| 00:00.01 | seanbright | 9 to 0... i'd still bet $10 that we end up tanking it |
| 00:01.13 | EvilRick | not sure where to see the version.. why is your nick familiar? |
| 00:01.49 | EvilRick | 2.4.0.1 core |
| 00:02.34 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-184-133.dsl.stlsmo.sbcglobal.net) |
| 00:04.10 | drmessano | Trixbox sucks |
| 00:05.12 | russellb | jbot: drmessano++ |
| 00:05.35 | drmessano | yay |
| 00:06.07 | drmessano | jjshoe works for Fonality, EvilRick.. you've probably swam in his "pool" |
| 00:06.36 | drmessano | Feel free to badger him with questions about RPMs and "Things in the color green" |
| 00:06.40 | EvilRick | I think we spoke about atheros stuff |
| 00:10.06 | jjshoe | EvilRick I've never used ce, and nope, no atheros stuff from me. |
| 00:11.57 | jjshoe | EvilRick what you want to do is about two clicks in pro, no idea on ce :\ |
| 00:14.19 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 00:17.04 | EvilRick | is there an upgrade path for pro or must I re-install? |
| 00:18.04 | Qwell | reinstall. another OS. |
| 00:18.54 | EvilRick | ouch.. not really an option then. I have no physical access to the boxes.. they are upcountry like 1300km away |
| 00:19.27 | drmessano | 1300km is not too far to drive to rid the world of another trashbox |
| 00:19.34 | drmessano | Maybe someone here will help |
| 00:19.42 | Qwell | I'd help, if you paid for travel expenses. |
| 00:20.04 | Qwell | drmessano: I mean, of course, if *you* paid. |
| 00:20.26 | drmessano | Not only would I reformat the box, but I would be glad to pee on any Trixbox install media you may have lying around |
| 00:20.29 | drmessano | Free of charge |
| 00:21.13 | drmessano | Consider it my contribution to "open sores" |
| 00:23.06 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:25.08 | EvilRick | hwats dundi? |
| 00:25.25 | seanbright | ~dundi |
| 00:25.26 | jbot | methinks dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 00:26.17 | seanbright | for the record, i don't know the answers to any questions... i just delegate to jbot for everything |
| 00:27.05 | EvilRick | sounds communist. |
| 00:27.12 | EvilRick | needs to be destroyed |
| 00:29.08 | hardwire | jbot or sean? |
| 00:29.09 | jbot | it has been said that sean is you name the original way of spelling it or mine( by the way its shaun for those of you who dont know) |
| 00:29.28 | hardwire | heh |
| 00:29.35 | seanbright | wow... that is the least coherent jbot message... EVER |
| 00:29.37 | hardwire | freak |
| 00:30.02 | seanbright | jbot, forget sean |
| 00:30.08 | hardwire | jbot or sean? |
| 00:30.08 | jbot | hmm... sean is you name the original way of spelling it or mine( by the way its shaun for those of you who dont know) |
| 00:30.18 | hardwire | jbot sean? |
| 00:30.19 | jbot | well, sean is you name the original way of spelling it or mine( by the way its shaun for those of you who dont know) |
| 00:30.29 | hardwire | jbot forget sean |
| 00:30.29 | seanbright | jbot: no, sean is just a name |
| 00:30.30 | jbot | okay, seanbright |
| 00:30.32 | hardwire | haha |
| 00:30.47 | hardwire | jbot no sean |
| 00:31.04 | seanbright | jbot: sean |
| 00:31.04 | jbot | sean is probably just a name |
| 00:31.11 | seanbright | good enough |
| 00:32.11 | EvilRick | jbot: whats your name? |
| 00:32.14 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-184-133.dsl.stlsmo.sbcglobal.net) |
| 00:32.27 | hardwire | jbot sean |
| 00:32.27 | jbot | i guess sean is just a name |
| 00:32.34 | hardwire | jbot shane |
| 00:32.34 | jbot | well, shane is confused. or enahs spelled backwards. |
| 00:32.40 | hardwire | thanks for noticing |
| 00:32.52 | hardwire | is enahs recneps |
| 00:32.58 | LemensTS | Anyone want to install asterisk prepaid/postpaid solution on gentoo? |
| 00:33.13 | hardwire | LemensTS: will you prepay or postpay? |
| 00:33.20 | LemensTS | halfpay |
| 00:33.26 | hardwire | upfruntay? |
| 00:33.57 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7c4ab26b4590b31b) |
| 00:33.57 | LemensTS | Yea |
| 00:34.05 | hardwire | pissiblay |
| 00:34.09 | hardwire | possiblay |
| 00:34.17 | hardwire | I don't even know what that product is, is it pre-built? |
| 00:34.53 | hardwire | you asked for "the" asterisk solution not "a" asterisk solution. |
| 00:35.02 | hardwire | I'm not "the" tax man, I'm "a" tax man. |
| 00:35.03 | LemensTS | There are alot of them out thier... |
| 00:35.22 | LemensTS | i dont see a "the" in my writings |
| 00:35.35 | hardwire | you implied "the" by not noting "a" |
| 00:35.57 | LemensTS | anyone else? |
| 00:35.58 | LemensTS | lol |
| 00:36.14 | wichitalineman | i'm found of "an asterisk solution" |
| 00:36.20 | hardwire | Sorry, I'm in friday mode. |
| 00:36.39 | LemensTS | wichitalineman: huh? |
| 00:37.02 | wichitalineman | LemensTS: nevermind me... i'm being a gramatical dick |
| 00:37.04 | hardwire | wichitalineman: but pointing out "an" instead of "a" reduces the context to only things that start with a vowel or are an acronym. |
| 00:37.26 | wichitalineman | hardwire: you have a point to make there |
| 00:37.38 | hardwire | I feel as though that's actually a question. |
| 00:37.49 | hardwire | I have no real point, ever. |
| 00:38.10 | hardwire | LemensTS: privmsg'ed you.. if you can put up with me I can probably do the work. |
| 00:38.31 | *** join/#asterisk techie (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
| 00:38.33 | wichitalineman | i'm just waiting for all the * crap i've tried to absorb presents a/an/the solution. i'll take any of them |
| 00:39.12 | wichitalineman | i'd ask for help but i'd only expose myself as a fraud or a lamer |
| 00:39.35 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:40.08 | seanbright | jbot, sean |
| 00:40.09 | jbot | rumour has it, sean is an irish derivative of the name john. other derivatives include shawn, shaun, and shane. |
| 00:40.24 | seanbright | ok. that is definitive. |
| 00:41.45 | hardwire | nobody believes me that shane = john |
| 00:42.00 | hardwire | my dad, john, did it on purpose. we also share the same middle name. |
| 00:42.15 | drmessano | ~seanbright |
| 00:42.15 | jbot | somebody said seanbright was a girl with standards |
| 00:42.21 | drmessano | lol |
| 00:42.24 | seanbright | weeee! |
| 00:42.32 | drmessano | You're a female? |
| 00:42.40 | seanbright | no |
| 00:42.45 | hardwire | hubba hubba |
| 00:42.47 | drmessano | Are you hot? |
| 00:42.49 | seanbright | that is why it is funny |
| 00:42.52 | seanbright | i am, yes. |
| 00:42.56 | hardwire | I am too! |
| 00:42.58 | hardwire | we should meet. |
| 00:43.06 | seanbright | um. i'm a dude. |
| 00:43.11 | seanbright | i'll pass. |
| 00:43.12 | seanbright | heh |
| 00:43.30 | hardwire | imagines a night at the roxbury moment. |
| 00:44.08 | drmessano | I wonder how many people have met in #asterisk, fallen in love, and made babies |
| 00:44.14 | drmessano | I would guess -2 |
| 00:44.20 | *** join/#asterisk ManxPower (n=manxpowe@12.sub-70-220-153.myvzw.com) |
| 00:44.30 | seanbright | seems like a fair guestimate |
| 00:44.42 | seanbright | considering i've seen exactly *1* female in this channel ever |
| 00:44.56 | seanbright | and by "seen" i mean "saw someone with a lady name" |
| 00:45.36 | drmessano | There's lots of females on IRC.. with male nicknames.. who could care less about your anatomy |
| 00:45.44 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
| 00:45.48 | drmessano | They know better |
| 00:45.50 | techie | MR TOTARO! |
| 00:46.18 | SteveTotaro | yes sir! |
| 00:46.23 | techie | :) |
| 00:46.32 | drmessano | <SomeFemale> Can someone help me with IAX2? |
| 00:46.48 | drmessano | <EveryUdderGuy> Sure, if I can have your IAX URI |
| 00:47.04 | drmessano | <EveryUdderGuy> Was that too forward? MARRY ME |
| 00:47.13 | drmessano | * SomeFemale has left IRC |
| 00:48.14 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:48.36 | SteveTotaro | what's up DRM? |
| 00:48.47 | SteveTotaro | still at the new jobby job? |
| 00:48.54 | drmessano | Just Digitially Righting Management |
| 00:48.55 | SteveTotaro | not new anymore i guess |
| 00:49.14 | SteveTotaro | sounds like grounds for termination |
| 00:49.19 | drmessano | Yep... All the newness has wore off, now I am just a drone |
| 00:49.31 | SteveTotaro | got asterisk in there yet? |
| 00:50.04 | drmessano | No, need to get them pissed off more with Cisco first |
| 00:50.09 | drmessano | "Oh, it cant do that?" |
| 00:51.11 | SteveTotaro | man i have been messing with this http://www.vmukti.com for a couple days, including a fresh M$ 2003 server, SQL2005, and IIS |
| 00:51.29 | SteveTotaro | hey DAHDI |
| 00:51.41 | DAHDI | Hello |
| 00:52.16 | SteveTotaro | you have a strange name, somehow associated with Adtran, Digium, and Iraq..... |
| 00:52.27 | SteveTotaro | WMD!!!! |
| 00:53.07 | SteveTotaro | oh, my mistake it is Iran |
| 00:53.26 | SteveTotaro | the next (oil rich) country that is a threat to our national security |
| 00:54.11 | DAHDI | gasoline cost $1.15 in Mexico man |
| 00:54.19 | DAHDI | who are they kidding!! |
| 00:54.20 | TrentCreek | that was a good movie |
| 00:54.21 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 00:54.23 | drmessano | DAHDI is a way for Digium to make money off people who already bought Zaptel |
| 00:54.28 | drmessano | It's all a scam |
| 00:54.37 | drmessano | First you pay for Zaptel |
| 00:54.40 | drmessano | Then, they change the name |
| 00:54.44 | drmessano | You pay for it again |
| 00:54.47 | TrentCreek | gasoline doe NOT cost 1.15 in Mexico |
| 00:54.48 | drmessano | I R NUT FOOLED |
| 00:55.19 | SteveTotaro | but there is no reason for the gas prices to be as high as they are |
| 00:55.23 | drmessano | TrentCreek: Everyone thinks the US has high gas prices.. let them have their moment |
| 00:55.29 | SteveTotaro | where is the grand jury investigation? |
| 00:55.42 | DAHDI | trentcreek: it does in Veracruz |
| 00:56.04 | TrentCreek | I live 20 miles from Mexico and go there a lot |
| 00:56.10 | SteveTotaro | why did exxon post the largest PROFIT of any company in history last year |
| 00:56.25 | SteveTotaro | ever go to the donkey show in tijuana? |
| 00:56.29 | DAHDI | well, tijuana and border cities may be a little different of course |
| 00:56.56 | TrentCreek | I dont always go to border cities. My wife lives in south central Mexico |
| 00:57.38 | SteveTotaro | that was the best $1.50 i ever spent |
| 00:57.39 | TrentCreek | While it is cheaper than in the US...I wish it was only $1.12 |
| 00:57.40 | drmessano | TrentCreek: Please stop debunking internet myths.. Your truth has no place here |
| 00:57.48 | TrentCreek | hehe |
| 00:58.14 | drmessano | TrentCreek: Gas in mexico is 59 cents a gallon or you don't get help with your asterisk problem, got it? |
| 00:58.21 | SteveTotaro | wonder what the price is in hawii |
| 00:58.37 | Qwell | high |
| 00:58.45 | SteveTotaro | hello Qwell |
| 00:58.47 | drmessano | eats his neiman marcus cookie |
| 00:58.47 | Qwell | they don't do much driving there, for obvious reasons |
| 00:59.10 | SteveTotaro | donkeys are good for more than just the donkey show |
| 00:59.11 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 00:59.31 | TrentCreek | gosh darn it |
| 00:59.32 | SteveTotaro | oh in hawaii |
| 01:00.02 | SteveTotaro | when i was there in ~98 it was $5 or $6 bucks a gallon |
| 01:00.06 | TrentCreek | it was 49 cents a gallon..no wonder there is always a long line of US plates going back into the US |
| 01:00.29 | SteveTotaro | it's for the donkey shows |
| 01:00.44 | SteveTotaro | and cheap meds |
| 01:00.47 | drmessano | TrentCreek: According to the left wing extremists, gasoline in the US is higher than anywhere else in the world.. While it SHOULD cost like 3 cents a gallon, we're getting screwed |
| 01:01.07 | drmessano | TrentCreek: So just go with it |
| 01:01.16 | SteveTotaro | just pump iraq dry already! |
| 01:01.30 | TrentCreek | <PROTECTED> |
| 01:01.52 | SteveTotaro | West africa there were no working gas stations |
| 01:02.09 | SteveTotaro | guys just sold it by the liter in water bottles, they colored it red |
| 01:02.19 | drmessano | TrentCreek: That's not possible.. I read on the internet that gas in the UK is so cheap, and with pound to dollar conversion rates, the oil companies pay THEM for gas |
| 01:02.22 | SteveTotaro | get a bad batch and the car runs rough |
| 01:02.34 | TrentCreek | lol |
| 01:02.49 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 01:03.27 | SteveTotaro | quiet about stealing the canadian's gas |
| 01:04.13 | drmessano | I need some truth.. off to watch O'Riley |
| 01:04.15 | SteveTotaro | i am going to import 10,000 locking gas caps from china |
| 01:04.25 | SteveTotaro | BBC news |
| 01:04.29 | Qwell | this is incredibly off-topic. |
| 01:04.30 | SteveTotaro | or Lou Dobbs |
| 01:04.48 | drmessano | Qwell: It's also complete BS |
| 01:04.51 | rob0 | J.R. "Bob" Dobbs |
| 01:04.59 | drmessano | Qwell: Want some $25 a gallon Asterisk? |
| 01:05.11 | *** join/#asterisk codehaxor (n=info@124.6.153.226) |
| 01:05.20 | SteveTotaro | nah, i will take the "opensource" version |
| 01:05.40 | drmessano | I'm not paying for an upgrade to Dahdi |
| 01:05.48 | DAHDI | come on! |
| 01:05.57 | SteveTotaro | it should be PAHDI |
| 01:05.59 | drmessano | Zaptel will work just fine |
| 01:06.10 | SteveTotaro | it's my PAHDI and i'll cry if i want to |
| 01:06.23 | drmessano | Who's your Dahdi? |
| 01:06.37 | SteveTotaro | sangoma |
| 01:06.43 | drmessano | Oh god |
| 01:07.20 | outtolunc | hands out pink shirts |
| 01:07.30 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 01:07.37 | drmessano | Yay.. another tornado |
| 01:07.59 | SteveTotaro | really, where? |
| 01:08.11 | drmessano | Baseball size hail earlier, tornadoes all over here |
| 01:08.20 | SteveTotaro | duck and cover |
| 01:08.24 | drmessano | About 10 miles from her |
| 01:08.26 | drmessano | About 10 miles from here |
| 01:08.39 | drmessano | We're about to get slammed |
| 01:08.40 | SteveTotaro | that is scary stuff |
| 01:09.02 | SteveTotaro | we get hurricaines more than tornados |
| 01:09.14 | drmessano | My wife got hail damage to her car earlier |
| 01:09.17 | SteveTotaro | and they are few and far between |
| 01:11.03 | *** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com) |
| 01:13.34 | SteveTotaro | i can't wait for hillary to lose |
| 01:13.38 | SteveTotaro | what a man's man |
| 01:16.08 | SteveTotaro | is asterisk 1.4.20 stable? |
| 01:17.15 | shmaltz | SteveTotaro, as stable as Hillary for president |
| 01:17.22 | shmaltz | just kidding |
| 01:23.13 | SteveTotaro | lol |
| 01:25.29 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 01:25.45 | maqr | of all the .conf files that 'make samples' generates, how can i tell which ones are actually going to be used by asterisk? |
| 01:27.04 | [TK]D-Fender | ~book |
| 01:27.04 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:27.05 | [TK]D-Fender | ^^^^^^^^^ |
| 01:29.25 | drmessano | Why does everyone ask that |
| 01:29.36 | drmessano | Is Libpri 1.4.9 stable? |
| 01:29.54 | drmessano | Ok, no |
| 01:29.57 | drmessano | 1.4.3 |
| 01:31.52 | jaytee | what is "stable"? |
| 01:32.06 | shmaltz | drmessano, I'm using 1.2.x only, havn't yet touched 1.4 |
| 01:32.21 | jaytee | Jethro Tull wrote a song about you |
| 01:32.21 | [TK]D-Fender | jaytee, when the system has finished crashing and has hit rock bottom. |
| 01:32.40 | jaytee | [TK]D-Fender, exactly! |
| 01:33.35 | SteveTotaro | 1.2 is sort of stable |
| 01:34.05 | SteveTotaro | only one 1.4 install to make bristuff play nice with wanrouter |
| 01:34.09 | drmessano | 1.4 is worth ditching 1.2 for now |
| 01:34.27 | *** part/#asterisk DAHDI (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
| 01:34.31 | drmessano | Not only is now stable, but it's leaving 1.2 behind in enhancements |
| 01:34.43 | drmessano | .. on the core pieces |
| 01:34.48 | drmessano | Not just fluff |
| 01:36.06 | jaytee | I couldn't find any fluff, I was digging around but nothing. however in one subfolder I found some pizzaz. |
| 01:36.16 | SteveTotaro | white paper please DRM? |
| 01:36.41 | SteveTotaro | pizzaz sells |
| 01:37.46 | SteveTotaro | give me three examples of core enhancements that are not fluff |
| 01:38.03 | SteveTotaro | and i don't care about iax2 |
| 01:38.14 | SteveTotaro | burned me enough to leave it alone |
| 01:46.05 | *** join/#asterisk TheX1le (n=thex1le@cpe-76-189-235-180.neo.res.rr.com) |
| 01:55.42 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 01:56.16 | TheX1le | Hi, I am quite familiar with linux < gentoo user, I am however new to pbx's and Astrisk, I am looking to set up a server with single number, and a talking clock, so one can call it and receive the time. Can this be done with a standard modem or do i need additional hardware? |
| 01:56.51 | jameswf-home | Can it be done Yes... Can you do it? no clue |
| 01:57.39 | [TK]D-Fender | TheX1le, you need special hardware if you want to use a physical line on your premises with *. |
| 01:59.31 | TheX1le | jameswf-home: haha thanks, i do network security for a living how about a little credit |
| 02:00.00 | TheX1le | [TK]D-Fender: thanks i assume i need somthing to the affect of a voip addaptor? |
| 02:00.26 | [TK]D-Fender | TheX1le, Everything depends on exactly what you want to interface with, and how. |
| 02:00.31 | jameswf-home | I support people with 2 dozen acronyms in their signature... no one gets a free pass |
| 02:00.44 | [TK]D-Fender | TheX1le, once the call get into * there is nothing special needed. |
| 02:00.57 | TheX1le | jameswf-home: well put and consider it noted |
| 02:01.03 | jameswf-home | if any of your certs start with MC then well |
| 02:01.26 | [TK]D-Fender | "Would you like fries with that, sir?" |
| 02:01.31 | TheX1le | jameswf-home: i cant think of one use for any cert that starts with MC |
| 02:01.51 | LemensTS | msg LemensTS test |
| 02:02.06 | jameswf-home | FAIL |
| 02:02.09 | LemensTS | heh |
| 02:02.34 | TheX1le | thanks for the help its off to google i go |
| 02:02.52 | drmessano | White paper on what? |
| 02:03.10 | LemensTS | msg [TK]D-Fender hey you do asterisk consulting right |
| 02:03.13 | LemensTS | ha lol |
| 02:03.21 | [TK]D-Fender | LemensTS, yes. |
| 02:03.24 | TheX1le | LemensTS: use a / |
| 02:03.25 | jameswf-home | I should reply to this symantic email... Dear sirs I use linux and have no idea what purpose your software serves |
| 02:03.56 | LemensTS | msg [TK]D-Fender this just going to you? |
| 02:04.06 | seanbright | jesus |
| 02:04.09 | seanbright | what is wrong with you? |
| 02:04.12 | seanbright | seriously |
| 02:04.12 | FreedomBI | jameswf-home: even if I used windows, I still would have no idea what useful purpose their software serves |
| 02:04.12 | jameswf-home | heh yes none of us can see it |
| 02:04.14 | [TK]D-Fender | LemensTS, no, you DID jsut PM me before |
| 02:04.15 | seanbright | are you retarded? |
| 02:04.30 | LemensTS | Sorry never used the PM fuction before |
| 02:04.37 | LemensTS | seanbright: slightly |
| 02:04.44 | TheX1le | jameswf-home: just for you http://www.xkcd.com/272/ |
| 02:04.50 | *** join/#asterisk plla (n=h@200.31.103.86) |
| 02:05.00 | seanbright | LemensTS: well you're in good company, then. |
| 02:05.16 | jameswf-home | sean showing the tact of my mother around any "EOE protected group" |
| 02:05.41 | seanbright | msg jameswf-home wanna go private and cyber? |
| 02:05.57 | jameswf-home | msg seanbright yes |
| 02:06.09 | seanbright | msg jameswf-home sweet... you're the girl |
| 02:06.12 | drmessano | msg seanbright Man, that LemensTS guy is a moron.. but don't tell him. |
| 02:06.16 | djs26 | Hey all. Trying to install Asterisk 1.4.19.2 on a fresh install of Centos 5.1. I am getting this error - http://rafb.net/p/Q6634T53.html . I have the kernel-devel and kernel headers installed, but I am still getting the error. |
| 02:06.25 | jameswf-home | lmao from now on every comment needs to have msg <username> |
| 02:06.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:06.47 | drmessano | msg jameswf-home Why bother, I use FreeSWITCH |
| 02:06.47 | seanbright | msg drmessano no, he's retarded. and i did tell him. |
