| 00:00.41 | *** part/#asterisk Cresl1n (n=matt@nat/digium/x-12344c8b6226c969) |
| 00:07.30 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-45-103.bchsia.telus.net) |
| 00:08.23 | *** join/#asterisk Penggu (n=me@203.213.102.59) |
| 00:09.19 | Penggu | hi all |
| 00:09.33 | Penggu | im looking at getting a 2-port PRI card with ECC |
| 00:09.42 | fujin_ | don't do itttt |
| 00:09.43 | JT | ecc? |
| 00:09.43 | Penggu | digium one, for use with * |
| 00:09.46 | fujin_ | buy a hardware deviiice |
| 00:09.49 | Penggu | echo cancellation |
| 00:09.53 | fujin_ | or a sangoma one at least |
| 00:09.55 | JT | HWEC |
| 00:10.01 | Penggu | is there anything to watch out for? |
| 00:10.04 | Penggu | jt: thanks |
| 00:10.17 | *** join/#asterisk Maliuta (n=nikolai@119.11.102.176) |
| 00:10.20 | Penggu | i heard there were issues with some of the cards (perhaps older ones) and something to do with vociebus.. |
| 00:10.29 | *** join/#asterisk mihinomenest (i=wVrh@66.255.220.17) |
| 00:10.57 | Penggu | we have to order through a VAR, so i want to make sure i ask the right questions - we cant touch the box ourselves (warranty) |
| 00:11.40 | JT | ecc means error correcting code |
| 00:12.00 | JT | Penggu: the new digium cards still have the same issues |
| 00:12.09 | Penggu | hmm |
| 00:13.21 | Penggu | IS there such a thing as a problem-free ISDN interface? |
| 00:13.35 | drmessano | ISDN is a problem |
| 00:13.40 | drmessano | There is only so much you can do about it |
| 00:13.40 | drmessano | lol |
| 00:14.16 | fujin_ | eh |
| 00:14.26 | fujin_ | use some dedicated isdn termination stuff |
| 00:14.30 | fujin_ | we have an AS5400 |
| 00:14.32 | fujin_ | works perfect |
| 00:14.48 | fujin_ | does E1 PRI<>SIP, haven't had any issues with * |
| 00:14.55 | drmessano | lol |
| 00:15.25 | JT | if i had a choice, i wouldn't buy a cisco voice gateway |
| 00:15.32 | JT | but if it was cheap, i wouldn't say no |
| 00:15.32 | drmessano | Troublefree ISDN circuits? |
| 00:15.38 | drmessano | I doubt it |
| 00:15.39 | drmessano | lol |
| 00:15.43 | JT | isdn is fine |
| 00:15.45 | JT | lololorofl |
| 00:15.48 | fujin_ | would you buy a voice gateway vs. a PCI(e|x) card? |
| 00:16.06 | JT | fujin_: depends on the size of the install |
| 00:16.11 | JT | and what use it would have |
| 00:16.14 | fujin_ | that's very valid, indeed. |
| 00:16.38 | JT | if i chose a voice gateway, it probably wouldn't be cisco |
| 00:16.54 | drmessano | But it costs more.. it has to be good |
| 00:16.58 | fujin_ | I don't doubt there are better, we're a cisco shop though and have super discounts |
| 00:17.40 | JT | drmessano: so you don't like isdn for some reason? |
| 00:17.41 | *** join/#asterisk a1fa (n=zZZ@unaffiliated/a1fa) |
| 00:17.52 | fujin_ | what the hell else are you going to use? :P |
| 00:18.09 | drmessano | I hate ISDN circuits in all their glory |
| 00:18.16 | *** join/#asterisk moprilo (n=nada@190.10.0.84) |
| 00:18.30 | JT | drmessano: why? |
| 00:18.48 | drmessano | For one thing, if you use ISDN, ma bell hates you |
| 00:19.16 | moprilo | hi, how do i get the the first 8 numbers out of a 9 number extension? i thought ${EXTEN:-1} but it just gives me the last digit. |
| 00:19.18 | drmessano | ISDN gets the lowest amount of support and priority from bell |
| 00:19.23 | JT | drmessano: what are you talking about? |
| 00:19.25 | JT | drmessano: BRI |
| 00:19.27 | JT | or PRI? |
| 00:19.40 | drmessano | Doesnt matter.. its how they handle ISDN period |
| 00:20.18 | JT | drmessano: i don't care what ma bell does |
| 00:20.21 | JT | isdn is fine |
| 00:20.23 | JT | especially pri |
| 00:20.24 | fujin_ | who the fuck is ma bell |
| 00:20.41 | fujin_ | no, but, seriously - what are you going to use isntead of PRI |
| 00:21.34 | drmessano | You dont? |
| 00:21.38 | a1fa | anyone have any experience with multicast paging? |
| 00:21.59 | drmessano | If the phone company doesnt care about keeping ISDN circuits running at 100%, shouldnt you care? |
| 00:22.29 | JT | drmessano: NO, BECAUSE WE DON'T HAVE MA BELL... |
| 00:22.31 | JT | seems simple |
| 00:22.44 | JT | not everywhere in the world is the same as the city where you are |
| 00:22.51 | JT | what is your proposed alternative to PRI? |
| 00:22.51 | drmessano | ROFL |
| 00:23.02 | drmessano | I know a hell of a lot more than just about the city I live in |
| 00:23.15 | drmessano | I can tell you in the US, ISDN is on a sharp downturn |
| 00:23.22 | fujin_ | so what do you use instead of ISDN? |
| 00:23.25 | JT | PRI, i doubt that |
| 00:23.25 | fujin_ | you're not answering me ;> |
| 00:23.28 | *** join/#asterisk ThatKidKel (n=Kelvin@64.221.169.232) |
| 00:23.31 | a1fa | fujin_: T1 |
| 00:23.38 | fujin_ | rofl |
| 00:23.40 | fujin_ | ;> |
| 00:23.40 | JT | ... |
| 00:23.42 | fujin_ | ok mate |
| 00:23.44 | fujin_ | t1 instead of ISDN |
| 00:23.47 | JT | a1fa: you're crazy |
| 00:23.48 | a1fa | t1 is cheap here |
| 00:23.49 | a1fa | :P |
| 00:23.52 | fujin_ | do you know what t1 is? |
| 00:24.08 | JT | a1fa: a north american PRI is just a T1 with PRI signalling |
| 00:24.18 | a1fa | :) |
| 00:24.23 | JT | and a PRI IS ISDN |
| 00:24.27 | fujin_ | i.e.; ISDN |
| 00:24.51 | ThatKidKel | is it possible to have asterisk jump contexts based on the incoming sip uri? in other words if i sent a call to sip:15552345555@contexta.mydomain.com --could asterisk send that call to 15552345555,1, in context contexta? |
| 00:25.17 | fujin_ | Yes. |
| 00:25.25 | fujin_ | You're thinking about it wrong, though. |
| 00:25.29 | a1fa | hehe |
| 00:25.45 | fujin_ | In the sip connection, define a context, handle your jump in there. |
| 00:25.53 | a1fa | i dont think of t1 like that, but it is what it is |
| 00:25.59 | fujin_ | t1==isdn |
| 00:26.05 | fujin_ | =~isdn |
| 00:26.12 | ThatKidKel | fujin.. i'm not sure i follow.. |
| 00:26.24 | ThatKidKel | are you talking about in sip.conf speicfy the context? |
| 00:26.50 | fujin_ | Yes. |
| 00:26.53 | JT | fujin_: well |
| 00:27.00 | JT | fujin_: a T1 CAN be ISDN |
| 00:27.02 | fujin_ | =~ isdn |
| 00:27.03 | JT | not always |
| 00:27.07 | ThatKidKel | that's all find and good; except the fact that my proxy is going to handle for multiple domains |
| 00:27.18 | fujin_ | okai |
| 00:27.18 | ThatKidKel | and i'd like those domains to be digested by * and jump to the appropriate context.. |
| 00:27.20 | JT | channelised T1 is not ISDN |
| 00:27.26 | drmessano | The term ISDN usually refers to a BRI |
| 00:27.28 | JT | uses RBS signalling |
| 00:27.30 | JT | drmessano: wrong |
| 00:27.34 | fujin_ | wrong wrong wrong |
| 00:27.35 | JT | drmessano: it refers to both |
| 00:27.35 | drmessano | uh huh |
| 00:27.41 | JT | BRI and PRI |
| 00:27.43 | ThatKidKel | so, when i specify the context in the declaration in sip.conf -- its all going to go to whatever context i specify |
| 00:27.49 | fujin_ | ThatKidKel: yes, indeed |
| 00:27.50 | JT | it refers to the use of Q.931 signalling |
| 00:27.58 | fujin_ | but you'll be able to match programatically in that context. |
| 00:28.02 | JT | and a dedicated D channel |
| 00:28.14 | fujin_ | ThatKidKel: setup multiple declarations in sip.conf for each domain that you're handling |
| 00:28.21 | JT | it has nothing to do with if there are 2 or 23 or 30 B channels |
| 00:28.53 | JT | only americans seem to have this weird misunderstanding |
| 00:29.31 | fujin_ | good old america |
| 00:29.40 | drmessano | Come to the states and order an ISDN circuit |
| 00:29.49 | drmessano | You'll get a BRI without even asking |
| 00:29.55 | JT | if i order a circuit, i will order it properly |
| 00:30.01 | JT | i will specify exactly what i want |
| 00:30.08 | a1fa | anyone have any experience with multicast paging? |
| 00:30.28 | drmessano | and the sales tech will say "Oh, so you want a T1 then?" |
| 00:30.29 | drmessano | lol |
| 00:30.32 | ThatKidKel | fujin_.. Like so? http://www.pastebin.ca/821257 |
| 00:30.36 | a1fa | JT:that was my understsnding |
| 00:30.51 | JT | channelised T1 (24 channels) = not ISDN, T1 PRI (23 B channels) = ISDN |
| 00:30.57 | *** join/#asterisk pepo-- (n=pepOSX@190.78.195.91) |
| 00:31.07 | JT | drmessano: then they will ask you what type of signalling you want on your T1, lololorofl |
| 00:31.09 | a1fa | 24 chann t1 |
| 00:32.46 | JT | depending on where you are in the US, you can get channelised (RBS), PRI, E&M |
| 00:32.53 | JT | on your T1 |
| 00:33.59 | JT | and a PRI is the best type of signalling, and it IS ISDN |
| 00:34.28 | *** join/#asterisk mihinomenest (i=3F1m@66.255.220.17) |
| 00:37.45 | *** join/#asterisk hijacked (i=SknP@66.255.220.17) |
| 00:39.06 | fujin_ | ThatKidKel: i'm not quite certain what you're tryign to achieve |
| 00:39.18 | fujin_ | you have four upstream sip connections that will send calls to your box |
| 00:39.24 | fujin_ | and they're all on the same IP address? |
| 00:39.37 | *** join/#asterisk beighto (n=chatzill@12.176.156.130) |
| 00:39.57 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 00:40.00 | ThatKidKel | i'm try to understand what you said.. here's the deal.. i want a call going to 5551234321@contexta.mydomain.com to be sent to extension 5551234321 in contexta |
| 00:40.29 | fujin_ | sorry, dunno. I've never handled sip URI's. |
| 00:40.34 | fujin_ | never found the reason to do such a thing. |
| 00:40.56 | fujin_ | perhaps if you define a declaration like [contexta.mydomain.com] |
| 00:41.04 | fujin_ | eh, I don't know - stabbing in the dark |
| 00:41.07 | fujin_ | perhaps read sip.conf.samples |
| 00:41.10 | fujin_ | it may have a relevant example |
| 00:42.00 | *** join/#asterisk salzh (n=salzh@124.77.5.10) |
| 00:42.25 | beighto | I can't get the buddy watch on a Polcyom 601 side car to show busy when a phone makes an outgoing call. It only shows busy for incoming calls. Help? |
| 00:47.42 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:47.42 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:55.40 | *** join/#asterisk mihinomenest (i=MWbp@66.255.220.17) |
| 01:27.50 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:28.43 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 01:31.35 | *** join/#asterisk chuckf (n=chuck@unaffiliated/chuckf) |
| 01:33.15 | chuckf | if I want to set up a home voip server on a dedicated box, is asterisknow the way to go? Or is there a better option out there? |
| 01:33.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 01:36.03 | JT | chuckf: this channel is mainly for asterisk itself |
| 01:36.21 | fujin | #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) |
| 01:36.43 | chuckf | okay. |
| 01:42.22 | *** part/#asterisk ThatKidKel (n=Kelvin@64.221.169.232) |
| 01:44.04 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:54.02 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:54.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:00.05 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:00.57 | *** join/#asterisk exvito (n=exvito@89.181.106.178) |
| 02:06.11 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:07.17 | *** part/#asterisk exvito (n=exvito@89.181.106.178) |
| 02:14.44 | *** join/#asterisk apocn (n=Apo@unaffiliated/apocn) |
| 02:15.16 | *** join/#asterisk cesar_CR (n=cr@celord.ice.co.cr) |
| 02:18.59 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:31.43 | *** join/#asterisk mihinomenest (i=GNqb@66.255.220.17) |
| 02:32.45 | *** join/#asterisk __freedom__lover (n=eduardo@200-148-114-24.dsl.telesp.net.br) |
| 02:32.58 | __freedom__lover | can someone help me with res_snmp? |
| 02:35.54 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:37.37 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 02:40.59 | *** join/#asterisk hijacked (i=FC4s@66.255.220.17) |
| 02:48.39 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 02:49.01 | [T]ank | having trouble with asterisk-addons and getting it to use cdr_mysql |
| 02:49.13 | [T]ank | i have installed mysqlclient and mysqlserver |
| 02:49.23 | [T]ank | and have installed asterisk with it running... |
| 02:49.35 | [T]ank | but asterisk does not load the cdr mysql stuff. |
| 02:54.55 | [T]ank | nothing is getting loaded for that matter, sip, iax, zap.. anything. |
| 02:55.13 | [T]ank | the modules are there in /usr/lib/asterisk/modules, and are all owned by root. |
| 02:55.16 | [T]ank | but nothing is there. |
| 02:55.19 | [T]ank | in the cli |
| 02:57.30 | [T]ank | lonely tonight |
| 02:57.48 | __freedom__lover | \q |
| 02:59.56 | [T]ank | "show modules " show 0 modules loaded. |
| 03:08.30 | [T]ank | this place is always hoppin' why so quiet tonight? |
| 03:11.12 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 03:11.48 | [T]ank | ok, how did I screw up and do this: No 'modules.conf' found, no modules will be loaded. |
| 03:17.00 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 03:19.53 | JunK-Y | use the use in sample |
| 03:30.23 | thehar | having some serious mic feedback into this phone headset with a pap2.. curious which fxs port input to change and which way. |
| 03:40.18 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 03:40.25 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:41.01 | a1fa | ahoy |
| 03:41.24 | *** join/#asterisk ReD-MaN (i=root-rox@172-220.static.golden.net) |
| 03:42.00 | *** part/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 03:42.44 | a1fa | anybod familiar with multicast paging? |
| 03:44.55 | Nugget | http://flickr.com/photos/ryjones/1746302727/ |
| 03:45.10 | *** part/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 03:46.27 | *** join/#asterisk jnc (i=lucent@outbound.silenceisdefeat.org) |
| 03:46.35 | a1fa | Nugget |
| 03:46.42 | a1fa | do you know what model is that? |
| 03:47.00 | Nugget | no clue |
| 03:48.26 | a1fa | jnc |
| 03:48.27 | a1fa | ask |
| 03:49.03 | jer | jnc, i hear ya =] |
| 03:49.38 | jnc | okay, was waiting patiently. I want to know (from a guru or somebody) what direction I need to go, I've got a trixbox setup that worked before... but that's not really even part of the picture yet... I want to know what sort of configuration I should get my Vegastream 50 (8fxs, 2fxo) hardware into |
| 03:50.15 | jnc | there's some options I don't understand, I can do SIP registration, SIP authentication, and/or SIP proxy on the hardware |
| 03:51.11 | jnc | do I want registration/authentication, or set it up as a proxy, or what ;/ |
| 03:51.44 | a1fa | did u read mfg docs? |
| 03:52.00 | jnc | yeah, I mostly grasp how to navigate the configuration details |
| 03:55.29 | jnc | if I'm dialing a call on one of the fxo ports, it hits the vega50's dialplan module and at this point I don't know what is the target... one SIP user per fxo/fxs port? dialing plan rules with only one SIP user to navigate |
| 03:56.50 | a1fa | how did u define it in sip.conf |
| 03:57.03 | jnc | heh, I'm not even there yet |
| 03:57.36 | jnc | what's a typical interaction with asterisk look like on these multi-fxo/fxs ATAs ? |
| 03:57.58 | jnc | multi-sip user or one sip user handling many #'s |
| 03:58.24 | a1fa | it should be registred like peer |
| 03:58.48 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:58.51 | a1fa | have you checked voip-info? |
| 03:59.00 | jnc | maybe I should |
| 03:59.17 | jnc | I only know how to search for docs on hardware and software I have - this doesn't cover SIP basics |
| 04:00.44 | riddlebox | does anyone use the 1.1.5.15 firmware for the Grandstream GXP's? |
| 04:01.45 | jnc | a1fa: know off-hand where I could read up on the difference between USER, PEER, etc. for my situation? |
| 04:01.56 | a1fa | voip-info |
| 04:02.06 | jnc | looking now |
| 04:02.16 | jnc | there's a lot of unrelated info, hard to know where to begin |
| 04:03.26 | jnc | http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14839 |
| 04:03.28 | a1fa | search |
| 04:03.29 | a1fa | for sip |
| 04:03.54 | jnc | so, if type=user will be phased out, why keep "friend" around? |
| 04:04.18 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 04:04.59 | fujin | lol |
| 04:05.06 | fujin | that's the first I've heard of that being phased out |
| 04:05.11 | fujin | and it's 2007 |
| 04:05.14 | jnc | so confused right now =( |
| 04:06.02 | a1fa | anyone know where u can get those emergency phones at? |
| 04:06.19 | jnc | the red ones? |
| 04:06.45 | a1fa | yes |
| 04:06.51 | jnc | take a black pre-configured one-number phone and spray paint it red |
| 04:06.54 | a1fa | with a blue light |
| 04:07.08 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 04:07.24 | jnc | a1fa: if you're hard up for one, www.sandman.com |
| 04:08.16 | a1fa | no dude |
| 04:08.25 | a1fa | i need those emergency phones u see at parkinglots |
| 04:08.30 | a1fa | with blue flashing lights |
| 04:08.39 | jnc | oh |
| 04:09.24 | *** join/#asterisk ethernalq (n=ethernal@S01060013461a92b3.ed.shawcable.net) |
| 04:09.28 | jer | a1fa, use the cover of darkness to your advantage =D |
| 04:09.41 | a1fa | its not rape, its only suprise sex? |
| 04:09.50 | jer | lol |
| 04:10.36 | jnc | some relavent pointers to peer / user info would be appreciated |
| 04:10.50 | jnc | seeing as the stuff I found is "old" |
| 04:11.54 | jnc | fujin: you use 'friend'? |
| 04:12.13 | fujin | uh |
| 04:12.15 | fujin | depends onw hat for |
| 04:12.16 | fujin | but yes |
| 04:12.21 | fujin | i use friend for the phones on everyones desks |
| 04:12.39 | fujin | and peer/user for my 'providers', fwiw |
| 04:12.45 | jnc | ah okay |
