00:02.21 | harryvv | Somone was telling me if you move a voip phone from one wifi router to another though a vpn that it will not drop the call is that true? |
00:02.45 | iosahdf | h3x do you have any 561 |
00:02.51 | tzanger | GO FISH! |
00:03.28 | harryvv | go fish :) |
00:03.41 | Qwell | harryvv: not likely |
00:03.41 | nDuff | harryvv, your VPN should mask the underlying network changes, generally. |
00:03.54 | harryvv | yea I had a hard time believing that. |
00:03.59 | h3x | iosahdf: what city |
00:04.10 | iosahdf | boca raton |
00:04.32 | antifuchs | oh, hi nDuff. interesting that we meet again here (: |
00:04.50 | nDuff | antifuchs, howdy. |
00:05.21 | harryvv | I dont think there is any handoff wifi voip technoligy at the moment. I wonder what cell site equipment use. |
00:05.22 | h3x | iosahdf: yep definately, multiple choices |
00:05.48 | nesys | there' s no one that could help me to understand my sip debug? I've problems with call-forward from ccme to * on sip trunk ... |
00:06.10 | h3x | iosahdf: just go on www.carrierone.net/dids and fill out the for more info form, and specify if you want to port your number or not |
00:06.11 | iosahdf | what's your url going to be again? |
00:06.12 | antifuchs | harryvv: ISTR a zyxel handheld sip phone thing that does 801.11b or g. |
00:06.14 | iosahdf | ah |
00:06.16 | iosahdf | thanks |
00:06.22 | *** join/#asterisk stustu (~stustu@fluffy.fatburen.org) |
00:06.34 | stustu | Any zaptel developers here? |
00:06.35 | antifuchs | the battery lasts for a whole 30 minutes or so |
00:07.15 | h3x | I count 5 options for underlying carriers in boca raton |
00:07.36 | Veryhot | know a good place to get some IAXy? |
00:07.44 | h3x | The cool thing is im going to allow customers to port numbers to another underlying carrier in case the voice quality sucks or anything. |
00:08.23 | h3x | we're also going to implement caller id name+number |
00:08.35 | h3x | most of the providers out there dont have that |
00:08.37 | h3x | it will be an option |
00:08.56 | h3x | anderiv, outgoing caller id name with a CNAM database record |
00:09.03 | h3x | damn nick complete. thats supposed to be "and" |
00:09.12 | nDuff | harryvv, if your VPN is able to adapt to the underlying network changes, packets traversing it don't know or care if the VPN just had to deal with a server handoff, routing differences, etc, so your connections should be unintterrupted. Now, if there's a change *after* the packets left the VPN, that's a different story; likewise if your VPN can't handle the handoff cleanly. (OpenVPN can, if properly configured). |
00:09.23 | harryvv | antifuchs thats cool |
00:09.56 | antifuchs | harryvv: yeah, but I don't think it's very useful in practical use (: |
00:10.04 | brc_ | h3x |
00:10.06 | brc_ | daaaamn |
00:10.07 | brc_ | h3x, |
00:10.08 | h3x | bee arr cee |
00:10.09 | harryvv | yea I see. there is probebly timmer options that also could be configured. |
00:10.10 | Godsey | anyone here use broadvoice? :) |
00:10.13 | brc_ | duuuude |
00:10.15 | h3x | DUDE |
00:10.27 | h3x | brc_: I'm going to have phoenix DIDs soon too btw hehehe |
00:10.35 | h3x | *turrets*fuck ELI*turrets* |
00:13.06 | *** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net) |
00:13.50 | greg_work | what am I doing wrong here? GotoIf($[${CALLERIDNAME:0:${LEN(RGPREFIX)}} != ${RGPREFIX}]?3:2) I'm trying to check if ${CALLERIDNAME} starts with ${RGPREFIX} |
00:14.05 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
00:15.18 | *** join/#asterisk feklee (feklee@genba.ffii.org) |
00:15.55 | feklee | I just transfered a setup from one machine to another. However, on the new machine my Asterisk configuration doesn't work anymore: When I call, I always get the busy signal. |
00:16.00 | feklee | How do I find out more? |
00:16.11 | patdk | logs |
00:16.19 | *** join/#asterisk djvoodz (~oli@host217-44-21-66.range217-44.btcentralplus.com) |
00:16.24 | feklee | Can't find anything interesting in them. |
00:16.40 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
00:16.49 | djvoodz | hi guys |
00:16.54 | djvoodz | got a few problems with an x100p card |
00:16.57 | feklee | It says something about a broken pipe, but that also happened with the original setup, IIRC. |
00:16.59 | djvoodz | anyone able to help me? |
00:17.17 | patdk | djvoodz, they don't make x100p anymore |
00:17.27 | feklee | The thing is, I want the incoming call to be streamed using IceS. |
00:17.51 | djvoodz | i believe its actually an x100p clone |
00:18.03 | djvoodz | i cant seem to get asterisk to pick up the line |
00:18.03 | patdk | good luck |
00:18.09 | djvoodz | or even notice there is any form of call coming in |
00:18.27 | *** join/#asterisk Dr-Linux (~sshah@202.163.69.3) |
00:18.30 | *** join/#asterisk Zipper_32 (~none@d209-121-36-44.bchsia.telus.net) |
00:18.38 | Dr-Linux | anybody can tell me please |
00:18.50 | Dr-Linux | about SIPPS softphone ? |
00:19.07 | Dr-Linux | i m using SIPPS softphone |
00:19.29 | Dr-Linux | how i can check its registerd or not ? |
00:19.35 | Veryhot | djvoodz: who the provider? |
00:20.39 | djvoodz | we dont have a voip provider yet - it is plugged into an analogue pstn |
00:21.09 | Veryhot | dj: which X100p clone? |
00:21.27 | djvoodz | ambient md3200 |
00:21.34 | Zipper_32 | I have a relatively general question; I've been put in the position of implimenting a PBX in a new warehouse/12 person office, the cement is being poured right now and I have until mid may to setup a system. I have just installed asterisk on a Redhat8 system and I am hoping somebody could explain what i need to emulate this office environment that I am going to setup. |
00:22.02 | Zipper_32 | ... Or point me in a useful direction. |
00:22.12 | Veryhot | zipper: check out Asterisk@home |
00:22.38 | Dr-Linux | anybody know about SIPPS softclient ? |
00:23.26 | bjohnson | Dr-Linux: sip show registry |
00:23.29 | bjohnson | or sip show peers |
00:23.34 | bjohnson | or sip show users |
00:23.40 | tzanger | I'm creating a patch for asterisk |
00:23.43 | tzanger | IAX2 GIMME BEER |
00:24.15 | djvoodz | when plugging the phone line into the x100p there is a fuzzy echo |
00:24.30 | bjohnson | Zipper_32: make a lan and set up some softphones on them to start configuring your system. sound quality will be better with hardware phone but will get you started |
00:24.43 | tzanger | djvoodz: shave it? |
00:25.27 | djvoodz | haha - cud the card be faulty or is this normal behaviour? |
00:25.32 | bjohnson | Zipper_32: decide what kind of phones you want .. for 12 people in a warehouse you might want some fxs+regular cordless phones, some voip phones, and maybe even a wifi sip phone (but they are not highly regarded) |
00:26.39 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
00:27.13 | bjohnson | Zipper_32: for 12 people you probably don't want enough lines for PRI or enough extensions for channel bank .. so you will likely want small numbers of fxo and fxs devices or pci cards (or a mix) |
00:27.54 | bjohnson | Dr-Linux: I don't have time to troubleshoot your problems .. I'm just answering your question |
00:28.12 | tzanger | bjohnson: actually 8-10 lines is where I start suggesting a channel bank or PRI |
00:28.41 | DEEZED | Is it ok to have the extensions.conf include another config which is full of includes for each one of my customers that will have there own config files. Such as asterisk.conf -> clients.conf -> account#.conf |
00:28.46 | Dr-Linux | bjohnson: SIPPS << is softphone like x-lite |
00:29.14 | DEEZED | this will result in numerous account#.conf files. will this slow down or hinder asterisk by having to read each file? |
00:29.47 | Dr-Linux | bjohnson: i'm using SIPPS softphone trail version, look for regisration option over it .. |
00:30.59 | JerJer[interop] | DEEZED: why would you need a whole file for each account? |
00:31.23 | tzafrir_laptop | I'm trying to build asterisk-addons. I get the following error from mkdep (in app_addon_sql_mysql.c) |
00:31.27 | tzafrir_laptop | /usr/include/sys/types.h:158:20: missing binary operator before token "(" |
00:31.33 | DEEZED | to make it easy for a php script to modify each account instead of creating one big file |
00:31.39 | tzafrir_laptop | Any idea what am I missing? |
00:32.01 | JerJer[interop] | a ( |
00:33.28 | tzafrir_laptop | DEEZED, I don't supposed it is much of an overhead. Anyway, you can use globbing in #include |
