| 00:00.16 | JerJer | like it has become with the Linux kernel and all of the various distros out there |
| 00:00.30 | ctooley | JerJer having 4 kids I can tell you that while I make decisions about some things that effect how my babies grow up, everyone else involved in their rearing (teachers, friends, etc) make decisions to. If I can't trust the people that are involved, I should have made better decisions about the people I associate with. |
| 00:00.35 | derek_1234 | Fragmentation is part of the course. |
| 00:00.37 | mstocco | code forking is something supported by Linus |
| 00:00.40 | JerJer | anthm: he deserves more than a heads up |
| 00:00.41 | ctooley | The same can be said about an open source project |
| 00:00.50 | derek_1234 | look at kde, simplekde, gnome etc |
| 00:00.57 | derek_1234 | all freagments of different X code |
| 00:01.03 | derek_1234 | eventually, one will "win" |
| 00:01.10 | JerJer | mstocco: notice I am not saying I am against it |
| 00:01.11 | derek_1234 | ideas will flow betweeen all X systems. |
| 00:01.12 | ctooley | whining, bitching, and telling people off serves no purpose and it certainly isn't being productive in here |
| 00:01.52 | JerJer | ctooley: we haven't gotten to telling people off yet |
| 00:01.55 | JerJer | close |
| 00:01.58 | derek_1234 | So JerJer, when will your h323 system for * be really stable ? |
| 00:02.01 | anthm | I call him, I say I am going to make X and do you have any preferences in the implmentation. |
| 00:02.07 | derek_1234 | I mean really stable. |
| 00:02.12 | JerJer | derek_1234: fuck you |
| 00:02.12 | anthm | that is what i mean by head's up |
| 00:02.22 | derek_1234 | you cannnot. |
| 00:02.27 | ctooley | see, now that was productive, wasn't it. |
| 00:02.30 | JerJer | use chan_woomera |
| 00:02.35 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 00:02.40 | JerJer | i wrote one months ago and was asked by mark to not release it |
| 00:02.41 | *** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3943638.sympatico.ca) |
| 00:02.44 | derek_1234 | apparently, chan_woomera is stable. |
| 00:02.46 | JerJer | so i did what i was asked todo |
| 00:03.13 | *** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo) |
| 00:03.26 | derek_1234 | So, jerjer, when will your h.323 channel be stable ???? |
| 00:03.37 | JerJer | never just because you brought it up |
| 00:03.51 | derek_1234 | never ? |
| 00:03.53 | derek_1234 | never ? |
| 00:04.02 | derek_1234 | what is up with your programming skills ? |
| 00:04.08 | JerJer | they fucking suck |
| 00:04.09 | derek_1234 | Why can you not ask for help ? |
| 00:04.27 | JerJer | i wrote chan_h323 because manousus told me i coudln't write a better H.323 driver |
| 00:04.42 | JerJer | i called mark and 48 hours later i released chan_h323 |
| 00:04.49 | JerJer | i dropped out of college |
| 00:04.56 | JerJer | and i failed CPS210 |
| 00:04.59 | jsaunders | Does anyone have h323 directly to Southwestern Bell in the Texas area, with commercial space available? |
| 00:05.12 | JerJer | i have never once stated i was a C or C++ programmer |
| 00:05.29 | Umaro | ok guys, i'm trying to configure a DID on my asterisk box to be busy |
| 00:05.37 | Umaro | don't I just do exten => did,1,Busy ? |
| 00:05.42 | JerJer | Umaro: yes |
| 00:06.00 | Umaro | when the remote side calls, they get a AST_CAUSE_NO_USER_RESPONSE (pri) |
| 00:06.11 | JerJer | derek_1234: then unlike manousus i disclaimed my code and Mark gladly accepted it |
| 00:06.26 | JerJer | then PCadach and snewpy came along and made it suck less |
| 00:06.45 | JerJer | hence why I have never asked that I be credited for H.323 shit |
| 00:06.56 | derek_1234 | Thanks. |
| 00:07.00 | derek_1234 | I appreciate the update. |
| 00:07.06 | derek_1234 | That does clarify things for me. |
| 00:07.09 | JerJer | then jsharp paid us to fix a few more things |
| 00:07.09 | derek_1234 | have a nice day. |
| 00:07.26 | JerJer | then learth (sp?) sent me a 7910 and chan_skinny was born |
| 00:07.50 | bkw_ | Why would Digium asked you not to release chan_woomera? |
| 00:08.12 | JerJer | Mark asked me as a friend not to release it |
| 00:08.19 | bkw_ | but why? |
| 00:08.29 | bkw_ | their has to be a reason behind that |
| 00:08.31 | jsaunders | Anyone have any commercial h323 voip routes available? Any country, I'm looking for them all. Lemme know please. |
| 00:08.36 | JerJer | it shows people how they can sidestep the GPL |
| 00:08.43 | JerJer | by putting things at arm length |
| 00:09.07 | anthm | doesnt agi do that too? |
| 00:09.12 | bkw_ | yes |
| 00:09.20 | anthm | and manager ? |
| 00:09.26 | bkw_ | but you can also sidestep the GPL other ways with asterisk |
| 00:09.30 | JerJer | that is specifically why agi and manager was created |
| 00:09.30 | bkw_ | its quite easy |
| 00:09.40 | JerJer | so people wouldn't have to sidestep |
| 00:09.46 | twisted | are you guys still arguing? |
| 00:09.53 | bkw_ | not really.. i'm just asking questions |
| 00:09.56 | twisted | ok |
| 00:10.02 | *** part/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 00:10.18 | MikeJ[Laptop] | where have all the flowers gone? |
| 00:10.37 | JerJer | i killed them |
| 00:10.43 | twisted | MikeJ[Laptop], i ate them |
| 00:11.15 | hardwire | so |
| 00:11.20 | hardwire | hows life everybody? |
| 00:11.25 | JerJer | shitty |
| 00:11.31 | hardwire | mine is too |
| 00:11.31 | bkw_ | why? |
| 00:11.36 | jsaunders | There's gotta be atleast one h323 provider in here with some room open. |
| 00:11.37 | MikeJ[Laptop] | it's happy day in #asterisk... |
| 00:11.42 | twisted | life == dealing with HEAT == shitty |
| 00:11.49 | hardwire | twisted: where are you? |
| 00:11.54 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 00:12.01 | MikeJ[Laptop] | hardwire, at his computer. |
| 00:12.01 | jsaunders | Nobody wants my money? Heh. |
| 00:12.07 | MikeJ[Laptop] | I want your money |
| 00:12.13 | MikeJ[Laptop] | what for |
| 00:12.13 | jsaunders | Well, of course you do. Who doesn't. |
| 00:12.15 | bkw_ | jsaunders, well my paypal is brian.west@mac.com |
| 00:12.17 | bkw_ | sure send me son |
| 00:12.18 | shmaltz | I want your money as well |
| 00:12.22 | twisted | hardwire, huntsville |
| 00:12.38 | jsaunders | Looking for h323 routes to the world, any country. |
| 00:12.42 | mstocco | harewire: he is like all of us, basking in the heat of this channel |
| 00:12.46 | hardwire | twisted: oh yeah |
| 00:12.47 | hardwire | doh |
| 00:12.49 | MikeJ[Laptop] | jsaunders, from where? |
| 00:12.54 | twisted | mstocco, no, i mean REAL heat |
| 00:13.00 | hardwire | so... interrupts.. is there a huge issue with eth's sharing an interrupt? |
| 00:13.02 | hardwire | for the most part |
| 00:13.06 | jsaunders | MikeJ: North America. |
| 00:13.10 | twisted | my office was 85 degrees INSIDE today |
| 00:13.18 | hardwire | can you get two full speed 100bt conns on a shared interrupt |
| 00:13.35 | mstocco | twisted: mine was too, 83F |
| 00:13.38 | Lathos42 | twisted: I'll share some of my Air conditioning with you.. my office was 68 degrees |
| 00:13.46 | hardwire | hmm |
| 00:13.49 | hardwire | I live in alaska |
| 00:13.52 | twisted | word |
| 00:13.54 | hardwire | heat is like.. 74 |
| 00:13.58 | hardwire | its really hot then |
| 00:14.02 | hardwire | unbearable even |
| 00:14.03 | twisted | my airconditioning at home rocks |
| 00:14.08 | twisted | it's a nice cool 66 degress in here |
| 00:14.09 | hardwire | 104 in my home in Colorado |
| 00:14.18 | shmaltz | 96 here |
| 00:14.21 | hardwire | I moved here to get away from the 80's |
| 00:14.25 | twisted | lol |
| 00:14.47 | hardwire | 80+F |
| 00:14.48 | shmaltz | hardwire, 104 in your home or 104 in your hometown? |
| 00:14.52 | hardwire | not 1980's |
| 00:14.59 | hardwire | shmaltz: outside temp in colorado |
| 00:15.25 | anthm | harryvv needs a good C book anyone know any titles? |
| 00:15.26 | shmaltz | oh, so you have air conditioning like the rest of us in the US |
| 00:15.53 | *** join/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 00:16.20 | twisted | hardwire, we're talking about indoor temps |
| 00:16.30 | anthm | how many hours ? |
| 00:16.42 | blitzrage | The C Programming Language, Second Edition, by Brian W. Kernighan and Dennis M. Ritchie, Prentice Hall Software Series. |
| 00:16.43 | twisted | teach yourself c in 24 hours....hahahaha.... |
| 00:16.48 | hardwire | twisted: well .. it gets to 79 |
| 00:16.55 | harryvv | funny twisted |
| 00:17.04 | QbY | we have phone numbers/termination services with broadvoice. everything works great, with the exception of the toll free number, callers complain about not being able to hear when they call the 800# although if you call the local it sounds normal..... is there a way that we can boost the outgoing volume on that particular connection (no one will ever call the local number) |
| 00:17.16 | Lathos42 | anthm: Its supposedly an hour a chapter, but i've read it for maybe an hour total so far :) |
| 00:18.00 | twisted | Lathos42, does it have a blurb about writing your own "hello world" program at the end of the book? |
| 00:18.45 | Lathos42 | twisted: Well, the first chapter consisted of his own version of Hello World |
| 00:19.49 | Umaro | JerJer: so, any idea why it would be giving me a AST_CAUSE_NO_USER_RESPONSE instead of AST_CAUSE_USER_BUSY ? |
| 00:20.01 | Umaro | JerJer: asterisk on both sides, both using PRIs |
| 00:20.43 | bkw_ | Umaro, recompile both libpri |
| 00:20.44 | JerJer | duno about that one |
| 00:20.45 | bkw_ | and asterisk |
| 00:20.54 | bkw_ | sounds like code skew |
| 00:20.59 | *** join/#asterisk Inv_arp (junya@adsl-8-230-143.mia.bellsouth.net) |
| 00:21.16 | Umaro | hmm.. ok |
| 00:25.57 | twisted | yay |
| 00:26.00 | twisted | tonight is going to be a fun night |
| 00:27.23 | sivana | bkw_: how long has /L been broke? |
| 00:27.31 | shmaltz | twisted, why? |
| 00:27.57 | twisted | shmaltz, because I'm going to be tunneling myself some IP's, and turning up across-the-board LDAP |
| 00:28.42 | shmaltz | twisted, are these yours or stolen? |
| 00:28.43 | MikeJ[Laptop] | sivana, Corydon posted to the dev list the commit that broke it |
| 00:28.44 | shmaltz | :) |
| 00:28.49 | MikeJ[Laptop] | looks like january |
| 00:28.50 | twisted | shmaltz, these what? |
| 00:28.56 | shmaltz | IPs |
| 00:29.04 | sivana | ok |
| 00:29.13 | twisted | yeah. I stole them. I went in and arp spoofed them from somewhere else |
| 00:29.39 | shmaltz | so you using like man-in-the-middle attack to steal them? |
| 00:29.40 | bkw_ | sivana, since jan 4th |
| 00:29.53 | twisted | shmaltz, sarcasm. learn it. love it. |
| 00:30.14 | shmaltz | I know LOL |
| 00:30.30 | shmaltz | I'm just laughing along |
| 00:30.53 | bkw_ | Ok guys .. I want you to see the truth here... I don't wanna hide anything so here is the personal talk JerJer and I just had http://pastebin.ca/18564 |
| 00:30.53 | shmaltz | bored waiting for that someone that might bring some chalange into the room |
| 00:31.02 | bkw_ | he attacked me |
| 00:32.01 | harryvv | bkw, i learn in bussines not to take things personally. |
| 00:32.08 | bkw_ | I didn't |
| 00:32.12 | bkw_ | I'm so not takign this personal |
| 00:32.16 | bkw_ | JerJer is |
| 00:33.14 | shmaltz | bkw_, JerJer, stop this, we will all suffer if this goes anywhere beyond this channel |
| 00:33.27 | derek_1234 | jerjer bkw_ that chat was interesting. |
| 00:33.32 | derek_1234 | crude, but silly. |
| 00:33.38 | shmaltz | I hope that what JerJer says overthere about Mark/Digium is not true |
| 00:33.44 | derek_1234 | a complete waste of time. |
| 00:33.48 | derek_1234 | nothing changed. |
| 00:33.56 | derek_1234 | and people just get sillier/angrier |
| 00:34.09 | derek_1234 | How about you write better /more stable code ? |
| 00:34.12 | JerJer | bkw_: your such a bitch |
| 00:34.40 | bkw_ | bitch? |
| 00:34.41 | bkw_ | how? |
| 00:34.43 | derek_1234 | children - be nice - and leave the slanging alonw |
| 00:35.22 | MikeJ[Laptop] | can anyone help with asterisk in here ;) |
| 00:35.49 | mtgh | bkw_: I am a UofM Grad student, what can you cut the registration rate to for cluecon |
| 00:36.22 | mtgh | bkw_: I can pick up the hotel room |
| 00:36.33 | *** part/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net) |
| 00:37.02 | JerJer | it is now personal bkw |
| 00:37.05 | derek_1234 | jerjer - I do not understand. All this silly bitching is advertising the cluecon ? |
| 00:37.06 | JerJer | i told you that in private |
| 00:37.17 | derek_1234 | jerjer - do you want to advertise cluecon ? |
| 00:37.18 | JerJer | because mark told it to me in private |
| 00:37.27 | JerJer | i want nothing to do with that con |
| 00:37.27 | derek_1234 | nothing we do on chat is private. |
| 00:37.35 | derek_1234 | nothing - it is all open to interception. |
| 00:37.39 | derek_1234 | get real. |
| 00:37.45 | derek_1234 | then leave it alone. |
| 00:37.50 | bkw_ | i'm being open and honest with everyone |
| 00:37.59 | bkw_ | I really don't know why people feel the need to attack ME personally |
| 00:38.00 | derek_1234 | if you really really want nothing to do with cluecon, don't mention it. |
| 00:38.01 | JerJer | no if they have the right to push it, i have the right to push against it |
| 00:38.07 | derek_1234 | don't even write bout it. |
| 00:38.10 | derek_1234 | move on. |
| 00:38.15 | JerJer | nope |
| 00:38.20 | JerJer | i am a little whiny bitch like that |
| 00:40.19 | JerJer | i picked the wrong week to stop sniffing glue |
| 00:40.57 | MikeJ[Laptop] | no no... bkw_ is the bitch... |
| 00:41.14 | MikeJ[Laptop] | ;) |
| 00:41.43 | bkw_ | JerJer would do the Honorable thing and tell the channel what you just said to me |
| 00:42.03 | JerJer | bkw better get a personal protection order on me |
| 00:42.11 | JerJer | cuz i will fucking kill him if i saw him righ tnow |
| 00:42.13 | derek_1234 | ok. ok. |
| 00:42.13 | JerJer | fuck you bkw_ |
| 00:42.15 | bkw_ | JerJer has threatened my life |
| 00:42.23 | derek_1234 | how crude jerjer. |
| 00:42.41 | JerJer | little bitch |
| 00:42.44 | postel | cripple fight |
| 00:42.45 | derek_1234 | I cannot believe that someone in your leadership posistion would be so crude. |
| 00:42.53 | JerJer | leadership? |
| 00:42.57 | derek_1234 | JerJer, I would be ashamed. |
| 00:42.58 | blitzrage | postel: lol |
| 00:43.02 | niZon | crude? |
| 00:43.13 | derek_1234 | what a terrible example you present to the world of the open source community. |
| 00:43.13 | niZon | he's such a happy person |
| 00:43.18 | postel | and a pepsi |
| 00:43.18 | JerJer | i took my issues private |
| 00:43.23 | JerJer | then he made them public |
| 00:43.26 | derek_1234 | Usage of the word "fuck" is crude. - R16 |
| 00:43.41 | bkw_ | JerJer you're accusing me of something that is not true... |
| 00:43.51 | bkw_ | I would rather you not go around telling people things that are not true |
| 00:43.57 | twisted | okay, bkw_, jerjer, calm down. I don't want to kick/ban either of you, but I will if this keeps up. You're both friends of mine, who are having a difference of opinion, please take it elsewhere. |
| 00:43.57 | derek_1234 | What was the comment, "open source programmers working together" ????? |
| 00:44.07 | JerJer | twisted i tried to |
| 00:44.11 | bkw_ | twisted I didn't start this |
| 00:44.17 | twisted | JerJer, point taken |
| 00:44.18 | JerJer | he brough it back publlic |
| 00:44.18 | derek_1234 | I don't care. |
| 00:44.21 | twisted | bkw_, it doesn't matter |
| 00:44.22 | derek_1234 | both be quiet |
| 00:44.25 | derek_1234 | please. |
| 00:44.26 | derek_1234 | please. |
| 00:44.35 | *** part/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 00:44.44 | MikeJ[Laptop] | with sugar on top? |
| 00:44.49 | twisted | the fact is, it's down right disturbing at this point |
| 00:44.52 | derek_1234 | anything. |
| 00:44.59 | twisted | and slightly scary |
| 00:45.03 | derek_1234 | and silly. |
| 00:45.10 | bkw_ | it was silly LONG ago |
| 00:45.13 | bkw_ | its down right stupid now |
| 00:45.14 | MikeJ[Laptop] | as scary as corydon on a loopy night? |
| 00:45.19 | twisted | MikeJ[Laptop], hah |
| 00:45.21 | bkw_ | doh |
| 00:45.38 | MikeJ[Laptop] | so, anybody here use asterisk.. |
| 00:45.39 | twisted | so let's all just simmer down a bit, drink a tall glass of 151, and relax |
| 00:45.58 | Ayano | Ooooo 151 |
| 00:46.16 | blitzrage | MikeJ[Laptop]: never |
| 00:46.33 | *** join/#asterisk santiago (~santiago@63.245.86.175) |
| 00:46.43 | twisted | thank you, that is all |
| 00:46.47 | twisted | move along now, nothing to see here |
| 00:47.26 | niZon | I use asterisk... ****** see? |
| 00:47.39 | postel | hey? what happened to the cripple fight? |
| 00:47.45 | postel | damn... |
| 00:48.06 | derek_1234 | it got cripplingly silly. |
| 00:48.14 | twisted | *ahem* |
| 00:48.17 | twisted | MOVE ALONG people |
| 00:48.27 | derek_1234 | I have. |
| 00:48.40 | twisted | okay. |
| 00:48.40 | derek_1234 | jUst wondering when we will find a new topic. |
| 00:48.54 | twisted | I have a great new topic |
| 00:48.58 | twisted | everyone go find a bug |
| 00:49.01 | twisted | http://bugs.digium.com |
| 00:49.10 | Darwin35 | your coming with us young man |
| 00:49.15 | derek_1234 | Question:: If I want to handle 10000 concurrent calls, will * help me ? |
| 00:49.20 | bkw_ | Ok everyone.. I apologize about the incident you have seen. I was just trying to defend my point against something that isn't the truth. For this i'll shut up knowing that Cluecon is here to truly help people this is why we gave away free passes to 3 area colleges. |
| 00:49.29 | blitzrage | everyone go write a document and submit it to www.asteriskdocs.org |
| 00:49.29 | Darwin35 | halling BKW off to Happy acres |
| 00:49.31 | twisted | bkw...... |
| 00:49.37 | bkw_ | and in my famous words... NEXT!!! |
| 00:49.42 | twisted | haha |
| 00:49.44 | twisted | ANYWAY |
| 00:49.54 | twisted | if we want to see 1.2 anytime soon |
| 00:50.12 | twisted | i'd suggest we focus some of the attention on the bugtracker, and knock some stuff out. |
| 00:50.14 | bkw_ | but don't bother trying to get anything actually commited you'll be called a whiney bitch... |
| 00:50.17 | JerJer | bkw_: shall i bring out the logs of my convo with mark just last night? |
| 00:50.18 | bkw_ | doh |
| 00:50.21 | twisted | GAH |
| 00:50.29 | bkw_ | JerJer why? |
| 00:50.33 | JerJer | "i feel obligated" |
| 00:50.38 | JerJer | mark's words |
| 00:50.39 | bkw_ | I have one that will trump that one |
| 00:50.45 | JerJer | sorry mark |
| 00:50.51 | Darwin35 | bkw go to your room |
| 00:51.28 | twisted | tell ya what |
| 00:51.33 | twisted | go create #asterisk-debate |
| 00:51.44 | twisted | then anyone who wants to see this can go there |
| 00:51.54 | JerJer | i took it private |
| 00:51.59 | twisted | i know you did |
| 00:52.04 | derek_1234 | Noh. |
| 00:52.09 | niZon | i think freenode likes ## for unofficial channels |
| 00:52.11 | twisted | i'm not pointing fingers |
| 00:52.19 | derek_1234 | why not have a #asterisk-pick-on-jerjer room ? |
| 00:52.29 | derek_1234 | which is not nice. |
| 00:52.52 | postel | twisted: It seems there's soem kind of disagreement, why do you try so hard to put it under the ground |
| 00:53.00 | twisted | postel, because this is not the place for it |
| 00:53.15 | *** join/#asterisk SarahEmm (~sarahemm_@2.35.220-216.q9.net) |
| 00:53.27 | blitzrage | SarahEmm: evening |
| 00:53.31 | SarahEmm | hihi! |
| 00:53.33 | twisted | postel, and, because bkw_ and JerJer are both good friends, and I _HATE_ to see them fighting |
| 00:53.34 | SarahEmm | sivana: you' round? |
| 00:53.38 | niZon | this is the place for asterisk talk, yes yes yes |
| 00:53.50 | niZon | has anyone use use asterisk to interface with real world hardware |
| 00:53.55 | niZon | such as X10 |
| 00:53.58 | niZon | and whatnot |
| 00:54.01 | sivana | yes |
| 00:54.04 | postel | twisted: true, just let both side paste what they got and truth will shine, why you're trying that hard? |
| 00:54.10 | postel | s/side/sides |
| 00:54.13 | SarahEmm | niZon: no, but what's the question? |
| 00:54.22 | twisted | postel, no, again, this is NOT the place for it |
| 00:54.34 | niZon | SarahEmm: just wondering what people have done |
| 00:54.38 | SarahEmm | niZon: ahh |
| 00:55.04 | postel | twisted: bkw_ is also an op and decided to bring it here, what makes tou more authoritative of whats that # for? |
| 00:55.11 | postel | s/tou/you |
| 00:55.21 | blitzrage | postel: #asterisk <-- topic. |
| 00:55.29 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 00:55.30 | twisted | postel, if you don't like it, go away |
| 00:55.34 | twisted | postel, simple as that |
| 00:55.40 | blitzrage | twisted: I think he does like it :) |
| 00:55.46 | postel | twisted: oh, so now the problem is me, any more ad hominem attacks? |
| 00:56.58 | twisted | postel, no, i have not attacked you, and I do not plan to. I'm simply stating that this is not the place for what was going on. This is a channel craeated to help asterisk, and people using asterisk - not to flame one another. |
| 00:57.15 | twisted | *help asteris users |
| 00:57.20 | twisted | **asterisk |
| 00:57.22 | niZon | +k |
| 00:57.25 | niZon | :P |
| 00:57.36 | JerJer | then why are there ads for two conf's in the topic? |
| 00:58.03 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 00:58.06 | postel | They're not ads, they're "information for the community" (tm) |
| 00:58.09 | twisted | JerJer, information resources. since both cons have to deal with asterisk, it's just an announcement |
| 00:58.14 | twisted | kram! |
| 00:58.52 | niZon | oh, he is here, now there's going to be some ass kicking |
| 00:59.06 | kram | i'm staying out of it |
| 00:59.26 | derek_1234 | So, can * be made to support 10000 concurrent calls ? |
| 00:59.32 | derek_1234 | or even 1000 ? |
| 00:59.41 | derek_1234 | or is it just a toy ? |
| 00:59.47 | MikeJ[Laptop] | heh... |
| 01:00.10 | MikeJ[Laptop] | hmmm... off to work on the bugtracker... |
| 01:00.17 | SarahEmm | have fun MikeJ[Laptop]! |
| 01:00.18 | MikeJ[Laptop] | it's at bugs.digium.com. |
| 01:00.19 | niZon | if your machine has the horsepower i bet it could do a few calls.. |
| 01:00.20 | derek_1234 | you mean it will do 10 concurrent calls ? |
| 01:00.41 | MikeJ[Laptop] | for those of you who don't know it.. go and look around... |
| 01:00.45 | SarahEmm | MikeJ[Laptop]: i'll look at the frame allocation bug i have open, when i opened it i said i didn't have enough code experience with * to be able to figure out the issue myself, and could someone else look into it |
| 01:00.49 | MikeJ[Laptop] | find a bug that might affect you. |
| 01:00.53 | MikeJ[Laptop] | test a patch |
| 01:00.54 | SarahEmm | but nobody else is, and it's been a few months now, so i have the experience :) |
| 01:00.56 | MikeJ[Laptop] | comment on the bug |
| 01:01.15 | MikeJ[Laptop] | SarahEmm, :) |
| 01:01.32 | SarahEmm | M4632, for example. that's a good starting point for people. ;) |
| 01:01.32 | niZon | derek_1234: http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 01:02.38 | derek_1234 | thanks |
| 01:02.43 | derek_1234 | i appreciate. |
| 01:03.27 | bkw_ | SarahEmm, yo |
| 01:03.31 | *** join/#asterisk ctjctj (ctjctj@192.55.203.130) |
| 01:03.45 | SarahEmm | hihi bkw |
| 01:03.48 | SarahEmm | what's up? |
| 01:04.05 | *** join/#asterisk Barmal (~info@c-24-30-75-206.hsd1.ga.comcast.net) |
| 01:04.19 | Barmal | exten => s,2,NVBackgroundDetect(welcome) |
| 01:04.19 | jsaunders | Any available H323/SIP commercial routes out there? Anywhere in the world? Looking for somewhere to put our traffic. |
| 01:04.29 | Barmal | what does welcome here mean? |
| 01:05.46 | hardwire | neat |
| 01:05.52 | hardwire | enabled the use of the "record" button on the snoms |
| 01:06.02 | hardwire | not a lot of good that does me. |
| 01:06.21 | colinm_ | Barmal: filename of the file to play to the caller. |
| 01:06.34 | ctjctj | Hello again. I'm having a problem with music on hold. FreeBSD 5.3-Release, Asterisk 1.0.7. mpg123 0.59r. After a day or so I had many mpg123 processes, all running and eating up CPU. Looking for a way to check MoH to see if it is actually playing anything. |
| 01:07.52 | bkw_ | exten => 999,1,MusicOnHold |
| 01:08.01 | Barmal | cilinm_, oh so you can play your greeting message at the same time waiting for fax detection? |
| 01:08.15 | ctjctj | bkw_: thanks, doing now. |
| 01:08.57 | Qwell | Barmal: many of the prompts are supposed to be used in conjunction with other ones |
| 01:09.24 | Qwell | Barmal: for instance "telemarketers are not" "welcome here" |
| 01:13.08 | Barmal | qwell: can you use nvfaxdetect(my_welcome_message) instead of nvbackgrounddetect(my_welcome_message)? |
| 01:13.27 | Qwell | dunno |
| 01:13.33 | Qwell | I don't do faxing |
| 01:13.57 | Barmal | anybody? |
| 01:19.05 | *** join/#asterisk SwK (~krice@newrso.suspicious.org) |
| 01:19.10 | MikeJ[Laptop] | ken! |
| 01:19.11 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 01:19.23 | SwK | damn its HOT outside |
| 01:19.32 | JerJer | jsaunders: $50,000 a month commit and i might be able to find you something |
| 01:19.39 | mishehu | SwK: how hot is hot? |
| 01:19.46 | SwK | and i'm pissed i forgot t bring my laptop home from work so i'm stuck on the wifes G4 |
| 01:20.05 | Barmal | Man I was pulling cables today with no condition in building was close to 100 man... |
| 01:20.07 | SwK | mishehu: well its hard out and still ~90 |
| 01:20.35 | bkw_ | SwK yo |
| 01:20.38 | SwK | Barmal: try programming in an office w/ the A/C AFU and 85+ sitting still at my desk |
| 01:20.43 | SwK | bkw_ |
| 01:21.17 | Qwell | SwK: afu? |
| 01:21.27 | SwK | AFU == All F'd Up |
| 01:21.30 | Qwell | oh |
| 01:21.41 | SwK | like SNAFU only with out the SN |
| 01:22.04 | SwK | MikeJ[Laptop] did you get someone to test that audiocodes patch? |
| 01:22.13 | Qwell | SwK: Palm Springs, 115F, hotel - ceiling |
| 01:22.17 | SwK | heh |
| 01:22.46 | SwK | oh and dont forget south east Humidity |
| 01:22.48 | Qwell | its funny... |
| 01:22.58 | Qwell | because 90% of you understand what I just said...but not many others would |
| 01:23.13 | derek_1234 | Association for Creation Riduculos Odd Names for Your Mystification (acronym) |
| 01:23.16 | MikeJ[Laptop] | SwK, not yet... |
| 01:23.30 | Barmal | yep I used to work in south florida doing cabling sometimes outside building.... |
| 01:23.35 | MikeJ[Laptop] | need those tested seperately, and together |
| 01:23.41 | SwK | MikeJ[Laptop]: i gotta run back to the office to grab my PB... I'll grab one of the MP108s while i'm there... |
| 01:23.48 | MikeJ[Laptop] | :) |
| 01:23.49 | Barmal | way too hot... |
| 01:23.53 | MikeJ[Laptop] | love ya !! |
| 01:23.59 | SwK | I still gotta figure out how to make that bitch boot normally with DHCP |
| 01:24.19 | SwK | the closest thing I have gotten so far is BOOTP and that puts the f'er into recovery mode |
| 01:27.08 | *** join/#asterisk riksta (~rick@84.93.243.170.broadband.plus.dyn.plus.net) |
| 01:27.09 | *** part/#asterisk ctjctj (ctjctj@192.55.203.130) |
| 01:28.11 | *** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net) |
| 01:30.33 | *** join/#asterisk znoG (~gs@200.115.216.109) |
| 01:33.24 | *** join/#asterisk lters (~lters@mrtcdsl-034.mis.net) |
| 01:36.44 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 01:37.08 | *** join/#asterisk fugitivo (~ajf@201.255.99.157) |
| 01:38.45 | niZon | hmm |
| 01:38.59 | niZon | i just had an urge to write a php interface to the asterisk manager |
| 01:39.04 | niZon | but it would lack realtimeness |
| 01:39.13 | ManxPower | niZon: resist |
| 01:39.24 | niZon | yes |
| 01:39.38 | Qwell | I'm gonna write an AGI interface to it |
| 01:39.47 | Qwell | just because it would be extremely pointless |
| 01:40.42 | hardwire | grr |
| 01:40.50 | hardwire | how do I change the callback extension for voicemail? |
| 01:40.55 | hardwire | instead of it just being "asterisk" |
| 01:41.43 | *** join/#asterisk Alecsandro (~ale@200.189.53.10) |
| 01:44.27 | hardwire | callerid=VoiceMail in sip general |
| 01:44.34 | hardwire | I wonder what all that will make weird things happen to |
| 01:48.12 | mishehu | there is a disturbance in teh force. |
| 01:49.48 | *** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com) |
| 01:55.35 | *** join/#asterisk vartanZT (~vartan@24-55-1-184.vnnyca.adelphia.net) |
| 01:55.36 | vartanZT | hi |
| 01:55.39 | lters | anyone test the te411 |
| 01:55.52 | shmaltz | mishehu, mah korah? |
| 01:56.00 | shmaltz | eizah machatz? |
| 01:56.15 | vartanZT | anyone can show me a sample incoming settings please ? |
| 01:56.25 | shmaltz | vartanZT, I'm sure the wiki can |
| 01:56.29 | shmaltz | ~wiki |
| 01:56.30 | vartanZT | wiki ? |
| 01:56.35 | vartanZT | url plz? |
| 01:56.39 | lters | voip-info.org |
| 01:56.56 | JunK-Y | ~wikis |
| 01:56.57 | jbot | it has been said that wikis is http://www.voip-info.org |
| 01:57.01 | shmaltz | try this http://www.voip-info.org/wiki-asterisk+tips+and+tricks |
| 01:57.33 | mishehu | shmaltz: shover et harosh... yesh baAyah Im eyzeh server... konan iomega rev nafal mibus ha'firewire. |
| 01:59.00 | shmaltz | mishehu, rak tishmor shelo tishbor et hiomega :) |
| 01:59.36 | mishehu | shmaltz: al da'agah... haserver eyno baHader... |
| 02:00.10 | shmaltz | mishehu, eich omrim lol bivrit? |
| 02:00.32 | mishehu | shmaltz: mitpotsets mitsHok |
| 02:00.47 | shmaltz | mishehu, MM |
| 02:00.50 | shmaltz | :) |
| 02:01.55 | shmaltz | mishehu, ata poh oi bearatz? |
| 02:02.07 | mishehu | b'artsot habrit. |
| 02:02.14 | mishehu | hayati b'arets lifnay kaHodesh |
| 02:02.56 | shmaltz | achoti amra li, sh'yesh shama colorwar achshav, katom neged vkachol ba'ad, shmata al zeh? |
| 02:04.37 | *** join/#asterisk jiro5281 (~anton@210.213.95.226) |
| 02:04.56 | mishehu | shmaltz: ra'iti harbeh katom sham. lo ra'iti kaHol.... ulay hakaHol yoter Hadash... |
| 02:05.41 | shmaltz | nu, nu, ata ba'ad or neged? |
| 02:05.42 | mishehu | Avarti et nitsanim, Adayin bonim sham. Hashavti shehigdiru et hakarkA kapark le'umi... |
| 02:05.53 | jiro5281 | hi guys...a friend of mine refer me to this channel about asterisk....has any one had use vicidial? im having a problem making it work |
| 02:06.07 | mishehu | vicidial? not heard of it. |
| 02:06.18 | mishehu | sounds like dial for vicodin |
| 02:06.29 | jiro5281 | its an autodialer |
| 02:06.39 | SwK | vicidial ehhehe |
| 02:06.46 | jiro5281 | how bout gnudialer? |
| 02:06.46 | SwK | who wrote that thing? |
| 02:06.47 | MikeJ[Laptop] | jbot: vicdial |
| 02:06.49 | *** join/#asterisk santiago (~santiago@63.245.86.175) |
| 02:06.50 | shmaltz | jiro5281, why you using vicidial? |
| 02:07.02 | mishehu | shmaltz: neged. efshar lenatsel et Azah b'tahalich hahitmakHut... (zeh lo memash tahalich hashalom) |
| 02:07.03 | shmaltz | ~vicidial |
| 02:07.15 | jiro5281 | to autmate dialing in our office |
| 02:07.50 | shmaltz | jiro5281, why don't you hire someone that knows how to set this up? |
| 02:07.54 | shmaltz | if thats your business |
| 02:07.56 | shmaltz | ? |
| 02:08.17 | jiro5281 | coz my boss wants to check on first... |
| 02:08.24 | jiro5281 | if its worth |
| 02:09.00 | jiro5281 | http://astguiclient.sourceforge.net/vicidial.html heres the link if anyone is interested also |
| 02:12.31 | shmaltz | mishehu, tagid li, ata dati? mesorati? |
| 02:12.41 | mishehu | Hiloni |
| 02:12.42 | harryvv | check on first what? |
| 02:13.21 | harryvv | jiro, do you have a pbx? |
| 02:14.03 | Coriantum | How can I exit a macro in a dialplan? |
| 02:14.27 | JerJer | Coriantum: don't have any more priorites ? |
| 02:14.32 | JerJer | priorities |
| 02:14.56 | Coriantum | Trying to *cough* get around a bug in AEL |
| 02:15.16 | Coriantum | and I know you're not a fan of AEL |
| 02:17.00 | shmaltz | v'ata nege, tov me'od, ani dati (shchor kacha.. :)) |
| 02:17.17 | opus_ | how can i switch context in an extension in realtime |
| 02:17.26 | *** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
| 02:17.32 | *** join/#asterisk likwid-- (~likwid@nc-69-34-157-158.dyn.sprint-hsd.net) |
| 02:17.36 | opus_ | Goto doesn't seem to find the right context |
| 02:17.42 | shmaltz | aval b'derech klal ani neged mah sh'hachriedim rotzim b'keneset |
| 02:18.30 | harryvv | what language is that? |
| 02:18.45 | JerJer | jibberish |
| 02:18.50 | shmaltz | harryvv, hebrew |
| 02:19.06 | JerJer | wholy wrong character set |
| 02:19.07 | harryvv | was salam is the onlything i know of |
| 02:19.11 | shmaltz | hem rotizm rak kesef v'od hapam kesef |
| 02:19.18 | *** join/#asterisk SkramX (~mark@mark-s.net) |
| 02:19.22 | SkramX | Hello All. |
| 02:19.23 | shmaltz | yep, JerJer |
| 02:19.24 | mishehu | harryvv: salam is the other language. |
| 02:19.37 | harryvv | okay |
| 02:19.39 | harryvv | ;) |
| 02:19.57 | shmaltz | haryvv, shalom is how you say it in Hebrew |
| 02:20.09 | shmaltz | ~shalom |
| 02:20.16 | SkramX | Heh |
| 02:20.22 | SkramX | Shalom, Y'all |
| 02:20.27 | SkramX | Mah Nishma? |
| 02:20.28 | Beirdo | shalom! |
| 02:20.40 | Beirdo | heya, mishehu, BTW :) |
| 02:20.41 | shmaltz | ~shalom |
| 02:20.41 | jbot | somebody said shalom was how you greet a jew in america, the isrealies don't use it ever |
| 02:20.54 | SkramX | Nope. |
| 02:20.59 | Beirdo | that's a typo |
| 02:21.02 | Beirdo | fix that |
| 02:21.03 | shmaltz | yep it is |
| 02:21.07 | Beirdo | Israelites |
| 02:21.08 | mishehu | shmaltz: atah dos? |
| 02:21.14 | shmaltz | yep |
| 02:21.18 | Beirdo | and I'm not even Jewish and I know that |
| 02:21.19 | shmaltz | ken, ani dati |
| 02:21.31 | SkramX | Its hebrew, many Israeli's dont talk proper Hebrew |
| 02:21.38 | SkramX | So.. I need help with my 12SP+ |
| 02:21.43 | shmaltz | shchor, mah shatah roah b'meah sharim |
| 02:21.58 | shmaltz | shchor = shachor |
| 02:22.14 | opus_ | can somebody tell me why my Goto doesn't work in asterisk? |
| 02:22.16 | opus_ | http://pastebin.ca/18569 |
| 02:22.25 | SkramX | I have it connected to my router (No Firewall) |
| 02:22.30 | SkramX | opus_ il look, one second |
| 02:22.51 | SkramX | I havent played with Realtime yet... |
| 02:22.52 | shmaltz | opus_, it can't find extension 1 |
| 02:22.59 | shmaltz | do you have exten => 1,1 there? |
| 02:23.16 | SkramX | So.. my 12SP+ I can talk and people can here me, but I cant hear anything but the dialtone... |
| 02:23.26 | SkramX | I have it on DMZ so its not a port problem.. I dont think |
| 02:23.34 | shmaltz | SkramX, where here in IRC? or on the phone? |
| 02:23.59 | SkramX | on the phone. |
| 02:24.38 | JerJer | SkramX: what channel driver? |
| 02:24.49 | SkramX | im just using skinny.conf |
| 02:25.00 | JerJer | do you have a valid bindaddr in there ? |
| 02:25.09 | SkramX | Yes. |
| 02:25.15 | JerJer | ulaw ? |
| 02:25.21 | SkramX | Not sure |
| 02:25.24 | SkramX | how do i define |
| 02:25.25 | JerJer | allow=ulaw |
| 02:25.30 | SkramX | want me to post my skinny.conf? |
| 02:25.37 | JerJer | to a pastebin yes |
| 02:25.38 | opus_ | shamltz, Yes |
| 02:25.40 | JerJer | and the CLI interfece |
| 02:25.42 | JerJer | er |
| 02:25.45 | JerJer | CLI details |
| 02:25.49 | SkramX | does that go in the fone's context or context:defaukt. |
| 02:25.55 | SkramX | JerJer: ^ |
| 02:26.02 | opus_ | shamltz, but it is in a database table with res_config_mysql/realtime |
| 02:26.44 | SkramX | JerJer: I am going to try to see if allow=ulaw fixes it.. |
| 02:26.53 | SkramX | but which context do i put it in? |
| 02:27.41 | *** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net) |
| 02:27.52 | SkramX | ok? brb |
| 02:28.08 | JerJer | general |
| 02:29.15 | SkramX | ooo general? |
| 02:29.18 | SkramX | lemme do it again |
| 02:29.28 | *** join/#asterisk HellAgony (~HellAgony@200.121.241.78) |
| 02:30.32 | SkramX | JerJer: no luck |
| 02:34.42 | SkramX | http://pastebin.ca/18570 |
| 02:34.51 | SkramX | Ok, JerJer, and anyone, PLEASE Help! |
| 02:35.44 | SkramX | haha |
| 02:42.40 | opus_ | hmmm |
| 02:42.46 | opus_ | Goto is broken for realtime in cvs 5 day ago |
| 02:44.53 | SkramX | thats not good |
| 02:44.59 | SkramX | I still need help :( |
| 02:46.42 | opus_ | Whats wrong |
| 02:47.08 | *** join/#asterisk vtsherwood (user@cpe-24-210-53-246.columbus.res.rr.com) |
| 02:47.26 | vtsherwood | Brandon Price, John Reyes, PM me and let's create a private room |
| 02:49.24 | *** join/#asterisk bprice20 (~bprice20@cpe-24-194-236-170.nycap.res.rr.com) |
| 02:49.59 | Alecsandro | what softphone is possible to using in freebsd with asterisk ? |
| 02:50.34 | harryvv | try xlite |
| 02:50.39 | *** part/#asterisk vtsherwood (user@cpe-24-210-53-246.columbus.res.rr.com) |
| 02:52.14 | Alecsandro | harryvv xten ? |
| 02:55.22 | *** join/#asterisk vtsherwood (user@cpe-24-210-53-246.columbus.res.rr.com) |
| 02:56.02 | *** join/#asterisk JKR (JKR@cpe-69-207-59-59.nycap.res.rr.com) |
| 02:57.19 | *** join/#asterisk bprice20 (~bprice20@cpe-24-194-236-170.nycap.res.rr.com) |
| 02:57.35 | SkramX | opus_ i cant hear output from Asterisk on my 12SP+ |
| 02:58.12 | vtsherwood | JKR join vtchat |
| 03:01.27 | opus_ | 12SP+ |
| 03:01.28 | opus_ | ? |
| 03:01.38 | *** part/#asterisk vtsherwood (user@cpe-24-210-53-246.columbus.res.rr.com) |
| 03:04.45 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 03:05.39 | opus_ | whoah |
| 03:05.50 | opus_ | buy one get one free pizza yeah |
| 03:11.15 | niZon | now if only they had that for IP phones |
| 03:14.08 | Moc | lol |
| 03:15.27 | JerJer | SkramX: sorry got sucked into a few different private conversations |