| 04:12.46 | fujin | I have a peer/user defined for my cisco as5400 |
| 04:12.55 | fujin | and a peer defined for an outgoing SIP connection |
| 04:12.59 | fujin | (I don't accept calls from it) |
| 04:13.02 | *** join/#asterisk trashguy (n=trashguy@MSFC.calarts.edu) |
| 04:13.02 | jnc | is peer/user possible for all instances when you have friend ? |
| 04:13.06 | fujin | indeed |
| 04:13.10 | fujin | just requires double definitions |
| 04:13.19 | jnc | ah okay |
| 04:13.23 | fujin | but obviously gives you the advantage of being able to handle (contextually) the phone calls |
| 04:15.03 | jnc | I'm farting through the FreePBX web UI to decipher this... there's Outgoing Settings... um making a screen capture of this it is long to explain |
| 04:15.15 | fujin | #freepbx |
| 04:15.18 | *** join/#asterisk troy- (n=troy@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 04:15.19 | jnc | good call |
| 04:15.25 | fujin | no one here is interested in crappbx |
| 04:15.30 | fujin | nor will they support you |
| 04:15.35 | fujin | piece of junk |
| 04:15.51 | jnc | yeah, I work for a friend who won't touch anything that lacks a web UI |
| 04:16.00 | troy- | i have 2 asterisk boxes connected via IAX, how can i call an extension on the second box from the first? |
| 04:16.07 | jnc | not really my choice, unless I want to quit working :P |
| 04:16.28 | jer | troy-, put in an extension, i.e., exten => 1234,1,Dial(IAX2/otherpbx/1234) |
| 04:16.35 | troy- | perfect, thanks |
| 04:18.33 | *** join/#asterisk Yourname` (i=Myztic@unaffiliated/yourname/x-837320) |
| 04:19.14 | jer | speaking of which, i need to get started on building another pbx to connect all our tower sites together and order another pri =/ bah |
| 04:21.18 | jnc | fujin: http://silenceisdefeat.org/~lucent/pictures/fpbx-trunk-config.png are the terms on the UI proper VoIP terminology? |
| 04:21.30 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 04:21.32 | jnc | trying to search for help but I need the right terminology :) |
| 04:21.58 | jnc | particularly how the default is no "username=" for Incoming section |
| 04:26.04 | jnc | in the config, "Trunk Name: foo" ends up as [foo], "USER Context: bar" ends up as [bar] |
| 04:27.19 | jnc | it's just strange to me that default is no username for incoming |
| 04:27.22 | jnc | should that seem strange? |
| 04:37.32 | troy- | are there any additional ports besides 4569 that need to be open for IAX? |
| 04:39.46 | nestAr | troy-: nope, i am running iax through nat with only that port open |
| 04:40.57 | troy- | does it change anything if you use the encryption? |
| 04:45.13 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:45.56 | MrTelephone | what kind of methods would be appropriate for grabbing a peers ip from asterisk |
| 04:46.00 | MrTelephone | use manager? |
| 04:46.55 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 04:47.41 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 04:53.24 | ethernalq | We currently have have an 8-line 24-port Nortel PBX/VM/Conf.Call, is there some sort of guide as to what would be required to replace this with an Asterisk based system? Asterisk.org doesn't seem to have guide that breaks down each peice in a more laymans term. |
| 04:57.32 | *** join/#asterisk mihinomenest (i=F5jX@66.255.220.17) |
| 04:59.25 | jnc | ethernalq: I'm fighting with a VegaStream 50 fxs 8 |
| 04:59.33 | jnc | 8 fxs, 2 fxo |
| 04:59.50 | troy- | do i need 4569 TCP & UDP? |
| 04:59.50 | jnc | works like a charm if you can make it work at all |
| 05:00.19 | jnc | my boss picked one up for $150usd at auction |
| 05:01.08 | jnc | initial asterisk config is frightening if you've never dealt with computers, telephony, or networking before |
| 05:01.27 | jnc | might think seriously about hiring a chum to do your initial configuration |
| 05:02.37 | coppice | s/chum/chump/ |
| 05:03.30 | JT | troy-: no tcp |
| 05:03.39 | troy- | JT, just got it working :) thanks! |
| 05:03.43 | JT | troy-: almost nothing uses the same tcp and udp port at once |
| 05:03.48 | JT | forwarding both is pointless |
| 05:04.03 | Yourname` | Hi. On eyebeam, when a call comes in, it shows 3457234702@65.22.22.129 |
| 05:04.09 | Yourname` | What's up with that? |
| 05:04.18 | Yourname` | Shouldn't it be just the name on the caller id and number? |
| 05:04.25 | Yourname` | What's with the number@IP deal? |
| 05:05.02 | *** join/#asterisk mihinomenest (i=YC0T@66.255.220.17) |
| 05:06.24 | *** join/#asterisk shido6 (n=shido6@74-130-53-46.dhcp.insightbb.com) |
| 05:11.53 | *** join/#asterisk mihinomenest (i=OEGU@66.255.220.17) |
| 05:12.30 | Yourname` | God I miss tkdefender! |
| 05:12.36 | Yourname` | !seen [TK]-Dfender |
| 05:12.36 | *** join/#asterisk CVirus (n=GoD@62.135.96.207) |
| 05:12.40 | Yourname` | seen [TK]-Dfender |
| 05:14.47 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 05:20.45 | voiper1 | why does the echo cancel show mg2 if i recomplied with mark2? |
| 05:28.28 | Qwell | voiper1: bug in old version of zaptel |
| 05:28.39 | Qwell | was in...umm...1.4.5? I think? |
| 05:29.55 | *** join/#asterisk mihinomenest (i=q33G@66.255.220.17) |
| 05:32.04 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:32.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:41.44 | *** join/#asterisk mihinomenest (i=fxHx@66.255.220.17) |
| 05:42.26 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 05:46.14 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 05:49.01 | voiper1 | qwell: thanks im doing some adjustments. one other thing, with fxotune when you run it like 'fxotune -i 9' it gets the dial tione but pretty quickly goes to a beep beep pattern, is this what fxotune expects? |
| 05:49.29 | *** join/#asterisk e66 (n=shiplu@202.56.7.137) |
| 05:50.34 | e66 | how to install |
| 05:50.44 | e66 | asterisk-0.9.6-x86.vmx |
| 05:59.46 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:01.22 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 06:01.37 | MrTelephone | does anyone here try and use Perl IO socket to send crap to asterisk? |
| 06:02.03 | MrTelephone | If I telnet to the manager interface and type the auth info it works.. but sending it with perl says auth failure :( |
| 06:02.03 | Nugget | telnet is eeeeeeevil! |
| 06:02.13 | MrTelephone | I'm using \r\n\r\n after the info |
| 06:04.10 | MrTelephone | $login_info = "ACTION: LOGIN\nUSERNAME: username\nSECRET: secret\nEVENTS: ON\n\r\n\r"; |
| 06:04.45 | russellb | you have \n\r |
| 06:04.47 | russellb | it's \r\n |
| 06:04.58 | russellb | also, you need \r\n after every line, not just \n |
| 06:05.19 | russellb | \r\n after every line, two at the end |
| 06:11.09 | *** join/#asterisk mihinomenest (i=sfzs@66.255.220.17) |
| 06:14.24 | *** part/#asterisk e66 (n=shiplu@202.56.7.137) |
| 06:14.57 | MrTelephone | thanks russ knox |
| 06:17.56 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:18.02 | bintut | hello all.. |
| 06:19.12 | bintut | anyone can confirm this? does the snom 300 phone supports ipv6 address? |
| 06:22.54 | *** join/#asterisk hijacked (i=ai13@66.255.220.17) |
| 06:25.23 | *** join/#asterisk zeeesh (n=zeeesh@203.215.176.22) |
| 06:26.16 | zeeesh | is there anybody .. trying to install asterisk at redhat9 .. getting this problem "make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. |
| 06:26.16 | zeeesh | make: *** [utils] Aborted |
| 06:26.16 | zeeesh | "? |
| 06:27.02 | *** join/#asterisk Defraz (n=tim@24-116-152-177.cpe.cableone.net) |
| 06:27.40 | *** join/#asterisk mihinomenest (i=vAim@66.255.220.17) |
| 06:28.41 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 06:28.48 | teknoprep | anyone know how to fix this problem i am having |
| 06:28.59 | teknoprep | [Dec 18 01:28:21] WARNING[25809] chan_sip.c: username mismatch, have <VP-SIP-NYC>, digest has <s> |
| 06:28.59 | teknoprep | [Dec 18 01:28:21] NOTICE[25809] chan_sip.c: Failed to authenticate user "RAWLINGS CHRIST" <sip:6107413324@64.61.93.90>;tag=as07851c2b |
| 06:29.26 | teknoprep | i don't understand why voicepulse is trying to authenticate with me |
| 06:37.30 | *** join/#asterisk Networkstudent (n=Sharagim@91.184.82.130) |
| 06:39.59 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 06:43.28 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 06:43.45 | Networkstudent | I have a problem with H323 on asterisk , is there anyone which could help me |
| 06:44.35 | jnc | kind of slow tonight, FYI |
| 06:44.45 | jnc | I'm * newbie |
| 06:49.42 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:55.36 | *** join/#asterisk _baconbuttie_ (n=bob@87.127.196.152) |
| 07:01.20 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 07:12.45 | *** part/#asterisk misc-- (n=misc@122.2.119.122) |
| 07:19.30 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 07:28.21 | *** join/#asterisk vyamba (n=chatzill@194.42.96.201) |
| 07:38.11 | *** join/#asterisk LakeSolon (n=blake@12-202-201-70.client.mchsi.com) |
| 07:39.10 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 07:53.03 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:54.19 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 07:56.05 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 07:57.10 | MrTelephone | matt riddell just added me to facebook |
| 07:58.10 | MrTelephone | does he come in this channel? |
| 07:58.28 | JT | what country is he from? |
| 07:59.10 | MrTelephone | new zealand |
| 07:59.29 | JT | ZX81 |
| 08:00.13 | JT | MrTelephone: that was your answer |
| 08:00.28 | MrTelephone | cool |
| 08:00.41 | MrTelephone | maybe i'll quit my day job and start working on some asterisk bounties |
| 08:01.03 | JT | heh |
| 08:03.10 | MrTelephone | i should goto bed though its 3am |
| 08:03.14 | MrTelephone | take it easy jt |
| 08:03.35 | MrTelephone | t1 got a red alarm tonight for the first time in 6 months :( |
| 08:04.25 | MrTelephone | exit |
| 08:04.28 | *** part/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 08:10.12 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 08:11.05 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:20.50 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 08:24.06 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 08:33.06 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 08:44.25 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 08:45.50 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 08:47.12 | vyamba | hello |
| 08:47.39 | vyamba | i have a sip client x-lite inside a firewalled network. i have a socks 5 and http proxy to the outside where asterisk is. is there any way i can register with asterisk with this setup? |
| 08:48.21 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 08:49.49 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 08:52.26 | *** join/#asterisk grEvenX (n=even@1elt2pn.ip.hipercom.no) |
| 08:57.56 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:05.36 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 09:11.19 | *** join/#asterisk bmg505 (n=leon@196.209.180.166) |
| 09:17.38 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de) |
| 09:19.29 | yang | Is this a correnctly formed extension: |
| 09:19.29 | yang | exten => _9.,1,Dial(IAX2/btctr1/${EXTEN:1}) |
| 09:19.29 | yang | exten => _9.,2,Hangup |
| 09:19.29 | yang | "btctr1" is the name of IAX2 context |
| 09:22.27 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:23.50 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:24.54 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:27.40 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 09:33.33 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 09:33.42 | ice_croft | hi all |
| 09:36.15 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 09:36.31 | ice_croft | is there any substring function to use in dialplan? |
| 09:37.10 | tzafrir | ${VARIABLE:1} |
| 09:37.17 | tzafrir | ${VARIABLE:1:2} |
| 09:37.23 | ice_croft | thanx |
| 09:38.04 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:38.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:40.46 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 09:41.18 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 09:42.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:45.32 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.113) |
| 09:54.36 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:11.20 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 10:15.26 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 10:25.30 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 10:26.21 | McDouglas | how can i make asterisk reply differently based on time of day? |
| 10:27.02 | *** join/#asterisk Odd_Bloke (n=oddbloke@compsoc.sunion.warwick.ac.uk) |
| 10:27.37 | af_ | if you look at dial plan, there is a specific syntax for that |
| 10:29.26 | Odd_Bloke | Hi guys, I've got a box free which I'd like to use to play around with Asterisk. I'm more comfortable with Debian-based distros than with RPM-based distros. Is trixbox still the best way to go, or is there a decent Debian-based alternative? |
| 10:35.16 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 10:40.20 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-e3de0baa6758bebb) |
| 10:40.20 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk vyamba (n=chatzill@194.42.96.201) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk Pagautas (n=bigman@83.171.14.250) |
| 10:40.20 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk km-- (n=pgrace@aeneas.fierymoon.com) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 10:40.20 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk Falle (n=falle@diana.falle.se) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk mocker (n=ksexton@198.247.173.227) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk deever (n=deever@217-20-127-116.internetserviceteam.com) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk CCFL_Man (i=CCFL_Man@argon.pureshells.com) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 10:40.21 | *** join/#asterisk lotho (n=lotho@217-20-118-126.internetserviceteam.com) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk BadPacket (n=NoCarrie@unaffiliated/badpacket) |
| 10:40.21 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 10:40.21 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk destructure (n=kwatz@66.193.229.254) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk lilalinux (i=e-trolle@fellatio.deswahnsinns.de) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk iCEBrkr (n=icebrkr@cyberdyne.org) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk hardwire (n=bip@rdbck-7085.palmer.mtaonline.net) [NETSPLIT VICTIM] |
| 10:40.21 | *** join/#asterisk dbtid (i=qjgns0eb@cpe-71-72-252-171.columbus.res.rr.com) |
| 10:40.21 | *** join/#asterisk exothermc (n=miles@izetta.office2-ww.wideideas.net) [NETSPLIT VICTIM] |
| 10:40.23 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) [NETSPLIT VICTIM] |
| 10:40.23 | *** join/#asterisk DavisGr (i=grikis@in.delfi.lv) [NETSPLIT VICTIM] |
| 10:40.23 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) [NETSPLIT VICTIM] |
| 10:40.23 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) [NETSPLIT VICTIM] |
| 10:40.23 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) [NETSPLIT VICTIM] |
| 10:40.23 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 10:40.23 | *** mode/#asterisk [+o Deeewayne] by irc.freenode.net |
| 10:42.13 | McDouglas | af_: any specifics? |
| 10:42.35 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 10:42.43 | af_ | look at the samples |
| 10:43.51 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 10:45.00 | McDouglas | well, i would, if oyu tell me which samples |
| 10:45.47 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:46.05 | af_ | conf sample coming with asterisk |
| 10:46.31 | *** join/#asterisk e66 (n=shiplu@202.56.7.137) |
| 10:56.23 | ai-a | McDouglas: read the book ? |
| 11:00.48 | *** join/#asterisk CrummyGummy (n=dude@dsl-242-62-219.telkomadsl.co.za) |
| 11:01.21 | CrummyGummy | Hi all, is there any equipment I can use to split an E1 line for redundancy? |
| 11:03.00 | waKKu | multiplexer |
| 11:04.02 | CrummyGummy | any recommended vendor? |
| 11:05.28 | e66 | this channel is dead |
| 11:05.41 | e66 | don't know why they don't talk |
| 11:05.53 | e66 | though 225 people in this room |
| 11:05.59 | JT | waaa |
| 11:06.05 | JT | possibly they're sleeping or otherwise afk |
| 11:06.36 | e66 | JT: how 224 people sleeps at a time in a room? |
| 11:06.52 | CrummyGummy | Easy, 224 yanks. |
| 11:06.53 | e66 | JT: 226 - (you and me) = 224 |
| 11:07.07 | JT | e66: irc is not everyone's life |
| 11:07.11 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:07.12 | JT | why expect them to be on 24/7? |
| 11:07.19 | JT | being quite unreasonable |
| 11:10.58 | e66 | JT: jUST THINKING don't they shut down their pc ?? |
| 11:11.22 | JT | no, most experienced technical people do not shut down their PCs |
| 11:11.27 | JT | or at least their servers |
| 11:11.32 | JT | a lot use servers to connect to IRC |
| 11:12.08 | tzafrir | Why shut down the PC? I want to get my own private log of the channel |
| 11:12.17 | JT | indeed :) |
| 11:13.18 | e66 | hmm |
| 11:13.30 | e66 | JT: I don't shut down my pc also |
| 11:14.05 | e66 | how can I customize the asterisknow interface |
| 11:14.05 | e66 | its a vmware image. |
| 11:14.35 | e66 | No one is telling me. |