| 02:07.07 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-184-133.dsl.stlsmo.sbcglobal.net) |
| 02:07.22 | drmessano | msg seanbright SHH, retardo montoban is back |
| 02:07.23 | djs26 | Oh, I should say this is when I am trying to do the newest zaptel prior to the asterisk install. |
| 02:07.35 | djs26 | 1.4.10.1 |
| 02:08.04 | seanbright | msg drmessano ixnay on the etardoray |
| 02:08.20 | drmessano | People still use Zaptel, I converted all my crap to DAHDI before 8AM this morning |
| 02:08.50 | *** part/#asterisk Wic3edOne (n=Wic3done@cpe-72-184-246-19.tampabay.res.rr.com) |
| 02:09.06 | LemensTS | msg drmessano hey whats up |
| 02:09.20 | seanbright | o.O |
| 02:09.20 | LemensTS | m/sg dremessano hey whats up |
| 02:09.34 | seanbright | head... going... to... explode |
| 02:09.37 | jameswf-home | ~msg |
| 02:09.38 | jbot | (1) Use private messages to the bots to reduce channel spam, but don't message people on #debian without asking permission first. Most questions should be asked on channel, so that others can benefit from the question and the answers received. (2) Always feel free to message freenode network staff. They're the people with hostnames ending in 'staff.freenode'. (3) Monosodium glutamate, a food additive (see http://truthinlabeling.org/). |
| 02:09.43 | LemensTS | lmao im just messing with you guys now |
| 02:10.00 | LemensTS | before i really didnt know tho |
| 02:10.48 | seanbright | LemensTS: best way to get IRC command help is to use /server help |
| 02:11.06 | seanbright | FTW! |
| 02:11.06 | drmessano | holy effin crap |
| 02:11.20 | plla | djs26: output of "uname -r; rpm -qa | grep kernel-devel" |
| 02:11.26 | drmessano | seanbright: FTW!!! 1996 Style! |
| 02:11.27 | seanbright | bows |
| 02:11.34 | seanbright | is old skool |
| 02:11.46 | drmessano | Holy crap |
| 02:11.51 | drmessano | I think I just peed |
| 02:12.11 | drmessano | errr.. I mean |
| 02:12.13 | drmessano | Good show |
| 02:12.27 | djs26 | drmessano: 2.6.18-53.1.19.el5 kernel-devel-2.6.18-53.1.19.el5 |
| 02:12.45 | seanbright | and on that note... i retire |
| 02:12.48 | seanbright | night folks |
| 02:12.58 | djs26 | Night seanbright |
| 02:13.29 | drmessano | nite |
| 02:13.47 | plla | djs26: It's trying to recompile the kernel. |
| 02:14.00 | drmessano | djs26: Why are you telling me that? |
| 02:14.34 | djs26 | Oh, sorry drmessano... I misaligned the user nick with the request. |
| 02:14.39 | plla | djs26: try using "make --nodep" |
| 02:14.46 | djs26 | plla: Ok. |
| 02:15.30 | jameswf-home | 2.6.24-17-generic |
| 02:17.01 | drmessano | wonders why no one brought up channel 0 |
| 02:17.26 | djs26 | plla: Unrecognized option, and the options it gave had nothing comparable. |
| 02:18.36 | djs26 | It is like it has the wrong path for building the kernel. |
| 02:19.03 | djs26 | And I made the linux ln for the src. |
| 02:21.22 | jameswf-home | Q:Do you know what the death rate around here is? |
| 02:21.22 | jameswf-home | A:One per person. |
| 02:21.29 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 02:21.58 | maqr | asterisk is telling me it's trying to connect to postgres, but i don't think i want it to do that (yet)... any idea why it's trying? it doesn't seem to be mentioned in my config |
| 02:22.59 | plla | djs26: "make -nodep" |
| 02:23.39 | djs26 | Awesome! Thank you!! |
| 02:23.49 | *** join/#asterisk GregorR (n=gregor@71.237.179.105) |
| 02:25.10 | GregorR | This may be a bit ... totally impossible. Is it possible to use asterisk (or a plugin for it, or a softphone that plugs into it) as a dial-up networking server? |
| 02:27.55 | *** join/#asterisk hsv-AL (n=ding@user-24-214-126-81.knology.net) |
| 02:30.51 | GregorR | Or, more sensibly, for it to emulate the modem such that a standard pppd can be run on it? |
| 02:32.28 | *** join/#asterisk excAliBuR (n=sales@207.134.8.33) |
| 02:33.00 | excAliBuR | i have ubuntu and i did the apt-get asterisk install .. where did it go? |
| 02:33.42 | drmessano | Why would you want to emulate a modem with asterisk? |
| 02:33.48 | drmessano | How would you connect to it? |
| 02:33.51 | maqr | oh, i guess just by having res_pgsql.conf, it was trying to connect to postgres |
| 02:34.03 | shmaltz | excAliBuR, I hope you realized this is not a ubuntu channel, but an Asterisk one |
| 02:34.15 | excAliBuR | yes and my question is asterisk related |
| 02:34.30 | shmaltz | excAliBuR, no it's not, its ubunto related |
| 02:34.34 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 02:34.35 | drmessano | WTF |
| 02:34.42 | drmessano | He asked about asterisk |
| 02:34.44 | excAliBuR | no because the package is not a default ubuntu one |
| 02:34.57 | shmaltz | excAliBuR, I ahve been doing asterisk for over 4 yrs now, never did I do apt get to run or install asterisk |
| 02:35.01 | jbeez | where can I download ubuntu? |
| 02:35.04 | jbeez | :P |
| 02:35.10 | excAliBuR | i added the resp for asterisk |
| 02:35.31 | shmaltz | I guess here #ubuntu |
| 02:36.08 | drmessano | shmaltz: STFU ---> https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 02:36.31 | drmessano | If you're not gonna help, stop being an ass pls |
| 02:36.57 | shmaltz | drmessano, I'm not an ass, its the wrong channel to ask |
| 02:37.12 | shmaltz | just like it's the wrong channel for swtichvox |
| 02:37.19 | drmessano | shmaltz: you are wrong.. he's asking about installing ASTERISK.. THIS, IS, #ASTERISK |
| 02:37.40 | drmessano | Look up ^^^^^ |
| 02:37.40 | shmaltz | no, he is asking about how a dabien clone installs asterisk |
| 02:37.43 | drmessano | #Asterisk |
| 02:37.49 | drmessano | So? |
| 02:38.15 | shmaltz | to install asterisk you do wget http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.20.tar.gz |
| 02:38.21 | shmaltz | then tar -vxf |
| 02:38.23 | excAliBuR | umm i can leave if that makes you happy |
| 02:38.30 | shmaltz | then cd asterisk-1.4.20 |
| 02:38.33 | shmaltz | then make clean |
| 02:38.33 | drmessano | excAliBuR |
| 02:38.34 | shmaltz | make |
| 02:38.36 | shmaltz | make install |
| 02:38.39 | drmessano | Try this link: https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 02:38.39 | shmaltz | make samples |
| 02:38.42 | drmessano | and ignore him |
| 02:38.45 | shmaltz | make configs |
| 02:38.50 | excAliBuR | ok... |
| 02:38.52 | jaytee | excAliBuR, you can use Synaptic Package Manager to install Asterisk from the repositories |
| 02:39.09 | jaytee | or apt-get if you're running server without a gui |
| 02:39.17 | drmessano | excAliBuR: You are in the correct place... no worries |
| 02:39.29 | excAliBuR | :) |
| 02:40.18 | drmessano | This may help as well: http://forums.digium.com/viewtopic.php?p=69476&sid=e4f50870373a1c55a505835e7b1a8a06 |
| 02:40.43 | excAliBuR | ok |
| 02:44.54 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 02:47.23 | maqr | what causes "No matching peer found"? |
| 02:49.54 | maqr | oh, i guess i didn't have an entry in sip.conf to match the username i wanted to use, that'd explain it |
| 02:50.49 | [TK]D-Fender | load chan_obvious.so |
| 02:51.08 | maqr | :/ |
| 02:52.27 | maqr | [TK]D-Fender: alright, that makes sense, but now i get "408 - Request Timeout", any thoughts on that? |
| 02:53.10 | *** part/#asterisk TheX1le (n=thex1le@cpe-76-189-235-180.neo.res.rr.com) |
| 02:53.18 | [TK]D-Fender | maqr, Now why would * not get an answer back do you think? |
| 02:55.15 | maqr | [TK]D-Fender: well, usually i'd suspect some kind of firewall issue or something.... but in this case, i know it was getting a 404 before, and now it's "timing out", which doesn't make sense to me |
| 02:55.29 | maqr | my sip phone couldn't be unable to reach the asterisk server, otherwise it wouldn't have 404'd before |
| 02:55.42 | [TK]D-Fender | maqr, perhaps you should evaluate the networking between * and your device |
| 02:55.43 | plla | I was already used to the Zaptel name... the new name Dahdi makes me cringe. |
| 02:57.01 | *** join/#asterisk jsolis (n=jimmy@201.240.212.214) |
| 02:59.48 | plla | Dah. Maybe DAH wouldn't make me think in daddy. |
| 03:01.49 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 03:02.10 | drmessano | Dahdi is a lame name |
| 03:02.22 | drmessano | I don't see where it needed an acronym |
| 03:03.47 | plla | dahdi show channels |
| 03:05.12 | drmessano | Digium Asterisk Hardware Management EmulatoR |
| 03:05.15 | drmessano | DAHMER!! |
| 03:05.15 | lanning | now we need "Mother" from "Alien" |
| 03:06.40 | drmessano | MAHMI |
| 03:06.51 | jaytee | we used mothers when I was in the USAF, Mobile Operational Three Hole Evacuation Receptacle (basically a long bench with 3 toilet seats bolted to it that was setup on 2 short saw horses over an open pit. |
| 03:07.29 | lanning | bench seating? |
| 03:07.33 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 03:08.05 | jaytee | yeah, with no privacy curtains |
| 03:08.26 | drmessano | Multi-Asterisk Handshake Management Interface |
| 03:08.31 | drmessano | YAY |
| 03:08.36 | drmessano | I have a name.. no product |
| 03:08.58 | lanning | but the name is the only important piece |
| 03:09.06 | drmessano | Try |
| 03:09.07 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 03:09.20 | drmessano | It's enough to get Venture capital, so screw it |
| 03:09.26 | plla | You just need a good marketing department. |
| 03:09.32 | plla | You don't need a product. |
| 03:09.47 | drmessano | Who needs a product.. it's an API |
| 03:09.51 | drmessano | Hmm |
| 03:10.09 | plla | It's warrantied people will buy/use it even if they don't know what it is. |
| 03:10.13 | jameswf-home | ~dahdijoke |
| 03:10.16 | jbot | somebody said dahdijoke was http://files.jameswf.info/dahdihorse.jpg |
| 03:10.50 | drmessano | Spoil sport |
| 03:11.10 | jameswf-home | :) that was some hard work to do in gimp |
| 03:11.51 | drmessano | Probably took 3 days to install gimp |
| 03:13.08 | jameswf-home | sudo apt-get install gimp |
| 03:13.24 | maqr | [TK]D-Fender: is there some generic 'sip test' tool for testing the connections? |
| 03:13.52 | jameswf-home | maqr: sipp |
| 03:14.24 | [TK]D-Fender | maqr, every SIP device is a means of testing. |
| 03:19.03 | jblack | Is it normal to get iax timeout errors on a machine that's 0.4 ms away? |
| 03:19.29 | [TK]D-Fender | jblack, if 1 side ignores the other... sure |
| 03:20.13 | jblack | I wonder if I'm running low on cpu time |
| 03:25.35 | *** join/#asterisk BajaEd (n=ednagy@72.168.135.210) |
| 03:29.44 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:33.35 | *** join/#asterisk jesselang (n=jesse@h75-100-163-14.mntimn.dsl.dynamic.tds.net) |
| 03:34.11 | jesselang | Does PlayTones(0) transmit silence to the channel? |
| 03:34.30 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 03:42.46 | maqr | jameswf-home: what exactly am i looking for with sipp to show me where the error is? |
| 03:43.13 | [TK]D-Fender | maqr, you don't need sipp |
| 03:43.55 | maqr | [TK]D-Fender: well all the ports that asterisk shows in use with netstat i have open |
| 03:45.27 | maqr | [TK]D-Fender: what else could be wrong? |
| 03:46.32 | [TK]D-Fender | maqr, whats wrong is you're running around like a blind mouse in a maze. PASTEBIN is your friend, as is "sip debug". Use them. |
| 03:46.53 | [TK]D-Fender | maqr, You've shown nothing, so you're nowhere. Entirely appropriate. |
| 03:46.58 | hsv-AL | heh |
| 03:47.00 | hsv-AL | fender is still on |
| 03:47.04 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-187-7.hsd1.wa.comcast.net) |
| 03:47.05 | hsv-AL | ThakillaZ did good tonight |
| 03:47.06 | hsv-AL | :) |
| 03:48.05 | maqr | [TK]D-Fender: oh, sip debug shows the problem, asterisk is trying to send packets back to my lan ip |
| 03:48.08 | maqr | it's a nat problem |
| 03:48.16 | [TK]D-Fender | ~sipnat |
| 03:48.17 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:48.40 | hsv-AL | d-fender |
| 03:48.45 | hsv-AL | im gonna join the DiG server |
| 03:48.52 | hsv-AL | can clan [TK] join? |
| 03:50.26 | maqr | [TK]D-Fender: my NAT issue is actually on the other end, unless i'm reading that wrong... my phone has NAT, but my asterisk server does not |
| 03:50.43 | *** join/#asterisk jameswf-home (n=James@ip72-223-0-183.ph.ph.cox.net) |
| 03:51.19 | [TK]D-Fender | maqr, then your peer needs "qualify=yes", "canreinvite=no", "nat=yes", and shouldn't have any forwarding. |
| 03:51.41 | *** part/#asterisk jameswf-home (n=James@ip72-223-0-183.ph.ph.cox.net) |
| 03:52.01 | hsv-AL | fender, |
| 03:52.06 | hsv-AL | take some sleeping pills,a nd get a good night sleep |
| 03:52.10 | hsv-AL | enough asterisking it for the day |
| 03:53.27 | maqr | [TK]D-Fender: SUCCESS! |
| 04:03.51 | *** join/#asterisk fiddur (i=fiddur@78.82.252.209) |
| 04:10.18 | maqr | how do i go about associating a "sip peer" (my softphone) with a mailbox? right now asterisk is telling me "Received SIP subscribe for peer without mailbox", but i definitely do have a mailbox defined |
| 04:10.27 | *** part/#asterisk jesselang (n=jesse@h75-100-163-14.mntimn.dsl.dynamic.tds.net) |
| 04:13.51 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:14.15 | [TK]D-Fender | maqr, Apparently not in your sip peer. |
| 04:15.11 | *** join/#asterisk jameswf (n=nobody@ip72-223-0-183.ph.ph.cox.net) |
| 04:16.17 | maqr | [TK]D-Fender: oh, you are correct, fixed |
| 04:16.47 | [TK]D-Fender | maqr, It is your job to associate everything inside of *. |
| 04:17.11 | hsv-AL | fender, your quite Tha Killa |
| 04:17.12 | hsv-AL | with "*" |
| 04:17.44 | hsv-AL | now I see the oriign of your name |
| 04:17.46 | hsv-AL | "Defender" |
| 04:18.00 | hsv-AL | you defended the RL box in quake 1 on dm3.bsp - the abandoned base :) |
| 04:18.32 | [TK]D-Fender | hsv-AL, Nope. Almost the right age though. |
| 04:19.02 | hsv-AL | i still play 4 vs 4 on dm3.bsp |
| 04:19.04 | hsv-AL | in quake 1 NQ |
| 04:19.08 | hsv-AL | vulcan.harvard.edu :) |
| 04:19.41 | maqr | [TK]D-Fender: i'm getting the hang of it :) |
| 04:25.46 | excAliBuR | does asterisk still have that testing thing when u install it to test delay and stuff? |
| 04:26.07 | excAliBuR | i got the addons.. and samples |
| 04:27.39 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:31.52 | *** join/#asterisk alrs (i=foobar@216.151.159.21) |
| 04:41.31 | maqr | i'm having some trouble getting my dialplan to work, following the book, "exten => X,1,SayDigits(${EXTEN})" should read any digit i dial from my sip phone, right? |
| 04:42.20 | TJNII | X by itself is not a pattern match, but extension X |
| 04:42.29 | TJNII | Try _X |
| 04:52.50 | *** join/#asterisk jameswf (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 04:53.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:54.48 | maqr | TJNII: it rings once and says "The person you are calling is unavailable, please try again." |
| 04:55.35 | maqr | TJNII: my sip user for my phone has "context=phones", and in my extensions.conf i have "[phones]" then "exten => _X,1,SayDigits(${EXTEN})" |
| 04:55.56 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585339.dsl.bell.ca) |
| 04:56.48 | trnzmeta | guys any good decent SIP phones out there? |
| 04:56.58 | maqr | i'm not sure why it's even playing that message, since it's not in my dialplan anywhere |
| 04:57.03 | trnzmeta | gradnstream sucks |
| 04:57.05 | trnzmeta | snom sucks |
| 04:58.20 | TJNII | maqr: Do you have a virgin dialplan that you wrote yourself from scratch, or are you modifying someone else's? |
| 04:59.44 | TJNII | trnzmeta: I've had decent luck with grandstreams, they're good budget phones. If you want better pay more than $50. I have a Snom that I got on eBay, but it resets all the time. It would be a nice phone if it didn't do that. My friends sing the praises of Polycom, but I find them to be cost prohibitive. |
| 04:59.53 | TJNII | So I guess what I'm saying is no. |
| 04:59.58 | TJNII | :) |
| 05:01.30 | FreedomBI | Polycoms also don't deal with NAT at all well. |
| 05:01.31 | maqr | TJNII: well, i did 'make samples', but it's pretty much empty |
| 05:02.00 | maqr | TJNII: http://pastebin.ca/1024570 |
| 05:02.10 | maqr | that's the whole extensions.conf |
| 05:03.40 | TJNII | maqr: Anything usefull on the cli? |
| 05:03.53 | TJNII | (vith verbose set >= 5) |
| 05:03.59 | TJNII | s/vith/with/ |
| 05:05.16 | maqr | TJNII: http://pastebin.ca/1024571 |
| 05:05.39 | maqr | i'm not entirely sure what that means |
| 05:06.06 | TJNII | Well, it looks like it's trying to say 1..... |
| 05:07.19 | TJNII | Odd. |
| 05:08.00 | maqr | well, i dialed '1' |
| 05:08.09 | maqr | but it isn't saying anything |
| 05:08.09 | maqr | heh |
| 05:08.18 | TJNII | As a test, put exten => _X,2,wait(1) followed by exten => _X,3,Hangup() in there |
| 05:08.48 | TJNII | I've never played with extensions that run off the end and let's make sure it isn't moving on to something strange. |
| 05:09.44 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 05:10.25 | maqr | TJNII: http://pastebin.ca/1024572 |
| 05:10.39 | maqr | TJNII: i still here the "The person you are calling is unavailable, please try again." message |
| 05:11.08 | TJNII | Okay. Well it's trying tp play digits/1 |
| 05:11.24 | TJNII | So next I would check that the digits/1 sound file actually says 1 |
| 05:11.44 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:15.19 | maqr | TJNII: where is that supposed to be, or what's the file name? |
| 05:16.00 | maqr | TJNII: oh wait, 1.gsm? i have that, yeah |
| 05:16.18 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 05:16.23 | maqr | it's in /home/asterisk/local/var/lib/asterisk/sounds/digits/1.gsm |
| 05:16.53 | maqr | but in asterisk.conf i also have: astvarlibdir => /home/asterisk/local/var/lib/asterisk and astdatadir => /home/asterisk/local/var/lib/asterisk |
| 05:16.56 | maqr | so that should be fine, right? |
| 05:17.30 | TJNII | I'd think so. |
| 05:18.51 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 05:18.55 | murdock_ut | Howdy all. |
| 05:19.20 | murdock_ut | Quick question, when a user places someone on hold I'm seeing the following message on the console: RTCP SR transmission error, rtcp halted |
| 05:19.33 | murdock_ut | My research says that this is related to a sip reinvite. |
| 05:19.54 | murdock_ut | However, i've got canreinvite=no set in my sip.conf as a global setting. |
| 05:19.58 | maqr | TJNII: could it be because it keeps saying "in new stack"? |
| 05:20.06 | TJNII | No, that's normal |
| 05:20.08 | maqr | oh |
| 05:20.33 | maqr | well, it says it's playing the files, but then i get the error sound |
| 05:20.42 | maqr | and it doesn't say that it's playing the error |
| 05:21.00 | TJNII | I would try something like exten => 101,1,Playback(digits/1) exten => 101,2,Hangup() |
| 05:21.08 | TJNII | That should say 1 |
| 05:21.25 | TJNII | And if it doesn't I would make sure your sound files are correct. |
| 05:22.23 | maqr | where '101' is my extension? |
| 05:22.49 | TJNII | Right. Just a basic extension. |
| 05:23.57 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 05:26.11 | *** join/#asterisk ManxPower (n=manxpowe@12.sub-70-220-153.myvzw.com) |
| 05:28.13 | maqr | TJNII: aha! |
| 05:28.34 | maqr | i think i got it now |
| 05:28.38 | TJNII | Figure it out? |
| 05:28.51 | maqr | TJNII: apparently asterisk comes with some rss_features thing that sets '700' to 'Park()' |
| 05:29.01 | maqr | i have no idea what that means or why it does it, but i just arbitrarily picked 700 to start my extensions at |
| 05:29.07 | maqr | so it must have been conflicting somehow |
| 05:29.13 | TJNII | Shouldn't |
| 05:29.32 | TJNII | Did that 101 extension say 1? |
| 05:29.33 | maqr | hmm |
| 05:29.59 | maqr | not quite, but i think it just hung up too fast |
| 05:30.24 | maqr | TJNII: oh, i also changed the order of my extensions.conf, but that shouldn't matter, should it? |
| 05:30.25 | TJNII | throw a wait(1) before and after |
| 05:30.48 | TJNII | No, the order shouldn't matter as long as you keep the right extensions in the right contexts. |
| 05:31.03 | maqr | ok, lemme try again, sec |
| 05:32.00 | TJNII | so exten => 101,1,Wait(1) exten => 101,2,Playback(digits/1) exten => 101,3,Wait(1) exten => 101,4,Hangup() |
| 05:34.37 | maqr | hmm, no it doesn't work, i'll paste the error |
| 05:35.06 | maqr | http://pastebin.ca/1024583 |
| 05:35.12 | TJNII | Does it play the same sound? |
| 05:35.19 | maqr | err, sorry, no sound at all |
