHoss | that one got me too, i was having your exact problem, half-way audio |
| 13:07.22 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 13:07.24 | yang | and localnet 10.105.2.3? |
| 13:07.37 | lirakis | morning |
| 13:07.49 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:07.54 | BBHoss | localnet woul probably be 10.105.2.0/255.255.255.0 depending on your subnet |
| 13:08.00 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 13:12.03 | FlatFoot | hey ho all |
| 13:12.36 | BBHoss | sup dog |
| 13:13.10 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 13:14.13 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:16.44 | ZaVoid | morning all |
| 13:17.57 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 13:19.24 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:22.54 | uwe | i lost "sip" from asterisk cli :( .... i see no errors on startup with -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 13:23.08 | mosty | uwe: is chan_sip.so loaded? |
| 13:23.09 | BBHoss | heh thats no fun |
| 13:24.14 | BBHoss | uwe, when you startuo do you see the sip module load? |
| 13:24.29 | BBHoss | also try reload chan_sip |
| 13:24.34 | mosty | check that it isn't disabled in modules.conf |
| 13:24.37 | uwe | mosty, it is loaded |
| 13:25.03 | mosty | how do you know it's loaded? |
| 13:25.38 | uwe | i compiled 1.4.18-rc4 and replaced 1.4.15 i had |
| 13:26.08 | uwe | show modules like chan_sip.so |
| 13:27.40 | mosty | enable full logging, set verbose and debug to 10, then try reload chan_sip.so |
| 13:28.09 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:29.45 | uwe | i think i found something ... one sec |
| 13:31.03 | [TK]D-Fender | uwe: You trying to run any other SIP software on your server? |
| 13:31.15 | jblack | uwe: There you go, feel better? |
| 13:34.05 | *** join/#asterisk _gm (n=mustafa@58.27.175.222) |
| 13:34.12 | ZaVoid | is there really a difference between debug 6 and 10? i've never noticed a difference |
| 13:34.24 | _gm | anyone here tried out ldap realtime driver bundled with asterisk 1.6-beta |
| 13:34.26 | _gm | ? |
| 13:34.48 | ZaVoid | nope |
| 13:35.05 | ZaVoid | what does that do? that in place of using realtime for accounts? |
| 13:36.56 | _gm | yeah |
| 13:37.06 | ZaVoid | interesting |
| 13:37.14 | ZaVoid | stupid quiestion... why would yo wanna do that? |
| 13:37.22 | _gm | ldap realtime driver was working fine with my previous install of asterisk-14 |
| 13:37.27 | _gm | hmm |
| 13:38.02 | _gm | we are writing a user management console |
| 13:38.03 | _gm | user/computer/mail |
| 13:38.14 | ZaVoid | ahh |
| 13:38.24 | _gm | you can say .. datacenter management gui |
| 13:38.29 | ZaVoid | right gotcha |
| 13:38.34 | _gm | AD for linux ;) |
| 13:38.50 | ZaVoid | so drones login to random terminals and their extension automaticall set up kinda thing via login? |
| 13:39.33 | _gm | yeah actually we want a centarlize authentication instead of creating user everywhere |
| 13:39.56 | _gm | for mail phone desktop and other things |
| 13:40.00 | ZaVoid | right i gotcha |
| 13:40.03 | ZaVoid | thats pretty slick |
| 13:40.10 | ZaVoid | all softlclents for phones then? |
| 13:40.19 | _gm | doesnt matter |
| 13:40.34 | _gm | u see the point is not a single terminal |
| 13:40.55 | _gm | single sign-on for all services like mail, desktop, phone and other things |
| 13:43.25 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 13:45.15 | BBHoss | SSO is a virtue :) |
| 13:46.15 | jblack | So is retaining one's virginity until marriage. |
| 13:46.21 | BBHoss | heh |
| 13:46.59 | tzanger | HAHAHA |
| 13:47.02 | tzanger | awesome! |
| 13:47.16 | jblack | ? |
| 13:47.22 | jblack | was it that funny? |
| 13:47.34 | lunaphyte_ | not having to create redundant users isn't really single sign on. |
| 13:47.58 | coppice | jblack: I think he just giggles when anyone says virgin |
| 13:48.00 | cesar_CR | hi guys a good service provider for voip |
| 13:48.08 | [TK]D-Fender | SSO = multipoint security risk. |
| 13:48.54 | lunaphyte_ | yeah, kinda. although with browsers remembering the credentials anyway, it's kind of mute. |
| 13:49.11 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 13:49.37 | lunaphyte_ | err, moot. |
| 13:49.56 | *** join/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 13:50.51 | *** join/#asterisk PepOSX (n=angeldav@190.79.246.105) |
| 13:55.28 | *** join/#asterisk nighty^ (n=nighty@p3125-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 13:56.47 | kyron | mvanbaak, well thanks a mill, I think I'll really enjoy it. |
| 14:03.26 | Uatec | Terminal Equipment and Network Terminator mean nothing to me. Which is which? |
| 14:03.50 | kyron | TE=your phone. NEtwork terminator...have a few defs in mind.. |
| 14:04.23 | Uatec | well my telco has a bunch of ISDN lines coming in to our office |
| 14:04.38 | Uatec | and i want to connect my asterisk box to them |
| 14:04.39 | tzafrir | Network Terminator is that small thing you put in the end of a coax cable, right? |
| 14:04.46 | [TK]D-Fender | Network Terminator : When Ahnold tries to legislate the web |
| 14:05.14 | Uatec | Can we use a client/server metaphor to describe it? |
| 14:05.25 | Uatec | isn't one the "server" at the telcos end |
| 14:05.29 | [TK]D-Fender | Uatec: NT = The telco |
| 14:05.37 | Uatec | and one the "client" at my end |
| 14:05.38 | Uatec | ok |
| 14:05.52 | Uatec | so i need to setup my sangoma a500 in TE mode then |
| 14:06.21 | Uatec | ok, great, thanks very much :) |
| 14:06.28 | *** join/#asterisk HeXeD (n=hex@87-194-8-43.bethere.co.uk) |
| 14:06.29 | tzafrir | TE - cpe . NT - net |
| 14:07.49 | *** join/#asterisk Greek-Boy (n=grb@41.221.58.4) |
| 14:07.53 | ZaVoid | is there any reason why this would not work in a .php file that i call from my dialplan... $agi->exec("Wait 7"); |
| 14:09.32 | [TK]D-Fender | ZaVoid: Why would you bother? |
| 14:09.35 | mvanbaak | kyron: good :) |
| 14:09.48 | [TK]D-Fender | ZaVoid: jsut wait in PHP |
| 14:09.48 | ZaVoid | why would i bother calling it from the script you mean fender? |
| 14:09.56 | [TK]D-Fender | ZaVoid: exactly. |
| 14:10.01 | ZaVoid | not wait 7 |
| 14:10.13 | mvanbaak | sleep() |
| 14:10.16 | ZaVoid | oh sorry you mean a php vesion of wait and not tell asterisk to wait |
| 14:10.31 | ZaVoid | ahhh thats a good question |
| 14:10.37 | ZaVoid | so sleep(7) |
| 14:11.50 | uwe | apparently something changed in reading the configuration files between 1.4.15 and 1.4.18-rc4 |
| 14:12.11 | Uatec | weird, this a500 has a molex conenctor on the back |
| 14:12.18 | Uatec | it needs it's own power? |
| 14:12.29 | Uatec | or is this just if you're using the daughterboards? |
| 14:12.44 | [TK]D-Fender | Uatec: Just plug it in already... |
| 14:13.13 | Uatec | lol |
| 14:13.17 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 14:14.01 | ZaVoid | hey fender this is where i was using it |
| 14:14.07 | ZaVoid | if (is_null($row['xxxxxxxxx'])) { |
| 14:14.07 | ZaVoid | <PROTECTED> |
| 14:14.07 | ZaVoid | <PROTECTED> |
| 14:14.07 | ZaVoid | <PROTECTED> |
| 14:14.07 | ZaVoid | <PROTECTED> |
| 14:14.19 | ZaVoid | so instead using a sleep(7); instead you think? |
| 14:15.23 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:15.34 | Frogzoo | Uatec: if it's an FXS port, it needs power to supply ring tone |
| 14:16.42 | [TK]D-Fender | ZaVoid: Silence = wait = waste |
| 14:17.01 | ZaVoid | = waste? |
| 14:17.12 | [TK]D-Fender | ZaVoid: Don't tell * to wiat... jsut WAIT |
| 14:17.27 | ZaVoid | right |
| 14:18.00 | [TK]D-Fender | Frogzoo: No such thing as FXS on that card. |
| 14:20.28 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 14:20.35 | _gm | <PROTECTED> |
| 14:20.36 | _gm | <PROTECTED> |
| 14:20.41 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:21.00 | Frogzoo | well I was guessing |
| 14:21.58 | *** join/#asterisk b1ch0 (i=b1ch0@static-200-105-150-33.acelerate.net) |
| 14:25.38 | tzafrir | Uatec, BRI phones take power from the NT |
| 14:26.12 | tzafrir | This is something you mostly don't need . But do need if you want to connect an ISDN phone |
| 14:26.12 | uwe | [TK]D-Fender, BBHoss , after installing 1.4.18-rc4, g729 doesnt crash anymore, and time is down to 2 and 10 ms |
| 14:26.44 | BBHoss | uwe, thats good |
| 14:26.57 | cjk | hi, is there a variable that tells me if the channel is in t.38 or not ? |
| 14:27.07 | BBHoss | cjk, no |
| 14:27.15 | cjk | BBHoss: htanks |
| 14:27.22 | uwe | yes, thats much better, thank you all :) |
| 14:32.54 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 14:33.28 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 14:41.01 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:41.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:44.22 | *** join/#asterisk e}{istence (n=uvedoble@140.Red-80-38-216.staticIP.rima-tde.net) |
| 14:44.30 | e}{istence | hello every body |
| 14:46.32 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 14:48.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:48.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:51.10 | e}{istence | anyone know any softphone for Windows Mobil 6 ? |
| 14:51.31 | file | it has a SIP client built in, unless your carrier has removed it |
| 14:55.04 | [TK]D-Fender | e}{istence: Ever been to www.google.com ? |
| 14:55.13 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 14:55.42 | [TK]D-Fender | e}{istence: You'd be amazed what you can find in under 1 minute. |
| 14:58.42 | Corydon76-dig | atis_work: in 1.6, argument delimiters are ',' not '|'. Realtime, too. |
| 14:58.57 | eric_hill | [TK]D-Fender: 1 MINUTE!? I don't type that fast!!! ;P |
| 14:59.57 | lmadsen | is a callid unique in asterisk? |
| 15:00.17 | lmadsen | i.e.... can I be relatively sure that a callid is going to be unique on a system? |
| 15:00.53 | file | lmadsen: it has to be eh |
| 15:00.59 | lmadsen | ok, thats what I thought |
| 15:01.01 | file | across "space and time" |
| 15:01.15 | lmadsen | :D |
| 15:01.19 | drmessano | Pest control guy shows up.. "Man, this must be the computer room, you work on them or something?" |
| 15:01.23 | denon | file_piccard |
| 15:01.36 | drmessano | "No, I am the hub of a terrorist cell.. I cannot let you leave now" |
| 15:01.37 | file | I just quote the RFC. |
| 15:01.39 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:02.25 | Qwell | drmessano: my favorite is "you must like computers" - "nope, never used one" |
| 15:02.31 | drmessano | lol |
| 15:02.49 | denon | hehe |
| 15:02.54 | denon | "no computers here, just PBXs" |
| 15:03.00 | drmessano | Hindsight is 20/20.. "I know nothing about this shit.. It's my wifes" |
| 15:03.30 | drmessano | I would have demeaned myself just for the reaction |
| 15:04.46 | drmessano | Someone needs to come up with a good Asterisk thug poster.. "I'm a memba of da Asterisk Pound, yo" |
| 15:05.19 | drmessano | Let them report that to their boss.. "I think I found some gang hideout... they call themselves the Asterisk Pound!" |
| 15:06.30 | drmessano | "Da Asterisk Pound - Too Hardcore fo' 9's" |
| 15:07.00 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:08.04 | *** join/#asterisk atop (n=user@oaktyres.force9.co.uk) |
| 15:08.22 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 15:10.00 | drmessano | 30 fatalities from the storms so far |
| 15:10.03 | drmessano | yikes |
| 15:11.32 | lirakis | this is an off topic question... but, does any one here work in publishing / editing? |
| 15:11.50 | eric_hill | lirakis: We have a publishing department... |
| 15:12.18 | *** join/#asterisk FciSoft (n=FabiOne@host107-144-static.59-88-b.business.telecomitalia.it) |
| 15:12.29 | lmadsen | lirakis: I've worked with O'Reilly and edited documents before.... |
| 15:12.36 | lmadsen | guess it depends what you need :) |
| 15:12.43 | lirakis | lmadsen: i knew that ;) ha ha |
| 15:12.47 | Qwell | I once edited a text file. |
| 15:12.53 | lmadsen | and then I fixed it |
| 15:13.08 | *** join/#asterisk Bob-_ (n=Bob_@fingerbottom.tekproj.bth.se) |
| 15:14.20 | lirakis | lmadsen: in all honesty, looking for some networking opportunities with editors etc. for my wife. |
| 15:14.30 | lmadsen | ahhh gotcha |
| 15:15.01 | drmessano | Qwell: I got 300 page book on Nano, if you want to borrow it |
| 15:15.05 | lirakis | lmadsen: i think oreilly does most of their work on the west cost right? |
| 15:15.16 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-124-120.tor.primus.ca) |
| 15:15.24 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:15.26 | lmadsen | lirakis: ya, in Sebatopal (or whatever that palce is called) in Cali |
| 15:15.38 | lmadsen | Sebastopal.... |
| 15:15.50 | Qwell | drmessano: 300...pages? |
| 15:15.52 | lirakis | lmadsen: yeah.. im east coast... was looking at wiley (wrox) hq in NJ |
| 15:16.00 | Qwell | how do you fit 3 commands on 300 pages? |
| 15:16.04 | drmessano | ROFL |
| 15:16.20 | drmessano | Nano: The Future Of Publishing |
| 15:16.22 | Qwell | no, but seriously, there's a book on nano? |
| 15:16.25 | drmessano | No lol |
| 15:16.29 | lirakis | Qwell: lol |
| 15:16.30 | drmessano | That would be.. sad |
| 15:17.12 | drmessano | I can make a PDF of the nano man pages if you like |
| 15:17.51 | Qwell | nano has a man page? |
| 15:18.17 | lirakis | nano man(1) .. open editor.. type.. close... |
| 15:18.39 | drmessano | yes |
| 15:18.45 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088937109.dsl.bell.ca) |
| 15:18.54 | Qwell | somebody should google me up a nanorc with C syntax highlighting |
| 15:19.28 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-82-154.d4.club-internet.fr) |
| 15:19.54 | Qwell | wow, nano actually has some useful cli options |
| 15:20.19 | Qwell | mouse support...autoindent...smarthome |
| 15:22.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:22.17 | L|NUX | Hello every one |
| 15:23.31 | drmessano | yeah |
| 15:25.40 | *** join/#asterisk Y0da^ (n=Bunny@70.159.118.70) |
| 15:25.48 | *** part/#asterisk Y0da^ (n=Bunny@70.159.118.70) |
| 15:26.14 | Bob-_ | I am trying to set up a system for sip conference calls on my nat, is there any way to do this without any specialized hardware? Asterisk will not have an external line, all calls will be internal. All data needs to be sent over standard ethernet as I am using software based clients. |
| 15:27.10 | lirakis | Bob-_: no special hardware is needed... |
| 15:27.20 | lirakis | Bob-_: well.. a pc |
| 15:27.37 | e}{istence | file my carrier remove it |
| 15:27.43 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 15:27.50 | e}{istence | i have a HTC TOUCH |
| 15:28.04 | Bob-_ | "Note that for technical reasons, you must have at least one Zaptel |