| 11:15.18 | JT | this isn't an asterisknow channel |
| 11:15.40 | e66 | JT: I know. |
| 11:16.13 | e66 | JT: this channel is atleast sleeping. but asterisknow is dead |
| 11:16.23 | JT | hardly anyone using it ;) |
| 11:18.51 | ai-a | e66: *now, is just a front end, that is beta and never working. |
| 11:19.19 | ai-a | e66: if you read though the book, 1 day, you'll understand how and why not to use the front end. and you'll feel a better man for it also. |
| 11:20.31 | tzafrir | e66, you can ask here asterisk-related questions |
| 11:20.49 | *** join/#asterisk beraldi (n=beraldi@200-181-171-114.ctame705.dsl.brasiltelecom.net.br) |
| 11:21.01 | tzafrir | if you're asking about the asterisk-gui: it is a bunch of static javascript files |
| 11:21.35 | tzafrir | located typically under /var/lib/asterisk/static-http/ |
| 11:50.59 | *** join/#asterisk egypcio (n=egypcio@200.150.142.210) |
| 11:51.38 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:55.38 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585332.dsl.bell.ca) |
| 12:03.34 | ice_croft | re all |
| 12:04.04 | ice_croft | plz check this - http://pastebin.ca/821903 |
| 12:04.25 | ice_croft | im dialin from sip to pstn through h323 provider |
| 12:09.09 | *** join/#asterisk duckz (n=duckz@85-204-47-228.etth.opensys.ro) |
| 12:11.37 | McDouglas | is there any way i can put some text on the lcd of an spa921 when there is an incomming call? |
| 12:12.33 | ai-a | McDouglas: dont like to see whos ringing you ? |
| 12:13.01 | McDouglas | no, i'd like to display the a work order number |
| 12:13.14 | McDouglas | which i'll ask fromt he caller before i make the call |
| 12:13.56 | ai-a | usually you get Tapi api and place it onto the desktop and even open the db ready for the caller. |
| 12:14.20 | McDouglas | well, we have a shitty program, so its kinda impossible to integrate it :\ |
| 12:14.40 | ai-a | nothings impossible. |
| 12:14.43 | McDouglas | ;) |
| 12:15.03 | McDouglas | the database was made in ms access, and the person who made it left the company 10 years ago |
| 12:15.37 | ai-a | look, i am not interrested in problems. |
| 12:15.44 | ai-a | neither is your boss. |
| 12:16.14 | McDouglas | well, jsut an insight why i'd rather display it on the lcd |
| 12:16.25 | coppice | ms access - a classic example of a name chosen to obscure rather than illuminate :-) |
| 12:16.32 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:16.38 | ai-a | its called "Caller Presentation" |
| 12:21.02 | McDouglas | hm, not too promising |
| 12:21.32 | McDouglas | i might actualloy make a simple script which could be called when there is an incomming call, and that would display the information |
| 12:21.40 | McDouglas | maybe a php page |
| 12:22.06 | McDouglas | do you have any example of such thing? |
| 12:27.20 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 12:39.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:41.19 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 12:44.24 | *** join/#asterisk ronr (n=ron@ip51cdd509.speed.planet.nl) |
| 12:44.42 | McDouglas | if i want to use did, and i have a nubmer field liek this: 566-600 -> 566-609, then what should i put in the dialplan, 09, or 9 ? |
| 12:45.04 | McDouglas | (last 2 numbers are sent) |
| 12:46.06 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 12:46.16 | ai-a | McDouglas: have you heard of the Asterisk Book ? |
| 12:46.23 | ronr | what is the right way to implement a dialplan that dial some channel for 5 seconds and add a second when those 5 passed, a third when another 5 passed etc. (so first sales gets the incoming call, if they don't answer, supports also starts ringing, if they don't answer tech is added) |
| 12:47.02 | McDouglas | ai-a: well, i have heard about it and also did read it, but i dont seem to remember what happens if the did number starts with a 0 |
| 12:47.12 | ai-a | ronr: dial command takes a timeout returning busy. then dial others and so on. |
| 12:47.27 | ai-a | McDouglas: download it and read it again then. |
| 12:47.38 | McDouglas | thanks, very helpful :\ |
| 12:47.47 | ronr | ai-a: that won't create hickups in the phones that ring, timeout and are then started again? |
| 12:47.57 | ai-a | what ? |
| 12:48.43 | ronr | ai-a: phone A rings 5 seconds, timeout and then phone A and B ring 5 seconds, if phone A is in the middle of a ring sound, won't that just 'break' and start again? |
| 12:48.55 | ronr | creating tring tring tr tring tring |
| 12:49.04 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 12:49.14 | ai-a | ronr: what has dialing got to do with a phone ringing ? |
| 12:49.23 | ai-a | the 'ringer' is just an indication of an incomming call. |
| 12:49.37 | ai-a | if you stop a dial, its stopped. |
| 12:49.56 | ai-a | and also SIP can support any number of ringing calls. |
| 12:51.20 | ronr | ai-a: ok, I guess I should just try it and fix any problems if they occur (I guess I was sorta expecting a BackgroundDial application, like BackgroundDial(SIP/1), Sleep(5), BackgroundDial(SIP/2), etc.) |
| 12:53.01 | ice_croft | people, where to look? |
| 12:53.25 | ai-a | ice_croft: look for what? |
| 12:53.40 | ice_croft | i need to dial through h323 gw from nated network |
| 12:53.53 | ice_croft | * is inside nat |
| 12:54.10 | ice_croft | udp transport cant reach h323 gw |
| 12:54.21 | ice_croft | how can i solve this? |
| 12:56.50 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:59.41 | *** part/#asterisk CrummyGummy (n=dude@dsl-242-62-219.telkomadsl.co.za) |
| 13:02.46 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 13:12.39 | e66 | in asteriskNow, is GUI created by only html?? |
| 13:14.04 | *** join/#asterisk gormux (n=julien@was59-1-82-226-112-70.fbx.proxad.net) |
| 13:14.08 | gormux | hi all |
| 13:14.22 | ice_croft | ðø |
| 13:14.23 | ice_croft | hi |
| 13:14.58 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 13:15.01 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:15.10 | gormux | i was searching for infos, i would like to automate the creation of SIP accounts on a server |
| 13:15.34 | gormux | i want a sip account for each user on a server, with the same user/pass |
| 13:15.38 | gormux | is it possible ? |
| 13:15.59 | e66 | I want to customize the GUI of Asterisk |
| 13:16.07 | e66 | sorry asteriskNOW |
| 13:16.28 | e66 | how to change the GUI? only images and some texts |
| 13:16.35 | McDouglas | any idea why do we only have 2 lines out of the 4? http://pastebin.com/m52359f88 |
| 13:17.47 | gormux | e66: maybe you could check in the directory where you put asterisknow files |
| 13:18.13 | badcfe | should * be able to receive DTMF on one of its own originated calls? |
| 13:20.36 | e66 | I need only to change the web files |
| 13:20.43 | e66 | not the whole engine |
| 13:21.06 | e66 | gormux: I installed the vmware image of asterisk image |
| 13:21.27 | e66 | gormux: mount it. |
| 13:21.30 | gormux | did you check in /var/www, for example |
| 13:21.49 | e66 | gormux: now checking the web files |
| 13:21.55 | kaldemar | e66: from this channel's topic: "#asterisknow or #asterisk-gui for AsteriskNOW" |
| 13:22.16 | gormux | yeah, probably a better place |
| 13:23.06 | e66 | but none of them are answering |
| 13:23.16 | e66 | join #asterisk-gui |
| 13:23.28 | e66 | sorry :D |
| 13:24.52 | *** join/#asterisk lftsy (n=lftsy@anj75-1-81-57-0-22.fbx.proxad.net) |
| 13:24.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:25.22 | McDouglas | "* Port 1 Type TE Prot. PTP L2Link DOWN L1Link:UP Blocked:0 Debug:0" -> if L1Link is up, could this still be a phisical problem? (with the nt box, or the cables?) |
| 13:26.20 | tzafrir | e66, I answered you. You haven't followed up |
| 13:26.50 | e66 | really |
| 13:27.10 | e66 | tzafrir: sorry. |
| 13:28.07 | tzafrir | Have you found those files? |
| 13:28.41 | e66 | after I mount it by vmware. I found lots of web files |
| 13:29.13 | e66 | tzafrir:there are in total 5076 files |
| 13:34.55 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585332.dsl.bell.ca) |
| 13:37.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 13:38.55 | *** join/#asterisk JaminCollins (n=jcollins@ptech7-171.acdmis.com) |
| 13:46.25 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 13:47.13 | VJFROMGT | http://pastebin.ca/822011 calls from ip 66.162.75.120 are failing |
| 13:49.38 | kaldemar | VJFROMGT: you're not using plain asterisk. |
| 13:50.04 | badcfe | i do an ami originate with "Variable: A=1". The exten in the dialplan where i place it awakes with no such variable set. any common mistakes or misunderstandings around this? |
| 13:50.13 | [TK]D-Fender | VJFROMGT: And where's the failure in there? |
| 13:50.14 | kaldemar | VJFROMGT: #freepbx is for you. |
| 13:50.54 | *** join/#asterisk [gnubie] (n=[gnubie]@cm71.gamma182.maxonline.com.sg) |
| 13:50.58 | VJFROMGT | ok, thanks |
| 13:51.26 | [TK]D-Fender | badcfe: pastebin your AMI call and your dilaplan. |
| 13:51.29 | [TK]D-Fender | ~pb |
| 13:51.30 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:51.32 | [TK]D-Fender | ^^^^^^^^ |
| 13:52.14 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:52.43 | *** part/#asterisk [gnubie] (n=[gnubie]@cm71.gamma182.maxonline.com.sg) |
| 13:59.53 | *** join/#asterisk seele_ (n=seele@static-adsl201-232-85-61.epm.net.co) |
| 14:01.30 | [TK]D-Fender | *crickets* |
| 14:02.12 | FlatFoot | [TK]D-Fender: ???? |
| 14:02.35 | FlatFoot | crickets ? |
| 14:03.14 | [TK]D-Fender | FlatFoot: Use your imagination.... or at least a DICTIONARY before just doing "????" and looking clueless. |
| 14:04.02 | FlatFoot | [TK]D-Fender: looking clueless ! HA i was born that way :} |
| 14:04.35 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:04.35 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:05.03 | badcfe | [TK]D-Fender: im terribly sorry to be a lame ignorant crab |
| 14:05.32 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:05.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:05.44 | badcfe | [TK]D-Fender: .. the name of the variable didnt correspond to the one i set in the originate |
| 14:06.15 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:06.20 | [TK]D-Fender | badcfe: well you just throw out a question with no backup, and when asked for some just sit around silently. SHOW US WHAT YOU"RE DOING. |
| 14:06.24 | badcfe | [TK]D-Fender: im just one of those errouneous humans |
| 14:06.51 | badcfe | [TK]D-Fender: i rather not show that publicaly |
| 14:07.16 | [TK]D-Fender | badcfe: Quick lesson : The little bits you tell us AREN'T going to let us know where you screwed up. We need to see the WHOLE thing. |
| 14:07.23 | [TK]D-Fender | badcfe: OH?! |
| 14:07.56 | [TK]D-Fender | badcfe: Thats like asking a mechanic "whats wrong with my car?!" and then not letting him LOOK AT IT. The is completely retarded. |
| 14:08.02 | [TK]D-Fender | That* |
| 14:08.23 | badcfe | [TK]D-Fender: thing is that there so much confidential stuff in my dialplan |
| 14:08.47 | [TK]D-Fender | badcfe: Then suffer silently. |
| 14:08.48 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 14:09.07 | [TK]D-Fender | badcfe: Because we can't help you when we can't see the parts surrounding your problem. |
| 14:09.16 | badcfe | [TK]D-Fender: i agree with you. im sorry. ill try to work harder before spitting out my crappy fragments on irc |
| 14:09.17 | Nugget | exten => jimmyhoffa,1,Dial(SIP/supersecrethideout) |
| 14:09.32 | rob0 | ah HA There he is!! |
| 14:09.36 | Nugget | ack! |
| 14:09.51 | rob0 | And Teamsters |
| 14:09.59 | rob0 | And Federales |
| 14:10.04 | apocn | Hello, I want to allow agents to make calls trough a softswitch using: Dial(Sip/EXT@myhost.com), I modified the /etc/hosts to map myhost.com with the remote asterisk server, but it doesnt work still. stays retrying, any help? |
| 14:10.41 | *** join/#asterisk oej (n=olle@81-224-166-188-o1036.telia.com) |
| 14:10.50 | [TK]D-Fender | apocn: PASTEBIN the entire call at verbose 10, and SIP DEBUG enabled. |
| 14:10.52 | [TK]D-Fender | ~pb |
| 14:10.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:10.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 14:11.02 | apocn | ok |
| 14:11.38 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:11.38 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:13.42 | optize | `openvz |
| 14:14.53 | mocker | optize: You use it? |
| 14:14.57 | optize | yep |
| 14:15.00 | optize | on about 10 boxes |
| 14:15.05 | mocker | Works well for you? |
| 14:15.26 | mocker | (I guess if you installed it 10 times then it better) |
| 14:15.36 | tzafrir | mocker, you need a custom kernel? |
| 14:15.57 | mocker | For OpenVZ? |
| 14:16.06 | mocker | Yes. |
| 14:16.14 | tzafrir | If not, I figure that openvz / linux-vserver will give you the separation you need in a cheaper way |
| 14:16.15 | optize | I've never had any problems |
| 14:16.28 | mocker | optize: You do Asterisk on it? |
| 14:16.35 | optize | yep |
| 14:16.49 | mocker | how do you handle timnig? |
| 14:16.53 | *** join/#asterisk agx (n=AGX@88.34.216.63) |
| 14:16.57 | mocker | timing even. |
| 14:18.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:20.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.03 | *** join/#asterisk mog (n=mog@nat/digium/x-b81ec320889be623) |
| 14:21.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:22.12 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 14:22.37 | apocn | [TK]D-Fender : http://pastebin.com/d3f973dd0 |
| 14:24.13 | [TK]D-Fender | apocn: I'd like to see the ENTIRE call as requested... and please include the [general] section of sip.conf and any relevent peers masking only passwords |
| 14:24.16 | apocn | I register to "172.18.1.104" with my softphone, when I call *9000 this one connects with the asterisk installed on 172.18.0.32 |
| 14:25.07 | apocn | then when I dial the sub-menu ext. 2, this asterisk must forward me to 172.18.1.104 to the ext: 9001 |
| 14:25.42 | apocn | [TK]D-Fender : the problem is that there are a lot of calls getting in right now, and when I start debugging I get a lot of messages (not relevant to this test). |
| 14:26.12 | [TK]D-Fender | apocn: And cutting off stuff that might be. Let US filter through to see whats relevent |
| 14:26.28 | apocn | ok |
| 14:27.30 | [TK]D-Fender | apocn: And while you're at it, and *nix CLI please doa DNS resolution of "apocn.com" and include in the next pastebin to include along with everything else you've been requested for. |
| 14:30.29 | [TK]D-Fender | apocn: and "ifconfig" might help as well |
| 14:30.57 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 14:31.14 | *** join/#asterisk dezenten (i=dezent@unixgeek.biz) |
| 14:32.22 | *** join/#asterisk oej (n=olle@81-224-166-188-o1036.telia.com) |
| 14:32.26 | *** join/#asterisk booip (n=_none@novik.TeNeT.Odessa.UA) |
| 14:33.49 | dezenten | hello, anyone got asterisk 1.4.13 + postgresql + voicemail odbc-storage to work ? i followed the guide in doc/voicemail_odbc_postgresql.txt and it seems like everything works just fine untill i try to listen on my voicemails.. i get a "app_voicemail.c:1111 in retrieve_file: SQL Get Data error! [SELECT * FROM voicemessages WHERE dir=? AND msgnum=?]" error.. im using debian sarge |
| 14:35.24 | apocn | [TK]D-Fender : http://pastebin.com/d62cb6365 |
| 14:35.39 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:36.12 | apocn | [TK]D-Fender: didnt see the ifconfig: but: inet addr:172.18.0.32 Bcast:172.18.7.255 Mask:255.255.248.0 |
| 14:36.16 | apocn | on eth0 |
| 14:36.22 | [TK]D-Fender | apocn: And |
| 14:36.34 | [TK]D-Fender | apocn: And "ping apocn.com" gives you...? |
| 14:36.58 | apocn | oh |
| 14:37.07 | apocn | icmp_seq=1 Destination Host Unreachable |
| 14:37.14 | apocn | let me see if they are blocking icmp 8 |
| 14:37.22 | apocn | its on the local network, but from another department |
| 14:37.39 | apocn | they can see me though, cause when I call *9000 they connect to my asterisk |
| 14:38.15 | [TK]D-Fender | apocn: You should not be adding an IP/host to you dialstring. It should be going through your registrara |
| 14:38.26 | *** join/#asterisk Maliuta (n=nikolai@119.11.96.246) |
| 14:39.12 | apocn | [TK]D-Fender: even if all is in the lan? |
| 14:39.20 | [TK]D-Fender | apocn: always |
| 14:39.43 | [TK]D-Fender | apocn: targeting a server in your dial isn't normal. You phone should always be going through its registered server |
| 14:39.56 | McDouglas | how can i initiate a call from the cli? |
| 14:39.58 | apocn | ok |
| 14:40.28 | ai-a | McDouglas: you read The Book ? |
| 14:40.50 | McDouglas | ai-a: i dont tihnkit mentioned it |
| 14:41.06 | [TK]D-Fender | McDouglas: "help" |
| 14:41.10 | *** join/#asterisk Networkstudent (n=Sharagim@209.51.201.170) |
| 14:43.00 | ai-a | McDouglas: Page 340 |
| 14:43.15 | Odd_Bloke | Is "The Book" the O'Reilly "Asterisk: The Future of Telephony"? |
| 14:43.32 | ai-a | or even google.com/search?q=asterisk+console+dial |
| 14:44.27 | ai-a | Odd_Bloke: yes. |
| 14:44.37 | Odd_Bloke | Cool. |
| 14:44.50 | apocn | [TK]D-Fender : where can I read about it? |
| 14:45.05 | ai-a | apocn: Sip Register |
| 14:45.22 | ai-a | written about a lot in "The Book" |
| 14:45.29 | [TK]D-Fender | apocn: jsut remove the domain suffix when you dial. And you seem to have issues resolving your target anyways. |
| 14:45.30 | ai-a | and on www.voip-info.org |