| 03:16.01 | JerJer | aparently tonights cage match motivated some people |
| 03:16.15 | JerJer | which was my intent |
| 03:16.56 | *** part/#asterisk bprice20 (~bprice20@cpe-24-194-236-170.nycap.res.rr.com) |
| 03:17.13 | *** join/#asterisk Lathos42 (~Lathos42@68.77.108.51) |
| 03:21.31 | opus_ | hmm i need more asterisk sounds |
| 03:21.52 | shmaltz | opus_, then do cvs co asterisk-sounds |
| 03:21.56 | *** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122) |
| 03:22.21 | *** part/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
| 03:23.15 | opus_ | hmm |
| 03:23.39 | *** join/#asterisk HellAgony (~HellAgony@200.121.241.78) |
| 03:27.28 | bkw_ | Lathos42, I have some i'll share at cluecon at are great too |
| 03:28.00 | MikeJ[Laptop] | :P |
| 03:28.16 | Darwin35 | lol |
| 03:28.37 | Darwin35 | Swk whats up |
| 03:28.52 | *** join/#asterisk bullweivel (~BullWeive@12.165.69.254) |
| 03:29.15 | bullweivel | what would you guys pay for a used cisco 7960 phone ??? just curious what the going used rate is. |
| 03:29.25 | blitzrage | $200 |
| 03:29.33 | blitzrage | $250 if it was refurbished |
| 03:29.41 | JerJer | with or without a cisco license :P |
| 03:29.44 | Darwin35 | look on ebay |
| 03:29.47 | JerJer | or power cube |
| 03:29.57 | MikeJ[Laptop] | or a monkey |
| 03:30.03 | blitzrage | fuck the power cube, I have a knock off and it works fine |
| 03:30.06 | MikeJ[Laptop] | ohhhh monkey... |
| 03:30.07 | bullweivel | who cares about the cisco license.... its going on a * box :) |
| 03:30.15 | JerJer | cisco :) |
| 03:30.19 | file | blitzrage: Leifffffff |
| 03:30.26 | blitzrage | or just get those $80 NetGear 4 port PoE adapaters |
| 03:30.42 | blitzrage | whats a cisco license? :) |
| 03:30.50 | bullweivel | realy ?? you have to get a license to use sip on the damn phone?? hold... those liceses are for call manager |
| 03:31.02 | bullweivel | it even reads on them for call manager...at least the ones we have at work do. |
| 03:31.06 | JerJer | sip license |
| 03:31.09 | Darwin35 | cool my new x401's will be here tomarrow |
| 03:31.10 | bkw_ | fuck cisco and its stupid license |
| 03:31.13 | bkw_ | I hate that idea |
| 03:31.30 | *** join/#asterisk Saaib (~nabudocon@ns1.ensenada.gob.mx) |
| 03:31.39 | bullweivel | cisco vs what?? what else has this kick ass lcd on it? (wish it was color) |
| 03:31.51 | bkw_ | the cisco is far from kick ass on the LCD |
| 03:31.57 | shmaltz | bullweivel, the 7970 has color |
| 03:31.58 | fugitivo | why you need a phone with an lcd like that? |
| 03:32.01 | JerJer | 7970 |
| 03:32.06 | blitzrage | I just wish it had a backlight |
| 03:32.06 | Darwin35 | what do you need the lcd for other then caller id |
| 03:32.07 | shmaltz | polycom has this lcd as well |
| 03:32.11 | Darwin35 | its a waste |
| 03:32.14 | fugitivo | just use a normal pc and a normal telephone |
| 03:32.15 | bullweivel | 7970 dont do sip (or so i was told) |
| 03:32.23 | blitzrage | anyone know how that guy made out with adding a backlight to his 7960? |
| 03:33.14 | JerJer | didn't hear about that hack |
| 03:33.33 | Lathos42 | I wish Cisco supported standard PoE with the 7940 and 7960 |
| 03:33.53 | JerJer | guess i need to probe around for a logic point to time out the light on idle |
| 03:34.05 | bullweivel | yeah... the poe crap is kinda screwed up on the damn phones! |
| 03:34.42 | JerJer | wonder if we could use one of those EL panels ? |
| 03:35.31 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca) |
| 03:35.35 | blitzrage | I'm thinking like the timex watches |
| 03:35.37 | SarahEmm | hihi |
| 03:35.40 | blitzrage | SarahEmm: good eve |
| 03:35.45 | SarahEmm | 'tis, mostly. |
| 03:35.59 | blitzrage | true |
| 03:36.26 | Strom_C | hi |
| 03:36.31 | bullweivel | lathos... you can use it... just have to change the wiring around LOL |
| 03:37.20 | Lathos42 | I was under the impression that your PoE switch had to be a dumb injector for the rewire trick to work |
| 03:37.28 | Lathos42 | and I dont know if the Dell switch is that way |
| 03:37.52 | TheEmperor | hi |
| 03:38.01 | TheEmperor | what's a good laser printer which works well with redhat 9? |
| 03:38.10 | SarahEmm | TheEmperor: this is SO not the right channel for that. |
| 03:38.16 | TheEmperor | o |
| 03:38.16 | TheEmperor | ok |
| 03:38.17 | shmaltz | TheEmperor, try #printers-rh9 |
| 03:38.19 | SarahEmm | TheEmperor: ask in a redhat channel or #linpeople or something |
| 03:38.24 | TheEmperor | ok thanks |
| 03:39.29 | vartanZT | can someone help me please iv tryed everything but when i get incoming calls they get fast busy |
| 03:39.32 | vartanZT | please someone ? |
| 03:39.39 | blitzrage | lol... laser printers in #asterisk? thats funny |
| 03:39.45 | JerJer | vartanZT: with what channel type? |
| 03:39.51 | vartanZT | SIP |
| 03:39.53 | shmaltz | vartanZT, how do they come in? what context |
| 03:40.09 | brookshire | TheEmperor: any network printer :) |
| 03:40.21 | brookshire | that supports ps |
| 03:40.35 | vartanZT | context should be context=from-pstn right ? |
| 03:40.41 | JerJer | SkramX: allow=ulaw,gsm is bad |
| 03:40.44 | JerJer | allow=ulaw |
| 03:40.50 | JerJer | chan_skinny only supports ulaw |
| 03:41.01 | fugitivo | TheEmperor: any printer with postcript chip |
| 03:41.03 | JerJer | and those phones only support ulaw and possibly g.729, depending on what version |
| 03:41.19 | JerJer | [global] |
| 03:41.23 | vartanZT | shmaltz context=from-pstn is what i set it to, is that incorrect ? |
| 03:41.23 | JerJer | include => skinny ?!?!?! |
| 03:41.29 | JerJer | what kind of crack is that ? |
| 03:41.41 | shmaltz | vartanZT, it realy depens on you |
| 03:41.54 | vartanZT | what do you mean please explain ? |
| 03:42.26 | *** join/#asterisk brettnem (~Brett@user-0ccsrag.cable.mindspring.com) |
| 03:42.38 | JerJer | vartanZT: type sip debug into the asterisk cli |
| 03:42.45 | brettnem | good evening all |
| 03:42.50 | vartanZT | ok |
| 03:43.25 | vartanZT | so you want to see the logs ? |
| 03:43.46 | brettnem | can anyone point me to a good place to buy some cisco phones in bulk?? |
| 03:44.28 | brettnem | anyone?? |
| 03:44.30 | Strom_C | ebay |
| 03:44.44 | bullweivel | so $200 and $15 in shipping isnt a bad price for a 7960 ??? |
| 03:44.48 | shmaltz | brethtnem, there was a post today or yesterday from cory on asterisk-biz about this |
| 03:44.52 | brettnem | I'm looking for a reliable source of phone that'll come usable.. |
| 03:45.03 | brettnem | bullweivel: where'd you see that deal? |
| 03:45.04 | blitzrage | not bad - great price if it comes iwth a power block (of any type) |
| 03:45.21 | brettnem | shmaltz: oh yeah?? I'll have to dig that up.. |
| 03:45.23 | *** join/#asterisk Innismir (~ratchet@dsl092-072-025.bos1.dsl.speakeasy.net) |
| 03:45.34 | Strom_C | hello ratchet :) |
| 03:45.38 | bullweivel | damn it |
| 03:45.42 | bullweivel | i hate ebay some time. |
| 03:45.45 | bullweivel | went for 187.50 |
| 03:45.54 | bullweivel | stupid thing wouldnt let me bid quick enough... oh well. |
| 03:46.17 | Strom_C | broadband + stopwatch == ebay fun |
| 03:46.32 | bullweivel | LOL... aint that the truth. |
| 03:46.34 | brettnem | I know I saw one of the main voip supply websites selling the 7960s in packs of 10 and 25.. but I can't find it.. anyone know?? |
| 03:46.41 | Strom_C | voipsupply.com |
| 03:46.54 | brettnem | yeah, that's what I thought, but I can't find the bulks anymore.. |
| 03:47.01 | bullweivel | a few of them have them on ebay in 10 and 20 packs.. i didnt think the prices were that good though. |
| 03:47.12 | shmaltz | bullweivel, I usualy put in my maximum (in this case it would've been $221) like < 1 min b4 end of auction |
| 03:47.37 | Innismir | I wait until about 15s |
| 03:47.42 | bullweivel | New 5pk Cisco 7960 Phones w/ USER LICENSE CP-7960 $1,425.00 |
| 03:48.00 | Innismir | I need to find a cheap 7905 |
| 03:48.02 | vartanZT | shmaltz ok now what ? |
| 03:48.03 | bullweivel | guess...that aint a bad deal... it has 11 hours left |
| 03:48.11 | shmaltz | gtg guys |
| 03:48.12 | shmaltz | c ya |
| 03:48.21 | bullweivel | shmaltz: yeah i put in 200 but didnt realize it only had 10 seconds left... oops |
| 03:48.31 | shmaltz | my wife is going to brake my * box if I don't come now |
| 03:48.37 | bullweivel | i hate that damn... confirm your bid crap too!!! when did they start that...very annoying. |
| 03:48.48 | Strom_C | they started that ages ago |
| 03:49.00 | shmaltz | gtg, c ya guys |
| 03:49.00 | Innismir | 2000 or so |
| 03:49.03 | shmaltz | gn |
| 03:49.30 | NewSole | hmmm she has him by the **** |
| 03:49.52 | bkw_ | GOOD NIGHT INKERNET |
| 03:49.55 | bullweivel | he has four * boxes ?? ;) |
| 03:50.09 | Innismir | RAI* ? |
| 03:50.13 | Innismir | :) |
| 03:50.21 | file | HARD! |
| 03:50.48 | bkw_ | its the dawn and drew show |
| 03:52.04 | bkw_ | every time someone says Internet, my mind substitutes Inkernet. Curse you Dawn Miceli! |
| 03:52.22 | brettnem | Inkernet |
| 03:52.24 | brettnem | ? |
| 03:52.35 | bkw_ | you need to listen to the dawn and drew pod cast |
| 03:52.40 | brettnem | hey bkw_ what's up? |
| 03:52.43 | brettnem | what is it? |
| 03:52.58 | bkw_ | http://www.dawnanddrew.com/ |
| 03:53.00 | *** part/#asterisk bullweivel (~BullWeive@12.165.69.254) |
| 03:53.27 | mishehu | bah. |
| 03:54.07 | brettnem | hmm.. I'll check it out |
| 03:54.16 | bkw_ | its adult only |
| 03:54.53 | Moc | lol |
| 03:54.59 | Moc | no thanks |
| 03:55.08 | Moc | Innismir, get Polycom phone instead.. |
| 03:56.06 | brettnem | oh, I'm probably too young for it then.. ;) |
| 03:56.14 | brettnem | ew polycom |
| 03:56.20 | brettnem | I can't find a damn phone I like |
| 03:56.30 | blitzrage | Polycom phones are nice |
| 03:56.32 | brettnem | wish I could get my 7920 to work with asterisk |
| 03:56.44 | brettnem | blitzrage: they are ok.. I use one in my office.. get weird problems sometimes. |
| 03:56.47 | Moc | brettnem, polycom only missing to be opensource firmware.. |
| 03:56.56 | blitzrage | brettnem: all phones have "wierd problems" :) |
| 03:57.03 | brettnem | yeah I guess so.. |
| 03:57.13 | blitzrage | only missing STUN and mini-browser (at least on my ip500) |
| 03:57.16 | brettnem | I'm pretty happy with my IP500 at 135.. heh.. don't know how I pulled that |
| 03:57.35 | blitzrage | I got a free one :) |
| 03:57.37 | JunK-Y | IP500 are great phones. |
| 03:57.39 | Moc | I got a IP 500 and IP 600... I love both of them... |
| 03:57.41 | remmo | has anyone got zaptel/libpri an a te100p working under freebsd? |
| 03:57.44 | brettnem | I get that weird problem sometimes where I keep hearing ringing when the call is actually conneted. |
| 03:58.01 | brettnem | I'm half baked on the IP500.. but my firmware is old.. so I'll stop knockin it |
| 03:58.14 | brettnem | oh.. free is good.. |
| 03:58.21 | Moc | brettnem, this is problem when you havent configured your nat correctly.. |
| 03:58.34 | brettnem | I would have ordered more than one if I got that pricing.. |
| 03:58.50 | Moc | IP 600 for 255$ US |
| 03:58.53 | brettnem | Moc: really? I didn't think NAT would do that.. the phone simply shouldn't ever do that.. |
| 03:59.03 | Moc | if you want 500 of them.. |
| 03:59.14 | brettnem | like the phone is playing ringback, but if I talk, the remote party hears me |
| 03:59.14 | vartanZT | Please someone help me, I have setup outgoing but my incoming gives a fast busy ( " Executing Congestion("SIP.....") |
| 03:59.18 | Moc | brettnem, it signaling that stop working correctly |
| 03:59.19 | vartanZT | anyone pleasE ? |
| 03:59.50 | brettnem | Moc: right, I'd expect that.. but either the signalling works or it doesn't.. why would a INVITE make it through NAT but the ACK doesn't?? |
| 04:00.02 | brettnem | vartanZT: you'll have to provide a little more info |
| 04:00.29 | vartanZT | brettnem what would you like ? |
| 04:00.45 | Moc | well I remember having this problem, and after fixing my nat, it went away.. |
| 04:01.05 | brettnem | Moc: it wouldn't suprise me.. but it still baffles me |
| 04:01.13 | brettnem | wish I had a trace of that.. |
| 04:01.25 | brettnem | vartanZT: you dialplan and the exact error text in a pastebin |
| 04:01.29 | brettnem | you+are |
| 04:01.33 | Moc | my cat (linux) want me to go to sleep... and I think I should .. |
| 04:01.42 | blitzrage | I'm going to bed soon too |
| 04:01.44 | brettnem | argh... |
| 04:01.52 | vartanZT | brettnem please give me url for pastebin |
| 04:01.59 | `Sauron | Hum, why's my regular pots phone continuing to ring after * has started the switch on zap/1 |
| 04:01.59 | brettnem | ~pastebin |
| 04:01.59 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 04:02.38 | `Sauron | Hum. |
| 04:03.27 | *** part/#asterisk Lathos42 (~Lathos42@68.77.108.51) |
| 04:03.36 | vartanZT | brettnem : http://pastebin.ca/18575 is that enough? |
| 04:04.02 | brettnem | lets see. |
| 04:04.48 | brettnem | no.. need more.. and show me your dial statement from extensions.conf |
| 04:05.11 | hardwire | hi |
| 04:05.17 | brettnem | hello |
| 04:05.22 | SkramX | anytone help me with my 12SP+? I cant hear what asterisk outputs! |
| 04:05.23 | hardwire | hi |
| 04:05.26 | SkramX | http://pastebin.ca/18570 |
| 04:05.32 | hardwire | 12SP+ ? |
| 04:05.41 | brettnem | "what asterik outputs" ?? |
| 04:05.50 | hardwire | oooooh skinny |
| 04:06.04 | brettnem | maaayyybe |
| 04:06.31 | vartanZT | brettnem: http://pastebin.ca/18576 |
| 04:07.14 | brettnem | vartanZT: ok, give me a hint.. what are you dialing to get this? |
| 04:07.30 | remmo | brettnem: i wish i could get my 7960 working too |
| 04:07.31 | vartanZT | you want my TElephone number ? |
| 04:07.33 | brettnem | and from what? |
| 04:07.41 | brettnem | remmo: 7960+sccp? |
| 04:07.52 | vartanZT | from my home phone i am dialing my VoIP service telephone number |
| 04:07.58 | brettnem | vartanZT: I want to know what is being dialed.. what hits the dialplan. |
| 04:08.01 | brettnem | the incomming digits |
| 04:08.02 | Innismir | SkramX: have you tried another handset? |
| 04:08.13 | Innismir | Occam's Razor |
| 04:08.14 | vartanZT | brettnem so the telephone number |
| 04:08.16 | vartanZT | ? |
| 04:08.48 | brettnem | vartanZT: you sent me 642 lines of code... I don't need to know your phone umber.. but I need to know what line you expect to be executed.. |
| 04:09.01 | JerJer | lolololol |
| 04:09.09 | vartanZT | brettnem: [from-trunk] |
| 04:09.18 | vartanZT | thats what i use for my context |
| 04:09.41 | brettnem | ok.. so whatever is dialing you lands in from-trunk.. |
| 04:09.49 | vartanZT | yes |
| 04:09.57 | brettnem | this looks like AMP.. eh? |
| 04:10.01 | remmo | brettnem: i was hoping for sip |
| 04:10.04 | vartanZT | yes correct |
| 04:10.14 | brettnem | remmo: 7960+ SIP should work great.. |
| 04:11.07 | brettnem | ok, well if it's a DID, then i need to see the ext-did context.. which you might want to filter before you post.. I need to see the attempted dial line.. perhaps you can give me more of the verbose debugging output |
| 04:11.10 | vartanZT | brettnem if you need any information i will tell you. im currently logged into amp |
| 04:11.42 | vartanZT | can i call you? |
| 04:11.48 | vartanZT | if you are in the states or ca that is |
| 04:11.58 | remmo | brettnem: had problems loading the image via tftp |
| 04:12.49 | vartanZT | i will post you my entire log one sec |
| 04:13.28 | SkramX | Innismir Yes. |
| 04:13.31 | SkramX | 23:08 < Innismir> SkramX: have you tried another handset? |
| 04:14.18 | vartanZT | brettnem: http://pastebin.ca/18577 hows that for info ? |
| 04:15.25 | *** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 04:16.51 | brettnem | ugh 2000 lines? |
| 04:16.56 | JerJer | lol |
| 04:16.58 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
| 04:17.01 | vartanZT | its the last 2000 lines |
| 04:17.09 | brettnem | oooohh the last 2000 lines |
| 04:17.14 | brettnem | is still |
| 04:17.16 | brettnem | 2000 lines |
| 04:17.25 | vartanZT | lol look at the last 50 :) |
| 04:17.29 | brettnem | ok, do you ever see the call hit your box? |
| 04:17.30 | vartanZT | i didnt want to miss anything |
| 04:17.35 | vartanZT | yes i do |
| 04:17.41 | vartanZT | even in the log it shows |
| 04:18.05 | Innismir | oh shit |
| 04:18.10 | Innismir | SkramX: Just try something else |
| 04:18.15 | vartanZT | 1994: Jul 27 00:12:29 VERBOSE[1550]: -- Executing Congestion("SIP/vartan1-c483", "") in new stack |
| 04:18.17 | brettnem | what line? give me a hint so I don't have to scan the whole thing |
| 04:18.23 | Innismir | SkramX: some audio just doesnt' get passed |
| 04:18.30 | vartanZT | thats showing it giving the busy signal |
| 04:18.38 | Innismir | SkramX: MoH is notorious for this in chan_skinny |
| 04:18.41 | brettnem | busy <> congestion |
| 04:18.45 | vartanZT | yes |
| 04:18.48 | Innismir | SkramX: It may just be that too |
| 04:19.44 | brettnem | you need to check your incomming context on that peer and see how it's comiing in.. |
| 04:19.45 | JerJer | notorious / |
| 04:19.47 | JerJer | ? |
| 04:19.54 | vartanZT | how do i do that ? |
| 04:19.57 | hardwire | blah |
| 04:20.13 | JerJer | smells like its time for someone to pay a consultant |
| 04:20.14 | JerJer | not me |
| 04:20.25 | brettnem | heh |
| 04:20.28 | vartanZT | lol |
| 04:20.40 | brettnem | I'm working on my karma here |
| 04:20.50 | vartanZT | brettnem please help me |
| 04:20.59 | brettnem | hey, I'm tryin.. |
| 04:21.07 | vartanZT | THank you i appreciate it |
| 04:21.23 | brettnem | look, it loos like you have your sip peer setup wrong..either that or your DID isn't programmed into your system correctly. |
| 04:21.42 | vartanZT | i didnt program any dids |
| 04:21.48 | brettnem | you should do a tethereal dump and see how the invite is coming into your box |
| 04:21.50 | brettnem | eh? what? |
| 04:21.58 | vartanZT | i need to setup a did ? |
| 04:22.02 | brettnem | oh right.. you don't have to do dids with AMP.. |
| 04:22.14 | brettnem | ok, so you have a IVR answer the calls? |
| 04:22.23 | vartanZT | IVR? |
| 04:22.27 | brettnem | like an "answer all calls" thing |
| 04:22.31 | vartanZT | yes |
| 04:22.31 | SarahEmm | sleepies time |
| 04:22.33 | SarahEmm | nini all |
| 04:22.41 | brettnem | vartanZT: what do you expect to happen when someone calls you? |
| 04:22.51 | *** join/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 04:23.12 | *** part/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
| 04:23.13 | vartanZT | brettnem: well call comes in they hear a message "press 1 for this 2 for that or the extension of the person" |
| 04:23.22 | brettnem | why would they hear that? |
| 04:23.33 | vartanZT | thats what i want to do |
| 04:23.35 | brettnem | have you programmed that? |
| 04:23.40 | hardwire | blah |
| 04:23.47 | vartanZT | not yet |
| 04:24.11 | brettnem | well.... um..so what do you expect it to do if it isn't programmed to do that; then.. ?? |
| 04:24.26 | vartanZT | well i just programmed a did and it shows busy again |
| 04:24.50 | brettnem | is it the DID that you are dialing?? that is incoming to the Box? |
| 04:24.57 | vartanZT | yes |
| 04:25.18 | brettnem | are you positive it is hitting your box the exact same way you typed it? |
| 04:25.40 | vartanZT | well if thats what im dialing isnt that how it would hit the box ? |
| 04:25.44 | brettnem | have you done a tethereal trace to be sure you get the call.. is your registration succeeding.. |
| 04:25.57 | brettnem | what makes you think it is even getting to you |
| 04:26.01 | vartanZT | yes my registration is succeeding |
| 04:26.11 | vartanZT | because it shows up in my call logs |
| 04:26.44 | brettnem | let me log into an AMP console real quick.. |
| 04:26.51 | vartanZT | ok can i call you? |
| 04:27.05 | brettnem | not unless you want to pay me. :) |
| 04:27.12 | vartanZT | :( |
| 04:27.50 | brettnem | really.. this is community support out of the goodness of my heart.. actually.. I do have another motive.. in helping the community, it is my hope they will help me when I need it. |
| 04:28.22 | vartanZT | may i pm you the information ? |
| 04:28.43 | brettnem | sure you can pm me |
| 04:29.18 | brettnem | how do you have "incoming calls" setup? |
| 04:29.35 | vartanZT | ? |
| 04:30.07 | vartanZT | to send to extension 200 |
| 04:30.21 | hardwire | how lame am I? |
| 04:30.33 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 04:30.57 | Nugget | watching it is bad. admitting it on irc is worse. :) |
| 04:32.27 | hardwire | its actually not that bad |
| 04:32.35 | hardwire | not as bad as some of the shit I have been watching recently |
| 04:34.55 | *** join/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 04:38.24 | jr352k | . |
| 04:43.15 | *** part/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 04:43.25 | *** join/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 04:51.57 | *** join/#asterisk brettnem (~Brett@user-0ccsrag.cable.mindspring.com) |
| 04:56.05 | *** join/#asterisk SwK_ (nwihve@12-219-156-206.client.mchsi.com) |
| 04:56.20 | sloPP | someone want to help me diagnose a clock slip problem? |
| 04:56.36 | sloPP | i'm running out of ideas |
| 04:56.59 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:02.42 | JerJer | sprinkle some crack on i t |
| 05:02.49 | blitzrage | crack! |
| 05:04.11 | blitzrage | ok, definately bedtime, night |
| 05:04.39 | hellop | gnite! |
| 05:04.57 | Qwell | He'll be back |
| 05:05.27 | JerJer | sloPP: details |
| 05:05.48 | MikeJ[Laptop] | crack.. hmmmm |
| 05:06.27 | sloPP | JerJer: PRI -> Vega -> Sangoma |
| 05:06.39 | sloPP | sangoma is detecting frame slips/crc errors |
| 05:06.48 | sloPP | about 40 slips/errors per day |
| 05:06.52 | sloPP | that's about all i've got :> |
| 05:08.54 | sloPP | Port Frames Bytes SLIPs Frames Bytes SLIPs CRC Error Bad Frames |
| 05:09.00 | sloPP | ISDN-2 17383 116842 25 16983 129693 25 6 3 |
| 05:09.35 | MikeJ[Laptop] | yay crack ;) |
| 05:09.37 | MikeJ[Laptop] | heh |
| 05:11.08 | Qwell | Does anybody ever not have problems with PRI? heh |
| 05:11.47 | MikeJ[Laptop] | I have periods of no problems w/ PRI |
| 05:12.02 | brenda | I haven't had problems |
| 05:12.09 | JerJer | sorry i do not support Sangoma |
| 05:12.25 | MikeJ[Laptop] | or crack habbits? |
| 05:12.38 | Qwell | crack habits are well supported here |
| 05:12.51 | brenda | JerJer: Do you support anything? All I ever see are smart ass answers from you. |
| 05:13.01 | sloPP | JerJer: it's not a sangoma question. |
| 05:13.03 | sloPP | it's a general T1 question |
| 05:13.04 | sloPP | :P |
| 05:13.13 | JerJer | brenda: then go away |
| 05:13.14 | sloPP | the sangoma is only there as an interop to the portmaster 3 |
| 05:13.23 | sloPP | since the PM3 is dense |
| 05:13.33 | brenda | JerJer: You'd like that, wouldn't you? |
| 05:13.41 | JerJer | i don't give a fuck |
| 05:13.57 | brenda | Yeah... I noticed |
| 05:14.14 | *** join/#asterisk bmay (~bam@snoopy.microcomaustralia.com.au) |
| 05:15.08 | brenda | I HIGHLY doubt it |
| 05:15.18 | sloPP | nah, jerjer likes me and my crack cookies |
| 05:15.23 | brenda | nothing negative to say about it |
| 05:16.26 | *** part/#asterisk jr352k (~jraborg@pcp03618028pcs.univde01.de.comcast.net) |
| 05:16.48 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
| 05:16.58 | hellop | So, if /quit and /connect are swimming in the ocean and /quit gets eaten by a shark, who's left? |
| 05:17.18 | dudes | Anyone know if having multiple host= in one context in sip.conf will work. |
| 05:18.05 | JerJer | dudes: try it |
| 05:18.17 | JerJer | perhaps separate with , ? |
| 05:18.24 | JerJer | host=1.2.3.4,4.5.6.7 ? |
| 05:18.26 | JerJer | guessing |
| 05:18.48 | dudes | JerJer - I figure "," or multiple will work. Can't test until tomorrow |
| 05:19.10 | dudes | or seperate contexts. |
| 05:19.16 | harryvv | is there any free dyanmic dns service i can use? |
| 05:19.23 | JerJer | gotdns.org ? |
| 05:19.27 | dudes | harryvv - myserver.org |
| 05:19.30 | harryvv | could do that one. |
| 05:19.31 | slePP | mine? |
| 05:19.31 | hellop | I wonder if wunderkin got the joke... |
| 05:19.52 | JerJer | slePP: who is doing the timing ? |
| 05:19.59 | slePP | carrier |
| 05:20.14 | JerJer | what's your span line(s) look like? |
| 05:20.35 | slePP | span=1,1,0,esf,b8zs |
| 05:20.35 | slePP | span=2,0,0,esf,b8zs |
| 05:20.48 | JerJer | what's span 2 ? |
| 05:20.57 | JerJer | and why no timing? |
| 05:21.27 | slePP | 2,0 is supposed to indicate 'clock master', is it not? |
| 05:21.34 | slePP | span 2 is linked off to the portmaster |
| 05:21.46 | JerJer | um isn't 1 master? |
| 05:21.55 | Sedorox | Hmmmmmmmmm |
| 05:22.02 | slePP | depends on wording, i suppose |
| 05:22.05 | JerJer | <PROTECTED> |
| 05:22.05 | JerJer | # source, then give it a value of "1". For a secondary, use "2", and so on. |
| 05:22.05 | JerJer | # To not use this as a sync source, just use "0" |
| 05:22.08 | slePP | 1 is the clock receiver |
| 05:22.10 | slePP | and it's a clock master on 2 |
| 05:22.17 | slePP | yes |
| 05:22.23 | slePP | so it's the clock receiver, mastering span 2 |
| 05:22.58 | hellop | whats the bitrate of Asterisk wav files? |
| 05:23.08 | JerJer | 16bit 8000hz mono |
| 05:23.09 | slePP | 8kHz, 16bit, usually |
| 05:23.12 | bmay | Anyone know why SIP sound stops working when I load the wcfxs module for the TDM400p card? |
| 05:23.24 | slePP | bad timing? heh |
| 05:23.55 | hellop | So that means 128kbits per second? |
| 05:24.12 | JerJer | eh? |
| 05:24.20 | slePP | ? |
| 05:24.31 | slePP | that means 16kbit/s |
| 05:24.40 | hellop | oh |
| 05:25.03 | harryvv | hhe |
| 05:25.14 | JerJer | slePP: have you mucked with the timing options? |
| 05:25.32 | slePP | oh wait, yes, 128kbit/s |
| 05:25.40 | JerJer | i do span = 1,1, span = 2,2, span = 3,3, but who knows I am on crack |
| 05:25.58 | slePP | i tried reversing it too, letting it try to clock off the span 2 |
| 05:25.59 | slePP | bad idea :> |
| 05:26.02 | JerJer | hmm |
| 05:26.05 | slePP | then nothing was too happy, lots of HDLC aborts'n'stuff |
| 05:26.49 | slePP | i'm starting to just think the cable or something is pooched |
| 05:26.57 | JerJer | 12 45 |
| 05:27.00 | JerJer | 45 12 |
| 05:27.09 | JerJer | or just use straight thru cat-5 cables |
| 05:28.02 | slePP | yeh, i tried some nice fancy ones (two) and a flat lead |
| 05:28.10 | slePP | they all behaved differently, but all errored out |
| 05:28.13 | Qwell | wow, JerJer just made me go braindead. |
| 05:28.16 | slePP | have you ever had interference problems? |
| 05:28.19 | Qwell | ethernet crossover is what, 12 36? |
| 05:28.38 | JerJer | no smoking guns |
| 05:28.40 | Qwell | or...13 26? |
| 05:29.05 | Qwell | bloody hell... |
| 05:29.09 | JerJer | lol |
| 05:29.15 | Qwell | JerJer: damn you :p |
| 05:29.17 | JerJer | T-1 is different |
| 05:29.20 | Qwell | yeah, I know |
| 05:29.38 | fearnor | sleep: your clocking is broken. |
| 05:29.42 | fearnor | (duh) |
| 05:30.14 | fearnor | make sure *one* end of circuit set up to provide timing, the other one to *accept line timing* |
| 05:30.22 | fearnor | that's all to it. |
| 05:30.30 | Qwell | 13 26...ah hah |
| 05:30.31 | slePP | you'd think so |
| 05:30.32 | fearnor | either that or you have irq issues |
| 05:30.34 | slePP | i've spent the last 3 days on it, fearnor |
| 05:30.41 | fearnor | ./zttest |
| 05:30.45 | slePP | it is definitely wired right |
| 05:30.50 | fearnor | if you aren't getting 99.98, you will have issues. |
| 05:31.27 | slePP | Best: 100.000000 -- Worst: 99.987793 |
| 05:31.30 | slePP | like that? :> |
| 05:31.32 | twisted | omfg |
| 05:31.35 | twisted | slePP is alive |
| 05:31.42 | fearnor | thats pretty decent |
| 05:31.46 | fearnor | fix your timing then |
| 05:31.50 | slePP | timing is right |
| 05:32.01 | slePP | it comes down to LBO or something else at this point |
| 05:32.11 | fearnor | no |
| 05:32.16 | fearnor | not possibol |
| 05:32.20 | fearnor | show me yo timing |
| 05:32.26 | fearnor | and where do you get hdlc errors |
| 05:32.44 | slePP | ok.. for those who missed it |
| 05:32.49 | harryvv | heard of a case where 4 admins tried to troubelshoot a new network that was working at a crawl. One of my sales guys I knew at fluke offered to hookup the analyser and thay said nooo we are trying to figure this out..3 hours later. Well, thay took up on his offer and in less then 1 min found the problem. All the wires at the terminal blocks where not twisted to within 1/2 at the termination point causing loads of cross talk. thay bought |
| 05:33.05 | slePP | PRI -> Vega -> Sangoma -> PM3 |
| 05:33.15 | slePP | harryvv: i believe it. i don't have a line tester, though |
| 05:33.16 | fearnor | whats vega |
| 05:33.21 | slePP | a SIP gateway |
| 05:33.24 | slePP | not important, actually |
| 05:33.35 | fearnor | where's asstricks? |
| 05:33.39 | slePP | it's something between the vega and sangoma that is out of whack |
| 05:33.40 | harryvv | slepp, what seems to be the problem? |
| 05:33.41 | slePP | sangoma |
| 05:33.42 | twisted | harryvv, congrats, they learned the value of TWISTED PAIR |
| 05:33.49 | slePP | harryvv: frame slips, crc errors |
| 05:34.02 | harryvv | twisted heheh I dont think it was the admins that installed it but the wiring guy. |
| 05:34.04 | twisted | harryvv, ;) |
| 05:34.05 | fearnor | vega and sangoma is t1? |
| 05:34.13 | twisted | well that makes more sense |
| 05:34.13 | fearnor | show proper links |
| 05:34.24 | fearnor | vega->(ethernet)->sangoma->PM3? |
| 05:34.50 | harryvv | slepp, yea a fluke analyser would tell you the problem. |
| 05:35.03 | twisted | slePP, what's the problem exactly? |
| 05:35.48 | slePP | fearnor: yeh |
| 05:35.50 | harryvv | guess mydns.org requres a subscription |
| 05:35.51 | harryvv | :) |
| 05:35.53 | fearnor | harry: i think they are full of shiznit, actually |
| 05:35.58 | fearnor | as far as xtalk |
| 05:36.03 | slePP | twisted: frame slips |
| 05:36.07 | slePP | the tming is off on the slave device |
| 05:36.13 | twisted | slePP, ahh.. which is the master? |
| 05:36.13 | harryvv | fearnor, what is full of shiznit? |
| 05:36.22 | slePP | Port Frames Bytes SLIPs Frames Bytes SLIPs CRC Error Bad Frames |
| 05:36.23 | slePP | ------ -------- ---------- ----- -------- ---------- ----- ---------- ---------- |
| 05:36.23 | slePP | ISDN-1 13033 97973 0 13033 100189 0 0 0 |
| 05:36.23 | slePP | ISDN-2 17723 118241 25 17322 131169 25 6 3 |
| 05:36.36 | slePP | carrier is the master clock, slaved at isdn-1, which is turned into the master clock on NT side isdn-2 |
| 05:36.54 | file[laptop] | Sleppy Boy! |
| 05:36.59 | fearnor | is pm3 set up to slave? |
| 05:37.00 | twisted | slePP, wait a minute |
| 05:37.09 | fearnor | isdn-1 goes to what? |
| 05:37.11 | harryvv | slepp, who did the wiring? |
| 05:37.11 | twisted | slePP, i thought you said the sip box was connected to an asterisk box |
| 05:37.14 | fearnor | isdn-2 goes to what? |
| 05:37.21 | fearnor | goddamn, help us help you |
| 05:37.31 | slePP | it is all one big chain |
| 05:37.32 | slePP | here we go |
| 05:37.33 | fearnor | instead of just whining 'OMG WTF TIMING SLIPS" |
| 05:37.42 | slePP | Carrier PRI into facility == a |
| 05:37.45 | harryvv | chill |
| 05:37.48 | slePP | Vega Port 1 (T1) == b |
| 05:37.52 | slePP | Vega Port 2 (T1) == c |
| 05:37.57 | slePP | Asterisk Port 1 = d |
| 05:38.01 | slePP | Asterisk Port 2 = e |
| 05:38.04 | slePP | Portmaster 3 port 1 = f |
| 05:38.08 | Qwell | harryvv: Your story earlier ended at "they bought". Was there much more to it? |
| 05:38.16 | slePP | a -> b, c -> d, e -> f |
| 05:38.21 | slePP | c->d has timing errors |
| 05:38.31 | slePP | e -> f feels these errors as well, but there are no interface errors between e and f |
| 05:38.34 | slePP | just c and d |
| 05:38.38 | twisted | okay, |
| 05:38.43 | twisted | that helps tremendously |
| 05:38.45 | fearnor | what is the set up on asterisk |
| 05:38.48 | fearnor | timing wise |
| 05:38.49 | slePP | ignore asterisk |
| 05:38.50 | fearnor | and on vega |
| 05:38.57 | slePP | a is NT/clock master |
| 05:39.01 | slePP | b is TE/clock receiver |
| 05:39.08 | harryvv | qwell, no...the fact the analyer digionosed the problem so quickly was the selling point of the analyer. It was just by chance, the fluke rep was stopping by when thay were having this problem. |
| 05:39.11 | slePP | b is sync source for NT/clock master at (c) |
| 05:39.11 | twisted | what happens if you tell c not to pull timing from b, and provide timing from c to b |
| 05:39.16 | slePP | d is TE/clock receiver |
| 05:39.19 | Qwell | ahh, heh |
| 05:39.33 | twisted | err |
| 05:39.34 | twisted | c to d |
| 05:39.40 | twisted | using the internal clock source |
| 05:39.41 | slePP | so a->b, c<-a |
| 05:39.45 | slePP | er. c<-d |
| 05:39.53 | harryvv | There top analyers back in 2001 cost like 25k |
| 05:39.55 | slePP | i lose clock sync between b and c, and thereby all the data calls suffer frame slip |
| 05:40.01 | slePP | d _must_ sync to c |
| 05:40.10 | slePP | and c must be timed to b |
| 05:40.28 | twisted | and you're sure that c is providing accurate timing? |
| 05:40.31 | slePP | but d slaving from c, gets framing errors |
| 05:40.59 | slePP | c is being synced off of a/b, and a/b is clean. if i take the asterisk box (d/e) out of the lop and go straight to f, it works ok |
| 05:41.01 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
| 05:41.10 | twisted | okay |
| 05:41.16 | slePP | i don't get any errors, but i also don't get any calls, since the PM3 doesn't speak the right protocols. but the T1 layer 1 is fine |
| 05:41.16 | harryvv | sounds like perhaps wires are crossed? |
| 05:41.25 | twisted | and you have the lbo and clock source set right for the sangoma card? |
| 05:41.34 | *** join/#asterisk clive- (~pirch@rndf-146-22-69.telkomadsl.co.za) |
| 05:41.35 | harryvv | what routed protocol you using? |
| 05:41.38 | slePP | harryvv: i think just some bad noise... i had a similar issue on the incoming T from the carrier, but i rerouted the wire and life was good |
| 05:41.45 | slePP | ESF/B8ZS/DMS100 |
| 05:41.54 | slePP | twisted: i've tried a bunch |
| 05:42.02 | slePP | a -> b == 220-330ft |
| 05:42.16 | *** join/#asterisk [1]jeremy (~jeremy@dsl-202-173-156-254.vic.westnet.com.au) |
| 05:42.16 | harryvv | only thing I am familliar with is frame relay and two others from cisco classes. |
| 05:42.24 | slePP | c -> d == 0-110, 110-220, 220-330, 7.5db.. all fail, and i think 7.5 just dies entirely (no carrier detected) |
| 05:42.38 | twisted | well yea, you're overcompensating ;P |
| 05:43.29 | slePP | yeh |
| 05:43.42 | slePP | it's pretty much just not going anywhere |
| 05:43.59 | slePP | so tonight's goal is to make a tiny cable, and if not, then make a really long one and run it in loops around the ceiling :> |
| 05:46.05 | harryvv | slepp, where are you |
| 05:46.12 | slePP | edmonton, ab, ca |
| 05:46.20 | harryvv | yea, same here on strike |
| 05:46.27 | harryvv | somone is cutting telus lines |
| 05:46.27 | slePP | i think it starte there :> |
| 05:46.29 | slePP | and moved here |
| 05:46.44 | slePP | i liek the 'we're locked out!' |
| 05:46.50 | slePP | well, sure you are, but only cuz you walked out first |
| 05:46.57 | harryvv | we also have a big truck strike because of the high fuel cost. vancouver ports are closed |
| 05:47.01 | harryvv | for almost a month now. |
| 05:47.11 | slePP | nice |
| 05:47.12 | harryvv | violence is happening because of it. |
| 05:47.30 | slePP | between whom? |
| 05:47.44 | harryvv | truckers and one company that has not paid them for a whole month. |
| 05:47.47 | Qwell | meh, unions piss me off. Strikes hurt nobody but consumers |
| 05:48.25 | slePP | unions are used for the wrong reasons now, i think |
| 05:48.33 | Qwell | indeed |
| 05:48.49 | harryvv | http://www.canada.com/vancouver/vancouversun/news/story.html?id=fcfc033b-868a-4a08-b66f-fc47a2d1b2e7 |
| 05:48.53 | *** join/#asterisk rjreb (~rjreb@greatwall.amer.net) |
| 05:49.00 | harryvv | thats the story |
| 05:49.01 | Qwell | like when grocery stores strike... |
| 05:49.06 | Qwell | you walk past, and they yell at you |
| 05:49.09 | harryvv | in laid back bc...this is unusuall |
| 05:49.12 | Qwell | "hey, fuck you buddy, I need to eat" |
| 05:49.26 | harryvv | thay can yell at me all thay want |
| 05:49.53 | harryvv | truckers are obviosly rough people |
| 05:49.53 | harryvv | :) |
| 05:50.19 | harryvv | but the bigger thay are the harder thay fall |
| 05:50.23 | *** join/#asterisk argos73 (~mike@65-85-207-101.client.dsl.net) |
| 05:50.26 | fearnor | you can't talk about unions unless you dealt with local 3 in NYC |
| 05:50.33 | harryvv | ive taken guys on bigger then me and got them on the ground ;) |
| 05:50.34 | fearnor | well, or maybe ILWU or similar unions |
| 05:50.46 | loud | fearnor, i was like, im sure i left the window on asterisk not nanog .. and then i saw you on both .. |
| 05:50.50 | fearnor | doh heh |
| 05:51.04 | fearnor | yeah well, nanog has been boring lately |
| 05:51.07 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 05:51.07 | *** mode/#asterisk [+o twisted] by ChanServ |
| 05:51.40 | loud | i miss the bgp talkings and stuffs |
| 05:51.47 | bmay | I asked before but maybe it got missed, Anyone know why SIP sound stops working when I load the wcfxs module for the TDM400p card? |
| 05:51.49 | twisted | damn |
| 05:51.52 | harryvv | never underestimate a scrawny quiet guy. friends brother has a temper like a bull if you push him hard enough. he charged this big jock and both went flying though a large window. |
