00:00.01 | ManxPower | One of our internal "standard patches" is to set it to 350 |
00:00.15 | blitz[laptop] | ManxPower: yo buddy... wuz up? |
00:00.36 | file[laptop] | blitz[laptop]: yay |
00:00.46 | blitz[laptop] | file[laptop]: zup |
00:00.57 | file[laptop] | blitz[laptop]: oh nothin' |
00:01.08 | iCEBrkr | Don't make me go get on my laptop |
00:01.12 | file[laptop] | oh oh oh |
00:01.20 | ManxPower | blitz[laptop], Other than the fact that god hates me? Not much. |
00:01.21 | file[laptop] | the soup nazi episode of Seinfeld is on |
00:01.22 | blitz[laptop] | iCEBrkr: TSN! |
00:01.24 | Bentley | ManxPower: so if this was a problem using a sip-pstn gateway, and not zap card, then I am prolly S.O.L., eh? |
00:01.34 | ManxPower | Bentley, Yup. |
00:01.36 | rene- | A quick question, is realtime supported in sub 1.2 versions of asterisk ( in 1.0.9 ? ) |
00:01.40 | file[laptop] | rene-: no |
00:01.41 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
00:01.44 | blitz[laptop] | nope |
00:01.46 | rene- | :( |
00:01.48 | ManxPower | rene-, no and it never will be |
00:01.53 | blitz[laptop] | screw 1.0.9 |
00:02.05 | file[laptop] | screw stable! up with 1.2.0! |
00:02.15 | BoRiS | stable....pfffffff |
00:02.34 | BoRiS | lol :-p |
00:02.41 | ManxPower | If you need Realtime wait a year or two until 1.2 is released. |
00:02.59 | blitz[laptop] | I'm getting ready to move things from 1.0 to 1.2 beta soon |
00:03.07 | blitz[laptop] | and then keep up with head |
00:03.12 | iCEBrkr | Hrrm. |
00:03.18 | blitz[laptop] | 1.0.9 is just giving me too many problems |
00:03.21 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
00:03.22 | iCEBrkr | I should upgrade my 1.0.9 stuff |
00:03.28 | rene- | I know, but my boss is having the idea of downgrading to 1.0.9 to see if that will help in our intermitent one way audio issue our realtime based GUI notwithstanding |
00:03.37 | ManxPower | I upgraded to CVS-HEAD a while ago. What a nightmare. |
00:04.06 | ManxPower | Eventually I found a CVS date that worked with only minor oddities. |
00:04.12 | asteriskmonkey | im tryign to creat a small context that lets me dial a 10 digit number out once dialed in the asterisk :P anyone know how i go about that? |
00:04.27 | MacRohard | Hey I wrote a 4 line patch to asterisk (res/res_agi.c - the end of the setup_env function) that exports variables to your AGI scripts.. someone might want to consider adding this to the main asterisk tree (I can't even believe it's not there already) |
00:04.27 | MacRohard | <PROTECTED> |
00:04.27 | MacRohard | <PROTECTED> |
00:04.27 | MacRohard | <PROTECTED> |
00:04.28 | MacRohard | <PROTECTED> |
00:04.28 | iCEBrkr | asteriskmonkey: IT'S IN THE WIKI |
00:04.50 | blitz[laptop] | ~docs |
00:04.52 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
00:04.52 | blitz[laptop] | ~thebook |
00:04.53 | jbot | somebody said thebook was Asterisk: The Future of Telephony by Jim Van Meggelen, Jared Smith & Leif Madsen, published by O'Reilly Media. It can be found at http://www.oreilly.com/catalog/asterisk for purchase, or FREELY AVAILABLE under the Creative Commons license at http://www.asteriskdocs.org |
00:05.28 | mog_work | wow blitz we get it |
00:05.31 | mog_work | you wrote a book |
00:05.33 | mog_work | it rocks |
00:05.36 | iCEBrkr | haha |
00:05.37 | mog_work | no need for spammination |
00:05.38 | ManxPower | MacRohard, submit it on bugs.digium.com You might have to submit a disclaimer. |
00:05.49 | e-milio | hello all |
00:06.19 | ManxPower | blitz[laptop], I read the first chapter or so. It's much better than I expected it to be. |
00:06.46 | e-milio | Have anybody made spandsp work stable ? |
00:07.03 | e-milio | I mean with 100% fax sending success ? |
00:08.00 | blitz[laptop] | mog_work: sorry... I wasn't really trying to spam for the book... I was really just trying to direct someone to documentation |
00:08.10 | e-milio | We are having probs with txfax. Page random cuts in halp |
00:08.10 | blitz[laptop] | I should really just change that link to show a link |
00:09.39 | mog_work | yeah... ^)^ |
00:09.57 | file[laptop] | you will never get 100% success |
00:09.59 | file[laptop] | nevar! |
00:10.10 | e-milio | i will set for 80% |
00:10.15 | alephcom | Not that you get 100% over the pstn anyways. :-) |
00:10.24 | file[laptop] | you can make it happen on digital stuff... a PRI |
00:10.31 | e-milio | we have a pri |
00:10.52 | e-milio | one fax is fine |
00:10.58 | e-milio | to not |
00:11.05 | e-milio | two not |
00:11.20 | blitz[laptop] | I'm not going to make any money on it... so I'm not spamming for that reason.. just trying to give the link out so people can use it |
00:11.47 | mog_work | i hear you blitz |
00:11.55 | mog_work | im not trying to reign on your parade |
00:12.06 | file[laptop] | Matttttt |
00:12.24 | e-milio | file[laptop]: Have you had success with spandsp |
00:12.25 | e-milio | ? |
00:12.29 | blitz[laptop] | mog_work: I know :) |
00:12.30 | file[laptop] | e-milio: yes. |
00:12.35 | file[laptop] | only for receiving though |
00:12.38 | e-milio | mmm |
00:12.43 | e-milio | no we need to send |
00:12.44 | mog_work | fileeeee |
00:12.45 | file[laptop] | sending use hardware. |
00:12.50 | blitz[laptop] | just changes the jbot tag... should be a lot shorter now |
00:12.58 | file[laptop] | mog_work: how are you? |
00:13.11 | mog_work | im actually a little iritated |
00:13.14 | blitz[laptop] | going back to watch the leafs game |
00:13.16 | mog_work | had a hell of a call |
00:13.25 | e-milio | file[laptop]: which hardware would you recommend ? |
00:13.35 | file[laptop] | brooktrout |
00:13.41 | file[laptop] | mog_work: :( |
00:14.35 | mog_work | it happens |
00:14.37 | MacRohard | ManxPower, ok. done. |
00:14.46 | e-milio | file[laptop]: with * or other pkg ? |
00:15.13 | file[laptop] | hylafax |
00:15.14 | mog_work | with asterisk get a t1 card |
00:15.26 | mog_work | with hylafaxits only that brooktrout right? |
00:15.42 | e-milio | we are trying with t406p and pri |
00:15.46 | e-milio | not good |
00:16.12 | e-milio | dont see much cpu usage neither irq activity but |
00:16.23 | mog_work | what seems to be the problem with that emillio |
00:16.24 | e-milio | simultaneous faxes just cut |
00:16.41 | mog_work | ouch |
00:16.45 | e-milio | ramdonly the faxes just get cut in the middle |
00:16.54 | mog_work | hmm using latest spandsp? |
00:16.59 | e-milio | 0.0.2 |
00:17.32 | e-milio | i tried on two dif machines |
00:17.36 | e-milio | with 2 dif cards |
00:17.40 | e-milio | with 2 dif pris |
00:17.47 | e-milio | with 2 dif providers |
00:17.56 | e-milio | same results |
00:18.23 | mog_work | yuck |
00:18.34 | mog_work | hylafax is probably way to go |
00:19.05 | e-milio | is hylafax and two ext modem stable ? |
00:20.16 | file[laptop] | very very hungry |
00:20.23 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
00:23.34 | alephcom | e-milio: I have nothing but good to say about hylafax when it comes to stability. |
00:24.07 | *** part/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
00:25.02 | e-milio | alephcom: using any hw in particular ? |
00:25.09 | *** join/#asterisk gbrowley (n=gbrowley@221.127.107.205) |
00:26.02 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
00:26.50 | *** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
00:27.48 | ManxPower | MacRohard, Be prepared to defend your patch to the death. |
00:28.48 | alephcom | e-milio: We used just a USR sportster. |
00:29.01 | gbrowley | I'm having a small problem with running my asterisk server on a dynamic IP address, when the IP address changes although asterisk reports them as registered when a converstaion takes place they can hear me but I can not hear the remote side. Anyone have any ideas? |
00:29.19 | alephcom | e-milio: we quite using hylafax because we needed a nicer windows interface than was available but that was the only problem. |
00:30.10 | e-milio | alephcom: i just need to send pre-prepared tiff files, 2 o 3 simultaneoulsly |
00:34.02 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
00:34.03 | MacRohard | ManxPower, why do you say that ? |
00:34.27 | ManxPower | MacRohard, you'll understand soon, grasshopper. |
00:34.32 | MacRohard | (i don't .. i don't really give a fuck.. I already patched *my* asterisk) |
00:34.50 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
00:35.08 | ManxPower | MacRohard, you can also submit your patch to the openpbx people |
00:35.14 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
00:35.25 | MacRohard | is that like some asterisk fork? |
00:35.44 | sahafeez | how do you set the outbound callerid for everyline. i want every call out the pri to say the same thing |
00:36.09 | ManxPower | sahafeez, Set it before your outbound Dial statement |
00:36.26 | sahafeez | ah, thanks. |
00:38.36 | ManxPower | MacRohard, Yes, it's an "fully GPL" fork of Asterisk. |
00:40.20 | asteriskmonkey | I can seem to get asterisk to do a callthrough thing properly |
00:40.34 | asteriskmonkey | ive tired usign the example online and that didnt work |
00:46.21 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
00:46.33 | e-milio | please help: How can I detect if a pri is having frame slips ? |
00:47.46 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
00:48.41 | *** join/#asterisk juice (n=juice@mo-67-77-189-131.dyn.sprint-hsd.net) |
00:48.42 | *** part/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
00:48.57 | e-milio | community: How can I detect if a pri is having frame slips ? |
00:50.20 | ManxPower | Looks like I need to get a new USA passport before they start putting RFID chips in them |
00:50.41 | Sedorox | you'll proably have to go get a new one |
00:50.41 | ManxPower | e-milio, you will hear occasional clicks in the audio. |
00:50.58 | *** part/#asterisk asteriskmonkey (n=phil@HSE-Windsor-ppp211407.sympatico.ca) |
00:51.06 | ManxPower | you want the 2nd field of your span= line to be 1 for the line you want to get your timing from. |
00:51.13 | e-milio | ManxPower, any log anywhere ? |
00:51.41 | ManxPower | e-milio, not that I am aware of. |
00:51.51 | ManxPower | HDLC abort errors are not frame slips |
00:52.01 | e-milio | ManxPower: mm |
00:52.53 | e-milio | ManxPower: any idea how to prevent them ? |
00:53.05 | ManxPower | e-milio, frame slips or HDLC abort errors? |
00:53.15 | e-milio | ManxPower: HDLC |
00:54.03 | ManxPower | e-milio, Read the Wiki, search the mailing list archives, replace your motherboard with a different brand/model, don't use SATA drives, don't use RAID controllers, enable IDE unmask interrupts, enable IDE DMA |
00:54.07 | ManxPower | ~docs |
00:54.08 | jbot | i guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
00:54.09 | ManxPower | ~mailinglist |
00:54.10 | jbot | somebody said mailinglist was Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
00:54.21 | tzanger | basically you want a good PCI subsystem |
00:54.57 | ManxPower | e-milio, HDLC Abort errors are basically cause by 2 things. 1) PCI IDE issues (the most common) and 2) line problems (uncommon) |
00:55.12 | e-milio | ManxPower: I am using scsi |
00:55.30 | e-milio | with raid |
00:55.41 | e-milio | not good 4 what you said |
00:56.03 | ManxPower | Results 1 - 10 of about 250 from lists.digium.com for HDLC error. (0.23 seconds) |
00:56.28 | ManxPower | e-milio, RAID, SATA, etc are all just things that commonly lock interrupts for so long that the zaptel card loses data |
00:56.52 | ManxPower | other common things are graphics modes, etc |
00:57.14 | e-milio | ManxPower: we are running very basic. only the raid might be |
00:57.18 | ManxPower | Results 1 - 10 of about 791 from lists.digium.com for HDLC. (0.14 seconds) |
00:57.32 | ManxPower | e-milio, HDLC Abort errors are some of the hardest things to fix. |
00:57.39 | ManxPower | Those and Echo |
00:59.14 | e-milio | ManxPower: do you think hdlc can be break fax trans with spandsp ? |
00:59.24 | tzanger | e-milio: fuck yeah |
00:59.30 | *** join/#asterisk PBXtech (i=nik@223.sub-70-213-183.myvzw.com) |
00:59.41 | tzanger | fax/modem calls are *extremely* sensitive to that kind of loss |
00:59.55 | ManxPower | e-milio, not can, WILL |
01:02.40 | e-milio | tzanger, ManxPower, thanks |
01:07.23 | *** join/#asterisk froguz (n=froguz@2-43-112.adsl.terra.cl) |
01:07.27 | froguz | hi |
01:08.41 | PBXtech | is there a skype to asterisk solution? |
01:08.58 | ManxPower | PBXtech, No. |
01:09.11 | PBXtech | not even external hardware? |
01:09.36 | ManxPower | Is there a POTS/Skype adapter that supports far end disconnect supervision? |
01:10.13 | PBXtech | got me :) |
01:10.26 | froguz | somebody can tell me why Asterisk@Home uses a macro called hangupcall wich first reset the CDR before doing a simple Hangup? is that necesary? (this macro is called from app-callforward) |
01:10.44 | ManxPower | froguz, that's really a question for the A@H channel |
01:11.09 | mog_work | yes yes it is ^_^ |
01:11.11 | PBXtech | or #amportal channel |
01:11.23 | froguz | oh i'm sorry i didn't know that channel exist |
01:12.10 | mog_work | its all good froguz |
01:12.18 | froguz | thanks |
01:12.21 | mog_work | you just wont find a lot of love for amp and a@h |
01:12.22 | mog_work | here |
01:12.59 | ManxPower | A@H is to Asterisk, as a McDonalds hamburger is to a quality steak. |
01:13.12 | *** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
01:13.49 | froguz | hahaha i can notice that, i don't love a@h neither, it's just i have to deal with it |
01:14.24 | mog_work | no you dont |
01:14.28 | mog_work | break free my friend |
01:14.58 | froguz | well yes, i have. work, you know. i'm not my own boss |
01:15.07 | froguz | =(not yet |
01:15.41 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
01:18.51 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
01:21.50 | DrukenHME | what is the best variable to use for the caller id number? |
01:22.17 | file[laptop] | calleridnum |
01:23.33 | DrukenHME | file[laptop]: in your understanding... this would do ? |
01:23.37 | DrukenHME | GotoIf($[${LEN(${CALLERIDNUM})} = 7]?6:8) |
01:23.58 | mover | why app_rxfax work on one box fine and on another box same asterisk (head) but diffrent hardware no? I user ztdummy |
01:28.04 | DrukenHME | ok, never mind... i finally got it to work... at least i think so |
01:29.31 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
01:35.54 | infinity1 | anyone use destar? |
01:36.59 | infinity1 | yay. beta2 packages for debian! |
01:38.49 | sahafeez | http://www.voip-info.org/wiki-Asterisk+cmd+Directory |
01:38.57 | sahafeez | Pressing "1" will exit setting up the channel to enter the extension selecte |
01:39.10 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
01:39.15 | sahafeez | what does this look like in extentions.conf. there is no example. |
01:39.44 | sahafeez | i have the directory working, but after hitting 1 i do not get transferd |
01:40.04 | infinity1 | tzafrir_laptop: hey! i c you :) ...can i get an opinion on destar? thanks for the deb packages btw. |
01:42.06 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
01:47.05 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
01:47.28 | harryvv | Has anyone seen Arial around? |
01:48.43 | PBXtech | does the sc430 have the same probs as the sc420? |
01:48.57 | infinity1 | no one has |
01:48.57 | infinity1 | ariel There was no such nickname |
01:49.52 | *** join/#asterisk paravoid (i=paravoid@void.photonics.ece.ntua.gr) |
01:49.55 | infinity1 | has anyone used amp ? does it work well with a custom extensions.conf? |
01:49.57 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
01:50.42 | *** join/#asterisk dsmouse (n=mouse@prism.datastacks.com) |
01:51.04 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
01:53.34 | *** join/#asterisk Knight_DKN (n=knight_d@61.95.68.85) |
01:54.23 | *** join/#asterisk seacode (n=anthm@adsl-69-215-147-152.dsl.milwwi.ameritech.net) |
01:54.50 | *** join/#asterisk darkpixel (n=darkpixe@209.216.179.194) |
01:56.57 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
01:57.24 | PBXtech | is pot really legal in denver now days? damn radio station foolin or what |
01:58.45 | [Airwolf] | PBXtech, I don;t have to worry about that :P |
01:59.58 | *** part/#asterisk gbrowley (n=gbrowley@221.127.107.205) |
02:00.42 | *** join/#asterisk znoG (n=gs@200.115.217.183) |
02:01.18 | e-milio | ManxPower: where ztclock.c can be found ? |
02:03.04 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
02:03.05 | e-milio | never mind |
02:05.38 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
02:18.05 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
02:20.31 | justinu | any polycom experts around? |
02:21.20 | justinu | bootrom 2.6.2 support HTTP? |
02:22.02 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-225.range86-138.btcentralplus.com) |
02:22.23 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
02:24.39 | BhaalWK | Anyone here had experience with Alcatel OmniPCX Office units? Just wanna know how the rack mount kit works? |
02:26.53 | *** part/#asterisk CoaxD (i=coax@shell1.cornernet.com) |
02:33.36 | sahafeez | question: in extension the s,1 bla is a cache all right? or not? |
02:34.51 | justinu | what? |
02:36.03 | sahafeez | well s = start and in all the examples then have exten => s,1,app |
02:36.37 | justinu | so what's the question? |
02:36.41 | justinu | i didn't understand |
02:37.06 | *** join/#asterisk axscode (i=axscode@210.213.106.189) |
02:37.20 | axscode | ~fxo |
02:37.22 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
02:38.07 | sahafeez | hum. not sure i know. i know what my inbound number is...but what if i want to make a rule that says any inbound calls do this. |
02:38.18 | justinu | oh |
02:38.25 | justinu | you mean a "catch all" |
02:38.27 | *** part/#asterisk sgorilla (n=tlp@cpe-24-160-119-179.houston.res.rr.com) |
02:38.29 | justinu | not a cache all |
02:38.31 | sahafeez | yes |
02:38.41 | sahafeez | sorry. having brain issues for the past week. |
02:38.45 | justinu | no, i don't think s is a catch all |
02:38.57 | justinu | you want _N. iirc |
02:39.14 | axscode | what do i need from my computer so that i can connect and dial to POTS or my existing TELCO? |
02:39.34 | justinu | exactly what the bot said |
02:39.56 | blitz[laptop] | _X. << catch anything starting with 0-9 |
02:40.00 | blitz[laptop] | N == 2-9 |
02:40.03 | justinu | oh, right. |
02:40.05 | justinu | der |
02:40.09 | sahafeez | fxs = subscriber, fxo = office |
02:40.11 | blitz[laptop] | Z == 1-9 |
02:40.18 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
02:40.22 | blitz[laptop] | fxs == station |
02:40.47 | sahafeez | blitz: depend on who you talk to and how long they have been doing telco;) |
02:41.02 | blitz[laptop] | :) |
02:42.59 | sahafeez | i have exten => 2004,1,Directory(default,incoming,f) |
02:43.07 | axscode | so i need an fxs,fxo card? is there any PCI that is around aside from digium? |
02:43.33 | blitz[laptop] | nope |
02:43.42 | sahafeez | and i can spell the person, find it, hit the number 1 and get nothing. is there another line i am missing? do i need to defince and extension for 1 |
02:43.43 | blitz[laptop] | unless you use an ATA or something |
02:43.50 | axscode | whats an ATA? |
02:43.52 | axscode | ~ATA |
02:43.53 | jbot | hmm... ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
02:43.59 | justinu | axscode: sip ata, or sip channel bank |
02:44.14 | justinu | why wouldn't you use an ATA? |
02:44.20 | justinu | why put something in the pbx? |
02:44.23 | sahafeez | its a hack |
02:44.29 | justinu | how is it a hack? |
02:44.35 | blitz[laptop] | someone showed a wicked mini-channel bank that connects via ethernet that looks very sweet |
02:44.39 | mog_home | you could put something in the pbx |
02:44.41 | blitz[laptop] | a hack? I disagree |
02:44.44 | mog_home | aka 2400p |
02:44.48 | mog_home | and be dominate |
02:44.51 | sahafeez | just my 2 cents. depends on what you are trying to do i guess |
02:44.52 | blitz[laptop] | aye |
02:45.10 | justinu | i think the ATAs are quite elegant, you don't have to restart the server to add them |
02:45.11 | blitz[laptop] | mog_home: do you remember what that mini channel bank thing was at astricon? |
02:45.15 | sahafeez | i just got a PRI and one TDM400 for the fax machine |
02:45.22 | mog_home | yeah the 8 porter |
02:45.23 | justinu | if computers stop having PCI busses, you can still use your ATAs |
02:45.28 | blitz[laptop] | mog_home: yah... that thing |
02:45.30 | mog_home | xorcoms |
02:45.34 | blitz[laptop] | ahhh right |
02:45.35 | blitz[laptop] | thanks |
02:45.37 | axscode | anyone have tried using "PANASONIC D1323 16SLC Digital Super hybrid system" |
02:45.38 | justinu | i'd go with the redphone TDMoE boxes for PRI |
02:45.42 | sahafeez | so anyone on the directory stuff? i read the faq, no examples |
02:45.59 | blitz[laptop] | back later -- going to try focusing on programming for a bit |
02:46.04 | justinu | good luck |
02:46.10 | blitz[laptop] | thx -- I'll need it :) |
02:46.15 | blitz[laptop] | E911 portal :) |
02:46.21 | justinu | yay! :) |
02:47.43 | axscode | so ATA = ADAPtER for POTS to ETHERNET? |
02:47.48 | justinu | basically |
02:47.58 | justinu | check out the SPA-3000 |
02:48.04 | justinu | one FXS port, one FXO port. |
02:48.13 | justinu | very slick |
02:48.51 | axscode | im here at philippines.. mostly i cant find one |
02:49.05 | justinu | hmm... asia is very hot on voip |
02:49.07 | axscode | unless ill order from outside.. which i need one today. |
02:49.14 | justinu | i'm sure you can get one from singapore or hong kong |
02:49.19 | axscode | asia = singapore for voip.. |
02:49.26 | justinu | yeah |
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02:49.49 | justinu | does air asia fly to to manilla? |
02:49.54 | justinu | their flights are really cheap |
02:50.01 | justinu | you could almost afford to go to singapore just to pick some up |
02:50.12 | axscode | oh... and im poor.. so i doubt i can afford one. |
02:50.23 | justinu | they cost about 100USD |
02:50.28 | justinu | some are as cheap as 45USD |
02:50.31 | axscode | and i dont have that.. |
02:50.58 | axscode | thats why im looking for a clone brand for PCI card to connect to pots |
02:52.02 | axscode | justinu: ASTERISK ===> SPA3000 ===> TELCO |
02:52.22 | axscode | is that right? how will the exten => looks like? |
02:52.23 | justinu | yeah... and you can plug a regular analog phone into the SPA3000 also |
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02:52.56 | justinu | extension for what, dialing out? |
02:53.15 | axscode | dialing out.. and how about recieving a call from outside? is that posible? |
02:53.19 | justinu | sure |
02:53.24 | orlok | why not just start off with a performance car? :) |
02:53.29 | mwgbc | What kind of problem would cause the error: |
02:53.29 | mwgbc | Nov 3 09:37:58 NOTICE[15327]: pbx_spool.c:243 attempt_thread: Call failed to go through, reason 1 |
02:53.37 | orlok | Doh, wrong channel. |
02:53.43 | justinu | haha |
02:54.23 | axscode | justinu: can u gimme example how the dialing out will look like? do i need a zapata for that? |
02:54.27 | mwgbc | orlok: This error started spontaneously without anyone changing any conf files |
02:54.29 | justinu | no zapata |
02:55.06 | mwgbc | It was working previously |
02:55.23 | justinu | something like exten => _X.,1,Dial(SIP/${EXTEN}@spa3000); |
02:56.10 | justinu | axscode: you own't need any hardware in your asterisk box to make that work, besides an ethernet card |
02:56.30 | axscode | oh. nice... so i can dial outside just having an ATA? |
02:56.34 | axscode | hehe. sweet.. |
02:56.35 | justinu | correct. |
02:56.37 | sahafeez | ok then what is the S extension they have in every example |
02:56.52 | justinu | it's just the starting point for macros and stuff, i guess |
02:57.08 | axscode | this is cool.. ok. so i dont need digium modems.. |
02:57.16 | justinu | not really |
02:58.00 | axscode | justinu: how about when they dial-in to a specific asterisk sip user? is that stated also in extensions? |
02:58.08 | justinu | yeah |
02:58.29 | justinu | if you have caller ID, you can even route based on that. |
02:59.35 | axscode | ok so it will be. .. exten=> 4324343,1,(ask-for-new-sip-number) <---where 4324343 is the number of the POTS line? |
02:59.57 | justinu | well, yeah |
03:00.03 | axscode | ok.. got that.. |
03:00.10 | justinu | sorta |
03:00.13 | axscode | ok so ATA then.. |
03:00.23 | axscode | hmm. problem is.. where can i find that.. haha.. |
03:00.24 | justinu | thats what i'd do |
03:01.16 | axscode | hey justin.. any ATA right? |
03:01.23 | justinu | it needs an FXO port |
03:01.26 | axscode | or some ATA dont work for asterisk. |
03:01.35 | justinu | as long as it's SIP |
03:01.41 | justinu | or iax |
03:01.48 | justinu | digium sells some cool iax ATAs |
03:01.49 | axscode | so ATA with SIP support? or iax support... |
03:01.51 | justinu | very small |
03:01.59 | justinu | yeah, and FXO port |
03:02.02 | justinu | ~fxo |
03:02.03 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
03:02.08 | axscode | FXO, ETHER port |
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03:04.11 | justinu | axscode: http://www.sipura.com/products/spa3000.htm |
03:04.37 | froguz | is there a variable wich tell me the context for the calleridnum? |
03:05.10 | froguz | or any certain extension? |
03:05.42 | axscode | justinu: yups ive seen that one.. |
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03:07.02 | axscode | thanks.. gud help.. |
03:07.08 | justinu | no prob |
03:07.17 | axscode | otherwise. if thers none exist.. have to check for digium pci modems |
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03:14.18 | file[laptop] | to all those who died in the netsplit, we will miss you! |