| 05:35.23 | maqr | it just sits there for a second and then hangs up |
| 05:35.48 | maqr | in that paste btw, my phone is '200' |
| 05:35.51 | maqr | which i guess you can tell anyway |
| 05:36.05 | TJNII | I think your soundfiles are broken. |
| 05:36.27 | maqr | but it seems to be able to play the error message okay |
| 05:36.32 | maqr | unless that's not asterisk sending that |
| 05:36.41 | maqr | maybe that's embedded in my softphone or something |
| 05:37.04 | TJNII | What I would do now is download a new soundpack (You can find them complete cnd online), move the current soundfiles to somewhere else, and put the new soundfiles in there place. Then try again. |
| 05:37.17 | maqr | TJNII: i just built this from source though, i don't see how it could be wrong |
| 05:37.24 | TJNII | According to the CLI output it is trying to do what it is supposed to. |
| 05:37.39 | TJNII | Do you have a sound card on that machine? |
| 05:37.47 | maqr | nope, it's just a virtual server |
| 05:37.53 | maqr | hosted at linode.com, Xen VPS |
| 05:39.12 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 05:39.43 | maqr | TJNII: there's a whole bunch of gsm files here, they must be right, it looks like they came in the source tarball |
| 05:40.29 | maqr | err, maybe not, i thought they did though |
| 05:40.44 | maqr | oh, yeah, they did, they're in a tarball in the source |
| 05:41.02 | maqr | TJNII: can i make asterisk show me what directory it's trying to play the file from? |
| 05:41.02 | TJNII | Try playing another sound file and see if it works |
| 05:41.07 | edibrac | hmm... my Cisco 7940 with firmware 8.x gets a busy signal when I go to speakerphone and in the handset -- but firmware 5.x-7.x is fine..except I can't dial over 4 digits if the 1st # is 5 (i.e. 54321) |
| 05:41.07 | maqr | maybe an absolute path to the file? |
| 05:41.29 | TJNII | Hmmmm.... I'm not sure... |
| 05:42.44 | maqr | TJNII: hello world didn't play either |
| 05:42.57 | maqr | Spawn extension (phones, 101, 4) exited non-zero on 'SIP/200-081e35d0' |
| 05:43.01 | maqr | same kind of error |
| 05:43.07 | maqr | 'non-zero' maybe means it can't find it? |
| 05:43.15 | TJNII | Well, you can do a quick test. Move 1.gsm somewhere else. 101 should then print an error about not being able to find the file |
| 05:43.41 | maqr | err, yeah, not finding it is a different error |
| 05:43.43 | TJNII | non zero is normal, at least in my experience. |
| 05:44.00 | maqr | alright, well, it's finding the files then |
| 05:45.11 | TJNII | So 101 doesn't play anything at all? |
| 05:46.07 | maqr | nope |
| 05:46.10 | maqr | just silence |
| 05:46.14 | maqr | no clicking sounds or anything |
| 05:47.39 | TJNII | Can I see your sip.conf? |
| 05:48.14 | maqr | sure, sec |
| 05:49.47 | maqr | well, lemme comment out the callcentric stuff and make sure i get the same result, then i'll paste |
| 05:49.50 | maqr | sec |
| 05:51.21 | maqr | http://pastebin.ca/1024592 |
| 05:51.30 | maqr | TJNII: nothing too complicated there |
| 05:52.27 | maqr | TJNII: oh, i did add a few noloads to make warnings go away, but i don't see how that'd be related |
| 05:52.33 | maqr | i'll post those too |
| 05:52.48 | TJNII | Please |
| 05:53.03 | maqr | http://pastebin.ca/1024594 |
| 05:53.11 | maqr | the chan_alsa one was disabled to begin with from the sample |
| 05:53.19 | maqr | postgres ones are disabled because i'm not using pg yet |
| 05:53.32 | maqr | i'm not using music on hold either, or ael configs |
| 05:53.49 | maqr | and i'm not sure what smdi is, but i think it has something to do with hardware |
| 05:54.37 | TJNII | Try adding disallow=all and allow=gsm to the [200] entry in your sip.conf as a test. |
| 05:55.05 | TJNII | Let's force the phone to gsm and see if it is a format conversion issue. |
| 05:55.17 | maqr | *CLI> [May 21 01:55:09] NOTICE[6843]: chan_sip.c:5481 process_sdp: No compatible codecs, not accepting this offer! |
| 05:55.23 | maqr | maybe this is related? |
| 05:55.55 | TJNII | You got this after adding the disallow and allow |
| 05:56.01 | maqr | yup |
| 05:56.07 | TJNII | I would say so. |
| 05:56.17 | maqr | i added those two lines to the bottom of my [200] sip section |
| 05:56.31 | maqr | does that mean my softphone is broken? |
| 05:56.42 | TJNII | Not broken, but it doesn't like gsm |
| 05:56.51 | maqr | it's x-lite, btw |
| 05:57.00 | TJNII | So * will have to convert those gsm sound files to whatever x-lite will talk |
| 05:57.15 | TJNII | Which chould be the weak link where the sound is being lost |
| 05:57.19 | maqr | i don't see "GSM" on my x-lite codec list |
| 05:57.28 | maqr | maybe i should download a new version of it or something |
| 05:57.32 | TJNII | Which ones do you see? |
| 05:57.42 | TJNII | Out of curiosity |
| 05:58.19 | maqr | BroadVoice-32, BroadVoice-32 FEC, G711 aLaw, G711 uLaw, iLBC, Speex, Speex FEC, Speex Wideband, Speex Wideband FEC |
| 05:58.22 | maqr | that's it |
| 05:58.42 | TJNII | Okay. You installed * from scratch, right? |
| 05:59.30 | maqr | yeah, in a /home/asterisk directory, using 1.4.19.2 |
| 05:59.53 | maqr | looks like i have the latest x-lite too |
| 06:00.16 | maqr | i could use a different softphone or something though, this was just the one that seemed most popular |
| 06:00.19 | maqr | (windows) |
| 06:00.21 | TJNII | Okay, as a test you can try installing the alaw or ulaw sound files as well. There should be an option in the make menu to do this. |
| 06:00.46 | maqr | http://lists.digium.com/pipermail/asterisk-users/2007-November/199842.html |
| 06:01.07 | maqr | it looks like i might just be able to add gsm here, sec |
| 06:01.44 | TJNII | That would probably be easier. :) |
| 06:02.09 | maqr | TJNII: btw, which codec sounds the clearest? audio quality is important to me |
| 06:02.25 | CCFL_Man2 | anyone know about cisco voice gateways? |
| 06:03.42 | TJNII | maqr: I'm not an expert on that, so I'm not sure. Though let's try straight gsm to gsm for now to figure out why your sound files won't play. |
| 06:04.29 | CCFL_Man2 | i'd imagine g711u sounds the best |
| 06:04.33 | maqr | TJNII: :) |
| 06:06.08 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 06:06.28 | maqr | TJNII: this might take a minute, i need to either find x-lite 2.0 somewhere or find a new sip phone |
| 06:07.07 | CCFL_Man2 | need to pick all the clams out of my teef |
| 06:09.40 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 06:12.37 | x86 | peeks in |
| 06:13.40 | CCFL_Man2 | if my teeth hurt i can tell there is something in between them |
| 06:14.13 | CCFL_Man2 | i got an efnora gsm1218a modem today |
| 06:14.24 | TJNII | maqr: Well, I need to go to bed, I have things to do in the morning. near as I can tell your dialplan is fine and * is trying to play the correct file. I think you have a problem with format conversion, but we haven't done enough tests to prove it. Perhaps someone else can take over from here. |
| 06:14.57 | maqr | TJNII: thanks very much :) |
| 06:15.04 | maqr | TJNII: that's definitely the problem |
| 06:15.14 | TJNII | You proved it? |
| 06:15.21 | maqr | not yet, but it has to be :) |
| 06:16.09 | *** join/#asterisk techie (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
| 06:16.13 | TJNII | Wellm it could still be broken files, but if there is no format conversion and it plays you'll know it is. Anyways, good luck. |
| 06:16.20 | maqr | thanks |
| 06:16.21 | maqr | :) |
| 06:17.33 | maqr | might be time to sleep here too, i can't seem to get x-lite 2 configured at the moment |
| 06:17.48 | maqr | i'm pretty confident that a gsm softphone is the solution though |
| 06:18.35 | maqr | oh wait, it does work |
| 06:18.44 | maqr | TJNII: that was it alright :) |
| 06:18.50 | maqr | that much is fixed |
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| 06:21.02 | maqr | asterisk can use speex, right? |
| 06:21.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:22.01 | x86 | yes |
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| 06:32.07 | *** part/#asterisk techie (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
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| 06:57.32 | FuriousGeorge | hey all |
| 06:58.09 | FuriousGeorge | so this is new: checking in on some servers today. havent been restarting them nightly just to see what happens. i had a channel that got hung between me and my iax2 provider |
| 06:58.19 | FuriousGeorge | i guess the iax2 equivalent of the 'BYE' got missed |
| 06:58.42 | FuriousGeorge | should i expect a few hundred dollars extra on my bill this month? |
| 06:59.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 07:00.11 | techie | Nah, no RTP |
| 07:03.13 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 07:06.10 | trnzmeta | ok guys |
| 07:06.26 | trnzmeta | what would stop my softphone from dialing outside |
| 07:06.42 | trnzmeta | when I change my account details to test account I can call my mobile |
| 07:07.00 | trnzmeta | but when I use my ordinary account, it just rings on soft phone |
| 07:07.27 | trnzmeta | but nothing on mobile |
| 07:13.26 | trnzmeta | Got SIP response 481 "Call/Transaction Does Not Exist" |
| 07:16.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:20.34 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 07:20.44 | casix | hello |
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| 07:23.38 | patrick-- | Hey, im using SpanDSP and rxfax to receive faxes. Im having the problem that some faxes that are received are not viewable. does anyone know why this could be the case? |
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| 07:51.16 | fiddur | Hi. Can I make asterisk send out a warning (through email, sendxmpp or whatever) when a certain interface has only X channels left? I want to know if we get low on ISDN-channels on our zap/g2... |
| 07:52.34 | Strom_M | fiddur: you could just analyze your call detail record |
| 07:52.41 | Strom_M | s/cord/cords/ |
| 07:53.09 | Strom_M | fiddur: but I don't see why you couldn't write some logic into your inboud/outbound call processing that checks the number of available channels |
| 07:56.51 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:58.30 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 07:58.32 | fiddur | Hmm, I guess I could analyze and find peak-usage of a channel from cdr yes... but now I rather wanted a quick warning when we're running out of channels... so we can wait with outbound calls that are non-urgent for example... |
| 07:58.47 | fiddur | Strom_M: How do I check number of available channels in the dialplan? |
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| 08:32.15 | falco_toadfoot | Anyone else noticed this problem? 2 pbxes, connect them with IAX2, call from sip phone on pbxA, via pbxB, via PBXa to another sip phone |
| 08:32.28 | falco_toadfoot | the cpu usage will rise to ~90% |
| 08:33.16 | falco_toadfoot | asterisk 1.2.13 |
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| 08:45.59 | daven | Hi. Do any of you have experience getting video support working in Asterisk? I tried for a few hours yesterday, but was unable to do anything |
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| 09:11.20 | MaartenB | does anyone have experience with XFER (call transfer) to a phone in another NAT? |
| 09:11.26 | MaartenB | my calls get dropped in that situation |
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| 09:32.22 | hoagie|ast | <PROTECTED> |
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| 09:51.46 | *** join/#asterisk Peterpen (n=stevem@212.57.232.254) |
| 09:52.35 | Peterpen | Lo, I just converted an MP3 to GSM file format (using Sox) and I'm on my linux desktop machine and I can't play it back - think I'm missing a codec or library or something... anyone know what I need just to listen to it? |
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| 09:59.54 | LuisTorres | Whan linux are you using? |
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| 10:08.54 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 10:11.44 | zeeesh | one of sip peer expires normaly after 3600 seconds .. and unable to receive incoming .. can asterisk server ping this client every after 20 minutes either its still alive or not ? |
| 10:14.38 | Oy90 | zeeesh: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify |
| 10:17.42 | Uatec | zeeesh, you can set the qualify parameter i think it is |
| 10:17.58 | Uatec | oh right lol, what Oy90 said |
| 10:23.09 | LuisTorres | Zeeesh what the error that you received? |
| 10:25.09 | LuisTorres | sometimes I have laods of Got SIP response 603 "Decline" back from x.x.x.x, will you think should be a qualidy issue as well? |
| 10:27.10 | zeeesh | <LuisTorres>: sometime channels not available and some time congestion .. |
| 10:29.20 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 10:30.13 | LuisTorres | did you got 603 decline messages oon the console as well? |
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| 10:40.08 | Georger86 | hi |
| 10:40.24 | Georger86 | anybody can help me with a2billing? |
| 10:41.01 | yang | Georger86: I think they got a paid support at a2billing website |
| 10:42.43 | Georger86 | yes paid.. |
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| 10:48.07 | zeeesh | <LuisTorres>: not receiving kind of 603 decline .. mostly channel unavailable or congestion |
| 10:49.07 | LuisTorres | thank you zeeesh |
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| 10:57.21 | redback | I keep getting messages such as - Channel 'IAX2/voiptalk' unable to transfer when making calls via my voiptalk account but calls to external extensions and the link to the other IAX box transact smoothly. Is this likely to be a problem their end or mine? |
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| 11:22.58 | Peterpen | LuisTorres, Debian |
| 11:23.35 | Peterpen | I've installed a good dozen or so extra packages and nothing can play these gsm files still :S |
| 11:23.45 | Peterpen | tryed VLC, totem, rhythmbox, audacious |
| 11:24.11 | Peterpen | put sox and all plugins on, gstreamter, ffmpeg, :( and the plugins into the players - no joy |
| 11:24.13 | Peterpen | and libgsm0 |
| 11:24.31 | *** join/#asterisk blinky42 (n=steveb@67.106.175.146.ptr.us.xo.net) |
| 11:25.02 | Assid | is still trying to figure out how and why is skype giving unlimited (10kmins) |
| 11:25.22 | tzafrir | Peterpen, Lenny? |
| 11:25.26 | tzafrir | install sox-all |
| 11:25.29 | Peterpen | yeah :) how did you get |
| 11:25.33 | Peterpen | already done that |
| 11:25.41 | tzafrir | play file.gsm |
| 11:25.44 | Peterpen | sox has converted my WAV to GSM already |
| 11:26.50 | Peterpen | 'play' works :S |
| 11:26.54 | tzafrir | can sox play wav/gsm files? (wav files with gsm payload)? |
| 11:27.03 | tzafrir | Can ity play gsm files from elsewhere? |
| 11:27.08 | Peterpen | when I was on Etch I found audacious could play the gsm files |
| 11:27.16 | Peterpen | like I already had a codec installed for it |
| 11:27.16 | tzafrir | If so: something is probably wrong with the conversion |
| 11:28.01 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:28.01 | Peterpen | well I took an MP3, got audacity to save as WAV then used .... |
| 11:28.02 | Peterpen | sox 3_track_courtesy.wav -r 8000 3_track_courtesy.gsm resample -ql |
| 11:28.26 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 11:29.53 | Peterpen | ok they play ok :) |
| 11:30.13 | Peterpen | I take it this 'play' is a part of sox and there should be a sox plugin for normal GUI players? |
| 11:31.38 | tzafrir | Peterpen, before install libsox-fmt-gsm: |
| 11:31.41 | tzafrir | play soxio: Can't open input file `/usr/share/asterisk/sounds/tt-monkeys.gsm': unknown file type `gsm' |
| 11:31.48 | Peterpen | I've already for that |
| 11:31.50 | tzafrir | after: plays it just fine |
| 11:31.55 | Peterpen | *got |
| 11:32.06 | tzafrir | (no extra switches needed for "play") |
| 11:32.06 | Peterpen | I had to have that to convert it in the first place |
| 11:32.45 | Peterpen | whenever totem or audacious etc open it they are unable to determine the mime type and fail |
| 11:32.52 | Peterpen | 'play' works fine tho |
| 11:32.56 | tzafrir | libsox-fmt-gsm is one of the packages installed when you install libsox-fmt-all |
| 11:33.05 | tzafrir | (sorry, got that name wrong |
| 11:33.07 | tzafrir | ) |
| 11:33.09 | Peterpen | yeah I've installed libsox-fmt-all |
| 11:33.26 | Peterpen | tried all this# |
| 11:35.09 | casix | I have two asterisks connectec via sip. The asterisk a make calls to asterisk b. I have defined a peer in asterisk a with host the name of asterisk b machine and in asterisk b I have defined a peer with host to the name of asterisk a machine. but when a calls to b it don't use the context of the user it uses de default context. If I define in asterisk b a friend with asterisk a it works well. what i'm doing wrong? |
| 11:35.13 | Rico29 | hello everybody |
| 11:35.56 | Rico29 | i need to get the brand (or model) of a phone registered onmy asterisk with "autocreatepeer=yes" |
| 11:36.10 | Rico29 | and I don't know how i can do it |
| 11:36.17 | Rico29 | any idea ? |
| 11:38.28 | casix | Rico29: I have never used autocreatepeer but if you know the contact header field that this phone use to register you could do a sip show peer <contact header field> and in the useragent will tell you the phone model |
| 11:38.56 | Rico29 | it must work automatically |
| 11:39.27 | Rico29 | i think an AGI would be a good way to do it, but i'm not soo good in AGI dev |
| 11:40.16 | casix | Rico29 but you know the username that will use?? |
| 11:40.22 | Rico29 | nop |
| 11:40.37 | casix | rico29 and is only for this user or for all users? |
| 11:40.45 | Rico29 | it uses the same username that the one which is in the phone |
| 11:40.56 | Rico29 | all users |
| 11:41.57 | Rico29 | when I plug a phone, after provisioning, it registers with autocreatepeer, |
| 11:42.14 | Rico29 | and here i want to get its brand / model |
| 11:45.41 | *** part/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 11:48.16 | casix | Rico29 you can use the manager interface: http://www.voip-info.org/wiki/view/asterisk+manager+events with peerstatus event. you can execute and script when you see that event that make an asterisk -rx 'show peer <new_peer> | grep useragent |
| 11:48.25 | casix | Rico29 or something like this |
| 11:49.02 | Rico29 | yes |
| 11:49.03 | casix | s/that make/then make/ |
| 11:49.25 | Rico29 | in CLI, sip show peer 1000 returns me "cisco" |
| 11:50.11 | Rico29 | i need to get this automatically ! |
| 11:50.12 | Rico29 | argh |
| 11:50.24 | Rico29 | i.e. with a perl script or something else |
| 11:50.39 | casix | with the manager and peerstatus event you can do this... |
| 11:51.52 | casix | and the command action you can make the sip show peer 1000 |
| 11:52.57 | casix | connect to manager interface, wait for a peerstatus event with a register option, then exec a command action with sip show peer <registered with peerstatus event> |
| 11:53.00 | casix | and you have it |
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| 11:58.07 | Rico29 | casix> can an applicatio connect to the manager automatically, each time a new peer registers with autocreatepeer ? |
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| 12:16.27 | daven | hi. I'm having some problem with video calling working correctly in Asterisk. Between two clients (I am using Ekiga), it works fine directly. I then try and use Asterisk, it connects for a second, and then descends into some form of feedback loop. I have specified in the sip.conf file that it shuold only use alaw, and h261 for the sake of simplicity. Would any of you know what else might be wrong |
| 12:16.44 | daven | Nothing odd is showing up in the asterisk console. |
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| 13:29.15 | Georger86 | hello |
| 13:29.31 | *** join/#asterisk netstv (n=stv@phoenix.u4eatechinc.com) |
| 13:30.17 | netstv | anyone seen this..? gcc -O -o mxmldoc mxmldoc.o -L. -lmxml |
| 13:30.17 | netstv | gcc: mxmldoc.o: No such file or directory |
| 13:30.27 | netstv | I get this on both the 1.4 and 1.2 branches. |
| 13:30.39 | netstv | both when I try to cross compile and just regular compile |