| 15:28.04 | Bob-_ | interface (of any kind) installed in your Asterisk system if you wish |
| 15:28.04 | Bob-_ | to use conferencing." |
| 15:28.14 | Bob-_ | http://www.digium.com/handbook-draft.pdf |
| 15:28.17 | e}{istence | [TK]D-Fender i have found one softphone for windows mobile 6 |
| 15:28.19 | Bob-_ | thus: Confusion |
| 15:28.45 | file | e}{istence: you can download a .cab that will install the parts you need for the Windows built in one |
| 15:29.19 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:29.32 | [TK]D-Fender | Bob-_: that doc is ANCIENT, and yes, you must have Zaptel installed. Use ZTDUMMY for your timer and go read a CURRENT book. |
| 15:29.34 | [TK]D-Fender | ~book |
| 15:29.35 | 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 |
| 15:30.46 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 15:31.21 | *** join/#asterisk BadHorsie (n=sebas@201.198.239.167) |
| 15:31.23 | Bob-_ | Zaptel is installed, cant get dialtone. I'll read the docs and troubleshoot some more. Thanks! |
| 15:31.54 | [TK]D-Fender | Bob-_: Dialtone from what? |
| 15:32.04 | freezey | how do i set a good audio codec? |
| 15:32.07 | BadHorsie | how can i generate a phone call besides originate in AMI ? |
| 15:32.13 | e}{istence | file where can i find it ? |
| 15:32.16 | [TK]D-Fender | freezey: allow=ulaw |
| 15:32.31 | file | e}{istence: I don't have the URL, Google can tell you |
| 15:32.36 | [TK]D-Fender | BadHorsie: .call file |
| 15:32.57 | [TK]D-Fender | BadHorsie: "originate" at CLI. |
| 15:33.21 | freezey | [TK]D-Fender: what conf file would that be in? and what does that exactly do? |
| 15:33.39 | [TK]D-Fender | freezey: depends on waht channel type you're working with. |
| 15:33.44 | Bob-_ | Dialtone might be the wrong word. Cant get a connection. Asterisk is up and running, its a wetware problem. |
| 15:33.46 | file | e}{istence: http://wiki.xda-developers.com/index.php?pagename=HTC_Vox read the VOIP section |
| 15:33.56 | [TK]D-Fender | freezey: Its looking like you don't even understand what a codec is.... |
| 15:34.09 | freezey | [TK]D-Fender: SIP |
| 15:34.14 | [TK]D-Fender | ~codec |
| 15:34.20 | freezey | [TK]D-Fender: i know what a codec is |
| 15:34.38 | [TK]D-Fender | freezey: Well go tell your device what codec(s) its allowed to use then |
| 15:35.18 | freezey | [TK]D-Fender: i was just wondering if you had to change a setting in asterisk somewhere.. or just use any old codec.. cause i right now i am testing this over sip and have these sip desktop phones and i get some delay |
| 15:35.18 | *** part/#asterisk nkoth (n=HP_Owner@bas4-ottawa23-1177649251.dsl.bell.ca) |
| 15:35.38 | eric2 | anyone have any luck with the app_rxfax application? |
| 15:35.41 | e}{istence | file thank you very much |
| 15:35.42 | [TK]D-Fender | freezey: Codec rarely has anything to do with delay |
| 15:36.05 | freezey | [TK]D-Fender: so what do you think the delay would have to do with? |
| 15:36.32 | [TK]D-Fender | freezey: Well you haven't mentioned anything informative about your entire setup. |
| 15:36.33 | *** join/#asterisk adjohn (n=adjohn@p1155-ipad91marunouchi.tokyo.ocn.ne.jp) |
| 15:36.40 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:36.43 | [TK]D-Fender | freezey: Am I to start guessing blindly? |
| 15:37.03 | freezey | [TK]D-Fender: well no you could just ask what you want to know in order to try and help me |
| 15:37.09 | *** join/#asterisk ManxPower (n=manxpowe@251.sub-70-223-214.myvzw.com) |
| 15:37.32 | [TK]D-Fender | freezey: You should know to describe all the devices you're using, networking involved, etc. |
| 15:37.33 | xheliox | Anyone have any issue with MixMonitor being out of sync on 1.4.18-rc4? |
| 15:37.46 | freezey | [TK]D-Fender: k i was typing that right now |
| 15:37.58 | [TK]D-Fender | freezey: Describe the delay itself. |
| 15:38.11 | [TK]D-Fender | freezey: is it a pure delay? More like echo? |
| 15:38.19 | freezey | echo |
| 15:38.41 | freezey | you can hear the echo that might just be comin in on the other persons speakers |
| 15:38.50 | freezey | and it seems like a delay |
| 15:39.03 | [TK]D-Fender | freezey: So you are hearing yourself come back? |
| 15:39.10 | freezey | [TK]D-Fender: let me think.. ok so this is strictly being done locally i have a point to point connection through my firewall to another office in manhattan |
| 15:39.16 | freezey | [TK]D-Fender: yeah |
| 15:39.35 | [TK]D-Fender | freezey: Describe the hardware / software involved. |
| 15:39.51 | freezey | [TK]D-Fender: i am using a sip phone called xlite and as well are they... i am using sip proxy for both people to register.. |
| 15:40.11 | freezey | [TK]D-Fender: the box i have asterisk on is just some dell machine its not high high quality but its not horrible either |
| 15:40.19 | freezey | [TK]D-Fender: the firewall is a gnet |
| 15:40.24 | drmessano | X-Lite with speakers and a mic? |
| 15:40.28 | [TK]D-Fender | freezey: odds are its acoustic echo caused by the mic picking up the speaker output |
| 15:40.28 | freezey | yeah |
| 15:40.32 | drmessano | Yeah.. |
| 15:40.36 | drmessano | Thats the problem |
| 15:40.52 | [TK]D-Fender | freezey: softphone+speaker+mic = suck |
| 15:40.57 | freezey | ahhh |
| 15:41.05 | freezey | [TK]D-Fender: work around? |
| 15:41.10 | drmessano | Headset |
| 15:41.11 | freezey | i am thinkin about just buying 2 ipphones |
| 15:41.13 | [TK]D-Fender | freezey: by real hardware |
| 15:41.15 | freezey | yeah |
| 15:41.18 | drmessano | or that too |
| 15:41.19 | freezey | thats what i was thinking |
| 15:41.28 | freezey | how about the cisco 7912's |
| 15:41.31 | freezey | they look pretty good |
| 15:41.42 | [TK]D-Fender | freezey: Crap |
| 15:41.48 | freezey | just for testing though? |
| 15:41.59 | [TK]D-Fender | freezey: Not even |
| 15:42.10 | freezey | any exact reasons why the softphone+speaker+mic would be crap? |
| 15:42.13 | [TK]D-Fender | freezey: What are you trying to accomplish in the long run? |
| 15:42.24 | drmessano | <[TK]D-Fender> freezey: odds are its acoustic echo caused by the mic picking up the speaker output |
| 15:42.32 | drmessano | That's why |
| 15:42.36 | [TK]D-Fender | freezey: Shitty mic, shitty EC, shitty speaker, being tied directly to a PC |
| 15:42.39 | freezey | [TK]D-Fender: eventually after done testing move everything over to VOIP and use asterisk to handle everything |
| 15:42.43 | L|NUX | can some one tell me how can i play mms:// when some one dial sip ? |
| 15:42.46 | eric2 | freezy, I've got a linksys spa942 and I'm happy with it (so far) |
| 15:43.00 | [TK]D-Fender | freezey: what is "everything". Please try to stop being so vague and give real details. |
| 15:43.05 | freezey | so pretty much your saying there is no delay its just the echo that screwing things up? |
| 15:43.22 | [TK]D-Fender | freezey: if you hear yourself coming back, its echo. |
| 15:43.36 | Qwell | unless it's sidetone |
| 15:43.40 | freezey | [TK]D-Fender: move over all the office people to voice over ip with asterisk handling it rather than using my old avaya system that is not compatible with asterisk |
| 15:43.48 | L|NUX | any one ? |
| 15:43.57 | [TK]D-Fender | Qwell: Oh, quite possible, but his scenario has the full friggen orchestra :p |
| 15:44.00 | drmessano | Often times causes by YOUR voice being picked up by the mic on the OTHER end as you're coming out the speakers on that end |
| 15:44.06 | drmessano | caused* |
| 15:44.22 | drmessano | Its a clusterf*** |
| 15:44.51 | [TK]D-Fender | freezey: So you plan on an * on your site, another on theirs? |
| 15:45.13 | [TK]D-Fender | freezey: and what kind of calls passing between the two? |
| 15:46.06 | freezey | [TK]D-Fender: i plan to remove the avaya system and basically throw it out in this office and remove manhattans phone sytem and replace it entirely with voip and have asterisk handle it... |
| 15:46.17 | DavisGr | I prefer Linksys SPA 9xx and Snom 3xx phones & no problems width echo! |
| 15:46.53 | [TK]D-Fender | freezey: and exactly what kind of lines at each site, and what kind of calls are you looking at passing over the internet? |
| 15:47.03 | freezey | [TK]D-Fender: so i will take my phone provider and let them know whats going on get the correct PRI from them... and then do the rest |
| 15:47.41 | freezey | [TK]D-Fender: well the calls are pretty much just business calls some confrences here and there... and when you say lines at each site? |
| 15:47.48 | [TK]D-Fender | Snom = not a cost effective / quality solution in most countries. Linksys = cost effective in just about everywhere that Polycom isn't. Polycom > ALL |
| 15:48.31 | freezey | yeah i heard snom and polycom where good |
| 15:48.39 | [TK]D-Fender | freezey: is the connection between these 2 boxes only for inter-office user calls? Or for providing PSTN access? |
| 15:49.14 | [TK]D-Fender | Snom is second rate and further down the "suggested list" |
| 15:49.32 | DavisGr | I had on client who have policom and there is problems - they are broking down the same ar granstreem ! |
| 15:50.01 | freezey | [TK]D-Fender: its going to be used for interoffice and making calls out recieving calls etc |
| 15:50.42 | drmessano | [Slashdot] Fifth Cable Cut, Iran Loses Net Connectivity <----- ut ro |
| 15:50.42 | [TK]D-Fender | freezey: so 1 site will not have any kind of actual phone lines in it at all? |
| 15:51.26 | freezey | [TK]D-Fender: well the faxes will have actual phone lines |
| 15:51.45 | L|NUX | did any one used mms_client with asterisk ? |
| 15:51.45 | BadHorsie | i'm curious if the best way to monitor a phone call would be to select a channel and use an originate Monitor so i can listen to it in another extension... |
| 15:51.56 | *** join/#asterisk Docfxit (n=none@cpe-72-129-118-135.socal.res.rr.com) |
| 15:52.07 | [TK]D-Fender | freezey: Ok, but no voice lines. So if you link is broken for any reason, they are stranded. |
| 15:52.12 | freezey | [TK]D-Fender: its pretty much like this... i want to put asterisk into my environment and you said at one point my avaya system is completely old.. which sucks cause i used to work on an avaya system with asterisk... so now i have to look into another solution |
| 15:53.00 | [TK]D-Fender | BadHorsie: "core show application chanspy" <- |
| 15:53.22 | DavisGr | freezey btw AVAYA U conected width h323 trunk or other way |
| 15:55.52 | freezey | DavisGr: i am not sure... |
| 15:56.00 | Docfxit | How can I change the root password? |
| 15:57.54 | ManxPower | Docfxit: Asterisk does not have a root password. |
| 15:58.05 | freezey | [TK]D-Fender: i got my cut sheet from my provider so answering some questions about the exact setup would be alot easier |
| 15:58.37 | [TK]D-Fender | Docfxit: log in as root, do "passwd" |
| 16:00.03 | asr33 | Docfxit: what version of Linux are you running? |
| 16:00.04 | freezey | [TK]D-Fender: unless i just get naother avaya system... i would rather do that |
| 16:00.36 | [TK]D-Fender | freezey: Ditch both, never look back. |
| 16:00.58 | [TK]D-Fender | asr33: "passwd" is pretty much *nix universal... |
| 16:01.37 | asr33 | ok I got ya.... |
| 16:02.02 | *** join/#asterisk _LoneCrow (n=ghfh@wdirect2.ADSL.mnsi.net) |
| 16:02.13 | freezey | [TK]D-Fender: so you say ditch both and continue to stick with VOIP |
| 16:02.20 | _LoneCrow | heya.. anyone know the default user/password for the Web-MeetMe Login? |
| 16:02.25 | DavisGr | freezey if U will make all on AVAYA that will cost a lot of $$$$ |
| 16:03.10 | drmessano | _LoneCrow: is that a trixbox? |
| 16:03.16 | _LoneCrow | yep |
| 16:03.33 | _LoneCrow | its built in.. .. maybe should go to #trixbox I just thought baout it |
| 16:03.56 | drmessano | Maybe need to ask on their forums.. its probably not standard |
| 16:06.27 | _LoneCrow | fresh trixbox install so I figured there would be docs for th at |
| 16:07.54 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 16:08.26 | [TK]D-Fender | freezey: its nto a question of "stick to VoIP", but rather stick to a solution that doesn't own your ass. |
| 16:08.40 | [TK]D-Fender | freezey: Right no each site has their own phone lines, right? |
| 16:09.10 | [TK]D-Fender | _LoneCrow: not here there isn't |
| 16:09.26 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 16:10.21 | drmessano | ~trixbox |
| 16:10.22 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 16:10.32 | *** join/#asterisk InsolentDreams (n=Insolent@62.157.211.194) |
| 16:11.22 | drmessano | Nothing they do is standard, and they've even started ripping out and changing pieces of FreePBX, so you won't even be able to rely on IT being standard in tb either |
| 16:12.19 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-82-154.d4.club-internet.fr) |
| 16:12.50 | InsolentDreams | Ok... so I have a dynamic queue, and I'm polling it's information about once every 5 seconds via a cli script to feed to queue members on a website to show them the status of the queue, members in the queue, etc. What I'm worried about is some slowdown from overdoing it here. Already I have 4-5 people simultaneously using this script, so techincally it _may_ be polled 5 times a second, then 5 seconds later, 5 more times in that second. These |
| 16:13.14 | InsolentDreams | I'm worried about performance... should I go down a different path and make a fifo daemon that I queue instead of asterisk like this? |
| 16:14.24 | InsolentDreams | As it is, when in the console with any debugging on it says Remote unix connection and disconnection around 1-2 times a second. :\ I have a feeling this is a bad idea to continue down this path... suggestions? |
| 16:16.14 | freezey | [TK]D-Fender: ok so i am thinkin this... maybe get to connections from my ISP so incase one goes down the other is a failover and if both go down just have some analog lines still coming in to back that shit up |
| 16:17.21 | drmessano | freezey, why not use analog lines anyway? |
| 16:17.52 | drmessano | Just because you have Asterisk, doesn't you you have to switch all your service over to a VoIP provider |
| 16:18.11 | drmessano | There are interface cards, or so I hear ;) |
| 16:18.18 | DavisGr | InsolentDreams if U giving once every 5sec info for website why are you not using website refreshing every 5sec for any conection - so if website is light so you can handle lotoff users |
| 16:18.25 | [TK]D-Fender | InsolentDreams: Make 1 process do all the lookups and create a static page for all the others to see |
| 16:18.35 | freezey | drmessano: yeah but TK was telling me that my avaya system is not compatible with asterisk its to old |
| 16:18.44 | drmessano | Yeah, ditch it |
| 16:19.04 | freezey | drmessano: so then where to go from there? |