00:33.45 | Dr-Linux | anybody know about SIPSS softclient ? |
00:35.16 | *** join/#asterisk advorak (~advorak@12-220-96-185.client.insightBB.com) |
00:36.48 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
00:37.00 | *** part/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147) |
00:38.46 | DEEZED | thanks tzafrir_laptop |
00:40.34 | *** join/#asterisk [shodan] (~shodan@216.113.99.160) |
00:43.25 | *** join/#asterisk Newbie___ (~me@218.208.232.24) |
00:43.37 | Newbie___ | hi, anyone familiar with perl ? |
00:43.55 | iq | ~perl |
00:43.56 | jbot | extra, extra, read all about it, perl is at http://www.handhelds.org/z/wiki/Perl or at http://www.perl.com, or a knitting stitch, or the Pathologically Eclectic rubbish Lister, or that other "P" language |
00:46.08 | tzafrir_laptop | Newbie___, ask your question, anyway |
00:47.43 | Newbie___ | tzafrir_laptop: all my calls are now channel to only one source, i would like * to channel to other provider if it matches the area code |
00:47.52 | Newbie___ | i think is written in perl |
00:48.12 | DEEZED | least cost routing? |
00:49.30 | Newbie___ | yes, a LCR http://pastebin.ca/8754 |
00:49.40 | Newbie___ | thats my original source |
00:50.04 | Newbie___ | right now , everything goes to span2 |
00:50.42 | Newbie___ | i like * to go to span1 ie if area code is 416 |
00:51.04 | *** join/#asterisk mariop (~mariop@201.133.224.253) |
00:51.09 | mariop | hi |
00:51.45 | DumbDude | hi mariop |
00:52.28 | mariop | hi |
00:52.39 | mariop | i'n newbie with asterisk |
00:52.42 | DumbDude | whats up mariop |
00:53.00 | mariop | exist some gui to administrate the asterisk |
00:53.21 | iq | mariop, yes there are few available |
00:53.22 | *** join/#asterisk tugalone (~tugalong@pcp0010303951pcs.avenel01.nj.comcast.net) |
00:53.47 | JerJer[interop] | interop question; does the message waitng indication on SIP need a valid type=peer? |
00:53.54 | mariop | what you recomend me i need one simple and very very easy to use |
00:53.59 | iq | mariop, http://www.voip-info.org/wiki-Asterisk+GUI |
00:54.10 | JerJer[interop] | i blieve so, just want to verify |
00:54.35 | tugalone | do sip phones work with asterisk? |
00:54.44 | JerJer[interop] | tugalone: um yes |
00:54.45 | Newbie___ | of course, tugalone |
00:55.06 | iq | tugalone, they need to be plugged in and properly configured ;) |
00:55.10 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
00:55.13 | tugalone | how about across nat's? |
00:55.18 | Newbie___ | hi, neopher |
00:56.03 | iq | tugalone, I never experienced any problem. But some people say that they do. |
00:56.30 | Newbie___ | grrr, GSM channel bank is still not available on ebay |
00:57.03 | JerJer[interop] | sip and nat can be made to cooperate but is not perfectly friendly |
00:57.36 | tugalone | iq: hmm okay. where do i find technical information on asterisk? i'm looking for - interfacing with the PSTN network, the protocol that's used for signaling and media and the likes? |
00:58.03 | iq | tugalone, http://www.asterisk.org |
00:59.36 | mariop | i have a project i need to know if posible withe asterisk i need to coneectr the pstn to the pbx create with asterisk and the extensions management via ip is posible? |
01:00.02 | h3x | mariop: only 100 people have done that already :) |
01:00.15 | iq | mariop, where are you located? |
01:00.52 | mariop | mexico |
01:01.08 | JerJer[interop] | better get a license then |
01:01.24 | mariop | in this case my PSTN need e1 card |
01:01.26 | iq | mariop, technically it is possible. Check to make sure it is not illeagle |
01:01.28 | mariop | license |
01:01.28 | h3x | s/license/beer, drugs, hookers for telmex/ |
01:01.52 | mariop | nop |
01:02.25 | iq | mariop, then start reading :) |
01:02.26 | mariop | no is ilegal meantime i'm not make me a carrier of telephony |
01:02.31 | mariop | like telmex |
01:02.36 | mariop | yep |
01:02.59 | mariop | the e1 i see the only company sell is diginum |
01:04.35 | *** join/#asterisk Rick_Hunter (~rhunter@03-111.008.popsite.net) |
01:07.09 | *** join/#asterisk AmaDEE0_ (~foo@209.52.90.78) |
01:07.28 | AmaDEE0_ | Does anyone have AMP working with Postgresql? |
01:07.42 | h3x | amp? |
01:07.58 | shmaltz | anybody want to take a look at this: |
01:08.00 | shmaltz | http://lists.digium.com/pipermail/asterisk-biz/2005-April/004017.html |
01:08.09 | Zipper_32 | apache mysql and perl?... |
01:08.30 | Newbie___ | anyone help me on LCR with perl ? |
01:08.41 | h3x | least cost routing? |
01:08.50 | Newbie___ | h3x: yes |
01:08.54 | h3x | international? |
01:09.12 | Newbie___ | internation and domestic |
01:09.22 | h3x | im working on a solution for US Domestic |
01:09.33 | h3x | you can't do a NPA/NXX table because of ported numbers |
01:09.50 | h3x | well you can, but 20% of numbers are ported now and you'll misroute calls |
01:10.21 | Newbie___ | i have not try anything yet, right now all my calls are going to one provider |
01:10.43 | tzanger | h3x: there's got to be a solution because it's already in use |
01:10.49 | tzanger | h3x: I'm guessing it's ss7 access |
01:10.52 | h3x | oh so you are going to send all domestic calls to one place |
01:11.09 | Newbie___ | for now, all domestic and internation going to one place |
01:11.13 | h3x | tzafrir: correct, but nobodys written a module for open source software like asterisk or ser yet |
01:11.39 | Newbie___ | need to sent few calls based on area code / country code to other provider |
01:11.53 | tzanger | would it not be possible, at least in interim to just dump the NANPA to a table every week or so and run off that |
01:12.00 | h3x | newl: aren't your domestic rates based on LATA by OCN Class? |
01:12.05 | h3x | silly nick completion |
01:12.20 | h3x | tzanger: that dosent help with ported numbers. |
01:12.41 | h3x | You can port wireless -> wireline, wireline -> wireline, wireless -> wireless, ilec to clec, etc. |
01:12.49 | tzanger | h3x: sure it would (maybe I have my terminology wrong) -- I'm looking at a table that has number mappings and who they should route to |
01:12.59 | h3x | if you misroute a call you could be paying $.03/Min instead of like <$0.01/minute |
01:13.05 | tzanger | h3x: I understand |
01:13.14 | h3x | tzafrir: you have to send a 10 digit query to a ss7 database provider |
01:13.26 | h3x | and it responds with the OCN of the place you are calling |
01:13.33 | tzanger | h3x: right but do they not publish "as of" lists of the numbers? |
01:13.37 | h3x | you take that OCN and figure out the LATA from the area code |
01:13.49 | h3x | tzanger: its not a block of numbers, every number could be ported |
01:13.52 | tzanger | I know |
01:14.04 | tzanger | but at the present there are only a small percentage of ported numbers |
01:14.07 | tzanger | compared to all of NANPA |
01:14.10 | h3x | take for instance people that port their number from uhm... bellsouth over to vonage, well vonage could be using XO as their underlying carrier for the DID |
01:14.18 | h3x | No way dude, its 20% |
01:14.21 | h3x | its terrible |
01:14.23 | tzanger | h3x: 20% is still only 1/5 |
01:14.24 | tzafrir_laptop | my errors were gone when I removed -I/usr/include/asterisk from cc's command-line |
01:14.39 | h3x | that 20% error could cost you tens of thousands of $ as a carrier |
01:14.46 | tzanger | h3x: I'm not saying ignore it |
01:14.55 | tzanger | I'm saying you have the "traditional" NANPA mappings to the oCN |
01:15.03 | h3x | The bigger problem is when you are billing your wholesale customers |
01:15.04 | tzanger | and then this table of the 1/5 of all NANPA that is "weird" |
01:15.06 | h3x | you could bill the wrong rate |
01:15.19 | tzanger | now you do a dump on Monday |
01:15.26 | tzanger | and one of those numbers changes on Tuesday |
01:15.30 | tzanger | you'll misbill for aweek |
01:15.39 | h3x | nanpa dosent give you ported number information |
01:15.50 | h3x | they just tell you thousands or ten thousands blocks of who can assign new numbers in those prefixes |
01:15.52 | tzanger | h3x: and it's not possible to get that information in an "as of this date" format? |
01:15.55 | h3x | NEW numbers |
01:16.16 | h3x | thats all NANPAs information means now |
01:16.21 | h3x | who can assign new numbers |
01:16.23 | tzanger | h3x: it's not possible to get a list of all ported numbers and their correct OCN in "as of this date" format? |