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03:14.27 | file[laptop] | ah who am I kidding, no we won't |
03:14.46 | file[laptop] | oh looksee they're back from the dead |
03:16.25 | justinu | you were the ones who died |
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03:21.31 | Lostfrog | wow.. that's one hell of a split. |
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03:23.47 | axscode | asl |
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03:24.56 | Lostfrog | lol.. asl on freenode. |
03:25.36 | iCEBrkr | wow and lilo isn't spamming us with a reason why and how we should hang in there? |
03:25.36 | iCEBrkr | UNFUCKINGBELIEVEABLE |
03:25.45 | Qwell | iCEBrkr: He's with them |
03:25.54 | iCEBrkr | huh? |
03:26.35 | Qwell | he's spamming the people that split |
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03:27.03 | iCEBrkr | hahaha |
03:28.26 | Qwell | took long enough |
03:30.22 | Lostfrog | I like getting packages. :) |
03:31.22 | DrukenHME | i personally suffer from sometimers |
03:31.48 | iCEBrkr | I dunno if I'm in a sober nuff state of mind to upgrade to 1.2.0 beta2 |
03:31.48 | Lostfrog | Mmm.. 50pk Verbatim 8x DVD-R |
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03:32.10 | Lostfrog | Gumbles? |
03:32.35 | nomazda | hmm |
03:34.07 | DrukenHME | jessus christ... 22C in the afternoon, 5C overnight.... it's just not right |
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03:35.37 | Qwell | funny |
03:35.38 | kuku5 | anyone max out the broadvoice line |
03:35.38 | kuku5 | like what is the limit |
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03:36.27 | Lostfrog | You are only supposed to use one incoming or one outgoing. |
03:36.44 | kuku5 | for the unlimited plan |
03:37.00 | Lostfrog | At least, that's how I read the service agreement. |
03:37.03 | Lostfrog | Ohh.. minutes. |
03:37.17 | Lostfrog | I heard if you use too many minutes, they bump you up to business use. |
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03:38.22 | Lostfrog | I won't ever get a residential plan from them.. |
03:38.25 | blitz[laptop] | iCEBrkr: :D |
03:38.25 | blitz[laptop] | rm -f /usr/include/asterisk/* and /usr/lib/modules/asterisk/* |
03:38.25 | blitz[laptop] | seems to help cause less problems once you do your make install |
03:38.27 | iCEBrkr | CoaxD: ?? |
03:38.27 | tzanger | CoolAcid: ? |
03:38.27 | tzanger | er CoaxD ? |
03:38.27 | CoaxD | gee. I'll leave my fucking toll-free DID customers hanging for 3 days without even posting so much as a news item on their website describing the outage |
03:38.37 | Lostfrog | I've read too my bad stories. |
03:38.54 | iCEBrkr | blitz[laptop]: I DID! I DID! |
03:39.08 | iCEBrkr | blitz[laptop]: Oh, it built just fine. |
03:40.00 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
03:40.04 | blitz[laptop] | now that I got java working in my browser... I forget why I needed it working in the first place... |
03:40.04 | iCEBrkr | blitz[laptop]: Just gotta mod-up extensions.conf |
03:40.04 | iCEBrkr | tzanger: Hah, I did the same, but corrected it |
03:40.05 | Lostfrog | Could someone explain todays Dr. Fun to me? |
03:41.02 | blitz[laptop] | nice |
03:41.02 | iCEBrkr | I'm a bit buzzed. |
03:41.05 | blitz[laptop] | can't wait till I get back to my house... I think I'll have a bunch of packages waiting for me |
03:41.07 | blitz[laptop] | some money too me thinks! |
03:41.29 | iCEBrkr | weeeeeeeeeeeeeeeeeewhooooooooooooooooooooooooo |
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03:42.05 | blitz[laptop] | mog_home: lol |
03:42.05 | blitz[laptop] | mog_home: come on over! |
03:42.21 | mog_home | i dont think i have enough gas to do it |
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03:42.54 | blitz[laptop] | mog_home: how far of a drive is KC to Huntsville? |
03:42.54 | tzanger | I'll meet ya there |
03:42.54 | kuku5 | i have the biz plan |
03:44.08 | iCEBrkr | dude, did System() change in 1.2.0 beta2 |
03:44.12 | CoaxD | if i wanna buy..say...a used car engine.. aside from ebay, where does one go? |
03:44.12 | iCEBrkr | ? |
03:44.33 | Qwell | CoaxD: pick a part? |
03:44.57 | mog_home | junk yard coaxd |
03:46.44 | MacRohard | CoaxD, a scrapyard? |
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03:50.39 | tzanger | CoaxD: scrapyard is my answer too |
03:51.09 | Lostfrog | IMHE, used engines are bad news. |
03:51.29 | iCEBrkr | oh WTF is this IAX / to - shit?! |
03:51.49 | ManxPower | iCEBrkr, I think it's to make it consistent with the other channels |
03:52.00 | mog_home | not always |
03:52.06 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
03:52.10 | ManxPower | i.e. so you don't have to have one regex for IAX and another for any other channel name when parsing channel instances |
03:52.11 | mog_home | i have a friend that is good friends with a junkyard owner |
03:52.18 | iCEBrkr | Ahh |
03:52.18 | mog_home | he gets loads of cool stuff |
03:52.21 | mog_home | everyweekend |
03:52.32 | harryvv | who does |
03:52.36 | mog_home | like he recently got a 2000 dollar projector for 100 |
03:53.01 | harryvv | CoaxD, uses engine for what reason |
03:53.04 | iCEBrkr | doh |
03:53.07 | iCEBrkr | it's hang'n up on me. |
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03:53.19 | harryvv | mog, who does? |
03:54.03 | mog_home | who does what |
03:54.03 | harryvv | get those prices |
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03:54.03 | mog_home | junkyard |
03:54.03 | harryvv | at deep discounts |
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03:54.04 | Lostfrog | ok.. that's annoying. |
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03:54.08 | mog_home | had bad powersupply i think |
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03:54.21 | harryvv | mog, a friend of yours got a projector at a junk yard for 100 dollars? |
03:54.29 | harryvv | makes no sence. |
03:54.30 | mog_home | yeah |
03:54.32 | Qwell | They have projectors at junkyards? |
03:54.39 | mog_home | they have everything |
03:54.44 | mog_home | it depends what kind it is |
03:54.48 | Vco | dead hookers, you name it.. |
03:54.54 | mog_home | he is outside of newjersey |
03:54.56 | mog_home | so maybe |
03:54.57 | Lostfrog | Dead hooker? where? |
03:54.57 | harryvv | you must be confused with electronic reclyclers right? |
03:55.04 | Lostfrog | lol |
03:55.16 | Lostfrog | I guess noone wants to explain Dr. Fun to me. |
03:56.33 | iCEBrkr | Wow.. DBPut/DBGet got ugly |
03:56.33 | harryvv | this h5n1 virus is very serios. Recent investigation dicovered that the 1918 flue pandemic was a bird flue. |
03:57.08 | Vco | telecommuters of the world rejoice |
03:57.28 | Lostfrog | Why so? |
03:57.40 | harryvv | Lostfrog, why so what? |
03:57.59 | Lostfrog | <Vco> telecommuters of the world rejoice |
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03:59.13 | harryvv | I was just watching a hour long special on the bird flue that is hitting south east asia. this is far more serios then I ever though. The h5n1 virus is constantly mutating and eventually, it will mutate into a human to human form. |
03:59.32 | Vco | it'll mutate into human form? |
03:59.41 | ManxPower | I hope tomorrow will be better than today. |
03:59.45 | Vco | like walking and talking and shit? |
03:59.46 | harryvv | human transferable form of a virus |
03:59.51 | Vco | i know i know |
03:59.54 | iCEBrkr | xten => s,1,Set(CIDNUM=${DB(callerid/number)}) |
03:59.57 | iCEBrkr | That's FUCKING UGLY |
04:00.01 | Vco | it'd be cooler the other way |
04:00.14 | ManxPower | Today started off with a BAT (Big Ass Truck) backing into the borrowed car, which we call the Death Mobile because it's brakes don't work well. |
04:00.20 | harryvv | it worrys me because we have a large chinese population here. |
04:00.35 | Vco | <in china> |
04:00.41 | harryvv | in vancouver bc |
04:00.46 | ManxPower | Then the person who's car we were supposed to borrow had a family emergency so we could not borrow their car to go to the bank and go to the dealer and get a damn decent car. |
04:00.47 | Vco | you'r ein richmond right? |
04:00.50 | harryvv | no |
04:00.53 | Vco | oh... |
04:01.09 | Vco | wait...no ...that was soeone at the office today... |
04:02.56 | ManxPower | iCEBrkr, no more ugly than exten => _XXXX,8,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?12:9) |
04:03.31 | ManxPower | Or my fave: exten => _XXXX,13,GotoIf($[${LEN(${DIAL_DEST[${INDEX}]})} = 0]?14:5) |
04:04.16 | *** join/#asterisk cripito (n=ncripito@c-67-161-130-59.hsd1.co.comcast.net) |
04:04.18 | cripito | hi |
04:04.31 | ManxPower | I must admit using "subscripts" with variables in extensions.conf can rock. |
04:04.48 | cripito | did anyone ever connect an oxygen pbx to asterisk? |
04:05.05 | mog_home | oxygen pbx? |
04:05.30 | cripito | yup 1 sec |
04:05.50 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
04:06.11 | cripito | OOH323/AltiServ VoIP Line |
04:06.11 | Brijn | Good evening all |
04:06.32 | iCEBrkr | ManxPower: It's just unfortunate that syntax kinda sucks ass. |
04:06.32 | cripito | any way to known what codec is coming in h323? |
04:06.35 | cripito | sorry in ooh323 |
04:06.40 | iCEBrkr | ManxPower: But ya get used to it.. I guess |
04:06.48 | mog_home | where is oxygen pbx |
04:06.57 | Katty | hihi |
04:06.59 | ManxPower | iCEBrkr, soon you will see beauty in the chaos, grasshopper. |
04:07.26 | cripito | snif snif |
04:07.29 | cripito | that all what i got.. |
04:07.42 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM] |
04:07.42 | Brijn | Katty: You have Polycom SP 501's? Do you happen to have the Admin PDF? They are not avail for download anymore (none that Google turned up at least) |
04:08.19 | cripito | this is what my asterisk is saying |
04:08.21 | cripito | 23:38:06:774 ERROR:Incompatible audio capabilities - Can not open audio channel |
04:08.21 | cripito | <PROTECTED> |
04:08.33 | Katty | Brijn: what makes you think i have that information? |
04:08.43 | iCEBrkr | hehe |
04:08.43 | Katty | Brijn: google. |
04:09.03 | Brijn | I thought that you had 501's (or at least one), and was hoping you have the PDF |
04:09.15 | Katty | Brijn: we have 500s. |
04:09.16 | Brijn | The lniks that Google turns up give 404's |
04:09.36 | cripito | i think the pdf for 500 |
04:09.44 | cripito | i have that here |
04:09.48 | Brijn | cripito: that would be nice! |
04:10.12 | cripito | yeap i got userguide and adminguide |
04:10.15 | cripito | where u want it? |
04:10.37 | Brijn | mail, or if you have website/ftp close by, also ok |
04:10.38 | ManxPower | Brijn, You mean like the ones here: http://www.freedomphones.net/polycom/files |
04:10.59 | ManxPower | The Wiki is your friend. Love it, hold it, buy it flowers. |
04:11.04 | Brijn | Urmmmmmm |
04:11.06 | cripito | :D |
04:11.07 | Brijn | Ooops |
04:11.13 | Brijn | Thanx :) |
04:11.15 | *** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net) |
04:11.35 | cripito | just in case..;) i don't plan to delete those ones ;) |
04:11.48 | Lostfrog | Can someone find me that question for 42? lol |
04:11.59 | Lostfrog | six times nine.. |
04:12.00 | Lostfrog | I know. |
04:12.03 | Lostfrog | Sheesh. |
04:12.05 | cripito | ok i need a h323 guru |
04:12.08 | cripito | ;) |
04:12.32 | cripito | snif snif this f&*&*# pbx don't connect |
04:12.41 | iCEBrkr | b00000000 |
04:12.42 | iCEBrkr | <PROTECTED> |
04:12.56 | Lostfrog | nice. |
04:13.05 | Lostfrog | Guess they need a new module for 1.2.0. |
04:13.09 | iCEBrkr | :( |
04:13.21 | iCEBrkr | size does matter! ;'( |
04:13.25 | ManxPower | Lostfrog, Should not need one. |
04:13.27 | cripito | ice did u put the g729 codec in the washing machine? |
04:13.34 | iCEBrkr | cripito: drier |
04:13.37 | cripito | :D |
04:13.39 | Lostfrog | Yeah.. it must of shrunk. |
04:13.47 | ManxPower | iCEBrkr, everyone knows that! |
04:14.10 | cripito | well manx it depends... |
04:14.27 | cripito | depends who hold the short one ;) |
04:16.12 | ManxPower | I downloaded and installed the G729 codec from Digium the other day into a CVS-HEAD of a month or so ago |
04:16.15 | Katty | right. so when's hte next conference? |
04:16.22 | Lostfrog | And where? |
04:16.28 | cripito | next astricon? |
04:16.32 | cripito | yeap... |
04:16.36 | cripito | good to known |
04:16.45 | Katty | file: pester bkw about conference. |
04:16.48 | cripito | please denver.. please please |
04:16.58 | Lostfrog | Please East coast. |
04:16.59 | Lostfrog | :) |
04:17.08 | cripito | and if we get another hurricane? |
04:17.11 | cripito | naaaaaa |
04:17.19 | cripito | better center or montain ;) |
04:17.30 | Lostfrog | I would like to eventually attend one, cripito. |
04:17.32 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) [NETSPLIT VICTIM] |
04:17.36 | Corydon76-home | Yeah, next conference in Huntsville! |
04:17.36 | iCEBrkr | welp, ulaw will have to do for now |
04:17.47 | ManxPower | I suggest Birmingham or Atlanta |
04:18.00 | FuriousGeorge | hey all |
04:18.01 | Lostfrog | Georgia would be cool. |
04:18.07 | cripito | yeap.. |
04:18.17 | cripito | still prefer colorado to georgia... |
04:18.22 | iCEBrkr | Georgia and Alabama are both in driving range for me.. :) |
04:18.35 | Lostfrog | It was in Atlanta in 2004 |
04:18.40 | hypa7ia | toronto! |
04:18.42 | cripito | :) well driver range.... is also from me |
04:18.51 | hypa7ia | astricon 2006 is going to be in baaaahston |
04:18.52 | cripito | just 900 miles away |
04:18.53 | Katty | it will probably be in chicago or vegas guys. |
04:19.00 | hypa7ia | i think i recall oej saying |
04:19.07 | cripito | VEGAS!!! |
04:19.24 | harryvv | Or vancouver BC the most livable city in the world. |
04:19.42 | Lostfrog | Boston would be cool too. |
04:19.48 | Juggie | ummmmm |
04:19.52 | Juggie | astricon 2006 is in dallas |
04:19.54 | harryvv | unfortunaly that is driving our realestate cost though the roof. |
04:19.58 | Juggie | it was announced already |
04:20.00 | cripito | i am in the next even if i have to walk to the conferece |
04:20.18 | iCEBrkr | show g729 |
04:20.18 | iCEBrkr | 0/0 encoders/decoders of 2 licensed channels are currently in use |
04:20.19 | iCEBrkr | w00t |
04:20.26 | Katty | Juggie: i'm talking about cluecon :) |
04:20.27 | cripito | nice ice |
04:20.31 | ManxPower | I just got an e-mail that I can't decide if it's spam or a comment on one of my web pages. |
04:20.43 | cripito | :D |
04:20.45 | Lostfrog | It can't be both? |
04:20.53 | FuriousGeorge | ManxPower: probably just clever spam |
04:21.04 | harryvv | im off |
04:21.06 | cripito | hi george... |
04:21.14 | cripito | mmm that's was fast. |
04:21.25 | FuriousGeorge | cripito: wadup |
04:21.31 | Katty | twisted[asteria]: can i drive to your house and then you drive to dallas for astricon next year? |
04:21.40 | cripito | :D |
04:21.49 | ManxPower | FuriousGeorge, I'm not sure. |
04:22.27 | Katty | twisted[asteria]: are you still at work? |
04:22.35 | Katty | twisted[asteria]: don't make me get my nextel and beepbeep you |
04:22.42 | Lostfrog | wow.. /me beats his boss till he pays his $3000 + dCAP fee and plane ticket and hotel to Miami. |
04:23.00 | blitz[laptop] | hypa7ia: Astricon 2006 is in Dallas, TX |
04:23.00 | FuriousGeorge | ManxPower: my mom got a bug in her laptop somehow that read her mail in ms office and sent her chopped up pieces of mail meant to look like actual correspondence. was eerie |
04:23.04 | Lostfrog | He'd probably die before he paid. |
04:23.09 | hypa7ia | ooh |
04:23.13 | cripito | furiosgeorge: no much.. trying to figure out how i can delete 3 querys that are the same in a realtime extension call |
04:23.16 | hypa7ia | i stand corrected :) |
04:23.28 | iCEBrkr | shit, I forget how to tell which codec a call is using |
04:23.28 | blitz[laptop] | I wish it was in Boston than Dallas |
04:23.28 | FuriousGeorge | cripito: I give up :) |
04:23.32 | cripito | snif snif 3 mega bucks |
04:23.38 | blitz[laptop] | I'm not such a fan of the red states :) |
04:23.40 | Lostfrog | blitz[laptop]: me too. |
04:23.46 | cripito | i think can be done... |
04:23.50 | file[laptop] | CANADA ASTRICON! |
04:23.56 | blitz[laptop] | file[laptop]: no kidding |
04:24.07 | FuriousGeorge | is it in that part of canada with the legal pot? |
04:24.14 | ManxPower | Gads, no. If Astricon was in Canada I'd not spend any time at the conference. |
04:24.17 | blitz[laptop] | there is no legal pot part |
04:24.35 | FuriousGeorge | blitz[laptop]: err, strongly decriminalized |
04:24.37 | blitz[laptop] | although BC is very liberal about it |
04:24.41 | Juggie | pot isnt lega |
04:24.43 | Juggie | er, legal |
04:24.46 | Juggie | but it isnt illegal eiher |
04:24.49 | cripito | by the way anyone known what is the max g729 channels that a soekris card can handle? |
04:24.51 | blitz[laptop] | FuriousGeorge: its not decriminalized either |
04:24.55 | file[laptop] | blitz[laptop] is an expert on this stuff |
04:25.01 | ManxPower | FuriousGeorge, Canada can't legalize pot because of international treaties, they CAN simply not enforce the laws that make it illegal. |
04:25.09 | FuriousGeorge | blitz[laptop]: is it even encouraged? |
04:25.15 | blitz[laptop] | FuriousGeorge: I'd not say encouraged :) |
04:25.22 | ManxPower | Which is annoying, because when the political climate changes they can start enforcing those laws again. |
04:25.27 | FuriousGeorge | or is its illegality merely a suggestion |
04:25.27 | blitz[laptop] | but no one really seems to care... depending on who you hang out with |
04:25.29 | Juggie | you will not go to jail for smoking pot |
04:25.38 | Juggie | selling 50,000$ worth |
04:25.39 | Juggie | yes |
04:25.47 | blitz[laptop] | you go to jail for dealing... not possession |
04:26.02 | Juggie | its not really, illegal to smoke it, but its illegal to buy it |
04:26.04 | Juggie | go figure :) |
04:26.07 | blitz[laptop] | technically you COULD go to jail for possession... but no cop is going to both you unless you're smoking it on the street and blow it in his face |
04:26.07 | FuriousGeorge | so i was installing ntp-client today and noticed its been on version 4.2.0 for 18 months |
04:26.26 | hypa7ia | blitz[laptop]: even then, you'd have to get spittle on him while blowing it |
04:26.30 | hypa7ia | -_^ |
04:26.31 | blitz[laptop] | lets just say its low priority for cops unless you own a grow house or something |
04:26.32 | hypa7ia | that sounded bad |
04:26.33 | FuriousGeorge | coincidence? I think not |
04:26.45 | blitz[laptop] | lol |
04:26.55 | blitz[laptop] | I filed bug 420 in the bug tracker :) |
04:27.03 | hypa7ia | haha awesome |
04:27.03 | cripito | sheez i am lost here.. pot and ntp-client don't mix well |
04:27.08 | FuriousGeorge | blitz[laptop]: stoner |
04:27.14 | blitz[laptop] | FuriousGeorge: I've never smoked it |
04:27.22 | blitz[laptop] | I just know lots of people who do |
04:27.33 | FuriousGeorge | blitz[laptop]: not that you remember anyway |
04:27.39 | blitz[laptop] | well... i tried it once, but I didn't inhale |
04:27.42 | Juggie | theres alot of other things cops are worried about |
04:27.44 | Juggie | besides pot |
04:27.46 | blitz[laptop] | smells too much like skunk |
04:27.49 | FuriousGeorge | blitz[laptop]: brownies? |
04:28.03 | blitz[laptop] | yah... brownies rock |
04:28.12 | blitz[laptop] | never had special brownies though |
04:28.16 | Juggie | hah |
04:28.23 | FuriousGeorge | and they can't be inhaled. trust me, ive tried |
04:28.24 | Juggie | blitz[laptop], only smokes pole. |
04:28.33 | Juggie | :P |
04:28.35 | blitz[laptop] | Juggie: ass |
04:28.35 | ManxPower | Any sane person knows how stupid the drug laws are and any sane person knows the "War on (Some) Drugs" has been a miserable failure. |
04:28.36 | blitz[laptop] | :) |
04:28.55 | blitz[laptop] | ManxPower: the war on drugs keeps people employed... and helps keep people in fear |
04:28.59 | FuriousGeorge | pots a gateway drug so blitz[laptop] went straight to crank |
04:29.08 | blitz[laptop] | hehehe |
04:29.09 | ManxPower | blitz[laptop], In that respect it's a success. |
04:29.29 | Juggie | ManxPower, what ammuses me more is 'the war on terror' |
04:29.42 | blitz[laptop] | "the war on f00" |
04:29.46 | ManxPower | Juggie, that just scares me. |
04:29.47 | Juggie | canada is all up in arms about a gov scandal which fucked around with 200million |
04:29.51 | blitz[laptop] | substitute f00 for whatever you want |
04:29.56 | Juggie | yet, bush blows TRILLIONS on iraq |
04:30.13 | blitz[laptop] | Juggie: that doesn't make the 200 million right though :) |
04:30.24 | Juggie | blitz[laptop], no, but it puts things in perspective |
04:30.29 | blitz[laptop] | it can |
04:30.37 | Juggie | we are cool up here ;P |
04:30.45 | blitz[laptop] | I'm a liberal though... and I'll continue to vote liberal |
04:30.54 | blitz[laptop] | Juggie: damn straight -- we rock :) |
04:30.55 | blitz[laptop] | lol |
04:30.56 | Juggie | conservative ;) |
04:30.59 | Juggie | but i don tvote |
04:31.01 | Juggie | *vote |
04:31.04 | blitz[laptop] | Juggie: and I thought we could be friend |
04:31.07 | blitz[laptop] | s* |
04:31.11 | blitz[laptop] | Juggie: good -- keep it that way |
04:31.14 | cripito | AltiServ from altigen... did anyone connects that to asterisk |
04:31.25 | Juggie | theres no point |
04:31.34 | Juggie | goverment is fucked no matter who they are |
04:31.38 | hypa7ia | true :) |
04:31.48 | cripito | :))))) |
04:31.48 | Juggie | as long as its not the bloc, its fine by me :P |
04:31.52 | hypa7ia | haha yes |
04:32.05 | cripito | i can't vote .. :P so anyway.... |
04:32.12 | hypa7ia | even then... if they want to go that bad, they're the ones who fuck themselves over |
04:32.14 | Juggie | hypa7ia, i finally got a pika board working in * |
04:32.28 | Juggie | it works ok i guess, for analog |
04:32.34 | hypa7ia | cool |
04:32.35 | Juggie | i'm used to pri now, so i'm spoiled. |
04:32.42 | hypa7ia | hehe |
04:32.54 | cripito | any driver accesible for a dialogic card? |
04:32.56 | Juggie | i like instant gratification ;) |
04:33.06 | hypa7ia | i may do a key system -> asterisk -> voip termination setup for my dad's law firm |
04:33.06 | Juggie | cripito, yes, if you buy * business edition |
04:33.12 | cripito | naaaaaaaaa |
04:33.14 | Juggie | it contains a dialogic driver |
04:33.19 | cripito | snif snif |
04:33.24 | Juggie | and this is due to intel licensing requirements |
04:33.32 | cripito | 995 bucks only b/c a driver? |
04:33.39 | ManxPower | You can, in theory, but the Dialogic driver from Digium |
04:33.39 | cripito | i need to find another wya |
04:33.41 | cripito | way |
04:33.42 | digime | most stable veresion of asterisk now? 1.0.9??? |
04:33.44 | ManxPower | $10/channel |
04:33.51 | Juggie | yah, you can try and buy it without the business eddition |
04:34.00 | Juggie | i might try and buy an unlimited license |
04:34.06 | Juggie | if they wil sell me one |
04:34.09 | digime | what version of * are people running in production? |
04:34.14 | blitz[laptop] | hypa7ia: I'm actually a socialist -- but unless I know the NDP can win my area, then I vote liberal |
04:34.16 | cripito | so a quad card apart of the 7K from intel is more or less 960 extras |
04:34.23 | blitz[laptop] | as long the conservatives don't win, thats all that I really care about |
04:34.26 | Vco | anything but harper |
04:34.27 | blitz[laptop] | but I'd like the NDP to win. |
04:34.31 | hypa7ia | blitz[laptop]: okay, i can like you again :-p |
04:34.36 | iCEBrkr | Well, shit, I can't dial out of Voicepulse. |
04:34.43 | blitz[laptop] | hypa7ia: haha... oh yah... I'm totally a socialist :) |
04:34.47 | Juggie | cripito, i dont understand the question |
04:34.50 | hypa7ia | i've been a dipper since i was like 16 |
04:35.02 | hypa7ia | worked on like 5 campaigns |
04:35.03 | Vco | o.O |
04:35.04 | cripito | was a comment more than anything juggy |
04:35.20 | Juggie | hypa7ia, i'm glad i know your a girl |
04:35.32 | Juggie | because if a guy had said "i've been a dipper since i was 16" |
04:35.35 | Juggie | i'd be concerned :) |
04:35.41 | blitz[laptop] | LOL |
04:35.43 | hypa7ia | hehe actually i'm going to try and leverage my political connections to get another user group some space |
04:35.53 | hypa7ia | i'm not sure what else "dipper" means |
04:35.58 | hypa7ia | don't think i wanna know :-p |
04:36.04 | iCEBrkr | Nov 3 23:35:54 WARNING[11951]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! |
04:36.06 | Juggie | :P |
04:36.08 | iCEBrkr | WHAT THE MOTHERFUCK |
04:36.09 | Juggie | bum dipper :P |
04:36.12 | blitz[laptop] | hypa7ia: I'm sure you could figure it out if you really wanted to :) |
04:36.35 | hypa7ia | hahaha |
04:36.39 | blitz[laptop] | hypa7ia: the one thing I like about the minority gov't is that it gives the NDP power |
04:36.44 | Vco | anyone use DID's from Voxbone ? |
04:36.51 | blitz[laptop] | hypa7ia: so the NDP actually gets shit done |
04:36.51 | Vco | still a terrible name if you ask me.... |
04:36.53 | hypa7ia | blitz[laptop]: me too :) |
04:37.00 | Juggie | blitz[laptop], did you hear layton's latest thing |
04:37.07 | blitz[laptop] | Juggie: no.. what did he say? |
04:37.14 | Juggie | he wants some shit done to health care |
04:37.15 | blitz[laptop] | wicked :) |
04:37.26 | Vco | he was actuallly pretty snappy on that |
04:37.30 | blitz[laptop] | yah, he wants a law put in that forbids privitazation of health care |
04:37.31 | Vco | totally out of character |
04:37.40 | blitz[laptop] | if I understand correctly |
04:37.40 | hypa7ia | oh man speaking of |
04:37.41 | Juggie | if the liberals dont comply, he'll partner with conservative to topple the liberal minority |
04:37.51 | blitz[laptop] | oh yah... he has a lot of power right now |