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| 13:31.45 | Georger86 | <PROTECTED> |
| 13:31.45 | Georger86 | <PROTECTED> |
| 13:31.53 | smps | netstv, there is no mxmldoc.o file |
| 13:31.55 | Georger86 | a2billing to trunk ? |
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| 13:32.37 | netstv | snmps, I get that, but I don't understand why the makefile isn't making it. I'm not doing anything special |
| 13:32.39 | lmadsen | netstv: looks like it's trying to compile some xml docs? |
| 13:32.44 | Georger86 | <but if i call from a normal extension calls> |
| 13:32.55 | lmadsen | although I haven't seen that happen on my boxes before -- but I also don't cross-compile |
| 13:33.00 | netstv | I'm just doing the make at the top level. It works its way down there. |
| 13:33.21 | netstv | it's inside menuselect |
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| 13:33.32 | netstv | menuselect/mxml |
| 13:33.42 | hsv-al | Hello fellow irc addicts, are we all looking forward to another long & glorious day of irc addiction? :) |
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| 13:35.00 | smps | netstv, find ./ -iname "*mxmldoc*" -print |
| 13:36.11 | netstv | smps, nadda. |
| 13:36.34 | netstv | which is what I was expecting... I believe the line gcc -O -o mxmldoc mxmldoc.o -L. -lmxml makes mxmldoc.o |
| 13:37.15 | netstv | do you have menuselect checked out? |
| 13:37.24 | smps | netstv, that line should make "mxmldoc" binary |
| 13:37.29 | smps | netstv, from mxmldoc.o |
| 13:37.55 | netstv | smps, yeah.. you are right.. now I need to go figure out why mxmldoc.o isn't being built... |
| 13:38.29 | netstv | smps, do you build the menuselect stuff? Or when you do a ./configure at the top level do you leave it out? I don't think I need it |
| 13:39.17 | smps | netstv, how do you mean "menuselect stuff" ? |
| 13:40.10 | netstv | well I just did a svn checkout of the 1.4 branch and 1.2 branch and the menuselect is part of the branch. |
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| 13:40.24 | smps | netstv, yep of 1.4 |
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| 13:41.28 | netstv | smps, and I'm not sure but I thought I read somewhere that menuselect is a helper for the gui somewhere.. I'm probably wrong however |
| 13:41.40 | SteveTotaro | anyone know how to get ahold of bkw? |
| 13:42.24 | smps | netstv, hmm, you use menuselect just to select what modules etc... should be compiled in that distro of asterisk |
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| 13:42.55 | netstv | smps, ah... i see... so why when I do a make at the top level does it try to build menuselect? |
| 13:43.19 | netstv | smps, I need to build it I'm sure |
| 13:43.25 | ManxPower | netstv: menuselect is the curses interface to setting the compile options and settings |
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| 13:43.59 | ManxPower | It was introduced in 1.4 |
| 13:45.45 | netstv | ManxPower/smps: This is what I did.. I do a checkout from svn ... then I do a ./configure... all is well, I get the cute asterisk logo at the end... and then I do a make.... and I run into these problems... what I am doing wrong? |
| 13:46.21 | ManxPower | netstv: since I arrived after you told us what your problem is..... |
| 13:47.57 | Oy90 | netstv: what linux/gcc are you using? |
| 13:48.13 | netstv | hang on..checking. |
| 13:48.30 | netstv | gcc version 4.1.2 20070626 (Red Hat 4.1.2-14) |
| 13:48.35 | netstv | I'm on CentOS |
| 13:48.53 | netstv | hold on.. I just did a make clean at the root and it seems to be working.. |
| 13:48.54 | netstv | crap |
| 13:49.34 | SteveTotaro | ./bootstrap, ./configure, make menuselect |
| 13:49.55 | SteveTotaro | anyone know if brian@bkw.org is still valid? |
| 13:49.55 | ManxPower | netstv: why are you using SVN Asterisk anyway? |
| 13:50.21 | SteveTotaro | why not |
| 13:50.23 | ManxPower | SteveTotaro: bkw is not involved in this channel very much anymore. |
| 13:50.37 | SteveTotaro | i see him here sometimes |
| 13:50.44 | SteveTotaro | but that was not my question |
| 13:51.26 | netstv | ManxPower: basically we are going to be putting Asterisk in our embedded system and before I get my developers to join the community and start working on this, I wanted to get the idea of how this works first. |
| 13:51.54 | netstv | I will have probably two of my guys join the community and contribute etc... |
| 13:52.03 | netstv | at least two. |
| 13:52.16 | ManxPower | netstv: I would think you would want to use a known released version of Asterisk rather than the version of Asterisk that is under development and may be broken at any time. |
| 13:52.30 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
| 13:52.39 | SteveTotaro | he is looking to do dev work |
| 13:52.56 | netstv | ManxPower: Not really. I like to live on the edge until we get close to a release. Then I will lock everything down. |
| 13:53.07 | netstv | SteveTotaro: you are correct |
| 13:53.10 | ManxPower | netstv: you must not live in the corporate world. |
| 13:53.39 | netstv | ManxPower: I do. But I run the Advanced Technology Group.. we are rebels in the corporate world! :) |
| 13:54.11 | ManxPower | At least you are not trying to put the SVN version into production |
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| 13:54.24 | SteveTotaro | here is a tip, get asterisk working and then move to the dev channel |
| 13:54.24 | netstv | LOL!!!!! |
| 13:54.45 | ManxPower | SteveTotaro: Wow, we agree on something. |
| 13:54.52 | ManxPower | Look! There's a flying pig! |
| 13:54.57 | netstv | SteveTotaro: I already have the asterisk working (1.2) on a linux box running CentOS... |
| 13:55.13 | netstv | Now I have to cross compile and put on my board |
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| 13:55.20 | SteveTotaro | we agree on alot, but our heads are too big and bump alot |
| 13:55.28 | ManxPower | netstv: now you are getting into more #asterisk-dev stuff |
| 13:55.37 | netstv | ah.. ok |
| 13:55.48 | netstv | thanks.. I'll join that channel. |
| 13:55.56 | SteveTotaro | not really if you don't know what make menuselect does |
| 13:56.14 | ManxPower | netstv: The mailing lists are also a great resource. |
| 13:56.15 | SteveTotaro | you need the basics down anyways or the devs are not too nice |
| 13:56.33 | ManxPower | SteveTotaro: Yes, obviously he needs to know the user stuff before he starts on devel stuff. |
| 13:56.36 | netstv | well that's true.. I just have to figure out what all the little things do. Oh.. I'm use to them.. :) Been doing this for 20 years.. I don't take it personally |
| 13:56.58 | ManxPower | netstv: I assume you read both the 1.2 UPGRADE.txt as well a the 1.4 UPGRADE.txt? |
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| 13:57.09 | netstv | Yep.. quickly however... |
| 13:57.34 | netstv | When a basic build doesn't work I'm usually doing something wrong which is why I go to the irc channels |
| 13:57.38 | SteveTotaro | docs are for people with low IQs |
| 13:57.55 | SteveTotaro | make clean |
| 13:58.03 | SteveTotaro | ./bootstrap |
| 13:58.07 | ManxPower | Docs are for people too polite to ask stupid questions on this channel. |
| 13:58.07 | SteveTotaro | ./configure |
| 13:58.09 | netstv | SteveTotaro: I agree.. I usually read the Makefiles and scripts.. and go from there. |
| 13:58.13 | SteveTotaro | make menuselect |
| 13:58.41 | SteveTotaro | i guess when your IQ is over 140 you would understand |
| 13:59.02 | SteveTotaro | then make && make install |
| 13:59.11 | netstv | according to the README at the top level of the 1.4 branch, it says nothing about running ./bootstrap which is why I didn't run it. |
| 13:59.26 | SteveTotaro | see, i don't read the stupid docs |
| 13:59.31 | netstv | LOL |
| 13:59.45 | netstv | off to try and get my cross compile to work |
| 13:59.46 | SteveTotaro | try bootstrap after a make clean |
| 14:00.37 | netstv | yeah I just did.. says my Autoconf 2.6 or higher is required.. ugh.. playing the upgrade game now |
| 14:00.47 | SteveTotaro | easy to fix |
| 14:00.56 | ManxPower | netstv: line 94 of the README |
| 14:01.08 | ManxPower | and you don't need bootstrap unless you are using SVN |
| 14:01.16 | SteveTotaro | he is using svn silly |
| 14:01.30 | ManxPower | SteveTotaro: Correct. |
| 14:01.56 | ManxPower | the make menuselect is optional according to the README |
| 14:02.25 | SteveTotaro | it is optional unless you want all the voice prompts in the various codecs like i do |
| 14:02.32 | SteveTotaro | and chan_mobile and such |
| 14:03.17 | SteveTotaro | why transcode prompts when you don't have to? |
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| 14:07.20 | smps | netstv, for what platform are you cross-compiling ? |
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| 14:11.02 | thepacmanfan | if i have a phone set up as [6001] in sip.conf, shouldn't i just be able to dial 6001@context? |
| 14:11.17 | ManxPower | thepacmanfan: no. |
| 14:11.48 | ManxPower | Dial(SIP/6001) i.e. Dial(SIP/sipconfentry) |
| 14:11.59 | thepacmanfan | well, i've got a global defining my phone as SIP/6001 |
| 14:12.00 | netstv | smps, mips |
| 14:12.01 | [TK]D-Fender | thepacmanfan: extensions.conf configures EVERYTHING you "dial" |
| 14:13.50 | casix | Rico29 sorry I was eating. Are you here? |
| 14:14.32 | hsv-al | d-fender |
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| 14:14.40 | Rico29 | yes |
| 14:14.52 | hsv-al | get your coffee, get your crackerbarrel breakfest,,,,,and get empowered, for another day of helping :) |
| 14:15.03 | casix | you can make a proces listening to the manager all the time |
| 14:15.11 | casix | and recieve the registering events |
| 14:15.55 | casix | and then see what kind of terminal is and do the autoprovisioning file |
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| 14:17.37 | Rico29 | ok |
| 14:17.43 | Rico29 | i've got another question |
| 14:18.12 | Rico29 | in my sip.conf, I have : autocreatepeer=yes and regcontext=sip_autoreg |
| 14:18.18 | Rico29 | the phone registers |
| 14:18.44 | Rico29 | in my extension.conf : |
| 14:18.44 | Rico29 | [sip_autoreg] |
| 14:18.44 | Rico29 | exten => 123,1,AGI(login.pl) |
| 14:19.05 | Rico29 | when I call 123 with my phone |
| 14:19.20 | Rico29 | it says in the CLI : "rejected because extension not found" |
| 14:20.01 | glaz | did you include sip_autoreg in your phone context? |
| 14:20.23 | SteveTotaro | is that all the cli output? |
| 14:20.27 | SteveTotaro | it should work |
| 14:20.41 | Rico29 | ... but it doesn't :) |
| 14:20.41 | netstv | sorry ManxPower and SteveToTaro.. I'm back now.. |
| 14:20.55 | Rico29 | did you include sip_autoreg in your phone context? <- phone context ? |
| 14:20.57 | netstv | thanks for your help.. I'm going to get started now on the cross compile stuff |
| 14:21.49 | Rico29 | what is the "phone context" ? |
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| 14:23.12 | Rico29 | glaz> SteveTotaro> an idea ? |
| 14:24.22 | ManxPower | Rico29: It should say much more than just "rejected because extension not found" |
| 14:24.39 | ManxPower | Rico29: what is the context= line in sip.conf for that phone? |
| 14:24.48 | ManxPower | (the phone PLACING the call) |
| 14:25.08 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
| 14:25.08 | Rico29 | [May 21 16:36:49] NOTICE[3679]: chan_sip.c:13952 handle_request_invite: Call from '' to extension '123' rejected because extension not found. |
| 14:25.28 | [TK]D-Fender | Rico29: regcontext isn't used as the context for autocreatepeer. They have nothing to do with each other |
| 14:25.29 | Rico29 | ManxPower> there is no line ! the phone registers with "autocreatepeer" |
| 14:25.30 | ManxPower | Rico29: now answer my 2nd question |
| 14:25.44 | Rico29 | [TK]D-Fender> ah ok, i thought it was... |
| 14:25.54 | ManxPower | Rico29: ah, I can't help you with any autocreating of peers |
| 14:25.55 | [TK]D-Fender | Rico29: your autocreatepeer use the "context=" from general |
| 14:26.06 | Rico29 | ok !!! thanks a lot |
| 14:26.15 | [TK]D-Fender | Rico29: Nothing says it does. You should reconsider how you come up with these assumptions |
| 14:26.36 | [TK]D-Fender | Rico29: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext |
| 14:27.07 | [TK]D-Fender | Rico29: While the WIKI might not always be accurate, if things aren't working you should be checking there. |
| 14:27.55 | Rico29 | ok |
| 14:28.15 | Rico29 | it works, thanks |
| 14:30.10 | thepacmanfan | what would cause my SIP phone to return 404 when asterisk sends it an INVITE (while dialing that extension) |
| 14:30.48 | Rico29 | 404, not found I think it's like "no matching peer found" |
| 14:30.51 | Rico29 | check your sip.conf |
| 14:31.15 | ManxPower | thepacmanfan: usually means extension not found in the searched context |
| 14:31.23 | thepacmanfan | but my phone responds... |
| 14:31.27 | ManxPower | thepacmanfan: that means nothing |
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| 14:31.50 | ManxPower | thepacmanfan: what is the context= for the calling sip.conf entry? |
| 14:32.27 | bobbym | how can i perform some operation after the call hangups using AGI? |
| 14:32.49 | thepacmanfan | ManxPower, the context for the phone, in sip.conf, is my outgoing dialplan |
| 14:33.07 | ManxPower | bobbym: "core show application deadagi" and research the "g" option of Dial and the "h" extension. |
| 14:33.40 | ManxPower | thepacmanfan: and the actual context= line in sip.conf for the calling device is? |
| 14:33.42 | bobbym | ManxPower: i'm using the deadagi but there is another option? |
| 14:33.47 | ManxPower | bobbym: no |
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| 14:33.53 | bobbym | ManxPower: i'm concern about the performance of it... |
| 14:33.55 | bobbym | ManxPower: a ok.... |
| 14:33.57 | bobbym | ManxPower: thanks |
| 14:34.44 | thepacmanfan | context=numberplan-custom-1 |
| 14:34.57 | ManxPower | thepacmanfan: what number are you dialing that causes that message? |
| 14:35.07 | thepacmanfan | ManxPower: here's the beginning of the response from the phone, as shown by SIP debug |
| 14:35.15 | ManxPower | thepacmanfan: put it on pastebin |
| 14:35.48 | [TK]D-Fender | ~pb |
| 14:35.50 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:35.57 | ManxPower | and if you don't want my help you could say so, rather than ignore my questions, which are designed to get a good understanding of your problem as well as find a solution.. |
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| 14:36.32 | thepacmanfan | ManxPower: here's the full output, then: http://pastebin.com/d152867a8 |
| 14:36.43 | thepacmanfan | sorry, i don't mean to ignore questions. |
| 14:36.49 | bobbym | ManxPower: just more one question dude, which is the best performatic way to create agi. i'm thinkin in php ? |
| 14:36.56 | bobbym | ManxPower: there is that asterisk-java tooo |
| 14:36.57 | thepacmanfan | i'm doing dial 6001@local in console |
| 14:37.19 | thepacmanfan | dial 6001@internal pretty much serves the same purpose right now |
| 14:37.29 | ManxPower | thepacmanfan: no, it does not. |
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| 14:37.55 | jjshoe | bobbym I perfer using perl for AGI's, but use whichever language you know. |
| 14:38.17 | thepacmanfan | ManxPower: they both end up dialing SIP/6001, so for the purposes of testing a call to an extension.... |
| 14:38.23 | bobbym | jjshoe: thanks! |
| 14:38.45 | muiro | I'm having trouble with my unixODBC connection through asterisk. Basically, the first call of the day, or the first call in a few hours, takes a long time when it makes the first query. Should I attempt to fix this by modifying a "keepalive" variable in my unixodbc setup, something in my res_odbc.conf? Also, is there an asterisk cli command I can use to poke the connection every now and again? |
| 14:38.47 | ManxPower | thepacmanfan: best of luck with your problem. |
| 14:39.20 | thepacmanfan | anyhow, the phone seems to be returning 404, and the console says busy/congestion |
| 14:40.02 | ManxPower | the phone is saying extension 1234 does not exist. |
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| 14:40.30 | thepacmanfan | how's that? |
| 14:40.36 | thepacmanfan | 1234 should be the username... |
| 14:40.48 | ManxPower | no. 6001 is the username |
| 14:41.06 | ManxPower | If 1234 was the username then you would Dial(SIP/6001) |
| 14:41.06 | tzanger | bought a merlin hsdpa expresscard. bleh. |
| 14:41.16 | [TK]D-Fender | ManxPower: INVITE sip:1234@192.168.1.175 SIP/2.0 <-- |
| 14:41.22 | [TK]D-Fender | ManxPower: he set the username. |
| 14:41.31 | ManxPower | [TK]D-Fender: looks to me that is the phone IP |
| 14:41.34 | [TK]D-Fender | thepacmanfan: And I'm betting you jsut set your phone up wrong. |
| 14:41.47 | [TK]D-Fender | ManxPower: To: <sip:1234@192.168.1.175> |
| 14:42.02 | [TK]D-Fender | ManxPower: pretty clear in his pastebin that 1234 is the username for SIP/6000 |
| 14:42.05 | thepacmanfan | [TK]D-Fender: but the phone can place outgoing calls... |
| 14:42.25 | [TK]D-Fender | thepacmanfan: feel free to show us that in a separate pastebin |
| 14:42.30 | thepacmanfan | i mean, once i have the extensions configured n'stuff :P |
| 14:42.42 | [TK]D-Fender | thepacmanfan: and thanks for not showing us your sip.conf entry for the phone... |
| 14:42.51 | ManxPower | thepacmanfan: incoming and outgoing calls are TOTALLY different. They could even be handled by two different servers on two different netowkrs |
| 14:43.03 | [TK]D-Fender | thepacmanfan: Yeah, and I'm rich... once I win the lottery. |
| 14:43.23 | [TK]D-Fender | thepacmanfan: please try to keep within THIS space/time continuum..... |
| 14:43.30 | ManxPower | [TK]D-Fender: thepacmanfan as you can see it's one of those that ignores your request for information then either argues with you or provides you with information you didn't ask for. |
| 14:43.31 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:43.35 | ManxPower | You can support the guy. |
| 14:43.48 | [TK]D-Fender | ManxPower: debatable.... |
| 14:43.51 | jjshoe | you guys could all stand to relax a smidge :) |
| 14:44.17 | jjshoe | will give free hugs |
| 14:44.22 | ManxPower | jjshoe: We are volunteers. It's rude to expect support and then not even follow the instructions of the person trying to help. |
| 14:44.54 | jjshoe | You're volunteers, rather then bitch, don't help. |
| 14:45.11 | ManxPower | jjshoe: *nod* Hence my "best of luck" comment. |
| 14:45.28 | ManxPower | That's ManxPower code for "you are an idiot, I'm not helping you anymore" |
| 14:45.41 | thepacmanfan | here's the result when i try to dial 6000 from the phone, via the console. |
| 14:45.41 | thepacmanfan | http://pastebin.com/d2d77689a |
| 14:46.22 | Rico29 | what is the syntax for "STREAM FILE" in an AGI ? i checked on voip-info, but doesn't work |
| 14:46.25 | ManxPower | Looking for 6000 in incoming (domain 192.168.1.27) |
| 14:46.30 | MaartenB | I am having trouble with disconnections when I transfer somebody to another phone, could anybody please help? |
| 14:46.38 | thepacmanfan | here's the phone settings as seen in sip.conf: http://pastebin.com/d753771e9 |
| 14:46.40 | ManxPower | MaartenB: perhaps if we had more information |
| 14:47.24 | MaartenB | ManxPower, it are SPA942 on two different NAT's, the server is a server with a public IP in a datacentre |
| 14:47.38 | thepacmanfan | ManxPower: my point here is that Asterisk isn't rejecting the phone due to incorrect settings, so i'm wondering if there's any way my phone could be set incorrectly. |
| 14:47.58 | ManxPower | thepacmanfan: best of luck with that |
| 14:48.08 | thepacmanfan | thanks. |
| 14:48.15 | MaartenB | ManxPower, clients are connected with SIP, nat=yes, canreinvite=nonat |
| 14:48.23 | Rico29 | <PROTECTED> |
| 14:48.30 | ManxPower | MaartenB: try canreinvite=no |
| 14:48.49 | Rico29 | i wrote : $AGI->STREAM FILE vm-login; |
| 14:48.53 | MaartenB | ManxPower, canreinvite=no and =yes produce the same results, already tried that |
| 14:48.53 | Rico29 | mmh ? |
| 14:48.56 | thepacmanfan | IP address, username, secret... i don't know what else there is to match. |
| 14:48.56 | ManxPower | Rico29: what language, which AGI library? |