| 05:51.52 | twisted | my dhcp server shit the bed |
| 05:51.58 | Qwell | twisted: gross |
| 05:52.11 | fearnor | bgp talking? not on #nanog surely ;) |
| 05:52.18 | tzafrir | bmay, this issue came p recently in the -users list |
| 05:52.30 | *** join/#asterisk fitzel (~flint@p50863291.dip0.t-ipconnect.de) |
| 05:52.34 | tzafrir | ISDN-related? |
| 05:52.40 | bmay | Can you post me a reference? I searched, but couldn't find it. |
| 05:53.09 | tzafrir | too lazy to search. but look for threads about ISDN and/or zaphfc |
| 05:53.24 | tzafrir | In the last week or two |
| 05:53.31 | bmay | Hmmm... This isn't ISDN... presumably it is still relevant? |
| 05:56.44 | harryvv | slepp |
| 05:56.47 | harryvv | http://www.flukenetworks.com/us/Telecom/default.htm |
| 05:58.01 | bmay | tzafrir, do you remember approx how many messages were in the thread? |
| 05:58.18 | tzafrir | bmay, something simpler: maybe you don't have a timing source |
| 05:58.59 | tzafrir | bmay, a simple test: try seding 'Milliwatt' to that SIP channel. Producing that sound does not involve any timing source |
| 05:59.28 | tzafrir | next, use zttest to see if your timing source is functional |
| 05:59.49 | tzafrir | s/seding/sending/ |
| 06:00.07 | bmay | ;-) I knew what you meant. Will try the milliwatt application. |
| 06:02.41 | bmay | Milliwatt works correctly; I didn't see any reference to timining sources in the documentation I found, do you have a good reference handy? |
| 06:04.53 | tzafrir | bmay, try the source of zttest |
| 06:05.25 | tzafrir | Alternatively, try reading from the file /dev/zap/pseudo |
| 06:06.13 | tzafrir | bmay, asterisk -rx 'zap show channels' |grep pesudo |
| 06:06.21 | tzafrir | pseudo, that is |
| 06:06.48 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 06:07.10 | bmay | Reading from /dev/zap/pseudo produces nothing. Just a sanity check, does it matter that I don't have any PSTN connections plugged into the card yet? |
| 06:07.38 | fitzel | anybody here that uses a pda as a voip-wlan-client and has some practical experience? |
| 06:07.52 | bmay | tzafrir, no output produced |
| 06:08.27 | bmay | tzafrir, I should clarify, cat /dev/zap/pseudo hangs, and the show channels command produced no output. |
| 06:09.13 | tzafrir | bmay, it is supposed to hang |
| 06:10.03 | tzafrir | try 'time head -c 8000 /dev/zap/pseudo >/dev/null' (or is it 8192?) |
| 06:10.12 | tzafrir | That's basically what zttest does |
| 06:11.31 | bmay | What results should I get? I get hangs for all three. |
| 06:12.30 | bmay | zttest hangs on read(3, (as per strace) |
| 06:17.07 | tzafrir | bmay, what do you use for timing source |
| 06:17.08 | tzafrir | ? |
| 06:17.20 | tzafrir | lsmod |grep ^zaptel |
| 06:18.30 | bmay | "zaptel 230340 13 wcfxs" I am suspect maybe my /etc/zaptel.conf is missing something important? |
| 06:20.11 | bmay | I assume I don't need a spannum= or timing= option? |
| 06:20.22 | bmay | or spam= option? |
| 06:20.37 | bmay | errr.. I mean... span= option? |
| 06:22.30 | *** join/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 06:22.44 | rjreb | hello |
| 06:22.47 | SoloFlyer | hey |
| 06:23.08 | SoloFlyer | ahh just finished modifying mec2 |
| 06:24.17 | SoloFlyer | i can now use agressive echo cancellation for the first x number of seconds and then once trained swap over to non aggressive :) |
| 06:24.30 | SoloFlyer | works very very well |
| 06:26.03 | rjreb | is it easy to change the musicclass for an extension? |
| 06:26.18 | SoloFlyer | for the hold music? |
| 06:26.22 | rjreb | yah |
| 06:26.30 | bmay | tzafrir, as far as I can tell my /etc/zaptel.conf has everything it needs, and I don't need span= because I am not using E1/T1 or ISDN. |
| 06:28.02 | SoloFlyer | u mean change it in the conf file or change it in realtime...? |
| 06:28.17 | SoloFlyer | its easy to change in the conf file... |
| 06:28.18 | tzafrir | bmay, a span= line is not required for TDM |
| 06:29.03 | tzafrir | rjreb, I use a custom player and give a .mp3 extensions to the wavs I play there |
| 06:29.22 | tzafrir | rjreb, ignore that |
| 06:29.31 | rjreb | ok... haha |
| 06:29.57 | rjreb | i just don't understand how to change the hold music for an incoming call |
| 06:30.18 | tzafrir | rjreb, anything wrong with SetMusicOnHold in the dialplan? |
| 06:30.27 | bmay | tzafrir, didn't think so. What is the expected result of running zttest? I assume it shouldn't hang? |
| 06:30.42 | tzafrir | bmay, no, it should not hang |
| 06:30.57 | rjreb | let me see... i only looked up musiconhold and musicclass |
| 06:31.03 | tzafrir | it should print a report about once per second |
| 06:31.46 | bmay | tzafrir, so presumably something is wrong either with the hardware or in the linux kernel? |
| 06:33.05 | rjreb | that looks like what i'm after... thanks |
| 06:33.56 | SoloFlyer | im still confused to as what the question u were asking was rjreb :P |
| 06:34.10 | SoloFlyer | but if u have a solution its all good :) |
| 06:34.23 | rjreb | just for certain extensions to change the hold music for incoming calls |
| 06:34.46 | SoloFlyer | oh i see |
| 06:34.57 | rjreb | i tried adding a line in their sip.conf but it kept playing the default |
| 06:35.21 | SoloFlyer | you could just dump them in a different context... |
| 06:35.53 | *** join/#asterisk DA-MAN (~DA-MAN@24-180-28-208.pas-mres.charterpipeline.net) |
| 06:36.06 | SoloFlyer | but setmusiconhold is probally nicer |
| 06:36.16 | hardwire | blah |
| 06:36.18 | hardwire | making stir fry fro scratch |
| 06:36.23 | hardwire | I am not a big fan of using apple juice like this recipe recommends |
| 06:36.28 | hardwire | now my meat will be crazy and 1/2 |
| 06:36.28 | brimstone | bmay: your card isn't taking interrupts, try it in a different pci slots, or in a different machine |
| 06:37.23 | drumkilla | rjreb: yeah, that option doesn't work like a lot might expect ... there is a bug on the bug tracker talking about that |
| 06:38.05 | drumkilla | what you actually are setting is the musiconhold class for that extension to *hear*, not for what its callers will hear |
| 06:38.05 | rjreb | i was looking on commpartners and got totally lost |
| 06:38.16 | bmay | Hmmm. According to /proc/interrupts "22: 23 IO-APIC-level wctdm", perhaps your right. |
| 06:38.25 | rjreb | that explains the silence |
| 06:40.46 | bmay | tzafrir, thanks for your help, I will try another computer. |
| 06:41.42 | clive- | I have a question regarding zaptel timming...is it enough to modprobe wcfxo and then ztcfg , I am using a x100p for timming only |
| 06:42.43 | harryvv | yes |
| 06:42.56 | harryvv | thats the right order |
| 06:45.34 | SoloFlyer | anyu cant modprobe wcfxo without modprobing zaptel |
| 06:45.40 | clive- | harry thanks |
| 06:45.55 | clive- | solo, I havent modprobed zaptel... |
| 06:46.11 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:48.30 | tzafrir | SoloFlyer, what do you mean? modprobe wcfxo loads zaptel as well. Works here |
| 06:48.35 | tzafrir | (1.0.9.1) |
| 06:48.40 | *** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) |
| 06:49.16 | SoloFlyer | sorry i was going on the assumption that it wouldnt automatically resolve its dependencies |
| 06:49.20 | SoloFlyer | but it does |
| 06:49.30 | SoloFlyer | :) |
| 06:50.07 | CyberSword | hi , i have a problem i cant do that external sip users log into my PBX only from my network can do it |
| 06:51.50 | shido | er? |
| 06:51.52 | slePP | anyone feel like sending me about 20 callsat once? |
| 06:51.58 | shido | nat issues? |
| 06:53.16 | SoloFlyer | do u see the reqests comming in? |
| 06:53.55 | SoloFlyer | ie does asterisk see them... if not do u see the request come in if u use tcpdump? |
| 06:54.17 | *** join/#asterisk Kernel_core (Raph@217.218.94.147) |
| 06:54.17 | SoloFlyer | in other words more info please :) |
| 06:54.25 | SoloFlyer | hi |
| 06:59.05 | Kernel_core | how do I define if user called , 1234 then send this to Zap channel ?! |
| 06:59.58 | limbique | morning |
| 07:00.11 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 07:00.59 | *** join/#asterisk Aze` (~aze@85.18.136.114) |
| 07:01.26 | SoloFlyer | u mean dial? |
| 07:02.53 | SoloFlyer | Dial(Zap/1/${EXTEN}) |
| 07:03.57 | limbique | can anyone help me with installing a tdm card in asterisk?? |
| 07:04.14 | bmay | tzafrir, if you are still around, it works in another computer. Thanks again. |
| 07:04.16 | SoloFlyer | what part!? |
| 07:04.30 | SoloFlyer | physical install |
| 07:04.37 | *** join/#asterisk Inv_arp (junya@adsl-156-143-182.mia.bellsouth.net) |
| 07:04.40 | SoloFlyer | zaptel driver install |
| 07:04.49 | SoloFlyer | getting asterisk to connect to zaptel? |
| 07:05.25 | limbique | how to check? |
| 07:05.41 | SoloFlyer | what have u done so far? |
| 07:05.44 | *** join/#asterisk bjohnson_ (~bjohnson@i216-58-62-102.igs.net) |
| 07:05.51 | limbique | machine has a clean asterisk install |
| 07:05.58 | limbique | build in the card |
| 07:06.14 | SoloFlyer | have u compiled the zaptel drivers? |
| 07:06.27 | limbique | how can i check that? |
| 07:06.39 | SoloFlyer | modprobe zaptel |
| 07:06.45 | *** join/#asterisk lefred (~fred@d5152D086.access.telenet.be) |
| 07:07.02 | SoloFlyer | if it says "WTF is zaptel" then its not done |
| 07:07.04 | SoloFlyer | :) |
| 07:07.36 | limbique | no output |
| 07:07.46 | SoloFlyer | it is installed then |
| 07:07.48 | tzafrir | lsmod |grep ^zaptel |
| 07:07.50 | limbique | cool |
| 07:08.15 | limbique | zaptel 206724 1 ztdummy |
| 07:08.15 | SoloFlyer | yeah actualyl do lsmod |grep zaptel |
| 07:08.29 | limbique | hi tzafrir |
| 07:08.51 | limbique | i got a other machine with a latest version |
| 07:08.55 | SoloFlyer | do modprobe wcfxs for your fxo modules |
| 07:08.57 | tzafrir | limbique, you don't need ztdummy if you have that card. But it seems Asterisk is using it right now, so you should stop asterisk |
| 07:09.06 | SoloFlyer | and modprobe wcfxo for your fxs modules |
| 07:09.13 | tzafrir | rmmod ztdummy |
| 07:09.22 | limbique | ok, i know that ztdummy is for timing if you don't have a card |
| 07:09.36 | SoloFlyer | actually |
| 07:09.39 | limbique | rmmod done |
| 07:10.09 | limbique | asterisk is not running right now |
| 07:10.42 | SoloFlyer | do modprobe wcfxs |
| 07:10.48 | SoloFlyer | and modprobe wcfxo |
| 07:10.59 | limbique | zaptel 206724 0 |
| 07:11.24 | limbique | i have only fxs modules |
| 07:11.34 | SoloFlyer | ok then only do modprobe fxo |
| 07:11.34 | limbique | (extension module?) |
| 07:12.14 | SoloFlyer | fxo drivers is for fxs modules (yes its confusing :P ) |
| 07:12.19 | limbique | no output, so it seem installed |
| 07:12.22 | limbique | checking |
| 07:12.37 | limbique | zaptel 206724 1 wcfxs |
| 07:12.56 | SoloFlyer | so tail /var/log/messages |
| 07:13.05 | SoloFlyer | see what it says |
| 07:13.08 | limbique | modprobe fxo? |
| 07:13.16 | clive- | so for a x100p I need to modprobe wcfxs ? |
| 07:13.19 | tzafrir | cat /proc/zaptel/1 |
| 07:13.31 | tzafrir | an X100P uses wcfxo |
| 07:13.38 | SoloFlyer | yeah |
| 07:14.03 | clive- | solo your last statement was confusing:) |
| 07:14.35 | limbique | No such file or directory |
| 07:14.38 | tzafrir | clive-, well, it should find nothing and fail to load |
| 07:14.42 | limbique | with that cat command |
| 07:15.13 | limbique | should i install both fxs and fxs modules? |
| 07:15.25 | opus_ | yo |
| 07:18.30 | *** join/#asterisk mrproper_ (~psynode@CPE-60-225-51-188.nsw.bigpond.net.au) |
| 07:18.58 | mrproper_ | im trying to compile asterisk and when i do either a make clean or make install it keeps looping on: include/asterisk/version.h.tmp |
| 07:19.06 | Kernel_core | Jul 27 09:18:04 NOTICE[3307]: app_dial.c:764 dial_exec: Unable to create channel of type 'ZAP' |
| 07:19.13 | Kernel_core | Jul 27 09:18:04 NOTICE[3307]: app_dial.c:764 dial_exec: Unable to create channel of type 'ZAP' <----- what does it mean ?! |
| 07:19.57 | SoloFlyer | u have zaptel channels defined in zapata.conf? |
| 07:20.17 | Kernel_core | SoloFlyer: yes I defined ! |
| 07:20.30 | SoloFlyer | and all drivers etc are running? |
| 07:20.51 | Kernel_core | defaultzone=us |
| 07:20.52 | Kernel_core | span=1,0,0,ccs,hdb3 |
| 07:20.52 | Kernel_core | bchan=1-15 |
| 07:20.52 | Kernel_core | bchan=17-31 |
| 07:20.52 | Kernel_core | dchan=16,32 |
| 07:20.52 | Kernel_core | loadzone=us |
| 07:20.53 | Kernel_core | context=acc2 |
| 07:21.08 | Kernel_core | yes , i checked with zttool , everything is OK! |
| 07:21.14 | tzafrir | Kernel_core, 32 is not a dchan |
| 07:21.22 | tzafrir | Leave only 16 there |
| 07:21.30 | Kernel_core | tzafrir: one dchannel is enough ?! |
| 07:21.48 | tzafrir | yes. |
| 07:23.35 | tzafrir | limbique, both wcfxs and wcfxo, you mean? |
| 07:23.43 | mrproper_ | im trying to compile asterisk and when i do either a make clean or make install it keeps looping on: include/asterisk/version.h.tmp |
| 07:24.39 | tzafrir | mrproper_, some patients? |
| 07:25.10 | hellop | the patients are also the doctors |
| 07:26.09 | tzafrir | hellop, well, doctors are the ones who only practice all the time |
| 07:26.42 | tzafrir | mrproper_, what version of Asterisk? |
| 07:26.44 | mrproper_ | tzafrir: its been sitting there for 5 hours looping the exact same information over and over |
| 07:26.59 | mrproper_ | tzafrir: -D 20050713 |
| 07:27.08 | Kernel_core | tzafrir: I removed channel 32 of dchannel but still get the same error " Unable to create channel of type 'zap' " |
| 07:27.09 | tzafrir | I meant: about repeating your question here |
| 07:27.35 | tzafrir | Kernel_core, are the bchannels defined in zapata.conf? |
| 07:27.52 | tzafrir | Are they shown in zap show channels ? |
| 07:28.20 | limbique | brb, rebooting |
| 07:28.43 | tzafrir | argos73, there are existing cards |
| 07:28.49 | mrproper_ | argos73: well your in luck, apparently they are making a 4 port bri card |
| 07:28.54 | Kernel_core | tzafrir: you mean this -------> http://pastebin.com/322110 |
| 07:29.32 | argos73 | yea, i know... just finding one is tricky. have a BRI line here that really doesn't need to be used for it's current purpose any more, but I hate to have it turned off... |
| 07:29.43 | tzafrir | Kernel_core, that's /etc/zaptel.conf . I asked about /etc/asterisk/zapata.conf |
| 07:29.52 | argos73 | mrproper_, : hmm - really? |
| 07:29.57 | *** join/#asterisk elzhov (~etv@nf034.jinr.ru) |
| 07:30.18 | mrproper_ | argos73: yep, just not sure if theyre making a 1 port version or if it will be like the wild cards etc |
| 07:31.11 | Kernel_core | tzafrir: you mean this ---- > http://pastebin.com/322112 |
| 07:31.12 | argos73 | hell - i'd buy a 4-port if i had to... just having a hell of a time finding a linux-friendly card that I can purchase in the US through "approved purchasing channels" at work. |
| 07:31.26 | argos73 | (Digium = approved) |
| 07:31.29 | *** join/#asterisk nounoursfr (~Direct@stardust.noc.frontier.fr) |
| 07:31.31 | mrproper_ | argos73: the fritz card no good? |
| 07:32.04 | argos73 | would work - know a good supplier? |
| 07:32.14 | mrproper_ | argos73: your in the states? |
| 07:32.16 | argos73 | yup |
| 07:32.27 | clive- | mrproper...do you know much about kernel compilling ? |
| 07:32.31 | mrproper_ | argos73: all my suppliers are in australia (im an aussie) |
| 07:32.42 | argos73 | hehe - that's the problem! :) |
| 07:32.55 | mrproper_ | argos73: why not just order it from over seas? |
| 07:32.55 | SoloFlyer | argos hang 5 |
| 07:32.59 | nounoursfr | hello all |
| 07:33.08 | clive- | argos, I am looking for a eicon 4BRI card... |
| 07:33.08 | mrproper_ | argos73: i didnt think yanks used bri connections? |
| 07:33.44 | SoloFlyer | yeah but they are slightly different from ours mrproper_ |
| 07:33.50 | argos73 | mrproper_, : they're pretty rare up here... the telco was really confused wen I ordered it... |
| 07:34.09 | mrproper_ | argos73: well i think your going to have a hard time sourcing a bri card in the states |
| 07:34.16 | SoloFlyer | argos73 my telco was pretty confused when i ordered mine lol |
| 07:34.45 | *** part/#asterisk pif (ldm@zenon.apartia.fr) |
| 07:34.52 | tzafrir | mrproper_, there is always http://www.junghanns.net/asterisk/page17.html |
| 07:34.55 | argos73 | "you want what?" ?:) |
| 07:35.23 | drray | a bri is just an ISDN line right? |
| 07:35.24 | argos73 | really threw the telco for a loop when I started ordering Centrex over ISDN |
| 07:36.10 | mrproper_ | drray: bri is just 1 or 2 64k b channels an a single d channel |
| 07:36.24 | argos73 | drray: yea - only two channels, vs. 23 for an ISDN-PRI |
| 07:36.38 | mrproper_ | argos73: hehe we have 30 here =P |
| 07:36.43 | SoloFlyer | we have out of band data channel they have inband channeling in the states |
| 07:36.51 | mrproper_ | argos73: still cant understand why the yanks still use T1 |
| 07:37.15 | *** join/#asterisk dacleric (~dacleric@p54829B97.dip0.t-ipconnect.de) |
| 07:37.27 | argos73 | hehe - can't understand why we still use Miles vs Kilometers, either! |
| 07:37.30 | SoloFlyer | cant find that link sorry argos |
| 07:37.33 | argos73 | no prob |
| 07:37.46 | mrproper_ | argos73: dont get me started on the 'imperial' measuring system |
| 07:37.52 | argos73 | :) |
| 07:38.02 | SoloFlyer | and whats with the left hand drive!? |
| 07:38.29 | mrproper_ | SoloFlyer: hehe |
| 07:38.31 | SoloFlyer | other than states the majority of places are right |
| 07:38.36 | argos73 | but we just love the fact that there's 63360 inches in a mile! |
| 07:39.02 | mrproper_ | lol |
| 07:39.08 | opus_ | hey |
| 07:39.10 | opus_ | whats up |
| 07:39.29 | SoloFlyer | yeah so u can almost fill a unsigned int unlike us we barely use it at all |
| 07:39.32 | SoloFlyer | :P |
| 07:39.34 | drray | next you'll be pimping metric time |
| 07:39.36 | mrproper_ | opus_: the imperial measuring system....apparently |
| 07:39.56 | SoloFlyer | 2400 hours :) |
| 07:40.05 | SoloFlyer | whats with the am pm crap! |
| 07:40.14 | argos73 | SoloFlyer, hmm - you might have stumbled upon a conspiracy... "MUST FILL UNSIGNED INTS!" |
| 07:40.24 | SoloFlyer | lol |
| 07:40.25 | mrproper_ | drray: hey dont be mocking metric time, it will take over the world eventually.... |
| 07:40.57 | opus_ | metric time. whoah |
| 07:41.06 | opus_ | does that mean i can go to work later |
| 07:41.17 | SoloFlyer | no |
| 07:41.27 | argos73 | I just love how a lot of places (my wife's work, for example) calls "decimal time" "military time" |
| 07:41.31 | drray | by then the robots will be in charge and you won't work at all |
| 07:41.39 | limbique | tzafrir: genzaptelconf doesn't work :( |
| 07:41.54 | drray | military is not metric time |
| 07:41.57 | limbique | i get : no such command 'zap' |
| 07:42.01 | argos73 | they seem to think the US military refers to fractions of an hour... |
| 07:42.07 | drray | heh |
| 07:42.10 | tzafrir | #include "zapata-channels.conf" |
| 07:42.12 | drray | zulu and lima time |
| 07:42.19 | tzafrir | Should be the last line of zapata.conf |
| 07:42.19 | argos73 | not the simple "just add 12 hours if it's PM" |
| 07:42.25 | SoloFlyer | military time is good! |
| 07:43.08 | argos73 | lima bean time... :) |
| 07:43.09 | limbique | the last line is ;channel 1 |
| 07:43.14 | limbique | the last line is ;channel => 1 |
| 07:43.42 | tzafrir | limbique, that script does not touch zapata.conf , so it won't edit-away any of your modifications |
| 07:43.47 | SoloFlyer | forget metrictime /military time/old skool time ... lets just do miliseconds since 1900 :) |
| 07:43.54 | opus_ | bush bots |
| 07:44.03 | tzafrir | It writes to /etc/asterisk-zapata-channels instead |
| 07:44.09 | tzafrir | It writes to /etc/asterisk-zapata-channels.conf instead |
| 07:44.20 | mrproper_ | lol |
| 07:44.23 | limbique | tzafrir, the zapataconf is a default file |
| 07:44.29 | opus_ | hmm |
| 07:44.36 | argos73 | hehe - try MUMPS time... # of days and seconds since Jan 1, 1870, IIRC |
| 07:44.51 | limbique | i see a zapata-c~onf.bak |
| 07:44.55 | argos73 | right now, mumps time is 60108,13480 |
| 07:44.57 | tzafrir | limbique, and I wouldn't want to run over modifications you've added there, hence the use of an #include |
| 07:45.23 | limbique | but there is no zapata-channels.conf file found |
| 07:46.07 | tzafrir | was there any error from genzaptelconf? e.g: asterisk was running? |
| 07:46.27 | limbique | yes asterisk is running |
| 07:46.33 | limbique | the error message |
| 07:47.17 | limbique | Shutting asterisk pbx |
| 07:47.26 | limbique | starting asterisk PBX |
| 07:47.36 | opus_ | tzafrir - dude, we should use speex wideband |
| 07:47.44 | limbique | no such command 'zap' (type 'help' for help) |
| 07:47.57 | limbique | that was the output of genzapataconf |
| 07:48.23 | limbique | brb |
| 07:48.53 | tzafrir | opus_, does it indeed "conflict" with meetme? |
| 07:49.00 | opus_ | of course |
| 07:49.06 | opus_ | but, we'd have to rewrite that as well |
| 07:49.44 | tzafrir | limbique, is asterisk running? |
| 07:50.08 | tzafrir | limbique, is there anything in /etc/asterisk/zapata-channels.conf ? |
| 07:50.27 | lefred | mmm, when I try gnophone with asterisk (1.0.9) I got this : *CLI> Jul 27 09:49:13 WARNING[3104]: chan_iax2.c:546 iax_error_output: Information element length exceeds message size |
| 07:50.27 | lefred | Jul 27 09:49:13 WARNING[3104]: chan_iax2.c:5336 socket_read: Undecodable frame received from '10.0.0.196' |
| 07:50.27 | lefred | any idea how to fix this ? |
| 07:50.43 | limbique | asterisk is running |
| 07:50.58 | limbique | there is no file called zapata-channels.conf |
| 07:51.41 | hellop | I am suffering from amotivational syndrome. |
| 07:51.59 | limbique | don't spread it then |
| 07:52.21 | limbique | :) |
| 07:52.28 | opus_ | time to smoke some wacky |
| 07:53.21 | hellop | I think I'll take a little vacation next weekend. |
| 07:53.43 | argos73 | off to bed... in ten minutes, my pager will go off, telling me this thunderstorm has knocked out power at work, which will wake my wife up in a bad mood, and start another wonderful day! |
| 07:53.52 | argos73 | later |
| 07:53.53 | limbique | opus_: jup |
| 07:54.12 | limbique | we have a lot :) (NL) |
| 07:54.19 | opus_ | whesh |
| 07:54.30 | hellop | You ever go on a nice long vacation and when you come back you just feel great for like 2 weeks? |
| 07:54.36 | opus_ | one of my coworkers picked up my pack of smokes and was like, thats ciggaweed fool |
| 07:55.06 | tzafrir | limbique, what exactly did you run? |
| 07:55.12 | limbique | um, |
| 07:55.34 | tzafrir | Do you run that script as root, or any other user with write permissions to the relevant files? |
| 07:55.41 | limbique | . /bin/bash genzaptelconf |
| 07:55.51 | limbique | jup, i'm logged in as root |
| 07:56.01 | limbique | nope, single user here |
| 07:56.07 | limbique | my private system :P |
| 07:56.39 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:01.46 | bkw_ | GOOD NIGHT INKERNET! |
| 08:06.29 | limbique | :) |
| 08:06.37 | mrproper_ | im trying to compile asterisk but i have a section of code that keeps looping: build_tools/make_version_h > include/asterisk/version.h.tmp |
| 08:07.44 | Piranha- | hmm anyone hear of Sjphone not recieving any audio inbound (outbound works), but x-lite works completely? |
| 08:16.41 | *** join/#asterisk pa (~Paolo@pa.user) |
| 08:29.24 | *** join/#asterisk _deg_ (~deg@201.22.29.38.adsl.gvt.net.br) |
| 08:31.26 | pa | is Asterisk able to do text-to-speech? |
| 08:31.54 | tzafrir | pa, not asterisk directly. But festival does... |
| 08:32.44 | tzafrir | apt-get install festival . Worked for me rather well here. Then try: |
| 08:32.48 | tzafrir | Festival(hello world) |
| 08:33.52 | pa | but unfortunately festival hasnt italian language :-( |
| 08:34.04 | *** part/#asterisk SoloFlyer (~jkl@61.29.7.18) |
| 08:35.19 | *** join/#asterisk grimse (~grimse@p5481E197.dip.t-dialin.net) |
| 08:42.49 | *** join/#asterisk dacleric (~dacleric@p5482A04E.dip0.t-ipconnect.de) |
| 08:43.19 | *** join/#asterisk CyberSword (CyberSword@nl135.globalpc.net) |
| 08:45.28 | *** join/#asterisk paski_31 (~paski_31@stardust.noc.frontier.fr) |
| 08:46.33 | *** part/#asterisk paski_31 (~paski_31@stardust.noc.frontier.fr) |
| 08:46.55 | *** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM] |
| 08:48.03 | *** join/#asterisk paski_fr (~paski_fr@stardust.noc.frontier.fr) |
| 08:48.11 | paski_fr | hi |
| 08:48.40 | paski_fr | i need help with eyebeam |
| 08:48.41 | nounoursfr | hi paski_fr |
| 08:48.48 | Piranha- | what would the command be called to call a voip number, then set it up to accept dial-outs (like for cellphones to call longdistance) |
| 08:49.54 | paski_fr | Someone know how to use video under NAT with ASTERISK and EYEBEAM? |
| 08:53.11 | *** join/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
| 08:59.41 | *** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM] |
| 09:02.20 | *** join/#asterisk tobiasWolf (~konversat@195.162.255.10) |
| 09:02.37 | mrproper_ | im trying to compile asterisk but i have a section of code that keeps looping: build_tools/make_version_h > include/asterisk/version.h.tmp |
| 09:03.34 | Piranha- | 1.0.9? |
| 09:03.51 | Piranha- | what OS |
| 09:05.43 | *** join/#asterisk af_ (~af@ip-161-107.sn2.eutelia.it) |
| 09:06.42 | *** part/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
| 09:06.54 | *** join/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
| 09:07.02 | X-Rob | Piranha- DISA possibly? |
| 09:07.25 | Piranha- | I shall check |
| 09:08.28 | Piranha- | you know off hand if voip in -> <code> -> voip out (on the same provider) would work |
| 09:12.04 | *** part/#asterisk nounoursfr (~Direct@stardust.noc.frontier.fr) |
| 09:12.19 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 09:12.49 | *** join/#asterisk MustDie (~Alik@h-67-101-134-52.nycmny83.dynamic.covad.net) |
| 09:13.17 | *** join/#asterisk nounoursfr (~nounours@stardust.noc.frontier.fr) |
| 09:13.32 | elzhov | Hello |
| 09:14.39 | elzhov | I've a guestion. Why to assignments symbols are used in Asterisk conf files, '=' and '=>'. According to docs there are no difference between them? |
| 09:14.51 | elzhov | Why to |
| 09:14.55 | elzhov | == why 2 |
| 09:19.59 | mrproper_ | im trying to compile asterisk but i have a section of code that keeps looping: build_tools/make_version_h > include/asterisk/version.h.tmp see http://pastebin.com/322142 for debug |
| 09:20.12 | *** join/#asterisk meppl (~mephisto@84.245.164.195) |
| 09:22.50 | *** join/#asterisk jiro5281 (~anton281@203.177.242.192) |
| 09:25.10 | *** part/#asterisk lefred (~fred@d5152D086.access.telenet.be) |
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| 09:37.02 | paski_fr | <PROTECTED> |
| 09:38.43 | Delvar | i have only ever done that using our outbound proxy |
| 09:39.01 | mrproper_ | im trying to compile asterisk but i have a section of code that keeps looping: build_tools/make_version_h > include/asterisk/version.h.tmp |
| 09:39.08 | Delvar | you should eb able to do it with port forwarding, dut i have no idea what ports it uses for video |
| 09:41.40 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 09:42.00 | ManxPower | You should never need to port forward unless Asterisk is behind NAT |
| 09:42.03 | *** join/#asterisk nounoursfr (~nounours@stardust.noc.frontier.fr) |
| 09:42.58 | *** join/#asterisk Ueberspasti (~kdajfjq@83-64-69-134.static.adsl-line.inode.at) |
| 09:43.25 | *** join/#asterisk AustroPretorian (~AustroPre@83-64-69-134.static.adsl-line.inode.at) |
| 09:43.49 | fenlander | video just uses another set of RTP ports - you should be able to use stun. (If audio works but not video I would guess that you are using symmetric rtp for audio which lets asterisk do the nat trick) |
| 09:44.40 | AustroPretorian | Hi, I want to do two outbound calls via AGI/Manager API - this already works fine - how can I connect those two calls together, so they can talk? |
| 09:44.48 | nounoursfr | there is of french on the channel? |
| 09:44.55 | Ueberspasti | Hi... Can someone help me setting up my asterisk for using different languages |
| 09:46.44 | nounoursfr | the change languages for asterisk in /var/spool/asterisk/sound |
| 09:47.26 | tzafrir | nounoursfr, /var/lib/asterisk/sounds , that is |
| 09:47.43 | nounoursfr | yes sorry /var/spool/asterisk/sounds |
| 09:47.45 | *** join/#asterisk [Jedi] (~fdsafasdf@213.162.200.226) |
| 09:47.49 | [Jedi] | Hello |
| 09:47.58 | nounoursfr | hi |
| 09:47.58 | tzafrir | /var/*lib*/ |
| 09:48.02 | [Jedi] | I'm having a really big problem with asterisk and AGI |
| 09:48.19 | [Jedi] | I'm developing a calling card application |
| 09:48.32 | nounoursfr | oh I am interesting |
| 09:48.53 | nounoursfr | i listen to you :) |
| 09:48.53 | [Jedi] | if the called party hangs-up the phone, I can read the ANSWEREDTIME variable with no problem |
| 09:49.24 | [Jedi] | but if the calling party hangs-up, I get a "Channel was hang up." error |
| 09:50.03 | ManxPower | [Jedi]: try DeadAGI |
| 09:50.15 | nounoursfr | your script is a realtime ? |
| 09:50.32 | [Jedi] | ManxPower: DeadAGI works with FastAGI? |
| 09:50.40 | [Jedi] | FastDeadAGI? hehe |
| 09:50.50 | *** part/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it) |
| 09:51.01 | AustroPretorian | Anybody go an idea on this: I want to do two outbound calls via AGI/Manager API - this already works fine - how can I connect those two calls together, so they can talk? |
| 09:51.06 | ManxPower | [Jedi]: No idea, but I seem to recall that getting channel variables only works in DeadAGI. I could be wrong |
| 09:53.42 | *** join/#asterisk Fabe (~spamhere@217.91.11.247) |
| 09:54.20 | [Jedi] | ManxPower: i really love you |
| 09:54.29 | [Jedi] | lol |
| 09:54.30 | [Jedi] | it works |
| 09:54.43 | [Jedi] | thankyou Very much. I was getting very very afraid |
| 09:55.09 | ManxPower | [Jedi]: 8-) |
| 09:55.40 | [Jedi] | I was starting to dig into java fastagi code to see if the problem was there |
| 09:56.33 | nounoursfr | do you have use queue in realtime ? |
| 09:57.29 | *** join/#asterisk ellvis (~evills@adsl-data-148.84-47-83.telecom.sk) |
| 09:57.53 | ellvis | hi people |
| 09:59.06 | ellvis | i am having troubles to call to the DDI line which is connected to asterisk. calls from asterisk to that line works just fine, calling from outside to that line is resultin in the operator sound that the line doesn't exist. anyone have any experience with that? |
| 10:01.41 | AustroPretorian | Can anybody help me? |
| 10:03.00 | clive- | pretorian? |
| 10:03.37 | Ueberspasti | where can i set language for prompts to "de" for sip an from pstn? |
| 10:04.09 | AustroPretorian | yes |
| 10:04.18 | AustroPretorian | my problem is this: I want to do two outbound calls via AGI/Manager API - this already works fine - how can I connect those two calls together, so they can talk? |
| 10:06.09 | clive- | check out the callback script, which is inthe wiki I think...should work lekker |
| 10:06.36 | AustroPretorian | do you have the url? |
| 10:06.39 | AustroPretorian | thx |
| 10:06.50 | clive- | howz ozz |
| 10:09.55 | *** join/#asterisk tango1 (~murgs@dsl-084-059-148-211.arcor-ip.net) |
| 10:14.13 | AustroPretorian | nope, that did not help - My problem is to connect to lines together, not the call itself |
| 10:19.14 | *** join/#asterisk Tili (~Tili@202-133-67-20-dialup.sat.net.pk) |
| 10:21.25 | RaYmAn-Bx | AustroPretorian: you prolly want something similar to click-to-dial then...There are some scripts mentioned in the wiki that does that (TACI is one of them)..Perhaps that'll give some hints as to how to do it? |
| 10:25.26 | AustroPretorian | thx, I'll take a look |
| 10:25.51 | mrproper_ | can anyone help me debug a compiling issue (compiling asterisk keeps looping |
| 10:29.52 | [Jedi] | nounoursfr: what do you mean by "use queue in realtime"? |
| 10:36.38 | AustroPretorian | I just took a look at it, but it seem that this just works for a channel and an extention - But what I need is to connect two channels |
| 10:37.27 | *** join/#asterisk otmar (~lendl@arachne.bofh.priv.at) |
| 10:37.38 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:37.45 | puzzled | hi |
| 10:39.15 | RaYmAn-Bx | AustroPretorian: as far as I can gather it works by first dialing one person and then going for the extension of another and hence dial "from" the first person to the second..if that makes sense..I think that might be the only way to archive what you want |
| 10:39.36 | RaYmAn-Bx | well..some kind of transfer or similar might work, but I don't know |
| 10:41.37 | otmar | anybody seen joshnet? |
| 10:42.43 | *** join/#asterisk Tili_ (~Tili@202-133-65-216-dialup.sat.net.pk) |
| 10:46.02 | nounoursfr | [Jedi], I wanted to know if somebody had already to test the queue in realtime and to have its opinion |
| 10:46.31 | *** join/#asterisk zoo (nobody@ip-36-16.travedsl.de) |
| 10:46.37 | zoo | good morning |
| 10:46.54 | zoo | anyone using asterisk-cvs? |
| 10:47.57 | nounoursfr | yes me using asterisk-cvs |
| 10:48.17 | AustroPretorian | My point is I do not have an extention to use. I want to connect to ZAP/ Channels |
| 10:49.41 | zoo | nounoursfr: please try to do this: grep -e 'qos=\"%s\"' /channels/chan_sip.c |
| 10:49.56 | zoo | nounoursfr: please try to do this: grep -e 'qoq=\"%s\"' /channels/chan_sip.c |
| 10:49.58 | zoo | damn |
| 10:50.10 | zoo | it must be qop=\"%s\" |
| 10:50.30 | mrproper_ | im trying to compile asterisk but i have a section of code that keeps looping: build_tools/make_version_h > include/asterisk/version.h.tmp |
| 10:50.45 | jiro5281 | any gnudialer user there? |
| 10:51.01 | otmar | I've just checked out a fresh copy and running make now |
| 10:52.07 | otmar | make ran through. |
| 10:54.00 | zoo | anyone of the developpers with cvs write access here? |
| 10:54.06 | zoo | i found a bug in 1.0.9 |
| 10:54.20 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 10:54.29 | mrproper_ | what bug? |
| 10:55.30 | zoo | in asterisk/channels/chan_sip.c, line 8046: qop=\"%s\" is wrong ans should be qop=%s |
| 10:56.17 | zoo | according to RFC3261 the auth parameter has no double quotes around! |
| 10:56.55 | puzzled | zoo: have you filed a bug on bugs.digium.com? if not, please do so the developers know about it and it can be fixed |
| 10:57.04 | mrproper_ | have you checked that file in the cvs version? |
| 10:57.16 | zoo | mrproper_: yes, i just checked it out from cvs |
| 10:57.58 | [Jedi] | my cvs hasn't quotes |
| 10:58.53 | mrproper_ | [Jedi]: mine does at -D20040713 |
| 11:00.07 | zoo | well, whatever, i just wanted to tell you that there are no double quotes allowed. |
| 11:00.58 | zoo | my chan_sip.c is Revision: 1.796 |
| 11:02.40 | [Jedi] | ASTERISK_FILE_VERSION(__FILE__, "$Revision: 1.778 $") |
| 11:02.47 | [Jedi] | ???? |
| 11:02.50 | mrproper_ | can anyone help me with a compile problem, compiling asterisk |
| 11:04.06 | mrproper_ | im compiling asterisk with: make clean....that runs fine, run make install...all seems to go well till it gets to a part and keeps looping this section of code (over and over, had it up to 5 hours before giving up) http://pastebin.com/322177 |
| 11:04.39 | *** join/#asterisk Patrick04 (Patrick04@cpc1-bahd1-3-0-cust51.renf.cable.ntl.com) |
| 11:06.59 | mrproper_ | anyone? |
| 11:08.38 | *** join/#asterisk riksta (~rick@84.93.243.170.broadband.plus.dyn.plus.net) |
| 11:09.05 | *** join/#asterisk wulfy814 (~lorentz@c-67-165-37-20.hsd1.pa.comcast.net) |
| 11:09.36 | wulfy814 | I'm getting this when I park a call: WARNING[17132] file.c: Failed to write frame - |
| 11:09.47 | wulfy814 | the parking works and the parked caller here's the on hold music |
| 11:10.02 | tzafrir | mrproper_, tried a more recent version? |
| 11:10.21 | mrproper_ | tzafrir: running the latest cvs |
| 11:10.31 | tzafrir | 200*4*0713 ? |
| 11:10.47 | mrproper_ | tzafrir: was using that, just recently updated it and tried again....same result |
| 11:11.18 | mrproper_ | tzafrir: whats even more strange that once this starts if i then kill it off and try and run make clean again, it immediately goes straight to this part of the code and loops |
| 11:11.53 | mrproper_ | tzafrir: i should probably also mention im using FC3 with smp |
| 11:12.32 | wulfy814 | running the lastest CVS |
| 11:12.47 | mrproper_ | yes im running the latest CVS |
| 11:13.12 | mrproper_ | tzafrir: have a look at the output: http://pastebin.com/322177 |
| 11:19.41 | *** join/#asterisk darkskiez (~darkskiez@194.247.78.146) |
| 11:21.01 | mrproper_ | great |
| 11:21.20 | mrproper_ | i love when absolutely no one even lifts a finger to give you a hadn |
| 11:21.23 | mrproper_ | hand even |
| 11:22.44 | ManxPower | mrproper_: I don't see any errors in your pastebin, but maybe i've not had enough coffee. |
| 11:23.07 | mrproper_ | ManxPower: exactly why im pulling my hair out, i have nothing to even go down |
| 11:23.45 | mrproper_ | ManxPower: whats even more ridiculous is once this starts happening, i kill off the compile then run a make clean and it starts straight back into that code loop |
| 11:23.47 | ManxPower | mrproper_: What is your specific issue? |
| 11:23.59 | darkskiez | I have a TDM card with 2 phones and a fax attached, two of the modules attached to the phone seemed to cease detecting the phone was picked up after a month of use, they'd keep ringing and click out the handset. stopping asterisk and unloading/reloading the module was needed to make them work again. Has this been seen before? |