| 16:19.16 | [TK]D-Fender | freezey: well I asked what kind of line ports you have on that system... |
| 16:19.31 | Qwell | InsolentDreams: use manager |
| 16:19.35 | InsolentDreams | Yeah, the users using this will get into the 30-40's when really deployed... and I don't think a simple refresh will do it. I looked into the PHPAGI and don't seem too impressed, and am considering writing a polling / collection daemon. |
| 16:19.37 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:19.37 | InsolentDreams | Manager? |
| 16:19.40 | Qwell | ~ami |
| 16:19.41 | jbot | ami is, like, the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 16:19.52 | DavisGr | freezey witch avaya U have Definity or som G6xx width S8xxx |
| 16:19.54 | Qwell | get your stats from there |
| 16:20.18 | freezey | [TK]D-Fender: ESF/B8ZS |
| 16:20.31 | [TK]D-Fender | freezey: You have a PRi on both sites? |
| 16:20.38 | freezey | [TK]D-Fender: yeah |
| 16:20.53 | InsolentDreams | Ahh Qwell, that might be what I should be doing it looks like. :) Nice. Thanks, and thanks for other comments all. |
| 16:21.05 | [TK]D-Fender | freezey: Then you could buy a T1 card for *, and use that to pass of calls to the other server. |
| 16:21.25 | [TK]D-Fender | freezey: And the remote one could remain with their Avaya |
| 16:21.31 | freezey | DavisGr: well my phones are 5410's i forgot what the othe hardware is i was about to go take a look again... ( i am new this site) |
| 16:21.37 | ManxPower | 5th fiber cut in 10 days. |
| 16:21.46 | [TK]D-Fender | freezey: so * will only be used as gateway from their PBX to yours. |
| 16:21.52 | drmessano | ManxPower: I beat you to it |
| 16:21.59 | drmessano | Took Iran down |
| 16:22.04 | *** part/#asterisk niekie (i=niek@bergnetworks.com) |
| 16:22.07 | coppice | ManxPower: these things become fashionable, and everyone wants one |
| 16:22.09 | [TK]D-Fender | drmessano: "mission accomplished" |
| 16:22.12 | ManxPower | drmessano: it isn't a contest 8-) |
| 16:22.18 | drmessano | Sure it is |
| 16:22.25 | freezey | [TK]D-Fender: which card do you recommend? |
| 16:22.32 | drmessano | The internet is a big contest, and I am gonna hit the end guy one day |
| 16:22.39 | Qwell | freezey: Digium T1 card, of course |
| 16:22.43 | [TK]D-Fender | 1st step of military invasion : cut off the enemy's ability to respond/communicate |
| 16:23.47 | coppice | How do you declare war in the 21st century? |
| 16:23.48 | coppice | Blow out all the oppositions early warning systems in one go :-) |
| 16:24.03 | [TK]D-Fender | freezey: Since you're going direct to VoIP I would suspect you wouldn't need HWEC, so a Sangoma A101 or Digium TE120P would do. |
| 16:24.07 | iratik | its odd that voicepulse doesn't support g729.... are g729 and speex unpopular among trunking providers? |
| 16:24.10 | drmessano | CIA |
| 16:24.14 | [TK]D-Fender | coppice: Wire cutters. |
| 16:24.32 | coppice | you need a saw for fibre |
| 16:24.34 | ManxPower | iratik: many providers that are NOT based on Asterisk support G729 |
| 16:24.34 | freezey | ok ok so... see this is what my old site was like... i had a t1 directly plugged into the back of asterisk |
| 16:24.51 | freezey | but i had a definity system over there |
| 16:24.52 | [TK]D-Fender | coppice: Sharks with friggen lasers on their heads |
| 16:24.59 | DavisGr | freezey U must look on "display system-parameters customer-options" and find out if You have h323 - so if You hav then find out asterisk.org cyril poust to configure asterisk width avaya |
| 16:25.01 | iratik | ManxPower: can you recommend a g729 or speex provider? |
| 16:25.21 | coppice | sharks don't attack the modern fibres. that was an 1980s thing |
| 16:25.25 | drmessano | [TK]D-Fender: Grizzly Bears with machines guns on their backs are much more effective |
| 16:25.34 | drmessano | -s |
| 16:26.07 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 16:26.08 | freezey | DavisGr: ok i am vnc'ing into my voicemail system now |
| 16:26.10 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 16:26.35 | drmessano | What's really scary is whether or not wackjob in Iran will see this as a threat, especially if it really is some coincidence |
| 16:26.47 | ManxPower | iratik: no speex, but try vitelity.net or teliax.com |
| 16:27.10 | DavisGr | freezey I am going Off now till tomorow |
| 16:27.18 | freezey | ok |
| 16:27.31 | freezey | DavisGr: i will talk to you when i see you tomorrow then |
| 16:27.35 | *** join/#asterisk abaci (n=IceChat7@ool-4b7fc532.static.optonline.net) |
| 16:28.15 | ManxPower | iratik: broadvoice.com |
| 16:28.24 | ManxPower | in fact every one I just checked have G729 |
| 16:28.27 | iratik | hmmm... odd that teliax doesn't seem to provide rates for US |
| 16:28.39 | ManxPower | iratik: 1.9 cents/min for per min plans |
| 16:28.59 | ManxPower | It might be less now. |
| 16:29.01 | InsolentDreams | broadvoice is great, if you can really do it on your own. Their customer service is terrible now though, just a fore warning. :) I use 'em though |
| 16:29.16 | ManxPower | I have personally used Teliax and Vitelity. |
| 16:29.21 | InsolentDreams | Just don't expect their help, come here instead |
| 16:29.22 | InsolentDreams | :P |
| 16:29.22 | freezey | [TK]D-Fender: ok let me try and make sense of this for myself... so i would have a t1 line directly plugging into the asterisk machine with this digium card.. this will come strait from the wall and asterisk will be my pbx and handle all the calls in my current office... and then a connection would go over to the other office they would keep their avaya system |
| 16:30.35 | [TK]D-Fender | freezey: your Avaya will be your PBX, all * will do is pass your external calls out the PRI over to your other * server to be actually terminated / processed |
| 16:31.10 | BadHorsie | i think i'm doing something wrong or several things not done in the right way, i monitor channels from a php that gets updated every 5 seconds, whenever i see an active channel i click to generate an originate call to a certain extension that is defined in extensions.conf that in the end points to chanspy with my extension so i can listen to it, now it takes a long time to start receiving audio even tho the call to my phone is done quickly, and most |
| 16:31.10 | BadHorsie | of the time the calls are too short so i get connected in the end of the call, is there a chance i can start the monitoring of the call first and then make the connection to my phone? maybe that would do it better... |
| 16:31.23 | freezey | [TK]D-Fender: hmm i but what if i wanted it to handle internally as well as externally |
| 16:31.26 | [TK]D-Fender | freezey: This is on the assumption that you will be using * as a pbx on the other side as their primary PBX. This machine should have a HWEC card |
| 16:31.48 | [TK]D-Fender | freezey: Fix your dialplan on your avaya to dial out your PRI for those DID's. |
| 16:32.09 | freezey | ahh yes very very true |
| 16:32.12 | InsolentDreams | Yash? Maxim? |
| 16:32.18 | ManxPower | PRI -> Asterisk -> T-1 or PRI -> Legacy PBX -> Legacy Phones. |
| 16:32.38 | freezey | [TK]D-Fender: i thought you were saying my avaya system is no good |
| 16:32.52 | InsolentDreams | BadHorsie: You sound eerily like someone at my work trying to do this. |
| 16:33.00 | ManxPower | freezey: the number one thing you MUST remember is that legacy PBXs were never designed to do anything USEFUL with regards to integrating with other systems. Don't expect the legacy box to do everything you want in regards to call routing. |
| 16:33.06 | InsolentDreams | BadHorsie: and having the same problems. :P |
| 16:33.07 | [TK]D-Fender | freezey: It isn't. What I was saying is that you can't have * run those phones and control your calls. |
| 16:33.32 | [TK]D-Fender | freezey: The setup I described doe not have * handle how your avay hardware processes its calls internally. |
| 16:33.33 | BadHorsie | InsolentDreams, huh... boss? :P |
| 16:34.06 | angryuser | how many concurrent call broadcom allow? |
| 16:34.09 | BadHorsie | InsolentDreams, well are you having any luck with developing a better solution? |
| 16:34.13 | angryuser | call's |
| 16:34.24 | [TK]D-Fender | freezey: angryuser Go ask them. |
| 16:34.30 | ManxPower | We actually have that setup except we also use Polycom phones hanging off Asterisk -- in fact all new or replaced phones are on the Asterisk box. |
| 16:34.31 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-6001939a172d9ba4) |
| 16:34.38 | angryuser | for one account <ManxPower> |
| 16:34.40 | freezey | [TK]D-Fender: hmm when i had an Avaya Definity PBX |
| 16:34.45 | InsolentDreams | BadHorsie: Well it's not me it's a colleague, and no they have similar problems. In fact when they monitor often they end up crashing our asterisk server. :) |
| 16:34.48 | ManxPower | angryuser: almost all providers allow unlimited calls on their per/min plan |
| 16:34.53 | freezey | [TK]D-Fender: asterisk did everything ha |
| 16:35.16 | freezey | just wanted to thank everybody again for the help your giving me |
| 16:35.23 | [TK]D-Fender | freezey: you saying that witht he definity * was in control of one internal phone talking to another? |
| 16:35.26 | InsolentDreams | BadHorsie: They have a bit older of a asterisk version, but crashing it outright scares the hell out of me. That is_no_ good |
| 16:35.30 | angryuser | <ManxPower> i saw thay have unlimited plans, do you know the limit? |
| 16:35.46 | [TK]D-Fender | angryuser: Go ask THEM. |
| 16:36.11 | angryuser | <[TK]D-Fender> it is not written on site |
| 16:36.22 | angryuser | <[TK]D-Fender> nor in terms & conditions |
| 16:36.23 | [TK]D-Fender | angryuser: They have a phone number. Get dialing. |
| 16:36.52 | [TK]D-Fender | "Hi, I'm thinking of becoming a customer. Answer my questions and you might win me over" |
| 16:37.08 | freezey | [TK]D-Fender: yeah actually a guy i set this up wrote up a documentation on it and presented it at a conference in las vegas its a powerpoint doc avaialble on the internet.. i dunno if you want to check it out what he did was pretty interesting |
| 16:37.17 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 16:37.28 | angryuser | <[TK]D-Fender> ?? calm down |
| 16:37.38 | [TK]D-Fender | freezey: Doesn't really impact your specific box though, so no matter |
| 16:37.44 | freezey | [TK]D-Fender: k |
| 16:37.58 | freezey | [TK]D-Fender: afk for a lil bit |
| 16:37.59 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 16:38.28 | coppice | remember. customers make pain days possible |
| 16:39.40 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:39.53 | hmmhesays | coppice: lol |
| 16:40.06 | hmmhesays | oh and mediatrix makes pain. |
| 16:41.04 | coppice | mediatrix has the worlds' funkiest protocol implementations |
| 16:42.01 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:42.11 | freezey | [TK]D-Fender: so if i wanted asterisk to handle everything i need a new avaya system... thats pretty much what i want to accomplish but its so dam pricey |
| 16:42.50 | freezey | [TK]D-Fender: and voip would be cool to have but if i have this system i would just like to improve the dam thing |
| 16:42.55 | hi365 | anyone know of any applications/products to create sefisticated IVR (data retrevial)? |
| 16:43.06 | ManxPower | freezey: make sure you have the worlds EXPERT on the Avaya handy as well. |
| 16:43.45 | freezey | ManxPower: just pisses me off its so dam old |
| 16:44.17 | [TK]D-Fender | freezey: ARGH |
| 16:44.32 | [TK]D-Fender | freezey: no NEW Avaya ANYTHING |
| 16:44.46 | [TK]D-Fender | freezey: I was showing a way to leave one you HAD in "service" |
| 16:44.47 | ManxPower | freezey: integrating different brands of PBXs is HARD. |
| 16:45.00 | ManxPower | Don't do it unless you have to. |
| 16:45.19 | freezey | [TK]D-Fender: oh yeah tahts right sorry.. but i am thinkin screw the other office let them be and just do this to mine |
| 16:45.58 | [TK]D-Fender | freezey: You are coming from so many direction its nigh impossible to get this story straight and advise you. |
| 16:46.23 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:46.37 | coppice | ManxPower: BlendTec can help :-) |
| 16:46.48 | *** join/#asterisk BugKhaM (i=BugKhaM@125-25-194-78.adsl.totbb.net) |
| 16:46.52 | freezey | [TK]D-Fender: ok so let set my goal on one thing for now.. cause you already explained how i can merge the two of them... i just want to move my current status in this one office with the use of asterisk handling all the calls incoming and outgoing |
| 16:47.11 | BugKhaM | anyone has an experience with using Linksys WIP300? |
| 16:47.31 | hmmhesays | yep |
| 16:47.32 | hmmhesays | stay away |
| 16:47.49 | [TK]D-Fender | freezey: You need to draw a complete picture of "now", and how you expect the new setup to function (without mentioning by what means you think it should work) |
| 16:48.02 | [TK]D-Fender | BugKhaM: Ditto |
| 16:48.02 | hmmhesays | short battery life, and it hangs a lot |
| 16:48.05 | sx|lappy | hmmhesays: that's pretty clear message ;-) |
| 16:48.21 | BugKhaM | it just doesn't get registered for more than 10 secs I guess |
| 16:48.22 | [TK]D-Fender | Indeed, battery doesn't last 4 hours with intermittant use |
| 16:48.23 | phillipk | BugKhaM: We use a couple of them here. They've been discontinued, I think. |
| 16:48.37 | hmmhesays | I use mine, but the charger is always near |
| 16:48.58 | [TK]D-Fender | I now have a UTStarcom F3000 that I'm RMA'ing for it locking up and losing link constantly |
| 16:49.09 | hmmhesays | I have yet to find a decent wifi phone |
| 16:49.10 | BugKhaM | phillipk: haven't you had registration problems with *? |
| 16:49.22 | file | [TK]D-Fender: have you crashed it yet? |
| 16:49.26 | hmmhesays | I just plugged my spa-942 back in yesterday, I like these phones |
| 16:49.35 | [TK]D-Fender | file: it crashes all by itself. Driver optional. |
| 16:49.43 | file | [TK]D-Fender: :D |
| 16:50.06 | drmessano | jbot: A wirenut is the solution to the problem of "Should I switch to Asterisk, should I keep my Avaya, should I keep my Cisco, or should I integrate them" |
| 16:50.06 | jbot | drmessano: okay |
| 16:50.10 | freezey | [TK]D-Fender: ok i am going to come up with a complete idea of what exactly is going on back there... maybe a little visio diagram.. and you might be so kind to analyze? =o) |
| 16:50.12 | drmessano | ~wirenut |
| 16:50.12 | jbot | wirenut is probably the solution to the problem of "Should I switch to Asterisk, should I keep my Avaya, should I keep my Cisco, or should I integrate them" |