| 14:49.08 | [TK]D-Fender | thepacmanfan: Found no matching peer or user for '192.168.1.175:50777' <--- your phone ISN'T getting matched |
| 14:49.10 | Rico29 | perl |
| 14:49.13 | ManxPower | MaartenB: you will want to leave canreinvite=no to avoid future problems |
| 14:49.15 | [TK]D-Fender | thepacmanfan: the call is coming in unauthed. |
| 14:49.19 | Rico29 | with asterisk-perl lib |
| 14:49.25 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 14:49.27 | [TK]D-Fender | thepacmanfan: From: "Andrew" <sip:line1@192.168.1.27>;tag=003094c2d043049c1b68c674-601a57dd |
| 14:49.32 | ManxPower | Rico29: then check the docs for that |
| 14:49.35 | MaartenB | ManxPower, ok, I do that, but it does not solve the problem :( |
| 14:49.48 | ManxPower | MaartenB: Is Asterisk behind NAT? |
| 14:49.49 | [TK]D-Fender | thepacmanfan: Your phone is not listed as 1234 like you claimed, but rather "line1" |
| 14:50.01 | [TK]D-Fender | thepacmanfan: Go reread the instructions on who to set it up. |
| 14:50.12 | [TK]D-Fender | how* |
| 14:50.35 | MaartenB | ManxPower, no, it is in a datacentre and on a public IP |
| 14:50.51 | ManxPower | MaartenB: Is there any firewall or packet filtering? |
| 14:51.17 | MaartenB | ManxPower, fyi, doing a blind transfer with the # works, doing an announced transfer with xfer fails |
| 14:51.45 | ManxPower | MaartenB: why are you using the DTMF # hack rather than the built in transfer feature of your phone? |
| 14:51.45 | MaartenB | ManxPower, yes, there is a iptables firewall, but the both IP's involved are both whitelisted and not blocked in any way |
| 14:51.50 | thepacmanfan | good catch [TK]D-Fender, thanks. |
| 14:52.05 | MaartenB | ManxPower, because it is the only one that works? |
| 14:52.16 | ManxPower | MaartenB: but it's not working, is it. |
| 14:52.20 | MaartenB | ManxPower, somehow my clients didn't appreciate being disconnected :) |
| 14:52.44 | MaartenB | DTMF # does work, using the transfer feature of my phone does not |
| 14:53.03 | MaartenB | I changed every setting on the phone, spent a day trying different things, but nothing seems to solve it |
| 14:53.16 | ManxPower | MaartenB: I hope you solve your problem. I don't really help people using the DTMF # hack for transfers. BTW, the native phone transfer failing is a symptom if a significant problem with your asterisk or your phone setup. |
| 14:53.39 | MaartenB | ManxPower, I do not want to use the # hack! |
| 14:53.49 | MaartenB | ManxPower, but it is the only thing that works at the moment, I want to get rid of it |
| 14:53.56 | ManxPower | MaartenB: good. Now what happens when you try to use the phone blind transfer? |
| 14:54.30 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 14:54.58 | MaartenB | ManxPower, line gets disconnected after a very short period (within < 2 seconds) |
| 14:55.25 | ManxPower | MaartenB: what is shown on the CLI? use pastebin.ca for pasting more than 2 lines. |
| 14:56.23 | MaartenB | <PROTECTED> |
| 14:56.30 | MaartenB | for both lines |
| 14:57.09 | ManxPower | MaartenB: and when you get that exact message, what number are you trying to transfer the call to? |
| 14:57.13 | Rico29 | excluding festival, is there another way to make asterisk working with a vocale synth ? |
| 14:57.22 | ManxPower | Rico29: yes |
| 14:57.30 | Rico29 | i heard one, but I can't remember it name |
| 14:57.40 | ManxPower | Rico29: perhaps Cepstral ? |
| 14:57.44 | Rico29 | yes ! |
| 14:57.47 | Rico29 | thanks |
| 14:57.59 | Rico29 | I heard it is better than festival |
| 14:58.03 | Rico29 | is it ? |
| 14:58.04 | ManxPower | Rico29: Odd that your google search of the mailing list archives did not come up with that info. |
| 14:58.39 | Rico29 | i don't understand, sorry |
| 14:58.43 | Rico29 | (fr...) |
| 14:58.48 | ManxPower | ~mailinglist |
| 14:58.49 | jbot | hmm... mailinglist is Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 14:58.51 | MaartenB | ManxPower, I am trying to transfer the call to extension 1, I get that message when the calls are disconnected, after the tranfser |
| 14:58.55 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:59.05 | Rico29 | ok |
| 14:59.12 | ManxPower | Rico29: next time search the mailing list archives first. There is an incredible amount of information there. |
| 14:59.21 | jameswf-home | ping tzafrir |
| 14:59.28 | tzafrir | jameswf-home, pong |
| 14:59.29 | glaz | Rico29: il a dit, que c'était triste que la recherche que tu as fait sur google ne t'as pas donné ce que tu cherchais. |
| 14:59.33 | Rico29 | ok, i didn't knew |
| 14:59.36 | ManxPower | MaartenB: pastbin the info for extension 1 in the context [local] in extensions.conf |
| 15:00.13 | ManxPower | Rico29: sources of documentaion /path/to/src/asterisk/doc the mailing list archives, the wiki, and finally this channel, in that order. |
| 15:00.24 | Rico29 | ok ! |
| 15:00.55 | thepacmanfan | [TK]D-Fender: thanks again! |
| 15:01.24 | MaartenB | ManxPower, http://pastebin.com/m3f1438cc |
| 15:01.58 | ManxPower | MaartenB: now show more of the failed CLI message |
| 15:02.19 | ManxPower | I need to see the information BEFORE the "Spawn extension (local, 1, 3) exited non-zero on SIP/blah |
| 15:02.29 | [TK]D-Fender | Rico29: Pas que c'est triste mais que c'est etrange qu tu ne le pas vue. plus a dire que il croit mem pas que t'as essaye |
| 15:02.38 | anonymouz666 | WTF |
| 15:02.43 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:02.45 | anonymouz666 | [TK]D-Fender speaking another language! |
| 15:03.09 | [TK]D-Fender | anonymouz666: Va-t'ens tabarnac :p |
| 15:03.31 | MaartenB | ManxPower, ok |
| 15:03.37 | Rico29 | [TK]D-Fender> je ne connaissais vraiment pas |
| 15:03.49 | TGALVIN | Hello all |
| 15:03.54 | [TK]D-Fender | Rico29: rien a savoir. Il faut CHERCHER <- |
| 15:04.11 | Rico29 | ok |
| 15:04.19 | [TK]D-Fender | Rico29: Au mois t'es pas un <<nul>> :) |
| 15:04.33 | glaz | Au moin. |
| 15:04.39 | Rico29 | au moins |
| 15:04.40 | Rico29 | :) |
| 15:04.40 | TGALVIN | I have an AA50 question, is the appropriate channel |
| 15:05.16 | [TK]D-Fender | Rico29: Orthographe est correct, clavardage.... ca c'est un autre affaire ;) |
| 15:05.35 | Rico29 | ok |
| 15:05.43 | [TK]D-Fender | TGALVIN: maybe, ask and find out |
| 15:05.49 | Rico29 | are you american or canadian ? or other? |
| 15:05.57 | [TK]D-Fender | Rico29: Canadian. |
| 15:06.01 | glaz | He's a Maple Leaf fan! |
| 15:06.16 | Rico29 | ok |
| 15:06.34 | TGALVIN | I am using it for a small office(8 people) on a MPLS T1 line, and would like to use G.729 |
| 15:06.53 | TGALVIN | Is it possible |
| 15:06.57 | [TK]D-Fender | glaz: Know what the difference is between the Montreal Canadians and the Toronto Maple Leafs? The Canadians last Stanley Cup photo is in COLOUR <- |
| 15:07.23 | glaz | hehe |
| 15:07.29 | [TK]D-Fender | TGALVIN: Last I checks the AA50 didn't do t1, only analog. |
| 15:07.40 | [TK]D-Fender | checked* |
| 15:07.45 | glaz | Wasn't it in 1976 the last Maple Leafs Stanley cup? |
| 15:07.52 | Rico29 | is there an admin of voip-info.org here ? |
| 15:08.14 | [TK]D-Fender | Rico29: doubt it. |
| 15:08.24 | Rico29 | in french ? |
| 15:08.32 | jameswf-home | admin? alot of us edit voip-info it is a wiki |
| 15:08.33 | Rico29 | j'en doute ? |
| 15:08.37 | Rico29 | ok |
| 15:08.42 | Rico29 | there is a dead link |
| 15:08.49 | jameswf-home | so remove it |
| 15:08.50 | Rico29 | http://www.voip-info.org/wiki/index.php?page=App_Cepstral |
| 15:08.52 | glaz | Yes doubt it means j'en doute. |
| 15:08.53 | [TK]D-Fender | Rico29: go fix it yourself. Anyone can register |
| 15:09.01 | Rico29 | ok |
| 15:09.07 | TGALVIN | The T1 is connected to a Cisco 2811 |
| 15:10.21 | TGALVIN | calls are only over Ethernet using SIP and G.711 at the moment |
| 15:10.28 | [TK]D-Fender | TGALVIN: Where is the Cisco located relative to the AA50? |
| 15:10.45 | [TK]D-Fender | TGALVIN: And where are you phones relative to *? |
| 15:10.47 | TGALVIN | Both are connected to a POE switch |
| 15:11.00 | [TK]D-Fender | TGALVIN: Then why on earth would you want to use G.729? |
| 15:11.11 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 15:11.57 | TGALVIN | to route calls to Global Crossing. The T1 is MPLS with Global crossing acting as our connection to the PSTN |
| 15:12.29 | mort_gib | TK I added G.729 to one of my clients install, without telling them, they told me a few days later that "Sound quality was better" -All on the same LAN/Switch |
| 15:12.40 | [TK]D-Fender | TGALVIN: So your T1 is DATA only? |
| 15:12.54 | TGALVIN | one T1 for intenet access, and SIP Trunks with Priority |
| 15:12.54 | [TK]D-Fender | mort_gib: Compared to? |
| 15:13.08 | TGALVIN | yep |
| 15:13.10 | [TK]D-Fender | TGALVIN: AH, thought you had split voice/data on that. |
| 15:13.10 | Rico29 | anyone of you has the app_cepstral_1.4.tar.gz file ? |
| 15:13.18 | Rico29 | I cant find it anywhere |
| 15:13.39 | MaartenB | ManxPower, I added some more info so it cost some time |
| 15:13.40 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 15:13.44 | [TK]D-Fender | TGALVIN: I might call Digium to see how many channels the AA50 can transcode. Its an embedded platform and G.729 is processor intensive. |
| 15:13.45 | MaartenB | ManxPower, http://pastebin.com/d3a625920 |
| 15:14.03 | TGALVIN | G.729 will allow for more efficient bandwidth utilization. |
| 15:14.10 | mort_gib | TK: Before :-) |
| 15:14.12 | mort_gib | ulaw |
| 15:14.25 | [TK]D-Fender | TGALVIN: I know, and I see its importance as you are going over the internet for VoIP. |
| 15:14.32 | TGALVIN | thanks |
| 15:14.38 | [TK]D-Fender | mort_gib: that is completely retarded |
| 15:14.53 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:15.01 | mort_gib | I didn't promise that I or my clients were sane.... |
| 15:15.05 | mort_gib | :-) |
| 15:15.18 | mort_gib | I'm as confused as you! |
| 15:15.48 | Rico29 | i find it |
| 15:16.08 | TGALVIN | Digium does not support the configuration. I figured I would ask here too. |
| 15:18.06 | MaartenB | ManxPower, still here? |
| 15:22.49 | [TK]D-Fender | TGALVIN: They SHOULD be able to answer "Does the AA50 support G.729 transcoding, and if so how many channels is it capable of supporting?" |
| 15:24.40 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 15:26.05 | kannan | for TE410P, can we safely use the software over ride as per the wiki zaptel config page, rather than set the card's jumper to E-1 ? or is it always advisable to se the jumper only |
| 15:27.12 | TGALVIN | Thanks |
| 15:27.49 | [TK]D-Fender | kannan: Se the jumper |
| 15:28.17 | kannan | [TK]D-Fender : thanks |
| 15:30.43 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 15:40.17 | *** join/#asterisk [manimal] (n=[manimal@69-196-131-215.dsl.teksavvy.com) |
| 15:45.50 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 15:49.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:53.40 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:54.49 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 15:56.10 | murdock_ut | Howdy All. |
| 15:56.11 | thepacmanfan | [TK]D-Fender: it sure was auth issues... got it fixed. thanks. |
| 15:57.00 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 15:58.24 | murdock_ut | Quick question, why when a person is on hold do I see this message show up on the * console? RTCP SR transmission error, rtcp halted |
| 15:58.31 | murdock_ut | I have canreinvite=no |
| 15:58.42 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:58.49 | murdock_ut | It doesn't seem to break anything. |
| 15:59.28 | murdock_ut | * version 1.4.20 |
| 16:02.10 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 16:03.04 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 16:03.50 | disposable | context A includes context B. Context C includes context A. does context C include context B as well? are includes recursive? |
| 16:04.01 | SteveTotaro | oil execs testifying now |
| 16:04.42 | *** part/#asterisk redback (n=noname@mail.datadream.co.uk) |
| 16:05.56 | *** join/#asterisk AndyGraybeal (n=AndyGray@128-177-27-78.ip.openhosting.com) |
| 16:06.18 | [TK]D-Fender | disposable: Yes C includes both |
| 16:06.35 | [TK]D-Fender | disposable: the term is "transitive inheritance" |
| 16:09.21 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584781.dsl.bell.ca) |
| 16:10.19 | *** join/#asterisk s0lid (n=s0lid@58.69.107.19) |
| 16:10.53 | *** part/#asterisk fiddur (i=fiddur@78.82.252.209) |
| 16:15.24 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:20.06 | maqr | so i set my sip phone to use ulaw only, and i downloaded the ulaw sounds from digium's ftp, but Playback still doesn't work... any suggestions as to what i could be doing wrong? |
| 16:20.12 | maqr | if i force gsm, it does work |
| 16:21.53 | hardwire | force gsm in sip.conf? |
| 16:25.22 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:25.58 | *** join/#asterisk s0lid (n=s0lid@58.69.107.19) |
| 16:28.18 | maqr | hardwire: err, no, i don't want to use gsm, because it sounds bad |
| 16:28.39 | CCFL_Man2 | Strom_M: you here? |
| 16:28.39 | [TK]D-Fender | maqr: I suggest you consider that you aren't showing us the problem and the configs related to it and shouldn't expect much help. |
| 16:29.13 | hardwire | maqr: how did you force GSM.. at the phone side or in sip.conf? Or by simply using .gsm files only. |
| 16:29.51 | disposable | [TK]D-Fender: thank you. is there any way to avoid 'transitive inheritance'? |
| 16:29.52 | maqr | hardwire: well, i forced gsm in the sip conf with "disallow=all" and "allow=gsm", but when i replace "gsm" with "ulaw", and replace the gsm sounds with ulaw sounds, i don't get any playback |
| 16:30.13 | [TK]D-Fender | disposable: No, you can merely choose better contexts for your includes. |
| 16:30.18 | znoG | anyone know the root cause of this error msg: Hanging up call xxxxx - no reply to our critical packet (this is after a Maximum retries exceeded on transmission error) |
| 16:30.23 | disposable | [TK]D-Fender: thanks |
| 16:30.35 | CunningPike | Does anyone have any opinions on SipX vs. SER for UDP to TCP conversion? |
| 16:31.05 | CCFL_Man2 | wow, the diaphram is glued in on this candlestick phone |
| 16:31.07 | maqr | [TK]D-Fender: i'll post my relevant configs, one sec |
| 16:31.37 | hardwire | maqr: does asterisk complain when it tries to read them? |
| 16:31.45 | hardwire | maqr: did you test it with .wav? |
| 16:32.05 | maqr | hardwire: asterisk says it's playing them, but no sound comes out |
| 16:32.06 | hardwire | CCFL_Man2: candlestick? |
| 16:32.15 | hardwire | maqr: is it isolated to just that phone? |
| 16:32.18 | maqr | hardwire: and no, i didn't test it with wavs, but it's easy enough for me to put them in the directory too, i'll do that |
| 16:32.30 | maqr | hardwire: no idea, this is my first attempt at setting up asterisk, i have exactly one softphone and no itsp :) |
| 16:32.36 | CCFL_Man2 | hardwire: vintage candlestick tehephone |
| 16:32.44 | hardwire | CCFL_Man2: I guess I still don't konw what that is. |
| 16:33.01 | hardwire | google ftw |
| 16:33.16 | hardwire | ahha.. cool |
| 16:33.24 | CCFL_Man2 | you never seen candlestick phones? |
| 16:33.25 | mort_gib | znoG: Have a look at network issues.... |
| 16:33.49 | hardwire | CCFL_Man2: I've never heard of it called that. |
| 16:34.03 | hardwire | CCFL_Man2: or I've never meant to remember it's called that - seeing as what century I live in and all. |
| 16:34.30 | CCFL_Man2 | i lived in the same century as you :P |
| 16:34.45 | hardwire | what are the odds that not everybody cares? |
| 16:34.53 | znoG | mort_gib: such as? i can make outbound calls just fine, inbound calls are the problem |
| 16:35.18 | hardwire | CCFL_Man2: so a real vintage one, or a newer model? |
| 16:35.20 | CCFL_Man2 | hardwire: fail |
| 16:35.50 | [TK]D-Fender | znoG: Just a stretch, let me guess... your phone is behind NAT relative to *? |
| 16:36.18 | CCFL_Man2 | hardwire: yep, western electric 51AL |
| 16:36.33 | CCFL_Man2 | i got two other vintage phones too |
| 16:36.52 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:37.43 | CCFL_Man2 | i'm going to connect them to voip using a channel bank |
| 16:38.07 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 16:38.19 | mort_gib | znoG: how do you make inbound calls?? |
| 16:39.40 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 16:40.40 | maqr | [TK]D-Fender: http://pastie.caboo.se/200913 |
| 16:40.51 | maqr | on the phone, i just hear nothing |
| 16:41.44 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 16:43.27 | [TK]D-Fender | maqr: Don't use pastie for your PB's (bleh), and next time increase the debug so we can see exactly which files are being called. Also where is your phone relative to *? |
| 16:43.58 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 16:44.16 | hardwire | maqr: which softphone? |
| 16:44.23 | maqr | [TK]D-Fender: i set verbose to 5, how else to i increase debug? and my phone is on my comcast account, it's an x-lite sip softphone, and my asterisk box is on a VPS at linode |
| 16:44.31 | maqr | hardwire: x-lite 3.0 |
| 16:44.43 | hardwire | maqr: tried another? |
| 16:45.02 | CCFL_Man2 | maqr: cumcast voip? |
| 16:45.55 | [TK]D-Fender | maqr: What setting do get you audio? |
| 16:46.08 | hardwire | [TK]D-Fender: what? |
| 16:46.09 | maqr | hardwire: well, i tried with x-lite 2.0 and forced gsm, and it worked |
| 16:46.43 | *** join/#asterisk aikona (n=soload00@dsl-242-254-56.telkomadsl.co.za) |
| 16:46.44 | maqr | x-lite 3.0 doesn't have gsm, but i'll downgrade again and try ulaw i guess |
| 16:46.50 | maqr | unless someone can recommend a better windows sip phone |
| 16:47.11 | jaytee | my x-lite has gsm |
| 16:47.22 | maqr | jaytee: weird, which version? |
| 16:47.27 | jaytee | 3.0 |
| 16:47.29 | jaytee | for linux |
| 16:47.31 | [TK]D-Fender | maqr: "core set debug 10" |
| 16:48.48 | FuriousGeorge | hey all |
| 16:48.53 | maqr | [TK]D-Fender: that doesn't really show anything, it just shows when it destroys the session |
| 16:48.59 | maqr | jaytee: i thought there was no 3.0 for linux |
| 16:49.04 | hsv-al | fender, does coffee through your veins |
| 16:49.07 | hsv-al | instead of blood? |
| 16:49.08 | FuriousGeorge | my server from hell is behaving very badly today |
| 16:49.15 | hsv-al | flow |
| 16:49.29 | cpm | kick it! |
| 16:49.33 | jaytee | maqr, well if that's the case then I wonder what I downloaded? :-) |
| 16:49.46 | *** join/#asterisk znoG_ (n=gs@host254.190-31-168.telecom.net.ar) |
| 16:50.04 | FuriousGeorge | a call comes in, its answered by *, wait 3, then &dials a bunch of extensions. one picks up, but the media is not connected between the parties, so the channel isnt hung up, but they dont hear eachother |
| 16:50.26 | [TK]D-Fender | hsv-al: I have a continuously validated status as NASA's first caffeine-based life-form. |
| 16:50.55 | FuriousGeorge | this may or may not be related to another problem where after a successful inbound call, asterisk fails to detect hangup, so the channel is left active indefinitely |
| 16:51.09 | hsv-al | I see, well my life liquid isn't blood |
| 16:51.12 | jaytee | maqr, yep it's 3.0. just doublechecked |
| 16:51.14 | hsv-al | its creamy and white |
| 16:51.18 | hsv-al | meth |
| 16:51.28 | hsv-al | wait, that sounded off base heh |
| 16:51.29 | *** part/#asterisk aikona (n=soload00@dsl-242-254-56.telkomadsl.co.za) |
| 16:51.59 | maqr | jaytee: i guess i'm wrong then :) |
| 16:52.09 | FuriousGeorge | to make matters worse the problem seems to persist independent of any hardware i take to the location. when any of these things happen, they call me and tell me the phones are 'haywire'. even restarting asterisk nightly doesnt seem to guarantee a whole day of stable operation |
| 16:52.41 | FuriousGeorge | of course, im still on 1.4.19 |
| 16:52.49 | jaytee | jeez, I roast my own coffee beans, drink about 8 to 10 cups of the stuff a day and yet [TK]D-Fender gets validated by NASA instead of me! |
| 16:53.22 | FuriousGeorge | luckily :) i actually got a core dump yesterday. while that of course will disconnect everyone, at least i have something to file a bug report on |
| 16:53.25 | [TK]D-Fender | jaytee: You drink coffe... I'm MADE of coffee. |
| 16:53.38 | maqr | argh, now gsm doesn't even seem to work |