04:37.52 | Vco | well..not 'snappy' , just super asseritve |
04:37.52 | Juggie | which should really be done anyway |
04:37.54 | Vco | haha |
04:37.54 | Juggie | to clean house |
04:37.55 | hypa7ia | now that is strong-arming :) |
04:38.03 | hypa7ia | rox0rs |
04:38.09 | cripito | chan_ooh323 is version h323v1 or h323v2 ? |
04:38.12 | blitz[laptop] | Juggie: guh.... the conservatives scare the hell outta me |
04:38.29 | hypa7ia | my parents' friend went to a private clinic in montreal for some experimental back surgery... no waiting lists |
04:38.33 | hypa7ia | makes me sad :( |
04:38.45 | blitz[laptop] | yah... its too bad people with money can do whatever they want |
04:38.46 | Vco | remember when that liberal dude tossed him that insult when he was at parliament.. |
04:39.00 | Vco | basically that he might be the leader of the party , but he doens't even have a seat there yet.. |
04:39.46 | blitz[laptop] | private clinics should be taxed heavily and that money put DIRECTLY back into public health care |
04:40.25 | hypa7ia | yeah |
04:40.29 | Vco | just like tobacco tax should ACTUALLY go back and (i think) go to smoking cessation stuff |
04:40.37 | hypa7ia | or we should just fix the public system |
04:40.43 | hypa7ia | and/or that is :) |
04:41.03 | hypa7ia | instead of spending $18 jillion on cisco callmanagers at the hospitals |
04:41.08 | Vco | heh..heh.. |
04:41.08 | blitz[laptop] | duh |
04:41.09 | hypa7ia | which is what they're doing :( |
04:41.19 | hypa7ia | asterisk needs to talk to nurse-call systems |
04:41.30 | iCEBrkr | This g729 shit has been pissing me off. I have YET to make the shit work and I bought a second license just to make sure I had all my bases covered. |
04:41.33 | Vco | nevermind the ludicrous money they spent at UBC and VGH just a couple of years back |
04:41.51 | file[laptop] | I think I'm going to go to sleep |
04:41.55 | Vco | doign deployment work, they bought a pile HP 4050n |
04:41.55 | hypa7ia | Vco: university health system in toronto was $5M of cisco gear |
04:42.00 | blitz[laptop] | I think I'm going to visit the garage |
04:42.02 | blitz[laptop] | then go to sleep :) |
04:42.09 | hypa7ia | the "garage" |
04:42.17 | Vco | there was 1 $3000 laser printer for every 2 patients in the hospital |
04:42.24 | hypa7ia | cripes. |
04:42.28 | blitz[laptop] | hypa7ia: I'm actually going out to the garage :) |
04:42.32 | Vco | evern the nurses were complaing they didn'tneed these things |
04:42.33 | blitz[laptop] | "what do you call it?" |
04:42.39 | blitz[laptop] | "a car hold" |
04:42.43 | Vco | hahaha |
04:43.25 | hypa7ia | carport |
04:43.39 | Juggie | hah |
04:43.40 | hypa7ia | Vco: the cisco wifi stuff actually gets well used |
04:43.43 | Juggie | reminds me of the simpsons |
04:43.52 | Juggie | hypa7ia, sure i have 1.5 million sitting on a lab floor |
04:43.59 | cripito | yes Altigen |
04:44.00 | blitz[laptop] | Juggie: uhhh... yah... thats kind of what I was trying to imply :) |
04:44.03 | Juggie | i have a 24port poe switch on my desk :) |
04:44.10 | Juggie | so i can power my phones :) |
04:44.12 | Vco | i have a dead hooker on my desk... |
04:44.23 | Vco | anyone have a van and some carpet they want to get rid of? |
04:44.30 | Juggie | Vco, your approaching it all wrong, what you want is a live hooker under your desk |
04:44.40 | Vco | well.. |
04:44.42 | cripito | :))))))))))))))))))))))))))))))))))))) |
04:44.47 | Vco | i've figurd that out now |
04:44.51 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
04:44.59 | cripito | ok altigen.... and asterisk |
04:45.06 | cripito | i can call to altigen |
04:45.13 | cripito | but those F*&*#$@#$@#)($)@#$ can't call to me |
04:45.27 | Qwell | So, sccp is actually kinda nice |
04:45.28 | Vco | is this with ooh323? |
04:45.33 | Vco | cripto |
04:45.37 | Vco | cripito |
04:45.37 | cripito | yeap |
04:45.39 | Vco | rather |
04:46.10 | cripito | i am thinking to install h323 and openh323 just to see changes anything |
04:46.10 | Vco | call to your * box, or to a 323 client? |
04:46.11 | cripito | but i doubt it |
04:46.28 | *** join/#asterisk XTR (n=xtr@staff-nat.netnation.com) |
04:46.37 | cripito | i can call from my sip phones in * to the phones in altigen |
04:46.43 | cripito | but altigen can't send a call to me |
04:46.56 | Pazzo | Hi guys! I've had a self-packaged 1.2.0-Beta1 working fine (together with ser, mediaproxy etc). With 1.2.0-Beta2 (package from debian experimental) I run into trouble: everything seems to work fine, but I cannot hear any sound produced by asterisk!? (echotest etc) |
04:47.06 | Pazzo | any idea? similar problems? |
04:47.07 | Vco | yea, i can't ring any h323 clients but they can call out fine.. |
04:47.08 | cripito | 23:38:06:774 ERROR:Incompatible audio capabilities - Can not open audio channel |
04:47.08 | cripito | <PROTECTED> |
04:47.18 | Vco | goes straight to unavailable. |
04:47.26 | Vco | eww....totally different issue than me |
04:47.28 | Pazzo | btw: no errors, even with -rvvvvvv |
04:48.15 | cripito | so vco they can call your *? |
04:48.25 | cripito | can we check versions , settings etc? |
04:48.28 | Vco | ok, so not a lot of feedback for people that use voxbone, anyone knwo where else to find Japan DID's? |
04:48.57 | cripito | there was something from didx.org offering that... |
04:49.10 | Vco | didx bough the japan ones from voxbone |
04:49.13 | Vco | :/ |
04:49.15 | *** join/#asterisk nobell (n=jdegraff@70.103.228.158) |
04:49.19 | cripito | but didx don't have a precisally good critic |
04:49.26 | Vco | the h323 clients can call my * box, or route out over IAX etcetc.... |
04:50.06 | cripito | yeap man.. that's exactly what i need.. but if i connect the phone h323 to my server words fine... |
04:50.14 | Vco | but i can't Dial(H323/soemuser) |
04:50.14 | cripito | magic... |
04:50.27 | cripito | using chan_ooh323? |
04:50.30 | Vco | ya |
04:50.40 | Vco | "oo"h323 |
04:50.44 | cripito | did u try dial(OOH323/soeuser)? |
04:50.49 | Vco | yea.. |
04:50.53 | Vco | channel unavailable. |
04:50.55 | iCEBrkr | make my g729 go!!! |
04:51.05 | Vco | show channeltypes |
04:51.11 | Vco | lists it as H323 |
04:52.10 | Pazzo | hmmm... no one running 1.2.0 beta2?? |
04:52.15 | cripito | Feature Feature Proxy Channel Driver no yes no |
04:52.15 | cripito | OOH323 Objective Systems H323 Channel no yes no |
04:52.31 | Vco | this japan DID i still haven't figured out if it's potentianlly something on my end or what....calls come in, rings a sip phone, i answer the phne i hear busy ..but the calling party is still ringing.. |
04:52.46 | Vco | hang up phone, rings right away again.. |
04:52.53 | Vco | answer, it's busy.. |
04:53.27 | Vco | thing it , it used to work, anbd i have other DID's with the same config and it all works fine.. |
04:53.49 | Vco | i'm jsut the provider for the .jp one is fubar |
04:54.00 | Vco | i'm just guessting that the.... |
04:54.06 | Vco | guessing even... |
04:54.09 | Vco | damn |
04:54.24 | Vco | i need to start using keyboards with the same layout.. |
04:54.36 | *** join/#asterisk kvit (n=kvit@203.209.56.65) |
05:00.17 | *** join/#asterisk Baph (n=Dave@dirobertson.plus.com) |
05:02.51 | *** part/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET) |
05:03.04 | iCEBrkr | *CLI> Nov 4 00:02:43 NOTICE[12528]: chan_iax2.c:7746 iax2_poke_noanswer: Peer 'vpconnect-t01' is now UNREACHABLE! Time: 0 |
05:03.08 | iCEBrkr | ANything else?! |
05:03.24 | Vco | Time: 0 ? |
05:03.50 | iCEBrkr | I've never had issues with VP like this. |
05:03.57 | iCEBrkr | Unless there were hurricanes :P |
05:05.54 | iCEBrkr | voicepulse-in-0 66.234.228.170 (S) 255.255.255.255 4569 OK (49 ms) |
05:05.57 | iCEBrkr | vpconnect-t02 66.234.228.166 (S) 255.255.255.255 4569 OK (46 ms) |
05:06.00 | iCEBrkr | vpconnect-t01 66.234.228.160 (S) 255.255.255.255 4569 UNREACHABLE |
05:06.03 | iCEBrkr | I love it |
05:06.19 | *** join/#asterisk Eyecon (n=matt@jumala.plus.com) |
05:11.19 | *** part/#asterisk bancus (n=treed@athena.crschmidt.net) |
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05:27.52 | *** join/#asterisk infinity1 (i=brendon@solara.netcal.com) |
05:28.00 | infinity1 | err ..anyone use the TORTURE feature? |
05:28.16 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
05:29.19 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
05:29.33 | shmaltz | how does robbed bit t1 signal DID? |
05:31.43 | infinity1 | how does the TORTURE status for the Dial application work? |
05:33.46 | *** join/#asterisk t0p (i=t0p@202.8.86.162) |
05:38.39 | Corydon76-home | shmaltz: depends upon the signalling |
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06:07.16 | Qwell | headline: 78 year old woman talking on her cellphone while driving makes an illegal right turn at a red light, and gets hit by a bus. |
06:07.22 | Qwell | Anybody see anything wrong with this picture? |
06:07.39 | Qwell | (or, more specifically, 4 things?) |
06:08.34 | Netgeeks | hrm, 1. H in headline not capitalized |
06:08.59 | hypa7ia | wrong side of the road unless it's english |
06:09.01 | Netgeeks | but I don't think you wanted to know the 4 things wrong in your typing... |
06:09.50 | _m_ | hypa7ia: or australian (or some other leftdriving place) |
06:10.00 | Qwell | nah, US |
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06:33.26 | FuriousGeorge | i use eyebeam and tbh i dont really like it. i like how firefly loosk but afaik it can only handle one call at a time |
06:33.48 | FuriousGeorge | can someone recommend a softphone for me to try with the little things, like support of multiple connections and perhaps a xfer button |
06:37.58 | infinity1 | FuriousGeorge: i use eyebeam. seems okay. |
06:37.58 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
06:38.56 | infinity1 | its quite here lately |
06:39.06 | infinity1 | whats the deal? everyone busy deploying beta2? |
06:42.27 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
06:42.59 | lehel | mornin' |
06:44.30 | FuriousGeorge | infinity1: you know what bothers me about eyebeam? it uses that simple messaging and i want to eventually use XMPP for messaging, so it annoys me that the "send an instant message" option wont go away |
06:44.39 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
06:45.28 | *** join/#asterisk Splas (n=sp@brooklyn.paravolve.net) |
06:46.05 | FuriousGeorge | id love to get my hands on a copy of X-Pro |
06:47.06 | infinity1 | FuriousGeorge: url? |
06:47.51 | infinity1 | simple vs XMPP? duno ...i just use the sip functinoality. |
06:50.24 | FuriousGeorge | infinity1: there is astjab.org and there is asterisk-im from jivesoftware.com |
06:50.33 | FuriousGeorge | xmpp=jabberd |
06:50.55 | infinity1 | FuriousGeorge: i think i'm using X-Pro actually. you mean the commercial x-lite? |
06:51.19 | FuriousGeorge | infinity1: right, but eyebeam supports video (which could be cool) and messaging (which i cant disable) |
06:51.42 | Vco | plus g729 ..no? |
06:51.43 | FuriousGeorge | it just bothers me that even though i disable presence and dont implement it on the server it still gives the user the option |
06:52.26 | FuriousGeorge | g'night all |
06:52.33 | infinity1 | l8 |
06:53.13 | infinity1 | FuriousGeorge: it appears the version of x-pro i'm using doesn't support messaging. i can't find an option fori t |
06:57.50 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
07:15.38 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
07:22.41 | hardwire | drumkilla: pinch poke |
07:26.36 | Netgeeks | I don't think he's alive, I tried him an hour ago |
07:26.53 | hardwire | mofo made a completely funky comment on one of my bug reports |
07:26.57 | *** join/#asterisk Poincare (n=jefffnod@dD577A88C.access.telenet.be) |
07:27.06 | hardwire | I strive to understand it.. but I don't think I am supposed to know its real meaning :) |
07:27.52 | Netgeeks | oooooh, which one! I want to read it! |
07:28.18 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
07:28.21 | hardwire | http://bugs.digium.com/view.php?id=5338 |
07:29.26 | Netgeeks | heh |
07:29.45 | Netgeeks | This option is not present in 1.0 |
07:29.46 | hardwire | what does that even mean? |
07:30.03 | hardwire | is it some sort of crazy statement I should spend the rest of my life studying? |
07:31.07 | Netgeeks | I think he's trying to say that the option 's' for chanIsAvailable doesn't exist in the 1.0 branch |
07:31.50 | hardwire | so I am not insane |
07:32.01 | Netgeeks | I know this because I slept in a Holiday Inn express last night |
07:32.08 | hardwire | heh |
07:32.20 | hardwire | I hate those commercials |
07:32.34 | Netgeeks | and, no, you are incorrect, we are all insane... |
07:33.08 | hardwire | so.. why he is posting that about a 1.2.x fixup? |
07:33.19 | hardwire | if s doesn't exist in 1.0.x whats the whoop? |
07:33.22 | Netgeeks | I'm beating myself to pieces trying to figure out why I'm getting this: |
07:33.27 | Netgeeks | Nov 3 19:13:34 WARNING[7850]: channel.c:713 channel_find_locked: Avoided deadlock for '0x6000000000154840', 10 retries! |
07:33.33 | Netgeeks | and it's got me very insane |
07:34.00 | Netgeeks | I dunno what the whoop is... maybe whardier or someone asked him about it on irc and he added a note to the patch so they would read it |
07:34.22 | hardwire | whardier == hardwire |
07:34.46 | Netgeeks | hrm, well now, I guess you would know if you asked in irc then |
07:34.51 | hardwire | possibly |
07:34.55 | hardwire | ok I am good now |
07:34.59 | hardwire | its meant to be confusing |
07:35.01 | Netgeeks | unless you are of the multiple personalities type of insane |
07:35.07 | hardwire | yeh. no |
07:35.10 | Netgeeks | then you may not know if you asked |
07:35.12 | Qwell | hardwire: It's because twisted set it to Pending Stable |
07:35.18 | Netgeeks | unless you talk to your selves |
07:35.32 | Qwell | So, drumkilla says "Not in 1.0. Status => Closed." |
07:35.34 | hardwire | Qwell: so pending 1.2.0 stable has to address 1.0 stable changes? |
07:36.31 | Qwell | was just a simple mistake... drumkilla wanted to clarify that when he closed it |
07:37.00 | hardwire | it was good enough to cause me a wonderin. |
07:37.02 | hardwire | so.. oh well |
07:37.15 | kuku5 | Anyone here storing credit card numbers? |
07:37.30 | hardwire | if they are they should be shot :0 |
07:39.11 | Netgeeks | I store mine in my wallet on pieces of plastic |
07:39.30 | kuku5 | :) |
07:39.33 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
07:39.42 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
07:39.48 | Netgeeks | I doubt thats what you were looking for, however :) |
07:40.15 | Netgeeks | if you find my locked channel, let me know, hardwire |
07:42.26 | *** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-50.cst.dccnet.com) |
07:42.46 | kb1_kanobe | evening all. |
07:44.53 | steff | hi all |
07:47.50 | *** join/#asterisk bon (n=bon@217.172.148.5) |
07:47.50 | bon | hey |
07:47.55 | bon | anyone using realtime aroung? |
07:47.56 | bon | around |
07:47.57 | bon | even |
07:47.59 | bon | :) |
07:48.08 | bon | Unable to load module cdr_addon_mysql.so |
07:48.17 | bon | Nov 4 08:47:43 WARNING[24690]: loader.c:314 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directory |
07:48.20 | bon | :( |
07:48.29 | bon | rerun ldconfig, mysql lib dir in ld.so.conf |
07:48.35 | bon | don't know what else to do |
07:49.59 | Qwell | Did you change mysql versions or something? |
07:50.09 | Qwell | looks like it's linked against 4.1.15 |
07:52.21 | Qwell | off to bed |
07:52.37 | *** join/#asterisk grimse (n=grimse@p5481DDC5.dip.t-dialin.net) |
07:52.52 | bon | Qwell: well fresh install mysql5 |
07:53.07 | bon | there has been no mysql4 before on that machine |
07:54.57 | steff | somebody using AVM C2 controller or other dual bri card? |
07:56.56 | steff | anyone? |
07:57.08 | kb1_kanobe | Behaviours you don't want to see at 11:30pm - playing a freshly recorded wav file oopses the Asterisk server... :-/ |
07:57.54 | Dr_Ray | <PROTECTED> |
07:58.20 | kb1_kanobe | That's true, but like every third person I meet in here at this time of night, I just want to go to bed... :-) |
08:00.28 | Dr_Ray | I'm one of the other 2 :) |
08:00.45 | Dr_Ray | I hate having to wait for calls to stop before I can reload the asterisk server |
08:01.12 | kb1_kanobe | Agreed. |
08:02.05 | kb1_kanobe | Alas, my issue appears to have been filesystem corruption, not asterisk... and now I have a mess on my hands. |
08:04.14 | *** join/#asterisk kiko69 (n=Keith@kauai.sys.pas.earthlink.net) |
08:05.05 | *** join/#asterisk tryfoss (n=morten@80.239.93.22) |
08:10.19 | steff | so... anyone use chan_capi ? |
08:11.44 | steff | nobody :-( |
08:11.49 | lehel | steff: me |
08:11.58 | steff | :-D |
08:12.28 | lehel | ? ;) |
08:12.37 | steff | lehel: my * box work with only 2 bchan, i can't get the 4 bchan working |
08:13.17 | lehel | me neither steff |
08:15.06 | zoa | steff |
08:15.11 | zoa | im using a c4 |
08:15.23 | steff | (i'm french, my english is poorly), i need some time to understood |
08:15.31 | steff | i have a C2 |
08:15.42 | zoa | ah mais je parle aussi le francais |
08:15.44 | zoa | pas de probleme |
08:15.52 | steff | super |
08:15.55 | skyen | mois nonplus |
08:15.57 | skyen | ;) |
08:16.02 | zoa | la meilleure chose a prendre, c'est chan_misdn pour l'instant |
08:16.24 | zoa | ou bristuff (si les drivers marchent pour votre carte) |
08:17.21 | steff | je vaistester avec chan_misdn, |
08:17.44 | steff | cela proviens de chan_capi le blocage a 2 bchan alors? |
08:17.46 | skyen | what does vaistester mean? |
08:17.56 | skyen | i'm getting the rest |
08:17.56 | zoa | oui je pense |
08:18.10 | zoa | its "je vais tester" |
08:18.12 | zoa | i will test |
08:18.20 | steff | tring to test -je vais tester) |
08:20.32 | steff | there was a version of chan_misdn know to work with avm ? |
08:21.07 | steff | i try 0.2.0 |
08:21.52 | zoa | yes |
08:21.54 | zoa | it works |
08:22.09 | zoa | take chan_misdn from asteriskbeta2 |
08:22.11 | zoa | its included |
08:22.14 | zoa | no need for patches |
08:30.07 | *** join/#asterisk stoffell (n=stoffell@59.46-201-80.adsl.skynet.be) |
08:30.08 | kb1_kanobe | any sox wizards around? I need to convert from prerecorded gsm files to asterisk-compatible 'wav' files. |
08:30.36 | kb1_kanobe | plain old 'sox ...gsm ...wav' results in 'not a wav49' in the console. |
08:31.01 | kb1_kanobe | But I was expecting * to be looking for a 16bit linear, not a wav49? |
08:32.15 | *** join/#asterisk BleedingMe (n=Bleeding@adsl-69-227-208-235.dsl.scrm01.pacbell.net) |
08:32.34 | kb1_kanobe | huh - asterisk-users is showing up in Google Groups. |
08:33.29 | lehel | long time kb1_kanobe |
08:33.42 | kb1_kanobe | I'm just a little behind the curve is all. :-) |
08:33.50 | lehel | :P |
08:34.16 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:34.22 | *** join/#asterisk vinko (n=zic@208.5.87.254) |
08:34.26 | lehel | kb1_kanobe, u've dealing with fax? |
08:34.43 | kb1_kanobe | not really, I do some detect and passthrough. |
08:36.09 | af_ | what dows the "n" priotity in the cvs asterisk? |
08:36.14 | af_ | does, even :) |
08:36.49 | *** join/#asterisk vinko (n=zic@208.5.87.254) |
08:37.14 | Dr_Ray | it's N for next, it replaces 1,2,3,4 in the extensions.conf |
08:37.37 | af_ | oh I see: like go to the next statement in this extension? |
08:37.49 | skyen | what about the +101-feature? |
08:38.18 | Dr_Ray | I think it is more for picking oreder |
08:38.23 | Dr_Ray | not teh goto |
08:38.26 | Dr_Ray | but maybe |
08:38.30 | *** join/#asterisk many (n=many@daheim.ukeer.de) |
08:38.48 | *** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
08:41.09 | tryfoss | is it normal that the cdr from a call going out from a queue to an agent is logged with billsec=0 ? |
08:54.21 | *** join/#asterisk shido6 (i=shido6@d221-68-216.commercial.cgocable.net) |
08:57.35 | *** join/#asterisk shanky (n=shanky@82.159.214.9) |
08:58.25 | shanky | good morning |
09:05.57 | pooh_ | morning |
09:06.39 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
09:10.50 | *** join/#asterisk Ariek1 (n=ariek@84-245-28-221.dsl.cambrium.nl) |
09:17.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
09:20.41 | lehel | phonenr=faxnr, could anyone look at my CLI, why am i unable to send fax? |
09:20.43 | lehel | http://pastebin.ca/27580 |
09:21.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
09:23.26 | pooh_ | lehel: Sending 58740 on 2/3 to ipaddress:4569 |
09:23.33 | pooh_ | lehel: where is the REAL address ? |
09:29.58 | Delvar | quick question do digium g729 licances expire after a period of time/ |
09:30.31 | zoa | no they dont |
09:31.33 | shido6 | :) |
09:33.28 | kb1_kanobe | Time to refrest the asterisk/festival interface - http://cslu.cse.ogi.edu/tts/flinger/ |
09:34.39 | kb1_kanobe | s/refrest/refresh |
09:34.57 | mazzanet | is it possible to play say an mp3 down a current open sip channel? |
09:36.00 | lehel | pooh_: the real address? i replaced the address with "ipaddress" |
09:37.13 | Delvar | col thanks |
09:38.12 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
09:39.18 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.171.106) |
09:39.24 | Delvar | weird this box, been working fine for months, now g729 doesnt seem to work at all |
09:39.46 | MuppetMaster | Helllo |
09:39.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
09:40.11 | MuppetMaster | I have just started receiving a new error after doing some upgrades to v1.2beta2: http://pastebin.ca/27581 |
09:40.19 | MuppetMaster | Anyone else seen this, know how to resolve? |
09:41.06 | Dr_Ray | which codec? |
09:41.33 | MuppetMaster | G711 beteween an Asterisk and a Sipura |
09:42.36 | shanky | anyone know the pinout schema for a E1 connector RJ45-BNC ? (I'm trying to connect a TE410P to BNC males connectors) |
09:42.43 | shido6 | i used grubinstall |
09:42.46 | shido6 | oops |
09:43.18 | Ariek1 | does anybody know some info about the mixmanager |
09:43.26 | Ariek1 | or where i can find it |
09:49.52 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
09:50.29 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
09:50.45 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-169-66.44-151.net24.it) |
09:51.28 | BladeRunner05 | Hi all anyone has used Controller ISDN AVM B1 v.4.0 |
09:55.23 | lehel | Nov 4 12:56:26 DEBUG[25577] chan_zap.c: Monitor doohicky got event Event 160 on channel 2 |
09:55.38 | lehel | <doohicky ? what this supposed to mean? |
09:55.44 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
09:55.59 | lehel | BladeRunner05, i'm using fritz |
09:57.30 | *** join/#asterisk ful|work (n=fulgas@209.8.233.67) |
09:57.43 | *** join/#asterisk jojo (n=jojo@c83-253-39-20.bredband.comhem.se) |
09:57.52 | ful|work | hey |
09:58.47 | BladeRunner05 | lehel: the avm b1 card ? |
10:05.16 | *** join/#asterisk Red5 (n=Red5@sodor5.ethertech.com.au) |
10:07.26 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
10:08.49 | BladeRunner05 | lehel: r u there ? |
10:09.39 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
10:10.07 | lehel | BladeRunner05: AVM FRITZ!Card PCI < this one |
10:12.09 | Red5 | hey all.. quick question. how can i place a direct internet call from my sipura to a friends phone via IP address ? |
10:12.47 | Red5 | its not possible is it ? unless he's registered with someone.. |
10:15.01 | RaYmAn-Bx | it is possible :) Assuming he has a SIP soft or hardware phone listening on that ip address...(check your sipura manual on how to dial ip's directly) |
10:15.11 | RaYmAn-Bx | obviously you can't dial his normal phoneline by ip address |
10:15.46 | Red5 | i understand that... but if my sipura is registered with asterisk dont i need a dialplan for ip based calls too ? |
10:16.03 | BladeRunner05 | Lehel: which kernel r using ? |
10:16.51 | RaYmAn-Bx | Red5: it depends..Sipura can (if setup correctly) dial directly to an ip address..So it simply just doesn't let the call setup go through the register (=asterisk) |
10:17.16 | Red5 | what about a 7940 ? |
10:17.44 | RaYmAn-Bx | never used that so I have no idea..I'd assume it would be possible though..Check documentation :) |
10:17.50 | Red5 | ok.. ta.. |
10:19.53 | *** join/#asterisk RGi_- (i=RGi@62.97.247.44) |
10:20.33 | RGi_- | anyone here using asterisk@home ? |
10:20.49 | shido6 | used to |
10:20.51 | shido6 | :) |
10:21.01 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
10:21.30 | RGi_- | I have problem making Digital Receptionist recordings... getting an error when it tries to copy the wave file... |
10:22.39 | puzzled | anyone seen this error when compiling ztdummy and know how to solve it? error: #error ztdummy requires 1000 hz jiffies |
10:23.38 | lehel | puzzled: my ztdummy works fine now;) and not seen that err |
10:24.03 | puzzled | good for you :) |
10:28.44 | heroine | hi |
10:35.59 | lehel | ahh, i can't manage to send a fax;( |
10:37.57 | ful|work | RGi_-: check your permissions... |
10:39.30 | nfi|ermes | hi all |
10:39.47 | nfi|ermes | anyone never used bristuff-CVS ? |
10:40.21 | *** part/#asterisk shanky (n=shanky@82.159.214.9) |
10:42.08 | RoyK | nfi|ermes: yep. works for me (tm) |
10:42.16 | MuppetMaster | Is Asterisk Thread Safe? |
10:42.37 | MuppetMaster | So, for example, if I am using a SetGlobalVar to increment a counter, what happens if two calls come in at the same time and try to increment that counter? |
10:42.44 | nfi|ermes | RoyK, havve you hfc based isdn caards ? |
10:43.03 | RoyK | nfi|ermes: one hfc-pci. just a sec, and I'll paste the lspci entry |
10:43.15 | RoyK | 0000:03:05.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
10:43.21 | nfi|ermes | ok |
10:43.29 | RoyK | works nicely |
10:43.30 | nfi|ermes | i have some problem to compile zaptel |