01:16.44 | h3x | Sure you can but the SS7 providers charge a setup fee of $2 Million for that information |
01:16.45 | h3x | seriously |
01:16.45 | h3x | heh |
01:16.50 | tzanger | ok |
01:16.52 | h3x | thats what SNET told me |
01:16.56 | tzanger | so that is certainly out of the question |
01:17.01 | h3x | Sentito or whatever |
01:17.03 | ManxPower | Mere mortals can't get that information |
01:17.10 | h3x | its a big ass database with millions of records |
01:17.10 | tzanger | how much trouble would you get into for "wardialing" the SS7 database and creating your own list? :-) |
01:17.11 | h3x | LIDB |
01:17.22 | ManxPower | A "mere mortal" is someone with less than $2 million in available money, of course. |
01:17.23 | h3x | Uh... you have to pay by the query |
01:17.31 | tzanger | h3x: I'm kidding |
01:17.41 | tzanger | if you can query the SS7 database you may as well just do it right and query it |
01:17.58 | h3x | yes, so what im doing is putting together a module for asterisk, ser, etc that will query |
01:18.06 | h3x | it queries by going through my ss7 gateways |
01:18.17 | h3x | im going to give it away for whatever my cost is so it drives my volume up and brings my cost down |
01:18.33 | tzanger | h3x: so what are you complaining about then if you're building it? :-) |
01:18.33 | h3x | the per query cost that is |
01:19.05 | h3x | I'm not complaining about anything, it just sucks that there isnt a solution out there for reasonable software yet |
01:19.07 | *** join/#asterisk santiago (~santiago@63.245.86.85) |
01:20.49 | tzanger | h3x: perhaps but you are carving out a business opportunity from that lack of solution |
01:20.56 | h3x | of course thats probably because until recently there weren't any IP interfaces to LIDB |
01:21.18 | h3x | well its a matter of aggregating to lower my costs than making money in this case :) |
01:22.31 | bkw_ | anyone here thats not in the US or Canada sms me? |
01:22.37 | bkw_ | I wanna test something |
01:24.15 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
01:24.59 | mithro | howdy people, anyone here know how FXS work electrically? |
01:26.02 | ManxPower | mithro: how much info do you want? |
01:26.13 | mithro | i'm intrested in finding out how the FXS cards provide the required 48V DC and 120V AC 25Hz ringer |
01:26.35 | ManxPower | actually it's 90VAC ringer |
01:27.06 | ManxPower | mithro: look at the datasheet for the TigerJet FXS chips |
01:27.07 | mithro | yeah, how do they do that from the 12V rail in the computer? |
01:27.31 | ManxPower | mithro: they prolly drop the amps and up the voltage. |
01:27.42 | ManxPower | I don't know much, but I do know that's pretty trivial |
01:28.02 | mithro | yeah, i know how to do everything apart from that in designing a FXS |
01:30.18 | *** join/#asterisk |nix (~inix@202.148.164.48) |
01:31.01 | *** join/#asterisk PBXtech (~nik@70-58-41-173.slkc.qwest.net) |
01:31.03 | *** join/#asterisk d00gster (~doughant@Toronto-HSE-ppp3661779.sympatico.ca) |
01:33.18 | |nix | hey PTG1234, you're around? |
01:36.09 | florz | mithro: I dunno how that chip does it or any other FXSes either, but at least some cheaper analog PBXes often simply use 24 V at 50 Hz or something (in europe, that is) simply taken from the grid using some transformer |
01:36.27 | JerJer[interop] | uses ohms law |
01:36.48 | *** join/#asterisk zhier (~nick@61.144.20.3) |
01:36.53 | JerJer[interop] | takes a lot of current with a transformer to create a lot of voltage |
01:36.58 | JerJer[interop] | with little current |
01:38.05 | harryvv | JerJer learned all of that in electronics collage. :) |
01:39.08 | zhier | how to register a user to my sever? that is say how to configure my conf file to receive the registratiom. |
01:39.12 | JerJer[interop] | more like high-school |
01:39.15 | florz | JerJer[interop]: Though, neither is 90 V a lot of voltage nor does it take a lot of current to make a telephone ring =:-) |
01:39.19 | JerJer[interop] | then self taught the rest |
01:39.21 | harryvv | mithro, stepping up 12 volts though a class a amplifier or transformer. |
01:39.37 | JerJer[interop] | florz: hence why there is a molex power connector on the TDM400P chassis |
01:39.42 | mishehu | ipx hehe |
01:39.59 | mishehu | deprecated protocol. |
01:40.06 | JerJer[interop] | and also why one cannot run a small power supply and expect all four phones to ring at the same time |
01:40.43 | tzanger | the bigger problem is that the 12V line on the PCI connector is not very beefy |
01:40.53 | ManxPower | mishehu: We are trying to track down and kill open wireless networks in our building |
01:40.59 | tzanger | thin traces and big current spikes don't mix |
01:41.05 | harryvv | whats the maxium current the pci will take |
01:41.13 | tzanger | that and those current surges are coupling to all the other lines nearby |
01:41.19 | tzanger | harryvv: you'd have to check the spec |
01:41.23 | harryvv | yea |
01:41.27 | tzanger | but even motherboards that are PCI2.2 compliant aren't |
01:41.32 | harryvv | probebly a few miliamps |
01:42.05 | tzanger | I'd be willing to suggest high dozens to maybe a hundred or so |
01:42.09 | tzanger | just from intuition |
01:43.01 | harryvv | yea probebly ;) |
01:43.37 | hermie | USB is ~500 milliamps |
01:44.24 | mishehu | ManxPower: your building meaning your residence or your employment? |
01:44.47 | tzanger | hermie: yeah but that's 500mA at 5V or 3.3V, not 12V |
01:45.16 | harryvv | manx why are you trying to kill it? unauthorized wireless routers being put on your network? |
01:45.18 | harryvv | brb |
01:45.34 | ManxPower | mishehu: the office building of my largest customer |
01:45.34 | harryvv | 5 volt ttl |
01:45.40 | tzanger | but the entire point is that you can't rely on it, which is why the molex connector's there. hard drives and tape backups are very power hungry devices on the 12V line, especially when starting up |
01:45.46 | tzanger | harryvv: USB is not TTL |
01:46.00 | harryvv | i see |
01:46.19 | harryvv | but mabey in the ps ? |
01:46.53 | tzanger | TTL is a logic level specification, it has nothing to do with power |
01:46.54 | tzanger | well |
01:46.59 | tzanger | okay it has something to do with power |
01:47.00 | ManxPower | harryvv: Once I figure it out I'll use arp spoofing and a web page to basically say "Your wireless network is not secured. Turn on the security features of your wireless devices. Someone sitting in the parking lot with a laptop and wireless card could be capturing all your data! |
01:47.03 | tzanger | but not how you're using it. :-) |
01:47.44 | harryvv | :) |
01:48.08 | ManxPower | harryvv: We are finding that our users with laptops and wireless cards are trying to send corporate data over other companies wireless networks and we are tired of it. |
01:48.23 | tzanger | ManxPower: bitchslap them |
01:48.30 | ManxPower | There is NO reason to have an unsecured wireless network in an office building. |
01:48.36 | tzanger | ManxPower: we do it |
01:48.38 | tzanger | but |
01:48.46 | tzanger | the wireless network attaches OUTSIDE the firewall, not inside |
01:48.50 | ManxPower | tzanger: The coffee shop is going to be pretty pissed if we implement this at other locations. |
01:48.56 | harryvv | not good. alot of that info is considered sensitive? btw dinner. I will tell you of a defence contractor that almost put my freind in prison for what one of the company employees almost did. |
01:49.01 | tzanger | so if you're on the wireless network you're also on the VPN to get in to the network |
01:49.09 | ManxPower | tzanger: You know that XP will BRIDGE wireless and wireline networks, right? |
01:49.16 | tzanger | ManxPower: we don't run XP |
01:49.40 | ManxPower | tzanger: The company does not own the computers. |
01:49.45 | ManxPower | The users own their computers. |
01:49.52 | tzanger | ManxPower: oh yeah I remember you told me that |
01:50.01 | tzanger | so they are ACTIVELY trying to send corporate data around? |
01:50.40 | ManxPower | tzanger: Whenever I refer to "my largest customer" just imagine someone that does a lot of LSD writing up how a company should be run and you'll have a pretty good idea... |
01:50.43 | ariel_ | anyone want to spend $ 2800.00 on a domain name? www.vitaphone.com hummmm expensive. |
01:50.46 | ManxPower | tzanger: Not that we know of...... |
01:51.04 | tzanger | ManxPower: so why are they hooking up both wireless and wired connections? |
01:51.14 | ManxPower | tzanger: because they are idiots |