| 11:24.00 | mrproper_ | ManxPower: it wont compile it just sits there looping |
| 11:24.12 | ManxPower | mrproper_: *shrug* You've seen all the changes to Asterisk in the past couple of days, right? |
| 11:24.39 | ManxPower | darkskiez: Yes. That issue is why I now avoid TDM400P cards. |
| 11:25.02 | mrproper_ | ManxPower: no i havent |
| 11:25.08 | mrproper_ | ManxPower: whats been happenin |
| 11:25.26 | ManxPower | mrproper_: You use CVS-HEAD and yet are not on the asterisk-cvs mailing list???????????/ |
| 11:25.38 | mrproper_ | ManxPower: nope hehe |
| 11:25.42 | ManxPower | CVS-HEAD is under heavy developement it WILL be broken occasionally |
| 11:25.54 | darkskiez | ManxPower: thats depressing, very depressing. |
| 11:26.13 | darkskiez | ManxPower: is it a hardware bug/software bug? is there a workaround for either? |
| 11:26.32 | zoo | puzzled: bug report issued |
| 11:26.34 | mrproper_ | ManxPower: yeah i know that but i built another box using an older version 20050713 and that was fine, i tried the same version on this box (only difference is this box is an smp box) and its broked |
| 11:26.53 | [Jedi] | well it's mostly frozen, right? |
| 11:26.53 | *** join/#asterisk Zeeek (~icechat5@Zeeek.active.supporter.pdpc) |
| 11:26.56 | ManxPower | darkskiez: since it takes a long time to happen we've not solved the problem. |
| 11:27.05 | ManxPower | We solve the problem by using a T-1 card and a channel bank |
| 11:27.15 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 11:27.27 | ManxPower | mrproper_: does 1.0.9 work? |
| 11:27.50 | puzzled | zoo: thanks |
| 11:27.53 | mrproper_ | ManxPower: cant go back that far, i have an issue with oh323 in 1.0.9 |
| 11:28.10 | [Jedi] | we have an oracle which stops working after 289 days of use |
| 11:28.11 | zoo | puzzled: http://bugs.digium.com/view.php?id=4822 |
| 11:28.15 | [Jedi] | I really hate long-time bugs |
| 11:28.31 | darkskiez | ManxPower: is there a bug id for this so i can monitor the situation? |
| 11:28.32 | ManxPower | mrproper_: I meant just to make sure that it builds on your system |
| 11:28.50 | darkskiez | ManxPower: or a way to reinit the devices without stopping * |
| 11:29.45 | ManxPower | darkskiez: There is no bug ID since 1) it's an issue for support@digium.com and 2) I didn't feel like spending a couple of weeks defending the bug to all the people on bugs.digium.com that think there are no bugs in Asterisk |
| 11:29.56 | mrproper_ | ManxPower: will try |
| 11:30.14 | [Jedi] | uhm |
| 11:30.55 | darkskiez | ManxPower: I see. |
| 11:31.04 | darkskiez | ManxPower: so would I get abuse for creating a bug report? |
| 11:31.21 | ManxPower | darkskiez: Yes, since it's really an issue for bugs.digium.com |
| 11:31.36 | darkskiez | ahhh, conflicting answer. |
| 11:32.00 | ManxPower | Digium support always tells me to use CVS-HEAD so I stopped even trying to get the problem fixed. |
| 11:32.18 | darkskiez | I was using cvs head, but after a month it borked |
| 11:32.47 | darkskiez | now its a month old :) |
| 11:32.58 | ManxPower | There's nothing wrong with using CVS-HEAD as long as it fits with your needs. It doesn't fit with my needs. |
| 11:36.17 | ManxPower | I need something that doesn't change behavour between updates, something that doesn't need a lot of testing before deployment of an update. CVS-HEAD doesn't fit the requirements. |
| 11:40.15 | puzzled | what's this VPM module that is mentioned in a couple of cvs fixes and on the -dev list? |
| 11:40.31 | ManxPower | puzzled: The EchoCan module, I think |
| 11:40.53 | puzzled | thanks, that makes total sense :) |
| 11:41.18 | puzzled | think I would have caaled it the ECM - echo cancellation module |
| 11:42.49 | ManxPower | One would think. |
| 11:44.00 | puzzled | then again ECM is prolly taken and stands for plasma conduit flux capacitor regulators |
| 11:44.07 | MrChimpy | is hardware echo cancellation actually worthwhile? |
| 11:44.16 | puzzled | more than software echo can me thinks |
| 11:44.23 | ManxPower | MrChimpy: It would not be if Asterisk's EchoCan didn't suck so much. |
| 11:44.25 | MrChimpy | seems to double the price of the 4x board |
| 11:44.31 | MrChimpy | ah |
| 11:44.50 | MrChimpy | given previous boards cost £14,000 to do the same I think we can splash out |
| 11:45.23 | puzzled | hehe |
| 11:45.25 | MrChimpy | where does the echo come from? the telephone handset? |
| 11:46.03 | MrChimpy | I'm just feeling my way round the terminology/hardware and stuff |
| 11:46.15 | puzzled | or anything in between when going from digital -> analog and back |
| 11:46.33 | puzzled | analog has a way of cross interference |
| 11:46.55 | *** join/#asterisk |dennis| (~dennis@200.32.215.82) |
| 11:47.07 | MrChimpy | our current PBX crashed this morning. it somehow relies on two windows desktop machines :( |
| 11:47.38 | MrChimpy | i think job #1 is replace that before attempting monster sized asterisk install |
| 11:48.14 | MrChimpy | if you have several asterisk servers does gig ethernet linking to share voice conferences across boxes actually work? |
| 11:48.33 | MrChimpy | I saw it mentioned in the handbook, but that doesn't really confirm anything :) |
| 11:48.46 | [Jedi] | MrChimpy: fast ethernet ant TDMoE or IAX2 is more than enough |
| 11:49.14 | [Jedi] | MrChimpy: how many channels do you need to carry over ethernet? |
| 11:49.28 | MrChimpy | not sure. project hasn't been specced yet. |
| 11:50.00 | ManxPower | MrChimpy: ulaw or alaw (the codecs that uses the most bandwidth) use about 8kilobytes/second |
| 11:50.12 | ManxPower | Which is something like .000008 of a 100Mbps Ethernet |
| 11:50.26 | MrChimpy | i'll either end up replacing a *large* IVR installation, or just building a voip gateway for it |
| 11:51.08 | MrChimpy | manx: sure |
| 11:51.15 | MrChimpy | current install has ATM for that |
| 11:51.35 | MrChimpy | but then it is several years old |
| 11:51.37 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 11:52.01 | *** join/#asterisk YveDeLarc (~user@i3ED6E0A0.versanet.de) |
| 11:56.47 | *** part/#asterisk Patrick04 (Patrick04@cpc1-bahd1-3-0-cust51.renf.cable.ntl.com) |
| 12:02.57 | MrChimpy | ah yes, another question |
| 12:04.55 | *** join/#asterisk dvh_bk (~dvh_bk@81.2.42.2) |
| 12:05.38 | *** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com) |
| 12:06.17 | MrChimpy | we have intertel 8520 phones |
| 12:06.23 | MrChimpy | they're not IP |
| 12:06.51 | MrChimpy | they do have CLID type displays etc, get the time from the remote end and suchlike |
| 12:07.07 | MrChimpy | i believe there's a standard for such things? what's it called? |
| 12:08.45 | dvh_bk | ÎÁ opennet ×ÙÛÌÁ ÓÔÁÔØÑ ÞÔÏ ÐÏÒÔÒÉÒÏ×ÁÌÉ FreeBSD ÎÁ XBox ! ÔÁËÏÊ ×ÏÐÒÏÓ ÎÉËÔÏ ÎÅ ÐÒÏÂÏ×ÁÌ ÐÒÉËÒÕÞÉ×ÁÔØ ÁÓÔÅÒÉÓË ÎÁ XBOX???? |
| 12:09.58 | tzanger | heh |
| 12:11.10 | MrChimpy | dvh_bk: yep. me too. |
| 12:16.24 | MrChimpy | ah, having looked through the spec it seems all the stuff that the phone supports can be done through CLID |
| 12:17.19 | *** join/#asterisk PG3 (CyberSword@dup-148-221-68-217.prodigy.net.mx) |
| 12:19.53 | ManxPower | MrChimpy: PBX phones do not have a standard. If there was a standard then other companies could make phone for the PBX and that would cut reduce the MASSIVE profit margins PBX vendors have on their phones. |
| 12:20.27 | *** join/#asterisk Kernel_core (Raph@217.218.94.206) |
| 12:21.06 | Kernel_core | hi all , is there any web based management for configureing , zapata.conf and zaptel.conf and extension.conf ? |
| 12:21.17 | ManxPower | Kernel_core: Zillions of them. |
| 12:21.21 | ManxPower | All of them suck. |
| 12:21.33 | ManxPower | But AMP seems to be the most popular of them. |
| 12:22.04 | Kernel_core | ManxPower: many of users are to lame to use CLI ! |
| 12:22.57 | Kernel_core | and they afraid of command line environment... |
| 12:22.59 | ManxPower | Asterisk is not designed to be managed by users, it's designed to be managed by admins |
| 12:23.16 | ManxPower | Users don't manage traditional PBXs |
| 12:23.35 | MrChimpy | arses. though I think if we did replace our shitty PBX we'd go full VOIP |
| 12:23.51 | ManxPower | MrChimpy: I would NEVER go vull VoIP. |
| 12:24.01 | ManxPower | ITSPs and the internet is just not reliable enough. |
| 12:24.24 | ManxPower | On my systems I've mostly brought a PRI into Asterisk, then connect phones to Asterisk using either SIP or Analog |
| 12:24.30 | MrChimpy | no, I mean VOIP to the desk |
| 12:24.45 | ManxPower | If we start running out of channels then maybe failover to VoIP to an ITSP |
| 12:25.07 | MrChimpy | we're a telco anyway, we have plenty of lines :) |
| 12:25.26 | ManxPower | That does help 8-) |
| 12:25.38 | ManxPower | We use Polycom Soundpoint IP 300s and 500s, BTW |
| 12:25.58 | *** join/#asterisk amir (~amir@195.226.9.186) |
| 12:26.41 | MrChimpy | looks cool |
| 12:27.09 | ManxPower | We don't use Cisco phones because the SIP firmware is an extra cost item and the power supply is an extra cost item |
| 12:27.10 | Kernel_core | ManxPower: it seems AMP isnot free ..... |
| 12:27.29 | MrChimpy | yep, i've heard you have to pay licence on cisco |
| 12:27.34 | PG3 | AMP is free |
| 12:27.36 | PG3 | :p |
| 12:27.51 | ManxPower | Kernel_core: Since AMP is included in several open source projects likes Asterisk@Home, it has to be free. |
| 12:28.02 | PG3 | the instalation cost |
| 12:28.05 | PG3 | but the AMP is cree |
| 12:28.22 | PG3 | those prices that u see is from the instalation cost if u want that those guys install amp in your pc |
| 12:28.22 | Kernel_core | good |
| 12:28.28 | ManxPower | MrChimpy: Polycom includes SIP firmware (if you get the right model) and a power supply. PoE cables are special and run about $30 |
| 12:28.31 | PG3 | i got confused too the first time |
| 12:28.39 | Kernel_core | is installation too complicated ?! |
| 12:28.59 | ManxPower | Kernel_core: Dude, this is VoIP and Telecom, of course it's complicated! |
| 12:29.10 | gordonjcp | Kernel_core: it's really easy to set up asterisk from the command line |
| 12:29.29 | gordonjcp | but anything that is as powerful and flexible as asterisk will by its very nature be complicated |
| 12:29.37 | Kernel_core | gordonjcp: yes , after 3 month playing hard with asterisk , it is easy for me too |
| 12:30.13 | Moc | ManxPower, your comming to cluecon ? |
| 12:30.23 | ManxPower | Moc: Hell no. |
| 12:30.38 | Moc | why ? |
| 12:31.06 | ManxPower | Moc: Um, because I just spend $8,000 on a month long trip to Europe. |
| 12:31.18 | Moc | lol yea your right... how did it goes ? |
| 12:31.26 | MrChimpy | weird. you mean the PoE cable at the phone end? |
| 12:31.33 | ManxPower | MrChimpy: yes. |
| 12:31.33 | MrChimpy | (ManxPower) |
| 12:31.36 | ManxPower | Moc: it was great |
| 12:31.39 | *** join/#asterisk darby_t (~tom@host-ip237-209.crowley.pl) |
| 12:31.40 | MrChimpy | bastards! ;) |
| 12:31.58 | RaYmAn-Bx | ManxPower: what countries did you visit? |
| 12:31.59 | *** join/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
| 12:32.04 | MrChimpy | what's with manxpower? you from the isle of man? |
| 12:32.11 | ManxPower | MrChimpy: The polycom 300 and 500 don't have PoE support in the phone, so a special PoE cable will convert from PoE into whatever the phone expects. |
| 12:32.17 | ManxPower | MrChimpy: no. |
| 12:32.25 | MrChimpy | heh. just checking :) |
| 12:32.34 | Moc | ManxPower, it wouldn't be so bad if those cable were 10$ !!! |
| 12:33.03 | ManxPower | RaYmAn-Bx: Stockholm Sweden (7 nights), Antwerp Belgium (6 nights), Delft Netherlands (1 night), Amsterdam Netherlands (8 nights), Eindhoven Netherlands (2 nights), Madrid Spain (7 nights) |
| 12:33.16 | *** join/#asterisk nitram (foo@superblob.com) |
| 12:33.17 | RaYmAn-Bx | okay |
| 12:33.38 | Moc | that alot of nights !! hehe |
| 12:34.24 | ManxPower | During my trip I also attended VON and Astricon |
| 12:35.41 | Moc | hehe |
| 12:38.57 | Moc | I'll probably do that someday.. |
| 12:39.41 | Moc | My travel so far are "Cuba, Toronto, and Quebec city..." I'll add Chicago next week |
| 12:39.58 | Moc | I love going to toronto in train.. |
| 12:40.06 | ManxPower | Toronto is awesome. We went there for the past 2 summers. |
| 12:40.07 | Moc | but it suck when I arrive there hehe |
| 12:40.11 | Moc | it BORING |
| 12:40.19 | gordonjcp | Kernel_core: it's difficult to fly a passenger jet plane, too. Ever wondered why? |
| 12:40.22 | Moc | come to MTL next time... alot more interesting hehe |
| 12:40.39 | ManxPower | I have little interest in French or Canadian French culture. |
| 12:40.43 | Moc | Quebec might be hard if you do not speech french |
| 12:41.21 | ManxPower | Moc: I'm an American, of course I don't speak French. |
| 12:41.47 | Moc | well do not go to Quebec city to talk with people then ;) but Montreal is as english as french.. |
| 12:42.05 | *** join/#asterisk lehel (~Lehel@82.79.20.17) |
| 12:42.11 | *** join/#asterisk virterm (~virterm@shiva.kanatek.com) |
| 12:42.25 | lehel | hello |
| 12:42.45 | Moc | It took me hours to find a McDonals in toronto |
| 12:44.32 | InfraRed | eh? |
| 12:44.36 | Moc | and damn the homeless people are expensive in Toronto... They want 5$, not 25cent, get get angry ! |
| 12:44.42 | InfraRed | why were you LOOKING for a McD? |
| 12:44.57 | InfraRed | just find a decent deli for decent food |
| 12:45.15 | clive- | anyone using a quad or octo-bri card? |
| 12:45.55 | Moc | InfraRed, blah... I got ennuf of "decent" food on the train.. |
| 12:46.03 | Moc | it ennuf for the week.. |
| 12:46.08 | Moc | gota run to work.. bbl |
| 12:48.52 | wintermute_ | Moc: Quebec yes, but not montreal |
| 12:49.38 | wintermute_ | i've been living here practically my whole life... i went to english school, i speak english nearly everywhere |
| 12:50.20 | fitzel | anybody here that uses a pda as a voip-wlan-client and has some practical experience? |
| 12:50.21 | wintermute_ | it's really not that bad |
| 12:51.21 | Zeeek | t"as qu'à faire attention en clase ! |
| 12:51.29 | *** part/#asterisk dvh_bk (~dvh_bk@81.2.42.2) |
| 12:55.03 | lathos42 | eh? :) |
| 12:55.05 | ManxPower | I don't hate french culture. I just have no interest in it. I also don't have any interest in the culture of Turkey either. Don't take it personally. |
| 12:55.47 | Zeeek | On s'en fout pas mal, ManxPower :) |
| 12:56.08 | Zeeek | (roughly, no one gives a toss) |
| 12:56.12 | lathos42 | Zeeek: Its like you're speaking a foreign language or something |
| 12:57.23 | wintermute_ | je m'en faout aussi, meme si mon pere est francais ;) |
| 12:57.25 | Zeeek | speaking of culture... Sony lost a ten million dollar lawsuit. THey were actually paying radio stations to play J'Lo's last album - horro of horrors! |
| 12:57.53 | Zeeek | "payola", a standard parctice since the 30's, has been illegal since the fifties |
| 12:58.05 | Zeeek | but everyone knows the music you hear is bought and paid for |
| 12:58.23 | Zeeek | that's why the internet is great. Vive asterisk ! |
| 13:00.10 | lathos42 | Zeeek: Well, they probably had to pay them to play it.. I mean nobody would actually want to listen to that :) |
| 13:01.10 | *** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net) |
| 13:01.34 | Zeeek | never hoid it! but they all pay, trips to Brazil, flat screen hi-def tvs, etc |
| 13:01.57 | Zeeek | like politicians, they're all a bunch of crooks, Nixon got caught |
| 13:02.17 | Zeeek | i gotta go, I've worked to hard today already |
| 13:02.34 | *** part/#asterisk Zeeek (~icechat5@Zeeek.active.supporter.pdpc) |
| 13:10.02 | *** join/#asterisk FunXioN (~nunya@mtnuser.icgws.com) |
| 13:15.42 | *** join/#asterisk jiro5281 (~anton@203.177.242.192) |
| 13:18.14 | *** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net) |
| 13:18.22 | *** join/#asterisk mut (~animenodv@65.111.201.79) |
| 13:22.20 | *** join/#asterisk malaiwah (~malaiwah@Quebec-HSE-ppp242302.qc.sympatico.ca) |
| 13:23.06 | malaiwah | hi everyone, i have some questions about asterisk+spandsn (txfax/rxfax)... any1 familiar with this |
| 13:23.46 | Darwin35 | anyone know of a good paybyphone creditcard system that works with asterisl |
| 13:24.37 | Darwin35 | asterisk |
| 13:24.48 | nDuff | malaiwah, maybe, maybe not. Ask your questions and wait -- someone who *is* familiar may come along and read their backlog, or there may be someone who knows enough to answer your questions but not enough to volunteer to answer any arbitrary spandsp question without knowing what you're going to ask. |
| 13:25.28 | malaiwah | nDuff: thanks |
| 13:26.13 | malaiwah | i have some problems sending faxes with the txfax application; once the call is answered, there's no audio coming out of asterisk.. the fax doesn't get sent |
| 13:26.16 | mut | these peppered beef steak nuggets are good |
| 13:26.42 | Darwin35 | spandsp |
| 13:26.49 | Darwin35 | what do you need to knpw |
| 13:26.52 | Darwin35 | know |
| 13:27.35 | Darwin35 | is the otherside initating the fax tone |
| 13:27.36 | *** join/#asterisk Jearil (colin@67.151.160.162) |
| 13:28.05 | malaiwah | i think this problem is related.. with rxfax, i can't receive any fax unless i playback() something on the channel first |
| 13:28.07 | jake1932 | just bought a few g729 licenses - but it looks like asterisk is tyring to play prompts in g729 and failing - did i miss some settings? |
| 13:28.15 | ManxPower | malaiwah: You have to specify the "caller" option to txfax |
| 13:28.32 | malaiwah | manx: did it, txfax(file.tif|caller|debug) |
| 13:28.52 | ManxPower | malaiwah: maybe you have run Answer() before receiving? |
| 13:28.55 | malaiwah | darw: the otherside is giving asterisk a lot of fax tones. |
| 13:29.01 | jake1932 | <PROTECTED> |
| 13:29.13 | malaiwah | manx: did it too ;-) answer, playback(dummy), rxfax() |
| 13:29.25 | ManxPower | jake1932: does vm-youhave.g729 exist? |
| 13:29.30 | Jearil | alright.. trying to install the mysql cdr addon.. asterisk is unable to connect to mysql and I'm unsure as to why. |
| 13:29.36 | ManxPower | jake1932: Looks like you didn't install the G729 codec correctly. |
| 13:30.01 | jake1932 | ok - but just wanted to know that * should be able to play prompts using g729... |
| 13:30.12 | jake1932 | (once everything is installed correctly) |
| 13:30.22 | ManxPower | jake1932: The G279 codec allows Asterisk to convert between G729 and other formats |
| 13:30.28 | jake1932 | ok - tnx |
| 13:31.39 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 13:32.42 | malaiwah | with de debug option, txfax outputs what it hears to /tmp/audio-xxxxxx.. but these files are 0 in size |
| 13:33.24 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 13:33.29 | malaiwah | what can i look for ? why must I playback something before rxfax kicks in? |
| 13:33.55 | malaiwah | maybe I should try to playback something before txfax but i guess that it is just hiding the problem |
| 13:34.16 | *** join/#asterisk FunXioN (~nunya@mtnuser.icgws.com) |
| 13:34.27 | ManxPower | malaiwah: playback answers the line before playing |
| 13:35.08 | jake1932 | ManxPower: i did everything right except move the module into the correct folder - it works now - tnx again |
| 13:35.09 | malaiwah | then.. with txfax, i should Answer() and then TxFAX() ? and not just TxFAX ? |
| 13:35.54 | malaiwah | my call is auto-dialed from a ".call" file in the outgoing directory |
| 13:36.33 | ManxPower | You don't need to answer for outgoing calls, only incoming calls |
| 13:37.05 | malaiwah | manx: then that can't be my problem with txfax.. outgoing call is made from the outgoing directory and passed to the application txfax |
| 13:37.34 | malaiwah | manx: with the file.tif|caller as a parameter |
| 13:37.43 | ManxPower | malaiwah: try sending a fax to your voice phone, do you hear the tones when you pick up the call |
| 13:37.57 | malaiwah | manx: already tried it, i don't hear anything |
| 13:37.59 | ManxPower | malaiwah: Rememebr fax doesn't usually work via VoIP |
| 13:38.15 | malaiwah | manx: yeah, i know.. but I thought i could hear tones anyway ? |
| 13:40.11 | malaiwah | manx: i know that getting the timing right is tricky, and only ulaw can be used for fax over VoIP, but receiving faxes works allright (when i playback something first) |
| 13:40.41 | ManxPower | As I said for receiving faxes just do an answer first |
| 13:41.11 | *** join/#asterisk bikokola (~amal@203.134.85.66) |
| 13:41.48 | malaiwah | manx: call is already answered, but do i need to pause before kicking in rxfax? |
| 13:42.26 | lehel | anyone CAPI ? |
| 13:43.28 | ManxPower | malaiwah: should not need to. |
| 13:44.01 | malaiwah | manx: thanks. |
| 13:46.55 | Darwin35 | man fax is outdated if you have email and a scanner |
| 13:46.58 | ManxPower | god hates me. Today is the first day of the MIS manager's vacation and today is the first time we started geting HDMC Abort errors |
| 13:47.12 | Darwin35 | and scanners are dirt cheap |
| 13:47.17 | ManxPower | ..er... HDLC Abort errors |
| 13:47.19 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
| 13:47.25 | ManxPower | Darwin35: Try it some time. |
| 13:47.48 | malaiwah | manx: i tried swapping the "txfax" application for the "playback" application in my scripts and then trying faxing.. asterisk calls my phone an when i answer i get the playback message.. why doesn't it work that way with txfax ? |
| 13:48.12 | malaiwah | darwin: but what about other people that don't have emails and just fax ? ;-) |
| 13:48.46 | bikokola | hey guys, i have a lil problem, after downloading asterisk to desktop, i get to console, get into its directory, after unpacking. Then i type make, followed by make install. Make completes succesfully, make install gives an error, help pls |
| 13:48.54 | ManxPower | 1) Scan, 2) convert into a format the receiver can read, 3) find file, 4) attach file, send file, 5) get bounce message because message is too big, 6) go to fax machine, 7) insert documents, 8) punch in phone number, 9) walk away |
| 13:49.08 | malaiwah | manx: right ! |
| 13:49.26 | ManxPower | Of course fax has it's own issues. |
| 13:49.35 | lehel | anybody capi Callback? |
| 13:49.36 | malaiwah | how can i get txfax to be more verbose in the asterisk console ? |
| 13:49.44 | ManxPower | Our users can't even figure out how to subscribe to an IMAP folder. |
| 13:49.57 | Darwin35 | thats sad |
| 13:49.59 | malaiwah | i tried set verbose 10 but txfax isn't more verbose.. |
| 13:50.00 | ManxPower | malaiwah: start asterisk as "asterisk -cvvvd" |
| 13:51.09 | malaiwah | manx: i guess the "d" parameter is what i was missing |
| 13:51.43 | Darwin35 | well here at this office we use a scanner and halifax for faxing and a laser printer |
| 13:51.45 | Katty | mew |
| 13:52.41 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
| 13:52.48 | astoria | Good morning, all. |
| 13:52.54 | Darwin35 | a freind wrote a script you hit the f1 key on the machine and it scans converts and faxes it getss the nmbr your calling and everything |
| 13:53.08 | Darwin35 | yet to test with asterisk |
| 13:53.18 | Darwin35 | if it works might bea answer |
| 13:53.25 | Darwin35 | or a add on |
| 13:54.19 | malaiwah | interesting.. i guess that txfax doesn't get the "|caller" option right |
| 13:54.33 | malaiwah | it seems to be waiting for something.. |
| 13:54.43 | malaiwah | Data: /tmp/myfile.tif|caller |
| 13:54.48 | malaiwah | is what i wrote in my ".call" file |
| 13:56.05 | ManxPower | can you send to a normal fax machine? |
| 13:56.45 | malaiwah | Urgent handler |
| 13:56.45 | malaiwah | File name is '/var/spool/asterisk/tmp/1122472576.tif' |
| 13:56.45 | malaiwah | Changed from phase 0 to 1 |
| 13:56.56 | malaiwah | when trying a normal fax machine |
| 13:57.02 | malaiwah | it gets stuck here |
| 13:57.22 | malaiwah | what is the "urgent handler" anyway? |
| 13:57.26 | bikokola | can someone please help me with an error i get during "make install", |
| 13:58.01 | DarthClue | bikokola: not unless you can tell us what the error is. and even then, it may not be possible. |
| 13:58.04 | malaiwah | oh right.. i think i found the bug |
| 13:58.26 | bikokola | ok, |
| 13:58.34 | ManxPower | ~pastebin |
| 13:58.34 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 13:59.05 | bikokola | the error is mkdir: Cannot create directory "var/lib/asterisk" PERMISSION denied |
| 13:59.43 | DarthClue | what user are you running make under? |
| 13:59.59 | bikokola | the only user, the admin |
| 14:00.17 | bikokola | make works fine, but make install stuff's up |
| 14:00.57 | otmar | sudo make install ? |
| 14:00.58 | clive- | ~seen kapejod |
| 14:01.00 | jbot | kapejod <~kapejod@e178053088.adsl.alicedsl.de> was last seen on IRC in channel #asterisk, 56d 50m 3s ago, saying: 'or a bristuff setting'. |
| 14:01.01 | DarthClue | bikokola: running as root? and getting that error? well, the error is pretty clear, it doesn't have permission to do what you want. |
| 14:01.21 | bikokola | but im root user |
| 14:01.40 | otmar | bikokola: what's the output of "id" ? |
| 14:01.51 | bikokola | simply typing id in console? |
| 14:01.55 | otmar | yes |
| 14:03.53 | bikokola | uid=500(bikola) gid=500(bikola) groups=500(bikola) context=user_u:system_r:unconfined_t |
| 14:04.23 | ManxPower | bikokola: you are not root |
| 14:04.33 | jake1932 | is there a way to get fxotune using stable? |
| 14:04.43 | bikokola | what |
| 14:04.45 | bikokola | how |
| 14:04.50 | pjz | can anyone point me to the changelog for the newest zaptel stuff? |
| 14:04.53 | jake1932 | (stable 1.07) |
| 14:04.57 | otmar | ~seen joshnet |
| 14:04.57 | jbot | otmar: i haven't seen 'joshnet' |
| 14:05.05 | tzanger | "I really want the features in the Cadillac but I don't want to pay for it" |
| 14:05.09 | bikokola | how do i create a root user |
| 14:05.12 | *** join/#asterisk mistik1 (rasta@ool-44c02704.dyn.optonline.net) |
| 14:05.14 | ManxPower | pjz: It's in the Asterisk Sorce |
| 14:05.20 | tzanger | ~seen my dick in 5 years |
| 14:05.20 | jbot | i haven't seen 'my dick in 5 years', tzanger |
| 14:05.21 | jake1932 | clive: I've been e-mailing him |
| 14:05.21 | ManxPower | bikokola: "su -l" |
| 14:05.23 | otmar | just do a "sudo make install" |
| 14:05.42 | bikokola | aight thx |
| 14:05.46 | mistik1 | morning folks |
| 14:05.53 | ManxPower | bikokola: And learn linux. |
| 14:06.48 | jake1932 | guess i should be more specific - I'm using asterisk stable 1.07 - is there any way to get fxotune without a full d/l and recompile? |
| 14:07.12 | mistik1 | I managed to get my asterisk server up and running, I can accept calls and all that but If i add an extension the calls my friend on his server no traffic at all gets to him |
| 14:07.28 | ManxPower | jake1932: no. |
| 14:07.51 | jake1932 | tnx |
| 14:08.01 | mistik1 | for example when I try to use the demo to test it also tries to call digium for the demo and just hangs, no connection is ever made |
| 14:08.21 | mistik1 | anyone have a clue what could be going on |
| 14:08.35 | ManxPower | mistik1: not without lots of additional information, which I do not have time to look at. |
| 14:09.10 | jake1932 | mistik1: you have to at least do a "iax2 debug" |
| 14:11.42 | tzafrir | jake1932, zaptel in HEAD is not that different ffrom zaptel in stable. Try grabbing fxotune.c from HEAD and building it |
| 14:12.05 | jake1932 | tzafrir: thanks i'll try that |
| 14:12.14 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 14:12.49 | jake1932 | <PROTECTED> |
| 14:12.52 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 14:13.07 | ManxPower | fxotune won't work with CAPI |
| 14:13.28 | jake1932 | no - i have a tdm400 i want to use that on (seperate deal) |
| 14:13.49 | jake1932 | sry bout that - wasn't clear |
| 14:13.52 | *** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
| 14:14.08 | lehel | jake1932: you have a CAPI working? |
| 14:14.10 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:14.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:14.21 | jake1932 | lehel: I'm darn close |
| 14:14.39 | jake1932 | lehel: Verizon says my SPID is not programmed properly |
| 14:14.55 | ManxPower | jake1932: what is your SPID? |
| 14:15.02 | ManxPower | same as your phone number? |
| 14:15.19 | jake1932 | ManxPower: 215XXXXXXX0101 |
| 14:15.29 | [Jedi] | when I was in stable I used "h323". Which is better for unstable, "oh323" or "h323"? |
| 14:15.37 | jake1932 | ManxPower: they say I have it programmed as 215XXXXXXX01 |
| 14:15.47 | lehel | jake1932: could you help me too to get closer |
| 14:16.12 | ManxPower | jake1932: try 215XXXXXXX010101 then |
| 14:16.20 | ManxPower | or 215XXXXXXX01 |
| 14:16.26 | ManxPower | Oh! Wait! |
| 14:16.41 | [Jedi] | also, which openh323 versions are required in HEAD? the ones I had installed for stable don't compile on head |
| 14:16.41 | jake1932 | ManxPower: is the SPID the same as MSN? |
| 14:16.50 | ManxPower | Verizon? I don't think you can use USA BRI with Asterisk |
| 14:16.57 | ManxPower | jake1932: I don't know. |
| 14:17.21 | jake1932 | ManxPower: If I get this - it's going in the wiki |
| 14:17.36 | jake1932 | I'm using debian _ asterisk 1.07 |
| 14:17.37 | ManxPower | [Jedi]: For CVS-HEAD use chan_h323 (included) or use the new H323 driver that Digium paid for in asterisk-addons |
| 14:17.49 | [Jedi] | uhm ok |
| 14:17.59 | Katty | Anyone framilier with Michael Leunig? |
| 14:18.04 | [Jedi] | I don't see anything in addons |
| 14:18.16 | lehel | i have three isdn numbers.. i put them in capi.conf (MSN=...) .. what else? |
| 14:18.22 | jake1932 | lehel: which distro? |
| 14:18.24 | lehel | debian |
| 14:18.29 | [Jedi] | ok using an older version |
| 14:18.39 | ManxPower | [Jedi]: hold on |
| 14:18.54 | mistik1 | jake1932: turned on debug and got nothing usefull from the call attempt, it just sits there and never ends, the only debug info I get is from my client heartbeat |
| 14:19.21 | [Jedi] | ManxPower: the directory tree is empty |
| 14:19.32 | [Jedi] | oh I don't understand cvs very well |
| 14:19.34 | Katty | DarthClue: new |
| 14:19.34 | ManxPower | [Jedi]: Well I see CVS comits for that directory |
| 14:19.35 | Katty | oh |
| 14:19.37 | [Jedi] | if I checkout into an empty dir, it works |
| 14:19.37 | Katty | DarthClue: mew |
| 14:19.39 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 14:19.39 | jake1932 | mistik1: and set verbose 15 or higher |
| 14:19.48 | Katty | DarthClue: http://www.geocities.com/rainforestwind/server_on_fire_md_wht.gif <- the windows box i'm fixing. |
| 14:19.48 | ManxPower | [Jedi]: cvs co asterisk-addons |
| 14:19.58 | [Jedi] | if I checkout into my already existing asterisk-addons, I get empty directory tree for that addon |
| 14:20.11 | [Jedi] | things like these make me love SVN :)) |
| 14:20.13 | ManxPower | [Jedi]: delete your local asterisk-addons then |
| 14:20.33 | *** join/#asterisk tinpot (~nick@217.145.120.198) |
| 14:20.36 | ManxPower | You should NEVER check out a different tree (i.e. HEAD .vs. -r v1-0) into the same directory |
| 14:20.44 | jake1932 | lehel: i can give you an idea of how I did it with Debian and AVM Fritz PCI |
| 14:20.51 | DarthClue | Katty: nice box, is that one of those new features implemented with the windows update authentication patches? |
| 14:20.52 | lehel | jake1932: you have any useful doc? wich maybe helps me.. on what distro is your asterisk running? .. i have a fritz |
| 14:21.06 | lehel | ok jake |
| 14:21.51 | Katty | DarthClue: nodnod |
| 14:22.19 | jake1932 | lehel - used this doc: http://www.voip-info.org/tiki-index.php?page=Asterisk+How+to+connect+with+CAPI and this http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install |
| 14:22.35 | tinpot | hello all |
| 14:23.09 | jake1932 | lehel: and used asterisk stable 1.07 |
| 14:23.25 | tinpot | when you set the gains in zapata I have a situation where it only sets the gains on inbound calls - this shouldn't be the case??? |
| 14:23.47 | ManxPower | tinpot: I have never ever heard of that. |
| 14:23.48 | jake1932 | lehel: I've e-mailed kapejod yesterday - hopefully he'll get back to me this week |
| 14:23.55 | ManxPower | tinpot: But there IS something you should know about gains |
| 14:24.07 | tinpot | go on |
| 14:24.36 | lehel | jake1932: i know that docs, my chan_capi is loaded well, my fritz is compiled ok.. i have problems mostly with the extensions .. dialing rules with capi :( |
| 14:25.14 | jake1932 | lehel: dialing rules? can you be more specific? |
| 14:25.24 | ManxPower | Let's assume you have a TDM400P with an FXO and an FXS port. A call comes into the FXO and goes to an analog phone on the FXS. Not he's say you put rxgain=5.0 and txgain=5.0. Well your total gain will be 10 because you get rxgain of 5 coming into the FXO and a txgain of 5 coming out of the FXS. |
| 14:25.53 | tinpot | I agree |
| 14:27.05 | tinpot | I set both the rxgain and txgain to -90, when I call in patch straight through to a sip phone - you can hear nothing, get the same sip phone to call out over that zap interface (kapes actually) and you can hear fine |
| 14:27.12 | mistik1 | darnit |
| 14:27.47 | ManxPower | um -90 would make you hear nothing. |
| 14:28.12 | tinpot | exactly - but this only happens if teh call is orifginated in one direction |
| 14:28.21 | ManxPower | - means softer, + means louder. In decibles, which is not a linear scale. |
| 14:28.34 | lehel | jake1932: i have 3 number (MSN) in capi.conf, but i dunno whatto add in extensions.conf (exten => ...Dial(CAPI/... ??)).. how to use those numbers? this kinda stuff :( |
| 14:29.05 | ManxPower | tinpot: I don't beliece you have a gain problem. comment out your rxgain and txgain and issue a unload chan_zap.so and a load chan_zap.so |
| 14:29.17 | ManxPower | tinpot: you have some OTHER issue. |
| 14:30.00 | tinpot | I have unconmmented the gains - and the voice comes back - but my concern is the gain levels are not setting the gain on outbound calls as a result of this test |
| 14:30.03 | ManxPower | -90 db would be like the sound of electrons in an atom. +90 db would be something like a jet engine. |
| 14:30.35 | tinpot | I know - I effectivly want to switch it off to ensure it was adjusting in all cases - but it wasn't |
| 14:30.59 | *** join/#asterisk w0w0 (~w0w0@14.Red-81-39-84.pooles.rima-tde.net) |
| 14:31.00 | ManxPower | *gain=0 would make asterisk not do any adjustment. |
| 14:31.08 | jake1932 | lehel: from what I can see (mine hasn't worked fully yet), you need to put in extensions.conf: Dial,CAPI/@MyMSN:${EXTEN} |
| 14:32.35 | lehel | jake1932: it is possible to dial virtually (ex.: from FireFly)? |
| 14:33.04 | jake1932 | lehel: of course |
| 14:33.18 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 14:33.19 | tzanger | none of y'all know if there's a specifically-formatted SMS I can send to my CDMA phone to make it tell me there's voicemail? I imagine that's how the carriers set the VM notification I just don't know the format |
| 14:33.25 | lehel | what is the dial_out number ? (9?) |
| 14:33.41 | jake1932 | lehel: it's whatever you configured in extensions.conf |
| 14:33.59 | jake1932 | tzanger: i believe it's a different message altogether |
| 14:34.16 | ManxPower | tzanger: There's supposed to be, but I can't find the technical details. There's a company in the usa that sells a toolkit for use with asterisk to do it, don't think it's open source. |
| 14:34.31 | lehel | jake1932: it is needed some AGI script? |
| 14:35.07 | jake1932 | lehel: no - just put the dial cmd in your firefly context |
| 14:35.31 | tzanger | ManxPower: well if they have an eval I'll just sniff the ethernet link :-) |
| 14:36.05 | *** part/#asterisk hapoteh (hapoteh@yossman.net) |
| 14:38.26 | tzanger | I found kannel |
| 14:38.33 | tzanger | which is an OSS SMS/WAP gateway |
| 14:39.35 | *** join/#asterisk _deg_ (~deg@200.146.0.254) |
| 14:40.22 | astoria | kannel is pretty cool. |
| 14:41.12 | lehel | jake1932: exten => 1551,1,Dial(${ANAME}:${EXTEN},40,Tt) . is this correct? |
| 14:41.45 | blitzrage | tzanger: can't you just send an email to your phone? (I know thats probably not what you're looking to do though) |
| 14:42.14 | *** part/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
| 14:42.16 | tzanger | blitzrage: yes I can |
| 14:42.17 | jake1932 | lehel: I would make it real plain first - i.e. have an extensions from firefly (1551) dial a CAPI number |
| 14:42.26 | tzanger | but I want the MWI to light up so I can have it dial the * server for voicemail |
| 14:42.40 | jake1932 | lehel: use hardcoded values first - then experiment with the variables |
| 14:42.43 | blitzrage | tzanger: you have a MWI on your cell? |
| 14:42.52 | tzanger | blitzrage: my gut instinct tells me that it's just a specific format of SMS message |
| 14:42.56 | tzanger | I'm trying to find tha tformat ;-) |
| 14:42.59 | tzanger | blitzrage: yes |
| 14:43.02 | *** join/#asterisk kshumard (~root@207.111.174.1) |
| 14:43.03 | blitzrage | tzanger: hrmmmmmmm, quite interesting |
| 14:43.07 | MicC_ | who here knows call center VOIP? |
| 14:43.07 | tzanger | all cells do |
| 14:43.10 | blitzrage | tzanger: good luck! |
| 14:43.18 | MicC_ | what would I need for 4000 lines |
| 14:43.19 | MicC_ | hehe |
| 14:43.21 | jake1932 | tzanger: is it CDMA, GSM, etc? |
| 14:43.23 | *** join/#asterisk CrashHD (crashhd@206-170-51-220.starstream.net) |
| 14:43.26 | blitzrage | tzanger: really? I've never seen one on my Nokia |
| 14:43.30 | CrashHD | hello |
| 14:43.31 | lehel | jake1932: dial the CAPI number identified in capi.conf as MSN? |