10:45.16 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
10:48.52 | nfi|ermes | RoyK, have u used bristuf patches or other patches ? |
10:49.11 | RoyK | nfi|ermes: no. just head |
10:51.31 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
10:51.55 | zoa | royk ? |
10:51.56 | zoa | you here ? |
10:57.41 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
10:59.08 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
11:00.29 | *** join/#asterisk sambal (n=sambal@213.148.236.189) |
11:01.15 | *** join/#asterisk johnm (n=johnm@gentoo/developer/johnm) |
11:06.37 | *** join/#asterisk JmGV (n=jgomez@217.11.102.178) |
11:09.34 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
11:09.58 | *** join/#asterisk bon (n=bon@217.172.148.5) |
11:10.10 | bon | any res_mysql.so user around please? |
11:17.16 | InfraRed | that's not in standard asterisk compile |
11:19.06 | BladeRunner05 | lehel: can u tell me wich version of kernel you are running on ? |
11:19.43 | bon | InfraRed: sure i know |
11:19.53 | bon | but is there another way to have data stored in mysql? |
11:20.14 | InfraRed | yes! |
11:20.24 | bon | that is? |
11:22.14 | lehel | BladeRunner05: 2.6.8-2-686 |
11:23.53 | BladeRunner05 | that kernel support your fritz card or u install the driver given by avm ? |
11:25.21 | InfraRed | cdr_addon_mysql.so |
11:27.06 | InfraRed | actually |
11:27.08 | InfraRed | ignore me |
11:27.18 | InfraRed | i just foundout what res_mysql does |
11:29.58 | lehel | BladeRunner05: i install the driver |
11:31.10 | bon | InfraRed: heh:) i have cdr_addon already |
11:31.13 | bon | that works flawlessly |
11:31.19 | bon | all the cdr is written to a mysql db |
11:31.30 | bon | but i need to have peers/users and sip.conf in a db as well |
11:31.33 | bon | can't figure out how |
11:43.35 | InfraRed | there is a page on voip-info |
11:43.43 | InfraRed | http://www.voip-info.org/wiki-Asterisk+RealTime |
11:43.57 | InfraRed | tells you the db scema |
11:45.27 | tryfoss | I wonder... is it normal that the cdr from a call going out from a queue to an agent is logged with billsec=0 ? (the cdr is written once the call is answered by the agent) |
11:47.27 | bon | InfraRed: i already have those dbs |
11:47.52 | bon | long long sip_buddies table |
11:52.36 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
11:53.22 | *** join/#asterisk potsboy (n=chrisg@196.34.241.242) |
12:00.48 | jalsot | hi |
12:01.00 | jalsot | does anybody use atxfer with success? |
12:01.56 | X-Rob | jalsot - put ',tT' on the end of your dial string. |
12:02.08 | X-Rob | both lower case and upper case. |
12:02.46 | jalsot | the transfer itself works, however I get zero-way audio |
12:03.25 | jalsot | sip1->sip2-atxfer->sip3, sip1 and sip3 are in connection, however both sides are quiet |
12:06.00 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
12:15.58 | jalsot | any idea? |
12:21.01 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
12:22.56 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
12:24.55 | *** join/#asterisk emdub (n=mike@graffiti.corp.ef.net) |
12:25.01 | emdub | anyone around that can answer a (hopefully) simple macro question? |
12:26.53 | emdub | for some reason a macro i have isn't falling over to the offset priority when a dial out via iax fails |
12:27.02 | emdub | not sure why |
12:28.24 | *** join/#asterisk neuwald (n=neuwald@weber.anpa.org.br) |
12:28.48 | neuwald | hi folks. does anybody knows where can I found some call-back examples? |
12:30.39 | *** join/#asterisk bmg505 (n=leon@rndf-146-60-157.telkomadsl.co.za) |
12:31.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
12:31.52 | *** join/#asterisk fugitivo (n=ajf@209.13.242.109) |
12:32.30 | lehel | neuwald: voip-info.. google [?] |
12:39.58 | DrukenHME | emdub: pastebin the macro... |
12:41.19 | emdub | pastebin? |
12:41.25 | DrukenHME | ~pastebin |
12:41.26 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
12:41.30 | emdub | oic |
12:42.42 | emdub | hmm for some reason i can't connect to that url from home |
12:42.56 | emdub | weird |
12:44.14 | emdub | ah here http://pastebin.ca/27589 |
12:44.48 | emdub | thats just the specific part, im trying to get the macro to use the other $TRUNK_IAX_VP2 if the first dial fails |
12:45.20 | emdub | but it never actually tries the other priorities... i tried doing a setvar and setting MACRO_OFFSET to 100 but that didnt work either |
12:46.46 | neuwald | lehel yes, but I only found things using AGI |
12:47.20 | *** join/#asterisk clive- (n=pirch@ndn-165-131-97.telkomadsl.co.za) |
12:47.36 | neuwald | let learn AGI... |
12:48.24 | DrukenHME | hehehehe |
12:48.36 | DrukenHME | embub it's +101 not +100.... |
12:48.51 | emdub | doh |
12:49.10 | emdub | do i need to setenv MACRO_OFFSET or does it default? |
12:49.18 | DrukenHME | also, you'll need a g in the dialstring |
12:49.37 | emdub | not sure i follow |
12:50.00 | DrukenHME | in the first dialstring, you need to have the dialplan continue even if it fails... |
12:50.01 | clive- | can someone help me with a silly newbie perl question...how do I include the words "user@account" in the dialstr= statement |
12:51.53 | BladeRunner05 | ....anyone has used Controller ISDN AVM B1 v.4.0 |
12:52.22 | DrukenHME | emdub : http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
12:53.46 | *** join/#asterisk Madounet (n=Madounet@AMontpellier-206-1-2-97.w80-11.abo.wanadoo.fr) |
12:58.09 | emdub | hmm, seems as though adding the g option just makes it go back to the context that included it... i also made the priorities offset by 101 and that didn't fix it either... i updated the pastebin --- http://pastebin.ca/27590 (not sure why pastebin made a new url heh) |
12:59.16 | DrukenHME | emdub: got the cli output? |
12:59.23 | emdub | yeah |
12:59.25 | emdub | one second |
13:00.26 | *** join/#asterisk newbien (n=e@119.241.33.65.cfl.res.rr.com) |
13:02.01 | newbien | hi, need info on setting up asterisk server for linux user group to provide free outbout calls to members |
13:04.05 | fugitivo | ~docs |
13:04.06 | jbot | extra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
13:04.18 | emdub | http://pastebin.ca/27595 -- drukenhmw |
13:04.24 | emdub | drukenhme, rather |
13:06.12 | newbien | know of any linux user groups that have an asterisk server providing outbound calls to lug members? |
13:07.36 | DrukenHME | hmm.... looks like the first dial (to voicepulse) isn't erroring... |
13:07.57 | emdub | something i've done or? |
13:08.28 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
13:08.48 | DrukenHME | WARNING: chan_iax2.c:1473 attempt_transmit: Max retries exceeded to host 66.234.228.160 on IAX2/66.234.228.160:4569/1 (type = 6, subclass = 1, ts=12, seqno=0) |
13:09.05 | emdub | nod |
13:09.22 | emdub | i assumed that was just a warning msg that was normal if the other side was pretty much dead :) |
13:09.48 | DrukenHME | i guess you don't have voicepulse watched? |
13:10.19 | DrukenHME | qualify= ? |
13:10.24 | emdub | not sure i understand what you mean by "watched" |
13:10.57 | lesouvage | I'm trying to write my first agi script, reading the first 10 characters stored in a file and return it to asterisk. Can somebody please help me or point me to some examples of python agi scripts on the internet. |
13:11.22 | emdub | druken - that looks to be for sip specifically |
13:11.25 | emdub | i connect to vp via iax |
13:11.57 | DrukenHME | well, all my iax connections have it... so i would assume it's not sip only :) |
13:12.48 | emdub | ah i see let me give it a whirl |
13:16.47 | lesouvage | I have the agi script that isn't working yet pasted to http://pastebin.com/417099 . Any help will be apriciated. |
13:17.02 | *** join/#asterisk Corndawg_ (i=whoisit@c-67-191-22-79.hsd1.fl.comcast.net) |
13:17.19 | *** join/#asterisk Astinus (i=iBook@freenode/staff/gentoo.astinus) |
13:20.11 | emdub | yeah qualify doesnt seem to work |
13:20.13 | emdub | bummer |
13:21.19 | ManxPower | lesouvage, check the docs directory in the asterisk source code. |
13:23.44 | *** join/#asterisk lilwookie (n=root@modemcable005.89-81-70.mc.videotron.ca) |
13:27.34 | synthetiq | is there way to limit asterisk log files? |
13:28.25 | ManxPower | synthetiq, Not really. Have the system rotate the log files, then issue an asterisk -rv "logger reload" |
13:28.37 | *** part/#asterisk clive- (n=pirch@ndn-165-131-97.telkomadsl.co.za) |
13:28.40 | ManxPower | sorry |
13:28.43 | ManxPower | synthetiq, Not really. Have the system rotate the log files, then issue an asterisk -rx "logger reload" |
13:29.13 | lesouvage | ManxPower: thanks. |
13:29.17 | emdub | cmd dial |
13:29.21 | emdub | oops |
13:30.53 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
13:31.59 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
13:33.00 | test34 | Can you have speech recognition with asterisk ? |
13:34.03 | mog_home | yes |
13:34.06 | mog_home | via sphinx |
13:34.19 | ManxPower | test34, That question was asked on the mailing lists yesterday, as well as an answer was posted. |
13:34.59 | test34 | ok thanks |
13:35.26 | test34 | ManxPower, which mailing list ? |
13:35.56 | test34 | asterisk-users@lists.digium.com ? |
13:36.02 | ManxPower | test34, I don't recall |
13:36.14 | ManxPower | The subject didn't have the word "speech" in it. |
13:36.47 | test34 | ok |
13:37.43 | mutilator | . |
13:38.17 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
13:38.37 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
13:39.54 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
13:41.12 | test34 | http://lists.digium.com/pipermail/asterisk-users/2005-November/132160.html |
13:43.55 | *** join/#asterisk dalabera (n=dalabera@pmr.pmrtechnologies.com) |
13:43.55 | docelmo | whadup!!!!!!!! |
13:45.11 | neuwald | does anybody knows some link that explain how to implement a callback service? |
13:45.46 | Ariek1 | exit |
13:48.04 | test34 | neuwald, here's one http://lists.digium.com/pipermail/asterisk-users/2004-October/065406.html |
13:53.50 | Exion | Does anyone know if this should work with SIP channel (not yet established): SetVar(PRI_CAUSE=17); Hangup(); |
13:54.05 | Exion | the packet comming back from asterisk include 503 Service Unavailable + X-Asterisk-HangupCause:: User busy |
13:54.18 | Exion | X-Asterisk-HangupCause seems to change but not the status type |
13:55.46 | Exion | idealy it should be status 486... |
14:01.25 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
14:01.44 | Katty | hihi |
14:05.10 | Katty | too quiet. |
14:05.18 | Lostfrog | Si.. |
14:05.20 | Lostfrog | Too quiet |
14:06.01 | *** join/#asterisk Eight (n=blake@12-227-165-54.client.mchsi.com) |
14:07.23 | Lostfrog | Anyone familiar with Avaya ACS systems? |
14:10.16 | *** join/#asterisk potsboy (n=chrisg@196.34.241.242) |
14:11.22 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
14:13.23 | jalsot | does anybody use atxfer with success? the transfer itself works, however no get no audio. |
14:19.20 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
14:22.09 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
14:24.14 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-225.range86-138.btcentralplus.com) |
14:25.13 | *** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com) |
14:25.29 | Ferrari | good day all |
14:25.38 | Lostfrog | Good morning, Ferrari. |
14:26.17 | Ferrari | anyone know the MAX length of a variable (using setvar/getvar) in asterisk 1.0X i am too lazy today to read source so i figured i would ask in here first |
14:26.34 | Ferrari | looking for contents of the variable not variable name |
14:26.45 | Ferrari | i recall something about maybe 80 chars |
14:27.21 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85) |
14:27.30 | MuppetMaster | Hello. |
14:27.51 | file[laptop] | Hello has been replaced by, "Blurgblah" |
14:28.24 | MuppetMaster | I have compiled and installed ooh323 in the CVS HEAD (as of minutes ago) from within the /usr/src/asterisk-addons/.. And, the modules loads, based on the settings in /etc/h323.conf, but when I do a show channel types it does not show up. http://pastebin.ca/27600 |
14:28.41 | MuppetMaster | I had this working in v1.2beta1 with no problem. Has something changed in v1.2-beta2 on this front? |
14:29.55 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
14:30.12 | Lostfrog | Blurgblah, file[laptop]. |
14:32.15 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
14:32.22 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85) |
14:32.44 | swm_ | Ok got a problem with asterisk meetme confrence. |
14:33.03 | swm_ | Zap calls moved into a confrence work fine |
14:33.12 | swm_ | sip calls from the outside into the confrence dont work |
14:33.35 | swm_ | 841 conf_run: Error getting conference |
14:33.52 | swm_ | could it be a codec issue? |
14:34.22 | swm_ | brb gotta chain smoke on the issue |
14:38.17 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
14:43.01 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
14:45.20 | swm_ | anyone able to help? |
14:46.29 | docelmo | Whats new in life? |
14:46.40 | docelmo | SSDD? |
14:46.48 | swm_ | SSDD? |
14:46.59 | docelmo | Same Shit Different Day |
14:47.11 | *** join/#asterisk coppice (n=chatzill@5.162.17.210.dyn.pacific.net.hk) |
14:47.26 | swm_ | I cant transfer sip callers/callouts into a meetme room... driving me nuts |
14:50.41 | Lostfrog | Make them drive/fly to the same locale and sit down and talk.. lol |
14:52.33 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
14:53.49 | docelmo | Has anyone has problems with keeping the GXP2000 registered? |
14:55.42 | ManxPower | swm_, what codec are the SIP phones using? |
14:56.01 | ManxPower | swm_, and can the SIP phones call voicemail or other things that require asterisk to play audio to the caller? |
14:57.32 | swm_ | It's a outbound call I cannot transfer into the conference |
14:57.53 | swm_ | say I call 5551212@asterlink and try to move it into the meetme room, it just dumps the call |
14:58.19 | swm_ | I'm moving to CVS-HEAD current and see if that changes anything. Other than that It might be a codec issue |
15:00.01 | Lostfrog | that is going to be a large bundle of paper. |
15:00.14 | *** join/#asterisk peletiah2234 (n=benke@81.223.28.196) |
15:00.34 | peletiah2234 | good afternoon everyone |
15:00.50 | swm_ | Actually printed it once for the hell of it, had like 80 reams of paper and free toner for a month, so... Yeah, it took about 3 hours to print it all |
15:01.58 | *** part/#asterisk JmGV (n=jgomez@217.11.102.178) |
15:02.02 | Lostfrog | I've printed most of a 1.0 linux kernel before. |
15:02.11 | tzanger | I did that way back in the day |
15:02.18 | peletiah2234 | does someone know if its possible to specify channel distinct signalling types, like pri_cpe for pri1 and pri_net for pri2(for making a stress test)? |
15:02.52 | swm_ | I wonder how long it would take to print the linux 2.6 kernel.... heh |
15:02.59 | jontow | about 6 years.. ;) |
15:03.01 | jontow | until 2.8? |
15:03.05 | jontow | one of the two.. :D |
15:03.25 | jontow | i'll bet on the former, given the quick turnaround of the linux folks |
15:03.37 | swm_ | anyone get a plantronics bluetooth headset to work with asterisk yet ? |
15:03.47 | Lostfrog | I doubt it would take more that a week., |
15:04.15 | Lostfrog | I would be dead before I finished.. |
15:04.20 | Lostfrog | My boss would kill me. |
15:04.29 | jontow | thats assuming you've got a fast printer.. but we were back at dot matrix above ;) |
15:04.51 | swm_ | I miss old dot matrix impact printers |
15:04.52 | Lostfrog | You notice, he said 'toner' |
15:05.01 | Lostfrog | I want a line printer.. :) |
15:05.10 | jontow | hehe |
15:05.11 | Lostfrog | Drive the neighbors crazy. |
15:05.13 | swm_ | I want a serial based laser printer heh |
15:05.20 | jontow | i miss my old panasonic KX-24 |
15:05.20 | jontow | :( |
15:05.28 | swm_ | or even a serial based line printer |
15:05.35 | newl | They used to be good printers. |
15:05.40 | swm_ | Neeeeerrrrrrrrrrrrrr Neeerrrrrrrrrrrrrr Neeeerrrrrrrrrrrrrr |
15:05.44 | jontow | that thing rocked.. never had a jam, never misprinted.. just worked until it ran out of paper or ink, fix, repeat |
15:05.45 | *** join/#asterisk wolfson (n=ggggg@65.174.122.198) |
15:06.56 | swm_ | I remember my IBM lab at school we had some Panasonic KX-somethin printers ... oh they were so loud without the noise hood on them |
15:06.56 | *** join/#asterisk anti (n=russ@gentoo/developer/anti) |
15:08.03 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
15:08.07 | znoG | how does one raise the volume on a Zap channel? |
15:08.16 | swm_ | Wonder if they have a tatoo gun that works like a line printer but on a 2D axis ... changes ink and all that... that could be fun... |
15:08.17 | znoG | most people ask me to talk louder, even though I'm talking prettttty loud! |
15:08.50 | Lostfrog | Read the wiki? |
15:08.53 | Lostfrog | txgain? |
15:09.00 | swm_ | txgain works |
15:09.10 | swm_ | but you can get echo tweaking it to hard |
15:09.27 | Lostfrog | Whoever invented those printer stands had to have made a fortune. |
15:09.40 | swm_ | printer stands? |
15:09.54 | Lostfrog | The ones with foam padding inside. |
15:10.13 | Lostfrog | For line printers. |
15:10.34 | peletiah2234 | does someone know if its possible to specify channel distinct signalling types, like pri_cpe for pri1 and pri_net for pri2(for making a stress test)? |
15:11.00 | swm_ | peletiah2234: IF YOU CODE IT YOURSELF |
15:11.03 | Inkubot | when i make a call with a analoge phone, asterisk wait 10 seconds aprox before make the call |
15:11.15 | Lostfrog | Inkubot: digitimeout? |
15:11.23 | Lostfrog | digittimeout? |
15:11.23 | Inkubot | i really don't know |
15:11.33 | swm_ | what are you using to call out with lnkubot |
15:11.38 | Inkubot | a POTS |
15:11.44 | swm_ | pots what? |
15:11.50 | swm_ | Sipura? Cisco ATA? |
15:11.53 | Inkubot | aaa |
15:11.56 | Inkubot | a cisco ata |
15:11.57 | Inkubot | 182 |
15:12.14 | swm_ | Is your dialplan setup properly in the cisco? Maybe it's waiting for a timeout? |
15:12.16 | Inkubot | and also a IMC AccessLinx |
15:12.20 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
15:12.26 | newbien | know of any linux user groups that have an asterisk server providing outbound calls to lug members? |
15:12.28 | swm_ | Try dialing the number then pressing # after it, and see if it dials quicker :) |
15:12.35 | Inkubot | yeps |
15:12.40 | Inkubot | that's works fine |
15:12.43 | Inkubot | with # |
15:12.58 | Katty | hmm. |
15:13.07 | Lostfrog | Hmm.? |
15:13.14 | Inkubot | but here at work they don't want the # key |
15:13.29 | docelmo | Does anyone know of a Radius billing suite? |
15:13.41 | Inkubot | gbill ? |
15:14.03 | swm_ | lnkubot: your cisco ATA is not setup properly then for a dialplan. |
15:14.03 | peletiah2234 | ok, thank you. didn't mean to be annoying... i was not sure if its possible to set it distinctively for a channel group(and i'm still not sure how to interpret your answer) |
15:14.05 | Inkubot | docelmo: gbill, i think |
15:14.14 | Inkubot | ok swm_ let me see |
15:14.36 | swm_ | lnkubot: I could show you what mine looks like |
15:14.42 | docelmo | nice thank ink: |
15:14.47 | docelmo | thanks |
15:15.18 | Inkubot | thanks swm_ |
15:15.24 | Inkubot | let me see yours |
15:15.34 | Inkubot | (sorry for my english) |
15:15.41 | swm_ | 9..........|8...|7..........|11.|18.........|3..|4..|5...........|6>#35...................... |
15:16.05 | Inkubot | damn! |
15:16.25 | Inkubot | how you configurate the cisco ata ? |
15:16.39 | Inkubot | through the web page ? |
15:16.42 | swm_ | http://ipaddr/dev |
15:16.46 | Inkubot | a ok.. |
15:16.49 | Inkubot | me too |
15:16.57 | Inkubot | and where you put that! |
15:17.03 | Inkubot | there's nothing like that here |
15:17.10 | *** join/#asterisk sung (n=sung@216.106.191.1) |
15:17.24 | swm_ | section called "Dialplan" |
15:17.31 | Inkubot | aa |
15:17.38 | Inkubot | ok.. |
15:17.44 | Inkubot | 0 is default |
15:18.01 | swm_ | You have to customize it to your way of dialing for your country tho. |
15:18.17 | Inkubot | a ok |
15:18.30 | Inkubot | i think i'm understanding how it's work |
15:18.50 | swm_ | If you need more help refrence the cisco ata manual at cisco.com |
15:18.50 | Inkubot | the first is 9+ 10 numbers ? |
15:19.04 | znoG | so the idea behind rxgain/txgain is to get both around 50% ? (in ztmonitor) |
15:19.20 | Inkubot | swm_: thanks men |
15:19.21 | Inkubot | :) |
15:19.31 | swm_ | For america, I crop the 1, so it would be 9 503 312 1234 10 digits after the 9 to get out |
15:19.42 | znoG | Lostfrog: is that right? |
15:20.06 | Lostfrog | znoG: I don't know.. I just tweak it little by little 'till it sounds ok. |
15:20.21 | *** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-226-241.central.biz.rr.com) |
15:20.52 | Lostfrog | I haven't had to do in a while.. I use a T1 now. |
15:21.15 | Lostfrog | Voice level seems to be perfect. |
15:21.40 | *** join/#asterisk oogle (n=jart@justin.ctlinc.com) |
15:25.35 | znoG | i can't get zero echo to work even tweaking [tx,rx]gain settings |
15:25.49 | znoG | i tried to raise/lower them to varying levels and.. nope |
15:26.13 | ManxPower | echotraining=900 |
15:29.00 | tzanger | eww |
15:29.02 | tzanger | echotraining is nasty |
15:29.25 | tzanger | I realize it works but ewwwwwwwwwwwwwwwwwww |
15:29.30 | tzanger | it works like cod liver oil works |
15:29.36 | tzanger | the cure is sometimes worse than the disease |
15:30.30 | Inkubot | swm_: thanks men again |
15:30.35 | Inkubot | works fine |
15:30.39 | Inkubot | :D |
15:31.19 | fugitivo | cvs [login aborted]: unrecognized auth response from cvs.digium.com: |
15:31.31 | fugitivo | it's me? |
15:31.53 | fugitivo | yes it's me |
15:31.55 | fugitivo | :) |
15:32.22 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-66-83.hsd1.fl.comcast.net) |
15:32.57 | fugitivo | it works from one server, it doesn't from another one |
15:33.38 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) |
15:33.41 | Madkiss | hi folks. |
15:33.45 | Madkiss | I need some asterisk developer. |
15:35.27 | Beirdo | have a call come in via IAX, go out via IAX to my cellphone... the caller can hear me fine, I can't hear squat |
15:35.38 | Beirdo | something's being a PITA |
15:37.02 | znoG | tzanger: why you say that? is it that ugly? |
15:37.22 | *** join/#asterisk Miggidy (i=user@dsl-202-72-180-171.wa.westnet.com.au) |
15:37.34 | tzanger | znoG: echotraining makes me wait a second before I can talk |
15:37.36 | tzanger | drives me nuts |
15:37.45 | tzanger | if you have to pick the phone up to your head to answer that's cool |
15:37.59 | tzanger | but 99% of the time when I place a call the phone is at my head already |
15:38.05 | tzanger | or I pick up the phone and hit the line t answer |
15:38.49 | coppice | tzanger: so, how much can you be trained to do in just one second? :-) |
15:39.05 | tzanger | :-) |
15:39.56 | znoG | ah so I should call with speakerphone, and once answered pick up :) |
15:40.03 | znoG | i just have echotraining=yes |
15:40.06 | fugitivo | tzanger: 1 second? i have to wait like 15 seconds with my x100p |
15:40.16 | znoG | i want to raise txgain so people can hear me a bit better, but the echo increases with txgain |
15:40.22 | tzanger | znoG: yep it does |
15:40.28 | tzanger | properly balance the gains :-) |
15:40.48 | fugitivo | znoG: what hardware? |
15:41.34 | znoG | just a std Zap card |
15:41.47 | fugitivo | what's that? x100p? |
15:41.55 | znoG | yup |
15:42.50 | file | tzanger: ANDREW |
15:43.52 | tzanger | file: what |
15:43.59 | *** join/#asterisk marrandy (n=marrandy@209.216.76.1) |
15:44.05 | file | tzanger: hi |
15:44.08 | tzanger | heh |
15:44.08 | tzanger | hello |
15:44.09 | *** join/#asterisk tryfoss (n=morten@ti131310a080-14221.bb.online.no) |
15:44.16 | file | tzanger: banana? |
15:44.20 | *** join/#asterisk dsmouse (n=nnnnnnnm@prism.datastacks.com) |
15:44.27 | marrandy | hello - nice long list of people |
15:44.31 | tzanger | file: no thanks, I ate |
15:44.41 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85) |
15:44.43 | dsmouse | hello, marrandy |
15:44.44 | file | tzanger: fine then! |
15:44.44 | MuppetMaster | Hello |
15:44.46 | marrandy | anyone here using a fanless setup ? |
15:44.53 | MuppetMaster | Upgraded to CVS HEAD and now getting this: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 20 scheduled tasks all at once |
15:44.55 | MuppetMaster | Any ideas? |
15:44.56 | tzanger | file: at least quote the entire message |
15:44.59 | MuppetMaster | A problem?? |
15:45.05 | devel | is anybody here using the alarmreceiver module/app? |
15:45.21 | tzanger | I have an unbelievably thick skull. |
15:45.23 | tzanger | MuppetMaster: yeah I know |
15:45.35 | tzanger | MuppetMaster: do me a favor and disable the jitter buffer... does it go away? |
15:45.41 | MuppetMaster | Will check |
15:46.07 | coppice | marrandy: nobody is a fan of my setup |
15:46.12 | marrandy | :-) |
15:48.06 | marrandy | I'm looking to find a system that can do 12 extensions, 4 or which are pots phones, the other 8 sip phones, probably 5 or 5 in use at any one time but I'm looking for a low power, fanless system if possible |
15:48.45 | fugitivo | mini-itx |
15:49.20 | devel | marrandy, i just started some testing with the via mini-itx platform as well (but not the lower power fanless ones yet) |
15:49.38 | marrandy | yes, I've been looking at the via eden 533MHz, 800MHz and 1GHz but keep reading about issues, so was wonering of anyone here had a system |
15:49.40 | fugitivo | marrandy: via mini-itx model C3 |
15:50.07 | coppice | marrandy: what kind of issues? |
15:50.17 | MuppetMaster | tzanger disabled, no errors yet |