| 14:47.37 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 14:50.42 | *** join/#asterisk infernix (n=nix@unaffiliated/infernix) |
| 14:51.12 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 14:51.29 | asteriskmonkey | is there a way to select which codec to use before dialing a sip peer? |
| 14:52.54 | dezenten | asteriskmonkey: you could choose that in your client |
| 14:53.46 | asteriskmonkey | no i dont need that |
| 14:54.02 | asteriskmonkey | i have to be able to choose between gsm and ulaw for one destination based on a call |
| 14:54.27 | asteriskmonkey | ie a did for a voice channel id set as ulaw and a did for a fax channel has to go ulaw |
| 14:54.34 | asteriskmonkey | sorry gsm/ulaw |
| 14:54.54 | apocn | [TK]D-Fender: yup... I was connecting to the wrong box. now when I connect to the right one: Got SIP response 500 "Server Internal Error" back from 172.16.104.27 |
| 14:55.22 | [TK]D-Fender | apocn: pastebin.... |
| 14:55.37 | [TK]D-Fender | asteriskmonkey: Yes.. yous et it IN your peer. |
| 14:56.18 | asteriskmonkey | [TK]D-Fender: ok so here is where im confused i have a peer setup,, ulaw and gsm how do i say in the dial code to choose gsm for on call and ulaw for another ... |
| 14:56.58 | [TK]D-Fender | asteriskmonkey: make 1 peer allowing only 1 codec, and another with a different codec, both witht he same auth user/pass |
| 14:57.28 | apocn | [TK]D-Fender : http://pastebin.com/d6ca50ec3 |
| 14:57.30 | asteriskmonkey | so no way of doing it in the dialplan |
| 14:58.15 | kaldemar | asteriskmonkey: once upon a time there was a variable called SIP_CODEC that could be set before the dial. |
| 14:58.35 | asteriskmonkey | is it still in 1.2x or is it gone? |
| 14:58.48 | kaldemar | go find out. |
| 14:59.27 | *** join/#asterisk Arcopix (n=Arcopix@pirus.securax.be) |
| 14:59.31 | Arcopix | hi all |
| 15:00.05 | [TK]D-Fender | apocn: Warning: 399 172.16.104.27 "Failed to Process INVITE for SCSCF" <- WTF |
| 15:00.16 | [TK]D-Fender | apocn: Thats an * on the other side of that call? |
| 15:00.28 | [TK]D-Fender | asteriskmonkey: Your peer chooses its codecs. |
| 15:00.51 | [TK]D-Fender | kaldemar: I believe that only applied to non-peer dials. |
| 15:00.56 | apocn | [TK]D-Fender : what do you mean? |
| 15:01.12 | [TK]D-Fender | apocn: You are calling direct from 1 * box to another? |
| 15:01.22 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 15:01.25 | apocn | [TK]D-Fender yes |
| 15:01.51 | [TK]D-Fender | apocn: You probably shouldn't be passing on un-authed calls between boxes... |
| 15:02.03 | [TK]D-Fender | apocn: Go lookup "asterisk dual servers" on the WIKI |
| 15:02.11 | apocn | ok |
| 15:07.14 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 15:09.00 | *** join/#asterisk dworkin (n=asdf@cpc2-barn4-0-0-cust142.brnt.cable.ntl.com) |
| 15:10.59 | muiro | hmm. The grammers I have in lumenvox.conf don't seem to be preloading. Am I missing something? |
| 15:16.14 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 15:17.20 | *** join/#asterisk gardo (n=gardo@121.97.104.33) |
| 15:17.52 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 15:18.23 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:18.46 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:20.29 | *** join/#asterisk GerbilTheNut (i=GerbilNu@72.48.176.41) |
| 15:21.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:21.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:22.48 | GerbilTheNut | is it possible to make a call through an asterisk system without being a registered user? |
| 15:22.49 | *** join/#asterisk bantu (n=Miranda@p54A32EEC.dip0.t-ipconnect.de) |
| 15:23.03 | GerbilTheNut | we're showing alot of calls yesterday from random usernames |
| 15:26.05 | [TK]D-Fender | GerbilTheNut: Show us something useful.... |
| 15:26.35 | *** join/#asterisk Vec (n=Vec@dsl-243-108-121.telkomadsl.co.za) |
| 15:26.37 | GerbilTheNut | i'm grepping the logs right not to try to find something useful, all i have right now are cdr's from Teliax and Voicepulse |
| 15:27.54 | [TK]D-Fender | GerbilTheNut: Well let us know when yuo have. |
| 15:29.03 | GerbilTheNut | Well here is one of the cdr entries, "","monglonasa","12515803047","default","monglonasa","SIP/monglonasa-40988570","IAX2/voicepulse02-3","Dial","IAX2/voicepulse02/12515803047","2007-12-17 23:15:40","2007-12-17 23:15:53","2007-12-18 00:11:25",3345,3332,"ANSWERED","DOCUMENTATION" |
| 15:29.48 | [TK]D-Fender | GerbilTheNut: And you have no [monglonasa] ? |
| 15:30.08 | GerbilTheNut | nope |
| 15:30.21 | [TK]D-Fender | GerbilTheNut: And thats your CDR is it? |
| 15:30.40 | *** join/#asterisk saftsack (n=saftsack@83.218.162.174) |
| 15:30.53 | GerbilTheNut | yes |
| 15:32.04 | dezenten | i got this bug with debian etch and asterisk 1.4.13 http://bugs.digium.com/view.php?id=8513 it says fixed by Corydon76 01-16-07 |
| 15:32.37 | GerbilTheNut | and this is the only one i see that also successfully made a call out that shows in the messages log |
| 15:32.38 | GerbilTheNut | Dec 17 23:52:42 NOTICE[12398] chan_sip.c: Registration from 'kokoko <sip:kokoko@72.xxx.xxx.xxx>' failed for '202.79.27.114' - Username/auth name mismatch |
| 15:33.57 | Vec | Does anyone know if its possible to adjust the dial plan and reload from someone dialing a number ? |
| 15:34.15 | mocker | Vec: Sure, look at System |
| 15:34.34 | Vec | mocker : thanks |
| 15:35.21 | mocker | Vec: Something like 1,1,System(asterisk -rx "sip reload") |
| 15:35.30 | mocker | Or extensions reload, etc.. |
| 15:35.36 | [TK]D-Fender | mocker: yeah, somehting LIKE that |
| 15:35.38 | VJFROMGT | can anyone tell me where to get firmware for my polycom 501? |
| 15:36.13 | [TK]D-Fender | VJFROMGT: From your reseller, or older versions at freedomphones.net/polycom |
| 15:36.24 | Vec | mocker : I would first have to modify extensions.conf somehow before the System(asterisk -rx) |
| 15:36.50 | mocker | [TK]D-Fender: ? |
| 15:36.53 | Vec | mocker : would I do that using awk, echo, grep etc, or are there better methods ? |
| 15:37.10 | [TK]D-Fender | Vec: Any way you feel like |
| 15:37.44 | [TK]D-Fender | Vec: "modify" is not a comprehensive description that gives us any idea what tool best serves the job. |
| 15:38.01 | Vec | [TK]D-Fender : what I mean is would I have to use linux console apps, or does asterisk have some other built in method, to include certain contexts and exclude others based on number dialed ? |
| 15:38.13 | mocker | Vec: You could even 'include' another conf file in your dialplan and just overwrite that. |
| 15:38.23 | [TK]D-Fender | Vec: "rm /etc/asterisk/extensions.conf" <- there's a nifty modification, in fact your file will be completely changed. |
| 15:38.41 | mocker | Vec: You might want to check out ~thebook |
| 15:38.42 | mocker | ~thebook |
| 15:38.43 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 15:38.45 | Vec | [TK]D-Fender : LOL, yeh |
| 15:38.48 | [TK]D-Fender | Vec: Taht will not impact the RUNNING call I hope you know... |
| 15:39.26 | Vec | [TK]D-Fender : yeh, don't worry, must be sounding dumber than I really am |
| 15:39.27 | *** part/#asterisk GerbilTheNut (i=GerbilNu@72.48.176.41) |
| 15:39.56 | *** join/#asterisk dswillia (n=me2@199.3.247.34) |
| 15:40.09 | Vec | mocker : I like your idea of includeing an external file in extensions.conf and then modifying that, less risky. |
| 15:40.33 | mocker | Vec: I don't know about that. :) |
| 15:40.47 | mocker | Just risky in different ways. |
| 15:42.20 | mocker | Vec: There are probably better ways to do what you want though, w/o dynamically writing a file. |
| 15:42.56 | dswillia | hey all, I am connecting an Avaya PBX to an asterisk server via a PRI into a Sangoma A102. I am wanting to use the * box as a conference bridge. I currently have no alarms on the T1 and when dialing the asterisk trunk I get "Extension '' in context 'from-zaptel' from '' does not exist. Rejecting call on channel 0/1, span 1" anyone have a clue what to look at, I find it strange the * box see's this as an extention |
| 15:43.44 | mocker | dswillia: did you configure your extensions.conf at all? |
| 15:43.46 | [TK]D-Fender | dswillia: Yeah, you set that context in zapata.conf as the context to send calls to and it does not xist in your dialplan. Fix it |
| 15:43.56 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 15:44.11 | mocker | dswillia: Also, sorry about the whole Avaya thing. |
| 15:44.15 | mocker | I feel your pain. :P |
| 15:44.23 | Vec | mocker : well thats what I'm trying to find out, all I want to do, have a message play when the bussiness closes, I can do it very easily time based using include, but the company wants to manually turn it on and off ? |
| 15:44.41 | [TK]D-Fender | Vec: then you are using the wrong aproach |
| 15:44.54 | dswillia | lol |
| 15:45.10 | [TK]D-Fender | Vec: Set a value on/off that states whether there is a manual override to your hours and check gfor it in your IVR |
| 15:45.35 | [TK]D-Fender | Vec: Where to start "show function DB", "show application gotoif" |
| 15:46.01 | Vec | [TK]D-Fender : Brillient, yeh, know how to do it, never thought of that, thanks alot. |
| 15:46.04 | dswillia | ok the from-zaptel does exist |
| 15:46.20 | dswillia | so do I need to create an extention ''? |
| 15:46.22 | [TK]D-Fender | dswillia: Well either make it exist, or pick a context that does |
| 15:46.32 | [TK]D-Fender | dswillia: And I think thats a resounding "duh!" |
| 15:48.20 | *** join/#asterisk angom (n=Angel@201.170.49.106) |
| 15:49.38 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 15:49.41 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 15:49.41 | asteriskmonkey | SetVar(SIP_CODEC=gsm) dosnt seem to work in the dialplan.. is this the right format in 1.2? |
| 15:51.29 | *** join/#asterisk toomba (i=phph@89.216.200.228) |
| 15:51.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:53.38 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 15:54.21 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 15:55.28 | tzafrir | asteriskmonkey, it is deprecated in 1.2 and doesn't work any more in 1.4 |
| 15:55.48 | tzafrir | use SET(VAR=value) instead |
| 15:56.04 | dezenten | seems like voicemail and odbc-storgage is deprecated also |
| 15:56.15 | asteriskmonkey | thanks tzafrir ill give that a try |
| 15:56.26 | mocker | dezenten: Ugh, don't say that. |
| 15:56.27 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 15:56.46 | tzafrir | dezenten, what makes you say that? |
| 15:58.34 | _x86_ | i've got a problem where sometimes analog stations just stop getting a dialtone or anything -- just dead air |
| 15:58.38 | dezenten | tzafrir: this bug is still valid http://bugs.digium.com/view.php?id=8513 |
| 15:58.51 | _x86_ | connected to a Rhino channel bank to a Sangoma A102D-x T1 card |
| 15:59.00 | _x86_ | anyone ever seen that? |
| 15:59.10 | asteriskmonkey | Set(SIP_CODEC=gsm) dosnt seem to work either :/ |
| 15:59.46 | _x86_ | multiple asterisk PBXs in multiple locations doing the same thing |
| 15:59.50 | _x86_ | same hardware setup |
| 15:59.58 | _x86_ | is this a bad channel bank or something? |
| 16:00.06 | tzafrir | dezenten, that is an issue that is listed as close. It has been resolved almost a year ago |
| 16:00.31 | dezenten | tzafrir: i got the same error here with 1.4.13 and 1.4.15 |
| 16:02.58 | _x86_ | happening quite a bit |
| 16:02.59 | _x86_ | this sucks |
| 16:03.02 | asteriskmonkey | is sip_codec removed in 1.2 |
| 16:03.06 | asteriskmonkey | I cant get it to work :/ |
| 16:04.12 | toomba | hi all, can somebody help me with a cisco AS5300 and T1 PRIs, I keep getting 500 Internal Server Error back from it when I send it a SIP call for PRI termination |
| 16:04.32 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 16:04.35 | shadebob | hi |
| 16:04.53 | _x86_ | this is a really bad situation, they're wanting to take asterisk out and put mitel in |
| 16:05.42 | asteriskmonkey | _x86_: call sangoma your using there hardware |
| 16:06.13 | shadebob | I have problem with some installation of asterisk 1.4.14 on Core Duo Mac Mini and Debian Etch. Randomly I have dropped calls. I haven't absolute timeout or other options... This problem happen on many installation... Some ideas? |
| 16:06.48 | Corydon76-dig | _x86_: yeah, it honestly sounds like your cards went bad |
| 16:07.11 | _x86_ | all at the same time, in geographically diverse locations? |
| 16:07.18 | _x86_ | in 3 different servers? |
| 16:07.22 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 16:07.30 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 16:07.38 | Corydon76-dig | Let me guess. Was it the day after your hardware warranty expired? |
| 16:07.49 | _x86_ | nope, it's all brand new, or fairly brand new |
| 16:08.00 | *** join/#asterisk Akira65 (n=Akira65@84.16.28.38) |
| 16:08.05 | Akira65 | shadebob, hi |
| 16:08.14 | Corydon76-dig | What else did you upgrade around that time? |
| 16:08.20 | _x86_ | nothing |
| 16:08.31 | Corydon76-dig | Could be the card, could be the driver |
| 16:08.34 | _x86_ | it was working fine, and all of a sudden three offices are having the same issue |
| 16:08.37 | Akira65 | shadebob, can you help me ? |
| 16:08.45 | Akira65 | shadebob, my asterisk doesn't work... :( |
| 16:08.50 | Corydon76-dig | In either case, you need to contact the manufacturer |
| 16:08.57 | _x86_ | wondering if it's a problem with the channel bank? |
| 16:09.16 | Corydon76-dig | 3 different channel banks went bad at the same time? |
| 16:09.24 | [TK]D-Fender | _x86_: Are you that think? just SWAPT THEM and find out. |
| 16:09.29 | [TK]D-Fender | thick* |
| 16:09.37 | [TK]D-Fender | blarg, can't type again today. |
| 16:09.42 | _x86_ | nearest office is ~1.5 hours away |
| 16:09.48 | [TK]D-Fender | _x86_: You have multiple CB, SWAP IT and see |
| 16:09.56 | Corydon76-dig | You could also swap out to a different brand of CB |
| 16:10.03 | Corydon76-dig | They are commodity goods, after all |
| 16:10.25 | [TK]D-Fender | _x86_: You can't play the hypothetical game until you're blue and face and be no further ahead that you are now. Go do something. |
| 16:10.45 | [TK]D-Fender | s/can't/can/ |
| 16:10.51 | [TK]D-Fender | asdasdklhdfldfg |
| 16:11.16 | [TK]D-Fender | shadebob: some DETAILS would be helpful......... |
| 16:11.29 | [TK]D-Fender | Akira65>shadebob, my asterisk doesn't work... :( <---- you too. |
| 16:11.29 | tzafrir | shadebob, what type of calls? SIP? IAX? Zap? |
| 16:11.38 | *** join/#asterisk d-k-t (n=dt@60.176.198.63) |
| 16:12.30 | shadebob | SIP or ZAP through Xorcom |
| 16:12.59 | shadebob | with SIP Phone SPA921/941, Thomson 2030 or SJphone |
| 16:13.16 | dezenten | Corydon76-dig: i got this bug http://bugs.digium.com/view.php?id=8513 with debian etch, postgresql-8.2.5, debian etch odbc packages.. and i been following the doc/voicemail_odbc_postgresql.txt |
| 16:13.18 | shadebob | I have good lans with 2ms status in SIP show peers |
| 16:13.30 | dezenten | Corydon76-dig: any idea ? |
| 16:13.55 | dezenten | and asterisk 1.4.15 |
| 16:14.06 | shadebob | but my calls are randomly dropped... in more than 10 differents servers.... |
| 16:14.12 | Corydon76-dig | dezenten: nope |
| 16:14.15 | dezenten | k |
| 16:14.45 | dezenten | we send a patch if we figure it out.. |
| 16:16.12 | shadebob | tzafrir, [TK]D-Fender : any ideas? |
| 16:16.33 | tzafrir | shadebob, A Xorcom Astribank, you mean? |
| 16:16.37 | [TK]D-Fender | shadebob: you've shown us NOTHING. |
| 16:16.53 | tzafrir | (Zap) |
| 16:17.06 | shadebob | tzafrir, Yes 8 Fxo but also with SIP telco |
| 16:17.37 | tzafrir | shadebob, anything in the logs regarding the disconnection? |
| 16:18.23 | shadebob | tzafrir : a normal hangup |
| 16:18.24 | tzafrir | a "random disconnect" with respect to a call that passes through FXO may be due to the use of busydetect |
| 16:18.40 | tzafrir | do you have busydetect enabled? |
| 16:18.55 | tzafrir | If so: do you actually need it? |
| 16:18.55 | shadebob | tzafrir :Sure, but SIP Telco laso disconnect.... |
| 16:19.33 | tzafrir | ok. If you want to know what really triggered things, enable verbose and debug logging |
| 16:19.45 | tzafrir | How often are those disconnects? |
| 16:20.25 | tzafrir | then post / pastebin a log snippet around the hangup (warning: it will be very verbose) |
| 16:20.29 | shadebob | tzafrir : OK. I will catch all the SIP and RTP debug. Randomly... Maybe on 1/6 |
| 16:20.39 | tzafrir | no need for sip debug |
| 16:20.43 | _x86_ | sangoma is no help |
| 16:20.53 | shadebob | tzafrir : just RTP ? |
| 16:20.54 | tzafrir | but do log "debug" as well in logger.conf |
| 16:20.58 | tzafrir | no. |
| 16:21.17 | shadebob | tzafrir : OK, I will now on a customer |
| 16:21.23 | [TK]D-Fender | _x86_: You have no physical access, you have little to test. |
| 16:22.41 | *** part/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:22.55 | tzafrir | [TK]D-Fender, any idea if anybody has already documented that bit of using asterisk debug logging to trace the source of a hangup? |
| 16:23.26 | tzafrir | I've explained it too many times here but I can't think of a generic way to write it |
| 16:23.31 | [TK]D-Fender | tzafrir : not a clue... |
| 16:23.53 | [TK]D-Fender | tzafrir : Frankly I like seeing CLI output with channel debug enabled |
| 16:24.38 | [TK]D-Fender | tzafrir : And CONFIGS. You still haven't seen his zapata, etc |
| 16:27.23 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 16:28.12 | *** join/#asterisk saftsack (n=saftsack@83.218.162.174) |