| 16:50.13 | phillipk | BugKhaM: Not really. The main problem I have is that the phone seems to go "idle" and loose its connection to the AP at random. |
| 16:50.19 | [TK]D-Fender | freezey: 2 of them |
| 16:50.40 | BugKhaM | phillipk: I use WIP300 to connect to my * within the same subnet with qualify=yes |
| 16:50.40 | phillipk | BugKhaM: I find myself constantly tapping the little joystick to keep the phone awake. |
| 16:50.49 | freezey | [TK]D-Fender: ok so one with how exactly its doing back there and the other with how this should work |
| 16:50.50 | hmmhesays | phillipk, yeah |
| 16:51.20 | BugKhaM | phillipk: and sip show peers just say "Unreachable" all the time |
| 16:51.51 | phillipk | BugKhaM: We use 2 in a ring group for mobile tech support folks. We haven't had any trouble with registration. |
| 16:51.51 | BugKhaM | phillipk: heard a few other people said that too |
| 16:51.58 | [TK]D-Fender | freezey: 1 comprehensive picture of what you have NOW at both sites and all links between. #2 with what you KNOW you're going to do for a site you have solidified on, and show its impact on connectivity betweent he two and PSTN |
| 16:52.28 | BugKhaM | phillipk: ok |
| 16:52.37 | phillipk | BugKhaM: We do have trouble with roaming between APs, though. We have to power cycle the phone to get it to pick up a new AP. |
| 16:52.43 | freezey | [TK]D-Fender: gotcha when i complete this you will hear from me =o) |
| 16:53.08 | BugKhaM | phillipk: yeah, I have that roaming problem too |
| 16:53.10 | drmessano | Is that a threat or a promise? |
| 16:53.14 | drmessano | :) |
| 16:53.53 | phillipk | I'd really like to find a decent wifi phone, because the convenience factor for our tech support folks is huge, but the phones just don't seem to be any good. Has anyone tried the WIP320? |
| 16:54.06 | drmessano | Why not use an ATA and a cordless? |
| 16:54.16 | Qwell | phillipk: go with DECT... |
| 16:54.30 | Qwell | even SIP+DECT |
| 16:54.40 | phillipk | I'm in a campus style situation. Tech support has to go too far. |
| 16:54.40 | Qwell | sip base station, DECT handsets |
| 16:55.01 | Qwell | they'll have to charge the phones a few times a day with wifi... |
| 16:55.08 | drmessano | yep |
| 16:55.10 | Qwell | wifi phones have *very* poor battery life |
| 16:55.14 | phillipk | Yeah |
| 16:55.18 | drmessano | Wifi chips burn jucice |
| 16:55.20 | Qwell | especially if you're using it a lot, like tech support would be |
| 16:55.20 | drmessano | Wifi chips burn juice |
| 16:55.42 | [TK]D-Fender | phillipk: the Hitachi seems to be the most highly regarded one so far |
| 16:55.50 | Qwell | [TK]D-Fender: true |
| 16:56.02 | [TK]D-Fender | Qwell: Everything in perspective... |
| 16:56.02 | Qwell | I think twisted had one of the Hitachi's for a while |
| 16:56.26 | *** join/#asterisk CrashSys (n=kumba@t1.databalance.com) |
| 16:57.45 | *** join/#asterisk aiurea (n=aiurea@arcadia.timisoara.roedu.net) |
| 16:57.49 | phillipk | This one: http://www.voipsupply.com/product_info.php?manufacturers_id=49&products_id=1688 |
| 16:57.54 | aiurea | hi |
| 16:59.17 | aiurea | I have a setup in which asterisk calls me(via asterisk/outgoing), I hit a button and asterisk Dials another number and connects me to it. However if I(the first person called) end the call the person's after me call isn't terminated |
| 16:59.18 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:59.37 | aiurea | confusing, right:)? |
| 17:00.31 | [TK]D-Fender | phillipk: I've heard more on the IP-5000. Not sure how the IP-3000 measures up. |
| 17:00.43 | aiurea | http://pastebin.com/m202a0933 this is a part of extensions.conf |
| 17:01.05 | phillipk | [TK]D-Fender: Thanks. I'll look for the 5000. |
| 17:01.25 | ManxPower | I'm getting REALLY REALLY REALLY sick of support departments that give you the WRONG info, send the wrong attachment, or just plain take 4 days to respond. |
| 17:02.38 | BadHorsie | i wonder if instead of having Application: chanspy call in an originate in the AMI would make any significant difference to having AMI trigger extension.conf rule for monitoring and calling chanspy inside of it... |
| 17:02.50 | [TK]D-Fender | jameswf: No, he prefers others... |
| 17:03.23 | jameswf | I wa sgoing to go bust some heads... guess not.. |
| 17:04.16 | drmessano | jameswf: I have an issue with your support |
| 17:04.33 | jameswf | drmessano: your opinion doesnt count |
| 17:04.34 | jameswf | :) |
| 17:04.35 | drmessano | I had a card that wasn't working.. why didn't you people know enough to CALL ME? |
| 17:04.52 | drmessano | I waited hours :( |
| 17:04.53 | jameswf | we would have called you but your card wasnt working |
| 17:04.58 | [TK]D-Fender | :p |
| 17:05.01 | ManxPower | drmessano: we frequently get e-mails sent to our support desk telling us that e-mail is not working |
| 17:05.04 | drmessano | Ohhhhhhhhh I see |
| 17:05.10 | iratik | is the default maxexpiry=3600? |
| 17:05.13 | [TK]D-Fender | ManxPower: SMRT |
| 17:05.16 | drmessano | Is that like the e-mail microsoft sent me to reactivate my outlook? |
| 17:05.24 | drmessano | bastards |
| 17:05.58 | *** join/#asterisk sx|lappy (n=sxpert@lgit-1225.obs.ujf-grenoble.fr) |
| 17:06.17 | drmessano | When we have a WAN outage, and someone calls me, every now and I then I tell someone "Im looking into it, please send me an email on that" |
| 17:06.30 | drmessano | Only the stupid ones though |
| 17:06.41 | drmessano | It's like a sick game I play, that only I have the chance to win |
| 17:06.46 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 17:07.10 | *** join/#asterisk sudhir492 (n=sudhir@adsl-146-123-10.mco.bellsouth.net) |
| 17:07.11 | jameswf | Sorry all IT issues must go through the web based ticket system.... But the networks down... NO EXCPTIONS!! |
| 17:07.27 | drmessano | TICKET OR IT DIDNT HAPPEN! |
| 17:07.43 | [TK]D-Fender | "We have a Zero Intelligence policy here!" |
| 17:07.47 | sudhir492 | Is it possible to use asterisk with VPN |
| 17:07.52 | Qwell | sudhir492: sure |
| 17:08.08 | jameswf | when our network goes down everyone wallpapers the IT guys door with postits |
| 17:08.17 | drmessano | lol |
| 17:08.35 | ManxPower | our entire network almost never goes down. |
| 17:08.40 | jameswf | he hates us :)) |
| 17:09.06 | sudhir492 | The Asterisk is on a public IP address where there is no restriction. However, the client's phone is in a place where the ISP is blocking all SIP traffic. What is the best way to setup things |
| 17:09.23 | jameswf | Our network never goes down except the few times that it goes down but other than when it is down it is always up |
| 17:09.27 | iratik | What is going here - I can't issue an originate command and am getting an odd error back - let me know if you need more info - http://pastie.caboo.se/148279 (trunk works, but issuing originate through AMI suddenly stopped) - asterisk 1.2 on slackware |
| 17:09.39 | drmessano | We had 3 sites on one connection into our corporate cloud.. It was mess whenever the hub site, highly prone to lightning problems, would get taken out |
| 17:10.04 | drmessano | Its only slightly better now, with one builing on its own circuit |
| 17:10.09 | drmessano | building |
| 17:10.17 | sudhir492 | Qwell: have you set up something like that |
| 17:10.22 | drmessano | But nights like lastnight where all of the southeast was out... sucked |
| 17:10.25 | Qwell | sudhir492: there's nothing to setup |
| 17:10.32 | Qwell | just setup the vpn like normal, and you're done |
| 17:11.18 | sudhir492 | Qwell: The asterisk box is also accessed by phones which do not use any vpn, |
| 17:11.44 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:13.42 | iratik | hmm congested.? |
| 17:16.09 | drmessano | No, runny nose |
| 17:16.28 | drmessano | But thank you for asking |
| 17:19.51 | jblack | holy fuck. |
| 17:19.51 | aiurea | if I have a small macro on a loop, if I terminate the call shouldn't it stop running? |
| 17:20.03 | jblack | a _fifth_ fiber cable was cut |
| 17:20.35 | jblack | Iran is now offline |
| 17:20.39 | Qwell | jblack: welcome to 5 hours ago |
| 17:20.41 | Qwell | then again 3 hours, and 1 hour |
| 17:21.16 | jblack | I think I can be forgiven for being 5 hours out of sync with the news. |
| 17:22.08 | *** join/#asterisk cpjosh (n=josh@cp120.cardplayer.com) |
| 17:23.04 | *** join/#asterisk smash- (n=smash@c-71-59-163-135.hsd1.wa.comcast.net) |
| 17:23.06 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:23.11 | generalhan | hey all ! |
| 17:24.02 | cpjosh | Morning all, im having a problem getting my context's to work properly.... |
| 17:24.05 | cpjosh | <PROTECTED> |
| 17:24.05 | cpjosh | 0/17, span 1 |
| 17:24.10 | cpjosh | i keep getting that error.. |
| 17:24.17 | jblack | The news is only 2 hours old |
| 17:24.23 | smash- | how do i setup my asterisk box |
| 17:24.36 | smash- | so i can have another asterisk box uplink to it through voip, and use the pri channels |
| 17:24.49 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 17:25.40 | generalhan | we are opening up a remote office, with 8 users. we want these 8 users to connect their SIP phones through another * server and connect that box to our local * box via IAX2 ... anyone have any adice on the bandwidth that i should get for this remote office to ensure good qaulity on thier phones ?? |
| 17:27.01 | generalhan | in what little experience i have, i've found that a general consumer cable connection works well for one phone, but calls start to get rather choppy with 2. and the ISP there is offering me a 15M up 2M down stream on a cable connection, but i fear that wont be enough |
| 17:28.35 | cpjosh | generalhan: use a sip phone and just record how much bandwidth your really using.. |
| 17:29.13 | generalhan | cpjosh: i dont really have that option ... this office has no connection, and i dont want to sign a year lease for a testing phase |
| 17:30.35 | generalhan | and in our local office we have a T connection, which isnt really comparable. i just want to know if anyone has used a cable connection successfully in a situation like this, and at what speeds they got it to work |
| 17:32.48 | [TK]D-Fender | cpjosh: Clearly you do not have an exten in your [incoming] context to match that DID. |
| 17:33.18 | *** join/#asterisk Jake[work] (n=jakep@pool-96-227-17-56.phlapa.east.verizon.net) |
| 17:34.16 | cpjosh | [TK]D-Fender: Im assuming this context is in the extensions conf or in the zaptel conf? |
| 17:34.31 | [TK]D-Fender | cpjosh: Extensions are in extensions.conf |
| 17:34.40 | cpjosh | ok thanks |
| 17:34.43 | [TK]D-Fender | one would think the name sorta said ti all |
| 17:35.39 | *** join/#asterisk ming_zy1 (n=ming_zym@123.103.29.252) |
| 17:35.45 | cpjosh | yeah it didnt, but thank god your around to help us figure it out |
| 17:35.53 | drmessano | This reminds me of a parody someone did years ago... "Bomb bomb bomb, bomb bomb Iran..." |
| 17:36.05 | cpjosh | i think it should be put_in_your_extensions_here.conf |
| 17:36.58 | *** part/#asterisk cpjosh (n=josh@cp120.cardplayer.com) |
| 17:39.38 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 17:40.32 | bsdwarrior | are voip phones drift time after several days,weeks. ntp is running on the server, I believe I have to add option ntp-servers to our dhcpd.conf. Am I correct? |
| 17:42.16 | [TK]D-Fender | bsdwarrior: depends where your provisioning tells them to get their time from and what resync period |
| 17:44.07 | bsdwarrior | tkd-fender when you say provisioning, do you mean in the phones or in asterisk ? |
| 17:45.11 | [TK]D-Fender | bsdwarrior: Only one "provisions" |
| 17:46.12 | bsdwarrior | so its in the phones ? |
| 17:46.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:46.45 | aiurea | it weird, I have an identical sip.conf on an asterisk 1.2 and 1.4. 1.2 detects a hangup, 1.4 doesn't |
| 17:46.48 | aiurea | it's* |
| 17:47.03 | [TK]D-Fender | bsdwarrior: Yes |
| 17:47.22 | [TK]D-Fender | aiurea: SIP does not "detect" hangups. It is an absolute |
| 17:47.42 | [TK]D-Fender | aiurea: hangup "detection" is an analog thing |
| 17:48.17 | aiurea | [TK]D-Fender, what could be different that would affect it? I am calling the same phone, 1.2 sees the hangup and exits, 1.4 continues |
| 17:48.36 | [TK]D-Fender | aiurea: show me something sueful and describe your scenario |
| 17:48.52 | aiurea | [TK]D-Fender, just a sec while I prepare it |
| 17:49.29 | VitoCorleon | I have a Cisco 7960 setup to a Asterisk box. Incoming works great but when i dial out i get "Reorder", any help please? |
| 17:49.38 | bsdwarrior | wierd I have the sntp_server set in the sipblah.cnf |
| 17:49.52 | Qwell | VitoCorleon: try repeating your question again - I don't think anybody saw it the first 4 times |
| 17:50.03 | drmessano | Effin Spammer |
| 17:50.35 | [TK]D-Fender | VitoCorleon: and I asked you for details CLI output with SIP debug enabled yesterday and you're revolving like a broken record. |
| 17:50.54 | VitoCorleon | did you? |
| 17:50.59 | drmessano | .............click.............click.............click.............click.............click.............click |
| 17:51.05 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:51.06 | VitoCorleon | i didnt type that lol sorry |
| 17:51.17 | VitoCorleon | im not sure why my xchat repeated itself |
| 17:51.20 | [TK]D-Fender | VitoCorleon: Yes, not if you want help, give us something we can help you based on |
| 17:51.25 | drmessano | Up arrow, enter |
| 17:51.26 | drmessano | Up arrow, enter |
| 17:51.29 | [TK]D-Fender | VitoCorleon: pastebin is your friend. |
| 17:52.24 | VitoCorleon | yep yep :) |
| 17:52.33 | VitoCorleon | im going over to my client again today |
| 17:52.42 | VitoCorleon | i will be back in here :) |
| 17:53.31 | *** join/#asterisk freezey (n=freezey@gw.mypublisher.com) |
| 17:53.49 | *** join/#asterisk PDani (n=pdani@dsl51B6FDFB.fixip.t-online.hu) |
| 17:53.50 | PDani | hi |
| 17:54.06 | VitoCorleon | hi |
| 17:55.32 | ZPertee | I have an overhead pager system and a cell phone. I would like to be able to dial a number if I'm in the office so that incoming calls will page me and if I leave dial another extension and have calls forwarded to my cell phone. this needs to happen for multiple people in my office. Any ideas? |
| 17:56.03 | VitoCorleon | *87 |
| 17:56.14 | aiurea | [TK]D-Fender, http://pastebin.com/m4c77e8d5 |
| 17:56.21 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:56.38 | bsdwarrior | tkd-fender if the phones are behind a firewall on pvt network, I doubt they could sync time with a ntp server with a public ip right ? |
| 17:57.19 | [TK]D-Fender | ZPertee: All dialplan stuff. "core show application gotoif", " "core show function DB" |