| 14:43.37 | MicC_ | I have an OC12 spec'd out...but what do I need for the TDM to VOIP conversion |
| 14:43.42 | tzanger | holy shit that was easy |
| 14:43.45 | MicC_ | I would had to use 20x Asterisk servers. |
| 14:43.49 | tzanger | kannel has it I think |
| 14:43.55 | CrashHD | I'm looking for a good multi-tenant (voip capable) pbx solution. Any suggestions? |
| 14:44.00 | tzanger | MicC_: OC12? jesus |
| 14:44.11 | tzanger | jake1932: CDMA |
| 14:44.13 | blitzrage | MicC_: you need some hardware - Digium TE411P ? |
| 14:44.14 | jake1932 | lehel: no use the capi.conf number as your msn, and use a known phone number hardcoded |
| 14:44.16 | *** join/#asterisk kshumard (~root@207.111.174.1) |
| 14:44.23 | tzanger | blitzrage: you don't have something that lights up on your phone when you have voicemail? |
| 14:44.26 | tzanger | I have a little mailbox icon |
| 14:44.28 | jake1932 | tzanger: you got it to work w/ CDMA? |
| 14:44.30 | astoria | i managed to get kannel to send out msgs through my t610, it was pretty sweet. |
| 14:44.31 | blitzrage | tzanger: nope, just a message on the screen |
| 14:44.33 | tzanger | er letter icon |
| 14:44.40 | tzanger | astoria: :-) |
| 14:44.42 | astoria | i've only seen kannel work with GSM |
| 14:44.47 | tzanger | astoria: I can just send emails to the email gateway |
| 14:45.06 | *** join/#asterisk mut (~animenodv@65.111.201.79) |
| 14:45.15 | astoria | tzanger: yeah. thats easier. kannel is good if you need to send large amounts of messages and need to connect to an SMSC. |
| 14:45.19 | tzanger | hmm this is too low level I have to see where it's putting this data |
| 14:45.23 | jake1932 | tzanger: does kannel + CDMA = sucess? |
| 14:45.38 | tzanger | I think I'd have to do it through an SMSC unless I can email their gatway with the right format |
| 14:45.41 | tzanger | jake1932: have not tried |
| 14:45.45 | tzanger | I *just* downloaded it |
| 14:45.52 | blitzrage | tzanger: slacker |
| 14:45.57 | jake1932 | tzanger: let me know - i'd like to do that also |
| 14:46.09 | blitzrage | tzanger: get me a mug of milk beotch! |
| 14:46.14 | tzanger | blitzrage: shut up and fetch me four beers, my address book and my conversation hat |
| 14:46.19 | CrashHD | <PROTECTED> |
| 14:46.32 | blitzrage | tzanger: lol, yes sir! |
| 14:46.37 | tzanger | CrashHD: repeating every minute will get you ignored faster than a fat chick at a runway show |
| 14:46.42 | CrashHD | lol |
| 14:46.54 | CrashHD | fat chicks wouldn't get ignored just laughed at |
| 14:46.59 | CrashHD | sorry tz |
| 14:47.05 | tzanger | CrashHD: asterisk is a voicemail capable voip capable pbx |
| 14:47.06 | nDuff | CrashHD, well, you might get that here too. |
| 14:47.14 | CrashHD | well we are an altigen dealer |
| 14:47.27 | CrashHD | we are looking for alternatives to their HPBX software |
| 14:47.37 | tzanger | I wonder if telus will give me the address of their SMSC |
| 14:47.49 | astoria | tzanger: they probably won't know what you're talking about. |
| 14:47.56 | jake1932 | tzanger: for enough money - i'm sure you could get it |
| 14:47.56 | CrashHD | I figured you fine gentleman might know of a good alternative |
| 14:48.05 | tzanger | astoria: :-) |
| 14:48.16 | tzanger | we spend $60k a year on mobile costs... they are eager for our business |
| 14:48.44 | astoria | tzanger: let me know if they will do anything for you. i'm not sure WHERE to call to get hooked up to t-mo's smsc. |
| 14:48.53 | tzanger | myself I'm just discovering that SMS != email gateway |
| 14:49.10 | tzanger | <PROTECTED> |
| 14:49.10 | tzanger | <PROTECTED> |
| 14:49.10 | tzanger | <PROTECTED> |
| 14:49.10 | tzanger | <PROTECTED> |
| 14:49.22 | tzanger | I need to go higher up, I found exactly what I want but I don't know what to do with it now :-p |
| 14:50.44 | lathos42 | Its a pity that probably wouldnt work through SNPP |
| 14:50.57 | tzanger | eh? |
| 14:51.22 | *** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com) |
| 14:51.53 | astoria | lathos42: wha? |
| 14:52.22 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:52.37 | lathos42 | astoria: Most providers have their SNPP gateways open for Text Messaging.. I use that to send pages to myself instead of email |
| 14:52.56 | lathos42 | astoria: but I doubt the SNPP gateway would understand SMS speak |
| 14:53.07 | tzanger | ok how the fuck does kannel get a message from the computer to thep hone |
| 14:53.14 | tzanger | I don't see any gateway cnfiguration |
| 14:53.19 | astoria | lathos42: thats what kannel does.. |
| 14:53.29 | astoria | tzanger: it's a pain in the ass. i don't remember very well, i haven't used it in months. |
| 14:53.38 | astoria | tzanger: there is a sendsms script that goes into a cgi directory. |
| 14:53.45 | *** join/#asterisk FunXioN (~nunya@mtnuser.icgws.com) |
| 14:53.49 | tzanger | astoria: I figured there'd be an IP address or a connection to a phoen required type of thing |
| 14:53.51 | astoria | it's a clusterfuck to configure kannel |
| 14:54.12 | astoria | tzanger: oh, if you want to connect to the phone, you have to setup a serial connection to a gsm modem or something |
| 14:54.26 | astoria | tzanger: i use bluez serial emulation to my t610 |
| 14:54.30 | astoria | i wish i still had my kannel box |
| 14:54.40 | tzanger | ohhhhhhhh |
| 14:54.59 | jake1932 | we need to do this without tying up another phone |
| 14:55.02 | astoria | tzanger: kannel is cool, but way difficult to setup |
| 14:55.05 | tzanger | so kannel goes from computer to phone over serial and SMSs another phone from there |
| 14:55.21 | astoria | tzanger: yeah, that is one configuration. |
| 14:55.34 | astoria | tzanger: you could also use kannel to send SMPP to an SMSC to your phone |
| 14:55.46 | tzanger | I thought kannel was given some remote host (SMSC) and send SMS messages to it over TCP/IP which then transmitted it to thep hone |
| 14:56.14 | astoria | tzanger: yeah, you can do that too |
| 14:56.22 | tzanger | oh okay that's perfect then |
| 14:56.29 | astoria | tzanger: thats the best way to do it, but for testing, it works well just to use my gsm phone |
| 14:56.56 | lathos42 | tzanger: Have you seen this list? http://www.activexperts.com/activsms/smsclist/ |
| 14:57.04 | tzanger | nope I haven't |
| 14:57.06 | tzanger | this is very new to me |
| 14:57.14 | tzanger | I've only used email gateways before |
| 14:57.21 | astoria | I came up with a big business plan for an sms gateway using kannel before. |
| 14:57.23 | astoria | It got me a 4.0 |
| 14:57.25 | tzanger | Telus 14032532266 TAP 1200 7,e,1 |
| 14:57.27 | tzanger | that's me |
| 14:57.37 | tzanger | but that's pager |
| 14:57.56 | astoria | wow, thanks lathos42!! I've been looking for a list like that. |
| 14:57.59 | tzanger | I could use asterisk to do it manually |
| 14:58.01 | astoria | Those are free gateways? |
| 14:58.03 | tzanger | over VOIP, no less |
| 14:58.10 | tzanger | 1200 baud will work just fine over VOIP ;_0 |
| 14:58.11 | tzanger | er :-) |
| 14:58.21 | lathos42 | astoria: As far as I know.. There was another list that I found before that i'll see if I can find again |
| 14:58.22 | astoria | tzanger: i'm not sure kannel does TAP though. |
| 14:58.22 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 14:58.33 | tzanger | yeah but I know TAP (kind of) |
| 14:58.38 | tzanger | ooh |
| 14:58.40 | tzanger | that's not 1200 baud |
| 14:58.48 | tzanger | that is clearly a 2400 baud tone |
| 14:59.04 | tzanger | I wonder if that would work |
| 14:59.07 | tzanger | I need a fucking modem now :-) |
| 14:59.21 | MrChimpy | as opposed to a frigid one? |
| 14:59.24 | lathos42 | Here's another one I found.. http://www.notepage.net/tap-phone-numbers.htm |
| 14:59.27 | tzanger | heh |
| 14:59.31 | *** join/#asterisk brettnem (~Brett@207.90.232.34) |
| 14:59.39 | tzanger | tha'ts out in BC though I'm sure they have an ontario # |
| 14:59.43 | astoria | kannel has a mailing list, like -users that you should sign up for |
| 14:59.51 | brettnem | good morning all.. or whatever it is.. |
| 15:00.02 | tzanger | hahah I was right too |
| 15:00.07 | tzanger | this list is newer and it's 2400 baud |
| 15:00.24 | brettnem | I think we need a fidonet board |
| 15:00.25 | lathos42 | the notepage.net was the one I used when I was setting up our Nagios Paging system |
| 15:00.26 | tzanger | kitchener |
| 15:00.29 | tzanger | they have a local # |
| 15:00.30 | DarthClue | lathos42: did you manage to get hold of your boss? |
| 15:00.31 | tzanger | for TAP anyway |
| 15:00.34 | jake1932 | can anyone tell me what the values are that fxotune.conf shows? |
| 15:00.56 | tzanger | brettnem: I was 1:221/10something.77 :-) |
| 15:01.05 | astoria | DarthClue: how many people are going to be at cluecon? |
| 15:01.10 | *** join/#asterisk RoyK (~roy@217.16.209.122) |
| 15:01.28 | brettnem | tzanger: heh, I absolutely can't remember what mine was.. so sad.. |
| 15:01.35 | astoria | tzanger: just to confirm, kannel does not support TAP. |
| 15:01.37 | DarthClue | astoria: we don't have final numbers yet, we won't actually know until the day of on quite a few of them. |
| 15:01.57 | astoria | DarthClue: okay, i was just wondering. |
| 15:02.04 | lathos42 | DarthClue: Yeah, and I registered last night.. we just need to hammer out the final details today and i'll be able to get you guys payment |
| 15:02.07 | tzanger | astoria: I understand |
| 15:02.10 | tzanger | TAP is really simple though |
| 15:02.21 | astoria | tzanger: it is? |
| 15:02.28 | astoria | tzanger: is there a good doc site on tap? |
| 15:02.32 | tzanger | astoria: at least it was when I was screwing with it years ago :-) |
| 15:03.34 | lathos42 | Sendpage has worked pretty well for me to be able to do TAP and SNPP |
| 15:03.51 | tzanger | sendpage eh? |
| 15:04.22 | astoria | I'm going to make a fax-sms gateway. |
| 15:04.54 | jake1932 | astoria: using a tiff to ascii converter? |
| 15:05.20 | lathos42 | That harkens back to the ascii porn days |
| 15:05.29 | *** part/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 15:05.40 | *** join/#asterisk gtigene (~chatzilla@70.89.216.41) |
| 15:05.48 | astoria | jake1932: maybe. I'm not going to give away my secrets. |
| 15:05.57 | gtigene | Is there a chat room for Asterisk Management Portal? |
| 15:06.05 | jake1932 | hehe |
| 15:06.25 | jake1932 | amportal |
| 15:06.37 | gtigene | jake1932: thanks |
| 15:09.48 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 15:09.55 | brenda | DarthClue: When is the last day I can sign up for cluecon? |
| 15:10.33 | *** join/#asterisk inv_arp (junya@adsl-156-141-145.mia.bellsouth.net) |
| 15:11.01 | DarthClue | brenda: if you need a place to stay, we need to know now. if you just want to crash the party (with cash in hand of course) then just show up. of course, we may have to charge more for the food then, but it all works out. |
| 15:11.36 | JerJer | more shameless self promotion i see |
| 15:11.48 | ManxPower | JerJer: it will be over with soon. |
| 15:11.57 | bkw_ | no |
| 15:12.01 | bkw_ | its not self promotion |
| 15:12.03 | anthm | wtf, she asked him |
| 15:12.07 | brenda | That wasn't self promotion... I asked |
| 15:12.08 | bkw_ | brenda, doesn't work for us |
| 15:12.11 | *** join/#asterisk hhh_ (Gone@i-195-137-50-162.freedom2surf.net) |
| 15:12.13 | DarthClue | JerJer: i'm going to be as polite as i can. I am a fucking bot and I just respond to peoples questions. If you don't like it, then put me on ignore. |
| 15:12.14 | ManxPower | Or we could all just promite our own businesses on the channel as well. |
| 15:12.19 | hhh_ | hi |
| 15:12.26 | tzanger | yup telus definitely uses TAP |
| 15:12.28 | bkw_ | ManxPower, go for it |
| 15:12.33 | DarthClue | good morning hhh_ |
| 15:12.35 | brenda | ManxPower: nothing wrong with that |
| 15:12.36 | tzanger | http://www.telusmobility.com/pdf/tap_v1p8.pdf |
| 15:12.42 | JerJer | this is not #asterisk-biz |
| 15:12.51 | hhh_ | does anyone here use ibs billing with gnugk ? |
| 15:12.52 | bkw_ | I see people pick off customers in here all the time for consulting jobs |
| 15:12.56 | fearnor | jerjer laying down the LAW |
| 15:13.01 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 15:13.10 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 15:13.11 | JerJer | that conf has very little to deal with Asterisk so its off topic at best |
| 15:13.12 | brenda | JerJer: this isn't #asterisk-nobiz |
| 15:13.17 | bkw_ | very little |
| 15:13.19 | lathos42 | DarthClue: I like a bot that isnt afraid to drop an f-bomb :) |
| 15:13.20 | bkw_ | more than 1 day is all about asterisk |
| 15:13.27 | fearnor | and tell me, jerjer, you never picked up a customar or two over at #asterisk or mentioned existance of noophone and its services? |
| 15:13.45 | fearnor | stop hatin', start participatin' |
| 15:13.47 | CoaxD | fearnor: In his defense, i've never seen it |
| 15:13.58 | JerJer | i have never told someone to use my company when someone asks "who's a good VoIP provider" |
| 15:13.58 | bkw_ | CoaxD, I have.. |
| 15:14.12 | mishehu | bah. |
| 15:14.18 | CoaxD | bkw: Hmm. k |
| 15:14.19 | anthm | well that's daft |
| 15:14.32 | tzanger | me either |
| 15:14.33 | brenda | JerJer: we can't all live by YOUR moral values |
| 15:14.39 | tzanger | I personally have recommended them but I've never seen jerjer do it |
| 15:14.41 | DarthClue | JerJer: then you really do need to re-evaluate your priorities. There is nothing wrong with pointing out that company z is a good company especially when i spend upwards of 12 hours everyday on the damn service. |
| 15:14.50 | CoaxD | <PROTECTED> |
| 15:15.04 | tzanger | DarthClue: no, but do you not think he should also disclose that he owns nufone if he's plugging it? |
| 15:15.11 | tzanger | I don't own it, I just use their service and I recommend it |
| 15:15.25 | tzanger | CoaxD: dammit where's my check??! |
| 15:15.39 | CoaxD | oops. my fingers slipped. |
| 15:15.46 | DarthClue | tzanger: i never hide the fact that i work for the company i promote. hell, most of them already know it. and no, i don't get a damn commission on it either. |
| 15:15.53 | tzanger | :-) |
| 15:15.56 | JerJer | bullshit |
| 15:16.04 | JerJer | pure microwaved bullshit |
| 15:16.05 | tzanger | I promote asterlink too, they both work very well for me |
| 15:16.18 | tzanger | microwaved bullshit? I don't think the gf'd appreciate that in the microwave |
| 15:16.19 | *** join/#asterisk Exstatica (exstatica@65.119.22.200) |
| 15:16.27 | fearnor | microwaved voice is so 1970s |
| 15:16.47 | Qwell | bkw_: Would you mind a quick msg? |
| 15:17.01 | FunXioN | lol |
| 15:17.03 | mishehu | JerJer: why do you think it's wrong that you not pitch your own company? |
| 15:17.05 | bkw_ | Qwell, shoot |
| 15:17.18 | JerJer | mishehu: this is not the place for it |
| 15:17.20 | JerJer | this is #asterisk |
| 15:17.20 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 15:17.24 | yaaar | word |
| 15:17.33 | *** join/#asterisk dasenjo (~dasenjo@208.195.214.7) |
| 15:17.36 | DarthClue | sentence |
| 15:17.46 | Ayano | paragraph? |
| 15:17.47 | yaaar | why do people always say that? |
| 15:17.47 | brenda | that's only a word |
| 15:17.50 | anthm | isn't the topic here that brenda asked when she can register and DarthClue answered and was accused of self-promotion ? |
| 15:17.53 | mishehu | JerJer: is there a place for it? I'm personally not familiar with any sort of forum, channel, or wiki about that type of stuff. |
| 15:17.57 | yaaar | brenda: exactly. |
| 15:18.11 | JerJer | anthm: she could look at the website |
| 15:18.13 | JerJer | or ask privately |
| 15:18.16 | DarthClue | JerJer: if someone ASKS then i will gladly answer them. if you think it's self promotion, then you must have failed more than just your computer science class. |
| 15:18.50 | JerJer | you always make a point to make sure you mention the name of that con |
| 15:19.01 | JerJer | yet brenda knows the name |
| 15:19.07 | fearnor | omfg, jerjer, set up a kickban on the word cluecon then |
| 15:19.08 | brenda | JerJer: It's a communication medium... I thought others would like to know since it is asterisk related. So I opened the question up for everyone. We can't all live by YOUR values. |
| 15:19.28 | bkw_ | well my finaly word is i'm shutting up about all this.. I have work to do... |
| 15:19.34 | *** join/#asterisk Turulo (~weed@125.Red-83-54-157.pooles.rima-tde.net) |
| 15:19.46 | brettnem | I actually come to the channel to listen to this bullshit |
| 15:20.01 | brettnem | :-D |
| 15:20.23 | brettnem | hey. with 288 people, you'll have a couple of soapboxes |
| 15:20.23 | tzanger | ok so telus uses TAP but there's nothing specific in the TAP protocol for MWI or voicemail callback |
| 15:20.32 | brenda | Personally... I don't like seeing the work 'fuck' in here all the time, cause I don't think it's Asterisk related. But I don't expect people live by my values. |
| 15:20.43 | fearnor | fuck the fucking fuckers, brenda. |
| 15:20.54 | brettnem | brenda: I seem to recall that there is a different asterisk channel reserved for that word |
| 15:21.00 | tzanger | app_fuck would be very popular I think. :-) |
| 15:21.09 | brenda | brettnem: #asterisk-fuck |
| 15:21.10 | lathos42 | tzanger: I had a feeling that TAP wouldnt be the answer to what you were looking for, but it was worth a look :) |
| 15:21.12 | tzanger | along with res_gf :-) |
| 15:21.12 | brettnem | hmm.. or is that a channel? |
| 15:21.34 | tzanger | lathos42: yeah... I'll just email my telus rep, they seem pretty keen on helping |
| 15:22.06 | *** join/#asterisk marv[work] (~timr@border0hsv.asterisksgi.com) |
| 15:22.19 | lathos42 | tzanger: Yeah, if you give them that much business, i'm sure they'd probably be willing to setup some sort of interface into their network for you |
| 15:22.50 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 15:23.00 | brettnem | hey so anyone know why sip channels hang in asterisk?? I do a 'sip show channels' and the list just keeps growing. until it hits about 500 dead (unknown) channels and the system reaches it's max open files and croaks.. all the unknown channels have the IP address of a single sipura.. about 450 channels of it. |
| 15:23.16 | fearnor | well, you just said your answer, brett |
| 15:23.19 | fearnor | broken sipoora |
| 15:23.33 | fearnor | dont blame asstricks, blame sipura |
| 15:23.39 | fearnor | or the broken firewall in front of it |
| 15:23.41 | brettnem | yeah, that's kinda what I thought.. but still, asterisk shouldn't croak because of a broken UA |
| 15:23.58 | gordonjcp | brettnem: you said it yourself |
| 15:24.02 | gordonjcp | it *isn't* |
| 15:24.14 | brettnem | it should relinquish those file descriptors... really |
| 15:24.20 | brettnem | I said why myself? |
| 15:24.36 | tzanger | http://about.telus.com/publicpolicy/pdf/id-0016.pdf |
| 15:24.37 | tzanger | ooh ooh |
| 15:24.39 | fearnor | yes, asterisk is *also* may be broken, something somewhere should finally time out and kill out those channels |
| 15:24.50 | brettnem | It's very apparent that the sipura may be broken, but how do I keep asterisk from dying because of it.. |
| 15:24.53 | tzanger | "network portability access service / universal voice messaging service" |
| 15:25.13 | dudes | How long does it take to get g729 licenses? |
| 15:25.24 | brettnem | I think you can get this right on the digium website |
| 15:25.33 | fearnor | thats not what he aksed |
| 15:25.35 | *** join/#asterisk santiago (~santiago@63.245.86.175) |
| 15:25.38 | tzanger | dudes: usually a day or two |
| 15:25.38 | mishehu | dudes: usually < 24 hours |
| 15:25.45 | dudes | We ordered 120 yesterday |
| 15:25.47 | fearnor | he aksed how long it takes for digium sales team to get off their butt and process |
| 15:25.57 | fearnor | which varies from 1 hour to a week. :P |
| 15:26.02 | lathos42 | tzanger: That doc looks interesting |
| 15:26.05 | brettnem | oh the mood in here is ripe today eh? |
| 15:26.23 | mishehu | brettnem: we just need a few bfg9000's and we'd all be set |
| 15:26.55 | mishehu | the anomosity in the air can be cut with a chain saw. |
| 15:27.07 | brettnem | and it should be... |
| 15:27.30 | lathos42 | Remember Kids, Dont Poke the Bear |
| 15:28.35 | brettnem | I don't have to outrun the bear.. I just have to outrun you |
| 15:29.00 | tzanger | that movie is awesome |
| 15:29.04 | tzanger | "does a bear shit in the woods?" |
| 15:29.04 | tzanger | hahahahah |
| 15:30.17 | dudes | so does digium work faster if you call them and ask them about these things? |
| 15:30.28 | brettnem | haha |
| 15:30.37 | mishehu | "sometimes you eat the bar... and well, sometimes the bar, well he's eats you..." |
| 15:30.42 | ManxPower | dudes: most of the time it's 24 hours |
| 15:30.54 | mishehu | dudes: sure, I suggest a heavy regiment of calling them every 30 minutes. |
| 15:30.54 | blitzrage | In Soviet Russia, the beer drinks you! |
| 15:31.09 | dudes | mishehu - that could work |
| 15:31.11 | mishehu | because people always get much more work done when somebody is calling them every few minutes |
| 15:31.27 | mishehu | blitzrage: that's vodka |
| 15:31.34 | blitzrage | well, I drink beer |
| 15:31.48 | mishehu | blitzrage: yeah, but russians drink vodka like water last I knew. ;-) |
| 15:31.49 | dudes | I start on beer and finish up with some wiskey |
| 15:31.56 | lathos42 | dudes: You may also want to call and ask the person who answers the phone if they accept "Monetary Motivation" |
| 15:31.59 | dudes | whiskey rather |
| 15:32.01 | mishehu | I skip the beer. can't stand it. |
| 15:32.01 | brettnem | you know what they say.. the squeaky wheel gets replaced |
| 15:32.26 | brettnem | haha.. that sounds like a "bribe".. Of course, digium does everything for free |
| 15:32.26 | blitzrage | the squeaky wheel gets shot |
| 15:32.52 | mishehu | they do? |
| 15:33.07 | brettnem | no.. they accept monetary contributions in the form of bribes |
| 15:33.10 | mishehu | then why on earth did I pay $600 for a te110p card??? |
| 15:33.19 | brettnem | that was a donation |
| 15:33.22 | lathos42 | mishehu: You dont know the secret handshake |
| 15:33.28 | brettnem | you have wonderful karma |
| 15:34.04 | dudes | nothing is free |
| 15:34.20 | dudes | digium just sent them ... holy crap |
| 15:34.32 | lathos42 | dudes: Ask and you shall receive |
| 15:34.33 | *** join/#asterisk startled (startled@d220-238-92-14.dsl.vic.optusnet.com.au) |
| 15:34.38 | ManxPower | mishehu: Especially when you can get it for $499 direct from Digium |
| 15:35.26 | lathos42 | dudes: I had that happen yesterday when I was waiting for a license for Pro/Engineer Wildfire .. As soon as I said something to someone about it, it showed up in my inbox |
| 15:36.12 | startled | hey guys, just staring to have a play and got my X100 on its way to connect things up. Can you clear something up for me. If I bought a E1 card, had an E1 coming into the building... I'd essentially have 30 lines running out of the asterisk box yeah? 2 E1 and a dual-port card is 60 lines, etc? |
| 15:36.35 | SwK[Work] | startled: thats 60 trunks |
| 15:36.51 | SwK[Work] | now how you have it configured that could handle a truck load of DIDs |
| 15:37.00 | *** join/#asterisk jimmybob46 (~jim@81.5.154.235) |
| 15:37.03 | SwK[Work] | ()depending on how that is) |
| 15:37.48 | jimmybob46 | well, 7 days work and research, and i can say that capi rules.. |
| 15:38.05 | shido | 2 E1 trunks |
| 15:38.08 | shido | 60 voice channels |
| 15:38.08 | startled | DIDs? (hits the wiki) |
| 15:38.29 | mishehu | ManxPower: oh whatever the price was, I was just tossing out a rough number. |
| 15:38.36 | startled | cool, thats what I thought |
| 15:38.50 | jimmybob46 | hoooray, finally got it working.. good feeling |
| 15:38.56 | mishehu | lathos42: is this the Free Mason Digium Secret Society? |
| 15:39.02 | *** join/#asterisk fugitivo (~ajf@201.255.99.157) |
| 15:39.03 | fugitivo | hello |
| 15:39.08 | *** join/#asterisk doughecka_ (~Miranda@doughecka.user) |
| 15:39.38 | shido | No, this isnt the free masons, this is th Illuminati Asterisk Group of Elders. |
| 15:39.45 | InfraRed | <PROTECTED> |
| 15:39.47 | jimmybob46 | can anyone answer a capi question for me~? |
| 15:39.57 | SwK[Work] | startled: DID is just a phone number when used in conjuction with a PBX like asterisk and a E1-PRI you can only have 30 calls active at the same time but a phone number isnt tied to a specific channel so you can have 200 DIDs on one E1 |
| 15:39.58 | InfraRed | failed to authenticate on invite |
| 15:40.04 | *** join/#asterisk paski_fr (~paski_fr@stardust.noc.frontier.fr) |
| 15:41.23 | paski_fr | I have a problem with my waiting queue in realtime |
| 15:41.43 | mishehu | shido: well, I hope I get to be the head of the Asterisk of the Elders of Zion branch. We have some protocols we'd like to set up. (iax2, sip...) |
| 15:41.48 | doughecka_ | shido: rofl |
| 15:42.04 | tzanger | doughecka_: :-) |
| 15:42.08 | fearnor | i have a boatload of zhone channel banks |
| 15:42.12 | tzanger | what's wrong with a pair of TDM04B? |
| 15:42.12 | fearnor | 8fxo/16fxs |
| 15:42.12 | shido | I think |
| 15:42.13 | paski_fr | when I try to call my waiting queue, the available agent who answers doen't heard me |
| 15:42.17 | shido | you can get them from jsharp |
| 15:42.20 | fearnor | just use them for fxo ;) |
| 15:42.21 | doughecka_ | tzanger: I had issues |
| 15:42.31 | tzanger | doughecka_: the FXO issues on TDM have apparently been *resolved* |
| 15:42.31 | doughecka_ | zhone? |
| 15:42.37 | tzanger | it was a simple sign error |
| 15:42.48 | doughecka_ | tzanger: when? |
| 15:42.51 | tzanger | (if your issue was "it seems to just quit after 25 days or so) |
| 15:42.56 | doughecka_ | noo |
| 15:42.56 | tzanger | doughecka_: on Monday |
| 15:43.08 | doughecka_ | my issue is pops, clicks, echo, and lost calls |
| 15:43.14 | brenda | <PROTECTED> |
| 15:43.14 | brenda | <PROTECTED> |
| 15:43.14 | brenda | Shameless self promotion |
| 15:43.17 | tzanger | doughecka_: hmm that is unusual |
| 15:43.24 | doughecka_ | very |
| 15:43.27 | doughecka_ | its the server me thinks |
| 15:43.33 | Beirdo | brenda: looks fair to me |
| 15:43.35 | Beirdo | ;) |
| 15:43.46 | brenda | Beirdo: I agree |
| 15:43.47 | doughecka_ | jsharp? |
| 15:43.51 | doughecka_ | whats his company? |
| 15:44.05 | Beirdo | doughecka_: JerJer's? |
| 15:44.27 | doughecka_ | whats the zhone channel banks? |
| 15:44.27 | mishehu | jerjer -> nufone. |
| 15:44.31 | Beirdo | aye |
| 15:44.31 | paski_fr | Is someone would like to help me about my waiting queue problem ? |
| 15:44.33 | lathos42 | First Rule of Asterisk Club, dont talk about Asterisk Club |
| 15:44.35 | doughecka_ | yea, whos jsharp? |
| 15:44.47 | mishehu | lathos42: remember, rule #1 of Asterisk Club - nobody talks about Asterisk Club... |
| 15:44.49 | Beirdo | JerJer's saving me bundles of money :) |
| 15:44.57 | mishehu | gah |
| 15:45.02 | mishehu | I can't type fast enough |
| 15:45.05 | bkw_ | mishehu, haha |
| 15:45.37 | nDuff | paski_fr, have you described your problem yet? If so, and you haven't gotten a response, then it's likely there's just nobody awake right now who knows the answer. On the other hand, if you haven't described your problem, it should be obvious why you're getting no response. |
| 15:45.47 | fearnor | doug: its a channel bank. |
| 15:45.56 | Turulo | does anyone know, if echo cancellation on zap can be disabled for some sip phones and enable it for others? |
| 15:45.59 | fearnor | it has T1 in, 24*voice out |
| 15:46.07 | fearnor | zhone is maker. |
| 15:46.15 | doughecka_ | ah, found it |
| 15:46.27 | doughecka_ | do they have FXO models? |
| 15:46.55 | fearnor | they have 16fxs/8fxo |
| 15:46.57 | fearnor | thats what i have. |
| 15:47.03 | fearnor | cheap as dirt, 100$ ;) |
| 15:47.07 | fearnor | about just as good though. |
| 15:47.47 | doughecka_ | hoyl crap |
| 15:47.48 | tzanger | Turulo: there's no need for echo cancel on SIP phones... it's a 4-wire network |
| 15:47.55 | doughecka_ | what if I bought it new? |
| 15:47.59 | tzanger | 16FXS/8FXO for $100??!! |
| 15:48.07 | fearnor | have been discontinued ahwile ago |
| 15:48.09 | tzanger | where?? |
| 15:48.09 | Turulo | tzanger i mean disable it on zap channel |
| 15:48.11 | doughecka_ | CRAP |
| 15:48.15 | doughecka_ | discontiued?! |
| 15:48.22 | fearnor | there are always sellers on ebay selling for 100$ or so |
| 15:48.23 | tzanger | Turulo: sure -- send a T.8 tone (I think that's the name) |
| 15:48.26 | Turulo | tzanger depending on the sip client, is a fax or a phone |
| 15:48.32 | tzanger | fearnor: I have never seen an FXO channel bank for that price |
| 15:48.36 | tzanger | FXS sure |
| 15:48.43 | tzanger | 24FXS can be had for under $150 if you're frugal |
| 15:48.45 | fearnor | standard config is 16fxs/8fxo |
| 15:48.52 | tzanger | but throw an FXO module in there and you're over $200 usually |
| 15:49.04 | fearnor | anyway, i have two i can sell :) |
| 15:49.09 | tzanger | nice |
| 15:49.10 | fearnor | at least two. |
| 15:49.15 | Turulo | tzanger there is no way using pass trough, instead of 7.38 ? |
| 15:49.18 | RoyKilt | ~seen inspired |
| 15:49.19 | jbot | inspired <mikael@213.197.167.61> was last seen on IRC in channel #asterisk, 54d 2h 23m 58s ago, saying: 'if it's made at the wood alcohol factory, you'll go blind ;)'. |
| 15:49.22 | RoyKilt | ~seen uninspired |
| 15:49.23 | jbot | RoyKilt: i haven't seen 'uninspired' |
| 15:49.24 | doughecka_ | fearnor: can you get one to me by tomorrow? :) |
| 15:49.28 | tzanger | I am channel-banked out at the moment (TR08 FXS and two Adit600s maxxed out) |
| 15:49.36 | Turulo | tzanger i meant T.38 |
| 15:49.39 | tzanger | Turulo: what are you talking about? |
| 15:49.43 | tzanger | T.38 is a fax thing |
| 15:49.44 | Beirdo | ~seen net-snmp that doesn't suck |
| 15:49.44 | jbot | i haven't seen 'net-snmp that doesn't suck', Beirdo |
| 15:49.48 | Beirdo | hehe |
| 15:50.08 | Turulo | tzanger, yes i want to disable echo cancell |
| 15:50.18 | Turulo | tzanger, for faxes |
| 15:50.19 | paski_fr | nDuff, I am going to explain you my problem. I'm trying to install a waiting queue in real time mode on my asterisk. Then, when I try to call my waiting queue, all phones of agents defined previously ring, but when I answer and speak, I don't heard anything. We can not talk together. |
| 15:50.22 | tzanger | Turulo: it is automatic |
| 15:50.25 | tzanger | it already does that |
| 15:50.27 | tzanger | alway shas |
| 15:50.29 | tzanger | you're overthinking |
| 15:50.41 | Turulo | tzanger but that is using faxdetection? |
| 15:50.45 | tzanger | either that or you decided to disable the tone detect in zaptel |
| 15:50.56 | tzanger | Turulo: you're overthinking |
| 15:51.08 | Turulo | tzanger tone detect? |
| 15:51.11 | tzanger | zaptel echo cancel automaticlaly turns off the echo can if it detects a tone |
| 15:51.18 | tzanger | anyway I ogtta get to lunch, I'll bbl |
| 15:51.26 | Turulo | ok |
| 15:51.28 | Turulo | thz |
| 15:52.05 | gtigene | Our phone company says there are framing errors on our PRI, average about one per second. They say there is an audible "pop" when this occurs. We replaced the TE405P card and it didn't fix the errors. Has anyone had this problem or have any suggestions? |
| 15:52.52 | *** part/#asterisk Turulo (~weed@125.Red-83-54-157.pooles.rima-tde.net) |
| 15:52.53 | _DAW | gtigene - has you telco done any intrusive testing on the circuit? |
| 15:53.21 | _DAW | ie looping back their smartjack to make sure the circuit is clean |
| 15:53.35 | ManxPower | <PROTECTED> |
| 15:53.41 | gtigene | _DAW: They say they determined that the circuit was clean. |
| 15:53.51 | *** join/#asterisk junbug (junya@adsl-11-73-177.mia.bellsouth.net) |
| 15:54.07 | ManxPower | gtigene: do you get HDLC errors on the Asterisk console? |
| 15:54.08 | Ayano | where is the list of sip providers on the wiki? I can't find it for some reason |
| 15:54.20 | paski_fr | <PROTECTED> |
| 15:54.43 | gtigene | ManxPower: Yes, more than once an hour I get them. |
| 15:54.52 | ManxPower | that's an indication of a problem. |
| 15:55.05 | ManxPower | Make sure you have done the IDE tuning |
| 15:55.09 | gtigene | ManxPower: Bad FCS |
| 15:55.44 | gtigene | ManxPower: The only thing I have done with IDE is setting UDMA 2. What is IDE tuning? |
| 15:55.56 | ManxPower | any HDLC error is a data corruption problem. Sometimes caused by a bad line, more often caused by the IDE controller or running graphics |
| 15:56.05 | ManxPower | gtigene: unmasq IRQ as well. |
| 15:57.00 | gtigene | ManxPower: where do you unmasq IRQ. I know what an IRQ is but not know about IRQ masking. |
| 15:57.50 | *** join/#asterisk paski_fr (~paski_fr@stardust.noc.frontier.fr) |
| 15:57.52 | brettnem | I think ManxPower is refering to a hdparam setting |
| 15:58.00 | paski_fr | <PROTECTED> |
| 15:58.14 | brettnem | also, you can use zttool to check interrupt slips |
| 15:58.40 | brettnem | make sure you arn't sharing irqs with anything.. |
| 15:59.15 | brettnem | I've even seen the wrong PRI protocol cause those kinds of problems (really) |
| 15:59.38 | shido | "/sbin/hdparm --help" |
| 15:59.41 | shido | should help you |
| 15:59.47 | shido | I had to enable UDMA in ubuntu |
| 15:59.51 | *** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net) |
| 15:59.59 | ChkDigit | Does anybody know what causes: Ouch, part reset, quickly restoring reality (2) with a TDM400P? |
| 16:00.02 | Hmmhesays | so I'm sitting on the phone, on a call that sounds like it is in a wind tunnel |
| 16:00.19 | astoria | Hmmhesays: get out of the wind tunnel :) |
| 16:00.22 | shido | turn off the vacuum blowing up your.. |
| 16:00.36 | Hmmhesays | shido: where is that lint |
| 16:00.48 | Hmmhesays | it was not delivered as expected |
| 16:00.54 | shido | good thing it was insured |
| 16:01.07 | lathos42 | Hmmhesays: Well, when i'm on a call, I usually sit on my chair, not on the phone itself |
| 16:01.18 | gtigene | brettnem: Thanks. |
| 16:01.23 | Hmmhesays | hrm, good point, let me try that lathos42 |
| 16:01.37 | *** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net) |
| 16:01.55 | Hmmhesays | I want my 7 pence |
| 16:02.21 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 16:04.52 | shido | . |
| 16:05.02 | mut | o_O |
| 16:05.04 | Katty | i would like someone to explain to me white it takes 7 pages of information to purchase a single hard drive. |
| 16:05.15 | shido | customs? |
| 16:05.19 | mut | uh |
| 16:05.23 | mut | first page, select product |
| 16:05.27 | DarthClue | sure, customs. |
| 16:05.27 | Katty | ... |
| 16:05.28 | mut | 2nd page enter username/password |
| 16:05.32 | [Jedi] | a Pentium II with SCSI disks and 256mbytes of RAM could serve as a SIP "proxy"/"registar" without doing any transcoding ? |
| 16:05.33 | Katty | mut: not on the website |
| 16:05.45 | [Jedi] | for 20-30 concurrent users |
| 16:05.45 | JunK-U | Katty: when u buy a car u want details? we're like that :) |
| 16:05.51 | Katty | For everyone who does not understand female psychology.... |
| 16:05.55 | shido | yeah |
| 16:05.56 | Katty | This Is A Rant Moment |
| 16:05.57 | shido | sounds good |
| 16:05.57 | Katty | kthxbi |
| 16:06.06 | shido | talk about bare minimums |
| 16:06.06 | mut | oh |
| 16:06.09 | mut | k.. |
| 16:06.12 | Katty | this is not a solutions explination moment! |
| 16:06.13 | Hmmhesays | hey Katty |
| 16:06.14 | shido | Jedi |
| 16:06.20 | mut | cause i think i could only have done 6 pages |
| 16:06.31 | [Jedi] | shido, do you think such a machine could perform that task? |
| 16:06.31 | Katty | DarthClue: :> |
| 16:06.35 | shido | yes |
| 16:06.45 | [Jedi] | great then |
| 16:06.59 | [Jedi] | how could I make my asterisk forbid any kind of transcoding? |
| 16:07.13 | Hmmhesays | set the codec per user |
| 16:07.18 | Katty | DarthClue: we don't speak about my mother :< |
| 16:07.23 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:07.23 | shido | sip.conf set a codec and disallow others |
| 16:07.24 | shido | so |
| 16:07.25 | Katty | DarthClue: she is a Jehovah Witness |
| 16:07.26 | shido | disallow=all |
| 16:07.27 | Hmmhesays | I haven't talked to my mother in like 17 years |
| 16:07.29 | shido | allow=ulaw |
| 16:07.34 | Katty | DarthClue: you don't realize how many psychological issues i really have, do you :P |
| 16:07.43 | *** join/#asterisk moy (~kvirc@201.135.113.46) |
| 16:08.00 | [Jedi] | hmmm well ok |
| 16:08.04 | lathos42 | Whenever I see the word therapist i'm reminded of the SNL skit with Sean Connery playing Jeopardy.. "I'll take The Rapist for $100 Alex" |
| 16:08.10 | astoria | Ha ha. |
| 16:08.17 | Hmmhesays | anal bum cover alex |
| 16:08.38 | DarthClue | Katty: i used psychology when i was in 8th grade to make the teacher leave me alone. it was priceless when i told her she was a paranoid schitzo with dulusional tendencies...too bad it was true. |
| 16:08.43 | Hmmhesays | "I believe that is "an album cover" |
| 16:09.05 | Hmmhesays | "a penis mightier!" |
| 16:09.10 | Hmmhesays | i love that snl skit |
| 16:09.19 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 16:09.54 | *** join/#asterisk creative83 (~creative@adsl-62-167-221-90.adslplus.ch) |
| 16:09.55 | Hmmhesays | Alex: you drink whater for this, Sean Connery: a leather glove! |
| 16:09.57 | creative83 | re |
| 16:10.04 | Hmmhesays | holy crap did I just type that |