| 16:54.08 | jaytee | [TK]D-Fender, are you a City, City+, Full City, Full City+ or Expresso roast? |
| 16:54.19 | maqr | can anyone recommend a different windows sip phone? |
| 16:54.42 | jaytee | ~softphone |
| 16:54.42 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 16:55.54 | maqr | downloads zoiper |
| 16:57.02 | outtolunc | mmmms coffee |
| 16:59.22 | jaytee | I'm drinking a custom 50/50 blend of Columbian Huila Valencia ($4.80 lb.) and Kona Kowali Farm ($17.00 lb) roasted to a City+ roast. |
| 16:59.40 | maqr | btw, how can i set verbose 5 and debug 10 in my config so i don't have to type it when i restart the server? |
| 17:00.26 | maqr | also, zoiper works fine with ulaw, and i guess my problem was entirely x-lite |
| 17:01.53 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 17:02.03 | hsv-al | fender |
| 17:02.18 | hsv-al | i bet you drink about 15 cups of "work" provided rocket-fuel tangy cofee, thats as potent as raw gasoline |
| 17:02.25 | maqr | [TK]D-Fender: problem == solved |
| 17:03.03 | TJNII | maqr: You figured it out? |
| 17:03.07 | mwalling | problem.solved = true |
| 17:03.08 | hardwire | is interested |
| 17:03.27 | [TK]D-Fender | hsv-al: in all honesty I go through maybe 4 a day of our free automatic "per-cup" machine prepped coffee. Nothing spectacular. |
| 17:04.07 | maqr | TJNII: it turns out it was all x-lite's fault |
| 17:04.24 | maqr | TJNII: i got the wav and ulaw sound packs, connected with zoiper, and it just works |
| 17:04.33 | maqr | who would have thought x-lite couldn't make a simple phone call? |
| 17:04.34 | TJNII | Oooh. Nothing like spending hours barking up the wrong tree..... |
| 17:04.41 | outtolunc | drinks at least 1 pot of not strong 'staples' brand (whichever it is) |
| 17:04.47 | maqr | TJNII: well, i learned a lot :p |
| 17:07.53 | maqr | one more thing i don't quite understand though: i'm trying to set up callcentric for outbound calls, and they tell me to use " exten => _XX,1,Dial(SIP/${EXTEN}@callcentric)", which i do, but the pattern doesn't seem to match when i dial "17771234567" (their echo teste number)... any idea what that rule should actually look like? |
| 17:08.48 | maqr | asterisk just tells me 'extension not found', which i guess means that the exten isn't matching what i'm dialing |
| 17:09.01 | maqr | _XX doesn't make a lot of sense to me for that, but what should it be? |
| 17:09.05 | outtolunc | only 2 digit dialing eh <G> |
| 17:09.23 | maqr | yeah, thats what i thought, but i figured there must be a reason that's why they said to use it |
| 17:09.38 | outtolunc | change it to _X. and test |
| 17:09.52 | outtolunc | try 1+10digits and just 10 digits |
| 17:10.07 | outtolunc | i've never used callcentric.. so no idea what they expect |
| 17:10.20 | hardwire | wot's callcentric? |
| 17:10.32 | outtolunc | wot's google |
| 17:11.10 | maqr | hardwire: an itsp |
| 17:11.20 | [TK]D-Fender | maqr: Go read the book, you clearly don't understand dialplan patterns. This is the most important part of *. |
| 17:11.28 | hardwire | interactive tickle selection processor? |
| 17:15.32 | hsv-al | callcentric, heh |
| 17:15.41 | hsv-al | isnt that the service that is advertised with EVB |
| 17:19.31 | *** join/#asterisk xenonex (n=xenonex@89.218.235.197) |
| 17:19.37 | maqr | i guess they typo'd in their help file |
| 17:19.59 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:29.53 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:30.05 | hardwire | press 1 for an armpit tickle |
| 17:30.17 | [TK]D-Fender | maqr: Doesn't matter if they "typod", the fact you don't know any better is alarming. |
| 17:30.50 | hsv-al | hardwire |
| 17:31.59 | maqr | [TK]D-Fender: well, i was expecting to see something else, because _XX looked like "two digits" to me, but i figured it was more likely i was wrong than them |
| 17:33.58 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 17:35.28 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 17:35.29 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 17:37.13 | maqr | alright, now i hear ulaw messages from my asterisk server, and i can complete incoming and outgoing calls with my itsp, but i can't hear anything for the itsp calls... the calls are definitely completed, and i know the codec is supported, but i don't get any audio from my sip phone... any thoughts on that? |
| 17:37.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:39.14 | [TK]D-Fender | maqr: make sure their entry in sip.conf is "nat=no" |
| 17:39.19 | *** join/#asterisk mags2 (n=egray@ampulex.whoi.edu) |
| 17:39.59 | maqr | [TK]D-Fender: i just set that, but it's still no audio |
| 17:40.21 | maqr | [TK]D-Fender: they're type=peer and insecure=port,invite |
| 17:40.27 | maqr | if that makes any difference |
| 17:40.41 | maqr | i think those are the right settings though, or the call wouldn't go "up" |
| 17:41.04 | [TK]D-Fender | maqr: test each end independently for audio in/out. |
| 17:41.21 | mags2 | dumb question (which I'm about to verify as soon as I get asterisk 1.0.7 built.... ) |
| 17:41.21 | mags2 | is IAX between asterisk 1.0 and 1.4 going to be compatible and if so are there any big issues? |
| 17:42.21 | maqr | [TK]D-Fender: well, i know my end is okay, because when i dial my test extension, i hear the Playback |
| 17:42.34 | maqr | [TK]D-Fender: so i'd think it's getting lost somewhere between me and callcentric |
| 17:42.43 | [TK]D-Fender | maqr: test audio BOTH ways, from BOTH ends, separate from each other. |
| 17:43.13 | maqr | [TK]D-Fender: how can i test from their end? |
| 17:43.34 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:43.36 | Assid | heya |
| 17:43.40 | Assid | anyone here using sipbroker |
| 17:43.48 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 17:43.49 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 17:44.18 | maqr | [TK]D-Fender: sorry if it's a stupid question, i'm just not sure how i'd go about doing that |
| 17:44.26 | [TK]D-Fender | maqr: Callcentri > *. forget about including your phone in that test. |
| 17:44.49 | maqr | [TK]D-Fender: well, i can tell callcentric to call asterisk, and it does... but i don't see how i can test audio without a phone |
| 17:45.00 | [TK]D-Fender | maqr: .... VOICEMAIL <- |
| 17:45.26 | maqr | oh |
| 17:45.36 | kannan | hello, i did a make progdocs , where are the docs located? |
| 17:46.06 | *** join/#asterisk DaneM (n=dane@adsl-76-236-27-148.dsl.chi2ca.sbcglobal.net) |
| 17:46.44 | russellb | doc/api |
| 17:47.59 | DaneM | hello. I recently installed asterisk-gui, and I've noticed that when I click on as user in the Users configuration section, it highlights ALL the users, not just the one I clicked on, and it won't let me change any of the user's settings. I've noticed this behavior in both *NOW and regular asterisk (on Ubuntu). Any help would be appreciated. |
| 17:48.22 | kannan | thanks, its there alright. a huge reading .. to be done |
| 17:48.50 | *** join/#asterisk timothy_jones (n=chatzill@207.90.4.189) |
| 17:50.08 | yang | !grandstreamm |
| 17:50.11 | yang | !grandstream |
| 17:50.13 | Qwell | ~ |
| 17:50.30 | Qwell | yang: ~ is the bot modifier |
| 17:50.35 | yang | ~grandstream |
| 17:50.36 | jbot | [grandstream] the Yugo of VoIP hardware. Run. Run away now. |
| 17:50.58 | sp00kz | ~qwell |
| 17:50.58 | jbot | qwell is probably a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 17:51.19 | sp00kz | oh really |
| 17:51.24 | Qwell | it is - look it up |
| 17:51.27 | Strom_C | ~strom |
| 17:51.27 | jbot | from memory, strom is Southern California's only residential ISDN BRI customer |
| 17:51.36 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 17:52.03 | tzafrir | what's this BRI thing? |
| 17:52.22 | Qwell | Strom_C: are you really the only one? |
| 17:52.24 | Qwell | and are you still? |
| 17:53.36 | Strom_C | Qwell: i don't know; it was a joke more than anything else. and no, I don't have the circuit anymore |
| 17:54.05 | Assid | anyone here using sipbroker? |
| 17:54.23 | Assid | and able to call voicepulse numbers? |
| 17:55.48 | km2 | is there a way to clear the * CLI aside from holding down enter? |
| 17:55.50 | znoG_ | hey, anyone using zoiper? |
| 17:55.58 | bkruse | km2: !clear |
| 17:55.58 | bkruse | ? |
| 17:56.04 | bkruse | or ctrl+l |
| 17:56.05 | hsv-al | 2600.com |
| 17:56.06 | hsv-al | heh |
| 17:56.12 | km2 | bkruse, ah-ha! great, thank you |
| 17:56.17 | bkruse | km2: np |
| 17:56.26 | bkruse | hsv-al: <3 2600 |
| 17:56.28 | hsv-al | plik, I wonder if they still sell those |
| 17:56.29 | hsv-al | in B&N |
| 17:56.34 | hsv-al | and books million |
| 17:56.36 | DaneM | nvm about my gui problem...looks like it's an issue with firefox 3. |
| 17:56.58 | znoG_ | i have this weird issue with zoiper that when I call outbound (ie. through Zaptel or my voip provider) .. they can't hear me, so I place them on hold then take them off hold and it works fine. Calling another extension on the Asterisk server doesn't require this. Anyone had this? |
| 17:57.45 | *** part/#asterisk DaneM (n=dane@adsl-76-236-27-148.dsl.chi2ca.sbcglobal.net) |
| 17:57.49 | lesouvage | znogG: do you use sip or IAX |
| 17:57.57 | znoG_ | IAX |
| 17:58.00 | plik | hsv-al: should still be around ;) |
| 17:58.24 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 17:58.39 | hsv-al | i last remember reading those like 5 years ago |
| 17:59.18 | plik | me too... longer probably ;) |
| 17:59.34 | hsv-al | i learned about a glitch in cablevisions PPV ordering system from them.........back in the day, if you called up a phone # to order PPV on cablevision, but did it from a phone that wasnt registered as having a cablevision account |
| 17:59.41 | plik | you can subscribe online |
| 17:59.43 | hsv-al | you could specify the phone # you want the PPV billed too |
| 17:59.58 | hsv-al | so i used to call a # to order spice/penthouse porn, and billed like 400 of them to my neighbor, from a payphone |
| 18:00.03 | hsv-al | specifying their phone # HEH |
| 18:00.20 | plik | pranktastic |
| 18:00.25 | km2 | haha, spice |
| 18:01.17 | hsv-al | 2600 still gives weird tips on how to do off the wall things like that? |
| 18:03.51 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 18:05.59 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 18:11.15 | *** join/#asterisk bartpbx (n=bartpbx@e181192019.adsl.alicedsl.de) |
| 18:11.19 | maqr | [TK]D-Fender: good tip on the voicemail, it works |
| 18:11.19 | *** part/#asterisk bartpbx (n=bartpbx@e181192019.adsl.alicedsl.de) |
| 18:12.20 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 18:12.30 | maqr | [TK]D-Fender: but when i me answer the call on my phone, i don't hear the test service (it's TELLME, btw) |
| 18:12.41 | maqr | [TK]D-Fender: if i let it go to voicemail, it leaves me a voicemail and i can hear it the way i should hear it if i answered |
| 18:13.51 | [TK]D-Fender | maqr: test your phone direct to * the same way. if that's OK, then it is likely a reinvite issue |
| 18:16.29 | *** join/#asterisk harryv (n=harry@67-207-147-205.slicehost.net) |
| 18:16.53 | harryv | hey, i have some 200 numbers configured on a zapata, and when i dial out i want to specify which callerid to show.. |
| 18:17.12 | harryv | right now it sends the base-number of the number series.. ie, 99400300 and not 99400389 |
| 18:17.14 | kannan | harryv : me too |
| 18:17.18 | [TK]D-Fender | harryv: you don't configure "numbers" in zapata. |
| 18:17.39 | kannan | i used setcallerid in extensions.conf |
| 18:17.40 | [TK]D-Fender | harryv: And you can set the CID before you dial if your telco lets you set it at all. |
| 18:17.59 | kannan | but i got some 100 entries before i come to priority 2 , which is dial |
| 18:18.14 | kannan | there mustbe away to avoid so many lines |
| 18:18.45 | [TK]D-Fender | kannan: That didn't make much sense. perhaps you should pastebin it |
| 18:19.11 | *** join/#asterisk vitasoy (n=vitasoy@adsl-69-109-124-91.dsl.pltn13.pacbell.net) |
| 18:19.18 | kannan | [TK]D-Fender , will do |
| 18:19.32 | maqr | [TK]D-Fender: other than my voice sounding pretty bad through my pc mic... it definitely works in both directions for me leaving myself voicemail and listening to those voicemails |
| 18:19.38 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 18:19.55 | anonymouz666 | is there any way to bypass the menuselect and force chan_zap compilation |
| 18:19.56 | anonymouz666 | ? |
| 18:20.38 | vitasoy | my experience with Cisco 7940 and 7960 have been not so great, but workable (a matter of finding the right firmware to load, and testing my patience waiting for it to load) -- they also seem to have a somewhat bad reputation. Are the 7960G or 7940G any better? Are the firmware's more stable? |
| 18:20.59 | Strom_C | vitasoy: the G is almost identical to the non-G model |
| 18:21.08 | [TK]D-Fender | vitasoy: Polycom <------- |
| 18:21.08 | Strom_C | i think the only difference is in the key labeling |
| 18:21.28 | maqr | [TK]D-Fender: and i know that when callcentric calls me, they can leave me clear sounding voicemail, so that works |
| 18:21.30 | Qwell | Strom_C: yep |
| 18:21.44 | maqr | [TK]D-Fender: but when i call callcentric, i can't hear them on my sip phone |
| 18:21.51 | Qwell | the G version has symbols instead of text on the buttons like settings |
| 18:21.54 | maqr | does that indicate anything in particular to you? |
| 18:22.10 | vitasoy | ah.. if had a Cisco Call Manager i guess my experience would be better using SCCP? |
| 18:22.16 | sp00kz | I have a 7960 on my desk, works like a champ. |
| 18:22.41 | vitasoy | i mean, the cisco's are cool - feel solid, and people like them. but from and admin-POV they suck ass |
| 18:23.08 | vitasoy | well, as i see it, you fight with them, then they are fine |
| 18:23.37 | Qwell | vitasoy: are fine - for a while. :) |
| 18:23.49 | Qwell | until they lose their config, and then sit in a reboot loop |
| 18:23.56 | sp00kz | haha |
| 18:24.05 | sp00kz | they are a pain, and they don't support everything |
| 18:24.12 | sp00kz | but once you get them up and running |
| 18:24.18 | sp00kz | they're pretty much rock solid |
| 18:24.25 | [TK]D-Fender | maqr: And callcentric >* has audio both ways? |
| 18:24.51 | sp00kz | I personally do prefer polycom 650's though ;o |
| 18:24.54 | harryv | exten => _X.,1,Set(CALLERID(num)=99400700) |
| 18:24.57 | harryv | [May 21 20:23:16] ERROR[12668]: pbx.c:1553 ast_func_write: Function CALLERID not registered |
| 18:25.02 | kannan | [TK]D-Fender , http://pastebin.ca/1025100 |
| 18:25.18 | maqr | [TK]D-Fender: i can't test *->callcentric, but it shouldn't make much difference anyway, since i don't need to send any audio to them in order to hear the greeting i should be hearing |
| 18:25.41 | kannan | [TK]D-Fender , i am doing this for 5 contexts for 100 DID numbers |
| 18:25.49 | [TK]D-Fender | kannan: Ah, so you selectively set your appearance based on the phone placing the call? |
| 18:25.56 | kannan | lol, i am sure you will advise the correct wau now |
| 18:26.11 | kannan | [TK]D-Fender , that is right |
| 18:26.56 | [TK]D-Fender | kannan: I would eith DB that somewhere, or use SetVar in their sip.conf entries to list what they should appear as. Sure its still 100 "lines", but better than putting it in the dialplan. |
| 18:26.58 | sp00kz | harryv: what i do is: exten => s,n,Set(CALLERID(all)=NAME<NUMBER>) |
| 18:27.36 | kannan | can we write a agi to grab the phone number (SIP user) , and store to a variable, will the next priority inherit that variable? |
| 18:28.09 | kannan | oh ok, SetVar in sip.conf |
| 18:28.16 | Strom_C | kannan: you don't need an AGI for that |
| 18:28.57 | harryv | sp00kz: it's probally a module i forgot to load. |
| 18:29.07 | kannan | Strom_C , is there a system variable for the sip user, whicjh i can keep as the phone number also |
| 18:29.10 | harryv | ccould that be it? |
| 18:29.36 | Strom_C | ${CHANNEL} perhaps? |
| 18:30.16 | kannan | Strom_C , is it not a long string? we have to trucate it the 4 digits after SIP/? |
| 18:30.45 | [TK]D-Fender | kannan: Yes, SetVar in sip.conf :) |
| 18:30.54 | Strom_C | yes...there are dialplan functions for that |
| 18:31.06 | [TK]D-Fender | kannan: means you can do a single "if not blank, use it, otherwise use X for default) in your outgoing extens |
| 18:31.20 | kannan | if we set callerid in sip.conf, can we call it as $(CALLERID) |
| 18:31.23 | shido6 | test |
| 18:31.40 | kannan | [TK]D-Fender , i am yet to read up on this |
| 18:31.49 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 18:31.56 | [TK]D-Fender | kannan: I suggest the SetVar approach as it will indicate to your outpbound extens what to use without interfering with your "INTERNAL" appearance |
| 18:32.15 | vitasoy | i don't have windows handy at the moment - anyone have complaints about snapanumber? where you can dial asterisk via thunderbird or firefox? |
| 18:32.16 | maqr | [TK]D-Fender: you said something about reinvites? maybe that's my issue? |
| 18:32.29 | kannan | [TK]D-Fender , will try it out |
| 18:32.32 | kannan | thanks a lot |
| 18:32.35 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 18:32.51 | [TK]D-Fender | vitasoy: I'm sure it works jsut fine. |
| 18:37.12 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:37.36 | kannan | [TK]D-Fender : if i do setvar=CUSTID=100 , for a sip user 100, the i can call ${CUSTID} in extensions.conf? is that correct? |
| 18:37.49 | [TK]D-Fender | kannan: Yup. |
| 18:38.22 | [TK]D-Fender | kannan: as I mentioned, you'd see if its set first so that you can do a "fall-back" number". |
| 18:39.05 | kannan | kewl, so for every sip user , we can give the variable name as CUSTID, and value as unique one (such as the sip username) |
| 18:39.46 | kannan | how to check is set or not in the extensions.conf? |
| 18:40.45 | *** part/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 18:42.25 | *** join/#asterisk xenonex (n=xenonex@89.218.235.197) |
| 18:43.28 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 18:43.40 | pjz | anyone have thoughts on polycom 301 vs 330 ? |
| 18:43.48 | Qwell | pjz: 301 is a dead end |
| 18:44.02 | Qwell | EOL, etc |
| 18:44.22 | kannan | can we use LEN with NoOP first to see if it is null? |
| 18:44.23 | Qwell | besides, I think the 301 is still more expensive than the 330 |
| 18:44.39 | pjz | does the 330 have a headset jack? |
| 18:44.44 | Qwell | [TK]D-Fender: ^? |
| 18:45.45 | kannan | oh ok, i think we have an ISNULL function itself to do that |
| 18:54.01 | [TK]D-Fender | pjz: IP320/330 have a 2.5mm headset jack |
| 18:54.03 | kannan | [TK]D-Fender , will it not be the best way to use the EXECIF <CUSTID> condition ? in the outgoing extensions.conf ? |
| 18:54.13 | [TK]D-Fender | IP 301 = dead end |
| 18:54.19 | [hC] | the 301 only had a half duplex speakerphone and a character based display |
| 18:54.23 | [hC] | its basically a tease. |
| 18:54.42 | [hC] | the 330 is great, or if you're looking for the other style, the 430 is also good. |
| 18:55.01 | [TK]D-Fender | kannan: many ways. a single over-complex line is short on code and low on readability. I'm sure you see the options. Pick your favourite |
| 18:56.19 | kannan | [TK]D-Fender , sure ok, thanks for the help. makes the extensions.conf easier by far to read andedit |
| 18:57.23 | [TK]D-Fender | IP 430 = budget phone for RJ9 headset environments, nothing else. |
| 18:59.25 | pjz | [TK]D-Fender: Thanks! you're helpful as always! |
| 19:01.14 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:01.22 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 19:02.32 | [TK]D-Fender | pjz: I typcially advise the IP 320. If you think you need the passthrough, then consider investing the cost savings between the IP 320 & IP 330 in doing extra cable drops. You'll diversify your infrastructure and reduce failure points. |
| 19:03.30 | pjz | [TK]D-Fender: existing (old, PITA) infrastructure that makes cable drops a real pain are why I was gonna go 330 |
| 19:03.57 | pjz | [TK]D-Fender: good point, though |
| 19:04.07 | [TK]D-Fender | pjz: Well you can see the math. factor that into your scenario and do what you will with it. |
| 19:07.26 | pjz | [TK]D-Fender: will do, thanks again! |
| 19:08.27 | *** part/#asterisk pjz (n=pj@zachs.place.org) |
| 19:11.39 | *** join/#asterisk Aughey (n=jha@64.219.54.125) |