| 17:57.29 | [TK]D-Fender | bsdwarrior: Sure they can |
| 17:58.05 | bsdwarrior | tkd-fender wierd, cause its not working. |
| 17:58.57 | [TK]D-Fender | aiurea: SIP/smartcall1-081ceaf0 <- what is this coming from precisely? |
| 17:59.48 | aiurea | asterisk is calling me (asterisk/outgoing) |
| 18:00.52 | *** join/#asterisk c4t3l (n=c4t3l@74.95.210.124) |
| 18:02.06 | PDani | how can i give empty string as parameter to a command called from System()? |
| 18:02.15 | c4t3l | hello guys, are there any polycom gurus in here. I'm having trouble with ver 2.1.1 and asterisk sending callerid to the line apearances |
| 18:03.41 | ManxPower | c4t3l: make sure the callerid contains nothing except for numbers and letters. no " not other special chars. |
| 18:03.54 | ManxPower | PDani: "" |
| 18:03.58 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 18:04.29 | walhala | may some one can explain me this message : "Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from xxx.xxx.xxx.xxx" ? |
| 18:07.16 | c4t3l | ManxPower: the specific problem I have is that older versions of the SIp firmware (ie 1.6) would display inbound callerID on the line appearance. The newer version 2.1.1 displays caller ID on the very bottom. do you know of a setting in sip.cfg that might help |
| 18:07.34 | *** join/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 18:08.37 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 18:08.56 | *** join/#asterisk AndyGraybeal (n=mind@casanueva.wifi.frognet.net) |
| 18:09.55 | [TK]D-Fender | aiurea: that does not tell me anything useful about the device |
| 18:10.34 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 18:10.59 | aiurea | [TK]D-Fender, I don't know what else to share, everything is routed via SIP to the provider |
| 18:11.06 | hmmhesays | ami callerid stuff is completely screwy |
| 18:11.37 | aiurea | [TK]D-Fender, there are not physical telephony devices used in this setup |
| 18:11.45 | aiurea | [TK]D-Fender, except the phone:) |
| 18:12.08 | [TK]D-Fender | aiurea: What is the device making that channel I asked about? |
| 18:12.24 | *** join/#asterisk atop (n=a@82-33-155-212.cable.ubr04.wiga.blueyonder.co.uk) |
| 18:13.28 | aiurea | [TK]D-Fender, it's a sip connection to a remote provider. I don't know what they are using. and the device to which it's going is a mobile phone |
| 18:14.03 | [TK]D-Fender | aiurea: This is the part where the suspicion of failure immediately falls on them |
| 18:14.24 | aiurea | [TK]D-Fender, yes, but why would asterisk 1.2 work in the exact setup? |
| 18:14.40 | [TK]D-Fender | aiurea: Show 2 calls with SIP debug enabled through each |
| 18:14.46 | aiurea | ok |
| 18:18.04 | PDani | ManxPower: doesn't work |
| 18:18.11 | bsdwarrior | tkd-fender im still stumped on something we talked about. im running a perl daemon that sends outbound calls to phones every 20secs (if any). my problem is I cant find out what happened to the call properly. you said I have to set the userfield for cdr in the dialplan, howerver im running a script. |
| 18:20.16 | PDani | ManxPower: if "" is the last parameter |
| 18:23.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:28.05 | *** join/#asterisk Maxous (n=Maxous@mobile-166-214-251-223.mycingular.net) |
| 18:36.42 | [TK]D-Fender | bsdwarrior: Set the userfield before dialing out the first lefg of your call. |
| 18:37.51 | aiurea | [TK]D-Fender, sorry for the delay, I screwed up something in the 1.4 setup and I haven't figured out exactly what |
| 18:39.11 | bsdwarrior | tkd-fender, so set a channel variable ? |
| 18:39.26 | [TK]D-Fender | bsdwarrior: No, set the CDR USERFIELD. |
| 18:40.11 | [TK]D-Fender | bsdwarrior: Setting a variable for tracking live, but you're going to want that var to match CRDisn't a bad idea either. |
| 18:40.15 | [TK]D-Fender | CDR* |
| 18:40.42 | *** join/#asterisk guillote_GNU (n=guillote@host63.201-253-22.telecom.net.ar) |
| 18:42.22 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 18:43.01 | bsdwarrior | tkd-fender ill give it another shot |
| 18:45.51 | ZaVoid | anyone know if g.729 license on asterisk supports g.729br8 ? |
| 18:48.24 | lmadsen | it's G.729a |
| 18:48.42 | ZaVoid | yeah just g.729a though right? |
| 18:52.31 | lmadsen | that's what I just said :) |
| 18:52.46 | lmadsen | I dind't say g.729a/b/ab/foo |
| 18:55.07 | aiurea | [TK]D-Fender, how should I send you the logs? |
| 18:55.41 | [TK]D-Fender | aiurea: Pastebin them and maybe someone else can follow up, because I ahev to head out for a few hours |
| 18:55.59 | aiurea | damn |
| 18:56.05 | aiurea | just when I finished them |
| 18:56.49 | [TK]D-Fender | aiurea: well it took you over 3/4 hour to do this so you'd better not waste the effort |
| 18:57.54 | aiurea | there was a nasty permission bug I managed to introduce while testing it:) http://pastebin.ca/894152 this is 1.2 |
| 18:57.57 | iratik | Can asterisk management interface support more than one person logged in under the same account? |
| 18:58.26 | aiurea | http://pastebin.ca/894155 this is 1.4 |
| 18:59.17 | [TK]D-Fender | iratik: Yes |
| 18:59.29 | [TK]D-Fender | ok, I'm off, BBL |
| 18:59.29 | iratik | hmm.. k thanks |
| 19:04.35 | cappiz | is there a way to make the voicemail announce a different number rather than the extension itself? |
| 19:05.51 | *** join/#asterisk tsabi (n=tsabi@gw.creditexpress.hu) |
| 19:07.29 | *** part/#asterisk Maxous (n=Maxous@mobile-166-214-251-223.mycingular.net) |
| 19:08.37 | *** join/#asterisk drmessano-LT (n=nonya@207.230.140.240) |
| 19:13.03 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 19:14.34 | adelas | anyone know of a simple program to convert files to .gsm files so i can replace the default asterisk greetings? |
| 19:15.19 | eric_hill | Just set up a voicemail box that you can record some messages into. Asterisk will save the VM in the gsm format for you. |
| 19:17.05 | hmmhesays | sox |
| 19:17.36 | *** join/#asterisk RoyK (n=roy@ip-197-29-149-91.dialup.ice.no) |
| 19:19.57 | *** join/#asterisk saftsack (n=saftsack@p4FC7460F.dip.t-dialin.net) |
| 19:24.52 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 19:30.24 | *** join/#asterisk theNOTO (n=biggs@cpe-75-179-187-170.woh.res.rr.com) |
| 19:30.35 | *** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 19:31.03 | shido6 | asterisk rocks |
| 19:32.36 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:35.38 | generalhan | anyone have any suggestions on upload/download bandwidth for a 10 SIP phone remote office connecting via IAX2 to the local Asterisk machine ? |
| 19:36.54 | BBHoss | generalhan, so im assuming you have two asterisk boxes? |
| 19:37.23 | generalhan | yes, one local connected to a PRI, and one that i am sending to the remote office to connect to the local one via IAX2 |
| 19:37.27 | *** join/#asterisk ctdam (n=ctdam@96.56.45.14) |
| 19:37.42 | BBHoss | what kind of codec? |
| 19:37.56 | ctdam | Anyone interested in a full time Asterisk-related position in the NYC area, please PM me |
| 19:38.03 | generalhan | but i need to setup the ISP in that building and i dont want to go with a cable connection and have it be a crappy connection |
| 19:38.48 | generalhan | BBHoss: i forget what im using .. one sec lol |
| 19:39.56 | generalhan | BBHoss: on my local machine here i allow gsm, ulaw, alaw ... in that order |
| 19:39.59 | J4k3 | generalhan: 160kbit * 10 |
| 19:40.03 | J4k3 | generalhan: for g711 |
| 19:40.15 | J4k3 | (or 80k per direction) |
| 19:40.20 | generalhan | hmm |
| 19:40.21 | BBHoss | generalhan, i would get g729 |
| 19:40.24 | J4k3 | figure 32k-ish per direction per call for g729 |
| 19:40.57 | BBHoss | an SDSL connection would handle the g729 calls |
| 19:41.21 | generalhan | that seems off to me ... maybe i need to do some adjusting ... when i was testing this server at my house on a 2M cable connection 1 phone worked well, but 2 phones started chopping up really baddly |
| 19:41.34 | BBHoss | J4k3, it can work though, i get like 0.3 ms jitter on my Comcast |
| 19:41.41 | xp_prg | anyone use perl with asterisk here? |
| 19:41.50 | generalhan | BBHoss: but that is for only one phone, right |
| 19:42.17 | BBHoss | generalhan, no, an SDSL connection should handle 10 g729 calls at least |
| 19:42.51 | J4k3 | BBHoss: you're in a very rare location |
| 19:43.00 | BBHoss | J4k3, you better believe it |
| 19:43.04 | generalhan | i would love to put in a full T-1 line at this office ... but it would be a difference of like $400/month over the 12M cabel connection |
| 19:43.13 | J4k3 | BBHoss: all it takes is one jackass kid with deepish-pockets parents and a giganews account and/or bittorrent to ruin that |
| 19:43.18 | BBHoss | i consistently get 38ms ping |
| 19:44.20 | drmessano-LT | I've run 4 calls, bidirectional, on a 8/768 Comcast line before |
| 19:44.22 | BBHoss | generalhan, DSL of any type is less susceptible to packet loss and jitter because your neighbors usage doesn't effect it as much as cable |
| 19:44.30 | tuxfoo | anyone try to get bluetooth working with asterisk 1.4? |
| 19:44.33 | drmessano-LT | 711 |
| 19:44.43 | drmessano-LT | After that it got bad |
| 19:45.07 | J4k3 | then I moved a few feet and lost 7/8ths of my bandwidth :) |
| 19:45.09 | generalhan | BBHoss: this office is in BFE though ... no where NEAR a DSL location |
| 19:45.09 | BBHoss | wow, EVDO? |
| 19:45.16 | drmessano-LT | So you should be able to handle more G729 than that |
| 19:45.21 | BBHoss | BFE? |
| 19:45.27 | drmessano-LT | LOL |
| 19:45.31 | drmessano-LT | ~bfe |
| 19:45.32 | jbot | [bfe] B F-ing Egypt...or like nowhere |
| 19:45.36 | drmessano-LT | Yes |
| 19:45.37 | generalhan | lol |
| 19:45.43 | J4k3 | it ain't bfe if you can get cable |
| 19:45.44 | J4k3 | I'm in bfe |
| 19:46.00 | J4k3 | the last F connector is 2.9 miles west, or 40+ miles east |
| 19:46.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:46.13 | J4k3 | 10 miles south, and 23 miles north |
| 19:46.28 | J4k3 | DSL... well the CO has it, but I'm 28kft from the CO |
| 19:46.32 | J4k3 | so haha, riiiight ;) |
| 19:47.38 | generalhan | so if i could get a 20M download and 3M upload on a cable connection for the same price as a 1.544 T1 connection, which would you recommend to run these phones |
| 19:47.50 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 19:47.55 | J4k3 | generalhan: depends on what you want |
| 19:48.02 | BBHoss | J4k3, yeah but they use fiber to the fiber termination box, then run it into a DSLAM |
| 19:48.02 | J4k3 | cable generally has at least a 3 day 'wait for repair' time |
| 19:48.35 | generalhan | im only looking for call reliability for the IAX2 transfer to my local machine ... that is the only thing i am concerned with |
| 19:48.55 | BBHoss | we're telling you DSL |
| 19:48.56 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 19:49.13 | generalhan | BBHoss: that isnt an option in the location of this office |
| 19:49.21 | BBHoss | i wouldnt touch cable without an SLA |
| 19:49.35 | BBHoss | generalhan, so you've checked then? |
| 19:49.36 | generalhan | BBHoss: i can either get Cox Cable out there or a T1 |
| 19:49.40 | generalhan | BBHoss: yes |
| 19:49.59 | BBHoss | you can try it |
| 19:50.41 | bsdwarrior | tkd-fender in my perl script Im sending this command to the manager before I make the call and it doesnt work. Set(CDR(userfield)=1234) |
| 19:50.49 | generalhan | BBHoss: see i REALLY dont want to try something ... i have to sign a year contract with any of these companies ... i even asked if i could do a trial period for assesment and was denied |
| 19:51.25 | BBHoss | generalhan, just get the business class service, if its not business class, then don't pay the bill |
| 19:52.35 | BBHoss | generalhan, g.729 has got a robust PLC algorithm, and IAX2 has a jitterbuffer you can use |
| 19:52.44 | BBHoss | i would definitely use that |
| 19:52.48 | BBHoss | or ILBC |
| 19:53.02 | BBHoss | but ILBC is not as efficient as g729 |
| 19:53.19 | generalhan | hmm, never played with the jitterbuffer before |
| 19:53.30 | BBHoss | it can help loads |
| 19:53.45 | BBHoss | i have a friend is au that was having trouble connecting over iax2 to the us |
| 19:54.07 | BBHoss | sound quality was horrible until he turned the jb=on |
| 19:55.19 | generalhan | do thats it ? just jitterbuffer=yes in iax.conf ? |
| 19:55.30 | generalhan | no manual configuration ? |
| 19:55.57 | BBHoss | yeah i think that was it |
| 19:56.05 | generalhan | sweet ill try that ! |
| 19:56.08 | Frogzoo | can't you tune the size of the introduced delay? |
| 19:56.18 | BBHoss | just make sure you disable the phones jb |
| 19:56.50 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 19:56.51 | anarcat | hello |
| 19:56.53 | mintee | I'm looking to have a voicemail box that must be setup before it's working by the user... The first time someone calls the number, they will get the intro with instructions to setup their voicemail, including recording their away (sorry) message and changing their PIN. After the voicemail has been setup, the user will never hear the intro message again. |
| 19:56.56 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:57.11 | mintee | is that something that's concievably easy to setup? |
| 19:57.12 | BBHoss | Frogzoo, yes, maxjitterbuffer=n (number) |
| 19:57.26 | Frogzoo | nice |
| 19:57.31 | anarcat | so i'm still pondering on buying a Polycom station for our conference room and i've been impressed by a cisco phone that had a speakerphone... |
| 19:57.52 | anarcat | has anyone here played with softphones with speakerphones that work well? |
| 19:58.00 | anarcat | (sorry for being offtopic...) |
| 20:01.03 | RoyK | is slin wideband? |
| 20:03.11 | BBHoss | RoyK, i don't think so, but i read somewhere that Asterisk can run up to 16k internally, so maybe it is :) |
| 20:03.53 | *** join/#asterisk ctooley (n=ctooley@doc-72-47-33-80.maryville.mo.cebridge.net) |
| 20:08.07 | annielou | I am trying to automate some calls from asterisk extensions - I've successfully gotten extension to extension calls going through /var/spool/asterisk/outgoing/ - however, I'm interested in having a particular extension auto-dial some other digits which are not necessarily extensions (e.g. *72/*73 for traditional PSTN Call Forwarding, etc.). Anyone have suggestions on how this can be accomplished? |
| 20:08.54 | xp_prg | annielou so you want to dial calls out? |
| 20:08.54 | sudhir492 | what are the softphones for windows that support iax2? |
| 20:09.14 | BBHoss | sudhir492, zoiper does i think |
| 20:09.54 | sudhir492 | thanks, let me try |