| 16:33.40 | _x86_ | [TK]D-Fender: rebooted the server, still same problem... rebooted the channel bank, still same problem |
| 16:33.49 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:34.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:36.00 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 16:36.13 | [TK]D-Fender | _x86_: SWAP |
| 16:36.32 | [TK]D-Fender | _x86_: And test the individual channels and wiring |
| 16:37.06 | *** join/#asterisk atlantia (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 16:37.23 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:38.18 | _x86_ | [TK]D-Fender: NOT POSSIBLE |
| 16:38.24 | atlantia | [TK]D-Fender: hey it's nny, having nick issues. Had an opportunity I wanted to discuss, was wondering if I could PM you... |
| 16:38.29 | _x86_ | IM 1.5 HOURS AWAY |
| 16:38.30 | _x86_ | ;) |
| 16:38.48 | [TK]D-Fender | _x86_: And you've wasted how much time here doing nothing? :) |
| 16:39.05 | [TK]D-Fender | atlantia: Sure |
| 16:39.17 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-f34238af5adac475) |
| 16:40.08 | asteriskmonkey | agggg SIP_CODEC dosnt work on iax channels does it |
| 16:40.44 | [TK]D-Fender | asteriskmonkey: make. a. PEER |
| 16:41.05 | asteriskmonkey | back to earler convo i dont want to have 2 peers for something i should be able to do in dialplan |
| 16:41.43 | asteriskmonkey | if thats my only op though grrr... feature request |
| 16:42.18 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:42.27 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:44.43 | *** join/#asterisk ExplodingLemur (n=change@72.29.166.25) |
| 16:45.46 | apocn | Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx. Any hints? |
| 16:45.57 | shadebob | Re, I have question on multi-registration on the same SIP server. For example, If I have 4 accounts on sip.digium.com all incoming calls come in the last registered context... For the moment I create a ugly patch wich rewrite the context on the fly regarding the p->domain. No other clean solution for this problem? |
| 16:46.12 | apocn | this is when asterisk tries to register |
| 16:48.17 | ExplodingLemur | shadebob, multiple registrations from different devices on the same account, or multiple different accounts registering from the same device? |
| 16:48.23 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:48.56 | shadebob | ExplodingLemur: Register one asterisk box with multi accounts on the same SIP provider |
| 16:49.51 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 16:51.58 | [TK]D-Fender | apocn: your 2 systems don't like the amount of time set for re-registration. |
| 17:00.51 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:01.42 | ExplodingLemur | shadebob: Not sure, I'm actually having a similar problem... |
| 17:02.03 | ExplodingLemur | Not sure if mine is a NAT issue, I have several Snom 320 phones behind NAT registering to an Asterisk box outside of NAT. |
| 17:02.14 | ExplodingLemur | One of the phones has three different accounts set up on it, all registering to the Asterisk box. |
| 17:02.41 | ExplodingLemur | Incoming calls get to the right line appearance on the Snom phone. Outgoing calls only work from the last registered account though. |
| 17:03.46 | ExplodingLemur | I have STUN and keepalives set on the phones. It appears that the registrations are coming from the same source port on the Snom phone, which is likely why Asterisk is only recognizing the last account as properly registered. |
| 17:04.00 | ExplodingLemur | Any ideas as to how to get the Snom phones to not behave like this, or is something else the issue here? |
| 17:05.33 | shadebob | ExplodingLemur: It's not a NAT issue. Problem is asterisk send incoming call in the context based on the domain... For example if account 1234 is registered to 84.16.20.20 and have incoming context [sip_operator], all the another account registered to 84.16.20.20 will be sent in this context. |
| 17:05.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:05.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:07.28 | shadebob | ExplodingLemur: I solve this problem with ugly code in chan_sip ( in get_destination() ) |
| 17:07.31 | toomba | anyone familiar with T1 PRIs and Cisco AS5300 and sending calls from SIP to PSTN through it? |
| 17:12.24 | shadebob | ExplodingLemur: my UGLYYYY patch... but working for multiregistration : http://www.pastebin.org/12053 |
| 17:12.39 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 17:12.42 | loompek | morning |
| 17:13.05 | shadebob | ExplodingLemur: I have to create a context with the same name of my sip account (for example account 1234 must have a context=1234) |
| 17:13.12 | *** join/#asterisk pepo-- (n=pepOSX@190.79.246.105) |
| 17:13.36 | loompek | anyone of you guys uses suncomm sc-385? |
| 17:13.38 | shadebob | ExplodingLemur: with this you will make multi registration work. But I search a "clean" solution |
| 17:14.15 | ExplodingLemur | shadebob: I think our problems may be different, then... |
| 17:14.26 | shadebob | ExplodingLemur: ok ;) |
| 17:16.05 | ExplodingLemur | I have a bunch of phones behind the same IP, all of the single-extension phones are working (and they show as the same IP to * but different ports), the phones with more than 1 account set up are not though (as they're sourcing from the same ports, so * thinks it's the same device registering 3 different times, and only the last registered account works) |
| 17:16.25 | Qwell | ExplodingLemur: type=friend |
| 17:19.16 | FlatFoot | ExplodingLemur: Which Snom phone and what Version ? |
| 17:19.35 | FlatFoot | ExplodingLemur: There are NAT issues with some versions |
| 17:20.22 | FlatFoot | ExplodingLemur: Snom 320 use 7.1.19 |
| 17:21.24 | ExplodingLemur | FlatFoot: Snom320, using 6.5.15...I'll try the 7.x non-beta release that just came out |
| 17:21.55 | jameswf | so fonality's explination of there phone home tool is to be used as a way to allow them to go to vendors and say hey people are using our stuff for your hardware pay up... I want whoever at digium who gets that call to record it and all that follows |
| 17:21.58 | FlatFoot | ExplodingLemur: I use 7.1.19 on all 20 320's and they work NAT or not very well |
| 17:22.16 | FlatFoot | *not , very well |
| 17:24.07 | *** join/#asterisk fskrotzki (n=fskrotzk@74.39.192.254) |
| 17:24.41 | *** join/#asterisk funxion (n=x@63.214.236.169) |
| 17:27.57 | *** join/#asterisk cappiz (n=cappiz@217.68.105.60) |
| 17:29.18 | cappiz | i've made a sip device, and i want to use that login as a trunk on another asterisk server i have... what is the correct register string for my trunk? i've tried with user:pass@host |
| 17:29.21 | cappiz | but it doesnt register |
| 17:30.13 | Qwell | jameswf: you really think they'd call us? |
| 17:30.24 | Qwell | step back and think about that for about 2 seconds :) |
| 17:31.23 | *** join/#asterisk markgreene (n=Mark_Gre@rrcs-24-227-107-130.se.biz.rr.com) |
| 17:31.38 | jameswf | you know they probably will thats just how they are |
| 17:32.15 | markgreene | Hey everyone. Where can I find a list of applications that were removed in 1.4, and a complimenting list of the applications that replaced them. I am moving from 1.2 to 1.4 |
| 17:32.49 | jameswf | I think philiepe or rob for that matter should call them and be like ok dude we are 80% of you so we want 8% of what you collect |
| 17:33.19 | jameswf | s/8%/80%/ |
| 17:33.37 | *** part/#asterisk trashguy (n=trashguy@MSFC.calarts.edu) |
| 17:33.49 | jameswf | ~botsnack |
| 17:33.49 | jbot | :), jameswf |
| 17:36.55 | drmessano | I was expecting this test box I have here to start dialing 911 and ordering pizzas after 12am last night |
| 17:37.02 | drmessano | Guess it really is anonymous |
| 17:38.10 | jameswf | there is a perlscript called pizzaparty you could tie in to asterisk to have it order dominos |
| 17:38.28 | drmessano | Hmmm |
| 17:38.34 | drmessano | Sure its not already in TB? |
| 17:38.44 | drmessano | with coupon.jpg |
| 17:38.59 | Qwell | jameswf: combine that with the Lumenvox grammar and script for pizza creation, and you've got a million^W$100 idea. |
| 17:39.52 | drmessano | lol |
| 17:40.27 | jameswf | I was seriously considering doing an AGI and makinf some punk kid record the prompts |
| 17:40.36 | Qwell | jameswf: gold |
| 17:40.39 | drmessano | lol |
| 17:40.49 | drmessano | I still want to do the 17 yr old girl |
| 17:40.53 | drmessano | err |
| 17:40.58 | drmessano | as in "create an IVR" |
| 17:41.17 | drmessano | "Yeah, like, so like" |
| 17:42.28 | drmessano | Record all the prompts, put them on the web somewhere |
| 17:42.29 | drmessano | use them for businesses that hire high schoolers, but without the tardyness and drama |
| 17:46.20 | citats | what a great quote. <drmessano> I still want to do the 17 yr old girl |
| 17:46.43 | mvanbaak | ut-oh |
| 17:46.49 | drmessano | rofl |
| 17:46.56 | mvanbaak | my asterisk box at home is making weird noises |
| 17:47.07 | drmessano | weird noises? |
| 17:47.12 | [TK]D-Fender | mvanbaak: WD-40 |
| 17:47.27 | mvanbaak | *screach* *clickclack* *screeeeeeeeeeeeeeeeaaaaaaaaaaaaaaaaaaaaaaaaaaach* *clickclack* |
| 17:47.29 | _x86_ | [TK]D-Fender: so i called Rhino, and they think it is an issue with the channel bank |
| 17:47.38 | _x86_ | [TK]D-Fender: the guy told me most likely it was a "surge" |
| 17:47.39 | drmessano | Ewwww |
| 17:47.46 | *** join/#asterisk RoyK (n=roy@ip-247-8-149-91.dialup.ice.no) |
| 17:47.57 | drmessano | You can probably reapir the HD |
| 17:48.00 | drmessano | repair |
| 17:48.16 | _x86_ | [TK]D-Fender: of course I've got the rhino plugged into a brand new 3000VA APC UPS, and the only thing plugged into the channels are regular analog phones.... |
| 17:48.19 | drmessano | When its clanking really bad, hit it with a rubber mallet |
| 17:48.21 | drmessano | it will stop |
| 17:48.42 | mvanbaak | stop working ? |
| 17:48.47 | _x86_ | [TK]D-Fender: he still insisted that it could be an "impendant surge" heh |
| 17:48.47 | drmessano | that too |
| 17:48.49 | Qwell | _x86_: in their defense, it isn't *that* unreasonable to say a surge did it. If, however, that's just their fall-back excuse, then... |
| 17:48.51 | drmessano | But no more noise |
| 17:48.58 | mvanbaak | ah |
| 17:49.03 | Qwell | (of course, it's *far* more likely to occur with fxo) |
| 17:49.08 | _x86_ | jameswf: ah :) |
| 17:49.13 | [TK]D-Fender | _x86_: and you still aren't on-site over 2 hours later and have nothing tangeable to show for it |
| 17:49.14 | _x86_ | jameswf: i still find it hard to believe, however |
| 17:49.26 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 17:49.39 | _x86_ | Qwell: yeah with FXO i could totally see a surge coming in from the CO |
| 17:49.42 | mvanbaak | hhmm, smart output and other sensors say it's fine |
| 17:49.57 | mvanbaak | ah well |
| 17:49.57 | Qwell | _x86_: lightning could quite easily hit the line from the phone... |
| 17:50.08 | mvanbaak | I have some spare parts, and it's all RAID-1 |
| 17:50.10 | _x86_ | Qwell: lightning hit the phone? ;) |
| 17:50.11 | Qwell | (that would assume there...you know...was lightning) |
| 17:50.22 | _x86_ | in the middle of winter? :) |
| 17:50.31 | Qwell | sure :p |
| 17:50.31 | drmessano | Thundersnow |
| 17:50.37 | Qwell | I mean, there would have to have been lightning, but...heh |
| 17:50.40 | _x86_ | ah right, that dangerous thundersnow ;) |
| 17:50.46 | Qwell | (of course, there are other ways - lightning is just more common) |
| 17:50.47 | mvanbaak | maybe some cableprotection froze off ? |
| 17:50.51 | drmessano | Kills 3 people every year |
| 17:50.52 | jameswf | I spent 5 years installing toshinba pbx's I never went on a service call for the toshiba always something the user did |
| 17:50.55 | drmessano | and one channel bank |
| 17:51.24 | _x86_ | jameswf: yeah, i'm starting to get to the point where i dont think i'll be recommending asterisk as a production-ready PBX |
| 17:51.46 | _x86_ | jameswf: i'm running 8 asterisk servers at work, and they work great most of the time, but support when there is a weird issue is tricky |
| 17:51.46 | *** join/#asterisk Absorto (n=user@189.141.90.177) |
| 17:52.12 | mvanbaak | _x86_: that's where ABE comes to play |
| 17:52.17 | _x86_ | jameswf: for example, this issue... sangoma makes the T1 card in the server connected to the Rhino channel bank |
| 17:52.30 | _x86_ | mvanbaak: we do not qualify for ABE |
| 17:52.31 | jameswf | I dont know asterisk is used by the dod and quite a few government agencies along with thousands of big companies so it isnt asterisk |
| 17:52.46 | drmessano | Half of supporting Asterisk is knowing how to isolate the hardware and software to know WHAT and WHO to ask |
| 17:52.47 | _x86_ | jameswf: i bet they use SIP phones too ;) |
| 17:52.52 | drmessano | Otherwise, youre not doing it right |
| 17:52.58 | _x86_ | i've never had this many issues with asterisk using SIP phones |
| 17:53.00 | jameswf | sounds like you have a fonality box |
| 17:53.11 | _x86_ | drmessano: right |
| 17:53.24 | drmessano | So thats an onsite IT problem |
| 17:53.29 | drmessano | Not an Asterisk issue |
| 17:53.36 | _x86_ | jameswf: HP DL380 G5 with Sangoma A102D-x cards (2) and Sangoma A20002D-x (1) |
| 17:54.02 | jameswf | ok normaly sangoma card + channelbank = fonality... |
| 17:54.03 | _x86_ | drmessano: nothing had changed and it's been working flawlessly for >2 months |
| 17:54.19 | mvanbaak | 18:54 < _x86_> drmessano: nothing had changed and it's been working flawlessly for >2 months |
| 17:54.20 | _x86_ | jameswf: ah, never used them... will they do support? |
| 17:54.22 | drmessano | Thats like telling me its so hard to figure out air conditioning problems when you cant figure out if the power is even on |
| 17:54.25 | mvanbaak | that's not possible ;) |
| 17:54.45 | mvanbaak | if nothing changes, a working system wont start acting up |
| 17:54.47 | _x86_ | mvanbaak: obviously it is... no configuration changes, not even any moves/adds/changes |
| 17:54.55 | drmessano | Something changed |
| 17:54.57 | _x86_ | mvanbaak: that's incorrect, because it happened today |
| 17:55.00 | drmessano | This isnt Windows and OS Rot |
| 17:55.04 | mvanbaak | so something changed |
| 17:55.17 | _x86_ | yeah, the thundersnow came in and struck the users' phones while they weren't looking! |
| 17:55.23 | _x86_ | that's exactly what happened |
| 17:55.38 | _x86_ | jameswf: tell me how this impendant surge thing works |
| 17:55.43 | drmessano | Did you reset the power on everything? |
| 17:55.44 | mvanbaak | maybe someone had sex in the office and hit a glass of water that was spilled on a phone |
| 17:55.47 | _x86_ | drmessano: i did |
| 17:55.54 | drmessano | Did you reboot the server? |
| 17:56.19 | drmessano | Was the company Xmas party last night, and did you find some stray panties anywhere? |
| 17:56.39 | jameswf | something changed :) bad cabling someone somewhere didn't stay far enough away from a power source or light fixture... reguardless we offered to swap out and inspect the channelbanks because even though I am a god and right 98% of the time there is still the other 3% of the time where I am wrong |
| 17:57.08 | _x86_ | drmessano: yes, i power cycled everything in the voice path |
| 17:57.31 | mvanbaak | are all components properly grounded and UPS protected ? |
| 17:57.31 | drmessano | 98% of the time you are right, and the other 2% you haven't figured out how to prove? |
| 17:57.32 | drmessano | lol |
| 17:57.42 | _x86_ | mvanbaak: yes, APC UPS... 300VA |
| 17:57.44 | _x86_ | 3000* |
| 17:57.57 | jameswf | drmessano: it was an ego joke |
| 17:58.01 | drmessano | I know |
| 17:58.03 | _x86_ | jameswf: hah ;) |
| 17:58.05 | drmessano | I was rephrasing |
| 17:58.24 | _x86_ | so what else could it possibly be? |
| 17:58.31 | mvanbaak | did you do tests with the minimal amount of hardware attached ? |
| 17:58.34 | drmessano | 98% of the time I am right and I havent figured out how to prove the other 2% |
| 17:58.37 | drmessano | nevermind |
| 17:58.38 | _x86_ | they get no power to the phone at all |
| 17:58.52 | _x86_ | no voltage coming from the wall jack, when it was working fine yesterday |
| 17:58.56 | drmessano | How old is the UPS |
| 17:59.00 | jameswf | oic I cant read the english so good |
| 17:59.06 | _x86_ | ~6 months or so |
| 17:59.11 | _x86_ | jameswf: lol |
| 17:59.28 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:59.39 | drmessano | UPS's aren't perfect.. 99.9% of the time they are |
| 17:59.48 | drmessano | APC's are little slow to clamp |
| 17:59.49 | jameswf | like me :) |
| 17:59.55 | mvanbaak | lol |
| 18:00.05 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 18:00.28 | *** part/#asterisk ExplodingLemur (n=change@72.29.166.25) |
| 18:00.33 | drmessano | Usually an APC makes a good fuse, but sometimes they're too slow and something ends up zapped |
| 18:00.42 | _x86_ | but if the UPS was the problem (at two geographically diverse sites, overnight), wouldn't other stuff connected to the same UPS also have been affected? |
| 18:00.49 | drmessano | No |
| 18:01.05 | mvanbaak | depends on the PSU in those other components |
| 18:01.13 | _x86_ | hmm |
| 18:01.17 | _x86_ | so Rhino PSU = fail? |
| 18:01.27 | drmessano | I work in broadcast.. ive seen spikes and lightning do shit that makes less sense than george bush |