01:51.17 | tzanger | ok |
01:51.22 | tzanger | that works. :-) |
01:51.38 | DEEZED | anyone running asterisk on adsl? |
01:51.46 | ManxPower | tzanger: The person that generates the most revenue is someone that sells most of the houses to people she meets at the country club. |
01:51.47 | ariel_ | DEEZED, yes |
01:51.59 | DEEZED | how many calls can you handle at once? |
01:52.10 | ManxPower | The male version meets clients while playing golf. |
01:52.22 | hermie | google thinks 12v@50mA |
01:52.29 | ariel_ | DEEZED, it depends on codec. But if I use gsm about 4 |
01:52.45 | DEEZED | =/ |
01:52.48 | hermie | wait, i've got it! |
01:52.53 | ManxPower | DEEZED: We have the same questions as the last 300 people that asked the same thing: What is the upload and download bandwidth, what codec will you be using, will you be using trunking? |
01:53.01 | tzanger | 12V@50mA for what |
01:53.13 | hermie | 500mA@12V,100mA@-12V for PCI 2.3 |
01:53.21 | DEEZED | 3000/384 |
01:53.26 | DEEZED | gsm |
01:53.28 | hermie | that's per slot max |
01:53.29 | tzanger | 500mA wow they claim that?? |
01:53.30 | DEEZED | iax trunk |
01:53.37 | ManxPower | hermie: As Digium discovered a lot of motherboards don't let you draw that much current |
01:53.49 | tzanger | I call bullshit, I don't think any motherboard can handle a half an amp on the 12V rail PER SLOT |
01:53.51 | hermie | tzanger: that's the max allowed |
01:54.02 | tzanger | maybe that's what the spec says but I'll bet there isn't a commodity mobo that handles it |
01:54.18 | hermie | tzanger: most can't handle >120 on the 12v, but that's what the spec says |
01:54.23 | tzanger | :-) |
01:54.30 | ManxPower | DEEZED: GSM codec is about 12k + IP overhead of about 18k |
01:54.51 | ManxPower | so 30 or 31. |
01:54.59 | *** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net) |
01:55.00 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
01:55.13 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
01:55.27 | ManxPower | DEEZED: so about 12 calls if you live in a PERFECT universe and you have no other traffic. |
01:55.42 | ManxPower | In the universe I live in expect about 6 calls |
01:57.04 | syslod | HEAD broken. /usr/lib/asterisk/modules/chan_iax2.so: undefined symbo |
01:57.04 | syslod | l: ast_memcpy_byteswap |
01:57.46 | ariel_ | GSM is 32 plus 16 to 18 for the ethernet overhead. g729 is about 12 plus. |
01:57.58 | DEEZED | ManxPower.. what do you use your pbx for? |
01:58.14 | ManxPower | DEEZED: phone calls |
01:58.23 | tzanger | hahaha |
01:58.26 | tzanger | I was just about to say that |
01:58.54 | robl^ | DEEZED: he also runs one of those pay per minute phone sex lines . :) |
01:59.07 | ManxPower | http://www.packetizer.com/voip/diagnostics/bandcalc.html |
01:59.29 | ManxPower | robl^ is my..er...biggest customer! |
01:59.51 | ManxPower | DEEZED: I personally use Asterisk to get free phone calls to my lovers. |
01:59.57 | robl^ | ManxPower: that's right! "Press 4 for Teltetubbies in heat." |
02:00.11 | ManxPower | I professionally use Asterisk for free phone calls to my customers and I use it as a PBX for customers |
02:00.50 | ManxPower | ariel_: please step away from the alcohol. |
02:00.55 | ManxPower | it's IP overhead, not ethernet overhead |
02:01.10 | DEEZED | ic. |
02:02.24 | harryvv | arial want to create a voip domain? |
02:03.13 | *** join/#asterisk tessier (~treed@222.253.72.192) |
02:05.52 | *** join/#asterisk |nix (~inix@202.148.164.48) |
02:06.22 | harryvv | lets not get layers confused :) |
02:06.59 | zack | is it impossible to use chan_bluetooth without rebuilding all of asterisk? |
02:08.47 | ManxPower | My cat is pretty nice to me when his supper dish is empty. |
02:09.17 | ManxPower | zack: I wasn't aware that chan_bluetooth even worked at all |
02:09.30 | JonR800 | it doesn't. |
02:09.36 | zack | ManxPower: hmm, i might care less, then. |
02:10.42 | *** join/#asterisk Slainte (~Slainte@66.55.112.85.ppp.northrock.bm) |
02:12.18 | Slainte | Anyone have a working example of SetAccount, in their extensions.conf? I want anyone who enters a long distance call to enter a four digit code, so we can back charge the client. |
02:13.07 | mishehu | iax2 trunking works for all codecs, no? |
02:13.11 | Slainte | I dont need to authenticate the code, it just needs to be entered. (validation to come later) |
02:13.25 | mishehu | somebody had posted on the mailing list back in february that he thought that you can't trunk g711ulaw |
02:13.33 | syslod | Anyone testing HEAD? |
02:13.38 | Zipper_32 | For a home setup, do I need a card for use of softphones over a DSL connection? |
02:14.08 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
02:14.13 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
02:14.31 | ManxPower | I'll be in #asterisk-stable if people have questions about 1.0.x Asterisk |
02:14.35 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
02:15.21 | h3x | reading scrollback.... |
02:17.36 | syslod | :( cvs update -D yesterday |
02:18.11 | Newbie___ | exit |
02:18.18 | h3x | Zipper_32: Im more pissed off about a customer i could have had that spent $5 Million on an asterisk powered call center%!# |
02:18.30 | Zipper_32 | sweet jebus... |
02:18.37 | h3x | no shit mang |
02:18.42 | Zipper_32 | That's nuts.. |
02:18.50 | Zipper_32 | How the... wow... just wow. |
02:18.58 | syslod | $5 Million??? |
02:18.59 | h3x | im sure they paid less for the digium commercial license than they did on dinner to take the people out , wine and dine them to buy it |
02:19.11 | h3x | Yeah it was some non profit orginzation in the midwest |
02:19.12 | jakepdev | h3x - I feel your pain |
02:19.27 | Zipper_32 | <3 for h3x |
02:19.42 | h3x | i dont have the <3 to charge |
02:19.44 | h3x | $5 mil |
02:19.44 | h3x | well |
02:19.57 | Zipper_32 | hehe, I"m sure you could... =) |
02:20.02 | h3x | I hope they gave all the 300 agents a color cisco phone for that price |
02:20.03 | h3x | damn |
02:20.16 | Zipper_32 | lol, |
02:20.20 | *** join/#asterisk tessier (~treed@222.253.72.192) |
02:20.31 | h3x | jesus, why dont i ever get leads like that before people buy stuff |
02:20.56 | *** join/#asterisk iq[laptop] (~iq@207-224-100-90.omah.qwest.net) |
02:21.01 | DEEZED | noob question.. what does 5555 mean in "210 => 5555,John Smith,jsmith@yourdomain.com" |
02:21.20 | PBXtech | your saying someone built a $5m * call center? |
02:21.39 | h3x | i didnt personally see the system but my counterpart paid them a visit and noticed it had te410 cards in it.. |
02:21.55 | h3x | PBXtech: yeah supposedly, well im sure it had other things in it too but they got screwed |
02:22.01 | PBXtech | for $5m? |
02:22.03 | PBXtech | damn |
02:22.16 | PBXtech | thats like 4.5 profit huh |
02:22.18 | PBXtech | :) |
02:22.24 | h3x | it aint the first multi million $ call center raping ive heard of though |
02:22.33 | h3x | i saw another one a couple years ago for century 21 here in vegas |
02:22.48 | h3x | they put in a whole 60 cisco 7960 phones |
02:22.50 | PBXtech | come up to SLC call center central here |
02:22.57 | h3x | ran that into cisco catalysts, and then cisco |
02:23.03 | h3x | er.. as5300s |
02:23.06 | DEEZED | I use to work for convergys in SLC |
02:23.11 | h3x | and then from the 5300s into shitty ISA dialogic based dialer |
02:23.22 | h3x | that system sold for $2.5M and the guys never got it working completely |
02:23.25 | PBXtech | ive walked into convergsys :) |
02:23.27 | h3x | and they all moved to hawaii |
02:23.32 | h3x | and disappeared |
02:23.37 | |nix | anyone using PT1234's SIP code? |
02:23.39 | h3x | the customer cant find em, and went under |
02:24.13 | h3x | you wouldnt believe the sales pitch they used to sell it having VoIP capability |
02:24.13 | PBXtech | youd think for 2.5m you'd do your homework |
02:24.27 | h3x | the reason they decided to buy it is because "you can transfer both voice AND data!%#!!#% to another party in another office!" |
02:24.30 | h3x | well no shit you can do that with tdm too |
02:24.37 | Zipper_32 | How much would you guys charge for a 9-line asterisk system with support for 20-25 phones? |
02:24.38 | h3x | but it dosent matter coz they never got it to work |
02:24.39 | PBXtech | i lose $100k deals cause i dont have enough employees (which is BS) |
02:24.50 | h3x | Zipper_32: Uhmm.. how does $1.9 million sound? hahah |
02:25.10 | h3x | PBXtech: just hire some india call center for $2 an hour |