| 20:10.03 | annielou | xp_prg: yes, dialing out to a VOIP service provide which supports *72 to set CF, so just want to automate calls to enable CF. |
| 20:10.50 | xp_prg | annielou can you tell me an example, I am confused |
| 20:11.05 | xp_prg | you want to call forward an extension to another phone number yes? |
| 20:13.44 | annielou | xp_prg: sure. take example of schedule CF - the user sets up a schedule through some web interface. at the start time of the scheduled event, i need asterisk to dial *72+NPANXXXXXX+# so that CF is enabled. At the end time of the event, i want asterisk to dial *72. i want to forward calls, not via asterisk logic, but via the existing logic for *72 and *73 which is implemented by the VOIP service provider. |
| 20:14.34 | defsdoor | annielou: use a prefix |
| 20:16.46 | *** join/#asterisk drako (n=thinkpad@nelug/coreteam/luisjose) |
| 20:16.52 | drako | whats the trick with ooh323 |
| 20:17.11 | drako | i compiled, its on modules but on help does not show anything about it |
| 20:19.06 | *** join/#asterisk sponger (n=sean@cf.kokuawireless.com) |
| 20:21.04 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 20:21.47 | annielou | defsdoor: thanks, i haven't heard of prefix - will read up on it. let me know if you have any examples. |
| 20:22.20 | defsdoor | annielou: dial 9*72 |
| 20:22.39 | defsdoor | remove the 9 in your dial plan |
| 20:28.44 | *** join/#asterisk smash- (n=smash@c-71-59-163-135.hsd1.wa.comcast.net) |
| 20:28.56 | smash- | Hey could someone give me a url or a couple of voip/sip uplink providers. |
| 20:29.28 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 20:29.44 | sponger | teliax.com |
| 20:29.53 | sponger | binfone.com |
| 20:30.02 | sponger | voipjet.com |
| 20:30.35 | ManxPower | ~itsp |
| 20:30.36 | 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 $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 20:30.45 | ManxPower | ~istlist-usa |
| 20:30.53 | ManxPower | ~isplist-us |
| 20:31.05 | ManxPower | *sugh* |
| 20:31.06 | ManxPower | ~itsplist-us |
| 20:31.07 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com |
| 20:31.23 | ManxPower | I use vitelity.net as well |
| 20:33.28 | kyron | ~itsplist-ca |
| 20:33.29 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 20:33.48 | kyron | ah, interesting |
| 20:34.25 | tzanger | what's the thought on thinktel? |
| 20:34.29 | tzanger | (thinktel.ca) |
| 20:36.35 | fiXXXerMet | How else can I stress test conferencing without getting 15 real people to call up? |
| 20:36.54 | annielou | defsdoor: i don't have a 9 in my dial plan. but your comment did help me to think of the solution to the problem, so thanks. |
| 20:38.58 | mintee | can anyone link me to a few samples of conditionals in asterisk? |
| 20:39.28 | mintee | exten => s,1,IF(system(touch helloworld)?ok:nok) isn't working for me it seems, and I don't know why. |
| 20:39.42 | tzanger | mintee: GotoIf($[ ${LEN(CALLERID(num))} = 10 ]?goodcid) |
| 20:40.30 | tzanger | mintee: GotoIf($[ ${MYVAR} < 3 ]?newcontext,someexten,1:greater_or_equal) |
| 20:41.05 | bsdwarrior | Im using Orginate with the manager and I can't set the userfield in cdr for the life of me |
| 20:42.07 | bsdwarrior | Variable: userfield=test doesnt work. nor does SetCDRUserField(1234) or Set(CDR(userfield)=1234) |
| 20:42.20 | mintee | tzanger, thanks. I'm looking to actually check if a file exist on the filesystem when a specific call comes in, and it it doesn't then create the file and do something else. |
| 20:42.47 | sponger | mintee: why not just write a simple agi to do that for you |
| 20:43.21 | tzanger | sponger: because agis suck? :-) |
| 20:43.34 | sponger | haha ok |
| 20:43.48 | mintee | because I don't know what an agi is... reading now... |
| 20:43.59 | mintee | and tzanger thinks they suck so there :P |
| 20:44.05 | sponger | tzanger: can you elaborate? |
| 20:44.09 | tzanger | mintee: try "core show function STAT" |
| 20:44.30 | RoyK | wtf is 'show translation' become in 1.6? |
| 20:44.36 | tzanger | sponger: I don't want to have a dozen little shell scripts hanging around waiting to get invoked |
| 20:44.45 | RoyK | ~translation |
| 20:44.51 | RoyK | ~lart himself |
| 20:44.51 | jbot | cuts himself into thin stripes |
| 20:44.53 | mintee | tzanger i don't have stat... working with an older version of asterisk |
| 20:46.09 | mintee | whenever i try and use IF i get No application 'IF' for extension (test-play-with-conditional, s, 1) |
| 20:46.23 | kyron | LOL |
| 20:46.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:46.48 | RoyK | mintee: it's a function, not an application |
| 20:48.32 | mintee | ah, so it has to be used inside an application? |
| 20:49.26 | *** join/#asterisk GrumpManAtWork (n=meanderi@pool-72-78-128-159.phlapa.east.verizon.net) |
| 20:50.01 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-72-48.tor.primus.ca) |
| 20:51.39 | mintee | ok, i changed the line to exten => s,1,set(myvar=${IF(system(touch /var/log/test-play-with-conditional.touched)?ok:nok)}) |
| 20:51.47 | mintee | and i got myvar=ok |
| 20:51.54 | mintee | but the file still doesn't exist |
| 20:52.40 | b11d | . |
| 20:53.39 | ctdam | Anyone interested in a full time Asterisk-related programming position in the NYC area, please PM me |
| 20:53.57 | RoyK | mintee: iirc system() returns true anyway, but it sets a variable indicating its return value |
| 20:55.01 | mintee | Returns -1 on failure to execute the specified command. If the command itself executes but is in error, and if there exists a priority n + 101, where 'n' is the priority of the current instance, then the channel will be setup to continue at that priority level. Otherwise, System returns 0. |
| 20:55.15 | tzanger | mintee: I don't think you can do it that way |
| 20:55.26 | tzanger | you can't execute an application inside an IF function block |
| 20:55.46 | mintee | did you see what I'm trying to do in the long run... I assume that I'm doing it the hard way anyhow |
| 20:55.48 | sponger | mintee: the other guys can rip me if they want but this can be done in a few line sof perl or php |
| 20:55.56 | b11d | how exactly does the #include <filename> stuff work for extensions.conf? can I setup a context like [faculty] and #include <ext.faculty.conf> and then follow with another context like [students] and include another like "ext.students.conf" ? |
| 20:56.05 | tzanger | mintee: show function STAT, is it there? |
| 20:56.12 | mintee | I'm looking to have a voicemail box that must be setup before it's working by the user... The first time someone calls the number, they will get the intro with instructions to setup their voicemail, including recording their away (sorry) message and changing their PIN. After the voicemail has been setup, the user will never hear the intro message again. |
| 20:56.22 | tzanger | if not, you'll just have to use a script or try to piss around iwth system |
| 20:56.24 | tzanger | oth are nasty |
| 20:56.37 | mintee | tzanger, oddly, STAT doesn't exist |
| 20:58.05 | sbingner | mintee, you could also use ASTDB to check if a voicemailbox is 'initialized' yet |
| 20:59.10 | *** join/#asterisk phsdshft (n=phsdshft@204.56.88.151) |
| 20:59.39 | phsdshft | How do I specify the DTMF tone duration on SIP channels (I'm using the sendDTMF() command, using a SIP outbound channel) |
| 20:59.57 | phsdshft | I dont see how to do it anywhere in the documentation for SIP channels.. just toneduration for a zap channel |
| 21:00.36 | xp_prg | sip protocol is the generic protocol that communicates with asterisk yes? |
| 21:00.58 | sponger | if you are using DTMF mode info or rcf2833 |
| 21:01.05 | sponger | that doesnt mattter |
| 21:04.56 | phsdshft | I'm using rfc2833.. |
| 21:05.25 | mintee | sbingner, is that something I'm going to have to code? Like add another field to the database and all, or is that already ready already? |
| 21:05.28 | phsdshft | but, since I'm making outbound calls using asterisk and senddtmf.. how do I control how long the duration of the tone is? Also.. I could be using inband as well.. |
| 21:05.30 | *** join/#asterisk d-k-t (n=dt@60.176.198.100) |
| 21:07.20 | sponger | phsdshft: your outbound calls are "sip" to the pstn via a provider? |
| 21:07.48 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:08.06 | phsdshft | my outbound calls are out to broadvoice via SIP, then out to the PSTN via them, correct |
| 21:08.38 | sponger | then broadvoice will be sending the dtmf for you... if you use inband expect it to suck |
| 21:09.41 | phsdshft | right.. so I'm trying it with rfc2833 |
| 21:09.57 | phsdshft | so asterisk controls how long the dtmf tones are w/ senddtmf |
| 21:14.36 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:15.03 | defsdoor | why would my telco not know when my sangoma a200 has hung up a call ? The line is staying open |
| 21:15.17 | sponger | hangup() |
| 21:15.53 | defsdoor | sponger: it's doing a hangup |
| 21:16.04 | defsdoor | just not being detected by the telco end |
| 21:19.35 | *** join/#asterisk JadZilla (n=Jadelrab@212.103.170.135) |
| 21:21.56 | bsdwarrior | tkd-fender |
| 21:24.44 | bsdwarrior | tkd-fender I've tried all over the following with the manager and still cant set the userfield. Variable: userfield=test doesnt work. nor does SetCDRUserField(1234) or Set(CDR(userfield)=1234) |
| 21:28.23 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 21:32.15 | *** part/#asterisk JadZilla (n=Jadelrab@212.103.170.135) |
| 21:34.58 | drmessano-LT | I guess Vito got wacked |
| 21:35.34 | drmessano-LT | fuggetaboutit |
| 21:39.25 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-82-154.d4.club-internet.fr) |
| 21:40.12 | sbingner | what is generally the problem when DTMF tone detection goes to shit? |
| 21:40.30 | BBHoss | defsdoor, are you using the correct signaling? |
| 21:40.59 | defsdoor | BBHoss: fxs_ks |
| 21:42.07 | defsdoor | hangup on outgoing calls works fine |
| 21:42.27 | defsdoor | it's only incoming that stay active until calling party hangs up |
| 21:42.27 | *** join/#asterisk d1mas (n=chatzill@ip195.117.adsl.wplus.ru) |
| 21:42.27 | *** join/#asterisk Spyder12345 (n=bob@169.139.217.48) |
| 21:42.52 | defsdoor | but that will obviously be a problem as next outgoing call will get that call again |
| 21:43.25 | BBHoss | have you tried fxs_ls? |
| 21:43.25 | defsdoor | I could limit the effect of it if I could get asterisk to round robin the channels - but it uses first available |
| 21:43.31 | defsdoor | BBHoss: no |
| 21:43.35 | defsdoor | I will now |
| 21:44.03 | BBHoss | if not i'll be back in about 1 hour and i will help some more, got a damn exam :( |
| 21:44.23 | defsdoor | my card/driver doesnt do fxs_ls |
| 21:44.30 | defsdoor | zap won't loa |
| 21:44.31 | defsdoor | d |
| 21:44.49 | defsdoor | <PROTECTED> |
| 21:45.01 | defsdoor | oo hangon |
| 21:45.08 | defsdoor | I need to change zaptel too |
| 21:45.18 | *** join/#asterisk Vco (n=Vco@S0106000625891ca5.cg.shawcable.net) |
| 21:45.23 | BBHoss | yep |
| 21:45.57 | BBHoss | you can also try gs too, but ls and ks are the only ones ive seen |
| 21:45.58 | defsdoor | hmm - no - still doesn't like it |
| 21:46.14 | BBHoss | bbiab |
| 21:46.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:46.27 | generalhan | ok so i got the final pricing on the connections at the office ... 1.5 T1 = $499/mo. and the 15M cable connection is $419/mo ... so now the question is: is it worth $80 for me to have that T1 in there over the cable |
| 21:46.37 | Spyder12345 | Semi newbie question here. I am trying out the devel 1.6 beta 2 and having problems with just a basic sip registration using xlite. Is there a know problem or a change for basic registrations or am I missing something really simple? |
| 21:47.52 | phsdshft | How do I specify the DTMF tone duration on SIP channels (I'm using the sendDTMF() command, using a SIP outbound channel) |
| 21:48.35 | d1mas | Spyder12345: why don't you use stable 1.4 ? |
| 21:48.59 | Spyder12345 | need sip tcp support and I guess from my understanding it has been added to 1.6 |
| 21:50.38 | d1mas | phsdshft: SendDTMF does not let you specify duration - it passes duration=0 to lower level API which means "use default" to it |
| 21:51.18 | d1mas | not a big deal to patch though |
| 21:52.06 | d1mas | Spyder12345: yep, you are right. Have you tried 1.4 ? Xlite supports UDP too... |
| 21:53.17 | Spyder12345 | I actually need to sip tcp to connect to an exchange 2007 Unified Messaging system. However, just for basic testing I was trying to get a Xlite client to register via SIP and the normal way in 1.4 seems like a no go in 1.6 unless I am missing something. |
| 21:54.18 | d1mas | Spyder12345: I understand you want TCP :) However if I were you, I would start with 1.4+Xlite(UDP) and IF it worked - update to 1.6 |
| 21:55.02 | husimon | generalhan, Is this under some year contract? If not I might just try the cable for a while and see how it goes. |
| 21:55.18 | d1mas | this way you will know if your basic config is wrong or the problem is 1.6 |
| 21:56.20 | husimon | is chan_local the best way to do forwards ? |
| 21:56.30 | husimon | i mean Dial(Local... |
| 21:56.43 | husimon | to change contexts to the outbound and then dial the new number |
| 21:56.47 | *** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble) |
| 21:57.21 | d1mas | husimon: why you need so? |
| 21:57.40 | d1mas | so=this. |
| 21:58.36 | husimon | how else would you forward a call |
| 21:58.48 | husimon | say you want to forward a call to your cell |
| 21:58.59 | husimon | other then of course using the call forwad option on the phone. |
| 21:59.45 | husimon | dlmas the idea was to have a user configurable call forward key combination. I found a macro to do it, just asking if anyone had thoughts on it. |
| 21:59.58 | d1mas | husimon: ah. Yes, just do Dial. You do not necessarily need to use Local - you can Dial(Zap/something), Dial(SIP/something) etc |
| 22:00.40 | husimon | yeah except the phones can be outside my system |
| 22:00.44 | nybbled | hey all, were timed includes (ie.include => context|<times>|<weekdays>|<mdays>|<months> ) removed from asterisk 1.6.0 beta 2 ? |
| 22:00.49 | husimon | so I assume dialing sip or zap sort of restricts it |
| 22:00.52 | *** join/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au) |
| 22:01.10 | d1mas | husimon: however if you some outbound context which dispatches calls to different providers depending on number - then yes, Local seems to be logical choice |
| 22:01.48 | dofear | Hi, can anyone please comment on if TE412P is compatible with Delll PowerEdge 2900 |
| 22:01.49 | d1mas | "if you some |
| 22:01.57 | d1mas | "if you some"="if you have some" |