| 18:01.51 | _x86_ | hmm |
| 18:01.52 | mvanbaak | drmessano: now that's a bit too much. that is plain impossible |
| 18:01.59 | drmessano | ROFL |
| 18:02.02 | _x86_ | kinda what i was thinking too |
| 18:02.38 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:02.42 | drmessano | I will tell you that an APC 120V unit will properly blow out when 208 is applied to it, and protect the connected equipment |
| 18:03.03 | mvanbaak | it will |
| 18:03.08 | mvanbaak | we found out the hard way |
| 18:03.12 | drmessano | yes, we did too |
| 18:03.14 | drmessano | ROFL |
| 18:03.34 | drmessano | miswired transfer switch? |
| 18:03.38 | mvanbaak | yup |
| 18:03.47 | drmessano | ha |
| 18:03.50 | mvanbaak | *p00f* |
| 18:03.52 | drmessano | same here |
| 18:03.59 | drmessano | Weekly test, 1am Monday |
| 18:04.02 | drmessano | KA... POW |
| 18:04.10 | mvanbaak | but after restoring the error everything booted again without issue |
| 18:04.12 | _x86_ | hah |
| 18:04.28 | _x86_ | well i have no idea what else it could be |
| 18:04.30 | drmessano | We lost about $3000 in misc crap |
| 18:04.42 | mvanbaak | ouch |
| 18:04.43 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:04.44 | _x86_ | it can't be the PCI bus, because all of the other channels work fine |
| 18:05.06 | _x86_ | i dont think the sangoma supplies voltage to each channel... isn't that a channel bank thing? |
| 18:05.30 | [TK]D-Fender | _x86_: You have multiple channel banks, right? |
| 18:05.45 | _x86_ | i've got ~16 of them |
| 18:05.49 | _x86_ | 2 at each site |
| 18:05.51 | mvanbaak | switch some |
| 18:05.52 | [TK]D-Fender | _x86_: then SWAP 2 of them |
| 18:06.00 | mvanbaak | to see if the problem moves with them |
| 18:06.07 | _x86_ | hmm |
| 18:06.09 | [TK]D-Fender | _x86_: Like I've been telling you for the last few hours |
| 18:06.10 | _x86_ | yeah good idea |
| 18:06.13 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:06.15 | mvanbaak | lol |
| 18:06.18 | _x86_ | [TK]D-Fender: indeed |
| 18:06.24 | mvanbaak | damn, you didn't even try that ? |
| 18:06.32 | _x86_ | i'm ~100 miles away |
| 18:06.34 | drmessano | ROFL |
| 18:06.35 | [TK]D-Fender | ~cluebat _x86_ |
| 18:06.35 | jbot | ACTION pulls out a ClueBat (tm) and thwaps _x86_. |
| 18:06.39 | drmessano | 100 miles? |
| 18:06.44 | drmessano | Youre wasting time |
| 18:06.46 | drmessano | Start driving |
| 18:06.49 | mvanbaak | want my 'troubleshooting for dummies' book ? |
| 18:06.52 | dswillia | what is the easiest way to force all incomming calls on a particular trunk to the meetme conference bridge? |
| 18:06.53 | _x86_ | was trying everything else before having to call the sales manager and walk him through swapping channel banks |
| 18:07.03 | [TK]D-Fender | drmessano: Told him that too hours ago... |
| 18:07.03 | drmessano | ROFL |
| 18:07.10 | _x86_ | [TK]D-Fender: two* |
| 18:07.22 | drmessano | Christ, youre not even on site and youre trying to fix it by arguing on IRC about it? |
| 18:07.24 | drmessano | FAIL |
| 18:07.30 | mvanbaak | lol |
| 18:07.35 | _x86_ | well if it required an all-day trip, i wanted to make absolutely sure that was the issue |
| 18:07.39 | [TK]D-Fender | _x86_: You mean swapping a measly wiring block connector & RJ48? Oh noes! |
| 18:07.46 | _x86_ | drmessano: not arguing about anything thus far |
| 18:07.46 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 18:07.51 | [TK]D-Fender | _x86_: Thad'd take like SECONDS! |
| 18:07.57 | _x86_ | [TK]D-Fender: normally yes |
| 18:08.10 | _x86_ | but this guy isn't the smartest one ;) |
| 18:08.13 | mvanbaak | 100 miles is nowhere near an all-day trip |
| 18:08.20 | mvanbaak | you can drive that in about an hour |
| 18:08.26 | _x86_ | 200 miles round trip, plus time spent on-site |
| 18:08.32 | mvanbaak | swap boxen, which takes another hour |
| 18:08.35 | [TK]D-Fender | _x86_: At least he doesn't have to worry about you talking over his head much then ;) |
| 18:08.38 | mvanbaak | and drive home, another hour |
| 18:08.38 | drmessano | [13:07] <_x86_> but this guy isn't the smartest one ;) <--- Right now you need to prove yourself too |
| 18:08.41 | mvanbaak | that's 3 hours |
| 18:08.45 | _x86_ | drmessano: *nod* |
| 18:08.48 | cpm | if it's in the mid-atlantic I95 corridor, it's all day. |
| 18:08.51 | mvanbaak | normal working days for the lazy ppl has 8 hours |
| 18:08.58 | _x86_ | drmessano: perhaps the blue smoke is seeping out of the servar! |
| 18:09.02 | drmessano | Some people work 8 hours? |
| 18:09.12 | drmessano | Isnt that part time? |
| 18:09.20 | mvanbaak | drmessano: I've heard the stories yes |
| 18:09.33 | _x86_ | hmm maybe the flux capacitor isn't receiving the full 1.21 jiggawatts |
| 18:09.36 | drmessano | Damn, I bet they sleep like 6 hours too |
| 18:09.47 | mvanbaak | drmessano: I heard 8 |
| 18:09.53 | drmessano | HA.. thats BS |
| 18:10.12 | mvanbaak | no, some health guru once told you need 8 hours of sleep every night |
| 18:10.17 | drmessano | Thats like a third of the day |
| 18:10.24 | mvanbaak | yup |
| 18:10.36 | mvanbaak | so most ppl waste 1/3 of their life by sleeping |
| 18:10.55 | drmessano | Thats at least two hours they could have spent on IRC |
| 18:11.04 | mvanbaak | 2 ? 7 |
| 18:11.09 | ice_croft | yeah.. the other are like zombies |
| 18:11.25 | drmessano | lol |
| 18:11.31 | badcfe | if you wait for a zombie it disapears |
| 18:11.32 | mvanbaak | moooooooooooo |
| 18:11.59 | *** join/#asterisk moprilo (n=jjohn@201.198.78.23) |
| 18:12.05 | mvanbaak | I wonder how much sleep my sip channels get. I see a lot of -<ZOMBIE> notices on my CLI from time to time |
| 18:12.13 | cappiz | i've setup my asterisk... i can call into it.. but i get no voice... what is the most common mistake? |
| 18:12.25 | drmessano | not speaking |
| 18:12.31 | mvanbaak | hahahaha |
| 18:12.32 | cappiz | haha... besides that.... |
| 18:12.35 | mvanbaak | NAT |
| 18:12.40 | drmessano | you said MOST common |
| 18:12.49 | moprilo | anyone know if i can get incomming international SMS..? i need to receive SMS.. |
| 18:13.01 | drmessano | Yes, NAT |
| 18:13.38 | mvanbaak | moprilo: ask your telco |
| 18:13.45 | drmessano | Thats going to be chapter 3 in my "Chronicles of VoIPia" book |
| 18:13.46 | cappiz | drmessano, let me try then |
| 18:14.26 | mvanbaak | NAT = No Audio for Telephony |
| 18:14.57 | drmessano | "... he spaketh to his his only son.. but no words cometh from his mouth.. "NAT" his son spaketh, and indeed it was" |
| 18:15.19 | Nugget | heh |
| 18:15.34 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 18:18.14 | dswillia | ok still getting Extension '' in context 'from-zaptel' from '' does not exist. Rejecting call on channel 0/1, span 1, what exactly am I looking for in the extensions.conf file to create extention '' |
| 18:18.24 | cappiz | Dec 18 19:17:17 DEBUG[23799] chan_sip.c: Setting NAT on RTP to 524288 |
| 18:18.39 | cappiz | do you think that has anyhting to do with it? |
| 18:22.14 | cappiz | found the error :P |
| 18:22.24 | [TK]D-Fender | dswillia: make sure you have "immediate=no" for your zaptel channel. |
| 18:22.58 | [TK]D-Fender | dswillia: And what signalling are you using? |
| 18:23.14 | drmessano | Digium needs to be working on a SIP <> IAX converter that fits in an RJ-45 Coupler |
| 18:23.20 | drmessano | Worlds problems solved |
| 18:24.54 | dswillia | this t1 is comming from an avaya pbx signaling is set to pri_cpe |
| 18:24.55 | cappiz | is there some global codes for disabling call forwarding by dtmf codes? i want to disable the forwarding from my SIP provider |
| 18:24.55 | russellb | ha |
| 18:25.15 | [TK]D-Fender | dswillia: Set the parameter I told you to and restart * |
| 18:27.35 | drmessano | Hmmm |
| 18:27.38 | drmessano | and I got a name for it too |
| 18:27.50 | drmessano | The SPIAX |
| 18:28.54 | mvanbaak | lol |
| 18:29.38 | drmessano | Thats almost as good as my FXO <> FXS converter |
| 18:29.38 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:29.45 | drmessano | Never buy the wrong ATA again! |
| 18:35.19 | cappiz | cant different devices hva different dialing plans? |
| 18:35.30 | cappiz | slash routes |
| 18:37.44 | mvanbaak | sure they can |
| 18:37.45 | [TK]D-Fender | cappiz: Yes |
| 18:38.12 | cappiz | [TK]D-Fender, is that configured in extensions_custom? |
| 18:38.33 | mvanbaak | ~freepbx |
| 18:38.33 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:38.38 | [TK]D-Fender | cappiz: Thats condfigured in you sip.conf entry for your device. you know "context=[my calls go here]" |
| 18:38.46 | cappiz | kk |
| 18:39.34 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 18:41.21 | mvanbaak | I'll stay away from the ~ key ;) |
| 18:43.29 | *** join/#asterisk dty (n=dertybiz@195.225.54.221) |
| 18:43.59 | cappiz | [TK]D-Fender, i think i was thinking of dial plan per extension |
| 18:44.24 | cappiz | i look slike it yeah |
| 18:44.44 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 18:45.03 | [TK]D-Fender | cappiz: a SIP device is NOT an EXTENSIONS. An extension is a number you can dial in your dialplan. And every SIP device can have its calls going to a context that is as unique from any other you have as you want it to be |
| 18:45.17 | cappiz | Roger that |
| 18:45.37 | [TK]D-Fender | cappiz: So when one device dials "123" its nothing like what happens when another device does the same. |
| 18:46.26 | [TK]D-Fender | cappiz: Of course FreePBX owns your sorry as and you have forked ofver control of your setup to it, so don't expect much. |
| 18:46.49 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 18:46.50 | cappiz | lol:> |
| 18:47.15 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 18:47.56 | [TK]D-Fender | </bile> |
| 18:49.16 | *** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com) |
| 18:49.51 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 18:51.54 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:01.29 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:04.55 | *** join/#asterisk Cresl1n (n=matt@nat/digium/x-95a3aa22a3f0042e) |
| 19:04.55 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:10.09 | _x86_ | jameswf: how long have to worked at digium? |
| 19:10.56 | jameswf | umm i dont |
| 19:11.13 | jameswf | I work at rhino |
| 19:12.15 | _x86_ | thats what i meant to ask |
| 19:12.19 | _x86_ | :P |
| 19:12.34 | _x86_ | for how long? |
| 19:12.48 | jameswf | about a year |
| 19:12.52 | _x86_ | cool |
| 19:12.53 | _x86_ | you like it? |
| 19:14.23 | jameswf | best job in the world |
| 19:15.12 | rob0 | rhino? |
| 19:16.06 | rob0 | Oh, you're not in Huntsville I guess. |
| 19:16.22 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 19:17.51 | russellb | no way, _I_ have the best job in the world. |
| 19:17.56 | russellb | :-p |
| 19:19.18 | twisted | HAHAHAHA |
| 19:19.45 | russellb | twisted: http://svn.digium.com/svn/asterisk/team/russell/smdi-1.4 |
| 19:19.46 | ai-a | nope, i have the best nob in the world. |
| 19:19.51 | russellb | *CLI> core show function SMDI_MSG_RETRIEVE |
| 19:19.53 | russellb | *CLI> core show function SMDI_MSG |
| 19:19.58 | ai-a | oh, job. sorry, thought you were comparing nobs. |
| 19:20.00 | twisted | russellb: sweet |
| 19:20.13 | russellb | twisted: yeah, just an fyi ... |
| 19:20.33 | twisted | russellb: since you were looking through the code |
| 19:20.34 | twisted | .... |
| 19:20.46 | twisted | do you know if multiple smdi interfaces will work? (multiple serial ports) |
| 19:20.51 | russellb | it should work |
| 19:20.57 | twisted | that's what I was thinking when I read it |
| 19:21.01 | twisted | i just wanted a second opinion :) |
| 19:21.05 | russellb | yeah, it creates a thread for each one to monitor it |
| 19:21.13 | russellb | 5 should work just as well as 1 |
| 19:21.23 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 19:21.33 | DeeJayTwo | I have some problems with moh on FreeBSD... |
| 19:21.34 | twisted | what we need, though, is a way to know which SMDI port sent the MD data so we can reply with the MWI on the correct one |
| 19:21.38 | DeeJayTwo | it sounds bad... |
| 19:21.44 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 19:21.54 | DeeJayTwo | is there any fix for this? |
| 19:22.03 | twisted | DeeJayTwo: use linux |
| 19:22.20 | DeeJayTwo | nothing else? |
| 19:22.21 | DeeJayTwo | ;) |
| 19:22.44 | twisted | DeeJayTwo: try downloading asterisk-addons, and see if format_mp3 works better than mpg123 |
| 19:22.53 | DeeJayTwo | I'm using native |
| 19:22.54 | DeeJayTwo | with ulaw files |
| 19:22.55 | twisted | oh |
| 19:23.17 | twisted | in that case, heh.... check your hw timer and make sure you're doing 1000 hz |
| 19:23.28 | DeeJayTwo | it is |
| 19:23.40 | *** join/#asterisk mog (n=mog@nat/digium/x-eae538bd6eefc25f) |
| 19:23.40 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:23.41 | twisted | then i dunno, considering it's bsd |
| 19:24.01 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:24.17 | DeeJayTwo | how do I ensure it's using timing? |
| 19:26.03 | ice_croft | DeeJayTwo> see kernel config |
| 19:26.07 | russellb | ZX81: can you please turn that off so it doesn't send a message to every channel you are in? |
| 19:26.26 | putnopvut | He can't since he's away fishing :) |
| 19:26.26 | ai-a | he might not get that message for a while ;) |
| 19:26.40 | twisted | i still don't see the big issue with away notifications, as long as they don't repeat |
| 19:26.50 | DeeJayTwo | ice_croft No I mean... timing source |
| 19:26.57 | DeeJayTwo | I loaded zaptel and ztdummy |
| 19:27.01 | russellb | it's just annoying, it makes like 5 channels show new messages, when it's just that |
| 19:27.08 | DeeJayTwo | how do I know the asterisk process is using it? |
| 19:27.38 | DeeJayTwo | is there a command that tells me "Yeah right, I'm using /dev/zap/timerX) |
| 19:27.51 | twisted | do you have something like lsof on bsd? |
| 19:28.21 | ai-a | DeeJayTwo: tried zap show status ? |
| 19:28.30 | ai-a | never used ztdummy :) |
| 19:28.47 | DeeJayTwo | ZTDUMMY/1 1 UNCONFIGUR 0 0 0 |
| 19:28.48 | ai-a | might come up in the description field.. |
| 19:28.50 | DeeJayTwo | is it good? |
| 19:28.50 | ice_croft | oh:) |
| 19:28.53 | ai-a | bingo :) |
| 19:28.58 | ice_croft | heck no :) |
| 19:29.03 | ai-a | unconfigured isnt good. |
| 19:29.12 | DeeJayTwo | how is it configured? |
| 19:29.33 | ice_croft | do u have any zaptel devices? |
| 19:29.37 | DeeJayTwo | no |
| 19:30.29 | ai-a | DeeJayTwo: read... http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 19:30.43 | ai-a | and read "The Book" |
| 19:31.14 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 19:32.14 | tzafrir | DeeJayTwo, zttest tests zaptel timing |
| 19:33.19 | *** join/#asterisk moprilo (n=jjohn@201.198.78.23) |
| 19:35.01 | DeeJayTwo | this is working |
| 19:35.04 | DeeJayTwo | it looks like asterisk doesn't use it |
| 19:35.41 | ai-a | used for voicemail and such. |
| 19:35.49 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.16 (2007/12/18), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.7.1 (2007/12/13), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 19:36.34 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 19:37.12 | [TK]D-Fender | russellb: ! |
| 19:37.13 | [TK]D-Fender | ~whee |
| 19:37.14 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 19:37.22 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 19:37.25 | russellb | :) |
| 19:38.51 | ice_croft | is it some weeman thing? :) |
| 19:39.22 | Mercestes | ... |
| 19:39.51 | Mercestes | I'm back less than 30 seconds, and this is what your talking about? |
| 19:39.51 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 19:40.32 | *** part/#asterisk myiagy (n=Jose@200.215.59.133) |
| 19:41.08 | rob0 | argh, a new release, just after I installed :( |
| 19:41.18 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:43.39 | *** join/#asterisk fatpelt (n=pfelt@mail.feltonline.com) |
| 19:45.26 | fatpelt | good afternoon all. i'm trying to help a customer figure out a problem with their veracity account. when we make a call from a sip phone outbound through veracity, the land (cell) phone rings, but when we answer the call doesn't get connected. i don't even know how to phrase this to search google. |
| 19:46.03 | stimpie | fatpelt, start with tracing/sniffing the sip session |
| 19:46.57 | *** join/#asterisk michael-i (n=michael-@208.53.198.95) |
| 19:47.17 | fatpelt | ok. i 've done that. let me open it up |
| 19:48.09 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:48.09 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 19:49.17 | blitzrage | when a call comes from a Zap channel (lets say analog phone), what would an example of the value of ${CHANNEL} look like? (I don't have any hardware setup here or I'd just look myself...) |
| 19:49.23 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 19:51.52 | russellb | Zap/1-1 |
| 19:53.30 | blitzrage | ok cool -- wasn't sure if there was a random string on the end or not |
| 19:53.33 | blitzrage | (like SIP) |
| 19:53.38 | fatpelt | stimpie: i see a 401 unauthorized that i'm tracking down. i assume that is part of my problem |
| 19:53.54 | russellb | the "-1" is the equivalent of the random string in SIP |
| 19:53.56 | blitzrage | russellb: Zap/span-channel is the format? |
| 19:54.06 | russellb | Zap/channel-...um...sub-channel |
| 19:54.17 | blitzrage | ah gotcha |