| 16:10.13 | Hmmhesays | *you drink water from this* |
| 16:10.18 | creative83 | Anyone with a hfc card and a swisscom ntba in here? |
| 16:10.30 | lathos42 | :) |
| 16:10.40 | Blackthorn | Hello. I have a sipura spa-2000 unit connected to the asterisk box over a wireless link. The spa has two lines and is setup with qualifty. One line went down and marked unrechable. I reset *, no change. reset the sap. and it came back up. |
| 16:11.01 | ManxPower | It's behind nat then |
| 16:11.05 | ManxPower | qualify=yes |
| 16:11.19 | Blackthorn | line one shows port 5060 line two shows port 1024. What can I look for to see what the problem is? yes MP this is the same unit we were working with the other day |
| 16:11.26 | JerJer | brenda: where does that simple statement show my company name? |
| 16:11.26 | Hmmhesays | its my life to live my way, so I'll keep day dreaming away |
| 16:11.32 | Blackthorn | worked great untill just a few minutes ago. and it's setup just like you suggested the other day. |
| 16:11.54 | [Jedi] | what's qualify for? |
| 16:11.59 | brettnem | Blackthorn: yeah, all those problems are typicall of NAT issues.. makes sure qualify is on |
| 16:12.01 | JerJer | brenda: and next time you see Jsaunders ask him who i referred him to |
| 16:12.04 | Blackthorn | thers no port fowarding anymore. |
| 16:12.06 | JerJer | in private |
| 16:12.18 | brettnem | [Jedi]: qualify sends keepalives which keeps the ports open |
| 16:12.50 | [Jedi] | so for using SIP in NAT'ed environments, you need qualify=yes and nat=yes ? |
| 16:12.58 | [Jedi] | I was only using nat=yes |
| 16:12.59 | brettnem | it is necessary for nat.. nat=yes should always be followed by a qualify=yes. |
| 16:13.06 | Blackthorn | quilifty is on for the device. And i was monitoring the keepalive over the last several days. and that was working fine. just that * said it was unreachable but only for the 2nd line on the ata |
| 16:13.25 | brettnem | Blackthorn: is qualify=yes set? |
| 16:13.33 | creative83 | I have the problem, that my hfc card with an NTBA on it gives no dial signal to the ISDN telephone. The isdn phone diplays an "Installation Error" |
| 16:13.36 | paski_fr | <PROTECTED> |
| 16:13.40 | Blackthorn | jedi: on the spa-2000 you should have nat = no and asterisk sip config qualfity = yes |
| 16:13.44 | creative83 | I use zaphfc on NT moder. |
| 16:14.12 | MicC_ | tzanger: you were right, I need OC-12 and one hell of a media gateway to handle 4000 lines |
| 16:14.13 | MicC_ | :P |
| 16:14.19 | brettnem | sipuras also need the nat and keepalive stuff turned on on the actual ata itself.. |
| 16:14.22 | [Jedi] | what does exactly "nat=yes" do? |
| 16:14.35 | Blackthorn | jedi and no port fowards or you break the rules.. |
| 16:14.51 | brettnem | I think it enables use of rport and ignores the ip address in the contact header and uses the actual source ip instead |
| 16:14.58 | [Jedi] | I wanted to avoid modifying the ADSL router configuration for port forwarding and that |
| 16:15.08 | brettnem | you shouldn't have to do port forwards.. |
| 16:15.46 | Blackthorn | I WAS doing port fowards... and nat=yes. now i do no port fowards.. nat = no. and qualfity=yes in the sip.conf file. |
| 16:16.01 | brettnem | that sounds backwards to me |
| 16:16.04 | Blackthorn | And when you do a sip show peers you get the ip address of the router, which I like. |
| 16:16.18 | Blackthorn | qualifty = yes tells * to do the nat translations. |
| 16:16.24 | brettnem | if you turn the port forwards off. you should have nat=yes and qualify=yes.. or maybe I'm on crack |
| 16:16.28 | brettnem | nooooo |
| 16:16.33 | JerJer | Blackthorn: um no |
| 16:16.37 | brettnem | qualify=yes just sends OPTIONS to the device |
| 16:16.56 | brettnem | qualify.. strictly speaking has absolutely nothing to do with NAT |
| 16:17.02 | JerJer | Blackthorn: registering to the proxy is how you get thru nat/firewall |
| 16:17.03 | Saaib | what configuration change i need to do in order that asterisk won't start mpg123 with the on-hold backgrouns music ? |
| 16:17.16 | jimmybob46 | hello! can anyone help me with a capi question? |
| 16:17.30 | brettnem | Saaib: what version of mpg123 are you running? |
| 16:17.42 | brettnem | oh.. you don't want it.. hmm |
| 16:17.49 | *** join/#asterisk cripito (~ncripito@67.154.143.190) |
| 16:17.51 | cripito | hi |
| 16:18.07 | brettnem | just remove your audio files from your moh dir |
| 16:18.10 | cripito | anyone have a res_mysql_config.so for astlinux? |
| 16:18.28 | Saaib | brettnem: mpg123-0.59r |
| 16:18.35 | Blackthorn | well i donno then you got me all confused. but I am 100% that i have qualfity = on, nat = off, and no prot fowarding. and it's been working for week with no problems. untill one side of the ata droped off a few mins ago and needed to be rebooted. |
| 16:19.15 | Blackthorn | If i turn nat = on (on the ata) then it fails to registerer. |
| 16:19.18 | Blackthorn | register. |
| 16:19.31 | brettnem | Blackthorn: when you have the config you described, it will work until the translations disappear.. the length is determined by how long your nat device holds it's translations and how often you use the device.. |
| 16:19.38 | brettnem | btw, that's nat=yes |
| 16:20.04 | brettnem | you need to set nat=yes in sip.conf AND set the nat stuff on the ATA |
| 16:20.16 | ManxPower | no, turning nat on in the device will screw up nat=yes in asterisk |
| 16:20.19 | Saaib | brettnem: could exist another solution... i'm running a test asterisk server on my desktop, but once i start the console, mpg123 blocks the sound devs... |
| 16:20.36 | brettnem | ah |
| 16:20.55 | *** join/#asterisk drbrown_ (~chatzilla@63.238.117.40) |
| 16:20.56 | Saaib | so i want to use a softphone to test the connection , but with mpg123 running cant |
| 16:21.05 | brettnem | you might be able to noload res_musiconhold.so |
| 16:21.23 | Saaib | on modules.conf right ? |
| 16:21.30 | Blackthorn | In sip.conf. Sip=1 is that the same as sip=yes? |
| 16:21.34 | brettnem | or there might be a way to get mpg123 not hook the soundcard.. |
| 16:21.35 | brettnem | wait |
| 16:21.45 | ManxPower | Saaib: mv musiconhold.conf musiconhold.conf-siabled |
| 16:21.47 | brettnem | are you sure mpg123 is getting the sound dev and it's not chan_oss?? |
| 16:22.00 | ManxPower | Blackthorn: sip or nat? |
| 16:22.07 | Saaib | brettnem: not really, how can i figure out ? let me do an lsof |
| 16:22.39 | ManxPower | Saaib: in /etc/asterisk/modules.conf put noload => chan_alsa.so and noload => chan_oss.so |
| 16:22.52 | Blackthorn | sip |
| 16:23.03 | Blackthorn | errror. |
| 16:23.08 | ManxPower | there is no option sip=yes or sip=1 |
| 16:23.17 | Blackthorn | i mean nat |
| 16:23.19 | Blackthorn | lol |
| 16:23.32 | ManxPower | Blackthorn: Please put down the beer and step away from the computer. |
| 16:23.36 | Blackthorn | i have qualfity = yes and nat=1 in the sip.conf file |
| 16:23.37 | Saaib | ManxPower: thanks |
| 16:23.39 | brettnem | Saaib: my mpg123 only shows up on ttys, not on the sounddev |
| 16:23.43 | junbug | which version of mpg123 is recommend again? |
| 16:23.49 | doolph | anyone know how to have h323 client like sip client? |
| 16:23.54 | Saaib | brettnem: the result from lsof is |
| 16:23.55 | Saaib | asterisk 21763 root 16u CHR 14,3 5310226 /dev/dsp0 |
| 16:24.01 | ManxPower | Blackthorn: 1 should mean "yes" and should mean "true". I always use "yes" |
| 16:24.19 | brettnem | Saaib: right.. and that's probably because of chan_oss and chan_alsa and proobably hs nothing to do with mpg123 |
| 16:24.20 | ManxPower | Saaib: the noload will fix that |
| 16:24.27 | Blackthorn | haha, i could use a good beer at the moment. |
| 16:24.30 | brettnem | right it should |
| 16:24.44 | brettnem | but understand that it isn't a mpg123 thing.. at least, it doesn't appear that it is. |
| 16:24.49 | [Jedi] | I should add a 'noload =>' for any module I'm not using? |
| 16:24.52 | [Jedi] | or it doesn't matter? |
| 16:25.03 | brettnem | [Jedi]: don't fix it if it ain't broke |
| 16:25.31 | moy | anyone here is using Asterisk on Gentoo Distro? |
| 16:25.32 | ManxPower | [Jedi]: There are MANY modules interdependencies. |
| 16:25.33 | brettnem | junbug: just do a make mpg123 |
| 16:25.40 | [Jedi] | hmm |
| 16:25.55 | *** join/#asterisk citats (~james@duff.gnuinter.net) |
| 16:25.58 | ManxPower | Generally nothing depends on chan_* so I noload => chan_iax.so chan_mgcp.so, etc |
| 16:25.58 | doolph | anyone know how to have h323 client like sip client? |
| 16:26.02 | brettnem | [Jedi]: really.. don't start unloading stuff until you know what you don't need |
| 16:26.41 | ManxPower | But, in the example of res_musiconhold.so many modules require it to be loaded, even if you are not using it. |
| 16:27.04 | bkw_ | res_musiconhold.so in cvs-head doesn't have to be loaded anymore |
| 16:27.14 | brettnem | fancy |
| 16:27.16 | bkw_ | tony did a patch to stub the functions in the core so you can noload that if you wish |
| 16:27.16 | Saaib | ManxPower: brettnem: there ya go ! problem fixed, thanks ! |
| 16:27.27 | brettnem | excellent |
| 16:27.50 | Saaib | Now executing X-Lite to do my testing |
| 16:28.45 | [Jedi] | I was thinking on disabling chan_mgcp and chan_skinny |
| 16:28.48 | [TK]D-Fender | Simple question : I'm trying to find out where to setup SQL storage of *'s queue log data, and the WIKI has me running around. Can someone link me more directly with docs on this? |
| 16:29.11 | [Jedi] | I don't think mgcp has any real use for me |
| 16:29.11 | Blackthorn | ok to clairfy i have nat=yes, qualfity=yes in sip.conf. The ata has nat=no and no port fowarding on the router. One line was working the other was marked as unreachable. Reset the ata got both to work. |
| 16:30.16 | [Jedi] | brettnem: mgcp and skinny are good candidates for being unloaded? |
| 16:31.56 | *** part/#asterisk santiago (~santiago@63.245.86.175) |
| 16:32.04 | *** join/#asterisk pifiu (~myassisbi@208.205.181.170) |
| 16:33.24 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 16:33.29 | riemensc | hello |
| 16:35.10 | *** join/#asterisk Cooltalk (~io@202.161.138.3) |
| 16:36.23 | cripito | anyone have a res_mysql_config.so for astlinux? |
| 16:36.34 | doolph | anyone good with gnugk? |
| 16:36.53 | *** join/#asterisk w0w0 (~w0w0@14.Red-81-39-84.pooles.rima-tde.net) |
| 16:37.08 | [Jedi] | I'm good at hating gnugk |
| 16:37.48 | riemensc | what is the error code -- Got SIP response 488 "Not acceptable here" back from 213.61.187.157 |
| 16:38.05 | *** join/#asterisk darkskiez (~darkskiez@194.247.78.146) |
| 16:38.22 | bkw_ | is this a sipura |
| 16:38.51 | riemensc | no voipbuster.com |
| 16:39.01 | bkw_ | voipbuster uses IAX |
| 16:39.08 | riemensc | and sip |
| 16:39.16 | bkw_ | sniff it .. crack it.. whip it good! |
| 16:39.23 | bkw_ | or something like that :P |
| 16:39.23 | *** join/#asterisk file (~jcolp@mctnnbsah25-142166093154.nb.aliant.net) |
| 16:39.37 | brettnem | riemensc: typically that's a codec incompatibility |
| 16:41.22 | bkw_ | ya ya |
| 16:41.33 | riemensc | i use the codec alaw |
| 16:41.33 | bkw_ | just wanting to see if it happened to be a sipura |
| 16:41.38 | brettnem | riemensc: are you only allowing alaw? |
| 16:41.45 | bkw_ | allow ulaw |
| 16:42.02 | riemensc | yes i´m only allow alaw |
| 16:42.14 | brettnem | yeah, you'll need ulaw in there most likely |
| 16:42.15 | bkw_ | well get a sip debug |
| 16:42.19 | bkw_ | and see what it wants |
| 16:42.20 | riemensc | in the sip.conf i have write |
| 16:42.23 | riemensc | allow=alaw |
| 16:42.24 | bkw_ | the SDP will show you the way |
| 16:42.25 | brettnem | uh oh |
| 16:42.26 | riemensc | disallow=all |
| 16:42.34 | *** join/#asterisk Hengei (~Hengei@196.203.53.57) |
| 16:42.35 | brettnem | put an allow=ulaw in there.. |
| 16:42.36 | riemensc | what is the correct codec for voipbuster |
| 16:42.59 | brettnem | you'll have to ask them.. ** if you are happy with alaw.. PLEASE just put ulaw in there. |
| 16:43.11 | riemensc | ive put ulaw and testing |
| 16:43.17 | brettnem | sip reload |
| 16:43.37 | Hengei | hello ! how to uninstall asterisk |
| 16:43.44 | brettnem | rm -rf / |
| 16:43.50 | brettnem | no not really.. please don't do that |
| 16:43.54 | *** join/#asterisk dos000 (~dos000@CPE00119572fd49-CM00137186e53a.cpe.net.cable.rogers.com) |
| 16:44.34 | riemensc | i´ve started asterisk -vvvvc again |
| 16:44.56 | brettnem | ohhh.. running asterisk in forground mode.......... |
| 16:45.09 | riemensc | i´ve got the same error Got SIP response 488 "Not acceptable here" back from 213.61.187.157 |
| 16:45.29 | brettnem | well do a sip debug and find out what they want |
| 16:45.32 | paski_fr | <PROTECTED> |
| 16:45.54 | Hengei | rm -rf remove the directory but how to uninstall the application ? |
| 16:46.17 | riemensc | i´ve i put sip show registry |
| 16:46.20 | brettnem | Hengei: what are you trying to acomplish? |
| 16:46.24 | riemensc | i see registered |
| 16:46.35 | brettnem | riemensc: sip debug |
| 16:46.35 | brettnem | make call. |
| 16:46.47 | Hengei | i want ti uninstall asterisk |
| 16:46.54 | brettnem | oh.. I see |
| 16:47.17 | brettnem | well look in your makefile and see where it put stuff and go erase it..I don't think there is an uninstaller |
| 16:47.25 | brettnem | not for the "linux version" heh |
| 16:47.51 | brettnem | anyone want to share a lunch?? |
| 16:48.24 | riemensc | sip debugging enabled |
| 16:48.27 | brettnem | 292 greedy people without extra lunches |
| 16:48.37 | riemensc | i call a number and waiting |
| 16:48.55 | riemensc | can i post this sip debugging in this room |
| 16:48.55 | Hengei | i want de uninstall it just for reinstalling it ... i have to do some things from the begining |
| 16:49.09 | brettnem | Hengei: you can reinstall over existing |
| 16:49.10 | Hengei | de= to |
| 16:49.32 | brettnem | oooh.. thanks.. |
| 16:49.42 | Hengei | it gives me some problems |
| 16:49.48 | brettnem | ugh pastrami |
| 16:49.54 | Hengei | i'd like to follow a tutorial |
| 16:50.08 | brettnem | just follow the tutorial.. you should be fine |
| 16:50.30 | loud | then find / -name asterisk or rm -rf /etc/asterisk and /var/lib/asterisk |
| 16:50.42 | brettnem | "uninstalling" wont' do anything really.. do a "make clean" if you want the compiled code to be erased (in the installed, but compiled) |
| 16:50.57 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 16:51.01 | riemensc | sorry |
| 16:51.01 | Hengei | ok |
| 16:51.04 | brettnem | loud: that'll miss installed binariares and maybe even some modules.. |
| 16:51.13 | loud | and codecs. |
| 16:51.22 | riemensc | brettnem have you recieve the debugging? |
| 16:51.27 | brettnem | no |
| 16:51.28 | loud | but he wants to wipe the install. |
| 16:51.41 | brettnem | riemensc: send the sip debug to a pastebin |
| 16:51.56 | brettnem | well then he'll have to go to each place the install puts stuff and get rid of it.. |
| 16:52.04 | loud | yepp. |
| 16:52.08 | brettnem | which I think is more than /etc/asterisk and /var/lib/asterisk |
| 16:52.12 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 16:52.14 | brettnem | like maybe /usr/lib/modules/asterisk |
| 16:52.18 | brettnem | depending on distro |
| 16:52.38 | brettnem | also /var/spool/asterisk.. there are lots of places for asterisk pieces |
| 16:52.41 | riemensc | i´ve open pastebin.com |
| 16:52.47 | brettnem | ~pastebin |
| 16:52.47 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 16:52.57 | brettnem | thanks jbot! |
| 16:53.52 | riemensc | http://pastebin.ca/18639 |
| 16:54.20 | riemensc | can you help me brettnem? |
| 16:54.31 | junbug | and set a timeout limit or it will ge cached by seach engines.... |
| 16:55.05 | riemensc | i think the problem ist the proxy authentication from voipbuster or what do you think? |
| 16:55.29 | *** join/#asterisk loick (~loick@APuteaux-151-1-21-108.w82-124.abo.wanadoo.fr) |
| 16:56.47 | brettnem | riemensc: I don't know why you are getting those 407s.. |
| 16:57.11 | riemensc | i think not 407s =60s or 90s |
| 16:57.26 | brettnem | well you know.. |
| 16:57.48 | brettnem | I don't see an actual call being placed here.. you invite sip:h ...which can't be right |
| 16:58.04 | brettnem | We're at 83.169.155.92 port 13946 |
| 16:58.04 | brettnem | Reliably Transmitting: |
| 16:58.04 | brettnem | INVITE sip:h@213.61.187.157 SIP/2.0 |
| 16:58.44 | *** join/#asterisk trickyrick77 (~rsegrest@207.111.174.1) |
| 16:59.09 | brettnem | ok... I think I need to get back to my paid job.. |
| 16:59.38 | riemensc | how can I solve this the problem? |
| 16:59.57 | brettnem | don't dial "h" |
| 17:00.14 | brettnem | you are picking up your phone at 30690116 and dialing "h" |
| 17:00.27 | dos000 | anyone can comment on proxying versus using nat traversal with sip ? |
| 17:00.33 | riemensc | i´m from germany what is h? |
| 17:00.41 | riemensc | my telephone is nr is 30690116 |
| 17:00.46 | brettnem | h is incorrect |
| 17:00.59 | brettnem | I don't see you making a real telephone call in the debug. |
| 17:01.28 | riemensc | i´m using bristuff |
| 17:01.49 | brettnem | doesn't matter |
| 17:01.57 | riemensc | asterisk server with bristuff with isdn hfc card connect to isdn telephone in nt mode |
| 17:02.10 | brettnem | LOOK in the TO: |
| 17:02.17 | brettnem | <PROTECTED> |
| 17:02.17 | brettnem | To: <sip:h@213.61.187.157>;tag=as6c00e71d |
| 17:02.23 | brettnem | that won't work |
| 17:02.41 | riemensc | i would link call a other number |
| 17:02.53 | brettnem | like I said, you didn't show the trace of a real phone call attempt |
| 17:02.55 | riemensc | how can i change this h? |
| 17:03.11 | brettnem | well perhaps you didn't send me the right debug... or maybe your dial statement is hosed |
| 17:03.18 | riemensc | i think it´s a error in extensions.conf what do you think? |
| 17:03.30 | brettnem | perhaps.. but I don't even see the attempt in your debug |
| 17:03.36 | riemensc | i´ve send you the right debug |
| 17:03.55 | *** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net) |
| 17:04.04 | brettnem | ok, then you are pushing "h" on your phone if that;'s the right debug |
| 17:04.36 | brettnem | I'm looking for a: |
| 17:04.36 | brettnem | -- Executing Dial("Zap/1", "SIP/voippeer/7135551212") in new stack |
| 17:05.01 | brettnem | make sure you set verbose high |
| 17:05.08 | brettnem | like: set verbose 100 |
| 17:05.38 | riemensc | sorry, i´ve got another configuration for sipgate |
| 17:05.52 | brettnem | really.. I must run.. |
| 17:05.57 | brettnem | good luck |
| 17:06.24 | InfraRed | is there a way to dial a number from the CLI ? |
| 17:06.35 | brettnem | grr |
| 17:06.44 | brettnem | This should be on a FAQ |
| 17:06.45 | riemensc | just a moment please i good of toilette |
| 17:07.01 | brettnem | InfraRed: chan_oss |
| 17:07.02 | InfraRed | brettnem: my Q? |
| 17:07.07 | brettnem | yes |
| 17:08.34 | cripito | anyone running gentoo and asterisk around? |
| 17:08.39 | *** join/#asterisk mkrufky (~mk@68.160.103.77) |
| 17:08.40 | harryvv | nope |
| 17:08.46 | Nugget | Asterisk doesn't care what linux you use. |
| 17:08.46 | harryvv | not yet |
| 17:08.52 | cripito | yeap true |
| 17:08.55 | Nugget | Asterisk doesn't even care if you use Linux at all. |
| 17:09.00 | cripito | but i need a res_mysql_config.so |
| 17:09.01 | harryvv | as long as the linux is stable i dont care. |
| 17:09.08 | cripito | for astlinux |
| 17:09.24 | cripito | and the best option to get 1 is from a gentoo distro |
| 17:09.31 | mmlj4 | ManxPower: you alive? |
| 17:09.33 | cripito | b/c the way of the libs |
| 17:10.01 | mmlj4 | ManxPower: you got my crazy voicemail yet? |
| 17:10.36 | *** part/#asterisk Hengei (~Hengei@196.203.53.57) |
| 17:11.20 | *** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net) |
| 17:11.34 | Darwin35 | ok this sucks |
| 17:11.42 | harryvv | so who has had the least supply problems with the purchace of ip500s? |
| 17:12.02 | Darwin35 | the netweb looks nice btu it is not upgradeable via tftp |
| 17:12.11 | Darwin35 | and it is not friendly to setup |
| 17:12.11 | DarthClue | harryvv: how many do you need? and would you settle for ip501s? |
| 17:12.19 | astoria | are there a lot of people having problems supplying ip501s? |
| 17:12.24 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 17:12.30 | tzanger | what's the difference between 500 and 501? |
| 17:12.46 | DarthClue | i believe it is just a memory upgrade and maybe a security upgrade as well. |
| 17:12.46 | harryvv | just a firmware security upgrade |
| 17:12.52 | brettnem | DarthClue: you got any Cisco 7960s? |
| 17:13.12 | harryvv | I dont know about the memory part |
| 17:13.25 | DarthClue | brettnem: i don't actually stock, but i know how to find them. how many are you looking for? |
| 17:13.38 | Katty | hi. |
| 17:13.38 | brettnem | DarthClue: about 25 |
| 17:13.49 | DarthClue | harryvv: it has a larger memory footprint for sure. that is well documented. |
| 17:14.36 | harryvv | okay |
| 17:15.17 | *** join/#asterisk fugitivo (~ajf@201.255.103.224) |
| 17:16.19 | DarthClue | brettnem: voipsupply is probably your best bet for the ciscos |
| 17:16.33 | blitzrage | brettnem: have you tried the -biz list? I see people selling refurbed Cisco's on there often enough |
| 17:16.47 | brettnem | DarthClue: yeah, that's what I thought.. they used to have 10 and 20 packs but they disappeared.. |
| 17:17.03 | brettnem | blitzrage: yeah. I need to go dig through that.. |
| 17:17.15 | *** join/#asterisk datagen24 (~steve+mir@londonderry-cuda1-68-171-193-26.lndnnh.adelphia.net) |
| 17:18.08 | [TK]D-Fender | harryvv : Not buying through CCP? |
| 17:18.23 | datagen24 | ok i have given up on amp, can(would) anybody help me write config files for asterisk |
| 17:18.26 | harryvv | nice thing about voip is the way to get around the out of area 1800 out of country restrictions that some companies create |
| 17:19.00 | Darwin35 | not always |
| 17:19.28 | Darwin35 | I have found with broadvoice you canot dial all 800/866/877 nmbrs |
| 17:20.25 | harryvv | Tk, yea I might as well sending a msg to you |
| 17:20.33 | *** part/#asterisk brettnem (~Brett@207.90.232.34) |
| 17:20.58 | datagen24 | ok i have given up on amp, can(would) anybody help me write config files for asterisk?? |
| 17:21.25 | dudes | In sip.conf I put permit=192.168.1.0/24 and deny=0.0.0.0/0.0.0.0 but it still allows me to dial using a non-permited IP. |
| 17:21.48 | dudes | I've also tried in permiet=ip/netmask |
| 17:22.11 | astoria | datagen24: what are you trying to do? |
| 17:23.13 | datagen24 | astoria: i have a router in my dorm room (shared with 3 persons) we all live in diferent states so the conclusion was to set up voip |
| 17:23.29 | astoria | do you use sip? or iax? or what? |
| 17:23.40 | riemensc | can everybody me write the command for sip debugging |
| 17:23.44 | datagen24 | astoria: iax or sip i am using voicpulse connect |
| 17:24.00 | dudes | sip debug ? |
| 17:24.05 | astoria | then edit your iax.conf to setup your iax, and use extensions.conf to do call routing.. |
| 17:24.12 | *** join/#asterisk Paul[NOC] (~paul@66.195.243.254) |
| 17:24.15 | Paul[NOC] | Yo |
| 17:24.31 | astoria | datagen24: i don't have time to write the confs for you. but if you look around in the wiki and mailing lists, you should be ablet o find enough information to get started. |
| 17:24.48 | *** part/#asterisk tinpot (~nick@217.145.120.198) |
| 17:25.28 | riemensc | okay sip debug |
| 17:25.35 | datagen24 | astoria: been trying but everybody i have been able to find is using amp. and i cant get it working |
| 17:25.46 | riemensc | who of you use voipbuster.com? |
| 17:25.48 | riemensc | via sip |
| 17:25.51 | Paul[NOC] | I got this kinda weird problem, I setup a IVR Menu for our call center. Now once the key has been pressed I get a few second delay+static before it completes action |
| 17:26.06 | astoria | datagen24: who have you been talking to. I only tlak to conf people :) GUIs are horribly overrated |
| 17:26.17 | Paul[NOC] | Eww GUI's ;) |
| 17:26.32 | Paul[NOC] | But I'm biasted. I run a all Linux Datacenter ;) |
| 17:26.55 | datagen24 | astoria: googling, and msg board at voip-info.org |
| 17:27.08 | astoria | datagen24: well, do you have a specific issue? |
| 17:27.15 | Paul[NOC] | Anyone have any suggestions for me? |
| 17:27.52 | Paul[NOC] | DOnt want answer, Just a hint in the right direction |
| 17:28.04 | datagen24 | astoria: more i need to know where to start, man does have any info on the conf files |
| 17:28.06 | riemensc | i can not call via voipbuster.com |
| 17:28.17 | astoria | sounds like some kind of DTMF issue paul. |
| 17:28.39 | astoria | datagen24: go to voip-info.org and search for iax.conf or extensions.conf |
| 17:28.47 | astoria | datagen24: there are a lot of good examples. |
| 17:29.12 | harryvv | By chance is there any 180 though 188 country code? if not good. |
| 17:29.21 | harryvv | actually |
| 17:29.27 | datagen24 | astoria: will do thanks,imight be back if i need more help |
| 17:29.27 | harryvv | 118 country code? |
| 17:29.48 | Paul[NOC] | dtmfmode=inband |
| 17:29.48 | Paul[NOC] | dtmf=inband |
| 17:30.18 | riemensc | i´ve got the error code -- Got SIP response 488 "Not acceptable here" back from 213.61.187.157 |
| 17:30.30 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 17:31.24 | riemensc | d |
| 17:31.25 | Strom_C | good morning |
| 17:31.25 | Paul[NOC] | Going to read up on this |
| 17:31.29 | Paul[NOC] | Thanks astoria |
| 17:31.52 | obsidian-studios | hello all, still having problems with ANI/CID info on zap channels. Less consistently with the use of callerid=asreceived. However just a moment ago I got a call that returned the ANI/CID info of the zap channel. Instead of possibly unknown and the #? |
| 17:32.09 | Darwin35 | anyone here have a netweb 401 aka x401 from iareaphone |
| 17:32.21 | Darwin35 | that has it working iax2 |
| 17:32.49 | astoria | Paul[NOC]: sorry i couldn't help you more, I don't have too much experience with dtmf stuff |
| 17:33.30 | Paul[NOC] | Thats why we have wiki ;) |
| 17:34.55 | riemensc | * oej has qu |
| 17:35.02 | riemensc | <PROTECTED> |
| 17:35.10 | riemensc | what is this for a error code? |
| 17:37.25 | Paul[NOC] | hmm |
| 17:37.42 | [Jedi] | there's a 50$ TE410P in ebay |
| 17:37.51 | [Jedi] | hehehe |
| 17:37.53 | Paul[NOC] | astoria, no dice. You might giving it a call and hearing it? |
| 17:37.56 | Paul[NOC] | Maybe it'll stand out to you |
| 17:38.21 | astoria | Paul[NOC]: sure, drop me a msg with the number, i'll call it |
| 17:38.32 | astoria | Paul[NOC]: are you on a PRI or IAX or what? |
| 17:38.59 | Paul[NOC] | astoria, It's all IP over SIP Channels |
| 17:39.19 | astoria | Paul[NOC]: are you having echo issues at all? |
| 17:39.38 | Paul[NOC] | astoria, nope |
| 17:39.46 | Paul[NOC] | Not that I have noticed |
| 17:40.54 | astoria | Paul[NOC]: it's giving me a busy signal |
| 17:41.20 | *** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET) |
| 17:41.24 | riemensc | <PROTECTED> |
| 17:41.29 | riemensc | please help me |
| 17:41.34 | riemensc | i use voipbuster |
| 17:42.00 | Paul[NOC] | astoria horribly sorry |
| 17:42.01 | Paul[NOC] | wrong number |
| 17:42.50 | astoria | Wow, you're right paul, but I didn't hear any static. |
| 17:42.59 | yaaar | can you have identical extensions in different contexts, headed to different phones? |
| 17:43.01 | astoria | I just heard a big delay between when I dialed and the next menu. |
| 17:43.08 | Paul[NOC] | Yea, I cant figure that out |
| 17:43.16 | Paul[NOC] | I'm using Background and Queue |
| 17:43.18 | junbug | riemensc: did you allow ulaw and alaw at the same time in sip.conf? |
| 17:43.31 | riemensc | yes im using ulaw and alaw |
| 17:43.41 | riemensc | allow=ulaw |
| 17:43.44 | riemensc | allow=alaw |
| 17:43.56 | astoria | Paul[NOC]: check out googling for "delay dtmf site:digium.com" |
| 17:43.58 | Strom_C | why the hell would you use both of those at the same time? |
| 17:43.59 | astoria | there's a ton of responses. |
| 17:44.19 | Strom_C | use ulaw in north america, alaw everywhere else |
| 17:44.21 | Paul[NOC] | astoria, Thanks for the help bro |
| 17:44.34 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
| 17:45.41 | junbug | Strom_C: all depends on what provider allows |
| 17:47.16 | riemensc | i live in germany |
| 17:47.47 | riemensc | i delete ulaw from sip.conf |
| 17:48.29 | *** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt) |
| 17:49.21 | mrtwister | Hello, who tested h323 / oh323 and ooh323 ? i have specific need - g723.1 (exten => ... , Answer, Wait, Playback, Dial etc ) |
| 17:50.11 | Paul[NOC] | Damn |
| 17:50.17 | obsidian-studios | ManxPower: if you are around, I added the callerid=asreceived, which seems to help and made a difference, but some are still coming through with the ANI/CID info instead of unknown and the #? |
| 17:50.26 | Paul[NOC] | Set them all to rfc, devices and both channels |
| 17:50.32 | Paul[NOC] | turned off jitterbuffer and tos=lowdelay |
| 17:50.33 | Paul[NOC] | Hmm |
| 17:50.37 | Paul[NOC] | Cant figure this out heh |
| 17:54.13 | riemensc | wer sprich deutsch? |
| 17:54.38 | harryvv | Does europe use alaw? |
| 17:57.28 | [TK]D-Fender | Simple question : I'm trying to find out where to setup SQL storage of *'s queue log data, and the WIKI has me running around. Can someone link me more directly with docs on this? |
| 17:59.13 | ChkDigit | riemensc: Die Deutsche Leute? |
| 17:59.52 | *** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
| 18:00.00 | ChkDigit | Und die andere Leute, die Deutsch sprechen. |
| 18:01.02 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 18:01.29 | *** join/#asterisk ctjctj (~Chris@peashooter.cyberpaladin.com) |
| 18:02.59 | *** part/#asterisk lehel (~Lehel@82.79.20.17) |
| 18:05.13 | Paul[NOC] | DTMF Payload Type: |
| 18:05.17 | Paul[NOC] | What in the hell is that |
| 18:06.03 | Paul[NOC] | Dont understand it, Dont touch it |
| 18:06.08 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 18:06.51 | *** join/#asterisk Jimme (~James@dsl-80-45-4-73.access.as9105.com) |
| 18:07.28 | jake1932 | is there any drawbacks to using this device with Asterisk - http://www.astricom.com/usi3500.htm? |
| 18:07.46 | *** part/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 18:08.10 | mut | <PROTECTED> |
| 18:08.10 | Jimme | hi there. interested in info on asterisk ability for recording calls costs and tennent abilities. I am looking at purchasing a copy with the required hardware to setup a PBX for a medium size shared facility |
| 18:08.15 | mut | *shrug* |
| 18:08.28 | jake1932 | waiting on the price |
| 18:08.44 | Paul[NOC] | Jimme, Asterisk itself is free |
| 18:08.51 | Paul[NOC] | It records calls pretty effective |
| 18:09.03 | Paul[NOC] | We do about 150 calls per week with no problem |
| 18:09.51 | jake1932 | mut: is there another way to connect asterisk to a USA BRI? |
| 18:10.28 | Jimme | Paul[NOC], yea we would be able to go down the route for paid support or whatever if asterisk does what we need it to |
| 18:10.53 | Jimme | when you say records calls, you get an output of who made which call and durations etc ? |
| 18:11.16 | MikeJ[Laptop] | Jimme, there are extensive CDR capabilities |
| 18:11.28 | *** join/#asterisk _deg_ (~deg@200.146.0.254) |
| 18:11.31 | MikeJ[Laptop] | Paul[NOC] thought you meant audio recording calls... |
| 18:11.36 | MikeJ[Laptop] | which it does too ;) |
| 18:11.56 | Jimme | ah yes |
| 18:12.04 | MikeJ[Laptop] | actual costing is a little more difficult |
| 18:12.27 | jake1932 | just import the cose table and do a join |
| 18:12.29 | jake1932 | cost |
| 18:12.40 | Jimme | well, all we would need is duration of calls, we could work it out from there. would it be in mysql or something easy to get to ? |
| 18:12.45 | Jimme | ah, sounds like it would be |
| 18:12.48 | jake1932 | yes |
| 18:13.07 | jake1932 | tyou can have asterisk output cdr to mysql |
| 18:13.39 | Jimme | perfect for what I want to do |
| 18:14.35 | ctjctj | Hello again. I'm attempting to get Festival to work with asterisk and having an unpleasent time of it. I've verified that festival |
| 18:14.37 | Jimme | so, in terms of ability to subdivide the PBX for tenants, is that something asterisk understands ? |
| 18:15.09 | [TK]D-Fender | Jimme : As is run it for use by multiple business units? |
| 18:15.10 | ctjctj | 'Saytext' works. I've verified that the macro does what is expected. But I get a quick, <1s, sound and but not the utterance. |
| 18:15.15 | *** join/#asterisk konfuzed (~kvirc@H129.C72.B0.tor.eicat.ca) |
| 18:15.16 | jake1932 | you have one file (extensions.conf) that holds all the extensions and contexts - you can subdivide however you see fit |
| 18:15.47 | Jimme | yes [TK]D-Fender |
| 18:16.00 | [TK]D-Fender | Jimme : Definately. |
| 18:16.16 | Jimme | ok, sounds good :) |
| 18:16.52 | Jimme | so what sort of size system can you run on a run of the mill rack mount server in terms of extensions / calls etc |
| 18:16.57 | [TK]D-Fender | Jimme : Asterisk can do a LOT, but how turn-key a solution are you looking for? Are you prepared to learn * inside out and do it yourself or are you expecting your solution to do a bit more hand-holding? |
| 18:17.23 | [TK]D-Fender | Jimme : Call density is variable depending on the technologies used. What did you have in mind |
| 18:17.35 | jake1932 | jimme - if you search for asterisk dimensioning on viop-info.com it'll give you a better idea |
| 18:17.42 | jake1932 | voip-info.com |
| 18:17.51 | Jimme | ok will do in a sec. |
| 18:18.06 | jake1932 | er voip-info.org |
| 18:18.51 | Jimme | um, were looking at maybe 50-100 extensions max.... no idea at all on other stuff at this stage |
| 18:18.57 | jake1932 | Jimme - here's the exact URL: http://voip-info.org/tiki-index.php?page=Asterisk%20dimensioning |
| 18:19.45 | ctjctj | Jimme, You might want to also take a look at asterisk@home just to learn more. It is a "turnkey" asterisk for small offices or homes. But it gives you an idea of what asterisk can do for you. |
| 18:20.04 | Jimme | i have no problem getting my hands dirty [TK]D-Fender, i dont expect to compile and go :) |
| 18:20.21 | Jimme | ok thanks, will look at that as well |
| 18:20.38 | ctjctj | I've been working with asterisk for about 4 weeks and am barely able to get it to do the things I want it to do. And I know there is lots more. It is a complex and powerful tool, and like any powerful tool, it takes time to learn out to do it "right" |
| 18:20.43 | [TK]D-Fender | Jimme : Ok, how many lines, and using what technology? |
| 18:21.08 | FarrisG | Does anyone have enough experience with Polycom 301/501 to tell me which ones I ought to get if given the choice? |
| 18:21.52 | DarthClue | FarrisG: 501 |
| 18:21.57 | [TK]D-Fender | 501 most likely. 501 has speakerphone, more call buttons, etc. But depends on your goal of course |
| 18:21.59 | konfuzed | I keep getting this error and can't find the cause. is it config on my side or config on providers side |
| 18:21.59 | konfuzed | CLI> Jul 27 14:34:53 NOTICE[12432]: chan_iax2.c:5888 socket_read: Registration of 'profx' rejected: Registration Refused |
| 18:21.59 | konfuzed | Jul 27 14:35:04 NOTICE[12432]: chan_iax2.c:5452 socket_read: Rejected connect attempt from 66.96.30.25 |
| 18:21.59 | konfuzed | i only get the second message when someone places a call to the number. |
| 18:21.59 | konfuzed | Oh this is setup via iax.conf and is Asterisk to asterisk |
| 18:22.01 | konfuzed | <PROTECTED> |
| 18:22.09 | konfuzed | this darn editor |
| 18:22.10 | DarthClue | i have one that i am using right now and it is just too cool |
| 18:23.07 | jake1932 | konfuzed: looks like an authentication error |
| 18:24.23 | *** join/#asterisk firestrm (firestrm@S010600095b829483.gv.shawcable.net) |
| 18:24.27 | FarrisG | Thanks. One of my execs' goals is to eventually have "call appearances" or whatever it's called so that the admin can tell who is on the phone and who isn't. Is this even possible with sip/Asterisk? I'm sure it could be done with some kind of web app, which I'm looking into, but they'd like to have it on the phone or a sidecar |
| 18:25.32 | [TK]D-Fender | FarrisG : Call Appearances works on Polycom, but NOT with Asterisk (to date) |
| 18:26.12 | FarrisG | [TK]D-Fender: Not with asterisk period, or not with asterisk and polycom? |
| 18:26.20 | [TK]D-Fender | (last I checked). SNOM 360 supposedly does though and is at a similar price-point. You could also run a Manager Interface GUI tool like IPSwitchboard for that effect |
| 18:26.49 | [TK]D-Fender | FarrisG : Believe its Asterisk's implementation or something to that effect |
| 18:26.50 | firestrm | i need 8 regular office desk type phones and 2 receptionist.. any opinions on what gear i should look at ? i havent a clue and i have to put the quote together in one day.. :( , cost not as important as ease of use (both user and administrator) and reliability/quality |
| 18:26.56 | FarrisG | Yeah, I'm considering IPSwitchboard. It's just hard to get these old fogies to use software instead of plastic |
| 18:27.01 | [TK]D-Fender | IPSwitchboard is the easiers way to see who's on calls. |
| 18:27.26 | Jimme | [TK]D-Fender, a single ISDN 30 (eventually with max channels) |
| 18:27.40 | DarthClue | firestrm: define receptionist. how many lines need to be supported? |
| 18:27.44 | [TK]D-Fender | Jimme : Euro E1? |
| 18:27.59 | firestrm | DarthClue, 4 lines, plus 1 transfer to remote office |