| 19:12.20 | Aughey | In Voice Mail, under advanced options there is a reply option. Where does Asterisk get the information to reply? What variable do I need to set to have the reply feature work? |
| 19:15.18 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 19:16.44 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 19:19.18 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:19.35 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:20.43 | nny_1 | hmm doing a bare install of zaptel and when i execute the init script it says "ztcfg not executable".. I can change the path to ztcfg to point to where it shuold be, etc, the question is "Is there a way during the install process to have zaptel automagically install ztcfg zttool, etc?" |
| 19:21.11 | nny_1 | right now those programs are only in the zaptel source directory |
| 19:21.49 | Qwell | make install installs ztcfg |
| 19:22.01 | Qwell | and zttool, if newt is installed |
| 19:22.05 | nny_1 | hmm didn't seem to the first time let me try again maybe pebkac |
| 19:22.50 | nny_1 | hmmph pebkac ftl |
| 19:22.57 | nny_1 | works now.. back to eating crayons ty |
| 19:23.24 | Strom_C | heh |
| 19:23.39 | Strom_C | nny_1 wins the "coolest nub ever" award |
| 19:23.53 | *** part/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 19:24.09 | nny_1 | yay that can go next to my special olympics medal for drooling |
| 19:24.33 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 19:25.31 | jaytee | yay!!!! I got * working with Exchange UM in both directions using sipX as a sip upd to tcp router |
| 19:27.28 | Damin | Hmmm.. |
| 19:31.12 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 19:32.07 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-230-58.unitz.ca) |
| 19:34.27 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 19:34.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:36.46 | eric2 | snom 300 phone, I call someone, and they can not hear me... happens at random, am behind a router, I know its a networking thing... but don't know what else to check |
| 19:37.00 | [TK]D-Fender | ~sipnat |
| 19:37.01 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:37.03 | [TK]D-Fender | ^^^ |
| 19:38.44 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 19:40.41 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 19:41.35 | eric2 | no, the ip phone is having issues, asterisk has a public ip |
| 19:41.47 | eric2 | linksys phones are ok, snom is giving me grief! |
| 19:43.35 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 19:44.38 | eric2 | maybe its the crappy linksys wireless router |
| 19:44.47 | eric2 | looks for a hammer |
| 19:46.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:47.58 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:51.14 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:51.33 | iamhrh | I'm tring to get Directory() to work in my dialplan, and whenever I call the extension it is set on, I get the output on the cli you see here: http://pastebin.ca/1025185 |
| 19:52.03 | iamhrh | am I missing a column from my database definition? |
| 19:55.06 | iamhrh | i tried looking for the sql statement that it is trying to run in app_directory.c, app_voicemail.c, and res_odbc.c ... no luck there. are those stored elsewhere or did I just miss something obvious? |
| 19:59.00 | [TK]D-Fender | iamhrh: "Unknown column 'mailbox'" <- doesn't this pretty much spell it out? |
| 19:59.39 | Corydon76-dig | res_odbc doesn't originate any statements, anyway |
| 20:00.15 | iamhrh | <PROTECTED> |
| 20:00.17 | *** join/#asterisk bmg505 (n=leon@196-209-78-202-tbnb-esr-2.dynamic.isadsl.co.za) |
| 20:00.57 | iamhrh | <PROTECTED> |
| 20:01.22 | Corydon76-dig | iamhrh: that's only half of it |
| 20:01.45 | Corydon76-dig | That's only for the storage of actual messages, not the index of what mailboxes are configured |
| 20:01.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:01.58 | iamhrh | Corydon76-dig: noticed that much, but when I tried digging into the other files i still couldn't find anything...sorry but i'm pretty new to poking around in the actual code |
| 20:02.15 | iamhrh | so am I missing a table entirely then? |
| 20:02.49 | Corydon76-dig | See contrib/scripts/vmdb.sql |
| 20:03.17 | Corydon76-dig | If you're in 1.4, it's even well-documented as to what each field does |
| 20:03.20 | iamhrh | thanks, looking there now |
| 20:03.44 | iamhrh | didn't even know that existed :-/ |
| 20:04.43 | Corydon76-dig | It's amazing what you find when you look around. |
| 20:05.18 | iamhrh | its also amazing what you find when you ask after looking around for hours on your own |
| 20:07.02 | lesouvage | Is it possible with Iksemel (gtalk) to send a message to the Asterisk server using gtalk and trigger something in the dialplan (like setting up a phonecall). The idea is to send two phonenumbers to the asterisk box with gtalk and have a phonecall setup between this numbers |
| 20:07.29 | Strom_C | lesouvage: no, that's not a gaping security hole at all |
| 20:09.52 | lesouvage | Strom_C: for now it's just a proof of concept thing without attention for security. It's just about what happens with a gtalk message send to an Asterisk server. Would it be possible to do something with the message in the dialplan? |
| 20:10.21 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:11.11 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 20:12.54 | spokra | just upgraded from 1.4.x to 1.6.beta9 and my sip stopped working.. getting password errors from my sip phone registering and can't make a call into box from pstn. any ideas what changed? something in sip.conf? |
| 20:15.01 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 20:18.15 | *** join/#asterisk bigdaddy2 (n=stv@phoenix.u4eatechinc.com) |
| 20:20.14 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:20.55 | jblack | I'm having an intermittant problem with "chan_zap.c: Call specified, but not found?" showing up in the log file. Google isn't being greatly useful right now; most of the posts go back to '03 or '05 or so. Any suggestions on what I can do to resolve it? Calls that get hit by it die. |
| 20:21.02 | jblack | (I'm using 1.4.17) |
| 20:22.58 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 20:24.17 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 20:24.40 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:25.16 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:25.52 | nny_1 | just installed my first Digium TE122B based system.. any fun way i can check and make sure all the hardware modules are loaded correctly, etc? |
| 20:26.08 | nny_1 | installed --> built |
| 20:26.11 | nny_1 | still testing in house |
| 20:26.27 | jblack | Dial the local pizza joint, and hand the boss the phone |
| 20:26.38 | nny_1 | lol not hooked up to a PRI yet |
| 20:28.06 | outtolunc | use a loop back plug and fire up zttool |
| 20:32.51 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 20:32.59 | hsv-al | 9th cup of coffee in 3 hours. . tight |
| 20:33.17 | hsv-al | now I need to take some ephedra pills |
| 20:35.07 | hardwire | hsv-al: where are you? whats a good 10 digit number for the local emerg4ency services. ? |
| 20:37.07 | *** join/#asterisk rignes (n=rignes@216-164-160-133.c3-0.atw-ubr3.atw.pa.static.cable.rcn.com) |
| 20:37.20 | hsv-al | I think theres some 256-755's available |
| 20:37.31 | hsv-al | s/n |
| 20:37.42 | Qwell | theren? |
| 20:38.16 | hsv-al | qwell, whats the chance that some microscopic asterisk will ever be developed |
| 20:38.22 | hsv-al | and loaded on a blackberry perhaps |
| 20:38.23 | rignes | Asterisk Noob question: How do I chance the number of rings before Asterisk will forward to voicemail? |
| 20:38.24 | hsv-al | :) |
| 20:38.58 | russellb | rignes: core show application Dial ... set the timeout parameter |
| 20:39.00 | Qwell | hsv-al: patches welcome |
| 20:39.04 | russellb | it's in seconds, not rings |
| 20:39.15 | Qwell | a "full" ring is 6s in the US |
| 20:39.18 | russellb | Qwell: gumstix |
| 20:39.20 | rignes | russellb: Thanks, I assume that is in asterisk.conf? |
| 20:39.27 | Qwell | russellb: it's not patched though. |
| 20:39.28 | russellb | extensions.conf |
| 20:39.33 | Qwell | he wants something where the binary is a few k, no doubt |
| 20:39.43 | rignes | russellb: Thanks a bunch. ;) |
| 20:39.59 | hsv-al | rb, are you on 2 hours of sleep? |
| 20:40.11 | hsv-al | waiting for the tornados to roll through last night, they just missed us, and killed people in NC |
| 20:42.01 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 20:43.07 | hsv-al | hardwire, where do you live |
| 20:43.26 | hardwire | anchorage, ak |
| 20:43.34 | hsv-al | oh, n/m, I thought you were in providence also |
| 20:43.42 | hardwire | why? |
| 20:43.50 | hsv-al | you said 256 in the new area |
| 20:43.57 | file | you should probably be glad hardwire is not in providence... |
| 20:44.04 | hardwire | I did? |
| 20:44.06 | russellb | hsv-al: there were not even storms last night ... |
| 20:44.12 | hardwire | You should? |
| 20:44.32 | hsv-al | hmm |
| 20:44.38 | hsv-al | was supposed to get us |
| 20:44.47 | hsv-al | but it died over the state, but killed some people |
| 20:44.54 | hsv-al | in nc |
| 20:45.30 | hsv-al | file, you another local? |
| 20:46.05 | file | is not |
| 20:46.16 | russellb | file lives in an igloo |
| 20:46.37 | russellb | in a place so cold that it's not suited for human life |
| 20:46.40 | russellb | called .... Canada |
| 20:46.46 | hsv-al | :) |
| 20:47.02 | sp00kz | the frozen tundra |
| 20:47.18 | file | russellb: :D |
| 20:47.31 | russellb | file: <3 |
| 20:47.32 | seanbright | thought it was Canadia |
| 20:48.47 | Juggie | pish |
| 20:48.52 | Juggie | its just eastern canada that sucks |
| 20:48.54 | seanbright | used to be funny... years ago |
| 20:48.57 | Juggie | ontario is fine :) |
| 20:49.10 | hsv-al | juggie, whats the origin, and significance of people saying eh? |
| 20:49.18 | hsv-al | the way people take jabs at southerners for saying yall |
| 20:49.21 | Juggie | i have no idea |
| 20:49.44 | seanbright | its from strange brew |
| 20:49.45 | hardwire | It sucks when people are offed by nature. |
| 20:49.48 | seanbright | that was the origin |
| 20:49.49 | seanbright | heh |
| 20:49.54 | Juggie | dont crap on canada until you visit |
| 20:50.10 | Juggie | you'll find it so much more relaxed then bush land |
| 20:50.16 | hsv-al | http://americansguide.ca/ |
| 20:50.17 | putnopvut | Juggie: and then when I visit, I can literally crap on canada. |
| 20:50.20 | hsv-al | ill just have a good read :) |
| 20:50.22 | seanbright | likes toronto |
| 20:50.23 | Juggie | exactally |
| 20:50.30 | Juggie | but we wont make you take your shoes off at the border |
| 20:50.41 | hsv-al | Most Americans know next to nothing about their neighbo(u)r to the north, except that Canadians play a lot of hockey, drink beer, and end sentences with "eh?" |
| 20:50.42 | maqr | http://pastebin.ca/1025259 <-- could someone help me figure out where i'm going wrong? i can place outgoing calls with my phone, but i don't hear any audio.... if i place an incoming test call, i get a voicemail and it sounds fine, and if i call my voicemail i can hear it fine, my provider does support ulaw and there's no errors.... why can't i hear anything on my sip phone (zoiper)? |
| 20:51.03 | seanbright | hsv-al: that pretty much sums it up for me |
| 20:51.09 | maqr | *why can't i hear anything on outbound calls on my sip phone? |
| 20:51.16 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20.1 (2008/05/21), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 20:51.22 | Juggie | seanbright, its a good place to visit :) |
| 20:51.50 | jer | hsv-al, that's all we want them to know, that way they'll never expect the invasion^Wliberation^Wgift basket we're going to send them |
| 20:51.51 | Juggie | i mean drinking age of 18/19 how can you go wrong :P |
| 20:51.58 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 20:51.58 | jer | =] |
| 20:52.54 | Juggie | hey if we got attacked like every country in the world would come to our aid :) |
| 20:53.00 | Juggie | everyone likes us |
| 20:53.25 | hsv-al | thats the first political discussion ive heard here in a few months heh |
| 20:53.44 | maqr | D-Fender suggested it might have something to do with canreinvite, but i've tried all the settings i can think of, and it doesn't seem to do anything |
| 20:56.33 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 20:57.17 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 20:59.52 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 21:00.41 | nny_1 | is the t1 card light supposed to be green/on? I have no lights, made a loopback cable (1-4,2-5) and rebooted.. zttool says "OK", bit no lights :\ |
| 21:02.04 | nny_1 | well there are red lights on the PCB of the card itself, but not on the panel front |
| 21:03.07 | sp00kz | brand/model? |
| 21:03.17 | nny_1 | hmm zttol details says TxB 1 TxD 1m RxB 1 RxD 0... looks like one of the wires isnt making proper contact? |
| 21:03.24 | nny_1 | <PROTECTED> |
| 21:03.55 | sp00kz | im my experiance with those cards, Red light should be on when the zap drivers are loaded and there is no link |
| 21:04.02 | sp00kz | and when your circuit comes up, it should turn green |
| 21:04.06 | nny_1 | yeah trying loopback testing |
| 21:04.29 | sp00kz | well |
| 21:04.39 | sp00kz | if zap is loaded correctly, and detects the card, it should be red |
| 21:04.45 | sp00kz | the led on the faceplate |
| 21:05.10 | sp00kz | can you start asterisk, and do zap show status? |
| 21:05.45 | Tebi | i remember it should be blinking red ;) |
| 21:05.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:06.06 | nny_1 | hmmm |
| 21:06.17 | nny_1 | zap show status no such command from asterisk CLI |
| 21:06.21 | nny_1 | which is also weird |
| 21:06.34 | sp00kz | indeed, i would bet you don't have zap compiled/loaded correctly |
| 21:06.37 | sp00kz | or maybe its not started |
| 21:06.41 | sp00kz | service asterisk stop |
| 21:06.44 | sp00kz | service zaptel start |
| 21:06.49 | sp00kz | service asterisk start |
| 21:07.00 | sp00kz | in a redhat enviroment |
| 21:07.33 | nny_1 | starting ,seems to hang for a sec after wct4xxp , but eventually loads |
| 21:07.53 | nny_1 | modules are loaded |
| 21:08.04 | sp00kz | start asterisk and do the zap show status |
| 21:08.17 | Tebi | ztcfg -vv ? |
| 21:08.37 | nny_1 | 1 channels to configure |
| 21:08.44 | nny_1 | guess that answers that question |
| 21:08.56 | sp00kz | pbx01*CLI> zap show status |
| 21:08.57 | sp00kz | Description Alarms IRQ bpviol CRC4 |
| 21:08.57 | sp00kz | Digium Wildcard TE110P T1/E1 Card 0 RED 18 0 0 |
| 21:09.23 | sp00kz | except in yours i hope red says No |
| 21:09.23 | sp00kz | :p |
| 21:10.12 | nny_1 | heh blinkin red light, orange light when plugge din loop back |
| 21:10.16 | maqr | anyone have any suggestions for why i would hear my PBX's messages on my softphone, but i get no audio when i dial out? i'm low on ideas |
| 21:10.16 | nny_1 | sorry this is my first t1 card :) |
| 21:10.28 | *** join/#asterisk kimo_sabe (n=nick@zappa.azrackspace.net) |
| 21:10.46 | Tebi | nny_1, sorry but your lookback ins´t working correctly ;) |
| 21:10.48 | sp00kz | no problemo |
| 21:10.51 | Tebi | loop even |
| 21:10.56 | sp00kz | it should go green |
| 21:10.59 | hardwire | lewp |
| 21:10.59 | Tebi | true |
| 21:11.30 | sp00kz | but at least its being detected now |
| 21:11.43 | Tebi | thats goodie :) |
| 21:12.23 | nny_1 | it is green now |
| 21:12.31 | Tebi | whee! :D |
| 21:12.33 | hsv-al | Is AN geared for the more experienced user? |
| 21:12.40 | sp00kz | Try the zap show status now |
| 21:12.41 | hsv-al | or is it simplified for a new user? im still what you consider "new" |
| 21:12.45 | nny_1 | orange when i plugged it in, ran ztcfg -vv, did Changing signalling on channel 24 from Clear channel to Slave channel |
| 21:12.49 | nny_1 | went green |
| 21:14.01 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 21:14.40 | edibrac | can you do "if/then" logic on extensions within extensions.conf? i.e. if extension is 4321, dothis() |
| 21:14.59 | sp00kz | You can indeed |
| 21:15.09 | sp00kz | make sure to check your syntax though |
| 21:15.18 | hardwire | syntax is is syntax does. |
| 21:15.27 | seanbright | edibrac: that kinda stuff is easier to do with AEL |
| 21:15.41 | seanbright | edibrac: and by easier, i mean 'cleaner' |
| 21:15.41 | edibrac | that's in newer/current versions right? i'm on 1.2 |
| 21:15.42 | seanbright | IMHO |
| 21:15.45 | edibrac | ah |
| 21:16.03 | sp00kz | http://www.voip-info.org/wiki/view/Asterisk+AEL |
| 21:16.05 | seanbright | edibrac: i think the original AEL is in 1.2. i might be lying though. |
| 21:16.40 | sp00kz | ~ael |
| 21:16.41 | jbot | hmm... ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
| 21:17.01 | edibrac | i see you can do this ${EXTEN:1} to chop off the first digit - is there some sort of substring() -- i'm not sure what i shoudl google for -- string manipulation in extensions.conf? |
| 21:17.48 | seanbright | edibrac: http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
| 21:18.08 | nny_1 | so i assume if nothing happens during the loopbacktest, it timeouts and ends, that it was successful? |
| 21:18.18 | edibrac | seanbright: awesome thanks |
| 21:18.24 | seanbright | nods |
| 21:22.10 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 21:23.40 | maqr | also, what's the prefered method for encoding a sound file to ulaw to be played as a message? i'd like to record my own phone treet messages |
| 21:23.40 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 21:27.03 | jblack | chan_zap.c:8784 pri_fixup_principle: Call specified, but not found? |
| 21:27.05 | jblack | Still! |
| 21:28.42 | jblack | Hmm. http://readlist.com/lists/lists.digium.com/asterisk-users/11/55173.html seems to say what the problem is, but I don't understand it |
| 21:28.51 | *** part/#asterisk laurier57 (n=drj33@gateway.centerpointe.com) |
| 21:39.16 | *** join/#asterisk Q3Man (n=Steve@wsip-24-234-155-62.lv.lv.cox.net) |
| 21:39.17 | hsv-al | fakhir |
| 21:39.20 | hsv-al | who the hell is that |
| 21:39.32 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 21:39.55 | fakhir | lol i have no idea |
| 21:40.00 | anonymouz666 | I don't remember if Asterisk 1.4 support queue log in RealTime engine? |
| 21:43.52 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:47.27 | *** part/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 21:50.11 | maqr | if i make a sip softphone connection directly to callcentric (my itsp), and force ulaw, i can send and receive voice without any problem.... if i try and do it through * though, i hear nothing, even though it's the same codec and the call is going through |
| 21:50.43 | maqr | http://pastebin.ca/1025259 <--- my configs are here, and i'm out of ideas, i know callcentric is working, and i know from my own Playback and voicemail that asterisk is working |
| 21:50.53 | maqr | somehow the connection Dial()'d connection isn't relaying the audio to my phone though |
| 21:51.57 | edibrac | hmmm in extensions.conf, how do you print() or echo() so that you can debug expressions and see the results in the *CLI (asterisk -rvvvvvvv)? |
| 21:52.20 | edibrac | like i have this, i'm not sure if it's right: GotoIf("${${CALLERIDNUM}:0:1}"="4"]?2:3) |
| 21:53.26 | edibrac | if the outbound caller id starts with a 4, then go to line 2, where I can set the callerId to that number. By convention, our DIDs start with a 4 |
| 21:53.44 | edibrac | then if it doesn't start with a 4, show the main office line instead for the callerid |
| 21:54.38 | maqr | [TK]D-Fender: could i get a kick in the right direction? i'm out of things to try |
| 21:56.18 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
| 21:57.46 | [TK]D-Fender | maqr, "canreinvite=yes" <- always "no". Why are you only ringing for 5 seconds? Next include SIP debug for an incoming call. |
| 21:58.34 | [TK]D-Fender | maqr, And specify your codecs in EACH peer/ |
| 21:59.01 | [TK]D-Fender | edibrac, NoOp <- |
| 21:59.21 | [TK]D-Fender | edibrac, And no, that isn't right |
| 21:59.45 | [TK]D-Fender | edibrac, broken variable reference and using a deprecated variable to boot. |