02:25.17 | Zipper_32 | h3x: .... How about no? =) |
02:25.32 | PBXtech | HaHaHa staff of 100 then huh |
02:25.33 | h3x | Damn I always lose to higher bidders!!! |
02:25.38 | Zipper_32 | hehe |
02:26.04 | Zipper_32 | I'm trying to learn how to set this stuff up, I just finished installing asterisk today. |
02:26.17 | h3x | where are you at |
02:26.43 | syslod | Next time someone sees a $5 million bid I'll be happy to send you all the staff you need. |
02:26.44 | harryvv | h3x on ebay? what you loose at the last second? |
02:26.48 | DEEZED | sorry to ask again. but what does 5555 mean in "210 => 5555,John Smith,jsmith@yourdomain.com". Is it the phone extension? If so, then can i just put any number if i don't want it linked to an actual phone? |
02:27.09 | PBXtech | DEEZED thats the password |
02:27.10 | Slainte | DEEZ password for voicemail |
02:27.16 | DEEZED | thanks guys |
02:27.41 | Zipper_32 | Well, so far h3x, I've setup asterisk on a redhat 8 box, and that's it. I am going to order some cards, but I'm reading up and trying to figure out what I need/. |
02:28.01 | h3x | oh telus. .. hmm |
02:28.11 | h3x | are you gonna use e1s? |
02:28.59 | harryvv | telus your complaints :) |
02:29.25 | h3x | shit, bitchx would never port to atari 2600 |
02:29.39 | h3x | they had enough problems getting it to bsd |
02:29.51 | harryvv | somone shot at a telus worker because we were suffenly hit by very poor service in this area about a year ago. |
02:30.03 | harryvv | suddenly that is :) |
02:30.06 | h3x | daamn |
02:30.33 | h3x | do you got decent aussie did providers down there |
02:30.36 | harryvv | yea it was affecting all of vancouver. Poor service month or more to get a phone line everything. |
02:30.43 | h3x | er canada |
02:30.43 | h3x | doh |
02:30.50 | h3x | smoking too much crack |
02:30.52 | Zipper_32 | hehe |
02:30.58 | Zipper_32 | I don't know what we're going to use yet |
02:31.06 | h3x | why dont you use primus did's |
02:31.11 | syslod | They sell guns in CA? :) |
02:31.15 | h3x | then you dont need phone lines |
02:31.37 | PBXtech | are there any providers that will carry 100+ dids for a comercial co? |
02:31.49 | h3x | PBXtech: where |
02:32.00 | PBXtech | huh i said are there any.. |
02:32.06 | h3x | (i should have just said 'yes') |
02:32.06 | h3x | heh |
02:32.18 | h3x | Yes, but its usually by the minute |
02:32.32 | h3x | since thats the only good way to measure the usage without physical lines |
02:32.37 | h3x | its cheap though |
02:32.44 | PBXtech | how cheep/ who |
02:33.12 | syslod | PBXtech. Any provider will do that. |
02:33.12 | h3x | unfortunately most of the telcos that actually do it with VoIP only do it under wholesale to resellers like us |
02:33.30 | h3x | and all of their coverage areas suck |
02:33.38 | *** part/#asterisk santiago (~santiago@63.245.86.85) |
02:33.39 | h3x | so you have to use them all if you want really good coverage |
02:34.37 | PBXtech | how do you find out who is providing local DIDs to these carrier? that possible |
02:34.51 | syslod | You want to find what? |
02:35.12 | h3x | the largest ones are level3, mci, and maybe xo |
02:35.14 | PBXtech | the local company who is providing DID blocks to like broadvoice |
02:35.30 | PBXtech | of IC |
02:35.33 | syslod | Do you have a NPA/NXX in particluar you want to find a carrier for? |
02:35.48 | h3x | you look up the NANPA assignment of the npa/nxx |
02:36.14 | syslod | LERG has a better listing. |
02:36.32 | h3x | yeah |
02:36.34 | *** part/#asterisk zack (~zack@sebastian.redhat.com) |
02:36.55 | PBXtech | what LERG |
02:37.08 | syslod | What are the resellers doing about 911 and CALEA |
02:37.18 | syslod | Local Excahnge Routing Guide. |
02:37.31 | h3x | it dosent really matter that much if you find out who it is anyway, many of the providers require a $25k a month committment to even get a DID product from them |
02:38.09 | syslod | Level 3 is almost impossible to deal with unless you are a larger carrier. |
02:38.15 | h3x | syslod: CALEA isn't required for VoIP yet |
02:38.42 | h3x | level3's prices suck anyway, i am going to use them but i found several other ones combined together that equals their footprint |
02:38.55 | PBXtech | h3x you new to this channel? |
02:38.57 | harryvv | h3x what to high? |
02:39.04 | h3x | nah i was just gone for quite a while |
02:39.09 | syslod | Well kinda. See if you purchase DID blocks and digital service you still are technically a CALEA product. |
02:39.23 | harryvv | syslod thanks for saying that about level 3 was going to give them a call. |
02:39.36 | h3x | syslod: the underlying carrier that converts to TDM provides CALEA |
02:40.02 | syslod | Yea but its all screwed up in the database. It has the wrong end user information. |
02:40.34 | h3x | harryvv: theres a couple problems with level3, they have the worst volume committment (you need to do $50k in business a month with them to get decent rates) and they have a proprietary voip gateway platform called Viper |
02:40.41 | h3x | it has some issues interfacing to asterisk |
02:40.42 | syslod | harryvv: What are you looking for? Nationwide or select markets? |
02:40.49 | h3x | one of my guys spent 3 months patching asterisk to make it work |
02:41.32 | PBXtech | you resell LD and * box's? |
02:42.09 | h3x | who me? |
02:42.12 | h3x | www.carrierone.net is my company |
02:42.14 | PBXtech | ya |
02:42.28 | h3x | ive been using asterisk for a couple years |
02:42.40 | h3x | but for the DIDs and core, we're deploying SER |
02:44.07 | h3x | among our market niches is giving away free voip gateways to customers doing a lot of traffic on tdm t1's |
02:44.31 | h3x | carrying trunked voip over private line |
02:45.00 | h3x | we might be switching to a DSP based product soon for CPE though |
02:45.37 | PBXtech | there are good non * gateways out there that are good price |
02:45.43 | shmaltz | anybody want to take a look at this: |
02:45.44 | shmaltz | http://lists.digium.com/pipermail/asterisk-biz/2005-April/004017.html |
02:45.47 | syslod | h3x: Compressed voice? |
02:45.55 | h3x | yes, compressed and trunked |
02:46.16 | h3x | we've got some deals with various IXCs that give us flat rate private line to anywhere in the US |
02:46.23 | h3x | its actually cheaper than dedicated internet most of the time |
02:46.24 | BoRiS | what about canada? |
02:46.28 | syslod | shmaltz: What about it? |
02:46.37 | PBXtech | h3x can you PM me your email ide like to talk about this further when im at work |
02:46.43 | shmaltz | I'm looking to pay for this |
02:46.47 | shmaltz | syslod |
02:46.54 | h3x | yes, we'll have canadian DIDs in the near future from a carrier out in one wilshire |
02:46.56 | syslod | shmaltz: We have this. |
02:47.04 | shmaltz | hmmmmmmmm, pm please |
02:47.24 | h3x | we've got our own datacenter here in vegas with private fiber cross connects to our carriers |
02:48.09 | *** join/#asterisk munchausen (~oihsafd@68.71.213-37.atlsfl.adelphia.net) |
02:48.10 | *** join/#asterisk esandeen (~sandeen@sandeen.net) |
02:49.33 | PBXtech | ide to biz with you just to have to come to las vegas for a tour :) |
02:49.44 | munchausen | newbie question: is it possible to "forward" a connected audio stream to a new destination so that the packets can go directly to the new destination instead of bouncing through the old? |
02:50.06 | h3x | hahaha |
02:50.10 | h3x | did i pick a good location or what. |
02:51.30 | BoRiS | hmm |
02:52.20 | bjohnson | munchausen: yes. reinvite in sip |
02:52.43 | *** join/#asterisk malverian (~malverian@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
02:52.50 | munchausen | thanks |
02:52.51 | malverian | Hmm.. |
02:52.55 | Dr-Linux | anybody know about SIPPS softclient ? |
02:53.03 | malverian | I'm having weird lockups when I use my sk98 with my w100p card. |
02:53.40 | malverian | I get lock-ups when I enable the wcfxo and my sk98lin modules together. I saw 2 posts on the mailing list regarding this (unanswered) |
02:53.57 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
02:54.14 | *** join/#asterisk Ferrari (~IPlexbyVe@216.196.255.42) |
02:54.24 | *** part/#asterisk Ferrari (~IPlexbyVe@216.196.255.42) |
03:00.02 | bjohnson | h3x: ny hints at what the costs are for your services? |
03:00.08 | bjohnson | nothing on the web site |
03:00.26 | h3x | it all depends on the location really |