| 18:28.14 | [TK]D-Fender | I find with auto-attendent's these days you hardly need receptionists. |
| 18:28.19 | *** join/#asterisk Kurtism (~IRC-Maste@h66-38-148-197.gtconnect.net) |
| 18:28.54 | Darwin35 | I hate phones you have to update with windows |
| 18:28.57 | firestrm | [TK]D-Fender, i know, but these guys cant seem to wrap their minds around autoattendant |
| 18:29.02 | DarthClue | firestrm: Polycom IP501 or IP600 will do and are highly recommended. If you want to spend the big bucks, look at Ciscos, but they aren't recommended simply because you have to pay for everything, the firmware, the ac adapter, the phone, etc. |
| 18:29.02 | [TK]D-Fender | Jimme : Asterisk will work just fine for that size, but aim for a 3 GHZ+ server... |
| 18:29.20 | firestrm | Darwin35, i agree with you on that :) |
| 18:29.25 | Darwin35 | I now have 2 phones that wont work because wrong flash ver |
| 18:29.41 | bkw_ | exit |
| 18:29.43 | bkw_ | doh |
| 18:29.47 | [TK]D-Fender | firestrm : Neither can my head office. They BOTH have their heads up their asses. "Sure, lets get the most advanced technology... and then pay someone to FULL-TIME ANSWER EVERY FRIGGEN CALL" |
| 18:29.50 | Darwin35 | I have 2 x401 aka netweb401 |
| 18:29.52 | firestrm | Darwin35, tried Direct flashing? |
| 18:30.14 | Darwin35 | no explain |
| 18:30.23 | DarthClue | i have an ip501 that can handle up to 24 calls on 3 line instances with the latest firmware. |
| 18:30.53 | Darwin35 | I dont know the port nmbrs on the phone for flashing |
| 18:31.01 | ctjctj | Does anybody have TTS working with Asterisk and Festival? I'm unable to get anything but a short burst of sound from the Festival application in my dialplan. |
| 18:31.21 | firestrm | Darwin35, all hardware has a way of getting at the flash for manufacturing.. it may be a ICP connector, or you may be able to desolder the flash part and program it in a prom burner.. |
| 18:31.30 | Darwin35 | did you add the patch for festival or using the perl script |
| 18:31.31 | Jimme | [TK]D-Fender, yea euro (uk).... i need to get back up to speed in the terms for this stuff :/ |
| 18:32.14 | ctjctj | Installed it from ports under FreeBSD, the festival command is there, it connects to the festival server, but it looks like it isn't in the right format? |
| 18:32.19 | firestrm | Darwin35, its just a matter of a little reverse engineering.. ive never yet been defeated by a bad flash.. |
| 18:32.49 | Darwin35 | well these are not mine to take apart |
| 18:32.52 | Darwin35 | yet |
| 18:32.57 | firestrm | :) |
| 18:33.02 | yaaar | can you have identical extensions in different contexts, headed to different phones? and then route the call to them based on the context it's coming from? |
| 18:33.12 | Darwin35 | 1 is mine but wait to get it working before moding it |
| 18:33.36 | Darwin35 | I have to mod it for a head set |
| 18:33.39 | firestrm | is cisco ip phones a good choice? or is there a better for the $$ unit out there? |
| 18:34.06 | FarrisG | Can you tell me which is better, SNOM 360 or Polycom 501? |
| 18:34.07 | nDuff | firestrm, personally, I prefer Snom |
| 18:34.23 | DarthClue | firestrm: Polycom IP501 or IP600 will do and are highly recommended. |
| 18:34.33 | nDuff | FarrisG, I haven't tried the Polycoms yet, but of the phones I *have* used, the Snom 360 is the best of them. |
| 18:34.36 | firestrm | snom.?. hmm easy to config? good quality? |
| 18:34.46 | _DAW | Polycom is a fantastic telephone |
| 18:34.53 | Darwin35 | once I get to working on the firmware for this phone I am going to change alot |
| 18:34.54 | astoria | I love the polycoms. |
| 18:35.09 | Darwin35 | make it more user friendly and more features |
| 18:35.13 | Mw3 | is it possible with asterisk to send the faxes to a zap device (it's working now) but _also_ save them to pdf/tif ? |
| 18:35.16 | DarthClue | i am beginning to think that maybe i am a bot and that i really don't exist...firestrm, can you see this? |
| 18:35.20 | nDuff | firestrm, the snom is easy to config, yes; nice hardware; good sound quality, including the speakerphone |
| 18:35.23 | *** join/#asterisk Craziman2 (~donnie@boromir.apid.com) |
| 18:35.26 | astoria | Mw3: look up spandsp |
| 18:35.35 | nDuff | firestrm, and on top of that the snom runs Linux underneath. :) |
| 18:35.38 | Mw3 | astoria: is it capable to do that ? |
| 18:35.45 | *** join/#asterisk hugo234 (~icechat5@83-65-72-2.berggasse-II.xdsl-line.inode.at) |
| 18:35.48 | DarthClue | FarrisG: i haven't used the snoms, but the Polycom I have sitting right here is really nice. |
| 18:35.54 | astoria | Mw3: i'm not sure how it works on a x100p, but its runs okay on my te110p with a PRI |
| 18:36.01 | firestrm | nDuff, which snom model would you recomend for a receptionist phone? and a desk phone? |
| 18:36.12 | Nugget | oh, spiffy. I didn't know polycom was in austin |
| 18:36.15 | nDuff | used to be literally right next door to them before we moved. |
| 18:36.36 | nDuff | Nugget, they're just next to the 360 bridge pictured on the phonebook. |
| 18:37.02 | Darwin35 | I think these x401/402 will be a good phone better then grandstream once firmware is fixed and the webinterface is cleaned up |
| 18:37.26 | firestrm | egad! there in germany... i would have to find a local stocking distributer.. |
| 18:37.35 | nDuff | firestrm, hmm. A receptionist for how big a facility? Call parking, particularly with the extra-line-buttons doohicky, doesn't work without the Snom Media Server. |
| 18:37.45 | DarthClue | firestrm: can you see me? |
| 18:37.51 | [TK]D-Fender | Both the SNOM and Polycom seem pretty nice. Depends what you're going to do with it. Polycom's have a LOT of serious features, but tatke a bit to set up. |
| 18:37.59 | hugo234 | Hello! I have already a running setup of asterisk. Everything works fine except the CID thing. The CID is transmitted to the pstn without problems but the other direction -> no way. I can't see the CID calling other VOIP-users on this server either. Any idea? |
| 18:38.40 | hugo234 | on the console it's no problem |
| 18:38.44 | nDuff | firestrm, our snom is for the head honcho of a nonprofit we're hosting, and she's pretty happy with it, but she's not necessarily as demanding as a receptionist might be wrt features for handling lots of calls at once (and having those features actually *work* wihout server support -- there's a bounty, but only $100 and nobody's filled it). |
| 18:39.12 | firestrm | DarthClue, no.. are you trying to pm me? |
| 18:39.13 | *** join/#asterisk ctooley (~ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
| 18:39.18 | FarrisG | I think it's come down to Polycom 501s and SNOM 360s. I have a quick and cheap source for the Polycom, but need to source the SNOMs to get a price comparison. Are the 501s really difficult to configure for SIP/Asterisk? |
| 18:39.43 | DarthClue | firestrm: no, just making sure you could see me in the channel. i was told that i didn't exist last night so i'm just a little paranoid today. |
| 18:39.53 | nDuff | FarrisG, figure about $240 for the Snom 360. |
| 18:39.54 | Darwin35 | grrr |
| 18:40.04 | Darwin35 | and now I cant reach anyone at the comany |
| 18:40.07 | FarrisG | I also need to get a hold of someone to support these shitty Grandstream BudgeTone 101s that automatically upgraded their firmware and are having codec issues |
| 18:40.07 | ctooley | FarrisG They're significantly easier to configure well for Asterisk and the Snom 360 |
| 18:40.11 | firestrm | DarthClue. lol.. i can see you.. |
| 18:40.13 | [TK]D-Fender | FarrisG : Before making any serious suggestion, what do you readlly EXPECT from your phones? Do you support PoE? |
| 18:40.19 | Darwin35 | I bet they can flash it remotllt |
| 18:40.40 | Darwin35 | just pisses me off they setup the flash to use IE |
| 18:40.50 | DarthClue | FarrisG: The Polycoms are easy to configure. And they will run you about $170 without shipping costs. |
| 18:41.11 | FarrisG | [TK]D-Fender: No PoE here. What we expect is GOOD speaker phone, long uptime and usage. |
| 18:41.34 | [TK]D-Fender | Polycom is probably your best bet there. They are the speakerphone kings.... |
| 18:41.45 | FarrisG | THen I think polycom it is |
| 18:42.19 | DarthClue | yeah, Polycom phones are great for speakerphones. |
| 18:42.31 | FarrisG | My source is $210 per unit shipped. Think I can get better elsewhere? |
| 18:42.33 | _DAW | FarrisG - Polycom is your choice here, but by from a certified reseller. You may pay a little more, but if not you will have a hell of a time with support.. ie firmware |
| 18:42.39 | yaaar | nDuff: the speakerphone on my cisco 7940 sounds great |
| 18:42.49 | DarthClue | FarrisG: on the polycoms? how many do you need? |
| 18:43.01 | firestrm | so im thinking for a 4 1b line in/ 10 extension system.. 8 x snom 190's and 2 snom 360's.. now all i have to do is figure out where i can buy em from.. |
| 18:43.09 | _DAW | FarrisG - What model, we are polycom certified. |
| 18:43.12 | [TK]D-Fender | FarrisG : For the 501 you should be able to get them for +/- 180$USD |
| 18:43.34 | nDuff | yaaar, we haven't tried Cisco; frankly, we mostly stuck around the low end of the market -- Grandstreams, Sipuras -- before trying the Snoms. |
| 18:43.40 | _DAW | FarrisG - How many do you need? |
| 18:43.52 | ctooley | FarrisG when I was working for my last company we were a Polycom partner selling 500/501 phones for $185 |
| 18:44.00 | FarrisG | Right now I guess I need 6 polycom 501s |
| 18:44.03 | yaaar | nDuff: I can understand. I'm probably going to get some polycoms or snoms in the near future myself |
| 18:44.12 | *** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net) |
| 18:44.12 | colinm_ | yaaar: supposedly cisco's using polycom tech for the speakerphone feature. so no big surprise there |
| 18:44.23 | *** part/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net) |
| 18:44.38 | *** part/#asterisk ellvis (~evills@adsl-data-148.84-47-83.telecom.sk) |
| 18:44.38 | yaaar | colinm_: but i hear the polycoms don't have the echo-cancelling feature these ciscos have... |
| 18:44.50 | DarthClue | FarrisG: where does it need to be shipped to? i might be able to get you a source that can get them to you for about 175 shipped. |
| 18:45.11 | FarrisG | Allen, TX |
| 18:45.29 | DarthClue | FarrisG: can i pm you? |
| 18:45.37 | FarrisG | DarthClue: Certainly |
| 18:46.46 | Paul[NOC] | Hmm anyone know of a problem with grandsteam and sending DTMF tones |
| 18:46.56 | Paul[NOC] | (using rfc on both sides) |
| 18:47.36 | Paul[NOC] | Anyone feel like going over my setup for a few bucks? |
| 18:48.05 | yaaar | does the snom 320 support PoE? |
| 18:48.31 | DarthClue | Paul[NOC]: use info on the grandstreams, works better. |
| 18:48.56 | DarthClue | Paul[NOC]: if that doesn't work, i'll take a look for a few bucks |
| 18:49.09 | Paul[NOC] | DarthClue, I heard info does not work with voicemail |
| 18:49.14 | Paul[NOC] | VoiceMailMain |
| 18:49.30 | [TK]D-Fender | yaaar : Apparently |
| 18:50.11 | DarthClue | Paul[NOC]: works with my bt101. |
| 18:50.29 | [TK]D-Fender | ~DTMF |
| 18:50.29 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 18:50.52 | lathos42 | I knew it was touch tones, but I didnt know what it stood for :) |
| 18:51.01 | Paul[NOC] | DarthClue Thanks |
| 18:51.07 | Paul[NOC] | Testing it now |
| 18:51.08 | lathos42 | and silly me for not asking jbot |
| 18:51.18 | DarthClue | Paul[NOC]: np. |
| 18:51.33 | *** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net) |
| 18:51.51 | harryvv | how many small bussiness really need a second public ip for a voip setup? |
| 18:51.58 | harryvv | is it even recomended? |
| 18:52.31 | Paul[NOC] | We use private and public, Public for our remote locations that arent on the VPN List |
| 18:52.36 | Paul[NOC] | and private for the call center and Operations Room |
| 18:52.37 | DarthClue | harryvv: depends on the type of business and what that ip gives them. |
| 18:52.46 | *** part/#asterisk ctjctj (~Chris@peashooter.cyberpaladin.com) |
| 18:53.18 | harryvv | So basicly for reliability and security is it even nessesary to use a second ip for voip? That never crossed my mind untill somone sugested it. |
| 18:53.20 | Paul[NOC] | Ok |
| 18:53.23 | Paul[NOC] | Info worked great |
| 18:54.10 | DarthClue | harryvv: not unless the second ip is on a different route or you can guarantee that the second ip will always be kept private. |
| 18:54.28 | DarthClue | Paul[NOC]: i accept donations at payments@snowprods.com |
| 18:54.36 | harryvv | is it even really nessesary for a small bussiness though? |
| 18:54.55 | Paul[NOC] | DarthClue, I have one problem. You fix this and you'll get a nice donation ;) lol |
| 18:55.03 | DarthClue | harryvv: no, unless those conditions that i outlined above exist. |
| 18:55.06 | *** part/#asterisk datagen24 (~steve+mir@londonderry-cuda1-68-171-193-26.lndnnh.adelphia.net) |
| 18:55.18 | DarthClue | Paul[NOC]: have a seat on the couch and let's discuss it. |
| 18:56.06 | Paul[NOC] | DarthClue, I setup a Menu for incoming calls, Now it all works. But after they make a selection there is a delay. Normally about 5 seconds before switching extensions |
| 18:56.34 | puowvip | zzzz |
| 18:57.12 | DarthClue | your digit / exten timeout is set and you haven't put in a specific enough extension to make it match right away. pastebin your extensions.conf and i'll have a look. |
| 18:57.28 | Paul[NOC] | I pasted part of it |
| 18:57.51 | *** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
| 19:00.01 | Craziman2 | Can I configured * to ring a 7960 different if the call is from a zap channel -vs- another sip call? |
| 19:01.03 | MRH2 | look for alert_info on the wiki |
| 19:01.08 | DarthClue | Paul[NOC]: was the DigitTimeout previously set at 5? that is what would have caused the delay. if you lower it to about 2, the delay shouldn't be as long. if it is still happening, i'll have to dig deeper, which means i need more of the conf file to see where it is actually going. although, i'm not sure you want line 14 in there on that pastebin. |
| 19:01.15 | Craziman2 | MRH2 thanks |
| 19:01.19 | *** join/#asterisk jdg (~jdg@CA03F867.adsl.mana.pf) |
| 19:01.36 | *** part/#asterisk jdg (~jdg@CA03F867.adsl.mana.pf) |
| 19:01.38 | harryvv | any reliable software based routers will pass sip? |
| 19:02.43 | MRH2 | do folk generally run asterisk with the -p option? |
| 19:03.02 | Paul[NOC] | Done, checking it DarthClue |
| 19:03.44 | Paul[NOC] | 7 Seconds |
| 19:03.45 | Paul[NOC] | it took |
| 19:03.51 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca) |
| 19:03.59 | SarahEmm | clear |
| 19:04.01 | SarahEmm | hihi |
| 19:04.14 | DarthClue | Paul[NOC]: HEAD or STABLE? |
| 19:04.36 | SarahEmm | sivana: still no luck |
| 19:04.45 | opus_ | the wiki is working against 1.2 |
| 19:05.58 | Paul[NOC] | central*CLI> show version |
| 19:05.58 | Paul[NOC] | Asterisk 1.0.7 |
| 19:06.03 | *** join/#asterisk pa (~Paolo@pa.user) |
| 19:06.16 | Paul[NOC] | I have 1.0.9 I can compile |
| 19:06.36 | DarthClue | Paul[NOC]: let me see if it does it on my HEAD box, it could be a bug in stable... |
| 19:06.49 | zoo | i am trying to register my sipura ATA at an asterisk via sip. I did not set localnet, since the phone is registering with a dynamic public ip. Asterisk does not answer my requests at all. The same locally did work. Any ideas? |
| 19:07.31 | dos000 | anyone can sugest nat traversal embeded box solution for sip ? |
| 19:08.38 | anthm | Paul[NOC] try something along the lines of http://pastebin.com/322655 |
| 19:08.47 | zoo | and my sipura comes from inside my local home network trough siproxd |
| 19:09.43 | yaaar | ok, so what are you folks using to manage a whole bunch of extensions? preferably dynamically, so i don't have to reload all my extensions every time i change one... |
| 19:09.51 | anthm | when you put the wildcard exten in the same context as the menu you are making it possible that you could dial a 10 digit exten |
| 19:10.07 | anthm | so it will not react when you dial just 4 |
| 19:10.19 | anthm | cos you may still dial 455555555 |
| 19:10.37 | harryvv | anthm what router do you use to pass sip though |
| 19:11.03 | anthm | you mean like ser ? |
| 19:12.14 | harryvv | are you using ser? |
| 19:12.21 | harryvv | I have not used it |
| 19:13.30 | Paul[NOC] | anthm, Thanks I am checking now. Just gotta redo the context |
| 19:13.54 | anthm | np |
| 19:14.01 | *** join/#asterisk Holos (~asdf@72.1.197.10) |
| 19:14.05 | Holos | I just downloaded CVS-Head to install on a machine and Zaptel "make linux26" is erroring with: zaptel.c:1745: warning: ignoring return value of ‘copy_from_user’, declared with attribute warn_unused_result |
| 19:14.14 | Holos | Anyone seen this before? |
| 19:14.26 | DarthClue | Paul[NOC]: you will also want to put a WaitExten() in there or it will just fall thru right after playing the message. |
| 19:14.50 | DarthClue | Holos: why are you using make linux26? make should work fine. |
| 19:15.09 | harryvv | is there somone who has had luck with say m0n0wal or ipcop or some other software based firewall that will pass sip and create a two way call? |
| 19:15.18 | *** part/#asterisk Craziman2 (~donnie@boromir.apid.com) |
| 19:16.23 | Holos | DarthClue: Hmm.. I thought with kernel 2.6 you had to do the make linux26.. Thats what the README.Linux26 says.. |
| 19:16.24 | Paul[NOC] | DarthClue, Jul 27 15:15:24 WARNING[8157]: pbx.c:1934 ast_pbx_run: Invalid extension '4', but no rule 'i' in context 'incoming' |
| 19:16.31 | Paul[NOC] | Hmm |
| 19:16.45 | DarthClue | Paul[NOC]: pastebin it again and msg me |
| 19:17.15 | DarthClue | Holos: i'm using a 2.6 kernel and i just use make |
| 19:17.39 | Holos | DarthClue: make has the same error, this is on FC4 if it makes a difference.. |
| 19:17.48 | DarthClue | Holos: one sec... |
| 19:18.11 | DarthClue | Holos: http://www.darthclue.org/categories/3-Chalkboard-Examples |
| 19:18.31 | *** join/#asterisk hardwire (~hardwire@209-112-147-72-cdsl-rb1.nwc.acsalaska.net) |
| 19:19.12 | DarthClue | Holos: keep in mind, FC4 uses the newer gcc and will have lots warnings, but it souldn't have any failure errors. |
| 19:19.34 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 19:20.02 | riemensc | everybuddy speak german? |
| 19:20.07 | zoo | i do |
| 19:20.23 | riemensc | i´m using voipbuster and have a problem with extensions.conf or sip.conf |
| 19:20.40 | Holos | DarthClue: I was having problems with chan_zap saying pri_cpe was invalid signalling.. (Sangoma A102) so I thought that may be the problem, it seems to be a warning though. |
| 19:20.45 | zoo | riemensc: as far as i know, you need to use IAX2 for voipbuster |
| 19:20.56 | pa | yesterday i was able to see "Receiving call to myMSN; entering the context [incoming-call]; etc etc" when i received calls. Now im not able any more. And i uses same number of "v" in the asterisk command line.. |
| 19:20.57 | zoo | riemensc: sip does not work correctly |
| 19:21.04 | pa | how can i see caller id? |
| 19:21.28 | riemensc | zoo i´ve read by voipbuster the company support sip and iax |
| 19:21.35 | riemensc | do you use voipbuster? |
| 19:21.52 | zoo | riemensc: use IAX2 with asterisk |
| 19:22.16 | riemensc | i´ve read the quality is not so good |
| 19:22.23 | Paul[NOC] | It's very good |
| 19:22.36 | *** join/#asterisk Cherebrum (cracka@64.72.146.24) |
| 19:23.02 | DarthClue | Paul[NOC]: try this...http://pastebin.com/322684 |
| 19:24.38 | *** join/#asterisk fifer (~sirfifer@207.202.227.161) |
| 19:25.21 | Cherebrum | Anyone here running Asterisk on a PPC or PPC64? |
| 19:25.21 | fifer | Anyone have access to the new firmware (FC-0032-01-03.st ) for the Aastra 480i? |
| 19:25.28 | *** join/#asterisk wrarrl (~Myself@200.46.209.163) |
| 19:25.42 | Paul[NOC] | DarthClue, pm me your paypal |
| 19:25.44 | mrtwister | Cherebrum, PPC = pocket pc? |
| 19:25.48 | Cherebrum | PowerPC |
| 19:26.11 | jontow | why, PurplePC, of course! |
| 19:26.34 | Cherebrum | IBM PPC970FX |
| 19:26.46 | mrtwister | i ran it on zaurus |
| 19:26.50 | mrtwister | it is linux pocket pc |
| 19:27.19 | *** join/#asterisk bankrupt (~bank@pcp08695394pcs.500ash01.tn.comcast.net) |
| 19:27.30 | Cherebrum | mrtwister: an Apple Xserve G5 |
| 19:27.43 | harryvv | anyone have a sip phone care to test my firwall sip passing capabilities? |
| 19:27.59 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
| 19:28.28 | Cherebrum | harryvv: Is this a NAT router? |
| 19:28.38 | harryvv | i was told that ipcop can pass it |
| 19:28.50 | Cherebrum | What's the sip URI? |
| 19:30.44 | harryvv | Cherebrum sent you a msg |
| 19:31.17 | Cherebrum | ok |
| 19:31.27 | pa | can i insert in an exten something like A=postgresql(execute query X and return the result)? |
| 19:31.49 | pa | so i can use for example then $A with gotoif() command |
| 19:32.24 | pifiu | where do you set what codec you want the phoent o use? |
| 19:32.27 | pifiu | in which .conf file? |
| 19:33.10 | dudes | the config file that reflects the context which the phone uses |
| 19:33.18 | pifiu | extensions.conf? |
| 19:33.18 | _DAW | pifiu - if sip phone then sip.conf |
| 19:34.41 | pifiu | ok |
| 19:34.42 | pifiu | looking |
| 19:35.11 | *** join/#asterisk L|NUX (linux@202.5.146.154) |
| 19:35.58 | bkw_ | FYI folks switch-07 supports encryption if you wanna try it out |
| 19:36.04 | bkw_ | er switch-07.asterlink.com |
| 19:36.24 | bkw_ | must have latest CVS |
| 19:38.02 | slePP | which sort of encryption? |
| 19:38.15 | file | IAX2 encryption. |
| 19:38.18 | dudes | pa - you you want to run a system command to query postgresql ? |
| 19:38.41 | slePP | ah |
| 19:38.57 | Cherebrum | Anyone know of a PSTN to VoIP provider that does toll free numbers that provide ANI? |
| 19:39.21 | slePP | ANI or CID? |
| 19:39.33 | SwK[Work] | bkw_: when did that quit dumping core? |
| 19:39.51 | Strom_C | ugh, silly, there's no such thing as ANI or CID. there's "charge number" and "calling party number" |
| 19:40.01 | yaaar | how many of you guys are using realtime? is it working pretty well? |
| 19:40.15 | pa | dudes: yes i think. |
| 19:40.27 | slePP | yaaar: crashes asterisk about twice a week :> but i've got some ancient version going |
| 19:40.28 | bkw_ | SwK[Work], when mark fixed it in paris |
| 19:40.31 | pa | i want to execute a pgsql query and put the result in an asterisk variable |
| 19:40.45 | yaaar | slePP: how ancient? |
| 19:40.49 | slePP | uhm |
| 19:40.50 | slePP | april |
| 19:41.04 | Cherebrum | no |
| 19:41.20 | *** join/#asterisk Craziman2 (~donnie@boromir.apid.com) |
| 19:41.25 | *** part/#asterisk Craziman2 (~donnie@boromir.apid.com) |
| 19:41.45 | dudes | pa - ok |
| 19:42.03 | yaaar | ~realtime |
| 19:42.03 | jbot | hmm... realtime is http://www.voip-info.org/wiki-Asterisk+RealTime |
| 19:42.18 | dudes | pa - you can do that |
| 19:42.53 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 19:43.22 | pa | dudes: ah, nice |
| 19:43.27 | pa | do you know how can i? |
| 19:43.59 | dudes | pa - I'll pm you |
| 19:44.38 | pa | oh, thanks very much! |
| 19:45.22 | *** join/#asterisk hans (fugalh@falcon.fugal.net) |
| 19:45.58 | hans | i can make calls through nufone, but not when I originate the call from my sipura device (analog -> sipura -> asterisk -> nufone) |
| 19:46.10 | pa | i have another question: in my "incoming-context" for incoming calls through i4l driven ISDN TA i put a Playback() command. It works but i can hear about a second of something other audio file before hearing my audio file. It looks like the last audio file heared.. is it normal? |
| 19:46.38 | hans | I think it might be caller ID problem: |
| 19:46.40 | hans | <PROTECTED> |
| 19:46.40 | hans | <PROTECTED> |
| 19:46.40 | hans | <PROTECTED> |
| 19:47.17 | hans | does that sound right, or completely bogus? |
| 19:48.21 | hans | after the new, I get authreq, authrep, accept, ack, voice, ack, hangup |
| 19:48.25 | hans | <PROTECTED> |
| 19:48.46 | Kurtism | how do I phone my home VOIP phone and home PBX server from a regular old phone line 300 miles away? |
| 19:48.59 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 19:49.04 | Kurtism | I can phone it with another voip phone but not with a real phone |
| 19:49.04 | yaaar | sorry to sound so ignorant (man i'm in the habit of that around here) but is realtime a stock part of cvs-head? or how/where do I get it? |
| 19:49.16 | hans | Kurtism: you'd need some connection to the pstn |
| 19:49.23 | Kurtism | how do I get that? |
| 19:49.28 | NewSole | anyone know who did that open g729 |
| 19:49.52 | Kurtism | pstn? |
| 19:49.55 | hans | there's many ways, but for example companies that provide termination include vonage, broadvoice, nufone, etc |
| 19:50.00 | hans | regular phone network |
| 19:50.05 | L|NUX | NewSole : i think its based on Intel IPP |
| 19:50.06 | Kurtism | ok |
| 19:50.14 | Kurtism | how can I set it up myself |
| 19:50.22 | Kurtism | without going through some company |
| 19:50.32 | hans | or if you have asterisk hardware on a regular pots line (or t1, etc.) |
| 19:50.51 | NewSole | it is... but I have a linux lib I would like to use insted |
| 19:50.57 | Kurtism | pots line? |
| 19:51.04 | hans | plain old telephone service |
| 19:51.07 | Kurtism | oh |
| 19:51.10 | Kurtism | interestin |
| 19:51.20 | Kurtism | so I get a phone line from my local telephone company |
| 19:51.26 | Kurtism | MTS for manitoba |
| 19:51.41 | Kurtism | and hook my astrisk hardware up to it |
| 19:51.42 | *** join/#asterisk Corydon76-home (beige@Corydon76-home.sustaining.supporter.pdpc) |
| 19:51.42 | *** join/#asterisk needlz (~needlz@adsl-69-109-166-205.dsl.pltn13.pacbell.net) |
| 19:51.45 | *** part/#asterisk fifer (~sirfifer@207.202.227.161) |
| 19:51.51 | hans | yeah, then bring it into your asterisk box with a digium card or the like |
| 19:51.56 | yaaar | Kurtism: if you want to attach to the public telephone system, you're going to pay *somebody* |
| 19:52.11 | Kurtism | but then I still don't see how they can dial my voip number from 300 miles away on their pots line |
| 19:52.12 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 19:52.28 | yaaar | Kurtism: they will dial the number of the phone line you plug into the asterisk box |
| 19:52.32 | hans | sounds like you'd like a toll-free number from one of those services |
| 19:52.43 | yaaar | or if you get termination from one of the voip providers, they will give you the number |
| 19:52.48 | netnameus | what needs to be placed in extensions.conf to dial a phone number when an extension is dialed? For example, when ext 123 is dialed, I would like 555-5555 dialed |
| 19:52.51 | Kurtism | hmmmm |
| 19:53.22 | *** join/#asterisk Craziman2 (~donnie@boromir.apid.com) |
| 19:53.29 | hans | exten => 123,1,Dial(<dial stuff>/${EXTEN}) |
| 19:53.30 | jarrod | netnameus: exten => <digits>,1,Dial(${TRUNK}/5555555) |
| 19:53.36 | Kurtism | I was told its possible to completely by pass all service providers with voip, and have plain old telephone people phone my voip number. |
| 19:53.37 | jarrod | as long as trunk is defined |
| 19:53.49 | netnameus | thanks guys |
| 19:54.15 | hans | Kurtism: to get on the POTS network, you have to pay someone |
| 19:54.27 | Kurtism | ok |
| 19:54.31 | Holos | Anyone have an idea on how to solve a "ERROR[4524]: chan_zap.c:10281 setup_zap: Unknown signalling method 'pri_cpe'" error on a new install of CVS? It's FC4 with a Sangoma A104 |
| 19:54.37 | hans | either MTS or a voip provider or somebody |
| 19:54.38 | jarrod | yup |
| 19:55.17 | Kurtism | who do the telephone companies setup connections with for international numbers to work? |
| 19:55.25 | jarrod | heh |
| 19:55.47 | *** join/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 19:55.53 | *** join/#asterisk meppl (~mephisto@87.193.5.244) |
| 19:56.10 | Kurtism | there must be some huge database somewhere similar to dns |
| 19:56.50 | harryvv | somone told me long ago its not safe to put asterisk on a firewall yet, its one of the ways to let sip pass with no problems behind the firewall. |
| 19:57.03 | jarrod | or run SER |
| 19:57.11 | Holos | Kurtism: Call up MTS and ask them how they do it :) |
| 19:57.16 | Kurtism | lol |
| 19:57.27 | harryvv | or run ser yea, but ser needs its own public ip or run on the firewall or how would that work? |
| 19:58.04 | ChkDigit | Telco providers own or lease connections between each other. |
| 19:58.16 | ChkDigit | (For Kurtism) |
| 19:58.36 | *** join/#asterisk essobi_ (kstone@75.137.26.216.host.teledvance.com) |
| 19:58.44 | *** join/#asterisk cripito (~ncripito@67.154.143.190) |
| 19:58.46 | cripito | hi |
| 19:59.00 | cripito | anyone using gentoo around? |
| 19:59.02 | essobi_ | Bye :) |
| 19:59.06 | SarahEmm | yep i use gentoo |
| 19:59.11 | cripito | :) |
| 19:59.27 | cripito | did u have res_config_mysql.so available? |
| 19:59.30 | essobi_ | You don't count. You arn't people. |
| 20:00.25 | SarahEmm | cripito: err, do you mean using asterisk emerge? |
| 20:00.37 | cripito | i am working in an astlinux installation |
| 20:00.41 | cripito | no emerge :( |
| 20:00.53 | Kurtism | so if I want to make my astrix box's VOIP phone number accessible to non-voip telephones I have to talk to a telephone service provider and tell them I'm a telephone service provider and want to have my people beable to talk to your people. |
| 20:00.53 | *** join/#asterisk irv999 (~irv999@67.105.188.178.ptr.us.xo.net) |
| 20:00.55 | irv999 | hey all |
| 20:00.56 | Holos | Anyone here using a Sangoma A10x card that can post their configs to a pastebin? |
| 20:01.11 | irv999 | I have some major issues (some of you may have heard) dropped calls happening randomly |
| 20:01.23 | Kurtism | how does one go about becoming a telephone service provider eh |
| 20:01.28 | SarahEmm | Kurtism: huh? |
| 20:01.30 | SarahEmm | Kurtism: telephone service provider? |
| 20:01.35 | irv999 | I have come to no resolution with digium and my programmer has given up.. has anyone else had this issue? |
| 20:01.35 | SarahEmm | Kurtism: err.. what exactly do you want to do? |
| 20:01.48 | Holos | Kurtism: Just get a incoming DID from a DID provider and have them forward it over IAX2 or SIP to your asterisk machine |
| 20:02.03 | Kurtism | hmmmm |
| 20:02.11 | Kurtism | ok |
| 20:02.20 | SarahEmm | cripito: err... on gentoo? *confused* |
| 20:02.37 | cripito | :) sarahemm astlinux is a reduced version of gentoo |
| 20:02.38 | Kurtism | DID? |
| 20:02.52 | lathos42 | Gento rocks my socks |
| 20:03.04 | *** join/#asterisk needlz (~needlz@adsl-69-109-166-205.dsl.pltn13.pacbell.net) |
| 20:04.21 | hardwire | ok |
| 20:04.23 | hardwire | I need ringtones |
| 20:04.25 | hardwire | that don't suck |
| 20:04.30 | hardwire | that are nice and business worthy |
| 20:04.36 | hardwire | cause I dunno what snom was thinking |
| 20:04.54 | lathos42 | hardware: I had the same thought with the Sipura 841 |
| 20:04.54 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 20:04.56 | Juxt | good day |
| 20:05.01 | Juxt | how can i avoid this WARNING[9142]: chan_sip.c:2316 sip_write: Asked to transmit fram |
| 20:05.01 | Juxt | e type 64, while native formats is 4 (read/write = 4/4) |
| 20:05.04 | astoria | irv999: what did you do ? |
| 20:06.15 | *** join/#asterisk fugitivo (~ajf@201.255.100.39) |
| 20:06.39 | Juxt | i get this when trying to monitor a call with chanspy |
| 20:06.46 | Holos | Kurtism: A DID is a phone number assigned (usually) to T1 circuts. A DID provider would get you a number in their serviced area code and accept incoming calls for the number. When a call came in it would bridge it to your asterisk server. |
| 20:06.49 | irv999 | astoria: not what I did.. myself and another compnay designed a phone system with asterisk and a 23 channel PRI.. Drops calls randomly |
| 20:07.06 | astoria | irv999: ok. why are you in big trouble? |
| 20:07.29 | Holos | irv999: did you try CVS-Head, and stable versions? Could it be your hardware or is it Software? |
| 20:07.30 | irv999 | astoria: we have a sonicwall, 2 managed switches.. We have gone through a lot of troubelshooting.. the reason why I am in big trouble is if it does not get solved by friday, I am out 15K |
| 20:07.33 | *** join/#asterisk xheliox (~jeff@user-0c6se1v.cable.mindspring.com) |
| 20:08.12 | astoria | What did you pay 15k for? |
| 20:08.22 | astoria | Have you called your PRI provider? what did they say? |
| 20:08.43 | mut | ds3 |
| 20:08.46 | irv999 | astoria: no.. If I dont solve the problem, I am out 15K to put towards another phone system.. to replace asterisk |
| 20:08.50 | astoria | Did you do a zap debug and see what the hangup cause was on those calls? |
| 20:08.54 | irv999 | astoria: they said everything is ok.. |
| 20:09.13 | astoria | every PRI call is given a HANGUPCODE when a call is terminated. |
| 20:09.16 | astoria | Find out what that code is. |
| 20:09.25 | irv999 | astoria: lots of testing done.. I am not the only one having this trouble.. so it is not the pri, however we thing it has to do with the interaction between asterisk and the pri |
| 20:09.26 | astoria | Then you can debug from there. |
| 20:09.30 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:09.43 | astoria | Yeah, find out why the call is being terminated. Asterisk does not randomly hang up your calls. |
| 20:09.53 | mut | who else is having that problem? |
| 20:09.59 | irv999 | astoria: well in this case it is.. and it can if it is monitoring for a hangup.. |
| 20:10.17 | SarahEmm | irv999: what was the hangup reason? |
| 20:10.19 | irv999 | mut: another one of my programmers clienhts |
| 20:10.27 | astoria | irv999: find out the HANGUPCODE.. do a zap debug |
| 20:10.33 | netnameus | how do you initate the On-Demand recording? Is there a normal key pressed? |
| 20:10.44 | astoria | Then you will know the ISDN cause-code and you can see if the problem is really you or your provider. |
| 20:10.47 | mut | does a basic setup work for you? |
| 20:10.57 | mut | w/o anything fancy at all.. |
| 20:11.02 | mut | call in one line to another |
| 20:11.11 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-77-210.dsl.pltn13.pacbell.net) |
| 20:11.30 | wulfy814 | anyone with experience with multiple zaptel cards |
| 20:11.34 | irv999 | mut: it only involves outgoing and incoming calls |
| 20:11.45 | essobi_ | hey irv |
| 20:11.48 | cripito | if anyone have res_config_mysql.so in gentoo give me a wistle.... |
| 20:11.49 | *** join/#asterisk dekerz (~dek@69.2.232.142) |
| 20:11.51 | mut | yeh thats what i meant, |
| 20:11.54 | mut | if a person makes a call in |
| 20:11.56 | wulfy814 | I have two, one with 4 incoming analog |
| 20:11.58 | astoria | irv999: figure out what the cause code is! I can't repeat this enough. |
| 20:12.08 | irv999 | astoria 1 sec ok |
| 20:12.29 | mut | must be a very easily repeatable problem then? |
| 20:12.53 | hardwire | there must be pleasent rings.. somewhere |
| 20:13.21 | mut | pleasent rings? |
| 20:13.29 | hardwire | yes |
| 20:13.48 | hardwire | like pissing in the woods.. the pleasent sound of a waterfall |
| 20:14.00 | hardwire | or a businessy ring. that doesn't suck or is to obtrusive. |
| 20:14.04 | hardwire | but gets your attention |
| 20:14.05 | *** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net) |
| 20:14.10 | hardwire | and I don't want the phone to vibrate off the desk either |
| 20:14.19 | irv999 | from what I remember (because we have debug on) it Message type: DISCONNECT (69) |
| 20:14.24 | irv999 | <PROTECTED> |
| 20:14.35 | Ayano | can asterisk be set up to just pass sip traffic to an IP without auth if the other side will allow? |
| 20:14.50 | jarrod | yes |
| 20:14.56 | irv999 | I am willing to pay $$ to figure this out |
| 20:15.25 | Ayano | jarrod; Just do a registy statement with no user and pass? |
| 20:15.33 | Kurtism | I wonder where I can find a DID provider in Winnipeg Manitoba Canada |
| 20:15.39 | Ayano | yes |
| 20:15.40 | jarrod | just send it to the SIP server from the Dial statement |
| 20:15.42 | jarrod | without a registry |
| 20:15.57 | yaaar | is realtime a stock part of cvs-head? |
| 20:16.07 | Ayano | But then I will loose the ability to keep track of the cdr? |
| 20:16.08 | irv999 | who here has heard of asteria solutions group? |
| 20:16.09 | jake1932 | hardwire: http://www.partnersinrhyme.com/soundfx/watersounds.shtml |
| 20:16.26 | netnameus | how do you initate the On-Demand recording? Is there a normal key pressed? |
| 20:16.27 | astoria | asteria.. ha ha. close to me! |
| 20:16.34 | mut | could make ya phone play some 50 cent when a customer calls |
| 20:16.40 | astoria | so, irv999, the isdn cause code is 16?? normal hangup? |
| 20:16.51 | astoria | irv999: what version of * are you running? |
| 20:17.03 | *** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985405.sympatico.ca) |
| 20:17.24 | irv999 | astoria: how do I check version? |
| 20:17.40 | Holos | Kurtism: Check the Wiki and google. |
| 20:17.48 | dudes | irv999 - asterisk -V |
| 20:18.05 | harryvv | so.. run ser in front of fireall so sip can pass? or run asterisk in front of nat on a firewall box. whats to suggest? |
| 20:18.08 | astoria | asterisk -V |
| 20:18.09 | jake1932 | or "show version" in the CLI |
| 20:18.17 | irv999 | Asterisk CVS-HEAD that does not make sense |
| 20:18.28 | bkw_ | show version files |
| 20:18.47 | harryvv | bkw what do you think |
| 20:18.56 | irv999 | Asterisk CVS-HEADAsterisk CVS-HEAD built by root@stanker on a i686 running Linux on 2005-07-14 21:13:31 UTC |