| 19:54.31 | Qwell | Zap/[span-]channel |
| 19:54.32 | Qwell | :D |
| 19:54.39 | Qwell | or something |
| 19:54.44 | blitzrage | haha... or something :) |
| 19:54.45 | russellb | the sub-channel is used in cases where you have multiple calls on the same zap channel |
| 19:54.52 | russellb | thing, call waiting, three way calling, that kind of stuff |
| 19:55.01 | outtolunc | i like to say tech/dev-occurance |
| 19:55.02 | blitzrage | oh ok -- then I don't care about that -- I just want to know what port I'm talking with |
| 19:55.10 | blitzrage | outtolunc: that makes sense |
| 19:55.20 | blitzrage | thanks all! |
| 19:55.55 | *** join/#asterisk mikecx (n=mikecx@pool-68-238-4-252.chi.dsl-w.verizon.net) |
| 19:56.37 | mikecx | i need help with an error |
| 19:57.44 | Mercestes | Is there more information forthcoming? |
| 19:57.47 | blitzrage | mikecx: just go ahead and ask :) |
| 19:57.49 | mikecx | yeah |
| 19:57.53 | mikecx | trying to get it on pastebin |
| 19:57.58 | Mercestes | Oh good. lol |
| 19:57.59 | mikecx | but pastebin.com seems broke |
| 19:58.04 | blitzrage | pastebin.ca |
| 19:58.08 | blitzrage | (Canadians are better) |
| 19:58.21 | Mercestes | s/better/hairier/ |
| 19:58.25 | mikecx | http://pastebin.ca/822518 |
| 19:58.47 | stimpie | fatpelt, sounds like you have found it |
| 19:58.58 | fatpelt | ok. so in my trace i see that the asterisk does an INVITE to the veracity system and it sends back a 200 OK which never seems to get acknoledged |
| 19:59.01 | Mercestes | mikecx: New to asterisk? |
| 19:59.07 | *** part/#asterisk moprilo (n=jjohn@201.198.78.23) |
| 19:59.07 | fatpelt | stimpie: i found further down that the phone registered later |
| 19:59.18 | mikecx | Mercestes: nah, just trying to get SLA working and it's being a pain |
| 19:59.24 | fatpelt | after veracity sends 8 200 OKs it terminates the call |
| 19:59.36 | stimpie | first 401 is then likely the trigger for password authentication |
| 20:00.07 | fatpelt | stimpie: yeah, it gets the 401 then registeres sending in an md5 hash |
| 20:00.37 | Mercestes | mikecx: Looks like you have a context defined in zapata.conf which doesn't exist in extensions.conf, or...if it does exist, then there isn't an exten => entry for the ANI being sent to it or for "s", nor are there appropriate entries under [default] for the ANI or s, so the call fails. |
| 20:00.40 | [TK]D-Fender | mikecx: Your zaptel channel was told to go to [line4] s,1 and that doesn't exist. Change your zapata.conf to match where you want it to go or go make the context its looking for. |
| 20:01.10 | Mercestes | mikecx: So if you make a context that matches the context= in zapata.conf, and put an exten => s,1,Playback(tt-monkeys), it should atleast provide you with more interesting results. |
| 20:01.23 | mikecx | [TK]D-Fender: probably have my contexts mixed up then |
| 20:01.37 | Mercestes | once you've had your fun with that, you may replace Playback(tt-monkeys) with whatever helpful code you would like to put there. |
| 20:01.54 | Qwell | you can't have tt-monkeys without the intro |
| 20:02.01 | Qwell | if you only need one sound, you must use tt-weasels |
| 20:02.15 | Mercestes | Ah, even better. |
| 20:02.19 | *** part/#asterisk markgreene (n=Mark_Gre@rrcs-24-227-107-130.se.biz.rr.com) |
| 20:02.33 | mikecx | at least now it looks like it's going into the right context |
| 20:04.11 | mikecx | woo, hits the right context now and can get a call from outside |
| 20:04.18 | mikecx | now to make the sip phones ring |
| 20:04.27 | fatpelt | i'm looking through docs on the sip protocol and it seems that after my asterisk gets the 200 ok from veracity i should ack it, or am i reading this wrong?) |
| 20:04.38 | *** join/#asterisk fujin_ (n=aj@unaffiliated/fujin) |
| 20:04.56 | Qwell | fatpelt: I believe that is correct |
| 20:05.29 | Mercestes | mikecx: Keep at it. :) |
| 20:07.17 | *** join/#asterisk anthonyww (n=anthony@71.216.32.58) |
| 20:08.27 | fatpelt | how does asterisk associate the incoming 200 OK with the outgoing sip request? (i assume it is the Call-ID field) |
| 20:09.06 | anthonyww | Newbie asterisk question: What's the easiest way to remove the default voicemail message. When forwarded to voicemail, I just want it to beep and start recording. |
| 20:09.34 | *** join/#asterisk ZX81_ (n=ZX81@202.49.106.158) |
| 20:10.01 | Mercestes | anthonyww, not so newbie. Instead of calling VM you could call Record(), or you could cp /var/lib/asterisk/sounds/silence/1.gsm to /var/spool/asterisk/voicemail/context/mailbox/unavail.gsm |
| 20:10.35 | Mercestes | If a greeting doesn't exist asterisk will always play the default message. If one exists then it will play *that* message. So you just make one exist that doesn't actually make any noise. |
| 20:10.58 | anthonyww | OK, well that sounds simple |
| 20:10.59 | Mercestes | I think it will still say "please leave a message after the tone." But you can mv that file and replace it with another silence file too. |
| 20:11.34 | anthonyww | great, I'll go try that |
| 20:11.36 | anthonyww | thanks much |
| 20:11.43 | dswillia | ok i guess i am a newb but when I get the error Extension '' in context 'from-zaptel' from '3161' does not exist (3161 is my calling ID) What exactly is it looking for in the extentions.conf file. the extention '' doesnt mean much to me |
| 20:11.49 | Mercestes | np |
| 20:12.34 | Mercestes | dswillia, what is 3161? |
| 20:12.49 | [TK]D-Fender | dswillia: I already told you to make sure to set "immediate=no" |
| 20:12.55 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 20:12.59 | [TK]D-Fender | dswillia: And restart *. have you don't this? |
| 20:13.03 | [TK]D-Fender | done* |
| 20:13.12 | dswillia | yes i have |
| 20:13.28 | dswillia | even rebooted the server to make sure i was getting it all |
| 20:13.30 | [TK]D-Fender | dswillia: in zapata.conf before your "channel" line, right? |
| 20:13.42 | [TK]D-Fender | dswillia: can you pastebin your zapata.conf please. |
| 20:13.42 | mintee | hello all, very very newbie here. Just got a sangoma card and a PRI installed and ready. Green light on the card, but I don't know what to do from here. |
| 20:13.43 | [TK]D-Fender | ~pb |
| 20:13.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:13.51 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 20:13.53 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 20:14.15 | Mercestes | mintee, Step 2. Read the book. |
| 20:14.17 | [TK]D-Fender | mintee: ... |
| 20:14.18 | [TK]D-Fender | ~book |
| 20:14.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 20:14.26 | [TK]D-Fender | Mercestes: This ones yours :) |
| 20:14.32 | Mercestes | mintee, actually, that would be step 1. Step 11 is "install PRI." |
| 20:14.36 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:14.41 | Mercestes | [TK]D-Fender, lol, Gee thanks. :P |
| 20:14.42 | mintee | well, i'm kinda stuck at |
| 20:14.43 | mintee | <PROTECTED> |
| 20:14.43 | mintee | <PROTECTED> |
| 20:15.03 | Mercestes | mintee, which is very well described in the book. |
| 20:15.07 | [TK]D-Fender | mintee: if you're pluggin in a PRI from the telco its PRICPE |
| 20:15.11 | ZX81_ | CPE == Customer Premises Equipment |
| 20:15.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:15.27 | mintee | Good, that's what I thought... ok. |
| 20:15.39 | Mercestes | Sangoma also has *very* good asterisk support. |
| 20:15.44 | dswillia | TK, does it need to be before channel? |
| 20:15.53 | Mercestes | atleast to the point of getting your card online and working. |
| 20:16.01 | russellb | which has nothing to do with asterisk ... |
| 20:16.05 | russellb | but hey |
| 20:16.07 | Mercestes | They won't for example, program your extensions.conf |
| 20:16.17 | Mercestes | russellb, True, figured i should elaborate. ;) |
| 20:16.23 | mintee | yeah, I wouldn't expect them too. |
| 20:16.35 | russellb | just wanted to clarify that they don't support asterisk in any way |
| 20:16.40 | mintee | I'm just working on a inbound voicemail system, so I've no need for any phones |
| 20:16.45 | Mercestes | I'm pretty sure that prompt is from the sangoma script, which generates a zaptel.conf |
| 20:16.45 | russellb | not with helping you set it up, nor with any contributions |
| 20:16.54 | muiro | what could cause a hangup not to be detected only within certain contexts? |
| 20:16.56 | russellb | :-X |
| 20:16.59 | mintee | yeah, setup-sangoma.... done |
| 20:17.02 | [TK]D-Fender | dswillia: YES |
| 20:17.14 | [TK]D-Fender | dswillia: Otherwise ti won't apply to them |
| 20:17.21 | mintee | zap show channels lists 48 channels |
| 20:18.52 | mintee | my pri span 1 is showing up and active... so, I'm getting somewhere |
| 20:20.08 | dswillia | TK > http://pastebin.com/d62ec86c9 |
| 20:20.13 | dswillia | is my pastebin |
| 20:20.32 | russellb | can someone try http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.4.16 ?? |
| 20:20.45 | russellb | i hit a local mirror so i need someone outside to test ... |
| 20:21.24 | Mercestes | I got a dir with pub/ |
| 20:21.59 | Mercestes | Last modified on August 24, 2007 btw. |
| 20:22.36 | tzafrir | russellb, don't you have a shell account elsewhere? |
| 20:24.13 | [TK]D-Fender | dswillia: channel => 1-23 should be at the BOTTOM |
| 20:24.32 | [TK]D-Fender | dswillia: Whats on the other end of this again? |
| 20:24.45 | russellb | tzafrir: yeah, heh ... |
| 20:25.18 | Mercestes | I'd give you access to my router but you might setup my iptables to forward all ips to goatse. |
| 20:25.21 | dswillia | an Avaya PBX, I think that is my issue |
| 20:25.46 | mvanbaak | russellb: I get redirected to the root of downloads.digium.com |
| 20:25.57 | russellb | thanks |
| 20:26.13 | mvanbaak | hhmm, what does sangoma support then ? |
| 20:26.23 | Mercestes | mvanbaak, Their cards. |
| 20:26.29 | mvanbaak | they dont support asterisk, they dont support openbsd |
| 20:26.49 | Mercestes | I don't think openbsd supports openbsd. |
| 20:27.01 | mvanbaak | Mercestes: try again in #openbsd ;) |
| 20:27.34 | Mercestes | mvanbaak, Yea, that's ok. I don't need my ego bruised anymore today. |
| 20:28.10 | mvanbaak | I have tried to get an ADSL nic from sangoma working |
| 20:28.19 | Mercestes | I think security in OpenBSD is akin to lining your hallway with razorblades. The level of security is inversely proportional to your intruder's pain threshold. |
| 20:28.22 | mvanbaak | the sales rep told me they support obsd |
| 20:28.27 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 20:28.32 | muiro | question about the generic speech engine (using lumenvox): voip wiki says that when performing SpeechCreate() that: If an error occurs are you are not able to create an object, the variable ERROR will be |
| 20:28.37 | muiro | set to 1 |
| 20:28.37 | mvanbaak | in the end they gave me some commands for freebsd and left me with it |
| 20:28.44 | muiro | but this variable doesn't seem to exist |
| 20:29.05 | Mercestes | mvanbaak, lol. Nice. an ADSL nic? |
| 20:29.16 | mvanbaak | yup |
| 20:29.22 | [TK]D-Fender | dswillia: ok, it looks like the Avaya simply isn't sending a DID even though you're using PRI signalling. That in mind, make a new context like this [stupidavaya] exten => _!,1,Goto(bettercontext,s,1) |
| 20:29.27 | CrazyTux | Hey guys, say I want to initiate something such as "call recording" while, IN Dialog (a call), is this possible? |
| 20:29.41 | [TK]D-Fender | dswillia: and set "context=stupidavaya" in zapata. |
| 20:29.50 | Mercestes | CrazyTux, features.conf |
| 20:29.55 | [TK]D-Fender | dswillia: Thats how to account for a null exten |
| 20:30.03 | *** join/#asterisk Inverted (n=codya@staff2.aus.datafoundry.com) |
| 20:30.22 | mvanbaak | I really hate the shitty buffers that they put in the dsl modems on the market. makes it hard to do queueing and prioritizing |
| 20:30.30 | Mercestes | mvanbaak, I have precious little experience in Obsd. |
| 20:30.33 | mvanbaak | so I wanted a nic in my obsd firewall |
| 20:30.42 | [TK]D-Fender | mvanbaak: Sangoma S518 :) |
| 20:30.48 | mvanbaak | [TK]D-Fender: yup |
| 20:31.07 | Mercestes | mvanbaak, NIce. That would be a sweet setup. |
| 20:31.46 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-58-204.lns10.syd7.internode.on.net) |
| 20:35.21 | nhuisman_work | are there any ways to build provisioning into asterisk? |
| 20:37.40 | [TK]D-Fender | nhuisman_work: Provisioning of WHAT? |
| 20:37.53 | nhuisman_work | well i'm talking about phones |
| 20:38.01 | nhuisman_work | like uploading the sip image, and phone settings |
| 20:38.07 | [TK]D-Fender | nhuisman_work: And what does * have to do with provisioning? |
| 20:38.31 | [TK]D-Fender | nhuisman_work: You server never uploads anything for provisioning. Phones download them. |
| 20:38.50 | [TK]D-Fender | nhuisman_work: And you can go right ahead and make a scrip that'll parse your sip.conf, etc and build them any which way you want. |
| 20:38.55 | [TK]D-Fender | nhuisman_work: But this is NOT *'s job |
| 20:39.08 | [TK]D-Fender | nhuisman_work: Thats the sort of stuff you see included in GUI setups. |
| 20:39.49 | nhuisman_work | wonder if I can find a gui that will work with asterisk business edition to make that easier for my junior sa's |
| 20:40.21 | [TK]D-Fender | nhuisman_work: Thats like asking your mechanic to cook you dinner. |
| 20:40.33 | [TK]D-Fender | nhuisman_work: You should jsut go buy a complete and supported setup. |
| 20:40.55 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:41.31 | *** join/#asterisk Dovid (n=Dovid@85.109.2.68) |
| 20:41.32 | nhuisman_work | I can config the server and setup myself I just don't want to have to edit configs day to day. |
| 20:41.38 | Dovid | can anyone help me with this error |
| 20:41.39 | Dovid | Unable to re-open DSP device /dev/dsp: No such file or directory |
| 20:42.26 | mvanbaak | hhmm |
| 20:42.32 | mvanbaak | xs4all is acting up again |
| 20:42.56 | mvanbaak | I think they have a loadbalanced platform where 1 sip box is sending weird traffic to asterisk |
| 20:43.04 | mvanbaak | [2007-12-18 21:42:30] WARNING[5034]: chan_sip.c:7537 determine_firstline_parts: Bad request protocol Packet |
| 20:43.16 | mvanbaak | it has been fine for days |
| 20:43.28 | mvanbaak | today my box reregistered because of a modem reboot |
| 20:43.32 | mvanbaak | and here we are again |
| 20:43.36 | mvanbaak | *sigh* |
| 20:51.15 | fatpelt | si there any good way to know why asterisk doesn't associate the incoming 200 OK from a SIP call to the INVITE that started it? |
| 20:52.05 | *** join/#asterisk jwh (i=jwh@scarlett.lon.rewt.org.uk) |
| 20:54.43 | jameswf | I just read fonality rewrote 10,000 lines of asterisk code to make it work right |
| 20:55.58 | mvanbaak | gheh |
| 20:56.02 | mvanbaak | what parts ? |
| 20:56.11 | dswillia | how would i show the status of a T1's individual channels (this is configured as a chennelized T1 not pri) |
| 20:56.24 | Dovid | can anyone help me with this error |
| 20:56.25 | Dovid | Unable to re-open DSP device /dev/dsp: No such file or directory |
| 20:59.07 | jameswf | http://pastebin.ca/822612 << chris from fonality |
| 20:59.27 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 21:00.08 | fatpelt | is there any way to have asterisk show all the packets it receives? (im' running a tcpdump and see that packets are getting to the box, but i'd like to see if asterisk is actually getting them) |
| 21:02.04 | jameswf | fatpelt: whats the ultimate issue |
| 21:03.49 | jameswf | ~book |
| 21:03.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 21:09.25 | nhuisman_work | trying to get that damn book to print |
| 21:09.27 | nhuisman_work | dunno why it isn't |
| 21:09.29 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 21:09.55 | [TK]D-Fender | fatpelt: "sip debug" |
| 21:10.23 | rantsh | hi guys |
| 21:11.39 | rantsh | is there anything else I need to do for asterisk to use speex BESIDES install libogg, install speex, recompile asterisk???? |
| 21:12.35 | nhuisman_work | adobe cs on linux via wine? |
| 21:12.54 | jameswf | no that failed misserably vmware |
| 21:13.42 | fatpelt | jameswf: i've got a packet sniff going watching my softphone dialing out to my cell via veracity. i see the invite go out, and i eventually see the 200 OK come back. however asterisk never seems to think the call goes trhough |
| 21:13.50 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 21:14.36 | [TK]D-Fender | fatpelt: PASTEBIN is your friend |
| 21:14.37 | [TK]D-Fender | ~pb |
| 21:14.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:14.53 | fatpelt | [TK]D-Fender: what would you like to see? |
| 21:15.01 | *** join/#asterisk digime (n=digime@141stb17.codetel.net.do) |
| 21:15.11 | [TK]D-Fender | fatpelt: How about the entire call attempt from beginning to end with sip debug & verbose 10 |
| 21:15.31 | etfonhomey | [TK]D-Fender, how many times a day do you think you type ~pb? |
| 21:15.47 | [TK]D-Fender | etfonhomey: "Carpal Tunnel Syndrome" |
| 21:15.59 | etfonhomey | :) |
| 21:16.39 | jameswf | irc needs mouse gestures |
| 21:18.00 | mvanbaak | jameswf: any idea wether fonality will send patches ? |
| 21:20.47 | jameswf | patches? |
| 21:22.06 | mvanbaak | yeah |
| 21:22.16 | mvanbaak | <PROTECTED> |
| 21:23.01 | jameswf | [TK]D-Fender: Qwell: how much has fonality contributed back.... |
| 21:23.05 | mvanbaak | any idea wether the asterisk community will get those 'fix reliably produced deadlock' patches ? |