03:00.35 | h3x | theres so many underlying carriers its difficult to determine fixed costs |
03:00.40 | h3x | we're mostly wholesale though |
03:01.03 | munchausen | bjohnson: do people successfully use reinvite to loop a call out through the same provider it came in on? |
03:01.06 | h3x | I don't think I'll really ever have a public DID service, but ill let you guys test it as markets become available though |
03:01.24 | PBXtech | so h3x for 1M minutes we comp'd to "splash" :) |
03:01.43 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-93.modem.logical.net) |
03:01.50 | h3x | haha, i think theres some better shows than that ;) |
03:02.00 | h3x | my friend's wife is the lead dancer in Jubilee... |
03:02.06 | Carp1 | how do you retrieve voicemail passwords? |
03:02.10 | PBXtech | nice |
03:02.20 | PBXtech | [Carp1]: voicemail.conf |
03:02.24 | h3x | The dumb thing is i still haven't seen that show |
03:02.39 | PBXtech | hah its your friends wife |
03:02.40 | Carp1 | Sorry, I havnt been able to use asterisk in like 4 months |
03:02.42 | opus_ | does anyone know about verisign ss7? |
03:02.43 | Carp1 | system went down |
03:02.46 | Carp1 | just got it back up |
03:02.46 | PBXtech | got to support that :) |
03:02.51 | h3x | well |
03:03.01 | PBXtech | :/ |
03:03.17 | syslod | verisign ss7 sucks |
03:03.22 | shmaltz | anybody want to take a look at this: |
03:03.24 | shmaltz | http://lists.digium.com/pipermail/asterisk-biz/2005-April/004017.html |
03:03.38 | PBXtech | shmaltz you on repeat mode? |
03:03.55 | opus_ | syslod - how does it work? do they have a price online, perhaps you might know it? |
03:03.55 | shmaltz | PBXtech, almost |
03:03.57 | bjohnson | munchausen: no |
03:04.09 | shmaltz | I'm a bit late to get this up and running |
03:04.09 | PBXtech | heh |
03:04.13 | bjohnson | munchausen: only works between sip clients you control |
03:04.29 | syslod | opus_: Its almost impossible to get in touch with them. Once you do they'll miss the first 6 months of FOC and you'll give up. |
03:04.32 | h3x | yeah well, verisign for ss7 is like using h323 for voip |
03:04.56 | PBXtech | if * had an interface that nice, then we wouldnt have anything to do in here |
03:04.58 | syslod | opus_: What are you looking for specfically from verisign? And why them? |
03:05.17 | opus_ | syslod - I'm trying to not buy a FXS card |
03:05.18 | h3x | I met a systems engineer with verisign that knew his shit |
03:05.18 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
03:05.22 | h3x | i have his card somewhere |
03:05.26 | syslod | Huh??? |
03:05.44 | syslod | Illiumnet |
03:05.49 | h3x | their solutions are too complicated though |
03:06.00 | opus_ | i mean, never mind. Thats what I am trying to do. I just came across verisign and wanted to know about it, completely seperate |
03:06.03 | syslod | opus_: Explain? |
03:06.25 | syslod | opus_: You trying to set cid? |
03:06.44 | bjohnson | opus_: trying not to buy a fxo? just use a normal voip provider |
03:06.56 | opus_ | No, I'm just looking for a cheap FXS card. sorry for the confusion |
03:07.21 | h3x | whats verisign ss7 have to do with fxs cards |
03:07.31 | opus_ | Nothing, sorry for the confusion. |
03:07.43 | bjohnson | opus_: spa 2000 is cheapest per port next to linksys pap2-na (if you can find them) |
03:07.47 | syslod | h3x: Are you a carrier or reseller? |
03:07.52 | opus_ | thanks |
03:07.52 | bjohnson | both are 2 port fxs |
03:08.06 | opus_ | why would the linksys pap2-na be hard to find? |
03:08.12 | h3x | syslod: Whats the difference? :) |
03:08.18 | PBXtech | unlocked |
03:08.25 | bjohnson | opus_: time for you to go shopping |
03:08.37 | h3x | after the incest i've seen in telcos here im not sure who exactly really carries the traffic anyway |
03:08.38 | opus_ | here is an off the wall question.. do all FXS cards support rotary? |
03:08.42 | syslod | h3x: You have to file a 499 or you don't |
03:08.56 | bjohnson | opus_: I don't know if ANY support rotary |
03:09.08 | h3x | you have to file a 499A exemption if you are an ESP :) |
03:09.23 | h3x | anyway, yes we're an enhanced service provider |
03:09.31 | h3x | we dont have any carrier licenses, we don't want any |
03:09.42 | syslod | Own any last mile? |
03:10.21 | h3x | noley, we use the rediculiously cheap wholesale prices of all the IXCs we're directly cross connected to |
03:10.25 | h3x | s/noley/no/ |
03:10.53 | h3x | thats a terrible business to be in |
03:11.06 | PBXtech | just ask XO :) |
03:11.07 | h3x | when I can get a T1 private line from vegas to new york with the local loop for under $500 |
03:11.20 | h3x | why the hell would i want to spend millions building a network |
03:11.21 | syslod | That doesn't work everywhere. |
03:11.23 | opus_ | nobody has been able to hack the linksys yet? |
03:11.28 | h3x | it works damn near everywhere |
03:11.32 | *** join/#asterisk Newbie___ (~me@218.111.224.175) |
03:11.41 | *** join/#asterisk CoffeeIV (rgr@cpe-70-112-100-20.austin.res.rr.com) |
03:11.50 | Newbie___ | hi, anyone help me on LCR with perl please ? |
03:11.52 | PBXtech | [opus_]: openWRT |
03:11.56 | h3x | i have had to get fiber constructed for customers before that are in BFE |
03:11.57 | opus_ | this sucks: **You must be buying at least 5 of these units to continue with the checkout process** |
03:12.09 | h3x | ordering fiber construction from a railroad company or something, but they had to pay for it |
03:12.11 | harryvv | h3x thats nice |
03:12.12 | harryvv | :) |
03:12.20 | bkw_ | grrrrrrrrrrrrrrreat |
03:12.32 | PBXtech | its tony the tiger |
03:12.33 | syslod | h3x: Most rural areas that are run by independants etc you'll pay $500 to get to customer from the local tandem. |
03:12.36 | bkw_ | wish I could solve this stupid issue with usb modems in linux |
03:12.36 | h3x | sometimes you can buy distressed fiber from a bankrupt company and pay to get it spliced |
03:12.43 | harryvv | h3x when you first started out how did you obtain the money to start the biz |
03:12.51 | h3x | syslod: I usually find a way to bypass the ILEC |
03:13.05 | syslod | Yea. Me too. |
03:13.06 | h3x | yes even in rural areas |
03:13.16 | h3x | but it really dosent matter that much with compressed voice anyway |
03:13.30 | h3x | if somebody need 120 channels of VoIP, if a loop costs $1000 a month who cares |
03:13.33 | h3x | it really isnt that bad |
03:14.05 | CoffeeIV | what is a softphone equivalent to XLite that I can use on linux ? will ohphone work with asterisk ? |
03:14.22 | h3x | harryvv: well we originally developed CTI software with aculab prosody cards on solaris, then we acted as a master agency selling telecom services for the past almost 5 years |
03:14.24 | harryvv | 120 channels for 1000? nice. I was quoted 23 for 600 bucks here in town. |
03:14.32 | h3x | commission checks from that -> carrier one facility |
03:14.41 | syslod | bkw_: Is your valetpark still supported/available? |
03:15.26 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
03:15.32 | PBXtech | except $1000 for 5/T1 equivelant isnt much savings. |
03:16.05 | h3x | Yeah but that $1000 isnt shit to a customer thats spending $10ka month in long distance |
03:16.31 | harryvv | 600 dollars for 1 T-1 here via thinktel/alstream |
03:17.27 | PBXtech | you got to be sub 2 cents |
03:17.30 | BoRiS | $650-$800 for a T1 via grouptelecom |
03:17.40 | file[laptop] | yay BoRiS! |
03:17.54 | h3x | you wouldn't believe it but many call centers etc. are still paying over .02/Min |
03:17.56 | h3x | for TDM |
03:18.04 | syslod | Your talking about different things. A T1 loop is cheaper usually than end to end service. |
03:18.08 | h3x | plus the taxes |
03:18.16 | PBXtech | plus the tax is KEY |
03:18.26 | PBXtech | F-in 14% ish |
03:18.28 | harryvv | h3x are these like call centers linking india with the states? |
03:18.28 | h3x | The point i was making is that long haul + local loop is usually cheaper than internet access on both sides plus a t1 loop |
03:18.44 | h3x | harryvv: no, india call centers want to pay like .000002/Min |
03:18.48 | harryvv | hahaha |
03:18.53 | PBXtech | h3x true |
03:18.53 | h3x | are you kidding |
03:18.56 | harryvv | your joking right? |
03:18.56 | h3x | they make $2 an hour |
03:18.59 | h3x | Yes |
03:19.00 | PTG1234 | did you guys see what happened with those soyo ip gateways? |
03:19.04 | harryvv | :) |
03:19.18 | *** join/#asterisk eaperezh (~eaperezh@200.75.242.202) |