15:50.25 | fugitivo | marrandy: i think you have to make some changes to the code, the via mini-itx is not 100% x86 |
15:50.32 | marrandy | I could use a tdm400 with one pots line module and 3 pots phones if the m/b only had one pci slot |
15:51.13 | zoa | i used it before |
15:51.14 | zoa | it works |
15:51.25 | marrandy | devel: would like to know what m/b you asre trying anf the results |
15:51.31 | marrandy | asre = are |
15:51.58 | coppice | fugitivo: you just need to treat the processor as an i586 |
15:52.04 | MuppetMaster | I am also using the ooh323 driver, it appears that it now references itself to the config file /etc/asterisk/ooh323.conf instead of h323.conf as indicated in the readme. |
15:52.16 | fugitivo | marrandy: http://www.voip-info.org/wiki/view/Asterisk+hardware+mini-itx |
15:52.17 | marrandy | coppice: transcoding issues apparently |
15:52.29 | devel | marrandy, i've got the mII-10000, and if you check back in a few days i should have some detailed results |
15:52.35 | *** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net) |
15:52.39 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
15:53.22 | coppice | marrandy: well, they have limited processing power for things like transcoding. FC4 won't install on one, because they screwed up the installer. generally, as long as you treat them as i586 machines they are fine. |
15:54.02 | paryl | i'm trying to understand something about faxing... i keep reading "Right now Asterisk does not have very good fax support", but then they go on to describe what seems to be something similar to trying to fax over a VoIP connection |
15:54.26 | file | did someone subscribe the asterisk-users list to the internic.at support e-mail? |
15:54.38 | Rav1974 | hi guys, I have a TE110p with an ADIT 600. I am getting terrible echo. I tried RX & TX gain in asterisk & ADIT 600. |
15:54.40 | marrandy | fugitivo: yes, thanks, I've read those. was hoping somone here might have some working experience on those units |
15:54.42 | synthetiq | if you delete the logfiles will they be written to anymore? |
15:55.46 | marrandy | devel: do you have my email address? can you email me when you have some reults |
15:55.48 | Rav1974 | digium wasn't a big help either. I looked in google etc. but still no good. |
15:56.14 | MuppetMaster | If I do an IAX -> IAX call from Asterisk A to Asterisk B, where Asterisk B simply returns a Busy() app, I simply get an originate fail over the Manager API. How do I emulate a busy from a signaling perspective as one might expect from the PSTN? |
15:56.26 | devel | marrandy, /msg me your email addr and i'll keep track of it. |
15:57.01 | marrandy | at the moment, I'm running it on a dell poweredge 2400 with raid 5 which is a bit of a veast |
15:57.09 | marrandy | veast = beast |
15:58.14 | paryl | does faxing work well enough if you have the fax plugged into a tdm400p and bridge it directly to a T1? |
15:58.38 | devel | paryl, we do faxing over g711u sip |
15:58.40 | coppice | paryl: sometimes |
15:58.47 | snitt | paryl: that's ok |
15:58.53 | fugitivo | paryl: it should work most of the times, but it depends |
15:59.13 | *** join/#asterisk marrandy (n=marrandy@209.216.76.1) |
15:59.18 | paryl | sometimes? it depends? |
15:59.27 | fugitivo | paryl: sometimes the tdm400 fails to get the fax, at least with my experience |
15:59.33 | *** join/#asterisk morale (n=russell@S010600111155e117.cg.shawcable.net) |
15:59.38 | coppice | paryl: for a long time faxing would not work with the tdm400, but someone told me they finally fixed the driver |
15:59.56 | fugitivo | coppice: really? |
15:59.57 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85) |
16:00.02 | kuku5 | tdm400 is the pri card? |
16:00.06 | morale | is asterisk beta2 cvs-head? |
16:00.13 | fugitivo | kuku5: no, analog |
16:00.13 | marrandy | aack.. |
16:00.22 | kuku5 | ah |
16:00.29 | kuku5 | is the pri card cool for faxing? |
16:00.50 | marrandy | devel: did you get it ? |
16:00.51 | paryl | the issue here is... we have 8 fax machines in the building |
16:01.15 | paryl | and all voice/fax traffic goes over the t1 |
16:01.26 | fugitivo | paryl: sending fax always worked for me, no fails |
16:02.18 | znoG | they fixed the zaptel driver for faxing over the FXS port? |
16:04.35 | devel | yep, marrandy, got it. i'll get my results to you in a few days. |
16:05.15 | paryl | fugitivo: k, thanks. i hope it works out |
16:08.53 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
16:09.13 | fourcheeze | can anyone recommend a BRI card for use with * ? |
16:09.38 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85) |
16:09.41 | *** join/#asterisk RussCC (n=face@216.157.205.211) |
16:10.23 | MuppetMaster | With ooh323, I need to have port 1720 open for the signalling. What about for the audio stream? |
16:10.44 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
16:11.37 | fourcheeze | are there BRI cards out there with open source drivers? |
16:11.40 | *** join/#asterisk BleedingMe (n=Bleeding@ppp-69-237-68-225.dsl.scrm01.pacbell.net) |
16:11.45 | fourcheeze | I keep finding closed source ones |
16:12.00 | RussCC | when I attempt to make an outgoing call I get it squaking about localprefixes.conf, could some one please help me? |
16:14.58 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
16:16.18 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.85) |
16:17.15 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
16:17.32 | *** join/#asterisk Domiplus (n=domiplus@66-202-165-66.rev.knet.ca) |
16:18.11 | Rav1974 | help me please guys, I am having terrible echo problems |
16:18.20 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
16:18.24 | Rav1974 | I have a TE110p with an ADIT 600. I am getting terrible echo. I tried RX & TX gain in asterisk & ADIT 600. |
16:19.08 | *** join/#asterisk Madounet (n=Madounet@AMontpellier-206-1-2-64.w80-11.abo.wanadoo.fr) |
16:19.26 | Rav1974 | I'm praying it can be solved, I spent over $2,000 on equipment already... |
16:19.46 | nfi|ermes | problem compiling zaphfc and other bristuff drivers |
16:19.49 | nfi|ermes | http://pastebin.com/417275 |
16:19.59 | malverian[work] | Rav1974, It won't be ;) |
16:20.00 | file | hey Mr. DJ jam all night long |
16:20.06 | file | hey Mr. DJ play that song for me |
16:20.44 | malverian[work] | Rav1974, I have the same, and no matter what I've tried, echo persists on some calls. |
16:20.49 | Rav1974 | malverian[work]: you are a real comic. I'm still praying though... |
16:20.52 | Rav1974 | :) |
16:21.04 | Madounet | Hi, is there someone using zaprtc for IAX trunking ? |
16:21.10 | malverian[work] | Rav1974, I suppose you could try the new MG2 echo canceller. |
16:21.18 | Rav1974 | malverian[work] so its the te110p thats the problem? |
16:21.49 | *** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net) |
16:22.27 | Rav1974 | does Sangoma have the same problems with echo? |
16:24.34 | tzanger | Rav1974: it can |
16:24.40 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
16:24.44 | tzanger | Rav1974: have you tried the MG2 echo canceller |
16:24.46 | tzanger | (CVS HEAD) |
16:26.33 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
16:29.07 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
16:29.57 | oogle | i'd like to write a voicechanger version of Dial()... but i'm relatively new to asterisk hacking. can anyone give me some tips on where to start? |
16:31.04 | tzanger | oogle: I'd start with echo() |
16:31.21 | tzanger | and what you read in, connect to your voice changer, and write hte result out |
16:31.56 | malverian[work] | tzanger, I just said that ;) |
16:32.08 | tzanger | malverian[work]: heh |
16:32.28 | malverian[work] | Oddly enough though, echo became _worse_ here when I used MG2 over KB1 |
16:34.22 | *** join/#asterisk ful|work (n=fulgas@209.8.233.129) |
16:34.23 | Rav1974 | ya that doesn't help. I was on the phone with digium & they did all sorts of echo cancelation tricks |
16:34.39 | Rav1974 | I only get echo from fxo to fxs ports |
16:34.50 | swm_ | cheap fxo card? |
16:34.52 | Rav1974 | sip/iax to fxs doesn't have any echo |
16:35.00 | Rav1974 | nope I'm on an ADIT 600 |
16:35.04 | tzanger | Rav1974: that doesn't make any sense |
16:35.09 | Rav1974 | maybe I should try the cheap one :) |
16:35.10 | tzanger | Rav1974: have you tuned your Adit600? |
16:35.14 | tzanger | what do you have the gains set to on there |
16:35.16 | tzanger | I have an Adit600 |
16:35.19 | *** join/#asterisk timscott (n=war@d198-166-226-205.abhsia.telus.net) |
16:35.32 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
16:35.37 | Rav1974 | tzanger: I tried, but i reset it to defaults as per the manual |
16:35.41 | timscott | Is there anyone here who could help me with a problem with Asterisk and the DISA function? |
16:35.43 | Rav1974 | it helped with the volume problem |
16:36.08 | tzanger | Rav1974: I find I have zero echo with the adit600 fxs or fxo... it's got very nice interfaces |
16:36.33 | Rav1974 | tzanger: tell me your gain & impedance settings on adit please |
16:36.46 | tzanger | it's all default |
16:36.53 | tzanger | Rav1974: what processor are you running on? |
16:37.05 | Rav1974 | pentium 4 2ghz I think |
16:37.17 | Rav1974 | i'm not sure if its 2ghz or 2.4ghz |
16:37.22 | Rav1974 | with 256mb ram |
16:37.35 | oogle | tzanger: thank you for the voice change tip |
16:37.36 | syle | cat /proc/cpuinfo |
16:37.46 | tzanger | ok |
16:37.48 | tzanger | that's fine |
16:37.51 | tzanger | now do this |
16:37.56 | tzanger | use CVS HEAD |
16:37.59 | tzanger | if you're not already |
16:38.01 | tzanger | just for zaptel |
16:38.08 | Rav1974 | Intel(R) Pentium(R) 4 CPU 2.40GHz |
16:38.09 | tzanger | you don't need to go to CVS HEAD for asterisk if you don't want to |
16:38.17 | tzanger | in zconfig.h |
16:38.22 | tzanger | enable MMX optimizations |
16:38.27 | tzanger | seelct the MG2 echo canceller |
16:38.33 | tzanger | (make sure all the ones above it are commented out) |
16:38.43 | tzanger | in the zaptel Makefile |
16:38.52 | tzanger | below the reference to "all the settings are in zconfig.h" |
16:38.54 | tzanger | add |
16:39.00 | Rav1974 | ya, the digium tech support did as you said |
16:39.02 | tzanger | CFLAGS+=-march=pentium4 |
16:39.09 | tzanger | KFLAGS+=-march=pentium4 |
16:39.16 | tzanger | make clean && make && make install |
16:39.31 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
16:39.35 | oogle | ok |
16:39.36 | Rav1974 | i don't think he did "CFLAGS" |
16:39.40 | Rav1974 | i'll try that |
16:39.52 | timscott | Is there anyone here willing to field a couple Asterisk questions about the DISA function? |
16:40.09 | tzanger | timscott: I can try |
16:40.10 | Qwell | timscott: only if you ask them |
16:40.15 | tzanger | generally though you just ask :-) |
16:40.17 | Qwell | off to work :D |
16:40.18 | timscott | Oh. |
16:40.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
16:40.44 | timscott | I'm trying to use the DISA functions in Asterisk, but my DTMF is not being passed from PSTN -> Asterisk. |
16:40.46 | timscott | What should I be doing? |
16:40.56 | tzanger | timscott: have you played with your zapata gains? |
16:40.58 | timscott | I've already tried like a million things. |
16:41.01 | timscott | Err |
16:41.04 | timscott | I'm using VoIP. |
16:41.09 | tzanger | timscott: you just said from PSTN |
16:41.12 | timscott | But I can check, I can't remember what they are set to at the moment. |
16:41.14 | timscott | Well, yeah. |
16:41.15 | tzanger | is this a SIP provider? |
16:41.18 | timscott | IAX. |
16:41.22 | timscott | Voipjet. |
16:41.26 | tzanger | bleh, voipjet |
16:41.29 | tzanger | try this |
16:41.30 | tzanger | iax2 debug |
16:41.33 | tzanger | and try to call in |
16:41.43 | tzanger | you should see messages about "DTMF event received: 5" or whatever digit was pressed |
16:41.51 | tzanger | if you do not see that then VoipJet is not sending it to you |
16:41.57 | tzanger | IAX2 DTMF is not a problem as it is with SIP |
16:42.16 | tzanger | but if VoipJet's upstream is SIP and isn't passing it properly to VoipJet you'll not see it in IAX |
16:42.17 | implicit | timscott, and only a problem with sip if you are stupid as a log |
16:42.32 | tzanger | implicit: not really |
16:42.37 | implicit | and don't understand the different dtmf modes |
16:42.43 | implicit | and what they are useful for |
16:42.54 | tzanger | there are lots of providers who won't give you RFC2833 but insist you can get inband DTMF with compressed voice codecs |
16:43.06 | file | ugh |
16:43.08 | file | don't remind me |
16:43.12 | file | <cough>Level3</cough> |
16:43.21 | fugitivo | tzanger: which providers do that? |
16:43.23 | implicit | and there are lots who will suck your dick and not complain |
16:43.24 | timscott | I'll check. |
16:43.53 | *** join/#asterisk lehel (n=mey@86.125.98.100) |
16:43.55 | implicit | it's not the problem of SIP, its the stupidity of people using it |
16:44.04 | sivana | where is * looking for sendmail? |
16:44.05 | implicit | and it's not the problem of RTP either |
16:44.18 | fourcheeze | file: level3 do that - I thought they were supposed to have some clue at least |
16:44.21 | paryl | is there a way to make the polycom 501 'dial instantly'... so that as soon as what i'm dialing matches a pattern asterisk recognizes... |
16:44.27 | file | fourcheeze: I know |
16:44.31 | mutilator | well implicit |
16:44.36 | Rav1974 | please recommend a reliable VOIP provider (only for outgoing) |
16:44.37 | mutilator | a phone isn't an advanced user interface |
16:44.39 | swm_ | Why can you not move a parked call into a queue? "Channel 'Zap/1-1' sent into invalid extension '350' in context 'parkedcalls', but no invalid handler ... Hungup 'Zap/1-1' ... no channel type registered for 'PARK' ... Unable to request channel PARK/350 |
16:44.48 | mutilator | it works or it doesn't |
16:45.04 | Chuji | paryl : Try selecting the line first, then dialing |
16:45.05 | implicit | Does anyone know how to or if you can enable SRTP on Cisco as5400? |
16:45.09 | implicit | for SIP calls? |
16:45.12 | shmaltz | swm_ what are you trying to do? |
16:45.21 | paryl | tzanger: you said "seelct the MG2 echo canceller"... i don't see an option for MG2 |
16:45.23 | swm_ | Move a parked call into a queue |
16:45.23 | Chuji | paryl : If I do that, and dial 10 digits, it dials |
16:45.30 | tzanger | paryl: CVS HEAD |
16:45.44 | paryl | Chuji: i want it to dial without waiting |
16:45.46 | swm_ | there appears to be no bridge between queue's and parked calls |
16:45.52 | paryl | tzanger: that what i'm in |
16:46.03 | paryl | it should be in zconfig.h? |
16:46.28 | tzanger | #define ECHO_CAN_MG2 |
16:46.30 | tzanger | in zconfig.h |
16:46.45 | [TK]D-Fender | Hey File, care to pass your contact info for while you're in montreal so we can coordinate? |
16:46.48 | timscott | tzanger: I'm not seeing it. |
16:47.00 | file | [TK]D-Fender: sure |
16:47.04 | shmaltz | wwm_, you will have to pick it up, or make the timeout extension goto that queue using parkandannounce |
16:47.15 | tzanger | timscott: well you've successfully diagnosed your first Asterisk IAX2 problem :-) Now go after VoipJet asking them why you can't see DTMF when you call your DID |
16:47.45 | coppice | #define ECHO_CAN_MG2 in not more than 25 words |
16:47.52 | swm_ | shmaltz. I'm using FOP (flash operator panel) Beeing both defined, It should be able to be dragged from a parked state to a queue with no problem tho. |
16:47.56 | timscott | tzanger: will do. |
16:47.59 | timscott | Gracias. |
16:48.03 | tzanger | no problem. |
16:48.05 | tzanger | coppice: ? |
16:48.35 | shmaltz | swm_, when you drag from a sip/zap channel to that queue does it work? |
16:48.58 | swm_ | shmaltz. Drops the call gives the errors stated earlier |
16:49.07 | implicit | coppice, do you know anything about srtp on ciscos or some sort of b2bua that can also do srtp and proxys media? |
16:49.14 | swm_ | it's because asterisk has no channel defined named "PARK" |
16:49.15 | shmaltz | swm_ even when doing it from sip/zap? |
16:49.29 | shmaltz | swm_ patebin your extensions.conf please |
16:49.30 | swm_ | shmaltz. Affirmative |
16:49.33 | shmaltz | are you using AAH? |
16:49.37 | coppice | implicit: ciscos have SRTP. very few other things seem to right now |
16:49.40 | swm_ | AAH ? WTF? |
16:49.48 | implicit | coppice, for SIP calls do you know how to get support on the as5400? |
16:49.59 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
16:50.42 | shmaltz | ~aah |
16:50.43 | jbot | aah is probably Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324 |
16:50.44 | coppice | implicit: I haven't used it. I just know its there |
16:50.44 | shmaltz | ~aah is Asteirsk At Home |
16:50.45 | jbot | ...but aah is already something else... |
16:50.45 | implicit | i know it's there but i don't know if it has support for negotiating keys over SIP |
16:50.45 | implicit | they say a lot of crap about mgcp |
16:50.45 | swm_ | What the question basically is. Why does asterisk specifically channel.c line 2525 ast_request have No channel type registered for 'PARK' |
16:51.06 | coppice | implicit: it wouldn't appear to be of much use with the key management part |
16:51.07 | swm_ | ~beat asterisk |
16:51.15 | implicit | coppice, exactly |
16:51.20 | implicit | i don't know if they support it with SIP |
16:51.24 | shmaltz | swm_ please let me see ur extensions.conf |
16:52.02 | Rav1974 | tzanger: please tell me where you added CFLAGS+=-march=pentium4 |
16:52.20 | swm_ | I dont think extensions.conf has anything to do with this issue, I can do everything but move a parked call to a queue |
16:52.41 | tzanger | in Makefile, under the comments about zconfig.h |
16:52.42 | Rav1974 | tzanger: at the bottom of zconfig.h? |
16:52.44 | tzanger | no |
16:52.48 | coppice | implicit: the srtp.sourceforge.net project is from cisco, and that is only just springing to life to become more polished and complete. perhaps cicsco's implementation in their own kit still has bits missing |
16:53.01 | implicit | i am sick of cisco |
16:53.09 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
16:53.16 | implicit | i've had so many problems with REFER's on their gw's and shit too |
16:53.49 | implicit | they are still smokin the sccp pot |
16:54.00 | implicit | who the FUCK would use it, i don't know |
16:54.19 | coppice | corporate users of CCM use it |
16:54.30 | shmaltz | swm_, OK, I know why |
16:54.32 | implicit | obviously the only ones |
16:54.50 | coppice | that's all they need. they pay a lot :-) |
16:54.51 | *** join/#asterisk js_birdboy (n=jselleck@60.68-224-177.reverse.smoothfusion.com) |
16:55.13 | shmaltz | swm_ what if you make a button in FOP for the other leg of the call (like zap or sip) that is parked and drag that? |
16:55.39 | shmaltz | swm_ there is a problem in the manager with redirects on parked calls |
16:55.42 | js_birdboy | does anyone know a good way of keeping track of peak trunk connection usage? |
16:57.32 | shmaltz | you can see that for yourself, fireup astman and park a call then try to redirect it |
17:00.46 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
17:00.46 | *** join/#asterisk gst (i=gst@eris.sysfrog.org) |
17:02.52 | ms345 | If I get the callerid number sent to * on my BellSouth PRI what do I do to get the CallerID name? Just ask the telco to send a name too? |
17:03.10 | shmaltz | ms345, you can ask, but they wont do that |
17:03.22 | shmaltz | you generaly don't get cname on PRIs |
17:03.22 | ms345 | how are you supposed to get it? |
17:03.33 | shmaltz | ms345, you don't |
17:03.40 | shmaltz | nobody does on PRI |
17:03.48 | ms345 | darn... |
17:04.14 | shmaltz | js_birdboy, I remember something like MRTG on the wiki |
17:04.15 | ms345 | I wonder if I can look it up on Google quick enough.... |
17:04.19 | shmaltz | search for it |
17:04.40 | shmaltz | ms345, sure, search the list there are some scripts on how to do it |
17:04.50 | ms345 | thx! |
17:05.06 | shmaltz | ms345, a quick hack would be |
17:05.30 | ms345 | I already send the CID # via jabber, but looking for a name... |
17:06.00 | shmaltz | ms345, you call a POTS that has callerID service with the callerID set to what you got on the incoming call, then make the dial statement to the final queue extension, you can then cache the name |
17:06.13 | christo | I get 'Could not create socket' whenever I attempt to connect to the manager interface on a * box. Netstat on the box shows nothing listening on 5038. How can I get the manager API to start accepting connections? |
17:06.37 | shmaltz | christo, /etc/asterisk/manager.conf |
17:07.27 | christo | shmaltz - I have my manager.conf all set up. |
17:07.40 | shmaltz | christo, then restart asterisk |
17:07.56 | shmaltz | christo, can you pastebin your manager.conf |
17:08.19 | ms345 | interesting... that is quite a hack - I'll dig further - thanks for the insight. |
17:08.19 | christo | shmaltz - yes, I've tried restarting *. My manager.conf is here: use Topdates::DB; |
17:09.04 | shmaltz | christo, WTF? |
17:09.28 | christo | doh - bad paste... http://pastebin.ca/27616 |
17:09.46 | christo | I nearly posted you my personal calendar :) |
17:10.15 | christo | you could've seen which night I'm having dinner with my mum next week heh |
17:10.25 | brettnem | I bet you've got a lot of juicy stuff on that! |
17:10.58 | js_birdboy | thanks shmaltz |
17:11.15 | shmaltz | js_birdby, for what? |
17:11.35 | *** part/#asterisk timscott (n=war@d198-166-226-205.abhsia.telus.net) |
17:11.42 | shmaltz | js_birdboy, for what? |
17:11.43 | js_birdboy | the MRTG bit about tracking peak trunk connections |
17:11.51 | shmaltz | oh I c |
17:11.52 | shmaltz | np |
17:12.21 | shmaltz | christo, you have enable=no, change that to enable=yes |
17:12.23 | *** join/#asterisk CoderCR (n=creyna@dsl093-157-131.phx1.dsl.speakeasy.net) |
17:12.37 | *** part/#asterisk CoderCR (n=creyna@dsl093-157-131.phx1.dsl.speakeasy.net) |
17:12.48 | Rav1974 | tzanger: how do you think it will sound without an echo canceler? |
17:12.51 | shmaltz | err, enabled=yes |
17:13.36 | tzanger | Rav1974: it will have very loud echo. the time between your voice going out and the echo coming back may be very short, or it may be a hundred ms or more depending on various factors |
17:14.21 | christo | shmaltz - I have played with the 'enabled' setting under the 'general' section in manager.conf and it doesn't make any difference.. |
17:14.29 | christo | I just hadn't set it back to 'yes' |
17:14.31 | Rav1974 | tzanger: where is the makefile (i'm new to this) |
17:14.42 | tzanger | zaptel source directory |
17:14.53 | *** join/#asterisk delmar (i=delmar@203.114.178.231) |
17:14.55 | shmaltz | christo, set it to yes, and restart |
17:15.05 | christo | I have |
17:15.09 | christo | a few times :S |
17:16.36 | christo | however, I can see it running now when I do a netstat -tpl |
17:17.00 | *** join/#asterisk rob314 (n=root@207.58.194.55) |
17:17.09 | delmar | sip --> asterisk -- iax2 -- asterisk ---> sip ... having some minor echo problems. locally at each asterisk ie sip ---> asterisk -----> sip .. no echo problems at all. what * config bits might I look into to help resolve this? |
17:17.23 | fugitivo | grat |
17:17.24 | fugitivo | great |
17:17.35 | fugitivo | i'm getting Illegal instruction running cvs version |
17:17.42 | *** part/#asterisk rob314 (n=root@207.58.194.55) |
17:17.43 | lehel | :P |
17:18.49 | *** join/#asterisk potsboy (n=chrisg@196.34.241.242) |
17:21.37 | *** join/#asterisk Defraz (n=t0tal@72.24.26.210) |
17:21.43 | shmaltz | delmar, try not using iax2 |
17:22.10 | delmar | shmaltz, oh? |
17:22.12 | Rav1974 | tzanger: sorry for sounding dumb, but there is no "makefile" in the zaptel source directory. only makdfw.c |
17:22.21 | delmar | shmaltz why not just make it work right? |
17:22.33 | lehel | Rav1974: ) u r serious? |
17:22.38 | shmaltz | delmar, because iax2 is not right |
17:22.39 | christo | Rav1974 - run ./configure first? |
17:22.52 | shmaltz | I wish it work as good as sip but it dosnt |
17:22.54 | delmar | shmaltz, how so |
17:23.07 | delmar | shmaltz, so iax2 has known echo problems? |
17:23.17 | shmaltz | delmar, one sec |
17:23.20 | shmaltz | im on the phone |
17:23.20 | drumkilla | that makes absolutely no sense |
17:23.21 | fugitivo | iax2 works ok for me |
17:23.22 | fugitivo | no echo |
17:23.30 | drumkilla | iax2 has *NOTHING* to do with echo |
17:23.32 | delmar | shmaltz, i have found it working great in other solutions... |
17:23.44 | delmar | right |
17:23.47 | drumkilla | nor does any other VoIP protocol |
17:23.47 | delmar | so where is my echo :P |
17:23.53 | file | drumkilla: !!!!!!!!!!! |
17:24.13 | delmar | ok so here is the solution that has echo..... |
17:24.13 | drumkilla | delmar: at some analog interface |
17:24.19 | fugitivo | delmar: using softphone? |
17:24.40 | drumkilla | yay |
17:24.49 | Chuji | drumkilla : what's the original bug that the wW options went in |
17:24.51 | delmar | sip --> asterisk -- iax2 -- asterisk ---> sip ... codecs are ulaw to g729 over iax... back to ulaw |
17:25.00 | delmar | hardphones |
17:25.07 | drumkilla | Chuji: 12312 |
17:25.11 | delmar | no analog interfaces involved in that |
17:25.11 | drumkilla | jk, I have no clue |
17:25.15 | fugitivo | codec_ilbc is broken on today's cvs? |
17:25.26 | drumkilla | delmar: there is still an analog interface at the endpoints |
17:25.30 | drumkilla | your sip phones suck :) |
17:25.31 | tzanger | Rav1974: there is a file called "Makefile" there or you are unable to build it. |
17:25.33 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
17:25.54 | fugitivo | i have to noload => codec_ilbc.so |