| 21:23.22 | russellb | jameswf: 0, of course |
| 21:23.29 | JT | i think fonality are more leechers than sharers |
| 21:23.40 | JT | their stupid trixbox pro grabage is proof of that |
| 21:23.45 | jameswf | mvanbaak: answer your question ? |
| 21:23.54 | JT | turning open source into proprietary |
| 21:24.20 | [TK]D-Fender | russellb: Do they have a commercial license for Trixbox Pro? |
| 21:24.28 | JT | s/grabage/garbage/ |
| 21:24.30 | [TK]D-Fender | russellb: (from Digium that is) |
| 21:24.31 | mvanbaak | jameswf: sadly yes |
| 21:24.51 | mvanbaak | digium is selling trixbox pro ? |
| 21:25.00 | russellb | mvanbaak: no ... |
| 21:25.04 | mvanbaak | thank god |
| 21:25.11 | mvanbaak | for a moment my world collapsed |
| 21:26.42 | rantsh | ~speex |
| 21:26.48 | JT | imho fonality is not a very ethical company |
| 21:27.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585332.dsl.bell.ca) |
| 21:27.43 | tzanger | http://bp1.blogger.com/_VsMY1jwbntQ/R2crj7lRG2I/AAAAAAAAAP8/uN9MMh8S7GE/s1600-h/1189626299294.jpg |
| 21:27.46 | tzanger | hahhaaha |
| 21:28.10 | mvanbaak | meh |
| 21:28.15 | mvanbaak | old but good |
| 21:28.27 | *** join/#asterisk FreePhone (n=phone@69-50-167-50.esthost.com) |
| 21:28.45 | *** join/#asterisk oppassum (n=op@adsl-76-249-13-6.dsl.klmzmi.sbcglobal.net) |
| 21:28.46 | toomba | anyone familiar with T1 PRIs and Cisco AS5300 and sending calls from SIP to PSTN through it? |
| 21:28.59 | shido6 | yes |
| 21:29.09 | fujin_ | we've got a 5400, and two e1 PRI's |
| 21:29.10 | fujin_ | but yes |
| 21:29.11 | shido6 | whats up? |
| 21:29.13 | oppassum | anyone familiar with trixbox and polycom phones? |
| 21:29.19 | fujin_ | oppassum: #trixbox |
| 21:29.19 | shido6 | yes |
| 21:29.22 | [hC] | oppassum: #trixbox |
| 21:29.27 | shido6 | whats up, toomba? |
| 21:29.28 | fujin_ | ;> |
| 21:29.35 | FreePhone | Hello, i have new service - web based voip calls (flash) anybody want test it ? MSG me |
| 21:29.53 | FreePhone | it based on asterisk and red5 |
| 21:30.13 | rantsh | I've recompiled asterisk about 3 times now, I can get it to show a translation from/to speex |
| 21:30.46 | [TK]D-Fender | Alrighty, I'm off. Later all |
| 21:32.30 | fatpelt | bummer. tk just left |
| 21:32.46 | fatpelt | jameswf: http://slexy.org/view/s2LTY1kRJT |
| 21:33.14 | fatpelt | the ultimate problem is that wireshark shows the 200 OK coming in but asterisk never ties the incoming sip request with the SIP call to veracity |
| 21:34.23 | *** part/#asterisk FreePhone (n=phone@69-50-167-50.esthost.com) |
| 21:37.09 | digime | looking for someone to update my asterisk IVR, I have the sound files already cut up, anyone game? would be willing to trade for some prompts from allison (theivrvoice.com) |
| 21:37.52 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 21:41.31 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:41.38 | JT | so the redfone fonebridges now can come with hardware echo cancellation |
| 21:41.56 | JT | all they need is SIP or H.323 then instead of TDMoE ;) |
| 21:42.36 | DeeJayTwo | Really it's hard getting ztdummy working on FreeBSD with asterisk... |
| 21:42.39 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:42.45 | DeeJayTwo | the module is loaded... |
| 21:43.11 | DeeJayTwo | asterisk.conf has internal_timing = yes |
| 21:43.29 | DeeJayTwo | zap show status shows: |
| 21:43.34 | DeeJayTwo | ZTDUMMY/1 1 UNCONFIGUR 0 0 0 |
| 21:43.41 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 21:46.28 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 21:47.37 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 21:47.58 | iamthelostboy | hi.. i seem to have a call stuck between a sip user and a zap line.. |
| 21:48.24 | iamthelostboy | the sip user can still make calls, its just really choppy |
| 21:48.48 | iamthelostboy | what would be the best way to kill the bridged call, short of resetting the machine? |
| 21:48.50 | hmmhesays | how's your cpu usage right now |
| 21:49.06 | mvanbaak | iamthelostboy: soft hangup <channelid> |
| 21:49.11 | rantsh | anyone knows how to install speex to asterisk? |
| 21:49.14 | mvanbaak | use tab completion to get the correct channelid |
| 21:49.30 | rantsh | I'm hitting myself against a wall now |
| 21:49.34 | mvanbaak | rantsh: I never tried it, so cant help you there |
| 21:49.39 | *** join/#asterisk w3pog (n=pgrace@aeneas.fierymoon.com) |
| 21:50.19 | iamthelostboy | mvanbaak: awesome.. thanks :) |
| 21:51.09 | mvanbaak | iamthelostboy: :) |
| 21:51.41 | iamthelostboy | i take it that is what would have been making the calls choppy ? |
| 21:51.54 | mvanbaak | could be |
| 21:52.38 | iamthelostboy | is there anyway to have the system time the call out? though i suppose it would have done that if it thought the call was over ... |
| 21:52.49 | Yourname` | How can I send a call using originate from the CLI via a new peer? |
| 21:53.36 | *** part/#asterisk atlantia (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 21:57.14 | hmmhesays | search the wiki for killing zombies |
| 21:58.53 | etfonhomey | Yourname`, you ever get that tftp stuff worked out? |
| 21:59.31 | Yourname` | etfonhomey: Nope, moved everyone to eyeBeam for now. |
| 22:00.02 | etfonhomey | Yourname`, does that have a good way of configuration provisioning? |
| 22:00.17 | Yourname` | hah, nope. |
| 22:00.17 | funxion | Yourname` what phones were u using? |
| 22:00.43 | jameswf | my asterisk box was rejected on e-harmony |
| 22:01.00 | funxion | lol |
| 22:01.07 | *** part/#asterisk klictel (n=klictel@atelka.info) |
| 22:01.25 | jameswf | people dont like open relationships :( |
| 22:04.14 | Yourname` | funxion: Vertial 9133is. :S |
| 22:04.24 | Mercestes | lmao |
| 22:04.59 | JT | jameswf: e-harmony? |
| 22:05.35 | Yourname` | haha jameswf, I was JUST talking to my wife about her friend who wants to have an open relationship. |
| 22:05.47 | mvanbaak | hahahaaha |
| 22:09.26 | fatpelt | ok, so in looking at the sip debug output it seems that the asterisk isn't ack'ing any incoming traffic. the debug log shows lines with "**** Received ..." but they are all from the phone (that is on the same side of the nat) |
| 22:09.40 | fatpelt | what in the crap is going wrong here? |
| 22:10.01 | funxion | Yourname` is her friend a guy? |
| 22:10.28 | mvanbaak | fatpelt: can you pastebin that ? |
| 22:10.41 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 22:11.07 | Yourname` | funxion: Nah, a married woman, heh |
| 22:11.25 | funxion | nice |
| 22:11.29 | funxion | <PROTECTED> |
| 22:11.44 | funxion | unless of course she means open with women |
| 22:12.04 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 22:12.15 | Yourname` | Yeah, that's kinda sorta what she meant. |
| 22:12.17 | Yourname` | lol |
| 22:12.38 | Putzz | hey guys is there a way to set how long a PAGE should be when paging a PA system? |
| 22:12.53 | *** join/#asterisk Qb3rt (n=eric@palpatine.privatedns.com) |
| 22:13.37 | funxion | well then shit |
| 22:13.52 | fujin_ | Putzz: doesn't page just last until you hang up? |
| 22:13.59 | funxion | is she scopin ur wife |
| 22:14.08 | twisted | she? |
| 22:14.13 | twisted | do i sense a 3 some? |
| 22:14.14 | Putzz | for some reason mine is timing out after 5 seconds I dont know if its the sound card or asterisk it self |
| 22:14.26 | funxion | twisted read up |
| 22:14.41 | twisted | meh |
| 22:15.14 | *** join/#asterisk fatpelt (n=pfelt@mail.feltonline.com) |
| 22:15.20 | fatpelt | mvanbaak: sure it'll take a sec |
| 22:15.54 | twisted | wheeeeee |
| 22:16.18 | muiro | lumenvox question: I have grammars set to preload but they never do unless I "module reload res_speech_lumenvox.so" after asterisk is already running |
| 22:16.49 | muiro | Is there anyway I can schedule asterisk to perform that command during setup? |
| 22:17.05 | muiro | or, during initialization, that is |
| 22:19.04 | muiro | any hints? scheduling asterisk to perform a command during startup? |
| 22:19.23 | muiro | or, well, nevermind, bash'll do it |
| 22:19.27 | Putzz | make a cronjob to do asterisk -rx module reload res_speech_lumenvox.so |
| 22:19.38 | fatpelt | mvanbaak: which part would you like? |
| 22:19.54 | mvanbaak | fatpelt: everything |
| 22:23.21 | fatpelt | mvanbaak: http://slexy.org/view/s20KC23mHc |
| 22:23.55 | fatpelt | via tcpdump i can see that packets are coming in from veracity, however asterisk doesn't seem to want to acknowleged that it sees them |
| 22:26.40 | mvanbaak | Dec 18 15:15:26 DEBUG[2857] chan_sip.c: Header 5: CSeq: 1 ACK (11) |
| 22:26.47 | mvanbaak | meh, it's sending ACK ok ? |
| 22:28.11 | fatpelt | mvanbaak: those are destined for the softphone on the same 10 network as the asterisk box |
| 22:29.16 | mvanbaak | you have 10.0.1 and 10.0.50 in the same network ? |
| 22:30.59 | fatpelt | mvanbaak: yes |
| 22:31.04 | mvanbaak | ok |
| 22:31.30 | mvanbaak | hhmm, I think the ammount of single malt scotch is acting up |
| 22:31.44 | mvanbaak | better ask someone else because I'm going to bed |
| 22:32.02 | mvanbaak | sorry |
| 22:32.48 | *** join/#asterisk SkramX (n=mark@HERCULES.sentiensystems.net) |
| 22:34.28 | fatpelt | anybody know why asterisk doesn't seem to want to ack any incoming packets from off network? |
| 22:34.48 | *** join/#asterisk Maliuta (n=nikolai@119.11.98.119) |
| 22:38.24 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:38.34 | *** join/#asterisk mihinomenest (i=SIXV@66.255.220.17) |
| 22:39.07 | mihinomenest | im that I've finally gotten all of the parts for my snowblower. |
| 22:40.19 | fujin_ | sweet, snowblower. |
| 22:41.17 | mihinomenest | oh, I wondered where that went. |
| 22:42.41 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 22:42.43 | fujin_ | fatpelt: have you done the relevant configs in sip.conf for these off-network devices? |
| 22:44.25 | fatpelt | fujin_: i do have a section for veracity yes. in fact when i place the call, my landline rings, however it never connects since the asterisk here doesn't ack the 200 OK |
| 22:47.14 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 22:47.45 | FuriousGeorge | hey all |
| 22:48.11 | FuriousGeorge | anyone using snom phones notice that calls between two snom 360's get "stuck" and must be soft hungup on? |
| 22:48.12 | *** join/#asterisk fittysix (n=six@047.216-123-202-0.interbaun.com) |
| 22:48.28 | FuriousGeorge | not always, just a lot of the time |
| 22:49.02 | FuriousGeorge | i am using 1.2 still... but im worried that switching to 1.4 will bring on the 'zap channels getting locked' i have on an identical system elsewhere |
| 22:49.11 | *** part/#asterisk fittysix (n=six@047.216-123-202-0.interbaun.com) |
| 22:49.16 | FuriousGeorge | identical except for that one is using 1.4 |
| 22:49.46 | FuriousGeorge | when sip channels get locked, the blf stays lit, when the zap channels get locked, we stop receiving incoming calls. not good at all. |
| 22:51.08 | FuriousGeorge | oh great, 1.4.16 is out today. last time the zap channels got locked, but i told myself i needed to upgrade from 1.4.12 (though i've stopped expecting it to help, just cuz thats the standard response to problems), but i had a feeling we were not long for another update |
| 22:56.06 | *** join/#asterisk saftsack (n=saftsack@83.218.162.174) |
| 22:56.25 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 22:56.25 | fatpelt | so um anyone else have any ideas on how to get my asterisk to correctly connect a softphone to veracity? |
| 22:57.37 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 22:57.44 | *** join/#asterisk sinanimam (n=sinan@81.213.69.63) |
| 22:59.40 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 22:59.45 | *** part/#asterisk pepo-- (n=pepOSX@190.79.246.105) |
| 23:04.39 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 23:04.51 | *** join/#asterisk Husk (n=husk@202.55.146.217) |
| 23:05.18 | Husk | Is there a way to restrict commands on the CLI? |
| 23:05.25 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585332.dsl.bell.ca) |
| 23:07.48 | fatpelt | what is the best way to debug why incoming sip calls don't work? |
| 23:16.29 | CrazyTux | Hello guys what are the available, AgentCallbackLogin() options? |
| 23:16.56 | *** part/#asterisk sinanimam (n=sinan@81.213.69.63) |
| 23:17.59 | *** join/#asterisk RoyKa (n=roy@ip-171-6-149-91.dialup.ice.no) |
| 23:19.27 | *** join/#asterisk mikecx (n=mikecx@cpe-76-181-117-188.columbus.res.rr.com) |
| 23:20.01 | mikecx | what controls what context outgoing calls go to? |
| 23:20.21 | mikecx | like when I pick up a phone and dial, how does it know what context to start in? |
| 23:20.54 | Qwell | the entry for that device in the channel tech config |
| 23:21.05 | Qwell | so zapata.conf, sip.conf, etc - context= |
| 23:21.34 | mikecx | alright, because i've got incoming SLA working well now, just can't make it get an external line yet |
| 23:22.00 | CrazyTux | ${CALLERID(num):${LEN{${CONTEXT}}} whats wrong with that? |
| 23:22.24 | Qwell | s/N{/n(/ |
| 23:22.29 | Qwell | and the matching ) |
| 23:22.58 | CrazyTux | ah ooopps. |
| 23:23.31 | Qwell | ${CALLERID(num):${LEN(${CONTEXT})} |
| 23:23.34 | Qwell | that should do it |
| 23:23.44 | Qwell | actually |
| 23:23.48 | Qwell | you need another } at the end too |
| 23:24.04 | CrazyTux | ${CALLERID(num):${LEN(${CONTEXT}) + 1} |
| 23:24.11 | Qwell | no |
| 23:24.19 | CrazyTux | I want to add 1, to the LEN() return now |
| 23:24.27 | Qwell | ${CALLERID(num):$[${LEN(${CONTEXT})} + 1]} |
| 23:24.39 | CrazyTux | oh [] for numeric operations |
| 23:24.40 | CrazyTux | alrighty. |
| 23:26.13 | *** join/#asterisk markgreene (n=Mark_Gre@rrcs-24-227-107-130.se.biz.rr.com) |
| 23:26.51 | fatpelt | grr! this is gonna kill me! i've got a box that sends a sip request out to a remote sip server (veracity) which then places the call via a landline. the landline rings and 25 seconds after we pick up, it disconnects us and the call never goes through |
| 23:28.10 | markgreene | can someone help me out. What are the first things I should check for a Digium card that is staying on red alarm no matter what I do. It's a new install. I did an lsmod and the zaptel and wct4xxp modules are loading, and when I run ztcfg I get no errors on the outout. But the spans STAY in red alarm no matter what. They don't even move to yellow for a seccond or anything |
| 23:28.38 | JT | markgreene: make sure it's connected properly and that the circuit is up |
| 23:29.20 | CrazyTux | What options are available for: AgentCallbackLogin([AgentNo|][exten]@context) |
| 23:29.20 | markgreene | JT: Yeah. Did that. If I unplug the spans and put them into a functioning box they come right up. I've also swapped the cards and verfied it's not a bad card |
| 23:31.02 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:32.56 | JT | markgreene: put zapata.conf and zaptel.conf into pastebin.com please |
| 23:33.06 | CrazyTux | How do I create variables in 1.4.* |
| 23:33.09 | CrazyTux | SetVar still? |
| 23:33.54 | JunK-Y | CrazyTux: read UPGRADE.txt file |
| 23:35.54 | markgreene | JT: http://asterisk.pastebin.com/822797 is the zaptel file, hold for the zapata |
| 23:36.34 | markgreene | [coorection] JT: http://asterisk.pastebin.ca/822797 is the zaptel file, hold for the zapata |
| 23:38.07 | markgreene | JT: Does it matter that this is a zaptel file that works with zaptel 1.2 and I'm putting it in 1.4? I know that SOUNDS stupid, but I did not turn up anything that needed to be changed |
| 23:41.21 | *** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au) |
| 23:41.27 | markgreene | brb JT - moving to diff LAN |
| 23:41.50 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 23:44.56 | Absorto | hello! I have two * boxes A and B and I want A to register as a peer to B, so that i can call extensions in A dialing 1234@B. Can this be done? |
| 23:49.50 | fujin_ | Absorto: of course |
| 23:50.13 | *** join/#asterisk markgreene (n=Mark_Gre@209.12.142.3) |
| 23:50.22 | markgreene | JT: You still in here? |
| 23:50.35 | Absorto | fujin_: how? I gather I must create a peer in sip.conf, but what should I type into the []? |
| 23:50.38 | JT | markgreene: can you use rafb.net? i can't access asterisk.pastebin.com for some reason |
| 23:51.23 | markgreene | Sorry I posted the correction - It's pastebin.ca |
| 23:51.25 | markgreene | no .com |
| 23:51.36 | *** join/#asterisk pepo-- (n=pepOSX@201.248.215.16) |
| 23:51.37 | markgreene | JT: http://asterisk.pastebin.ca/822797 |
| 23:51.44 | markgreene | Notice the .ca on that one |
| 23:51.59 | JT | yeah sorry that doesn't work for me either |
| 23:52.39 | markgreene | JT: one sec then |
| 23:54.33 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585332.dsl.bell.ca) |
| 23:55.31 | markgreene | JT: http://rafb.net/p/K5Ld0G57.html |
| 23:55.44 | markgreene | Did you see my comment about zaptel 1.2 and 1.4? |
| 23:56.10 | *** join/#asterisk Psychobilly (n=fuzz@194.219.217.138) |
| 23:56.17 | JT | yeah don't think there were any notable changes in the conf file |
| 23:56.49 | JT | markgreene: you have 2 spans, what are each for? |
| 23:57.18 | markgreene | One is a PRI and one is a Channel bank. But this conf works on 1.2 - I don't think it's a configuration issue. |
| 23:57.46 | markgreene | Also because ztconfig dosn't through anything. |
| 23:58.01 | JT | so swapping the card with a known good one made no difference? |