03:19.21 | h3x | they probably look at voipjet's .013/Min advertised price and email to ask for .007/Min flat |
03:19.28 | h3x | with a 50/50 rboc ratio |
03:19.34 | file[laptop] | HA |
03:19.37 | file[laptop] | a 50/50 rboc ratio |
03:19.40 | h3x | yeah hahahaha |
03:19.47 | opus_ | PTG - yeah |
03:19.49 | Slainte | anyone in here happen to know postfix? |
03:19.51 | opus_ | PTG -- what the fuck?? |
03:19.56 | harryvv | h3z i see voipjets lag at 2000 ms |
03:19.59 | harryvv | alot of times. |
03:20.03 | h3x | indians have to call BFE rural areas to sell widgets |
03:20.09 | PTG1234 | they freaken told me they had the wrong specs up.. did they with you opus? |
03:20.12 | h3x | harryvv: holy cow |
03:20.20 | PBXtech | i have a 30ms ping to their LA side |
03:20.22 | opus_ | PTG -- yup, shentech.com ?? |
03:20.23 | PBXtech | fast |
03:20.35 | h3x | 30ms? |
03:20.42 | PTG1234 | opus_: yep.. i said your bait and switching me which is illegal in the US :P) |
03:20.42 | h3x | I have 3ms time to Qwest's voip gateway |
03:20.45 | h3x | from my DS3 on Wiltel |
03:20.52 | h3x | er wait, i think its like 6ms |
03:21.02 | opus_ | PTG -- i'm having my friends in china locate the factory. |
03:21.04 | PBXtech | thats from ELI T1 |
03:21.09 | PBXtech | in SLC |
03:21.11 | h3x | ew, eli.. hmm |
03:21.15 | PBXtech | heh |
03:21.20 | PTG1234 | opus_: let me know what you find out |
03:21.20 | h3x | they used to be the bomb in data |
03:21.21 | opus_ | PTG -- I still want one, and I still want that price |
03:21.25 | PTG1234 | opus_: no one seems to sell them here |
03:21.33 | PTG1234 | opus_: i am just curious what the real price of them is |
03:21.39 | h3x | vegas is the only market they dont have voice in which sucks |
03:21.40 | opus_ | $399 i think |
03:21.45 | PBXtech | i got a killer rate on local PRI. $450 with FX from 3 rate areas |
03:21.48 | h3x | they are cheap as hell on voice, wholesale anyway |
03:21.58 | PTG1234 | opus_: thats insane |
03:22.02 | harryvv | PBX who |
03:22.04 | PBXtech | ELI |
03:22.16 | harryvv | thats about as cheap as it gets for t1 right? |
03:22.17 | PBXtech | err $550 |
03:22.18 | h3x | type II loop? |
03:22.22 | opus_ | PTG - what I want to know is how 3 different vendors all had the same problem |
03:22.24 | PBXtech | yea |
03:22.34 | harryvv | 600 here for one company thats CDN rates |
03:22.36 | h3x | noley, xspedius sells type II looped vegas T1s here for as low as like $300 |
03:22.38 | PTG1234 | opus_: told you they are all resellers of same place :) |
03:22.40 | PBXtech | 400 base then 150 for the extra rate centers |
03:22.40 | h3x | damn nick complete |
03:22.44 | opus_ | whats a Type II loop PRI? |
03:22.46 | h3x | voice |
03:22.47 | h3x | yes |
03:22.56 | h3x | type II means they have to use the ILEC to deliver your circuit |
03:23.04 | PBXtech | ELI uses Qwest to deliver |
03:23.07 | h3x | type I means you are on-net |
03:23.17 | harryvv | I see |
03:23.18 | *** join/#asterisk scubasteve (~steve@cpe-024-088-248-113.nc.res.rr.com) |
03:23.27 | *** join/#asterisk mezzmor (~mezzmor@adsl-068-209-180-119.sip.mco.bellsouth.net) |
03:23.51 | h3x | my building has xspedius fiber in it, i have a DS3 cross connect to their OC-3 ADM |
03:23.51 | opus_ | h3x that means they are going through some other company? |
03:23.51 | PBXtech | never heard of xspedius |
03:23.52 | harryvv | h3x data in for voip pstn out and that is called type II also? |
03:23.53 | h3x | opus_: yes which they have to do 99% of the time because they dont have fiber everywhere |
03:24.12 | mezzmor | I am having crazy problems with voicemail. Anyone else having weird problems? |
03:24.16 | opus_ | if I purchase a PRI, how do I update the DNIS? |
03:24.25 | PBXtech | from the carrier |
03:24.46 | opus_ | hmmm. nobody will let me update myself, say like, at 2am on sunday? :) |
03:24.48 | harryvv | mezzmor, let me guess it sound like the comedian mail female voice does not finish speaking one word before she speaks the next one? |
03:24.50 | h3x | update ? |
03:25.01 | mezzmor | When MWI works, I cant retrieve messages. When I can retrieve messages, MWI doesnt work. |
03:25.09 | mezzmor | Its crazy. |
03:25.31 | harryvv | mmm |
03:25.35 | harryvv | not seen that one. |
03:25.57 | opus_ | hmmm. it looks like all nufone does is that they have a PRI in michigan, and thats it... does anybody use them? is that the case? |
03:26.15 | PBXtech | probably for inbound DID |
03:26.28 | opus_ | then, they spoof caller id |
03:26.39 | file[laptop] | they don't spoof caller id |
03:26.46 | PBXtech | dont see why they would |
03:26.47 | h3x | opus_: thats probably because they get paid for inbound calls in michigan from being a CLEC, why encourage anything else? :D |
03:27.45 | Carp1 | I dont really know linux, I downloaded a tar.gz file...how to I uncompress and install? |
03:27.53 | PBXtech | HaHaHa |
03:27.58 | opus_ | If I just bought a PRI, sold SIP/IAX accounts, and let everyone spoof caller id .. I think thats their business, am I wrong? |
03:28.00 | h3x | honestly though, jerjer insists on TDM handoffs from everybody, so its really cost prohibitive to have DIDs all over the place |
03:28.02 | harryvv | ohh boy |
03:28.13 | esandeen | carp1: tar xvzf <file>... and then start reading |
03:28.26 | h3x | youd need colo space, private line, etc etc. |
03:28.27 | PBXtech | less INSTALL |
03:28.29 | PBXtech | :) |
03:28.37 | opus_ | oh yeah, INSTALL |
03:28.39 | mishehu | tzanger: you around? |
03:28.58 | h3x | I was going to do all that, and even do all VoIP to TDM transitions for my major IXCs |
03:29.17 | esandeen | hey I have a slightly less newbie question... is there any sort of asterisk howto, or a "what you can do with asterisk" kind of page.... |
03:29.17 | opus_ | well, it seems like an obvious business plan.. |
03:29.20 | h3x | but then ive found out that the telcos out there got some spendy $$$$$$$$$ gateways that do a hell of a lot better job than asterisk |
03:29.28 | h3x | from doing interop tests |
03:29.34 | h3x | its a pain in the ass to set up initially |
03:29.53 | PBXtech | esandeen read the wiki |
03:29.56 | h3x | but i think we're so much better off relaying voip traffic over private line to the carrier's SBC |
03:30.14 | h3x | For instance, I can G.711 fax at 14.4K to Qwest over the public internet |
03:30.24 | h3x | whereas going to my own asterisk gateway over a LAN it dosent work |
03:30.28 | h3x | even with everything set up right |
03:30.33 | opus_ | why is that? |
03:30.37 | esandeen | PBXtech, found it, thanks |
03:30.38 | h3x | well it does work but it renegotiates to 9600 |
03:30.52 | |nix | PTG1234: you're around? |
03:31.08 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
03:31.17 | *** join/#asterisk Dovid (~hirisk@pool-151-198-15-126.mad.east.verizon.net) |
03:31.18 | blitzrage | http://pastebin.com/266765 |
03:31.26 | smurfix | Anybody with zaphfc experience around? |
03:31.27 | h3x | I'm not quite sure, i think maybe the other load from doing many channels screws it up |
03:31.38 | blitzrage | for some reason, I have a phone that won't register. The proxy is sending a 401 back... |
03:31.51 | blitzrage | I have checked the settings in sip.conf, and the phone, and I'm stumped as to where to look next |
03:32.03 | h3x | Furthermore, T.38 fax will eventually work once the carriers turn it on and theres a decent ATA that supports it |
03:32.07 | opus_ | h3x - if asterisk was just doing that call, and only that call, would it wowkr? |
03:32.21 | harryvv | what phone |
03:32.30 | blitzrage | fax will work with spandsp at about 9600 and ulaw |
03:32.33 | h3x | opus_: Come to think of it i did do it in the middle of the night once with no calls |
03:32.41 | h3x | blitzrage: I meant sending a fax from an ATA |
03:32.43 | PTG1234 | nix: yah i am whats up |
03:32.45 | h3x | to an asterisk box |
03:32.45 | PTG1234 | msg me |
03:32.47 | blitzrage | h3x: ah :) |
03:32.49 | h3x | that goes to TDM |
03:32.54 | h3x | with a zaptel device |
03:32.58 | h3x | like a t1 card |
03:33.07 | harryvv | I need to test my spa for fax capability |
03:33.08 | h3x | perhaps i need to adjust some things like the um |
03:33.09 | blitzrage | I have a SIP device that is getting rejected, what can I do to debug why? |
03:33.16 | BoRiS | h3x: I emailed sipura and received a response from them (suprisingly quickly) and they said they expect T.38 support for their spa-2100 in Q2 of 2005. (atleast thats what they said) |