| 22:02.04 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 22:02.17 | dofear | PowerEdge comes with 1 PCI Express X8, 3 PCI Express X4, 2 64-bit/133MHz PCI-X |
| 22:02.25 | funxion | is there something wrong with zaptel branch 1.2? |
| 22:02.42 | dofear | But TE412P is a 33MHz 3.3 Volt PCI card |
| 22:02.46 | funxion | Im getting /usr/src/zaptel/Makefile:106: /usr/src/zaptel/Makefile.kernel26: No such file or directory when trying to make clean |
| 22:02.51 | d1mas | funxion: AFAIK it won't work with latest asterisk |
| 22:03.00 | d1mas | latest 1.4 I mean |
| 22:03.01 | defsdoor | dofear: it will work in the PCI-X slot |
| 22:03.07 | funxion | Im using 1.2 |
| 22:03.07 | tzafrir | funxion, not that I know of |
| 22:03.08 | phsdshft | d1mas: How do I change the default DTMF duration then (for SIP channels) |
| 22:03.13 | funxion | weird |
| 22:03.16 | nybbled | anyone have an idea? (timed includes with asterisk 1.6.x) |
| 22:03.22 | funxion | how do I get around that |
| 22:03.30 | funxion | goto 1.4? |
| 22:03.37 | [TK]D-Fender | husimon, A call is a call is a call. You "forward" (stupid term in dialplan) the call to somewhere the same way you'd call it normally. Its just that you end up in a different place that you might "normally" |
| 22:03.40 | dofear | defsdoor: even if the PCI-X slots are atleast of 66MHz |
| 22:03.50 | tzafrir | funxion, all the drivers should be available for zaptel 1.2, if you really need that |
| 22:03.58 | d1mas | phsdshft: you can not. There is only one constant specifying default DTMF duration for every channel. And you have to recompile asterisk in order to change it |
| 22:04.04 | funxion | tzafrir |
| 22:04.11 | phsdshft | d1mas: is it in channels.h? |
| 22:04.26 | funxion | tzafrir Im getting a lot of dropped calls over a t1 pri |
| 22:04.34 | lirakis | later all |
| 22:04.36 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 22:04.41 | funxion | I found that there is a bug in the version that Im using and was looking to upgrade |
| 22:04.43 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 22:04.45 | d1mas | tzafriri: will latest 1.4 compile and work with Zaptel 1.2 ? |
| 22:04.55 | d1mas | sorry, tzafrir. |
| 22:04.58 | dofear | Quoting from dell forum |
| 22:04.59 | dofear | PCI-X slots run at 66 or 133MHZ not 33MHZ. |
| 22:05.00 | dofear | if the card does not tolerate that speed it wont work. |
| 22:05.00 | dofear | All 64 bit slots by default are 66mhz 3.3V OR 133MHZ 3.3V only. |
| 22:05.03 | husimon | [TK]D-Fender, i'm just going to Dial(Local/<number>@maincontext) |
| 22:05.19 | husimon | where maincontext is where all my phones are, and where all the incoming and outbound contexts are included |
| 22:05.23 | husimon | is that the correct way to do it? |
| 22:05.34 | husimon | i think I should show my extensions.conf. sec |
| 22:05.37 | [hC] | bkruse: any ideas on the vlan issue with the aa50? |
| 22:05.43 | dofear | Anyone running these 3.3 volt 33MHz digium cards on pci-x? |
| 22:05.49 | [TK]D-Fender | husimon, total waste. |
| 22:05.53 | d1mas | phsdshft: if you really want change the duration, I would suggest you patchin app_sendtmf instead. You do not need to change defaults then |
| 22:05.55 | tzanger | nice, 5th cable cut |
| 22:06.03 | tzafrir | funxion, what card? |
| 22:06.08 | funxion | te210p |
| 22:06.14 | defsdoor | dofear: pci-x runs at the speed of the slowest card |
| 22:06.24 | bkruse | [hC]: no idea, if it is not dropping packets on the iptables side of things, then I really do not know |
| 22:06.32 | husimon | [TK]D-Fender, what do you mean total waste? |
| 22:06.34 | bkruse | I am going to try to mess around with it a little though, because I too, want to know :] |
| 22:06.58 | [TK]D-Fender | husimon, you are creating extra channels, more CDR, etc for nothing. Just do the REAL Dial. |
| 22:07.07 | [hC] | bkruse: have you done it before? Its very strange. tcpdumping the other side (since i dont have the pleasure of tcpdump on the aa50) shows weird unknown DSSP packet types |
| 22:07.07 | defsdoor | dofear: pci-x is backwards compatible with pci |
| 22:07.14 | defsdoor | dofear: http://en.wikipedia.org/wiki/PCI-X |
| 22:07.18 | funxion | will 1.4 zaptel work with asterisk 1.2? |
| 22:07.25 | nybbled | has include => time thing been replaced with GotoIfTime? |
| 22:07.31 | [hC] | bkruse: i thought it might be an mtu issue with the vlan tag on the packet... Shrinking the MTU on both sides didnt help. It will open the socket, just not send any data. |
| 22:07.52 | [hC] | bkruse: I even tried older firmware just incase there was an issue in the recent release. |
| 22:08.13 | defsdoor | dofear: "newer 3.3-volt PCI cards will work in a PCI-X slot" |
| 22:08.22 | d1mas | phsdshft: I told you complete bulshit. there IS timeout parameter for SendDTMF |
| 22:08.34 | husimon | [TK]D-Fender, i'm doing something similar to : http://www.voip-info.org/wiki-Asterisk+call+forwarding |
| 22:08.45 | husimon | are you saying that is the wrong way to set that type of forwarding up? |
| 22:09.11 | funxion | will 1.4 zaptel work with asterisk 1.2? |
| 22:09.13 | dofear | defsdoor: thanks |
| 22:09.21 | d1mas | phsdshft: You do not need to do anything - just SendDTMF(123456789,300) where 300 is in milliseconds |
| 22:09.24 | [TK]D-Fender | husimon, stop calling is "forwarding". its jsut friggen dialplan. And you do not need to recurse back into the dialplan and spam your CDR jsut to dial something DIFFERENT |
| 22:09.29 | tzafrir | funxion, basically yes. But you have to provide two symlinks from /usr/include/zaptel/*.h to the old places of zaptel 1.2 |
| 22:10.14 | funxion | o |
| 22:10.40 | funxion | I would like to get 1.2 to work which it did in the past but I redownloaded it thinking it would be a newer version and now it doesnt werk |
| 22:10.45 | dofear | Is there any free callshop application for asterisk? |
| 22:11.42 | funxion | I keep getting usr/src/zaptel/Makefile:106: /usr/src/zaptel/Makefile.kernel26: No such file or directory |
| 22:11.42 | funxion | make[2]: *** No rule to make target `/usr/src/zaptel/Makefile.kernel26'. Stop. |
| 22:12.17 | [hC] | oh here's something interesting |
| 22:12.22 | [hC] | tcpdumping the packet, |
| 22:12.27 | [hC] | i DO see the response |
| 22:12.28 | *** join/#asterisk erojasv (n=erojasv@190.43.97.39) |
| 22:12.45 | mvanbaak | hhmm, the ppl at http://www.the-asterisk-book.com only install zaptel and asterisk |
| 22:12.50 | [hC] | but tcpdump is classifying the packet type as "Unknown SSAP" |
| 22:12.52 | mvanbaak | why not libpri ? |
| 22:12.56 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 22:13.00 | [hC] | and I suppose, misinterpreting it. |
| 22:13.28 | husimon | mvanbaak, do you need libpri if you are just using sip? |
| 22:13.41 | husimon | mvanbaak, sorry I mean just an fx0 to a phone line. |
| 22:14.08 | bsdwarrior | tkd-fender I've tried all over the following with the manager and still cant set the userfield. Variable: userfield=test doesnt work. nor does SetCDRUserField(1234) or Set(CDR(userfield)=1234) |
| 22:14.25 | mvanbaak | husimon: guess not, but pri needs it for sure |
| 22:14.41 | mvanbaak | and since isdn10 is dirt cheap here |
| 22:14.42 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-72-48.tor.primus.ca) |
| 22:14.53 | mvanbaak | erm |
| 22:14.55 | defsdoor | isdn10 ? |
| 22:14.56 | mvanbaak | isdn15 |
| 22:15.47 | funxion | FYI the branches version of zaptel isnt working right now |
| 22:16.02 | funxion | I was getting the errors I posted ablove |
| 22:16.09 | mvanbaak | ISDN15, -20, -30 |
| 22:16.17 | funxion | I just got 1.2.9.1 and it werx fine |
| 22:16.19 | mvanbaak | that's the pri lines we have in .nl |
| 22:16.46 | defsdoor | mvanbaak: seems like a tariffing choice not a hardware/system solution |
| 22:16.54 | mvanbaak | indeed |
| 22:16.58 | funxion | someone may want to look into that |
| 22:17.04 | mvanbaak | isdn15 is E1 with 15 channels disabled |
| 22:17.06 | mvanbaak | that's it |
| 22:17.16 | defsdoor | in the uk you get isdn 30 and choose the number of channels - minimum 8 |
| 22:17.32 | mvanbaak | yeah, this is the same |
| 22:17.54 | mvanbaak | isdn 30 and you can get 15, 20 or 30 channels |
| 22:18.24 | asr33 | Hello guys; Should I be worried "SIP/2.0 404 Not Found" error message in my "sip debug" output? http://www.pastebin.ca/894376 |
| 22:18.37 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 22:18.56 | funxion | did anyone get that |
| 22:18.57 | mvanbaak | asr33: depends |
| 22:19.20 | mvanbaak | asr33: if you want to reach exten 39 in context default you should be worried indeed |
| 22:19.43 | asr33 | i'm having difficulty search google for the answer |
| 22:19.46 | mvanbaak | funxion: it it a clean checkout |
| 22:20.14 | asr33 | mvanbaak: the phone works perfectly |
| 22:20.31 | mvanbaak | Looking for 39 in default (domain xxx.112.72.48) <--- pretty clear |
| 22:20.56 | funxion | tzafrir |
| 22:20.58 | funxion | <PROTECTED> |
| 22:21.19 | asr33 | what should I put where to fix that? mvanbaak? |
| 22:21.58 | tzafrir | funxion, which version of zaptel didn't build? |
| 22:22.08 | asr33 | it's like a failed DNS lookup? |
| 22:22.21 | *** join/#asterisk DrAk0 (n=thinkpad@nelug/coreteam/luisjose) |
| 22:22.27 | funxion | the braches |
| 22:22.31 | funxion | branches |
| 22:22.41 | funxion | which I thought was latest stable no? |
| 22:22.42 | mvanbaak | asr33: in [default] add 39 |
| 22:22.50 | tzafrir | funxion, branches/1.2 ? |
| 22:22.59 | asr33 | just 39? |
| 22:23.16 | mvanbaak | exten => 39,1,Verbose(1,${CALLERID(num) tried to reach 39 in default) |
| 22:23.16 | funxion | tzafrir yes branches/1.2 |
| 22:23.18 | funxion | sry |
| 22:23.23 | mvanbaak | exten => 39,n,Hangup() |
| 22:23.26 | tzafrir | On what platform? |
| 22:23.28 | mvanbaak | something like that |
| 22:23.30 | funxion | debian |
| 22:23.32 | funxion | etch |
| 22:23.51 | funxion | 2.6.8-3 |
| 22:23.53 | asr33 | ok i'll try that thankyou! |
| 22:24.21 | phsdshft | d1mas: cool, thanks |
| 22:24.35 | mvanbaak | asr33: I forgot a } in the first line |
| 22:24.48 | mvanbaak | exten => 39,1,Verbose(1,${CALLERID(num)} tried to reach 39 in default) |
| 22:24.52 | asr33 | got ya |
| 22:25.29 | asr33 | i'll be right back |
| 22:25.41 | mvanbaak | hb sweety |
| 22:25.43 | mvanbaak | ;) |
| 22:26.11 | tzafrir | funxion, this is a clean checkout? |
| 22:26.18 | funxion | yes |
| 22:26.21 | funxion | 100% |
| 22:26.27 | [hC] | bkruse: so i have a tcpdump now illustrating exactly whats going on. the problem is that its not immediately clear. Maybe ill toss it up on the embedded list. |
| 22:26.38 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 22:27.56 | tzafrir | ah, with Sarge kernel. That makes sense |
| 22:28.12 | tzafrir | There's a problem there indeed, but will only be exposed with older kernels |
| 22:29.02 | stochastik | I want to accomplish DIAL(SIP/10@confserver) with Transfer... is TRANSFER(SIP/10@confserver) the correct syntax? |
| 22:29.38 | bkruse | [hC]: I would suggest that, and to try that. This is really stumping me |
| 22:29.58 | [hC] | bkruse: my first hunch is that its padding the tag past the end of the ethernet frame and confusing things |
| 22:30.05 | [hC] | bkruse: ive seen it before, but its been years and years |
| 22:31.10 | d1mas | ppls, could someone do me a favor - I have a movie on YouTube (Axe commercial - 2 mins) in which I cannot understand just one sentence. I need someone with native English (or just good listening skills) :) I know, it is complete offtopic, sorry. If someone can do that, please contact me in private chat. Thanks |
| 22:31.13 | bkruse | [hC]: what causes that? A possible driver issue? |
| 22:31.44 | [hC] | bkruse: yeah, I fixed a bug for this in the old eepro100 driver in like... kernel 2.4 - it was just a matter of hte kernel driver not expecting to ever have to deal with it |
| 22:32.09 | asr33 | mvanbaak: it worked, I can't thank you enough |
| 22:32.25 | asr33 | mvanbaak: is my yoda |
| 22:32.44 | mvanbaak | asr33: remember it and when you are near me buy me a bottle of single malt irish whisky |
| 22:33.00 | asr33 | will do |
| 22:33.04 | mvanbaak | cool |
| 22:34.41 | bkruse | [hC]: that is really old though, i would not imagine it popping up here unless it IS driver specific |
| 22:35.30 | [hC] | bkruse: yeah.. Its strange. I do see the data i am expecting in the response packet, but something about the packet gets treated not as a TCP packet but as a logical link control packet, which means something is being malformed somehow. |
| 22:35.55 | bkruse | OH, so it IS getting through and responding? |
| 22:36.10 | mvanbaak | I like the density of 'somehow' and 'something' in that sentence |
| 22:36.12 | [hC] | yep! I did a tcpdump and I see the response come back, but the packet is not being interpreted properly |
| 22:36.17 | bkruse | That would make it seem that the initial device (aa50) is not responding correctly because it does not know the initial frame |
| 22:36.25 | bkruse | hmm |
| 22:36.55 | [hC] | bkruse: i'll attach the pcap file to my email on the list if you want to look? I'm going to look some more too, but since i am not much with C, it may not get very far! :P |
| 22:37.13 | mvanbaak | heh, the joys of linux |
| 22:39.18 | *** join/#asterisk scr (i=lubo@zatwor.sk) |
| 22:41.17 | b11d | how exactly does the #include <filename> stuff work for extensions.conf? can I setup a context like [faculty] and #include <ext.faculty.conf> and then follow with another context like [students] and include another like "ext.students.conf" ? |
| 22:41.34 | mvanbaak | yup |
| 22:41.39 | b11d | sweet.. thanks |
| 22:42.07 | mvanbaak | #include just puts the content of the included file at that position |
| 22:42.54 | tzafrir | funxion, please make update |
| 22:43.00 | mvanbaak | :) |
| 22:43.04 | tzafrir | 1.2 should build again |
| 22:43.18 | b11d | thanks mvanbaak.. thats what I figured, nice to get verification on that. |
| 22:43.57 | mvanbaak | tzafrir: you use the 'expect' wrapper or do you have your cert password in your .subversion/servers file ? |
| 22:44.08 | tzafrir | yes |
| 22:44.26 | mvanbaak | erm, yes to 1 or 2 ? |
| 22:44.36 | tzafrir | expect |
| 22:44.39 | mvanbaak | ah |
| 22:45.06 | mvanbaak | I added svngpg to repotools. maybe you like that better ;) |
| 22:45.07 | tzafrir | What about gpg-agent? |
| 22:45.14 | mvanbaak | </commercial> |
| 22:45.32 | tzafrir | I have a little patch of my own to repotools |
| 22:45.42 | tzafrir | Just didn't know where to submit it |
| 22:46.08 | mvanbaak | I can commit there |
| 22:46.10 | *** part/#asterisk RoyK (n=roy@ip-197-29-149-91.dialup.ice.no) |