| 20:18.57 | irv999 | Asterisk CVS-HEAD built by root@stanker on a i686 running Linux on 2005-07-14 21:13:31 UTC |
| 20:19.01 | cripito | if anyone have res_config_mysql.so in gentoo give me a wistle.... |
| 20:19.13 | bkw_ | show version files |
| 20:19.16 | bkw_ | thats what you need |
| 20:19.21 | *** join/#asterisk stkn_ (~stkn@stkn-active-pdpc.developer.gentoo) |
| 20:19.26 | bkw_ | it shows you the versions of all files in the build |
| 20:19.39 | irv999 | which one should I tell you? |
| 20:19.50 | bkw_ | what are you having problems with? |
| 20:19.57 | irv999 | bkw pri |
| 20:20.01 | astoria | bkw_: his zap channels are hanging up randomly |
| 20:20.11 | bkw_ | show file version chan_zaip.c |
| 20:20.13 | bkw_ | er zap.c |
| 20:20.56 | irv999 | 1.474 |
| 20:21.06 | bkw_ | update |
| 20:21.16 | bkw_ | its at like 1.479 I think |
| 20:21.42 | irv999 | last time we did an update on the that.. it dropped calls a lot because the programmer tried to get cute with the pri |
| 20:21.45 | bkw_ | 1.482 |
| 20:21.46 | bkw_ | thats it |
| 20:21.54 | Ahewes | Question about alsa console driver: is it possible to use a sound card as a regular extension with the alsa driver? If so, is this documented in the CLI or does someone have a pointer? |
| 20:24.53 | dudes | Ahewes - Dial(Console/default) |
| 20:25.19 | Ahewes | thanks, dudes, lookin it up. |
| 20:27.13 | *** part/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com) |
| 20:29.46 | SarahEmm | Ahewes: yes it is... |
| 20:29.52 | SarahEmm | err, oops. too late ;) |
| 20:30.36 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-135-112.red.bezeqint.net) |
| 20:32.05 | *** join/#asterisk irv999 (~irv999@67.105.188.178.ptr.us.xo.net) |
| 20:32.43 | zoo | Why do I get this, When I get a call via iax2? Rejected connect attempt from 213.61.187.156, request 's@default' does not exist |
| 20:33.28 | essobi_ | zoo: s is the default for when a number isn't found |
| 20:33.29 | Holos | zoo: You need to create a "s" extension in the [default] contect of extensions.conf |
| 20:33.41 | zoo | okay thanks |
| 20:33.45 | essobi_ | and default is the default context you have iax2 pointed at |
| 20:34.01 | essobi_ | s is the catchall is everything else fails context. |
| 20:41.46 | Kurtism | New Now Know How |
| 20:43.10 | *** join/#asterisk da_ve (~d@ip24-254-117-53.pn.at.cox.net) |
| 20:43.36 | *** join/#asterisk paski_fr (~paski_fr@stardust.noc.frontier.fr) |
| 20:43.42 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 20:44.40 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 20:47.35 | *** join/#asterisk nounoursfr (~Direct@stardust.noc.frontier.fr) |
| 20:49.07 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 20:49.50 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
| 20:52.37 | paski_fr | Good Evening. I ask you because I have a problem with a real time waiting queue. When we try to contact the waiting queue, all the agents phones ring, but when we answer, nobody is on the phone : just the silence. Is someone able to help me ? But when we call directly the phone, we can convers normally. |
| 20:53.46 | paski_fr | here are the logs of my asterisk during a call to the waiting queue |
| 20:53.48 | paski_fr | - Called SIP/103 |
| 20:53.48 | paski_fr | <PROTECTED> |
| 20:53.48 | paski_fr | <PROTECTED> |
| 20:53.48 | paski_fr | <PROTECTED> |
| 20:53.48 | paski_fr | <PROTECTED> |
| 20:54.00 | DarthClue | no! use pastebin. |
| 20:54.01 | paski_fr | <PROTECTED> |
| 20:54.01 | paski_fr | Jul 27 22:44:31 WARNING[4607]: res_musiconhold.c:870 local_ast_moh_start: No class: default |
| 20:54.01 | astoria | use pastebin or something |
| 20:54.03 | DarthClue | ~pastebin |
| 20:54.03 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 20:55.27 | paski_fr | I'm really sorry about that but I really would like to find a solution to this problem, and I think that I will be easier if I give you more information |
| 20:55.30 | brookshire | ~fxo |
| 20:55.30 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 20:55.46 | MikeJ[Laptop] | fxo? |
| 20:55.51 | brookshire | lol |
| 20:55.56 | brookshire | i need a better definition |
| 20:56.05 | Cybertoy | yeah .. that sounds like finance... |
| 20:56.12 | brookshire | ~fxs |
| 20:56.12 | jbot | fxs is, like, foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 20:56.41 | *** join/#asterisk derek_1234 (~derek@203.167.203.10) |
| 21:01.16 | MikeJ[Laptop] | ~fx-brooksisweird |
| 21:01.23 | MikeJ[Laptop] | hmmmm |
| 21:02.13 | paski_fr | is someone here able to answer to my question ? |
| 21:03.12 | MikeJ[Laptop] | paski_fr, I see the details, what is the problem? |
| 21:03.31 | paski_fr | <PROTECTED> |
| 21:03.51 | paski_fr | MikeJ, can i speak to you by PV |
| 21:03.56 | MikeJ[Laptop] | no |
| 21:03.59 | MikeJ[Laptop] | but you can here |
| 21:04.03 | MikeJ[Laptop] | they are sip calls, |
| 21:04.11 | MikeJ[Laptop] | are they traversing any nat? |
| 21:04.15 | paski_fr | yes |
| 21:04.30 | paski_fr | we can call directly the phones |
| 21:04.30 | MikeJ[Laptop] | ok, and canreinvite=no, nat=yes? |
| 21:04.49 | paski_fr | it s OK |
| 21:05.07 | MikeJ[Laptop] | call directly is traversing nat as well? same peers, or different? |
| 21:06.11 | paski_fr | we can call directly the phones but when we use the waiting queue, the voice doesn t pass |
| 21:06.41 | MikeJ[Laptop] | ok, you are using the same phones to call both ways, same peer defs and all? |
| 21:06.56 | paski_fr | yes |
| 21:07.17 | MikeJ[Laptop] | ok, and these are not using local or agent ? |
| 21:07.22 | paski_fr | without waiting queue all work |
| 21:08.05 | MikeJ[Laptop] | can you create a sip debug, including verbose 4 and debug 4 and pastebin it |
| 21:08.13 | MikeJ[Laptop] | of the broken call through queue |
| 21:08.35 | paski_fr | oki one minute |
| 21:08.42 | paski_fr | we are doing the test |
| 21:08.55 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 21:09.18 | paski_fr | do you want all the logs? |
| 21:10.20 | MikeJ[Laptop] | the debug and verbose of that one call only if possible |
| 21:10.20 | *** join/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 21:10.24 | MikeJ[Laptop] | in pastebin |
| 21:10.29 | MikeJ[Laptop] | ~pastebin |
| 21:10.29 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 21:11.38 | paski_fr | Ok we are creating a new window of pastebin |
| 21:11.52 | outtolunc | .. |
| 21:12.21 | harryvv | man, shaw is a rip off for static ip rates |
| 21:12.38 | harryvv | $84 dolars per month for a static ip address. |
| 21:12.44 | harryvv | for commercial rate |
| 21:13.15 | harryvv | time to go |
| 21:13.31 | SwK[Work] | MikeJ[Laptop]: |
| 21:13.36 | SwK[Work] | wheres that audiocodes patch |
| 21:13.56 | MikeJ[Laptop] | one sec |
| 21:14.13 | *** join/#asterisk pifiu (~myassisbi@208.205.181.170) |
| 21:14.38 | *** part/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
| 21:14.45 | paski_fr | here is the link in pastebin |
| 21:14.46 | paski_fr | http://pastebin.ca/18656 |
| 21:14.49 | jarrod | what |
| 21:14.51 | jarrod | what city are you in |
| 21:14.56 | *** join/#asterisk postel (~zz@postel.user) |
| 21:15.09 | paski_fr | thanks for your attenttion Mike |
| 21:15.43 | Darwin35 | yes I have a iax2 ohone |
| 21:15.46 | Darwin35 | phone |
| 21:15.49 | Darwin35 | 2 of them |
| 21:15.53 | Darwin35 | and it rocks |
| 21:16.30 | Darwin35 | was a pain had to go to a friends and barrow a windows laptop to flash the phone |
| 21:16.41 | Darwin35 | but once flashed it works great |
| 21:16.57 | *** join/#asterisk fr_soft (~fr_sofr@ven34-1-82-238-185-27.fbx.proxad.net) |
| 21:17.26 | yaaar | anybody know which of the gentoo packages {asterisk, zaptel, libpri, zapata, asterisk-perl} I should emerge -C before installing asterisk from cvs? |
| 21:18.02 | yaaar | i'm thinking 'all of them' but i'm not sure |
| 21:19.10 | DarthClue | yaaar: if you are using asterisk from cvs-head then you shouldn't be using the gentoo packages. |
| 21:19.11 | xheliox | libpri and zaptel |
| 21:20.03 | *** join/#asterisk sangee (~rkuru@207.188.77.86) |
| 21:20.15 | yaaar | DarthClue: uh, right. I've got 1.0.8 installed from the gentoo ebuild right now. But i'm about to move to cvs-head. so i just wanted to make sure i had all the packages tracked down which i need to remove |
| 21:22.35 | *** part/#asterisk mkrufky (~mk@68.160.103.77) |
| 21:22.49 | Darwin35 | this rocks |
| 21:22.59 | Darwin35 | I have a new phone |
| 21:25.23 | fr_soft | Hi, do you know if there is a dsp software library for V.22 bis protocol with asterisk ? (i know spandsp but it's seems not functionnal) |
| 21:25.47 | file | spandsp works great |
| 21:26.15 | fr_soft | with V.22 bis have you made some tests ? |
| 21:27.05 | fr_soft | in the src there is a comment from coppice that said it's a work in progress |
| 21:27.08 | paski_fr | MikeJ are our logs correct ? |
| 21:27.36 | file | everything is a work in progress with asterisk |
| 21:28.36 | fr_soft | but have you made some tests ? |
| 21:29.04 | MikeJ[Laptop] | paski_fr, sorry, trying to look at them and getting distracted.. |
| 21:29.14 | MikeJ[Laptop] | and there seems to be a lot of noise in that file. |
| 21:29.25 | MikeJ[Laptop] | can you try with the inbound call being forced to ulaw |
| 21:29.28 | *** join/#asterisk xlyz (~xl@213-140-17-96.fastres.net) |
| 21:29.35 | MikeJ[Laptop] | disallow=all and allow=ulaw |
| 21:29.42 | *** part/#asterisk xlyz (~xl@213-140-17-96.fastres.net) |
| 21:30.36 | paski_fr | ok we contact you as soon as it will made |
| 21:30.42 | MikeJ[Laptop] | ok |
| 21:30.58 | *** join/#asterisk patrick^ (~patrick__@birch4.mountaincable.net) |
| 21:31.13 | *** part/#asterisk Craziman2 (~Craziman2@boromir.apid.com) |
| 21:33.20 | *** join/#asterisk bjohnson (~bjohnson@i216-58-65-53.igs.net) |
| 21:33.23 | pifiu | Does anyone know where in the Polycom config files I set the configuration server? |
| 21:34.14 | paski_fr | THANKS MIKE!!! IT'S WORKING |
| 21:34.40 | *** join/#asterisk jackfiber (~jack@66.96.209.21) |
| 21:34.57 | jackfiber | hi all, has anyone used tdm400P on FreeBSD ? |
| 21:35.06 | jackfiber | any wildcard on FreeBSD? |
| 21:35.16 | Darwin35 | yes |
| 21:35.18 | Darwin35 | on 5.4 |
| 21:35.35 | Darwin35 | I have the tdm40b |
| 21:35.40 | paski_fr | We have forced the G711 and the voice pass in the waiting queue |
| 21:36.09 | jackfiber | Darwin35> I got wildcard to work under Linux but on FreeBSD FXS modules are not being detected but FXO modules are detected |
| 21:36.21 | jackfiber | I have two cards, TDM03B and RDM40B |
| 21:36.25 | jackfiber | TDM40B |
| 21:36.30 | MikeJ[Laptop] | what I didn't see in that file was the codec the original call was in. |
| 21:36.36 | Darwin35 | mine are working fine |
| 21:36.41 | *** join/#asterisk wunderkin (kev@24.137.147.163) |
| 21:36.43 | MikeJ[Laptop] | this may be the translate via sln bug |
| 21:36.46 | paski_fr | Mike: Thanks for all |
| 21:36.53 | jackfiber | both FXS and FXO? |
| 21:36.57 | Darwin35 | yes |
| 21:36.59 | MikeJ[Laptop] | you shouldn't have to do that. |
| 21:37.14 | Darwin35 | are you running the svn ver of the driver ? |
| 21:37.16 | jackfiber | what kernel modules do you load? also do you use 5.4 RELEASE? |
| 21:37.22 | MikeJ[Laptop] | was the call coming in 723 or 729? |
| 21:37.32 | paski_fr | we test |
| 21:37.49 | jackfiber | ProSLIC 3210 version 2 is too old |
| 21:37.49 | jackfiber | ProSLIC sanity check failed |
| 21:37.53 | jackfiber | I got those |
| 21:37.53 | paski_fr | G711 and G729 OK |
| 21:38.17 | Darwin35 | not seen that before |
| 21:38.31 | Darwin35 | looking at the mailing list now for those issues |
| 21:38.33 | jackfiber | Darvin NO I used zaptel from /usr/ports/misc/zaptel |
| 21:38.37 | Darwin35 | not finding anything |
| 21:38.39 | jackfiber | should I use SVN one? |
| 21:38.46 | Darwin35 | get the svn |
| 21:38.55 | jackfiber | where? |
| 21:39.15 | Darwin35 | the freebsdasterisk site |
| 21:39.28 | Darwin35 | look in the wiki I dont have the page marked |
| 21:39.36 | Darwin35 | mine svn every night |
| 21:40.16 | Darwin35 | http://www.voip-info.org/wiki-Asterisk+FreeBSD |
| 21:40.21 | jackfiber | this: ? https://svn.bluezbox.com/repos/zaptel-bsd |
| 21:40.45 | jackfiber | so you don't use the zaptel from port collection? |
| 21:41.00 | *** join/#asterisk sudoer (~toy@c-24-60-183-102.hsd1.ma.comcast.net) |
| 21:42.03 | sudoer | can someone help me with fwd, it has suddenly stopped working, I am connecting via * and iax, there is no connection at all to fwd, and there are no failure or error messages at all, but when I can my fwd number, it says the number is busy |
| 21:42.14 | Darwin35 | http://www.voip-info.org/tiki-index.php?page=FreeBSD+zaptel |
| 21:42.32 | *** join/#asterisk foo8ar (~foo8ar@81.233.231.129) |
| 21:43.09 | DarthClue | sudoer: what do you see when you do 'iax2 show registry' on the cli? |
| 21:43.24 | jackfiber | thanks Darwin, should I use nightly snapshot or current? |
| 21:43.32 | sivana | tzanger: ping |
| 21:43.56 | sudoer | 65.39.205.121:4569 fwdnumber <Unregistered> 60 Request Sent |
| 21:44.25 | Darwin35 | svn co --username svn --password svn https://svn.bluezbox.com/repos/zaptel-bsd |
| 21:44.25 | zoo | What do I do to make voipbuster/iax not say this? Call rejected by 213.61.187.146: No authority found |
| 21:44.32 | Darwin35 | that one |
| 21:44.58 | sudoer | DarthClue: what can I do? I can't see any error messages at all regarding fwd |
| 21:45.03 | foo8ar | Zyxel 2000w version2 and asterisk, anyone experienced? |
| 21:45.08 | sudoer | can someone call my fwd to tell me what they get? |
| 21:45.13 | DarthClue | sudoer: run 'iax2 reload' on the cli, does it register? |
| 21:45.50 | *** join/#asterisk gordonjcp (~gordonjcp@cpc3-broo2-5-0-cust232.renf.cable.ntl.com) |
| 21:45.52 | jackfiber | yeah Darwin your help was very great, I found the port one is older!! and the porter does have his own location |
| 21:46.02 | sudoer | DarthClue: same thing |
| 21:46.08 | jackfiber | but Darwin, do you use asterisk from port collection or from cvsup? |
| 21:46.29 | Darwin35 | I use head and patch it |
| 21:46.35 | Darwin35 | on the make file |
| 21:46.35 | jackfiber | oh |
| 21:47.02 | Darwin35 | I set the install dir |
| 21:47.10 | jackfiber | fine seems I need to remove ports and starts my own |
| 21:47.10 | Darwin35 | but it compiles and works fine |
| 21:47.36 | Darwin35 | ports need to have a every 3 day snapshot update |
| 21:47.41 | Darwin35 | lol |
| 21:47.45 | jackfiber | the asterisk on port collection works fine however (manually mpg123 should be disabled) but not zaptel driver |
| 21:47.49 | DarthClue | sudoer: it sounds like fwd isn't replying or the reply isn't being received. iax2 show registry should show Registered. |
| 21:48.02 | sudoer | DarthClue: is there a way to tell if my port is being blocked? |
| 21:48.12 | Darwin35 | and make bristuff a _DWITH |
| 21:48.15 | sudoer | my isp seems to have been blocking ports lately |
| 21:48.28 | Darwin35 | so it does not make it automaticly |
| 21:49.06 | DarthClue | try calling me at IAX2/guest@70.244.228.13 and see if it gets thru. |
| 21:49.36 | Darwin35 | now that I know these phones work I can bundle them with te embedded system I built yesterday for a good cheap soho solution |
| 21:49.52 | sudoer | oh, the server is on the public internet, i am not actually there |
| 21:50.45 | DarthClue | the server where the output came from? if so, then it would be a port block at the server and not at your location. |
| 21:51.50 | sudoer | DarthClue: yeah, I mean asterisk is on the internet at a isp |
| 21:52.39 | *** join/#asterisk zotz (~zotz@24.231.36.100) |
| 21:54.28 | DarthClue | then you should be able to add a dial command on the asterisk box that will dial me from the asterisk box. if you can't connect via iax then it would seem to indicate that they may be blocking the port. alternatively, you could just call them or run a tracedump to see if it is blocked. |
| 21:55.17 | sudoer | oh ok |
| 21:55.47 | sudoer | DarthClue: wait, I just forgot, I am connected to nufone fine though\ |
| 21:55.52 | sudoer | so it cant be port blocking, right? |
| 21:56.01 | sudoer | also how do I dial you from CLI? |
| 21:58.05 | DarthClue | sudoer: via IAX? if so, then it is an issue with fwd and fwd isn't a guaranteed service. |
| 21:59.22 | sudoer | DarthClue: yeah, but my fwd has been like this for a few months, I just give up after an hour everytime of trying to find thre problem and I am trying to fix it this time |
| 22:00.32 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 22:02.29 | *** join/#asterisk ginvent (~joseph@adsl-67-121-208-105.dsl.sndg02.pacbell.net) |
| 22:02.38 | ginvent | Can someone help me with AMP... |
| 22:02.53 | sudoer | any ideas? |
| 22:03.12 | ginvent | I did something and now get this message in outbound routing: Warning: Missing argument 5 for addroute() in /var/www/admin/functions.php on line 1300 |
| 22:07.27 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) [NETSPLIT VICTIM] |
| 22:07.27 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 22:07.27 | *** join/#asterisk newl (~newlook@203-59-217-50.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 22:07.27 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 22:07.27 | *** join/#asterisk CdtDelta (~CdtDelta@dsl081-225-161.chi1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 22:07.39 | hardwire | ok |
| 22:07.42 | hardwire | creating ringtones can bite me |
| 22:07.53 | hardwire | I have used timidity to sample some 800hz ones |
| 22:07.55 | hardwire | and boy do they blow |
| 22:08.52 | *** join/#asterisk meppl (~mephisto@87.193.4.171) |
| 22:10.55 | ginvent | Anyone use amp a lot? |
| 22:10.58 | ginvent | I need some help. |
| 22:13.56 | Sedorox | hardwire: just upgrade your phone to a mp3 phone |
| 22:13.56 | Sedorox | :p |
| 22:14.21 | hardwire | Sedorox: eh? |
| 22:15.12 | sudoer | DarthClue: if i rconenct via another box to fwd, it registers fine, but not with the other box :( |
| 22:15.31 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 22:15.41 | Sedorox | ringtones... for a cellphone?? |
| 22:15.48 | hardwire | no |
| 22:15.53 | hardwire | for my VoIP phone |
| 22:16.38 | ManxPower | YES! The fired someone that desperatly deserved it! |
| 22:16.46 | gambolputty | ? |
| 22:16.50 | yaaar | bling! |
| 22:16.52 | gambolputty | who |
| 22:16.58 | hardwire | bling? |
| 22:17.13 | yaaar | hehehe |
| 22:17.38 | yaaar | just finished moving from gentoo's 1.0.8 to cvs, everthing just went smooth as silk |
| 22:17.46 | Sedorox | ooo ok... |
| 22:17.49 | Sedorox | nm then :p |
| 22:18.16 | hardwire | jesus the snom rings are amazingly ugly |
| 22:18.22 | hardwire | one of them sounds like ap olice siren |
| 22:18.27 | hardwire | another an evacuation siren |
| 22:18.34 | yaaar | now....the realtime wiki page is sorta light......is this stuff all included in the stock cvs? or do i need to go get stuff from someplace else? maybe somebody can point me to a better doc that explains how to set it up> |
| 22:18.35 | yaaar | ? |
| 22:18.43 | hardwire | theres one thats really made for making sure somebody picks up the fucking phone right goddamn now |
| 22:19.28 | hardwire | one of them sounds like headers for a slow scan fax transmission |
| 22:19.29 | ManxPower | gambolputty: Some woman, that instead of doing work, spent most of the day on bridal and baby sites. |
| 22:19.34 | hardwire | another is psk31 |
| 22:19.37 | hardwire | just.. amazingly bad |
| 22:19.44 | ManxPower | She also got her PC infected with Spyware every few days. |
| 22:20.09 | twisted[asteria] | mahna mahna |
| 22:20.24 | gambolputty | strange |
| 22:20.40 | hardwire | ManxPower: heh |
| 22:20.42 | hardwire | so she is pregnant? |
| 22:20.49 | ManxPower | twisted: this is the same woman that remarried her ex-husband. |
| 22:20.50 | gambolputty | the snom phone can have a new ringtone |
| 22:20.56 | hardwire | gambolputty: indeed |
| 22:20.57 | gambolputty | instead of the ones you don't linke |
| 22:20.58 | twisted[asteria] | ManxPower, huh? |
| 22:20.59 | ManxPower | hardwire: I would assume so. |
| 22:21.01 | hardwire | I have come up with 5 new ones so far |
| 22:21.10 | yaaar | ginvent: you can check in #amportal. i'm in there too and maybe (not just terribly likely) can help |
| 22:21.15 | ManxPower | twisted: she got divorced from the guy, then about a year later married him again. |
| 22:21.16 | hardwire | ManxPower: hope you are sheilded from the forces of pregnant revenge :) |
| 22:21.23 | hardwire | Bananaphone.. heh. |
| 22:21.25 | twisted[asteria] | <PROTECTED> |
| 22:21.28 | twisted[asteria] | ManxPower, i'm totally lost |
| 22:21.35 | hardwire | I just want a.. "Boooooong" |
| 22:21.37 | hardwire | simple |
| 22:21.39 | ManxPower | twisted: the woman that they fired today at my largest client. |
| 22:21.39 | hardwire | percussive |
| 22:21.41 | hardwire | happty |
| 22:21.49 | ManxPower | She was nice, but caused massive amount of work for MIS |
| 22:21.49 | twisted[asteria] | ManxPower, oh... okay.... |
| 22:22.01 | ManxPower | and she DID no work. |
| 22:22.04 | yaaar | hardwire: i know where you can get a pretty good bong..... |
| 22:22.11 | twisted[asteria] | sorry, i just interjected a mahna mahna and got jerked into a conversation i didn't know anything about ;P |
| 22:22.28 | hardwire | yaaar: oh ha |
| 22:22.42 | twisted[asteria] | i got a pocket bong |
| 22:23.13 | twisted[asteria] | hahaha |
| 22:23.25 | ManxPower | "Is that a bong in your pocket, or are you happy to see me?" |
| 22:24.36 | *** join/#asterisk Cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 22:26.23 | *** part/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net) |
| 22:26.56 | *** part/#asterisk jackfiber (~jack@66.96.209.21) |
| 22:31.15 | hardwire | wow KDE has some crazy ass soudns |
| 22:31.22 | *** join/#asterisk pbnj (~pwinkeler@69-171-130-62.clvdoh.adelphia.net) |
| 22:32.48 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 22:33.04 | yaaar | hey guys....in the asterisk-addons README, when it says "Using res_config_mysql at the same time as res_config_odbc can create system instability. Please load only one or the other" ...is that as simple as just making sure you only have one or the other of the .conf files for those in /etc/asterisk? or do i need to edit something else to make sure that the actual modules don't get loaded? |
| 22:34.08 | DarthClue | yaaar: you probably want to noload one or the other. |
| 22:34.31 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 22:35.17 | *** join/#asterisk wunderkin (kev@24.137.147.163) |
| 22:36.57 | *** join/#asterisk zoo (nobody@ip-36-16.travedsl.de) |
| 22:37.10 | yaaar | in modules.conf |
| 22:37.12 | yaaar | ? |
| 22:39.04 | yaaar | well, actually it looks like there's no res_config_odbc.so in /usr/lib/asterisk/modules anyway, so i guess it's probably moot |
| 22:41.29 | DarthClue | yaaar: yes, probably so. |
| 22:42.16 | hardwire | hey |
| 22:42.17 | hardwire | neat |
| 22:42.18 | hardwire | sip disabled |
| 22:42.20 | hardwire | in my snom |
| 22:42.23 | hardwire | isn't that nice? |
| 22:42.31 | hardwire | the 4.0 firmware can now go to hell |
| 22:43.03 | *** join/#asterisk laserfox (~jimbob@81-179-127-14.dsl.pipex.com) |
| 22:43.34 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 22:43.56 | *** part/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 22:43.58 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
| 22:45.47 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net) |
| 22:46.35 | *** part/#asterisk trickyrick77 (~rsegrest@207.111.174.1) |
| 22:46.35 | yaaar | can i only make one mysql connection to a given sock file? i'm just wondering, because i'm getting 'couldn't connect to db cdrdb on localhost' and i've got the same sock file specified for both the cdr and config db's |
| 22:47.15 | hardwire | hmmphmmphmm |
| 22:47.19 | hardwire | who where has snom 360's ? |
| 22:48.05 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 22:49.27 | obsidian-studios | does anyone have suggestions about how to go about integrating credit card payments via *? I use the Monetra CC engine http://www.mainstreetsoftworks.com/, basically a API engine. I am thinking about writing a small program in C or etc and using the system commands of *. But I am not sure about getting data back into * or etc? |
| 22:50.13 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
| 22:51.02 | ManxPower | BTW, we solved the "zaptel card driver hardlocks the system when unloaded" |
| 22:51.13 | ManxPower | We disabled hyperthreading and used a non-SMP kernel |
| 22:51.43 | obsidian-studios | ManxPower: did you get my message to you earlier? Still seeing zap channel info on CID/ANI, not as often but it happened instead of getting unknown caller and the #? |
| 22:53.52 | *** join/#asterisk kimosabe (~kimosabe@216.60.60.103) |
| 22:54.11 | kimosabe | what is the best fxo gateway that will interact with asterisk |
| 22:54.33 | kimosabe | or best device that has several fxo and will interact with asterisk |
| 22:55.18 | dudes | te110p and a channel bank ? |
| 22:56.26 | ManxPower | always T-1/E-1 card + channel bank. |
| 22:56.53 | dudes | better yet, a sangoma A102 and a channel bank |
| 22:57.02 | obsidian-studios | kimosabe: if you are asking about a card typically the Digium cards like a TDM400 with either 4 fxo ports or 4 fxs |
| 22:59.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 22:59.59 | yaaar | hrm. anybody wanna lend a hand with realtime? i can't get it to connect to the database for some reason. |
| 23:00.14 | yaaar | it'll connect to to the cdr database, but not the realtime one. |
| 23:00.22 | *** join/#asterisk hypa7ia (~leigh@fcf7010d2327670a.session.tor) |
| 23:00.38 | dudes | are you using sql? |
| 23:00.41 | yaaar | both databases are on the same mysql server, which is local, and connect with the same user/pass, which has grant for everything |
| 23:01.16 | yaaar | the asterisk database (the one for configs) is empty so far.....do i have to have stuff in it before it'll admit to making a connection? |
| 23:01.18 | fugitivo | check your mysql logs to see if asterisk is trying to connect or not |
| 23:01.41 | *** join/#asterisk Paski (~paski_fr@stardust.noc.frontier.fr) |
| 23:02.53 | dudes | yaar - so you did grant select, insert, update, delete on realtime_table.* to asterisk_user@localhost identified by 'password'; |
| 23:03.27 | yaaar | fugitivo: hrm...no real help. the mysql logs are effectively empty, and i know that it's making successful connections to the cdr database, because it's putting cdr's in there |
| 23:03.47 | yaaar | dudes: yes, the asterisk mysql user has full privs on both databases. |
| 23:03.54 | dudes | yaaar - you have set logs in my.cnf in /etc/mysql |
| 23:04.27 | fugitivo | yaaar: in order to find out the problem, you should have useful logs, if not, it'll be difficult and a waste of time |
| 23:04.36 | dudes | log-bin = /var/log/mysql/mysql-bin.log |
| 23:04.38 | yaaar | yeah |
| 23:04.45 | yaaar | log-bin is what i want? |
| 23:04.50 | yaaar | i've got log-err but that's it |
| 23:05.32 | dudes | I have /var/log/mysql/mysql-bin and mysql.log and in just /var/log/mysql.err |
| 23:08.24 | yaaar | i think i may have found the prob |
| 23:09.24 | yaaar | yep |
| 23:09.32 | yaaar | socket file was wrong.... |
| 23:09.36 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 23:09.47 | yaaar | apparently cdr_mysql and res_mysql have different default socket files |
| 23:10.33 | yaaar | cdr_mysql seems to default to /var/run/mysqld/mysqld.sock (which is correct on my system) while res_mysql defaults to /tmp/mysql.sock |
| 23:10.44 | yaaar | so, changed that and now it seems to have loaded. |
| 23:11.08 | *** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net) |
| 23:26.31 | *** join/#asterisk Error_X (~Error_X@217-131-211.5001.adsl.tele2.no) |
| 23:26.46 | Error_X | is it possible to set up meetme with a fwd number? |
| 23:27.07 | Error_X | I tried but it says: That is not a valid conference number |
| 23:28.22 | hardwire | you still at it eh? |
| 23:28.37 | Error_X | yes? |
| 23:28.42 | Nugget | try harder. |
| 23:28.45 | hardwire | use pastebin.ca and give me your extensions.conf and meetme.conf |
| 23:28.50 | Error_X | k |
| 23:29.01 | hardwire | and your root password while you are at it |
| 23:29.12 | Error_X | okey :p |
| 23:31.59 | Error_X | http://pastebin.ca/18663 |
| 23:32.07 | Error_X | there ya go |
| 23:33.28 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:33.34 | Ariel_ | hello everyone |
| 23:33.49 | Delta34 | anybody know what why i am getting this msg when starting asterisk Cannot allow unknown format 'wav |
| 23:34.26 | *** join/#asterisk SwK_ (hjeigx@12-219-156-206.client.mchsi.com) |
| 23:34.28 | Error_X | hardwire: Are you reading? :p |
| 23:34.30 | SwK_ | j #redbull |
| 23:34.54 | *** join/#asterisk _santiago_ (~santiago@63.245.86.175) |
| 23:34.59 | hardwire | Error_X: I am working |
| 23:35.06 | Error_X | ok |
| 23:35.09 | hardwire | I will be reading soon |
| 23:35.14 | Error_X | k, thanks |
| 23:35.20 | ManxPower | Delta34: you don't have allow=wav or something silly like that in one of your .conf files, do you? |
| 23:35.34 | Delta34 | yeh i do, i greped for it and found it =) |
| 23:35.37 | Delta34 | doh |
| 23:35.47 | Delta34 | its not needed in sip.conf right? |
| 23:36.07 | hardwire | Error_X: tried it w/o a pin? |
| 23:36.12 | Ariel_ | Delta34, it's not a codec no it's not needed |
| 23:36.18 | Delta34 | thxs |
| 23:36.23 | ManxPower | "wav" is not a valid CODEC name. It's a file format name. |
| 23:38.11 | hardwire | SwK: ? |
| 23:38.27 | ManxPower | WAV is ulaw (maybe supports alaw too), wrapped in a Microsoft header. WAV49 is GSM wrapped in a Microsoft header. |
| 23:39.01 | SwK_ | wut? |
| 23:39.16 | Error_X | hardwire: What? |
| 23:39.29 | hardwire | Error_X: tried setting it to just 6666 in meetme.conf |
| 23:39.31 | hardwire | no pin needed |
| 23:39.35 | hardwire | just to test |
| 23:39.44 | Error_X | yes, I tried out that too |
| 23:39.52 | hardwire | and it says no such conference eh? |
| 23:40.07 | Error_X | no it says: This is not a valid conference number |
| 23:40.11 | hardwire | try using MeetMe(6666) in your dialplan.. vs MeetMe,$(blah) |
| 23:40.20 | SwK_ | ${blah} |
| 23:40.27 | Error_X | Tried that too |
| 23:40.53 | hardwire | ok |
| 23:40.56 | hardwire | then you are screwed |
| 23:41.03 | hardwire | sucks to be you I guess |
| 23:41.19 | Error_X | Hmm, I thought I couldn't use the FWD number for meetme |
| 23:41.25 | Error_X | or something like that |
| 23:41.27 | hardwire | no |
| 23:41.38 | hardwire | asterisk won't do crap like that |
| 23:41.41 | Error_X | hehe |
| 23:41.47 | hardwire | how would FWD know? |
| 23:41.54 | hardwire | why would Mr. Pulver even care? |
| 23:41.54 | SwK_ | you can use a FWD number for meetme... |
| 23:41.56 | Error_X | right, right... |
| 23:41.57 | Error_X | :P |
| 23:41.58 | Error_X | hehe |
| 23:41.59 | SwK_ | you just need a timing source |
| 23:42.03 | hardwire | hey |
| 23:42.04 | Error_X | yes |
| 23:42.05 | hardwire | that could be it |
| 23:42.14 | hardwire | and now he is on his own again |
| 23:42.19 | Error_X | I know. But I can't compile ztdummy :s it comes with alot of errors |
| 23:42.25 | hardwire | ah |
| 23:42.28 | hardwire | then thats why it doesn't work |
| 23:42.33 | SwK_ | if you dont have a zap card (any zap card) use a 2.6 kernel and ztdummy |
| 23:42.53 | SwK_ | Error_X: ztdummy should compile just fine |
| 23:43.09 | SwK_ | unless you have a 2.4 kernel and the wrong USB hardware |
| 23:43.28 | Error_X | Oh, I'm using a 2.4 kernel :s |
| 23:44.07 | hardwire | I hate my job |
| 23:44.20 | hardwire | in the middle opf configuring a mass deployment of phones |
| 23:44.28 | hardwire | and now I have to buy a laptop for some dick and fix another one for another |
| 23:44.30 | Error_X | hehe |
| 23:44.45 | Error_X | btw, what ports are asterisk running on? |
| 23:44.56 | hardwire | check iax.conf sip.conf and rtp.conf |
| 23:45.02 | Error_X | k |
| 23:45.11 | hardwire | and most importantly (www.voip-info.org) the wiki |
| 23:45.28 | *** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net) |
| 23:46.47 | *** join/#asterisk Nukemizer (~Nuke@67.137.27.114) |
| 23:48.06 | SwK_ | IAX is 4569 SIP is 5060 and RTP is 10K to 20K by default |
| 23:48.08 | *** join/#asterisk datagen24 (~steve+mir@londonderry-cuda1-68-171-193-26.lndnnh.adelphia.net) |
| 23:48.53 | datagen24 | i am back i have been working on my own conf files and they do not work can i get some help? |
| 23:48.58 | hardwire | lead a horse to water SwK |
| 23:49.21 | hardwire | Error_X: might I recommend a 2.6.x kernel and Debian/Gentoo :) |
| 23:49.30 | SwK_ | yeah... but thats just those 3 things ;) he'll need to figure out skinny, h323, mgcp, and unistin on his own |
| 23:49.40 | *** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) |
| 23:49.47 | hardwire | heh |
| 23:50.04 | hardwire | SwK_: did you guys have Snom deployment php/sql setups? |
| 23:50.06 | ManxPower | SwK: You should have given him those port numbers and forced himto research SIP, IAX2 and RTP ports 8-) |
| 23:50.31 | Error_X | hardwire: I am using debian, but I forgot to upgrade the server where I have asterisk installed |
| 23:50.45 | SwK_ | hardwire: nope |
| 23:50.48 | SwK_ | we dont do snoms |
| 23:51.11 | hardwire | damn.. it was somebody else then |
| 23:51.15 | hardwire | I have no brain |
| 23:51.30 | ManxPower | At one time SNOMs were some of the best phones out there, but they have not kept up with current pricing or current technology. |
| 23:51.46 | hardwire | the 360 w/ 4.0 firmware is OK by me |
| 23:51.57 | SwK_ | Polys, Cisco's Linksys and sipura are another story tho |
| 23:51.57 | hardwire | it could use a few less star trek ring tones |
| 23:52.02 | hardwire | and oh.. an LCD that doesn't suck |
| 23:52.36 | *** join/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 23:52.53 | PyroSteve | HEY GUYS |
| 23:52.56 | PyroSteve | OPPS |
| 23:52.58 | PyroSteve | sorry |
| 23:52.59 | SwK_ | the GXP-2000 grandstream actually has a nice LCD |
| 23:53.02 | SwK_ | backlit even |
| 23:53.02 | ManxPower | For any deployment you should standardize. We standardized on Polycom phones. |
| 23:53.04 | PyroSteve | a littl drunk here |
| 23:53.08 | PyroSteve | sorry bout the caps |
| 23:53.12 | SwK_ | i wish poly's had that |
| 23:53.24 | SwK_ | and yeah wnat ManxPower said about standardizing |
| 23:53.28 | PyroSteve | ManxPower: Yeah, I love the Polycoms ! |
| 23:53.47 | PyroSteve | hey guys |
| 23:53.50 | SwK_ | only reason we support all those phones is we're a Vendor/Development house |
| 23:53.55 | PyroSteve | im looking at the channel variables |
| 23:53.58 | ManxPower | We could have standardized on Cisco or SNOM as well. We decided that Polycom had the best product lineup for our needs. |
| 23:54.05 | PyroSteve | and I see the dial status variables |
| 23:54.13 | Cybertoy | standardizing is good |
| 23:54.22 | PyroSteve | and have this example code from he wiki: (just a few lines) |
| 23:54.23 | PyroSteve | exten => s,1,Dial(${ARG2},20,r) |
| 23:54.24 | PyroSteve | <PROTECTED> |
| 23:54.24 | PyroSteve | <PROTECTED> |
| 23:54.24 | PyroSteve | <PROTECTED> |
| 23:54.24 | PyroSteve | <PROTECTED> |
| 23:54.25 | PyroSteve | <PROTECTED> |
| 23:54.25 | Cybertoy | the nice thing about standards is that there's so many to choose from .. :) |
| 23:54.32 | yaaar | catch you guys tomorrow... |
| 23:54.41 | ManxPower | PyroSteve: Please put the beer down, step away from the computer and use PASTEBIN |
| 23:54.43 | ManxPower | ~pastebin |
| 23:54.43 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 23:54.57 | PyroSteve | well im not flooding the channel.. just a few lines |
| 23:55.09 | hardwire | ManxPower: I was at 50% full steam ahead w/ polycoms |
| 23:55.37 | hardwire | there should be a pastebin bot. |
| 23:55.54 | hardwire | where you msg it.. and after a few seconds of inactivity it scrounges up a URL |
| 23:55.55 | ManxPower | PyroSteve: Did you have a specific question? |
| 23:56.00 | SwK_ | PyroSteve: look at macro-stdexten in the sample dialplan |
| 23:56.03 | PyroSteve | so how does the s-BUSY exten ever get excuted if there isn't a 'j' option in the dial statement |
| 23:56.33 | ManxPower | PyroSteve: If Dial wants to jump to n+101 and there is no n+101 it will got ot n+1 |
| 23:56.35 | PyroSteve | if the call is busy, the asterisk will jump to n+101 which skips the s-WHATEVER |
| 23:56.46 | PyroSteve | Ahhh !!! |
| 23:56.56 | PyroSteve | thanks ManxPower ! |
| 23:57.17 | PyroSteve | at priority n+101, if it exists |
| 23:57.27 | PyroSteve | keyword, if it exists' |
| 23:57.30 | PyroSteve | hehe |
| 23:57.33 | SwK_ | and jumps are depreciated BTW |
| 23:57.43 | PyroSteve | really ? |
| 23:57.48 | datagen24 | i am back i have been working on my own conf files and they do not work can i get some help? |
| 23:57.58 | datagen24 | here are y .conf files |
| 23:57.58 | datagen24 | http://pastebin.ca/18667 |
| 23:58.48 | Ariel_ | SwK, the jumps are depreciated?? why? what's replacing them? |
| 23:58.53 | SwK_ | it's going the exten,1,foo exten,n(bar),foobar way and you can just to n(bar) with goto(bar) |
| 23:59.20 | MicC_ | I am pleasantly suprised |
| 23:59.29 | SwK_ | jumps as in n+101 type things |
| 23:59.38 | MicC_ | my VOIP is working well over a craptacular Microsoft VPN |
| 23:59.46 | SwK_ | hah |