| 22:00.07 | [TK]D-Fender | edibrac, And lacking a proper expression syntax |
| 22:00.20 | edibrac | ah an extra $ |
| 22:00.34 | hardwire | mmmm $ |
| 22:00.38 | Yourname` | When I hit "reload" on the CLI, suddenly EVERYTHING stops. And I have to restart asterisk. What's wrong? |
| 22:00.53 | Yourname` | Only reload does that. |
| 22:00.56 | hardwire | blah. |
| 22:00.56 | [TK]D-Fender | edibrac, GotoIf($["${CALLERID(num):0:1}"="4"]?2:3) |
| 22:01.29 | [TK]D-Fender | edibrac, $[] <-- expression. ${} <- variable reference or function call. |
| 22:01.32 | maqr | [TK]D-Fender: i changed that to 'no' (doesn't seem to have an impact), i was only ringing for 5 seconds to test voicemail after the first ring (that's not the problem though, because i get no audio calling in or out), and how do i SIP debug for an outgoing call, i feel like outgoing will be easier to debug first? (sip set debug gets me tons of packets outputted, if that's what you mean) |
| 22:01.56 | [TK]D-Fender | maqr, yes, I want to see the whole thing. |
| 22:02.09 | maqr | [TK]D-Fender: can i make that log to a file? it'd be too much to copy/paste |
| 22:02.24 | [TK]D-Fender | maqr, pastebin it, no, it won't be too much. |
| 22:02.33 | muiro | Here's a fun question. I use the M() option in Dial() to run a macro after a dialed out call in answered. I use Playback() in the macro, however the audio is all choppy and doesn't play through the whole way (I've tired wav and gsm, both files play fine when calling -in- to asterisk). Afterwards the call fails to bridge. Hold while I gather a pastebin |
| 22:07.51 | maqr | it's huge, sec |
| 22:08.04 | muiro | Sound played back to other side, tried both gsm and wav, both horribly choppy and do not finish playing all the way through before the channel drops on both sides. http://pastebin.org/37614 |
| 22:10.34 | muiro | it will bridge if I run SendDTMF, Playback and Background cause it not to bridge. Will also bridge correcting if using the D() option or if no options are given. |
| 22:11.46 | muiro | curious as to weather a problem with zaptel (ztdummy) could be responsible for this. Upgraded kernel and have been having a zaptel error on boot (but it hadn't changed asterisk's behavior up until now so I didn't mess with it) |
| 22:12.19 | [TK]D-Fender | muiro, if your kernel changed you have to recompile zaptel. Go do that now. |
| 22:12.37 | muiro | can do |
| 22:14.35 | maqr | [TK]D-Fender: http://pastebin.ca/1025340 |
| 22:14.50 | maqr | [TK]D-Fender: i filtered out a few user names and domains and whatnot, but that's all of it |
| 22:15.27 | maqr | also, i'm surprised how much sip looks like a mail header |
| 22:16.37 | [TK]D-Fender | maqr, its based on HTTP |
| 22:18.34 | maqr | neat |
| 22:20.19 | [TK]D-Fender | maqr, Ok, I'm not sure what I'm seeing there. are you calling OUT callcentric only to come right back in? |
| 22:20.36 | maqr | [TK]D-Fender: umm, nope, i'm calling out to callcentric and expecting to hear a tellme service |
| 22:20.39 | maqr | [TK]D-Fender: instead i hear nothing |
| 22:21.01 | maqr | when i tell callcentric to call in, it's also supposed to connect me to the tellme service, but i also hear nothing |
| 22:21.02 | [TK]D-Fender | maqr, ok, not sure what to advise. |
| 22:21.13 | maqr | hrm |
| 22:23.31 | muiro | [TK]D-Fender: zaptel recompiled. Any hints with the dial macro playback thing? Would my sip.conf be helpful? |
| 22:23.47 | muiro | (recompiled and started) |
| 22:24.33 | maqr | [TK]D-Fender: the "packet bridging" is the point where my phone is supposed to connect to the Dial()'d service, right? |
| 22:24.42 | maqr | err, "packet2packet bridging" |
| 22:25.15 | maqr | [TK]D-Fender: does that mean that my phone is actually directly connecting to the sip service instead of proxying through *? maybe that's the issue? |
| 22:25.41 | [TK]D-Fender | maqr, don't believe so. |
| 22:25.55 | [TK]D-Fender | muiro, not sure... |
| 22:26.45 | maqr | maybe i need to disable packet2packet? |
| 22:27.16 | maqr | google says: "Packet2Packet Bridging = Audio is not going through the Asterisk core, |
| 22:27.19 | maqr | it comes into the RTP stack and goes directly out. This decreases the |
| 22:27.21 | maqr | oops, thought that'd be one line |
| 22:27.23 | maqr | amount of memory allocation that happens, and things require less |
| 22:27.26 | maqr | processing. |
| 22:27.28 | *** join/#asterisk techie (n=techie@adsl-76-214-2-200.dsl.lsan03.sbcglobal.net) |
| 22:27.28 | maqr | " |
| 22:29.16 | hardwire | maqr: redirects are teh awesomer |
| 22:29.38 | maqr | hardwire: could that maybe be why i don't hear audio on my itsp calls? |
| 22:29.58 | hardwire | maqr: I haven't been paying attention. |
| 22:30.00 | maqr | i know it's not a codec issue, and i know calling directly through the itsp with my phone works, but when i try to Dial() out through asterisk, i get no audio |
| 22:30.41 | hardwire | when you Dial() from asterisk to your softphone does that work? |
| 22:30.49 | maqr | hardwire: http://pastebin.ca/1025259 <-- configs .... and http://pastebin.ca/1025340 <--- sip debug |
| 22:31.01 | hardwire | oh, I'm only surface level right now, thanks tho. |
| 22:31.15 | maqr | hardwire: you mean when a call comes from the itsp and i Dial it back to my extension? no, i don't get audio there either |
| 22:31.20 | maqr | it does ring and i can answer though |
| 22:31.39 | hardwire | nevermind |
| 22:31.44 | maqr | :\ |
| 22:32.07 | hardwire | I misread something |
| 22:32.09 | hardwire | no biggie |
| 22:32.39 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20.1, 1.2.28.1 (2008/05/21), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 22:37.00 | *** join/#asterisk Weasel[DK]_ (n=Weasel[D@0x503fbea6.kd4nxx16.adsl-dhcp.tele.dk) |
| 22:38.17 | maqr | if it authenticates, the codec is right, and the call is completed... what could cause no audio to come through? |
| 22:38.34 | maqr | is there some other config file or maybe module or something that controls whatever part of asterisk does the forwarding? |
| 22:38.48 | maqr | err, not 'forwarding', i guess its 'relaying'? |
| 22:39.23 | *** join/#asterisk grovest (i=grovest@ginoc.telcom.oregonstate.edu) |
| 22:41.17 | *** part/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 22:42.28 | maqr | 'bridging' |
| 22:42.44 | *** part/#asterisk timothy_jones (n=chatzill@207.90.4.189) |
| 22:42.47 | maqr | anyone know how to turn off packet2packet and use something else for bridging? i think that might be where my problem is |
| 22:43.01 | Weasel[DK]_ | i have some trouble playing some sound .wav files... the error looks like -> http://pastebin.com/d42ad1f4c |
| 22:43.27 | lesouvage | I just did a svn checkout of Asterisk and there is no res/res_features.c available in the res/ dir. What happens to res_feautures.c? I used the instruction on http://www.asterisk.org/developers/get-source |
| 22:46.36 | maqr | [TK]D-Fender: i tried calling out again, and i heard a little 'blip' sound, if that maybe indicates something? i can't make it blip again though, it only happened once |
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| 23:10.58 | maqr | should my use count for all modules in 'modulue show' be 0? |
| 23:11.03 | maqr | how can i tell which ones are actually being used? |
| 23:11.04 | *** join/#asterisk ionix (n=ionix@124x38x141x182.ap124.ftth.ucom.ne.jp) |
| 23:12.40 | ionix | Hey, I have an asterisk machine behind NAT. I know it sucks but can't help it... I have setup externip and localnet in sip.conf, I can make calls and get audio both ways but I can't receive calls. I have configured my sip.conf correctly I believe (insecure=very, etc) but I don't even see calls coming in on the asterisk console. I tried NMAP from the internet to the machine and it shows port 5060/udp as open. any ideas? |
| 23:13.50 | daven | ionix: can your client use IAX? |
| 23:14.08 | [TK]D-Fender | ionix, read up : |
| 23:14.11 | [TK]D-Fender | ~sipnat |
| 23:14.12 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:14.13 | [TK]D-Fender | ^^^^ |
| 23:14.16 | [TK]D-Fender | ionix, its all in there. |
| 23:17.09 | ionix | hmm no... |
| 23:17.21 | ionix | SIP only I am afraid, the gateway is SIP only |
| 23:17.25 | ionix | stupid Japan |
| 23:18.15 | ionix | Basically clients and asterisk server are behind NAT |
| 23:18.29 | ionix | Gateway is of course on the public internet |
| 23:18.38 | [TK]D-Fender | ionix, go read ---^ |
| 23:18.55 | ionix | I read... but I know I can make it work without SER |
| 23:19.13 | [TK]D-Fender | ionix, first link make NO reference to SER. |
| 23:19.24 | ionix | #1 works with a NAT-supporting proxy as SIP Express router as the outside proxy. (Get an account at IPtel.org and try!). Fails with Free World Dialup. |
| 23:20.25 | ionix | but since asterisk is behind a NAT but DMZ, my situation is different, isn't it? |
| 23:20.39 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:20.42 | ionix | I was hoping that someone could help me track the SIP stack |
| 23:21.11 | [TK]D-Fender | ionix, http://www.aocomputing.net/?p=3 <--- 1st LINK |
| 23:21.34 | ionix | oh ok |
| 23:21.56 | maqr | does digium offer some kind of free sip connection to test asterisk with? |
| 23:22.18 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-40-133.lns10.syd7.internode.on.net) |
| 23:22.59 | ionix | maqr: Sip no, IAX yes |
| 23:23.02 | ionix | SIP, you can try FWD |
| 23:24.02 | rob0 | last time I messed with FWD (long time ago) their SIP was broken, but IAX worked fine. |
| 23:24.10 | maqr | yeah, their site only has iax directions |
| 23:24.26 | lesouvage | I have lots of errors from eagi-test.c while do a make on a svn checkout. Is this because of the code or am I missing something on my system. (see: http://www.pastebin.be/11472 ) |
| 23:26.56 | ionix | thx [TK]D-Fender, however it did not help... Any other ideas? |
| 23:27.22 | [TK]D-Fender | ionix, pastebin your sip.conf masking only passwords |
| 23:27.24 | [TK]D-Fender | ~pb |
| 23:27.25 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:27.26 | [TK]D-Fender | ^^^^^ |
| 23:28.50 | craigk | greetings all. I am looking at adding a BRI ISDN line to my asterisk system. From what I understand, I need to choose between chan_misdn and chan_capi ... is there any reason I would pikc on over the other ? |
| 23:29.07 | craigk | s/pikc on/pick one/ |
| 23:29.20 | ionix | http://pastebin.ca/1025395 |
| 23:29.25 | craigk | wow - i did not expect jbot to do that - nice :) |
| 23:30.00 | maqr | what exactly is RPT? |
| 23:30.14 | [TK]D-Fender | ionix, ... |
| 23:30.28 | [TK]D-Fender | ~freepbx |
| 23:30.28 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:30.51 | ionix | ok right |
| 23:30.51 | maqr | err, RTP, not RPT, that's apparently something else |
| 23:30.51 | ionix | but |
| 23:30.51 | ionix | the issue is simple here |
| 23:30.56 | ionix | it doesn't reach the dialplan |
| 23:30.56 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:31.05 | [TK]D-Fender | ionix, and you need to forward 5060,10000-20000 and have all the other setting listed in that guide. |
| 23:31.28 | ionix | [TK]D-Fender: As mentioned earlier, the machine is on DMZ and ports are open |
| 23:31.55 | ionix | At least I know that 5060/udp is listening. So I was hoping someone with an asterisk machine can help me trace the stack |
| 23:31.56 | SteveTotaro | hence the name "DMZ" |
| 23:32.10 | ionix | correct SteveTotaro |
| 23:32.25 | SteveTotaro | what exactly do you need? |
| 23:32.32 | SteveTotaro | maybe i can help |
| 23:32.47 | kimo_sabe | is there a channel for IAXModem? |
| 23:32.50 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 23:33.07 | [TK]D-Fender | kimo_sabe, highly doubt it... |
| 23:33.12 | ionix | Well, the asterisk server is behind NAT on the DMZ... I can make calls and head audio both ways... The only problem is that I can't receive calls |
| 23:33.22 | kimo_sabe | [TK]D-Fender: damn |
| 23:33.23 | maqr | SUCCESS! |
| 23:33.30 | SteveTotaro | so what do you need? |
| 23:33.37 | ionix | I set verbose to level 9 but I don't see any incoming calls |
| 23:33.42 | maqr | [TK]D-Fender: note: tell newbies to drop their firewall when experiencing odd problems with audio loss |
| 23:33.42 | ionix | I need incoming calls to work :) |
| 23:33.54 | maqr | [TK]D-Fender: i forwarded the wrong ports for RTP, i thought it was TCP, it's UDP |
| 23:34.16 | kimo_sabe | for the record, IAXModem doesn't quiet get along with the Microsoft Fax service |
| 23:34.31 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 23:34.34 | maqr | hardwire: problem solved, i'm bad at managing a firewall |
| 23:35.02 | hardwire | another trouble ticket closed, awesome work maqr |
| 23:35.05 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
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| 23:35.10 | [TK]D-Fender | maqr, SMRT |
| 23:35.17 | maqr | I AM SO SMART, S-M-R-T! |
| 23:35.28 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
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| 23:35.30 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 23:35.30 | maqr | does the RTP dance |
| 23:35.30 | hardwire | wb. |
| 23:35.51 | SteveTotaro | sniffing RTP |
| 23:36.04 | hardwire | new drug? |
| 23:36.20 | lanning | "Some good shit, man." |
| 23:36.28 | SteveTotaro | nah, just listening in on people |
| 23:36.43 | SteveTotaro | cable internet is great |
| 23:37.00 | maqr | SteveTotaro: lol? |
| 23:37.08 | maqr | what cable network isn't switched these days? |
| 23:37.24 | ionix | maqr: Hmm, all of then? |
| 23:37.27 | ionix | them |
| 23:37.30 | maqr | that can't be |
| 23:37.48 | ionix | you think they have a dedicated loop to your house? |
| 23:38.05 | SteveTotaro | why do you think bandwidth drops during peak hours? |
| 23:38.21 | maqr | ionix: well, you can't very well put a cable modem into promiscuous mode, can you? |
| 23:38.24 | SteveTotaro | why do you think i can see what pages my neighbors are visiting? |
| 23:38.40 | ionix | maqr: Actually you can :) |
| 23:38.42 | ionix | Anyway |
| 23:38.48 | SteveTotaro | no but you can put your nic into proiscuous mode |
| 23:38.51 | ionix | what about an asterisk server behind NAT on DMZ |
| 23:39.10 | SteveTotaro | turn off iptables |
| 23:39.17 | ionix | ah wtf |
| 23:39.25 | ionix | SteveTotaro. That's wrong |
| 23:39.35 | ionix | I guess there's nobody skilled at these hours to help me. |
| 23:39.45 | SteveTotaro | drop your firewall and test |
| 23:39.46 | ionix | I'll post on google forums, damn living in Japan |
| 23:39.54 | SteveTotaro | if it works then your firewall is wrong |
| 23:39.57 | *** join/#asterisk rizwank (n=rizwank@dsl093-116-242.lax1.dsl.speakeasy.net) |
| 23:40.01 | maqr | ionix: really? |
| 23:40.08 | maqr | ionix: what cable modem allows that? |
| 23:40.19 | maqr | ionix: when i put my nic in promiscuous mode connected to my cable modem, i only see my packets |
| 23:40.30 | ionix | maqr: Modem yes, some tools exists to unlock your modem. Pretty illegal and allows you to uncap your modem etc. |
| 23:40.51 | maqr | ionix: ah |
| 23:40.59 | SteveTotaro | the cable modem downloads it's config at restart |
| 23:41.00 | ionix | maqr: SteveTotaro is either wrong or he has a shit ISP. |
| 23:41.00 | rizwank | Hi there. Quick question - I've got some folks going to use an Asterisk box as a 'hold music' server that plays music live from the net. Is there a guideline to server specs available? I was just going to get a standard dell 2950 with 2gb of ram, but Iwasn't sure if Asterisk was greedy on some resource that might be needed .. |
| 23:41.14 | SteveTotaro | trick it to get it's config from you |
| 23:41.26 | maqr | yeah, i remember seeing people tftp their config down, edit and put it back up |
| 23:41.29 | maqr | years ago though |
| 23:41.43 | maqr | i'm still surprised any cable modem chipset has promiscuous mode as an option though |
| 23:41.57 | [TK]D-Fender | rizwank, How is * a "MoH Server"? You looking to make that a back-end app server for SER or something? |
| 23:42.22 | SteveTotaro | even some cell phones have "promiscousous mode" ;-) |
| 23:42.35 | [TK]D-Fender | SteveTotaro, filthy slut! |
| 23:42.52 | SteveTotaro | lol |
| 23:42.53 | kimo_sabe | [TK]D-Fender: so? They're so cute and small |
| 23:43.43 | SteveTotaro | app_ices |
| 23:43.58 | maqr | SteveTotaro: seriously? |
| 23:43.59 | rizwank | MoH server? |
| 23:44.05 | maqr | SteveTotaro: name one, i gotta see this |
| 23:44.22 | rizwank | We're going to have a phone number people can call to listen to music. Aside from why... |
| 23:44.38 | rizwank | it'll be receving sip streams, so no external cards needed |
| 23:45.01 | SteveTotaro | seriously, yes, name one, no |
| 23:45.24 | *** part/#asterisk Corydon76-dig (i=black@pdpc/supporter/bronze/Corydon76-home) |
| 23:45.35 | SteveTotaro | i don't want N SA to up me on their lists |
| 23:45.53 | maqr | unlikely |
| 23:46.05 | SteveTotaro | i already get SSSSSSS at the bottom of every boarding pass |
| 23:46.21 | *** join/#asterisk Corydon76-dig (i=green@pdpc/supporter/bronze/Corydon76-home) |
| 23:46.21 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 23:46.31 | maqr | SteveTotaro: some googling reveals that since DOCSIS1.1, there's been an option for 56bit 3DES encryption for all cable modems communicating to the switch (or whatever it's called in cable speak) |
| 23:46.53 | SteveTotaro | look into A5/1 A5/2 |
| 23:46.55 | maqr | SteveTotaro: so, unless the ISP is thoroughly incompetent, or you've cracked 56bit 3DES (which maybe is possible?), then you can't sniff anything off a cable modem |
| 23:46.59 | hardwire | maqr: CMTS |
| 23:47.12 | maqr | thanks hardwire , that's it |
| 23:47.16 | ionix | maqr: He is fcking toying with you. Stop feeding him. Just whois him, look at his hostname, think and understand why he can't be hacking his cable modem |
| 23:47.19 | hardwire | maqr: that mostly means nothing |
| 23:47.45 | maqr | lol verizon |
| 23:47.50 | hardwire | encryption on the wire to a CMTS has nothing to do with encryption ala IP packets |
| 23:47.57 | ionix | can we go back to Asterisk now? |
| 23:48.08 | [TK]D-Fender | maqr, 56bit = DES :) |
| 23:48.09 | SteveTotaro | wanna see me use half a dozen ips in a few minutes? LOL 8-) |
| 23:48.21 | maqr | hardwire: wouldn't it mean that capturing the encrypted packets heading to the CMTS would be useless? |
| 23:48.34 | hardwire | maqr: if you want to call them packets |
| 23:48.35 | maqr | ionix: and hold on, i've always wondered about this, so i'm going to pick hardwire's brain while i have the chance :) |
| 23:48.35 | ionix | SteveTotaro: Your identd is on. Better luck next time |
| 23:48.40 | [TK]D-Fender | rizwank, streaming isn't the most stable thing with *. Not sure how it scales. |
| 23:48.58 | rizwank | Got it. Thanks. |
| 23:49.25 | hardwire | maqr: /msg me if you want more info |
| 23:49.39 | [TK]D-Fender | rizwank, since it'd be an external app I think * might just "hook in" to a single instance at which point I'm not sure it would really represent any real load. |
| 23:49.47 | SteveTotaro | http://cgi.ebay.com/USB-24dbi-Yagi-Antenna-802-11-Wireless-Wifi-2-4ghz_W0QQitemZ170219557698QQihZ007QQcategoryZ74954QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 23:50.00 | rizwank | seems as much, I just wanted to get a sanity check =) thanks |
| 23:50.01 | SteveTotaro | that bad boy with a servo on the roof |
| 23:50.33 | ionix | SteveTotaro: Nobody believes you anymore. |
| 23:50.50 | SteveTotaro | that's fine, and you cannot get inbound to work, LOL |
| 23:51.04 | ionix | at least I am not lying :) |
| 23:51.10 | SteveTotaro | neither am i |
| 23:54.01 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:54.27 | SteveTotaro | my inbound works 1.888.777.1888 :-D |
| 23:54.58 | *** part/#asterisk rizwank (n=rizwank@dsl093-116-242.lax1.dsl.speakeasy.net) |
| 23:55.45 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 23:55.54 | *** join/#asterisk sack (n=sack@33.Red-81-32-160.dynamicIP.rima-tde.net) |
| 23:56.35 | SteveTotaro | darren, what's up? |
| 23:57.57 | *** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner) |
| 23:59.10 | *** join/#asterisk plla (n=h@200.31.103.86) |
| 23:59.37 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-aebf6b6617565d78) |