| 22:46.53 | mvanbaak | and svngpg is my project. so if it's about that send it to svngpg@vanbaak.info |
| 22:50.47 | b11d | if I start a context like [faculty] with an #include beneath it, I should NOT be having [faculty] in the actual included file, correct? it would be redundant.. |
| 22:51.13 | anarcat | sounds right |
| 22:51.21 | b11d | I guess I'll find out soon enough :) |
| 22:51.28 | b11d | im not including it... so.. we'll see. |
| 22:54.01 | tzafrir | mvanbaak, sent |
| 22:54.21 | mvanbaak | ok |
| 22:55.33 | defsdoor | BBHoss: you back yet ? |
| 22:57.51 | mvanbaak | hhmm |
| 22:57.57 | mvanbaak | makes sence to me |
| 22:58.41 | husimon | Question: I have a cell phone and an office phone. Say i want to ring both at once. I would use Dial(<BLAH>&<BLAH), but if my cell phone has no coverage it instantly goes to voicemail. This of course picks up and ruins the idea of calling two places at once. Can anyone think of a solution to this? |
| 22:58.46 | mvanbaak | tzafrir: I'll talk to kevin about it when he returns |
| 22:58.55 | tzafrir | thanks |
| 22:59.12 | husimon | because I can be in my office and have no coverage and then people can't contact me. |
| 22:59.25 | anarcat | husimon: turn off voicemail on your cell? :) |
| 22:59.26 | husimon | the obvious solution is to not forward while i'm in my office, but I walk around the building so much it would be a pain. |
| 22:59.27 | mvanbaak | I did the same over and over again to my copy because of the same reason |
| 22:59.34 | d1mas | husimon: you want easy one? Turn off voicemail on cell :) |
| 22:59.35 | husimon | anarcart laugh |
| 22:59.39 | anarcat | hehe |
| 22:59.50 | mvanbaak | I put ${HOME}/bin/repotools in my ${PATH} as well |
| 22:59.52 | anarcat | i'm serious, actually |
| 22:59.57 | husimon | I was thinking maybe dial my office phone for 3 rings first, and then forward to cel |
| 23:00.13 | anarcat | husimon: then you'll have callers wait for a while... but it's a good idea |
| 23:00.25 | husimon | maybe just two rings |
| 23:00.26 | husimon | i dunno |
| 23:00.37 | cappiz | someone has norwegian sound files for asterisk? |
| 23:00.38 | husimon | i wish there was a way for asterisk to know if a caller picking up was real or voicemail |
| 23:00.51 | mvanbaak | husimon: there is |
| 23:00.54 | mvanbaak | app_amd |
| 23:00.57 | husimon | yeah but for a cell phone? |
| 23:01.09 | husimon | i can see that working for a sip phone connected to * |
| 23:01.15 | d1mas | husimon: you can do it manually. But it requires experienced person |
| 23:01.16 | mvanbaak | answering machine detection |
| 23:01.47 | mvanbaak | husimon: asterisk*CLI> core show application AMD |
| 23:02.01 | d1mas | omg |
| 23:02.02 | mvanbaak | <PROTECTED> |
| 23:02.02 | mvanbaak | [Synopsis] |
| 23:02.02 | mvanbaak | Attempts to detect answering machines |
| 23:02.06 | husimon | yeah i'm reading it |
| 23:02.17 | d1mas | the key here is _attempts_ :) |
| 23:02.23 | mvanbaak | uhhuh |
| 23:02.30 | husimon | can you think of a way to use that with my situation? |
| 23:02.34 | anarcat | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD |
| 23:02.39 | mvanbaak | best way is to disable voicemail |
| 23:02.48 | mvanbaak | husimon: yup |
| 23:02.56 | husimon | yeah that's not a solution I like :P |
| 23:03.14 | anarcat | interesting stuff anyways |
| 23:03.21 | mvanbaak | Dial(Local/mycell@outbound/n&Local/myofficephone@internal/n) |
| 23:03.22 | d1mas | mvanbaak: actually he can call cell and play something like "press 1" to accept the call. And only when 1 is pressed - do bridging |
| 23:03.29 | mvanbaak | make the local stuff execute a macro |
| 23:03.50 | mvanbaak | d1mas: thats......evil |
| 23:03.56 | stochastik | I want to accomplish DIAL(SIP/10@confserver) with Transfer... is TRANSFER(SIP/10@confserver) the correct syntax or is that now how Transfer works? |
| 23:04.02 | mvanbaak | I would hangup on that |
| 23:04.20 | d1mas | mvanbaak: that is the only way AFAIK (except for disabling voicemail) |
| 23:04.23 | mvanbaak | stochastik: core show application TRANSFER |
| 23:04.25 | husimon | mvanbaak, so you are saying use a macro in the dial statement instead of a real dial |
| 23:04.27 | husimon | dail |
| 23:04.34 | mvanbaak | husimon: yup |
| 23:04.41 | husimon | the & can handle that? |
| 23:04.46 | mvanbaak | sure |
| 23:04.52 | mvanbaak | I use it all the time |
| 23:04.54 | d1mas | mvanbaak: it works with any kind of answering machine and does not require any "training" and configuration |
| 23:05.00 | stochastik | mvanbaak: Yes... already know that. The syntax isn't working... that's why I asked specifically. |
| 23:05.06 | husimon | so how do you tell it ok now it's really picked up on one side? |
| 23:05.31 | husimon | after you do the answering machine detection |
| 23:05.36 | mvanbaak | Dial(Local/employee_laptop_exten@internal/n&Local/employee_gsm_nr@route-employee-gsm/n) |
| 23:06.08 | d1mas | there is no reliable way of detecting answering machine Im aware og |
| 23:06.10 | phix | G'day |
| 23:06.10 | d1mas | of |
| 23:06.11 | mvanbaak | husimon: check the variable AMDSTATUS |
| 23:06.24 | mvanbaak | d1mas: app_amd does a great job |
| 23:06.27 | phix | any one awake that can assist me with a zaptel / TDM400p issue I am having? |
| 23:07.05 | d1mas | mvanbaak: from its description it looks like I have to tune it |
| 23:07.10 | phix | Where can I paste? |
| 23:07.26 | husimon | phix www.pastebin.com |
| 23:07.32 | phix | nah too slow to load up |
| 23:07.32 | stochastik | I want to accomplish DIAL(SIP/10@confserver) with Transfer... is TRANSFER(SIP/10@confserver) the correct syntax? I'm trying to accomplish a SIP REFER with Asterisk. |
| 23:07.41 | husimon | phix wtf connection are you on |
| 23:07.55 | husimon | that pastebin is too slow |
| 23:07.56 | phix | husimon: I will try again |
| 23:08.27 | phix | Wow they must of upgraded their end, it used to take ages to load |
| 23:08.29 | phix | http://pastebin.com/m74a3229b |
| 23:08.47 | phix | I am on 4mbit / 1Mbit or something |
| 23:09.21 | phix | that is my /etc/zaptel.conf and /etc/asterisk/zapata.conf configuration files |
| 23:09.22 | mvanbaak | ~pb |
| 23:09.23 | 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 |
| 23:09.23 | anarcat | there are lots of pastebins around.. pastebin.ca is also pretty good |
| 23:09.28 | mvanbaak | there you go |
| 23:09.36 | phix | mvanbaak: thnx |
| 23:10.04 | phix | mvanbaak: Well I already pasted at pastebin.com, the link is a few lines up :) |
| 23:10.38 | stochastik | Ahh... maybe Transfer is broken: http://forums.digium.com/viewtopic.php?t=20246 |
| 23:10.58 | phix | transfer? does that help me? |
| 23:12.10 | phix | so any way, the problem is only the first two FXS modules seem to work (I have three). It doesn't matter if I switch them around, only the first two ports have a dial tone, the other ones sounds like zaptel is not running at all (no dial tone, and it amplifies whatever you say on the phone) |
| 23:12.36 | phix | when the wctdm module loads it picks up allthree |
| 23:12.47 | phix | the other two work great |
| 23:12.48 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-212-194-228-238.d4.club-internet.fr) |
| 23:13.00 | phix | so what else could it be? what can I do to try and resolve this issue/. |
| 23:14.10 | JT | i'll have a look if you put it on rafb.net :P |
| 23:14.23 | phix | JT: lol it is on pastebiN!!!! |
| 23:14.39 | JT | i can't access it. |
| 23:14.59 | mvanbaak | phix: move around the physical lines |
| 23:15.05 | mvanbaak | maybe it's a dead line |
| 23:15.17 | phix | JT: http://rafb.net/p/y1l9ih85.html |
| 23:15.23 | phix | mvanbaak: I did |
| 23:15.53 | phix | mvanbaak: the lines plug into a PBX system (giving the PBX system three new lines to allow its telephones to ring ppl on) |
| 23:16.21 | phix | mvanbaak: It isn't a dead line, they all work on the PBX system |
| 23:16.34 | JT | fxoks=1,2,4 |
| 23:16.35 | phix | except for the third one that is :) |
| 23:16.46 | JT | you left a gap? |
| 23:17.05 | phix | JT: no, I tried moving the modules around to see if it was the TDM card at fault |
| 23:17.16 | phix | well yes I guess I did leave a gap |
| 23:17.19 | phix | I have 3 modules |
| 23:17.27 | phix | FXS |
| 23:17.29 | JT | so if the module that was not working is moved to a slot that is working |
| 23:17.31 | JT | it works? |
| 23:17.37 | phix | yes |
| 23:17.42 | JT | it's not 1-3 just due to testing? |
| 23:17.54 | phix | that is correct |
| 23:17.58 | JT | well |
| 23:18.05 | JT | try a different version of zaptel and asterisk |
| 23:18.08 | phix | it was 1-3 when I got it |
| 23:18.11 | JT | if that does the same thing |
| 23:18.18 | JT | return the card |
| 23:18.18 | phix | zaptel from debian testing |
| 23:18.21 | JT | as it is defective |
| 23:18.22 | JT | compile. |
| 23:18.27 | phix | hmmm |
| 23:18.34 | phix | I would like a debian package |
| 23:18.46 | JT | i don't care, you need to test, please compile. |
| 23:18.47 | phix | I should get the debian package source? |
| 23:19.03 | phix | hmmmm, ok sir :( |
| 23:19.10 | phix | ~zaptel |
| 23:19.11 | jbot | hmm... zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 23:19.15 | *** join/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 23:19.21 | phix | jbot: a url would be great |
| 23:19.45 | *** part/#asterisk tuxfoo (n=tuxfoo@pool-72-65-149-192.chrlwv.east.verizon.net) |
| 23:19.59 | JT | asterisk.org |
| 23:20.08 | phix | JT: so it is def not Asterisk? I am using version 1.2something |
| 23:20.15 | phix | from deian stable |
| 23:20.17 | phix | debian even |
| 23:20.52 | *** part/#asterisk DarWin_vcch (n=daryl@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 23:20.59 | ManxPower | phix: We do not consider precompiled packages to be "asterisk". At best it would be "asterisk customized and built in ways we have no way of knowing about" |
| 23:21.16 | ManxPower | And if you want help with the package then talk to the package manager. |
| 23:21.47 | Daviey | "Hi apt-get, how are you?" |
| 23:22.04 | Daviey | s/manager/maintainer :() |
| 23:23.48 | anarcat | ManxPower: isn't "no way of knowing about" a bit of a stretch? |
| 23:24.00 | anarcat | debian packages include detailed patches and a changelog, for example |
| 23:26.07 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:26.41 | mvanbaak | anarcat: like we have the time and resources to keep track of all them distribution patches out there |
| 23:26.58 | mvanbaak | every person that creates a package for $distribution does it their own way |
| 23:27.06 | mvanbaak | they maintain their own patches |
| 23:27.06 | mvanbaak | etc |
| 23:27.13 | mvanbaak | we cant keep track of that |
| 23:27.18 | anarcat | i agree |
| 23:27.29 | anarcat | i just wanted to correct a little tidbit :) |
| 23:27.37 | mvanbaak | so you have 2 ways to go |
| 23:27.53 | anarcat | there are ways to know about the changes in the packages, it's just it's too much trouble to keep track of all configurations |
| 23:27.58 | mvanbaak | either install the package from your distribution and ask them about bugs |
| 23:28.04 | anarcat | and anyways, (for example) debian has its own BTS |
| 23:28.08 | anarcat | exactly |
| 23:28.19 | mvanbaak | or install the official version and contact bugs.digium.com for support |
| 23:28.58 | mvanbaak | yeah, debian and friends have their own BTS |
| 23:29.08 | mvanbaak | but some distros tell you to comlpain upstream |
| 23:29.14 | mvanbaak | and that's just plain wrong |
| 23:29.24 | anarcat | yeah |
| 23:31.42 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 23:33.21 | ManxPower | anarcat: And we would have to study them extensively to help someone with a debian package. |
| 23:33.26 | eric2 | is there anyway to increase the audio gain on a call? |
| 23:33.40 | ManxPower | eric2: rxgain= and txgain= in zapata.conf |
| 23:33.49 | ManxPower | if you are not using zap, then you are out of luck |
| 23:34.01 | eric2 | I'm not using zap devices... :( |
| 23:34.18 | ManxPower | eric2: then the gain must be changed where the call is converted between PSTN and VoIP |
| 23:34.24 | mvanbaak | eric2: almost 100% of the devices can handle volume settings |
| 23:34.37 | eric2 | ok, everything here is without direct pstn connectivity |
| 23:34.41 | anarcat | ManxPower: indeed |
| 23:34.44 | ManxPower | unless you just want to change the gain on the phone, but if that was the case you would have given us more information like what damn phone you are using |
| 23:34.45 | eric2 | so zapata is no good for me |
| 23:34.56 | eric2 | ya, stupid phone! |
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| 23:35.10 | mvanbaak | eric2: use the volume buttons on the phone |
| 23:35.11 | eric2 | n, phone works fine... its the fax I'm trying to get going |
| 23:35.18 | ManxPower | eric2: VoIP does not have ANY volume loss, so it would be silly to compensate for it, even if all the other things were perfect. |
| 23:35.18 | eric2 | got 1 fax out of 5 |
| 23:35.20 | mvanbaak | hell, even the BT100 has volume buttons |
| 23:35.36 | ManxPower | eric2: that's pretty common with FaxOverVoiceOverIPOverInternet |
| 23:35.45 | eric2 | ya, I'm all depressed now |
| 23:35.52 | eric2 | been looking for a work around |
| 23:35.56 | ManxPower | eric2: get yourself an analog fax line |
| 23:36.02 | eric2 | bah |
| 23:36.03 | mvanbaak | fax over wan is _NOT_ going to work |
| 23:36.09 | eric2 | that would mean defeat |
| 23:36.19 | eric2 | but I know you're right |
| 23:36.36 | ManxPower | ALL other solutions are orders of magnitude more complicated |
| 23:36.44 | mvanbaak | yup |
| 23:37.00 | mvanbaak | and they will never guarantee you that it's 100% |
| 23:37.08 | eric2 | hylafax, iax modems and the rest of the crap.. astrafax |
| 23:37.27 | mvanbaak | faxovervoip over the internets is not ok |
| 23:37.46 | mvanbaak | in a local lan it works great |
| 23:37.54 | eric2 | timing issue? |
| 23:38.13 | mvanbaak | but there you dont have to worry about packetloss, variation in latency etc |
| 23:38.35 | eric2 | using tcp, is there not a buffer that can be used? |
| 23:38.49 | mvanbaak | * does not do tcp |
| 23:38.52 | eric2 | udp? |
| 23:39.07 | mvanbaak | no way to know for sure wether your packets arrived or not |
| 23:39.10 | mvanbaak | tadaaaa |
| 23:39.22 | eric2 | hmm, so much for beating this dead horse |
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| 23:40.12 | mvanbaak | get a landline |
| 23:40.22 | mvanbaak | and enjoy faxes |
| 23:40.51 | mvanbaak | I'm off to bed |
| 23:40.53 | mvanbaak | latero |
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