17:26.03 | delmar | drumkilla, no, there is no analog interface dude. unless you count the mic/speaker at each hardphone |
17:26.10 | fugitivo | i'm getting Illegal instruction |
17:26.16 | drumkilla | delmar: yes, i do :) |
17:26.19 | drumkilla | because that's what it is |
17:26.45 | delmar | drumkilla, well if thats your best effort in here... im just gonna laugh :P |
17:27.04 | drumkilla | best effort? |
17:27.15 | lesouvage | Will somebody please look at http://pastebin.com/417327 to see what I'm doing wrong with my first AGI script. |
17:27.38 | fugitivo | anyone else having problems with codec_ilbc.so? |
17:28.07 | tzanger | Corydon-w: re: 3360 |
17:28.22 | tzanger | Corydon-w: I understand what you're doing but now this requires a change with all VOIP providers I use to be able to set this |
17:28.22 | mutilator | haha |
17:28.25 | docelmo | •lesouvage• its written in python ??? |
17:28.30 | mutilator | this guy in our techsupport chatroom is a riot |
17:28.38 | mutilator | he's bashin on the supporrt dude hardcore |
17:28.50 | mutilator | and support guy is trying to be nice as possible |
17:29.08 | mutilator | <dyoderindustries-Net4Kids> no just nevermind...I'll get Carla to help me, she's nicer than you |
17:29.12 | mutilator | heh |
17:29.31 | mutilator | guess ya gotta see it and know who in support hes talkin to to find it as funny as i do |
17:30.54 | Corydon-w | tzanger: it would require all service providers you use to update to the latest code after a fix goes in ANY way |
17:31.09 | lesouvage | docelmo: from what I understand from the o'reilly book that should be working. I'm open for every advice and direction to info how read a first line of a flat file. |
17:31.24 | delmar | siphardphone(ulaw) ---> Asterisk ----> siphardphone(ulaw) works mint... siphardphone(ulaw) ----> Asterisk--->PSTN(TDM) .. works mint... siphardphone(ulaw)----->Asterisk ---->VoIPProvider(g729/IAX) .. works mint... siphardphone(ulaw) ---->Asterisk----->VoIPProvider(IAX2/g729) ... works mint.... so you see... most things are working fantastic... just not working so great when it comes to siphardphone(ulaw) --> Asterisk -->iax2/g729---->Asterisk----->sipha |
17:31.25 | tzanger | Corydon-w: nonsense |
17:31.58 | tzanger | on MY asterisk system if a peer's qualify has expired they become UNREACHABLE (hell it even says UNREACHABLE!) -- Dial should fail immediately with CHANUNAVAIL |
17:32.01 | tzanger | not CONGESTION |
17:32.03 | Corydon-w | tzanger: well, then, you don't need a change in the Asterisk source code |
17:32.13 | tzanger | Corydon-w: ?? |
17:32.28 | Corydon-w | tzanger: you either need a change in the source code or you don't |
17:32.32 | tzanger | I do |
17:32.45 | tzanger | I need the damn Dial() to return CHANUNAVAIL if the qualify shows them as unreachable |
17:32.46 | Corydon-w | If you need a change in the source, you're going to need your providers to update to that change |
17:32.48 | tzanger | I can do it |
17:32.50 | tzanger | no |
17:32.52 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
17:32.54 | tzanger | it's on my system |
17:32.54 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
17:32.56 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
17:33.10 | Corydon-w | Then why not just change your system and be done with it? |
17:33.26 | Rav1974 | tzanger: I feel silly to say it but, the "Makefile" not makefile was in the zaptel source directory. I forgot the case *sorry* |
17:33.31 | fugitivo | should i post a bug for codec_ilbc? anyone else with the illegal instruction error? |
17:33.42 | tzanger | Rav1974: no worries |
17:33.50 | KranZ | ilbc is crappy anyways |
17:33.59 | delmar | so back to my original question... what are the main things to look at with Asterisk when it comes to echo problems in a solution like siphardphone(ulaw) --> Asterisk -->iax2/g729---->Asterisk----->siphardphone(ulaw).. .anyone? |
17:34.07 | fugitivo | yes, but it's not working for me at all, asterisk is crashing |
17:34.24 | KranZ | the quality is not worth it |
17:34.33 | docelmo | •lesouvage• I dont know python nor do I want to learn it. I was more or less making a crack at you for using python. In theory yes.. Python should work.. But I dont know how to develop in it. |
17:35.26 | KranZ | delmar: latency and if echo cancellation is enabled at both phones |
17:35.41 | potsboy | delmar, check you have ztdummy compiled if you dont have a card to provide timing |
17:35.53 | docelmo | •lesouvage• Do you know PHP? That I do know.. |
17:36.00 | Rav1974 | Features are now configured in zconfig.h |
17:36.09 | KranZ | docelmo: do u use php in the dialplan? |
17:36.15 | docelmo | Yes |
17:36.18 | docelmo | in a round about way |
17:36.19 | delmar | potsboy, im not using trunking.. thought u only had to worry about that if u are trunking and need a timing source? |
17:36.22 | Rav1974 | tzanger: below that line I add the C&K flags |
17:36.27 | tzanger | yes |
17:36.38 | KranZ | docelmo: what mod did u install for that? |
17:36.44 | docelmo | about 75% of the time I tell the dialpaln to dump to a phpAGI file and continue running the dialplan |
17:37.04 | lesouvage | docelmo: What I try to do is run an agi script that reads the first line of a file (this is a phonenumber) and return it to asterisk for further use. A suggestion on how to this in php is more then welcome. |
17:37.14 | KranZ | so you dont do any call directing with php? |
17:37.41 | docelmo | I do everything in php.. I can get more accomplished 90% of the time |
17:37.53 | docelmo | •lesouvage• hold on I will make a quick agi for it. |
17:38.44 | lesouvage | docelmo: thanks a lot. I spend hours on it without any result. |
17:39.02 | docelmo | You want just 1 line from a text file? |
17:39.03 | *** join/#asterisk nitram (i=foo@superblob.com) |
17:39.15 | shmaltz | delmar, I'm back to that iax2 argument |
17:39.17 | lesouvage | docelmo: yes |
17:39.23 | docelmo | The first line? |
17:39.30 | delmar | shmaltz oh? |
17:39.43 | shmaltz | delmar, iax2 does not work very nicely |
17:39.51 | shmaltz | it has terrible sound quality |
17:40.08 | shmaltz | including ticks and distorted sound |
17:40.17 | KranZ | timing |
17:40.28 | delmar | shmaltz, dude.. sound quality is really a codec thing. i have no problems except this one solution and I believe its a case of it not working because its not setup right and it can be fixed. |
17:40.45 | shmaltz | your echo is however more related to something else and not iax2 |
17:40.45 | fugitivo | the new echocan is worse than the old one |
17:41.07 | shmaltz | delmar, codec, so which codec you recommend that I havnt tried yet with iax? |
17:41.10 | delmar | so in regards to this timer.... |
17:41.16 | lesouvage | docelmo: yes the first line. It's a file that holds a number that wasn't picked up. I will use it for a retry by asterisk. |
17:41.33 | shmaltz | I have a digium t110 with that iax problem and still |
17:41.33 | delmar | potsboy, there is a X100 clone card in one box and a TDM card in the other... do I still need ztdummy? |
17:41.54 | shmaltz | delmar, timing will not change echo problems |
17:41.57 | delmar | shmaltz, im using g729 atm. |
17:42.25 | shmaltz | delmar I have tried alaw, ulaw, gsm, g729, g723 and no luck |
17:42.37 | shmaltz | anyhow, back to ur echo problem |
17:42.48 | shmaltz | what sip devices you using when you have echo? |
17:43.21 | delmar | so as I said above... |
17:43.22 | delmar | siphardphone(ulaw) ---> Asterisk ----> siphardphone(ulaw) works mint... siphardphone(ulaw) ----> Asterisk--->PSTN(TDM) |
17:43.22 | delmar | <PROTECTED> |
17:43.24 | delmar | <PROTECTED> |
17:43.30 | delmar | all working... |
17:43.36 | delmar | and working really nicely |
17:43.52 | delmar | siphardphone(ulaw) --> Asterisk -->iax2/g729---->Asterisk----->siphardphone(ulaw) .. has minor echo |
17:44.20 | delmar | and i mean.. its not nasty loud stuff.. but its there.. and its at a level where it's kinda tolerable but .. can be a pain |
17:44.39 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
17:45.31 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
17:45.59 | delmar | shmaltz, so in answer to your question.. sip devices are hardphones. |
17:45.59 | docelmo | So now KranZ what did you want to know about AGI and PHP? |
17:47.14 | shmaltz | delmar, try using ulaw or g729 all the way, with no transcoding, you are doing double transcoding which be the source of the echo |
17:47.26 | docelmo | delmar have you tried SIP->SIP for your IAX connection? |
17:47.39 | docelmo | That too |
17:48.22 | fugitivo | damn, echo is higher with the new echocan |
17:48.24 | docelmo | I had the problem with echo on sip.. and it was all the codecs.. I cleared it up by g711u in the office to TDM out Cisco or PSTN PRI |
17:48.27 | delmar | shmaltz, yeah i had thought of this but the transcoding time is very low on each * box, ok so something to try when someone is at the remote office.. 7am here so a little early.. what else... |
17:49.00 | docelmo | 7am.. Where are you? |
17:49.09 | delmar | ok so SIP between * boxes rather than IAX2... |
17:49.20 | docelmo | I use sip unless I am trunking |
17:49.23 | delmar | New Zealand |
17:49.27 | docelmo | ahh |
17:49.49 | shmaltz | delmar, try that transcoding before you don't eliminate the obvious you cant complain |
17:50.04 | delmar | :P |
17:50.45 | delmar | im interested in this timing thing anyway... whats the story... if there is ANY Zaptel hardware, there is a timer? ie X100 clone, or TDM card? |
17:51.16 | docelmo | If you have Zaptel card timing is slaved from the card.. |
17:51.22 | docelmo | No card use clone or Ztdummy |
17:51.40 | docelmo | Have to watch with Ztdummy.. Have to have right USB chipset for it to work |
17:51.43 | veto | Anyone played or have a SPA-941? |
17:51.50 | delmar | ok |
17:52.31 | delmar | so on a system that has no Zaptel hardware whatsoever, I should setup ztdummy etc huh? |
17:52.47 | delmar | or is it not necessary unless trunking on IAX? |
17:53.05 | fugitivo | you only want ztdummy for things like meetme where you need a timer |
17:53.10 | shmaltz | delmar, you need for lots of other things as well |
17:53.12 | fugitivo | you don't need if for iax |
17:53.23 | shmaltz | fugitivo, moh also works better with timing |
17:53.47 | fugitivo | shmaltz: using native moh too? |
17:54.00 | shmaltz | fugitivo, yep |
17:54.07 | delmar | well I have a system up and running that is just * with no Zaptel hardware.. and it works great.. so since it aint broke.. i wont fix it :P |
17:54.13 | fugitivo | didn't know that :) |
17:55.08 | shmaltz | delmar, I agree with you, my mother always says about my littel baby girl, if she is happy don't try to make her happier ;) |
17:55.09 | delmar | moh? |
17:55.15 | fugitivo | music on hold |
17:55.16 | potsboy | correct me if i am wrong but ztdummy only comes into play with multiple calls |
17:55.16 | shmaltz | ~moh |
17:55.23 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
17:55.40 | shmaltz | potsboy, on iax trunking yeah, but scrollup to read more |
17:55.48 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
17:55.49 | potsboy | oops..k |
17:55.53 | delmar | potsboy, ok so u suggested timing issue.. but there is Zaptel hardware in both systems. |
17:56.08 | shmaltz | ~moh |
17:56.09 | jbot | [moh] Music On Hold |
17:56.34 | potsboy | k... i though you said was only a sip network with no break out...why have cards then? |
17:56.43 | delmar | i guess im wondering if there could be anything wrong with my IAX config between the * boxen... i dunno.. jitterbuffer on/off? dropcount? what is all that shit good for? :P |
17:57.03 | delmar | potsboy, the cards are used.. and when they are used things are fine.... |
17:57.16 | delmar | potsboy, scroll up... i detail whats working mint. |
17:57.45 | delmar | and whats not... |
17:59.35 | FuriousGeorge | im looking to use something other than eyebeam |
17:59.37 | delmar | ok so something else.. has anyone played with PA1688 based phones.. specifically any experience with IPPH203 (TigerNetCom aka TigernetCrap) because for the life of me I just cannot get these phones to do a simple call transfer.. got a couple of Grandstream BT101/102 phoens here and they are just awesome.. hold, transfer, hook-flash etc all works nice... |
17:59.42 | FuriousGeorge | any suggestions |
18:00.57 | iCEBrkr | Whutup! |
18:01.13 | docelmo | Shutup! |
18:01.18 | docelmo | :) |
18:01.53 | docelmo | hay ice.. When do you want that box? |
18:02.07 | FuriousGeorge | i dont care if its isax or sip, but im wanna try something different than the xten/counterpath clients ive been using |
18:02.15 | iCEBrkr | docelmo: Hang a sec. |
18:04.01 | iCEBrkr | docelmo: Crap. I'm gonna have to hold off on that box. |
18:04.16 | *** join/#asterisk Inv_arp (i=junya@adsl-144-17-25.mia.bellsouth.net) |
18:04.46 | Inv_arp | w00t I GOT POWER!!! |
18:05.15 | Inv_arp | only south fl residents would understand |
18:05.21 | FuriousGeorge | ive tried firefly but it doesnt seem to implement a transfer which kinda bothers me, nor does it seem to have the ability to take or make multiple calls |
18:05.26 | docelmo | INV ya ya ya |
18:05.38 | docelmo | I didnt have power last year.. I dont feel bad for you |
18:05.39 | docelmo | :) |
18:05.50 | Inv_arp | :) |
18:05.57 | docelmo | Im in Tampa BTW |
18:06.29 | Inv_arp | ahh k you guys werent hit too hard? |
18:06.50 | kuku5 | <PROTECTED> |
18:07.10 | docelmo | me me! |
18:07.15 | docelmo | I terminate A-Z |
18:07.20 | kuku5 | i need poland |
18:07.24 | kuku5 | whast your fee |
18:07.56 | docelmo | lemme go check my rate deck |
18:07.59 | docelmo | how much traffic |
18:08.07 | kuku5 | v. little now |
18:08.30 | docelmo | 0.0234 |
18:09.08 | docelmo | http://www.plainvoip.com/?page=showrates |
18:11.43 | kuku5 | ouch |
18:11.46 | kuku5 | too high |
18:11.53 | *** join/#asterisk demetrio (n=demetrio@62.173.180.182) |
18:11.56 | kuku5 | i get it for 1.9 now - high quality |
18:11.58 | demetrio | hi everyone |
18:12.06 | Inv_arp | demetrio: sup |
18:13.24 | demetrio | I just spent 2 hours trying to find out how blind transfer works in asterisk, and found nothing. does anybody here have a clue? |
18:14.11 | kuku5 | ergh |
18:14.14 | kuku5 | it just transfers |
18:14.40 | js_birdboy | as with many things in asterisk... it just works |
18:14.43 | demetrio | sure, but how do they work? the only thing I know is that the # key is involed, that's all the documentation says. |
18:14.49 | delmar | demetrio, i have a couple of grandstream phones here attached to * and can xfer either hook-flash or use the transfer button |
18:14.51 | docelmo | •Inv_arp• Your happy to have your power arent you.. :) |
18:15.14 | demetrio | let's put it this way: what do I have to do to transfer a call? |
18:15.16 | *** join/#asterisk Jake1932-mobile (n=Jake1932@18.sub-70-202-114.myvzw.com) |
18:15.16 | delmar | demetrio, sounds like you are not using the native transfer etc. |
18:15.22 | docelmo | flash |
18:15.27 | demetrio | nope, I'm using SIP blind transfer |
18:15.37 | js_birdboy | like #extension does a blind transfer... is that what you are talking about? |
18:15.41 | delmar | demetrio, what device? |
18:15.59 | demetrio | js_birdboy: maybe, I don't know how it works, since it's not documented in any way |
18:16.08 | FuriousGeorge | i gotta run, anyone wanna recommend an alternative to eyeBeam (with a xfer button) for me? |
18:16.44 | demetrio | so if I set the "t" option in dial, the dialled phone will be able to transfer a call to exten simply pressing #exten ? |
18:16.50 | js_birdboy | on your sip device make sure your tones are set to allow DTMF (I think I got the abbreviation right) |
18:17.41 | js_birdboy | are you talkig about the t extension? That is used for response/digit timeout |
18:17.45 | demetrio | ok, but can you please confirm that what I've just said is correct? |
18:18.01 | delmar | actually there is T and there is t. |
18:18.05 | demetrio | well.. the t option shoud be "allow the called person to transfer the call" |
18:19.06 | demetrio | nevermind, I have to go now |
18:19.12 | delmar | doh |
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18:22.37 | *** join/#asterisk diazonin (i=diazonin@114.222.8.67.cfl.res.rr.com) |
18:22.40 | diazonin | hey folsk |
18:22.51 | diazonin | I like asterisk, buts kind of hard to configure |
18:23.01 | diazonin | got it running on a Debian dual ppro 200, |
18:23.07 | diazonin | wonder if that is fast enough or not |
18:23.36 | Jake1932-mobile | One sure way to find out |
18:24.09 | diazonin | what kind of Asterisk configuration tools do folks use? |
18:24.17 | diazonin | a perl script or what |
18:24.23 | Jake1932-mobile | Vi and nano |
18:24.28 | IronHelix | nano and notepad |
18:24.30 | diazonin | I figured as such |
18:24.49 | diazonin | you folks use it actively? |
18:24.57 | Jake1932-mobile | Yes |
18:25.04 | diazonin | what do you think of it |
18:25.13 | Jake1932-mobile | Love it |
18:25.18 | IronHelix | asterisk? |
18:25.24 | diazonin | yeah, asterisk |
18:25.34 | IronHelix | asterisk is quite possibly the best thing since sliced bread. |
18:25.38 | IronHelix | actually scratch that |
18:25.41 | diazonin | you guys use softphones or hardphones and who do you all call with it |
18:25.43 | IronHelix | asterisk is better than sliced bread |
18:25.52 | Jake1932-mobile | Hardphones |
18:25.53 | Lostfrog | nano? eww. |
18:25.59 | docelmo | NANO RULES! |
18:26.02 | diazonin | what is your preferred hardphone |
18:26.12 | Lostfrog | Might as well use dosemu and edit |
18:26.12 | docelmo | If one can afford it.. Cisco |
18:26.13 | diazonin | like one of those good looking ciscos or a cheaper brand |
18:26.16 | Jake1932-mobile | 7960 cisco |
18:26.18 | IronHelix | SNOM phones are nice, polycom is nice too but good luck getting support |
18:26.32 | diazonin | we have those ciscos at my school, NICE |
18:26.36 | IronHelix | personally i use a grandstream 2000, its cheap, it works, no major complaints |
18:26.45 | docelmo | Iron! |
18:26.51 | IronHelix | docelmo! |
18:26.52 | diazonin | what protocol you all use, SIP? |
18:27.03 | docelmo | Dude.. Have you had any issues with them loosing registration after 1 day or something? |
18:27.11 | IronHelix | SIP for connecting to phones and providers if the use it, otherwise i try to use IAX whenever possible |
18:27.13 | Lostfrog | That is a religious war, diazonin. Mostly SIP or IAX |
18:27.19 | *** join/#asterisk Tobias76 (n=tk@01230.info) |
18:27.20 | IronHelix | yeah |
18:27.26 | diazonin | my bad, just wondering what I should use |
18:27.27 | Tobias76 | hi |
18:27.33 | IronHelix | docelmo- never had any reg problems, but they are all on a LAN |
18:27.42 | IronHelix | no worries diaz |
18:27.47 | docelmo | Are you DHCP or Static? |
18:27.56 | IronHelix | static mapped dhcp |
18:28.00 | diazonin | how do you like using the phones over the internet, any problems with WAN use? |
18:28.21 | diazonin | I had vonage for a while, but just using an ATA they provided not using Asterisk, it was ok, but had some problems |
18:28.27 | IronHelix | config issues can take some working otu, especially when there is NAT involved but overall once configured correctly they are very reliable |
18:28.36 | Jake1932-mobile | I use it everyday for hours at a time over the internet - works fine |
18:28.46 | IronHelix | diaz try broadvoice or quantumvoice both will support asterisk |
18:28.57 | diazonin | what are the rates like? |
18:29.12 | IronHelix | both sell 'lines', about $20/mo for unlimited us/canada |
18:29.14 | docelmo | Iron what are your peers setup like? Can I see one to compair? I am loosing my after about 24/48 hours then they will not reregister |
18:29.24 | diazonin | thanks for the advice folks |
18:29.26 | IronHelix | docelmo- same LAN? |
18:29.30 | docelmo | yep |
18:29.38 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
18:29.48 | IronHelix | diaz- if you want to buy minutes wholesale, there are a great number of companies that will sell you minutes for 1-3 us cents each |
18:30.02 | Jake1932-mobile | 2c a minute with asterlink |
18:30.30 | IronHelix | also diaz check out http://www.voip-info.org anything and everything about voip can be found there |
18:30.36 | IronHelix | doc- sure stand by |
18:31.58 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
18:31.58 | kuku5 | was anyone able to change the ring on a gxp 2000? |
18:32.11 | IronHelix | yeah |
18:32.12 | docelmo | no |
18:32.13 | IronHelix | its easy |
18:32.19 | kuku5 | ergh |
18:32.23 | IronHelix | use their crappy ringtone generator tool |
18:32.25 | kuku5 | can you send me a tone thats ready ? |
18:32.36 | IronHelix | if you have the latest firmware 1.0.1.9 or 1.0.1.12 it will work |
18:33.23 | IronHelix | sure, stand by |
18:38.33 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
18:38.40 | IronHelix | kuku check dcc send |
18:40.36 | *** join/#asterisk ronn (n=ronn@62-249-247-240.no-dns-yet.enta.net) |
18:42.13 | *** join/#asterisk rubble (n=rubble@wbs-146-186-81.telkomadsl.co.za) |
18:43.49 | ronn | is there anyway to stop asterisk from generating ringing tone? even when not using the r optin in Dial() ? |
18:44.14 | docelmo | No.. if no r is present then your getting progress from the endpoint |
18:44.20 | ronn | i have the following Dial(Zap/g1/${EXTEN}|60|g||300) but still * generates ringing |
18:45.20 | IronHelix | m? makes music on hold |
18:45.20 | ronn | docelmo: with the above command i'm getting ringing both from * and the end point |
18:45.36 | docelmo | No.. Just the PSTN |
18:45.49 | hardwire | ok |
18:46.00 | hardwire | would I be retarted to just lock down production to 1.2.0-beta2 |
18:46.11 | hardwire | and gradually phase it up as betas are released.. vs CVS weekly? |
18:46.21 | ronn | can u force asterisk from generating the ringing tone? |
18:46.28 | docelmo | yes |
18:46.29 | docelmo | add the r |
18:46.51 | hardwire | ronn: from using it.. play a silence MoH? |
18:47.32 | ronn | hardwire: how do i do that? |
18:48.05 | ronn | docelmo: i'm getting the ringing with and without the r option. |
18:48.30 | docelmo | the PSTN PSTN PSTN is giving you progress which is RING in telco terms |
18:48.31 | hardwire | ronn: read up on the musiconhold.conf |
18:48.34 | hardwire | and cmd dial |
18:48.58 | hardwire | docelmo: taking the teletubbies learning approach? |
18:49.09 | ronn | thanks |
18:49.18 | docelmo | if it helps him understand.. |
18:49.31 | IronHelix | lol |
18:49.39 | hardwire | thats so mean |
18:49.51 | IronHelix | ouch |
18:50.00 | hardwire | ugh |
18:50.10 | hardwire | I hate it when I become a brown noser due to the hole at the top of coffee lids |
18:50.26 | IronHelix | easy to fix |
18:50.35 | IronHelix | drink your coffee thru one of those stirrer straws |
18:50.38 | IronHelix | the tiny thin ones |
18:50.43 | IronHelix | cools it too on the way up |
18:50.54 | hardwire | IronHelix: Ah.. thats how I get it into my eye. |
18:50.59 | hardwire | I avoid if at all possible |
18:51.33 | IronHelix | hehehe |
18:51.34 | hardwire | the elastic suction vortex I create when wolfing down coffee tends to keep the flow going well after I have released the straw from my lops |
18:51.46 | hardwire | lops/lips |
18:52.01 | Beirdo | straw? |
18:52.09 | IronHelix | yeah stirrer is not recommended if you drink coffee fast |
18:52.11 | Beirdo | wow |
18:52.26 | hardwire | ok |
18:52.27 | hardwire | I don't care |
18:52.30 | Beirdo | I punch a second hole in the lid to get airflow |
18:52.35 | hardwire | I am using digitally imported sky.fm streams on my music on hold |
18:52.39 | *** join/#asterisk svenna_ (n=svenna@p548D2D1E.dip0.t-ipconnect.de) |
18:52.45 | Beirdo | a venti latte will be drained in about 1 min |
18:52.59 | hardwire | Beirdo: thats my original problem.. the tilt required to wolf at such speeds quickly sends coffee out the upper hole onto my nose |
18:53.07 | Beirdo | heheh |
18:53.11 | Beirdo | true enough |
18:53.13 | Beirdo | hmmm |
18:53.18 | hardwire | there must be a better way |
18:53.29 | Beirdo | could drink it like college students drink beer |
18:53.31 | hardwire | to avoid such coffee injuries as in the eye via straw.. and embarassing brown noser moments. |
18:53.35 | hardwire | Beirdo: hahahaha |
18:53.36 | Beirdo | coffee bong time |
18:53.43 | hardwire | JUST MIX IT AS IT GOES DOWN! |
18:54.28 | hardwire | a new coffee transport system will still however never address the coffe out of the nose at a humorous slashdot post delimma |
18:54.36 | Beirdo | true |
18:54.39 | Beirdo | I got it. |
18:54.40 | Beirdo | IV |
18:54.47 | IronHelix | hehehe |
18:54.50 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
18:54.54 | hardwire | who needs coffee then.. extract caffeine and plug in |
18:55.01 | *** join/#asterisk moles (n=martin@85.194.2.201) |
18:55.57 | hardwire | ok.. this channel is putting me to sleep |
18:56.28 | Beirdo | hehe |
18:56.59 | hardwire | I really want to get some radio stations as the MoH here |
18:57.01 | hardwire | not a problem |
18:57.08 | *** join/#asterisk juanjoc (n=jcomella@OL48-53.fibertel.com.ar) |
18:57.10 | hardwire | I just need to get my radioshark a linux driver |
18:57.29 | hardwire | or find a rackmount tuner :) |
18:57.49 | *** part/#asterisk moles (n=martin@85.194.2.201) |
18:59.57 | hardwire | so.. I need to use rec to go to sox to resample the souund from say 22khz to 8khz with its quadratic resample filter |
19:01.54 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
19:01.56 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
19:04.18 | *** join/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
19:04.22 | MuppetMaster | Hello. |
19:04.30 | hardwire | herro |
19:04.30 | Flauto | hello |
19:04.48 | MuppetMaster | When using the Async setting in the Manager API Originate action, how does one tie the call results to the original action requests? |
19:05.22 | MuppetMaster | Does the ActionID return for every response/event to an originate action? |
19:06.44 | jalsot | hi |
19:07.06 | *** part/#asterisk MuppetMaster (n=MuppetMa@81.184.73.169) |
19:07.18 | jalsot | how can I use res_features options [like atxfer] on a channel called from queue to an agent? |
19:08.08 | jalsot | when I set mapping for #1, I receive them, however they are ignored, no atxfer applied |
19:08.13 | jalsot | any idea? |
19:09.18 | *** join/#asterisk santiago (n=santiago@208.195.215.124) |
19:17.06 | justinu | is manx here? |
19:17.52 | hardwire | rawr |
19:18.22 | justinu | muppetmsater: try asking on #asterisk-dev |
19:18.39 | justinu | they're like smart and stuff |
19:21.08 | Lostfrog | Has anyone had a problem with the hookswitch on snom 3x0s? |
19:21.37 | Lostfrog | I just plugged in two of them in a row in which the hookswitch doesn't work. |
19:21.54 | Lostfrog | They're from two different boxes from the factory. |
19:22.46 | *** join/#asterisk algorithmn (n=na@ool-44c29ac5.dyn.optonline.net) |
19:23.55 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
19:24.53 | iCEBrkr | Hey, if I put a callfile in outgoing and I'm using the GroupCount() stuff-- and the group is 'full' will that callfile get deleted or will it 'queue' it for late when some ports are free?? |
19:25.09 | algorithmn | my current asterisk/realtime/mysql setup crashes when i have an extension of 900 registered with the system.. it crashes on any outgoing calls.. |
19:25.30 | iCEBrkr | I'm assuming maxretry= should 'retry' the file if no ports available. |
19:26.40 | Damin | twisted? |
19:26.48 | file | is a twisted individual |
19:27.14 | iCEBrkr | file: I think we're all a bit twisted. |
19:27.22 | Lostfrog | A bit? |
19:27.23 | Lostfrog | lol |
19:27.24 | iCEBrkr | file: Especially that Damin guy. |
19:27.56 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/files/NACS_NET-Fake_Jerky_Boys.mp3 |
19:29.09 | Lostfrog | No help on broken hookswitches on Snom 320s? |
19:29.26 | *** join/#asterisk javo (i=1000@144.13.53.210) |
19:31.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
19:31.19 | iCEBrkr | W00t! |
19:31.34 | iCEBrkr | I'll just put some insane amount of retry times in here to make sure the call is made. |
19:32.09 | *** join/#asterisk FuRR_ (n=meeps@bko29.chapman.edu) |
19:32.46 | javo | I have asterisk 1.0.9, and it will run just fine. Problems arise when I try to use zaptel. If chan_zap exists, asterisk will not load. when i try to modprobe wcfxo, i get Unable to open master device '/dev/zap/ctl' |
19:33.00 | javo | i have udev, and there is no /dev/zap at all |
19:33.11 | Lostfrog | ninety-nine million, nine hundred and ninety-nine thousand, nine hundred and ninety-nine retries. |
19:33.32 | Lostfrog | modprobe zaptel? |
19:33.41 | javo | nope, zaptel is loaded |
19:33.59 | KranZ | javo, what version of kernel? |
19:34.05 | javo | they both show as loaded, actually |
19:34.06 | iCEBrkr | Lostfrog: Yes. |
19:34.06 | javo | 2.6.11 |
19:34.32 | KranZ | you got the latest cvs? |
19:34.44 | javo | i read about needing rules in /etc/udev/rules.d but they are already there |
19:35.05 | javo | latest cvs of *? |
19:35.08 | KranZ | what does dmsg say when u try to load it |
19:35.10 | Lostfrog | ok.. I'm not running udev anywhere.. can't help. |
19:35.10 | KranZ | yeah |
19:35.18 | javo | no i have stable 1.0.9 |
19:35.34 | javo | wcfxo: DAA mode is 'FCC' |
19:35.34 | javo | Found a Wildcard FXO: Generic Clone |
19:35.38 | KranZ | make sure you have 1.0.9.2 zaptel |
19:35.43 | javo | i do |
19:35.44 | KranZ | they fixed a compile bug |
19:35.47 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
19:36.02 | ms345 | can anyone suggest why this doesn't set my outbound callerID? from pri debug span: Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) '4044257828' ] |
19:36.29 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
19:36.29 | javo | zaptel-1.0.9.2 |
19:36.40 | ms345 | but it shows the base number of my pri, not 7828 |
19:37.18 | KranZ | your telco might not allow the number you're trying to set |
19:37.28 | javo | it seems that the module loads fine, but it doesn't give me anything in /dev |
19:37.37 | *** join/#asterisk makhtar (n=ageller@mail.bulletinnews.com) |
19:37.38 | ms345 | i was able to set it with my nec pbx (now behind *) |
19:37.58 | *** join/#asterisk aetg_bob (n=aetg_bob@70-35-141-64.ironoh.adelphia.net) |
19:37.59 | KranZ | javo, u googled? |
19:38.02 | javo | ya |
19:38.06 | ms345 | i tried 11 digits, 10 digits, 7 digits and 4 digits - all the same |
19:38.09 | KranZ | ms345: what method u using to set the callerid? |
19:38.23 | javo | all i can find is stuff about /etc/udev/rules.d |
19:38.31 | javo | which appears to be correct on my system |
19:38.40 | ms345 | Executing SetCallerID("Zap/92-1", "4044257828") in new stack |
19:38.59 | *** part/#asterisk aetg_bob (n=aetg_bob@70-35-141-64.ironoh.adelphia.net) |
19:39.08 | KranZ | set the ani |
19:39.17 | KranZ | with the ,a flag |
19:39.39 | KranZ | or set(callerid(ani) = 4044257828) |
19:39.50 | *** join/#asterisk Corydon-w (i=red@pdpc/supporter/sustaining/Corydon76-home) |
19:40.15 | ms345 | KranZ - will try it - thx a bunch! |
19:40.38 | *** join/#asterisk Math` (n=math@modemcable157.3-81-70.mc.videotron.ca) |
19:42.21 | KranZ | ms345: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid |
19:42.59 | brettnem | Hey anyone know why I can't get asterisk to STOP sending 407s to SER? It looks like asterisk is trying to authenticate the original caller who placed the call TO ser. I'm using insecure=yes |
19:43.03 | Math` | I got * setted up on a router with multiple internet connections, is there any way to tell asterisk to bind to a certain ip for *OUTGOING* connections? |
19:43.13 | *** part/#asterisk makhtar (n=ageller@mail.bulletinnews.com) |
19:43.53 | Math` | that is for sip, bindaddr= listens on that ip (and I still need to listen on the lan, even if Im using an external interface to register to an sip server |
19:44.36 | Math` | (or would the best way to do it be iptables?) |
19:44.46 | *** join/#asterisk lmergen (n=Leon@grib.btcnet.nl) |
19:44.56 | brettnem | ugh.. i can't get rid of this 407 :( |
19:45.00 | lmergen | hello, i have a question |
19:45.27 | KranZ | brettnem: are calls coming through * to ser, or ser to * |
19:46.05 | lmergen | at http://www.voip-info.org/wiki-Asterisk+auto-dial+out , it says i can use the "Data: " parameter in /var/spool/asterisk/outgoing files to pass data to the application - first of all, is this "application" Asterisk, and if so, how can I read that variable ? At the "Asterisk variables" page, nothing can be found about a "Data: " parameter |
19:47.19 | Math` | lmergen: I usually pass data as an extension to a context, tho this depends on what you want to pass |
19:48.41 | *** join/#asterisk pun` (n=jeod20@adsl-067-034-127-246.sip.mia.bellsouth.net) |
19:49.06 | pun` | hi |
19:49.26 | file | brettnem: I bet it's matching a user entry in sip.conf and asterisk is challenging :) |
19:49.57 | *** join/#asterisk Darwin35 (n=richard@208.139.193.178) |
19:50.22 | pun` | hey guys, any wifi iax phone available yet? |
19:50.22 | Darwin35 | ok who here has asterisk on fbsd and has a /usr/local/etc/rc.d asterisk.sh they can show |
19:50.42 | Darwin35 | I need to see what I am missing |
19:51.31 | Darwin35 | anyone ? |
19:51.33 | Darwin35 | pls |
19:51.38 | Math` | pun`: not that I know |
19:51.47 | Darwin35 | I know someone here is using fbsd |
19:51.48 | Math` | Darwin35: whats the problem your having |
19:51.54 | javo | ok, i just read in README.Linux26 that I need to have CRC-CCITT enabled in the kernel. However, when I go to enable it in make menuconfig, it does not give me an option to select it. It only gives me --- |
19:51.55 | drumkilla | Darwin35: keep spamming the channel, and it might change my mind |
19:51.57 | lmergen | Math`, but that wouldn't allow for variable data, right ? I mean, i have to make dialplans for specific extensions, and can't do a "catch all" dialplan for all extensions, right ? |
19:51.59 | Math` | Im not using fbsd but maybe I can help |
19:52.07 | *** join/#asterisk brookshire (n=nubb@gateway.digium.com) |
19:52.13 | javo | does anyone know what has to be enabled to be able to enable CRC-CCITTZ |
19:52.17 | Darwin35 | drum |
19:52.23 | Darwin35 | I have not been on in awhile |
19:52.35 | Darwin35 | I just need to look at a working asterisk.sh |
19:52.46 | Math` | u tried googled? |
19:52.48 | Math` | google* |
19:53.07 | Darwin35 | you get the asterisk.sh.in file |
19:53.21 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
19:53.22 | Darwin35 | I need to see the changes made to the file |
19:54.18 | Lostfrog | java: modprobe crc32c before loading zaptel and wcfxo |
19:54.53 | Lostfrog | javo, even. |
19:55.07 | Lostfrog | I put crc32c in /etc/modules |
19:55.30 | javo | Lostfrog, i do not have the crc32c module because it will not let me select it in make menuconfig |
19:55.49 | javo | it shows --- instead of < > |
19:56.00 | sahafeez | what should i use in order to record custom greetings, etc.. |
19:56.30 | javo | Lostfrog, do you know what i need to enable before i can enable crc32c? |
19:56.38 | Lostfrog | One moment. |
19:57.31 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
19:57.59 | brettnem | Hey is there anyway to get asterisk to NOT look in the From: header for authentication?! |
19:58.09 | Lostfrog | Did you enable crc32c in Cryptographic options. |
19:58.10 | Lostfrog | ? |
19:58.14 | *** join/#asterisk jonarild (n=vircuser@195.28.172.4) |
19:59.44 | jonarild | someone with experience in zaphfc? |
20:01.11 | Lostfrog | I don't know javo. |
20:01.17 | Lostfrog | I have to run. |
20:01.50 | jonarild | insmod: error inserting 'zaphfc.ko': -1 Invalid module format |
20:01.50 | jonarild | Im getting ths error: insmod: error inserting 'zaphfc.ko': -1 Invalid module format |
20:02.08 | pun` | Math, thanlks for your reply |
20:02.18 | javo | ok thanks anyways |
20:02.23 | pun` | anybody that knows about a wifi iax phone? |
20:03.41 | jonarild | or someone that knows about a SIP wifi phone that supports WPA encryption |
20:03.46 | brettnem | why does asterisk assume I want to authenticate if the from sip uri matches a valid peer? that's insane!! |
20:06.20 | *** join/#asterisk jonarild (n=vircuser@195.28.172.4) |
20:07.50 | azzie | what's the code to pick up an extension which is ringing? using AMP... |
20:12.26 | Madkiss | Uhm, I still need an asterisk developer. |
20:14.45 | *** join/#asterisk mxmasster (n=mxmasste@ppp-71-138-119-74.dsl.irvnca.pacbell.net) |
20:14.47 | mxmasster | hi all |
20:17.00 | mxmasster | i need to configure some time based rules in asterisk - i.e. if it is between 8-6 m-f ring phone a, otherwise ring phone b |
20:17.09 | mxmasster | where can i find an example of how to do that |
20:17.24 | delmar | well that was easy... #1 and *2 for blind and attended xfers works fine. ill just use that and stop fighting with the silly tigernetcrap transfer button :P |
20:18.02 | GXTi | mxmasster: iirc the include directive does that |
20:21.22 | mxmasster | GXTi: you don't happen to have an example i can look at? |
20:22.06 | iCEBrkr | blah blah blah |
20:22.16 | pun` | someone here mentione some time ago that converted a sip wifi phone to a iax |
20:22.20 | pun` | mentioned.. |
20:22.37 | docelmo | so ice you want that server? I now have 30Mb down and 5Mb UP!?!?!?!? YES!!!!!!!!!!! |
20:22.45 | iCEBrkr | LOL |
20:22.52 | iCEBrkr | docelmo: Talk to me next week :P |
20:23.14 | iCEBrkr | docelmo: I'd hate to give you half now, half later.. That's just shady on my part. |
20:23.32 | iCEBrkr | Like I said, I gotta make sure I have $$$ incase something else comes up. |
20:24.24 | docelmo | this is true.. I am moving the boxes down now.. Also I noticed the guy who did the code for X10 on linux is giving up his development position.. I emailed him about it. I am gonna make app_x10.so for asterisk out of it. |
20:24.37 | iCEBrkr | hahah |
20:24.53 | docelmo | What? |
20:25.03 | docelmo | I think its a good idea.. Then you can control it from the dialplan |
20:25.05 | docelmo | :P |
20:25.07 | iCEBrkr | exten s,1,x10(N) |
20:25.26 | iCEBrkr | turn on all devices in housecode |
20:25.36 | tzanger | iCEBrkr: sweet |
20:25.37 | iCEBrkr | I guess that'd be more like.. |
20:25.43 | iCEBrkr | exten s,1,x10(A,N) |
20:25.50 | docelmo | tz, Im the one going it.. Not ice.. haha |
20:25.52 | iCEBrkr | Turn on ALL 'A' devices. |
20:26.04 | iCEBrkr | docelmo: Pff, I gave ya the idea. |
20:26.07 | lmergen | hey... say i have "Channel: Zap/g1/1234567890" in a /var/spool/asterisk/outgoing/file , how can i retrieve the number being dialed inside my extensions.conf ? |
20:26.09 | iCEBrkr | :-| |
20:26.19 | tzanger | lmergen: ? |
20:26.20 | iCEBrkr | docelmo: Make it happen! Make it happen! |
20:26.59 | docelmo | Once he says I can run with it. I am gonna start beta development on it. 2 trees 1 for asterisk and 1 for cLI |
20:27.16 | iCEBrkr | docelmo: You using the bottlerocket code? |
20:27.18 | [TK]D-Fender | docelmo: I have an CM11A package I use for * to control my stuff at home... |
20:27.21 | docelmo | yes |
20:27.24 | iCEBrkr | sweet |
20:27.31 | lmergen | tzanger, i want to fix an auto-dialing script, but in order to execute the Dial() command in dialplan, I need the number to be dialed... how can I retrieve that number ? |
20:27.36 | [TK]D-Fender | if youwant I can send it to youl later |
20:27.38 | docelmo | I also checked.. The bottlerocket will control more than that.. |
20:27.41 | iCEBrkr | docelmo: I should make that a PHP module :P |
20:27.44 | docelmo | ya would like to check it out |
20:27.47 | tzanger | lmergen: you should be able to pick up ${EXTEN} |
20:27.58 | docelmo | I am gonna make it internal to asterisk |
20:28.10 | iCEBrkr | mod_x10 |
20:28.21 | iCEBrkr | No more system() calls!!! |
20:28.40 | [TK]D-Fender | docelmo: It installs a single bin file really and I use System(heyu2 a1 on) for example from my dial plan. is that sufficient for your needs? |
20:28.58 | lmergen | tzanger, but in my dialplan I define "exten => s,1,Dial(Zap/g1/<<number>>,10)" for example... then my $(EXTEN) always is "s" ... or is there some way to define a dialplan for a "catch-all", all numbers ? |
20:29.19 | iCEBrkr | docelmo: Start the Home Automation with Asterisk project! |
20:29.22 | docelmo | I need the source code which is what ice and I are talking about. I am gonna make a app_x10 to control all devices on yoru network from the asterisk dialplan |
20:29.35 | docelmo | Hell ya dude.. Wanna help? |
20:29.38 | iCEBrkr | For sure! |
20:29.43 | iCEBrkr | I still have all my X10 shit |
20:29.49 | iCEBrkr | I know C. |
20:29.56 | iCEBrkr | ok, I can hack away at C |
20:29.58 | docelmo | Im gonna buy some.. I wanna setup my house.. I know of it. Not fluently yet.. |
20:30.33 | tzanger | lmergen: then set a channel var for the call file to pick up |
20:30.55 | lmergen | hmmm k |
20:31.22 | [TK]D-Fender | What we really need for X10 is to help the guys working on the USB one. Thats where the focus should be. I'm still using my old serial one for the time being and won't pay for new stuff until they get it working decent |
20:31.31 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
20:31.40 | lmergen | but say I set a "Channel: Zap/g1/1234", how can I pick up that "1234" in my extensions.conf then ? |
20:31.49 | *** part/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
20:32.07 | iCEBrkr | [TK]D-Fender: Is there any effort for the USB version? |
20:32.18 | docelmo | We can start on that.. USB is treated almost the same as serial |
20:32.26 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
20:32.33 | [TK]D-Fender | iCEBrkr: Yeah, but its rusty right now. they are reverse engineering it since X10 isn't helping much (while not hurting) |
20:32.54 | iCEBrkr | [TK]D-Fender: It should be the same shit as the serial version |
20:32.57 | [TK]D-Fender | Lazy asses..... |
20:33.01 | iCEBrkr | Oh, I guess you have to talk to the USB device. |
20:33.05 | iCEBrkr | hrrm. |
20:33.16 | [TK]D-Fender | iCEBrkr: No they changed quite a bit since it accepts all house/unit codes, and the extended versions. |
20:33.26 | iCEBrkr | and on the serial device you blip the terminal ready line and shit |
20:33.36 | [TK]D-Fender | They should use a model like SIP/HTTP :D |
20:33.55 | Blackthorn | Hello. My * is on a real ip. I have a sipura which is behind a linksys nat router. * is set nat=yes and notify=yes. On the console I get error message that one of the two lines on the sipra won't register. and the ip is 192.168.1.2 |
20:33.57 | iCEBrkr | Hrrm. Mine accepts all house/unit codes. |
20:34.08 | [TK]D-Fender | that way you could telnet it.... in fact, thats the next thing they should do! A networked version! |
20:34.20 | [TK]D-Fender | iCEBrkr: Your serial version? |
20:34.26 | iCEBrkr | [TK]D-Fender: Yeah |
20:34.43 | iCEBrkr | br a on |
20:34.47 | iCEBrkr | br b on |
20:34.49 | docelmo | Well initially we will work with the serial version then move to usb down the road |
20:34.53 | [TK]D-Fender | Well you serial one should only work on one housecode at a time. Thats last I recall from the manual dial on it. |
20:35.31 | docelmo | Well how many house codes would you need for 1 house? |
20:35.36 | iCEBrkr | [TK]D-Fender: I'm using bottlerocket. I specify the house code/unit number on the command line. |
20:35.40 | [TK]D-Fender | Look for the HEYU2 homepage and DL it from there. Works dead simple and should be enough for you to make an app_x10 out of.\ |
20:35.46 | iCEBrkr | docelmo: Um, you can have a lot!! |
20:35.48 | [TK]D-Fender | Google = good |
20:36.08 | iCEBrkr | A-F, 0-9 |
20:36.10 | docelmo | HEYU2 is USB? |
20:36.28 | docelmo | ohh.. Well if 1 at a time then serial should be fine.. But the interface is interesting.. |
20:36.40 | [TK]D-Fender | iCEBrkr: yeah I need a whole whack since I intend to virtualize my codes so I can use the slim-line switches for custom controls remapped by * for me. |
20:37.08 | [TK]D-Fender | HEYU2 is serial for the older controller. I don't have any links for the USB one yet since I don't own it yet. |
20:37.27 | delmar | ok i have had little sleep so i might not be thinking streight.. anyway.. looking at attended and blind xfers using Dial(Tt) ...I want to have extensions to have both T and t, so that a call received OR made can be transferred etc...but doing this seems to make it that say.... an incoming PSTN caller could transfer themselves... what can i do about that. |
20:37.32 | iCEBrkr | [TK]D-Fender: You have that little serial port 'pass-thru' thing? |
20:37.40 | docelmo | bottlerocket is fine for that then.. I have looked at the code its not very hard to understand. |
20:38.10 | iCEBrkr | docelmo: yeah, it's pretty simple looking |
20:39.06 | iCEBrkr | ${VARIABLES} are available to [macros] right? |
20:39.55 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
20:40.18 | docelmo | Found linux drivers for the CM15A computer interface |
20:40.21 | docelmo | for USB |
20:40.29 | docelmo | Can you say USB app_x10.so? |
20:40.34 | iCEBrkr | :) |
20:40.47 | falz | anyone with cisco 7960's know if there's a way to force the caller ID on transferred calls to be the ORIGINATOR instead of the TRANSFERING PARTY? |
20:40.55 | *** join/#asterisk Flauto (n=zhao@71.194.39.175) |
20:40.56 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
20:41.04 | [TK]D-Fender | docelmo: I found that source too, but never actually tried it. They have some of the basics right now, but a lot is still missing. |
20:41.05 | falz | (as 7960's do with blind xfer) |
20:41.24 | docelmo | ok going to smoke.. brb |
20:41.44 | Nugget | I quit smoking 8 years, 5 months, 4 weeks, 2 days, 1 hour, 41 minutes, and 44 seconds ago. During that time, I would have smoked 68,247 cigarettes. (That's like smoking a 3.23 mile-long cigarette) By quitting, I've saved $11,943.23! I've avoided inhaling 1.77 kg of tar, 109 grams of nicotine, and 1.09 kg of carbon monoxide. |
20:42.31 | [TK]D-Fender | And now have a lot of free time to calculate your savings! |
20:42.50 | [TK]D-Fender | You should run a ticker like the Jerry Lewis Telethon! |
20:43.21 | iCEBrkr | [TK]D-Fender: he has that in a frame hung on his cubical wall. |
20:43.50 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
20:43.56 | Blackthorn | Hello. My * is on a real ip. I have a sipura which is behind a linksys nat router. * is set nat=yes and notify=yes. On the console I get error message that one of the two lines on the sipra won't register. and the ip is 192.168.1.2 |
20:44.02 | [TK]D-Fender | Could have saved a few bucks on the frame and increased his savings ;) Use recycled paper and Scotch tape next time :D |
20:44.12 | *** join/#asterisk vitamintwo (n=vitaminm@wza.us) |
20:44.45 | iCEBrkr | Blackthorn: um ok? |
20:44.50 | vitamintwo | I've got a snom 320, and the programmable keys' status lights are behaving erratically. Anyone have experience with this? |
20:45.25 | Blackthorn | heheh - oops.. I'm guessint it won't register because it is coming up with 192 address. and i'd like to know what I need to to "fix" this isse |
20:45.31 | falz | Blackthorn: if asterisk is on a public IP, why is NAT=yes? |
20:45.44 | falz | so your asterisk box is dual homed? |
20:45.54 | iCEBrkr | falz: I think he meant the sip entries are set to yes. |
20:45.54 | Blackthorn | nat=yes in the respective sip config for that user only. |
20:46.07 | Blackthorn | no not dualhomed and the *box is not nat'ed |
20:46.34 | shido6 | nat=yes should be in the user/peer/friend |
20:47.27 | delmar | uh wtf shido6 ? |
20:47.27 | iCEBrkr | Blackthorn: Your 'Proxy' setting in the Sipura is set to the IP of your asterisk box, right? |
20:47.27 | falz | so is 192.168.1.2 the "outside" ip of the NAT device? |
20:47.27 | Blackthorn | correct ice. |
20:47.37 | docelmo | ok Im back |
20:47.50 | iCEBrkr | Blackthorn: UserID is equal to [whatever] in your sip.conf? |
20:47.51 | delmar | docelmo ... |
20:48.05 | docelmo | yes? |
20:48.08 | delmar | docelmo, do u know anything about using T or t with Dial command etc.. ? |
20:48.18 | docelmo | What do you want to know? |
20:48.23 | docelmo | Its in the wiki.. |
20:48.25 | delmar | ok here is whats happening... |
20:48.32 | shido6 | careful with T and t , you give the caller more access to your dialplan |
20:48.40 | Blackthorn | See what is wierd is that line2 registers using the ip address of the linksys router. line1 fails registration from 192.168.1.2 which is the sipura unit behind the linksys. |
20:48.43 | iCEBrkr | delmar: T and t don't work the way you're thinking.. |
20:48.43 | delmar | sure it is.. but im tired and this might be easily fixed or its just tough shit.. im not sure... |
20:48.55 | delmar | ok sec.. here is whats happening... |
20:49.25 | falz | Blackthorn: asterisk shouldnt know 192.168.1.2 exists if it really is NAT'd though |
20:49.32 | delmar | Extensions are setup with T and t .. so they can transfer the call made.. OR received... |
20:49.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
20:49.49 | iCEBrkr | Blackthorn: Proxy should equal the public IP of your asterisk box. |
20:50.11 | iCEBrkr | delmar: Bzzzt |
20:50.14 | Blackthorn | falz: I agree. Icebbkr: yes proxy on both lines are set to the * box. |
20:50.20 | delmar | I have a line to take care of incomign calls... with just t which calls other extensions.... when one answers... it can transfer.. and the caller on the PSTN.. cannot transfer... |
20:50.31 | delmar | iCEBrkr let me finish :P |
20:50.46 | delmar | but when the extension transfers the PSTN caller to another extension.... |
20:51.09 | delmar | at that point... the PSTN caller OR the new extension.. can make a transfer.. with *2 or #1 |
20:51.17 | iCEBrkr | delmar: T allows any caller transfer the call, which is something you really don't want. |
20:51.21 | delmar | now using Tt on extensions works perfectly... |
20:51.23 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
20:51.30 | delmar | iCEBrkr, exactly... |
20:51.52 | delmar | iCEBrkr i want this to be available to the extensions.. but not an outside caller... mind you... |
20:51.54 | iCEBrkr | delmar: So you want anyone to be able to hit * and arbitrally transfer themselves?? |
20:52.03 | delmar | what can they do... make a transfer and get disconnected.. LOL |
20:52.21 | delmar | iCEBrkr, err ok it works like this.... |
20:52.25 | iCEBrkr | delmar: or accidentally hit * and got WTF? |
20:52.33 | iCEBrkr | s/got/go |
20:52.50 | delmar | iCEBrkr, no its the default of *2 attended or #1 blind right now... so not simply * |
20:53.04 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
20:53.09 | lesouvage | docelmo: I have some strange problem downloading phpagi version 2.14 from sourceforge. It's not the full size (21 kb instead of 35kb) and I can't untar the archive. s there anither url where I can download phpagi? |
20:53.09 | iCEBrkr | delmar: Is that something new? |
20:53.15 | iCEBrkr | delmar: cuz it's always been * |
20:53.32 | docelmo | Download from CVS |
20:53.59 | iCEBrkr | phpagi sucks! :D |
20:54.09 | vitamintwo | So is anyone experienced with snom 320 wonkiness? |
20:54.12 | delmar | iCEBrkr, the default in the features.conf is *2 and #1 i believe... |
20:54.26 | delmar | iCEBrkr, ok just checked the sample.. and there isnt even anything in that.. so its whatever u like :P |
20:54.33 | delmar | anyway... that asside... |
20:54.39 | docelmo | ice eat me |
20:54.42 | iCEBrkr | docelmo: hehe |
20:54.59 | iCEBrkr | docelmo: Don't make me go through that thing and clean it ALL up! |
20:55.08 | iCEBrkr | Since I'm a PHP nut, I just might have to. |
20:55.18 | docelmo | Do what you want.. I dont care.. I only did a few parts in there |
20:55.26 | iCEBrkr | Sweet |
20:56.03 | delmar | iCEBrkr, lets say extension1 makes a call via the PSTN, then that called party wants to speak to the boss on extension2... :P |
20:56.09 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
20:56.24 | Darwin35 | exit |
20:56.44 | delmar | iCEBrkr having t only means u can transfer the call that is received... having T means u can also transfer the call you placed.... so extensions need to do both... |
20:56.56 | delmar | iCEBrkr, or am i looking at it all wrong? how can i overcome this? |
20:57.02 | *** join/#asterisk jarrod (i=jarrod@juniperyour.net) |
20:57.16 | delmar | <--- not enuf sleep |
20:57.21 | jarrod | where is cripito |
21:04.45 | brettnem | hey anyone know how to add a sip header on a call that originates on a PRI and goes OUT SIP? |
21:05.23 | brettnem | aparently sip_addheader only works on calls that originate SIP |
21:07.13 | mxmasster | what the dial string to ring multiple lines at the same time? |
21:07.53 | falz | Dial(SIP/103&SIP/203,8) |
21:08.01 | falz | that'll dial x 103 and 203 for 8 seconds |
21:08.18 | mxmasster | thanks |
21:10.21 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
21:11.34 | mxmasster | how do i answer a call, and play music while i ring extensions? |
21:12.10 | file[laptop] | mxmasster: show application dial, type that on the CLI and read all the available options |
21:12.10 | falz | while on hold? |
21:12.47 | mxmasster | well, maybe hold isn't the right word |
21:13.04 | mxmasster | i want to play music to the caller while the phones are ringing |
21:13.13 | file[laptop] | mxmasster: do what I said, there's an option to do it |
21:13.16 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-11-9.w81-248.abo.wanadoo.fr) |
21:13.56 | file[laptop] | 'm[(class)]' -- provide hold music to the calling party until answered (optionally with the specified class.) |
21:14.14 | mxmasster | thanks |
21:14.17 | mxmasster | just reading that |
21:22.42 | *** join/#asterisk mflorell (n=mattf@rrcs-24-173-158-34.se.biz.rr.com) |
21:24.08 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
21:24.35 | *** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3) |
21:24.50 | iCEBrkr | delmar: I'd make sure your inbound and outbound contexts are separate so you can put T on outbound from internal extensions. |
21:25.15 | iCEBrkr | delmar: ...and I've never gotten attended transfers to work. |
21:25.42 | delmar | iCEBrkr, ok ill take a look at that and think about it... |
21:25.51 | delmar | iCEBrkr, working mint here. |
21:26.12 | *** join/#asterisk daddy_23 (n=daddy@200.93.234.1) |
21:26.17 | iCEBrkr | delmar: Which version are you using again? |
21:26.46 | delmar | to be honest.. its been a while since i updated.... and i have no idea :P |
21:26.48 | delmar | sec. |
21:26.52 | iCEBrkr | show version |
21:27.21 | *** join/#asterisk _DAW (n=bob@adsl-156-91-219.msy.bellsouth.net) |
21:27.22 | delmar | Asterisk CVS-HEAD-04/14/05-17:26:18 |
21:27.41 | _DAW | Hello |
21:27.52 | delmar | time for an update for sure.. but nothing much is b0rked atm. |
21:28.02 | delmar | so no real need. |
21:28.22 | delmar | i might update if i get too frustrated with that * to * via IAX echo issue |
21:29.05 | _DAW | Does anyone know if it is possible to pass args to a macro via the M option in the Dial command? |
21:30.44 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
21:30.59 | sahafeez | question: what app would people recommend to record greetings (not for voice mail but the for the menus). can I use some function in asterisk? |
21:31.14 | iCEBrkr | delmar: F Bleeding-edge. |
21:31.19 | _DAW | sahafeez - record command |
21:31.23 | iCEBrkr | delmar: My 'production' machine was 1.0.9 |
21:31.30 | sahafeez | _DAW where is that? |
21:31.54 | iCEBrkr | _DAW: um.. what?? |
21:31.55 | _DAW | http://www.voip-info.org/wiki-Asterisk+cmd+record |
21:32.22 | iCEBrkr | _DAW: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
21:32.25 | sahafeez | _DAW: thank you. reading now. |
21:32.33 | delmar | iCEBrkr yeah, u gotta stay a bit behidn the times sometimes.. to stay stable and reliable etc. |
21:32.49 | iCEBrkr | delmar: I JUST upgraded to 1.2.0 Beta2 last night |
21:33.03 | iCEBrkr | It wasn't as smooth as I thought it'd be. |
21:33.06 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
21:33.11 | _DAW | iCEBrkr - for the M option you mean? I read up on it there but there is no mention of it passing args like the Macro command. |
21:33.12 | iCEBrkr | But whatever, I got it all working |
21:33.14 | delmar | iCEBrkr, i was gonna do that on my mess about box here that i just showed u the version... |
21:33.22 | iCEBrkr | _DAW: Re-read, it's there. |
21:33.22 | delmar | iCEBrkr, hows the beta running.. any good? |
21:33.39 | delmar | iCEBrkr, does it have rx_fax / tx_fax support? |
21:33.45 | iCEBrkr | delmar: Works like a charm. I don't do faxing, so I dunno |
21:34.00 | delmar | iCEBrkr, yeah i wanna play with receiving fax's next |
21:34.10 | iCEBrkr | _DAW: You can now add args to the macro by using a '^' char |
21:34.20 | delmar | iCEBrkr, last time i tried to setup a macro and stuff it was a pain and didnt work so i gave up but ill have another crak |
21:34.36 | iCEBrkr | delmar: <3 Macros |
21:34.49 | _DAW | iCEBrkr - just saw it. should be with the M command documentation not all over the page.. |
21:35.15 | iCEBrkr | _DAW: Um, RTFM, K/THX/BYE |
21:35.21 | *** join/#asterisk Corydon-w (i=grey@pdpc/supporter/sustaining/Corydon76-home) |
21:35.38 | iCEBrkr | All I gotta say is.. *I* found it.. and I'm half-retarded. |
21:35.40 | mxmasster | is there a place i can post my dialplan for help - i am trying to do a time of day include and then dial extensions based on that but it is not happy |
21:35.51 | iCEBrkr | mxmasster: http://pastebin.ca |
21:35.54 | *** join/#asterisk marc324 (n=marc3234@206-248-157-247.dsl.teksavvy.com) |
21:36.58 | mxmasster | http://pastebin.ca/27641 |
21:37.08 | mxmasster | i am missing something simple with this i am sure |
21:38.15 | jarrod | is cripi on? |
21:39.21 | iCEBrkr | mxmasster: Where'd you find the syntax for including by date/time? |
21:39.29 | mxmasster | voip-info |
21:39.31 | iCEBrkr | mxmasster: I don't see it in the wiki |
21:39.38 | mxmasster | hold on |
21:39.40 | mxmasster | i will give link |
21:40.04 | iCEBrkr | Personally, I'd do a GotoIfTime() and set up the different 'includes' in Contexts |
21:40.04 | mxmasster | http://www.voip-info.org/wiki/index.php?page=Asterisk%20tips%20openhours |
21:40.29 | mxmasster | iCEBrkr: the time works fine |
21:40.36 | brookshire | prank calls #1 |
21:40.38 | mxmasster | the problem is that it won't dial my internal extensions |
21:40.39 | brookshire | :( |
21:41.04 | iCEBrkr | mxmasster: adding those includes breaks your internal dialing? |
21:41.20 | mxmasster | no, when the user presses 1 for support |
21:41.31 | mxmasster | it works correctly to the date/time include |
21:41.43 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
21:41.43 | mxmasster | but the first command has the dial sip/3874&... |
21:41.49 | mxmasster | it cannot connect to the SIP extensions |
21:41.57 | mxmasster | so it fails and starts calling the external numbers |
21:42.11 | mxmasster | but those extensions work properly off of the main context |
21:42.21 | jarrod | how can i collect a key sequence in the middle of a call in order to have asterisk start recording a conversation? |
21:43.04 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
21:43.45 | pooh_ | jarrod: you can not |
21:44.47 | iCEBrkr | jarrod: supposedly you can use *1 with the 'w' command in Dial() |
21:45.26 | mxmasster | iCEBrkr: what do you think? |
21:46.05 | iCEBrkr | mxmasster: I'm really not sure, the logic is a bit confusing to me and I haven't tinkered with that stuff at all. |
21:46.17 | iCEBrkr | So unless I actually try it. I Dunno |
21:46.21 | docelmo | yay! |
21:46.27 | mxmasster | iCEBrkr: you said using gotoiftime |
21:46.32 | mxmasster | can you show me an example |
21:46.33 | iCEBrkr | docelmo: it's not 5 yet, you can't be YAYing |
21:46.54 | iCEBrkr | mxmasster: GotoIfTime(<times>|<weekdays>|<mdays>|<months>?[context|]extension|]pri) |
21:47.05 | docelmo | I am leaving now.. I can yay all I want. |
21:47.10 | iCEBrkr | Yea, me too |
21:47.15 | iCEBrkr | I just printed my timesheet. |
21:47.18 | iCEBrkr | BEEER O'CLOCK! |
21:47.31 | iCEBrkr | `(_8_(|)===> MmmmmmmmmmmBeeeeeeeeeeerrrrrrr! |
21:48.47 | docelmo | eww.. |
21:48.54 | docelmo | ok on that note.. I am going home.. CYA! |
21:50.15 | delmar | iCEBrkr, ok i got sidetracked... so i need to look at incoming and outgoing contexts for the Tt thing... ok... |
21:51.25 | brettnem | hmmm.. nice homer thre |
21:51.51 | delmar | iCEBrkr, i do have separate contexts for as much as possible.. ie incomingfxo outgoingfxo blah blah |
21:52.51 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
21:53.07 | mxmasster | how do i echo from te dialplan to the console |
21:53.36 | Corydon-w | mxmasster: show application verbose |
21:55.45 | mxmasster | instead of having gotoiftime switch to a different context, is it possible to have it change to a different priority? |
21:58.31 | falz | yea, drop the context |
22:00.32 | Chuji | Is there any reason that my Sip conversations would get jittery when monitor is on, but fine when it's off? |
22:00.39 | Chuji | Sip to PSTN |
22:00.51 | Chuji | CPU load is low |
22:02.26 | kuku5 | how do i do a blind transfer |
22:03.20 | Damin | Anyone used AudioCodes Gateways w/ Asterisk? |
22:04.06 | Inv_arp | Chuji: prob a cheap power supply |
22:04.42 | Chuji | Inv_arp : Power Supply? |
22:04.45 | Chuji | Huh? |
22:05.18 | Inv_arp | Chuji: are you saying your pc monitor? |
22:05.54 | Chuji | Inv_arp : No app_monitor |
22:06.01 | Inv_arp | lol |
22:06.03 | Chuji | ie when I'm monitoring calls |
22:06.34 | Chuji | technically, res_monitor |
22:07.10 | justinu | lol, wtf? |
22:07.10 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
22:07.14 | Chuji | Maybe I should try Muxmon |
22:11.44 | kuku5 | how do i do a blind transfer |
22:11.48 | kuku5 | ona gxp 2000 |
22:12.03 | sahafeez | queston about dial plans |
22:12.40 | sahafeez | if i have exten => 2004,1,Backgound(closed) ; play msg saiding enter extension |
22:13.14 | sahafeez | whats the next line. all the examples have the next stuff with a list of exten => dial() |
22:13.30 | lesouvage | docelmo: It's not working yet. Will you please take a look at http://pastebin.com/417591 |
22:13.37 | sahafeez | i do not want a list, i just someone to beable to type the exten that they know and go there |
22:15.07 | sahafeez | what am i missing? |
22:15.50 | justinu | next line would be exten => <digitsentered>,1,Dial(SIP/whoever) |
22:16.11 | Aughey | he wants exten => 2004,2,WaitExten |
22:16.24 | sahafeez | ah, thats it. and then? |
22:16.33 | sahafeez | will it transfer? |
22:16.47 | Aughey | if you dial an extension, it will go to whatever extension you had dialed |
22:16.59 | sahafeez | ok. can I add a time out? |
22:17.18 | Aughey | yes |
22:17.24 | Aughey | show application WaitExten or read the docs |
22:17.26 | justinu | ah |
22:17.45 | sahafeez | Aughey: thanks that is what i was looking for. i will go and read now.. |
22:18.08 | Aughey | its in the sample extensions.conf too |
22:18.10 | sahafeez | Aughey: all the examples so what justinu said so... |
22:18.39 | sahafeez | Aughey: I should look at that. i started with blank files for each *.conf and went from the book, wiki. |
22:20.12 | *** join/#asterisk pooh_ (n=hfwang@cust.15.241.adsl.cistron.nl) |
22:20.27 | Aughey | you'd usually do a WaitExten(15) and then Goto(whatever step plays the extern-ext-of-person line) to have it loop the prompt |
22:20.52 | sahafeez | Aughey: i just want it to dail the exten. thats it |
22:21.33 | Aughey | right, but if they don't dial within 15 seconds, you might want to play the prompt again |
22:22.03 | sahafeez | ok. one sec. let me work out the seq.. |
22:23.27 | marc324 | can someone help with SIPP? |
22:24.15 | sahafeez | exten => 2004,1,Background(afterhoursmsg) ; 1 for dir by 1st name, 2 for last |
22:24.18 | sahafeez | exten => s,n,WaitExten(15) |
22:24.21 | sahafeez | exten => 1,1,Goto(2005,1) |
22:24.21 | Aughey | Survey of Income and Program Participation? |
22:24.23 | sahafeez | exten => 2,1,Goto(2006,2) |
22:25.27 | Aughey | doesn't work, does it? |
22:25.40 | sahafeez | well besided from the typos |
22:26.21 | sahafeez | no |
22:27.12 | justinu | ~sipp |
22:27.14 | jbot | Single In-Line Pin Package: The last "standard" PC RAM configuration before they started making SIMMsA lot like SIMMs, but they have little pins instead of contacts. SIPPs are to VLB what SIMMs are to PCI.. A suicide tool for geeks |
22:27.32 | justinu | lol |
22:27.42 | sahafeez | Aughey: got that out of the examples |
22:27.47 | *** join/#asterisk mxmasster (n=mxmasste@ppp-71-138-119-74.dsl.irvnca.pacbell.net) |
22:28.15 | marc324 | why is ./sipp -ap returns "invalid argument -ap" |
22:28.36 | kuku5 | how do i pickup a parked call?' |
22:28.45 | Aughey | dial the extension |
22:29.12 | sahafeez | kuku5: for the parking lot you setup |
22:29.23 | sahafeez | when you part a call if should tell you |
22:29.30 | sahafeez | http://www.voip-info.org/wiki-Asterisk+call+parking |
22:30.57 | Aughey | sahafeez: if you need more help, pastebin your extensions.conf |
22:32.03 | kuku5 | it doesnt |
22:32.09 | sahafeez | ok. i am going to try to see if i can figure it out. |
22:32.12 | kuku5 | i do a blind transfer to extensiton 7000 |
22:32.13 | kuku5 | 700 |
22:32.20 | jontow | don't do a blind transfer. |
22:32.26 | kuku5 | then what should i do |
22:32.32 | jontow | a 'warm transfer' |
22:32.35 | kuku5 | how do i do that |
22:32.38 | kuku5 | on a gxp2000 |
22:33.02 | *** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net) |
22:34.02 | kuku5 | ... |
22:35.32 | *** part/#asterisk cypromis (n=michael@83.149.70.59) |
22:35.47 | delmar | kuku5, i have just been setting up blind/attended/parked calling etc... whats wrong? |
22:36.37 | Aughey | s,1,Background(file) |
22:36.43 | Aughey | s,1,WaitExten |
22:36.51 | Aughey | er, s,2,WaitExten |
22:36.57 | Aughey | 1,1,Goto(whatever) |
22:37.02 | Aughey | 2,1,Goto(whatever) |
22:42.39 | *** join/#asterisk riksta (n=rick@62.6.163.90) |
22:48.41 | Aughey | what context does it start in? |
22:48.45 | Aughey | incoming I assume? |
22:49.12 | sahafeez | yes |
22:49.39 | Aughey | and I assume you're dialing 3600 from some sip phone to enter into this? |
22:49.42 | sahafeez | no |
22:49.53 | sahafeez | dailing from my celling inbound on my pri |
22:50.50 | Aughey | and what do you dial after it plays closed-msg? |
22:50.56 | sahafeez | the DNIS is the _3600. i have the routing in place for during hours, etc.. |
22:51.07 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
22:51.12 | bweschke | damin: contact netxusa for the tdm2400 cards |
22:51.24 | bweschke | they're a distributor. they're supposed to ship the 3rd week in november |
22:51.35 | sahafeez | well nothing. the closed msg say, hit one 1 for dir, 2 for dir last name or enter and extension. if they screw up i want pbx-invalid and place closed-msg again |
22:53.27 | sahafeez | make ssense? |
22:53.58 | Aughey | so it does play the message right? |
22:54.28 | sahafeez | yes. and transfers to the 4xxx right. the dir trasfer does not work. |
22:54.39 | sahafeez | also the jump back for invalid does not work right |
22:54.45 | sahafeez | exten => _3600,1,Answer |
22:54.45 | sahafeez | exten => _3600,2,Background(closed-msg) |
22:54.45 | sahafeez | exten => _3600,3,WaitExten(15) |
22:54.47 | sahafeez | exten => i,1,Playback(pbx-invalid) |
22:54.47 | sahafeez | exten => i,2,Goto(incoming,3600,1) |
22:54.49 | Aughey | it plays the message and then goes to 4xxx? |
22:54.59 | Aughey | you're not making sense |
22:55.07 | sahafeez | no, plays and if you enter 4201 or such at anytime it works. |
22:55.37 | Aughey | I don't think it should do anything else because internal isn't included in the incoming extension |
22:56.04 | sahafeez | true, but the 4xxx transfer works and it is in interal |
22:56.54 | Aughey | if that's truely your full extensions.conf file, it shouldn't do that. There should be no other valid extensions in the incoming context |
22:57.52 | sahafeez | ah, i do have that in my inbound |
22:58.10 | Aughey | and take off the _ on the extensions that don't actually use patterns |
22:59.16 | sahafeez | on all? |
22:59.43 | Aughey | please post your actual extensions.conf file, otherwise we're just going to go round in circles |
23:00.10 | sahafeez | ok. but it is not quite complete as i am trying to test |
23:00.12 | Aughey | you don't need it. Shouldn't REALLY matter, but it implies there some sort of pattern for that extension |
23:00.12 | sahafeez | will do it now |
23:01.16 | sahafeez | http://pastebin.com/417663 |
23:01.17 | Aughey | but you have other things in the incoming context that you didn't show. So it's behaving in ways that it shouldn't if you actually used what you posted, but you didn't, so me telling you what it should do doesn't mean anything if I'm not actually looking at your actual configuration |
23:02.22 | *** join/#asterisk alvariux (n=unky@201.138.135.221) |
23:02.24 | alvariux | hi |
23:02.40 | alvariux | does somebody has build iaxclient on mac? |
23:03.00 | alvariux | im getting this error ld: Undefined symbols: |
23:03.00 | alvariux | _fprintf$LDBLStub |
23:03.01 | alvariux | _vsnprintf$LDBLStub |
23:03.01 | alvariux | _printf$LDBLStub |
23:03.01 | alvariux | _snprintf$LDBLStub |
23:03.02 | alvariux | _sprintf$LDBLStub |
23:03.04 | alvariux | libtool: internal link edit command failed |
23:03.06 | alvariux | make: *** [shared] Error 1 |
23:04.08 | Aughey | sahafeez: well your [internal] extensions aren't part of your [inbound] context. Move [internal] up above [inbound] and then include => internal in your inbound context |
23:04.51 | sahafeez | Aughey: i thought that was bad. |
23:05.24 | Aughey | well, the way it is now, anyone could dial 5616 and get your _5615 extension |
23:05.30 | mxmasster | is there a way to make the dialplan timezone aware - i.e. the server is in utc, but i want to make gotoiftime decisions based on the local timezone |
23:06.14 | Aughey | you need to partition your dialplan into more contexts so that you get what you want |
23:06.14 | jontow | mxmasster; any reason why you can't just set the server to the local timezone? |
23:06.38 | jontow | its a quick fix.. rm /etc/localtime ; ln -s /usr/share/zoneinfo/.../... /etc/localtime |
23:06.41 | jontow | ? |
23:06.50 | mxmasster | except i have multiple locations in different time zones |
23:07.14 | Aughey | so put the extensions tha tyou want available in their own context, and include it in the incoming context. Actually, you probably want an incoming_night context to only have the extensions you want there and have it go into that context |
23:07.46 | jontow | ok, so there is no 'local timezone' anyway.. its on a per-location basis |
23:07.55 | jontow | just translate all your times before you put 'em in the dialplan |
23:07.59 | sahafeez | Aughey: ok. that seems to make sense |
23:08.55 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
23:10.18 | mxmasster | jontow: that's what i need to know, how do i translate the time? |
23:10.24 | mxmasster | oh you mean by hand |
23:10.30 | mxmasster | any automated way? |
23:11.12 | jontow | writing scripts? |
23:11.16 | jontow | perl, C, shell? |
23:11.22 | jontow | "automated" is relative :) |
23:11.49 | mxmasster | automated as in internal to asterisk |
23:11.54 | jontow | no |
23:11.56 | *** join/#asterisk Flauto (n=zhao@71.194.39.175) |
23:12.09 | jontow | there isn't anything like that to my knowledge, currently |
23:12.13 | jontow | it will involve code-writing |
23:12.23 | jontow | i haven't seen everything thats out there, but.. to my knowledge. |
23:12.43 | *** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net) |
23:13.05 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166116148.nb.aliant.net) |
23:15.34 | DrukenHME | ~seen sivana |
23:15.38 | jbot | sivana is currently on #asterisk (9h 44m 45s). Has said a total of 1 messages. Is idling for 6h 31m 34s |
23:17.38 | sahafeez | does the exen => i,1 get jumped to in each contect that it is defined if something does not work right |
23:20.25 | justinu | ~seen blitz[laptop] |
23:20.30 | jbot | blitz[laptop] is currently on #asterisk-doc (2d 5h 25m 49s) #asterisk (2d 5h 25m 49s). Has said a total of 357 messages. Is idling for 1h 47m |
23:21.30 | Aughey | yes |
23:23.01 | *** part/#asterisk alvariux (n=unky@201.138.135.221) |
23:23.47 | sahafeez | Aughey: ok i have it working |
23:24.07 | sahafeez | Aughey: one thing left. i have the directories at ext2004/2005 |
23:24.42 | sahafeez | but in my msg i want to have them hit the exten, or 1 for dir1 or 2 for dir2 and have it jump to 2004/2005. can i do that |
23:25.32 | Aughey | sure, just create a 1 extension and Goto(2004) |
23:25.49 | Aughey | 1,1,Goto(2004) |
23:26.02 | sahafeez | ok. makes senses. thanks. each time i get something else working the whole picture makes more sense |
23:26.59 | sahafeez | OK, I Know what the exten =. i,1 is. but what is it when it is exten=> t,1 is t timeout? |
23:27.39 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
23:29.14 | *** join/#asterisk cripito (n=ncripito@ip67-154-143-190.z143-154-67.customer.algx.net) |
23:29.16 | cripito | hi |
23:29.34 | cripito | i ws in the wrong one :D |
23:30.32 | justinu | sahafeez: t is when they don't enter enough digits, I think |
23:31.51 | Aughey | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
23:32.16 | sahafeez | thanks all. ready for my cut over. |
23:32.20 | *** join/#asterisk BigJoe1 (n=BigJoe1@dsl-20-177.cofs.net) |
23:32.23 | sahafeez | heres hoping xo gets it right |
23:32.27 | BigJoe1 | hello everyone |
23:33.00 | syle | JOE! |
23:34.13 | syle | don't know who you are but sounded good , like norm from "cheers" |
23:34.46 | Flauto | is there a way i can setup the server to give the internet trafic to voice calls first and then others? |
23:36.00 | Aughey | iptables can probably help. It won't be necessarly easy |
23:36.43 | *** join/#asterisk Starmaker (n=magnus@85.8.2.169) |
23:37.01 | Flauto | aughey, it is not easy for setting up trafic? |
23:37.29 | cripito | flauto in the wiki u can find a few things about qos |
23:37.37 | cripito | some easy to do .. somes not so easy |
23:39.00 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
23:39.01 | Starmaker | I'm using alaw/ulaw right now for audio coding, but I'm thinking about trying G.729 instead, are there ANY advantages with it except it using less bandwidth? |
23:39.32 | Flauto | cripito, what topic it unders? |
23:39.35 | Aughey | not compaired to u/alaw |
23:39.39 | oogle | i wrote a hack to channel.c that manipulates voice frame data and spent an hour trying to figure out why it wasn't working. i loaded up gdb, froze the process and was like, huh? how come the phones are still bridged?? DUH they're sip phones! |
23:40.35 | Starmaker | ah, guess i'll stick to alaw/ulaw then, bandwidth isn't really a a problem |
23:42.07 | cripito | flauto: http://www.voip-info.org/wiki/view/VOIP+Routers |
23:42.26 | cripito | starmaker: me too |
23:42.58 | cripito | specially if i can make tricks with the invite etc etc for save bandwith too.. |
23:43.32 | Starmaker | now i just need to set up CDR with mysql :) |
23:44.55 | cripito | piece of cake ;) |
23:45.00 | *** join/#asterisk grimse (n=grimse@p5481DDC5.dip.t-dialin.net) |
23:45.01 | Starmaker | heh |
23:45.03 | oogle | i recommend using ODBC |
23:45.06 | oogle | makes life easier |
23:45.09 | Starmaker | i'm trying :) |
23:45.21 | cripito | cd_addon_mysql is not soo dif to install |
23:45.22 | Starmaker | it still keeps logging to csv files :) |
23:45.59 | Starmaker | cripito, well, I havn't compiled asterisk from source, i used debians asterisk-package |
23:46.22 | cripito | mmm... |
23:46.29 | oogle | heresy! |
23:46.33 | justinu | lol |
23:46.34 | Starmaker | heh |
23:46.36 | cripito | i prefer the source code.. ejem... |
23:47.04 | Starmaker | i like apt-get's dependency solving :) |
23:47.29 | cripito | well must me asterisk-addons-pkg or similar there |
23:47.41 | Starmaker | cripito, no, actually there isn't |
23:48.01 | Starmaker | but I should be able to set it up with odbc |
23:49.41 | oogle | i'm writing a voice changer, can anyone give me tips on ways i can change ast_frame data to accomplish this? |
23:50.05 | justinu | oogle, you might want to ask on #asterisk-dev |
23:50.14 | oogle | justinu: right-o |
23:50.16 | Starmaker | i mean, it's really strange, if i do odbc show, it shows that it is connected to one of my dsns |
23:50.46 | *** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il) |
23:51.51 | voipjoy | anybody can advice why i can't compile zaptel driver http://pastebin.ca/27578 ? |
23:53.12 | Starmaker | yes! i got it working :) |
23:55.36 | Starmaker | just found a really strange thing |
23:57.06 | Starmaker | skype -> sip-calls are actually voip all the way |
23:57.26 | Starmaker | but skype doesn't allow it's users to use SIP |
23:57.42 | Starmaker | that's just plain evil :) |
23:58.04 | ManxPower | voipjoy, You either have a bad CVS checkout or beta version, or you don't have the linux kernel source and headers installed. |