03:33.27 | h3x | harryvv: my spa-2100 is what i was testing fax with ulaw, bt it does not have T.38 yet |
03:33.27 | *** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.hsd1.ca.comcast.net) |
03:33.31 | h3x | BoRiS: yeah i know |
03:33.32 | PTG1234 | blitzrage sip debug |
03:33.34 | opus_ | if ( fax) then use_dedicate_p4_asterisk_just_for_fax() ; :) |
03:33.45 | h3x | If I run the call through asterisk it will probably not support T.38 |
03:33.46 | harryvv | 2100? |
03:33.58 | harryvv | I know the 1k and 2k are almost identical |
03:34.06 | BoRiS | 2100=2k + router |
03:34.10 | harryvv | ahh |
03:34.16 | h3x | yeah it can do nat to the PC port |
03:34.19 | h3x | and QoS |
03:34.29 | h3x | what i dont know is why didnt they put a ethernet switch in it too |
03:35.07 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
03:35.21 | blitzrage | PTG1234: I'm doing that: http://pastebin.com/266765 - I just see 401 Unauthorized and I don't know why its not authorized. I can't see the authorization credentials |
03:35.36 | blitzrage | so I'm stuck as to where to debug next |
03:35.53 | h3x | by the way in my test scenario the 14.4k fax on ulaw was going spa2100 -> same asterisk box -> qwest voip |
03:35.56 | harryvv | btw, I still need somone to test a remote softphone client with my system. all rtp stun and sip ports open but end user could not hear my voice. Could see him accessing vm but he could not hear anything. I suspect his xp firewall may be blocking it by default? |
03:36.10 | h3x | on sip |
03:36.15 | PTG1234 | blitzrage: sounds like password or login is incorrect |
03:36.16 | harryvv | yea |
03:36.23 | h3x | The IAXy is HORRID with fax |
03:36.26 | h3x | it wont even work on a lan |
03:36.35 | blitzrage | PTG1234: yah, thats what I thought... but I've checked the username and password.... very simple... |
03:36.41 | harryvv | h3x thanks for that info :) |
03:36.43 | blitzrage | I even re-entered them in the phone to verify |
03:36.47 | h3x | i think maybe the jitter buffer needs to be adjusted or something |
03:36.53 | BoRiS | I could do faxing with G711 within a lan using my SPA2100 no problems...Going out to the internet...died 95% of the time |
03:37.18 | h3x | BoRiS: I can do 14.4k fax over the sipura to asterisk to voip providers, as long as its not on a cablemodem |
03:37.20 | h3x | DSL works just fine |
03:37.21 | harryvv | boris ever get that resolved? |
03:37.41 | h3x | I bet maybe my problem coulda been fixed with gain settings |
03:37.43 | blitzrage | and the context for the friend exists |
03:37.44 | ubergoober | Can * randomize music on hold? |
03:37.47 | file[laptop] | which it then went to LA to TDM |
03:37.56 | h3x | file[laptop]: what, spandsp? |
03:38.02 | file[laptop] | nope I used a fax machine |
03:38.06 | file[laptop] | I hooked it up to my PAP2-NA |
03:38.06 | h3x | oh ok |
03:38.25 | file[laptop] | I had to fax in some forms for the box that I just got in San Jose ironically |
03:38.26 | h3x | One guy I have coloed at my DC wanted fax, and he spent forever messing with spandsp |
03:38.32 | h3x | hes using asterisk obviously |
03:38.49 | h3x | instead, after trial and many errrors (mostly problems with compatbility with fax machines on the other end) |
03:38.59 | h3x | i talked him into putting a modem in his box and using a pots line off my channelbank :) |
03:39.03 | h3x | hylafax rules |
03:39.07 | |nix | hey PTG1234, i've messaged you |
03:39.20 | opus_ | spandsp has compatability issues? I kind of thought so... what kind of problems did he run into? |
03:39.39 | h3x | certain fax machines it wont negotiate with and of course theres that streaking problem with HP fax machines |
03:40.00 | opus_ | weird. |
03:40.21 | opus_ | isn't there only one guy who works on spandsp? The orig coder himself..? |
03:40.31 | opus_ | perhaps if you ordered him a pizza and sent him the fax he'd do it :) |
03:40.34 | h3x | Yeah its steve, hes a very talented guy |
03:40.35 | h3x | coppice |
03:40.49 | h3x | i think hes probably just busy with other things |
03:40.55 | h3x | fax takes a hell of a lot of work to perfect |
03:41.00 | PBXtech | spandsp pre11 solved a lot of my problems but still a tiny amount of problems. but MUCH better than previous |
03:41.34 | h3x | well im sure its fine if you use it with your own fax machines, but somebody trying to blast out faxes for like say, a food.com clone it dosent work too well |
03:42.02 | PBXtech | blast with spandsp hmmm :) |
03:42.14 | h3x | I wish there was a good t.38 soft modem |
03:42.18 | harryvv | hylafax can work with windows bases fax clients? |
03:42.20 | h3x | besides t38modem in openh323 |
03:42.24 | h3x | yes |
03:42.26 | h3x | theres lots of them |
03:42.29 | harryvv | okay |
03:42.30 | harryvv | :) |
03:42.39 | PBXtech | hylafax rocks |
03:42.52 | harryvv | what modems does it work with |
03:42.56 | h3x | as sad as it is, i am really thinking about setting up a dozen modems on a channelbank and selling hylafax ports |
03:42.56 | h3x | hehe |
03:42.56 | harryvv | i have a number of modems |
03:43.01 | PBXtech | i got a NMS board with 64 DSP's that would be a good blaster box :) |
03:43.12 | h3x | heh |
03:43.31 | h3x | theres a lot of modems it works with even some $4 winmodems |
03:43.38 | harryvv | okay thats cool |
03:43.39 | h3x | im prob gonna use like USRs |
03:43.50 | bkw_ | PBXtech, duh |
03:44.10 | bkw_ | I also luv rxfax |
03:44.22 | h3x | we all love bkw |
03:44.27 | PBXtech | i love rxfax just not 100% solid |
03:44.58 | opus_ | dude, there is a $4k bounty on t.38 |
03:45.17 | h3x | shit, it needs a $40k bounty to put up with that crap |
03:45.19 | h3x | ehehhe |
03:46.45 | opus_ | how does hylafax work |
03:46.47 | opus_ | ? |
03:46.52 | PBXtech | hylafax.org |
03:47.26 | *** join/#asterisk finejava (~abc@218.208.119.98) |
03:47.32 | opus_ | can you receieve a fax with it? |
03:47.36 | finejava | hi guys |
03:47.41 | PBXtech | yup |
03:47.45 | finejava | any1 out there can give me a hand |
03:47.52 | PTG1234 | just use app_rfax |
03:48.10 | Newbie___ | hi, anyone help me on LCR with perl please ? |
03:48.10 | finejava | how can i check how many user which r logon to the asterisk PBX? |
03:48.16 | opus_ | app_rfax is a hylafax module? I thought it was spandsp |
03:48.24 | PTG1234 | no asterisk module |
03:48.29 | PTG1234 | no need to use hylafax really |
03:48.35 | PBXtech | login to what console? |
03:49.14 | Zipper_32 | One quick question, how does one use their home DSL connection with asterisk to place calls? (Isn't this similar to an office using an E1?) |
03:49.21 | finejava | i need a count of how many user is logon...anyway or command which i can use??? |
03:49.32 | PBXtech | [Zipper_32]: pick a VoIP provider |
03:49.51 | PBXtech | login to what? SIP? CLI? |
03:49.59 | finejava | CLI |
03:50.08 | finejava | i mean logon to SIP |
03:50.15 | finejava | but how i can i check from CLI |
03:50.16 | PBXtech | sip show peers |
03:50.24 | finejava | then will show all the user |
03:50.28 | PBXtech | yup |
03:50.32 | opus_ | pbxtech what is a nms board? |
03:50.49 | finejava | i just wan to know how many user is logon with the status 'OK' |
03:50.50 | PBXtech | expensive T1 boards |
03:50.55 | h3x | well, the max tnt 11.0 TAOS supports T.38 on SIP |
03:51.03 | h3x | that is encouraging |
03:51.03 | opus_ | hmmm. |
03:51.13 | PBXtech | [finejava]: show sip peers |
03:51.18 | h3x | at least i could set up a endpoint to test |
03:51.26 | opus_ | pbxtech -- somebody should hack up a t1 board/fpga and put it on opencores.org |
03:51.53 | opus_ | is the circuitry used in a FXS really that complicated? |
03:52.02 | h3x | its called the zapata T400P/E400P :P |
03:52.03 | finejava | show sip peers???? |
03:52.03 | malverian | Hey guys.. if I'm using a W100P card with the analog line plugged into the card and a line from the card to a normal phone. |
03:52.05 | opus_ | Isn't it just an FXO with a ringer/ circuit? |
03:52.09 | finejava | is there such command |
03:52.15 | malverian | Is there any way to make an extension ring through to the phone? |
03:52.18 | Majestik | I've got some codec weirdness, It doesn't seem to behave with the priority I'm trying to set.. |
03:52.19 | PBXtech | finejava type that in the CLI |
03:52.30 | opus_ | I thought the T400P was FXO |
03:52.37 | finejava | yeah...i did...no such command |
03:52.38 | opus_ | nvmind |
03:52.39 | PBXtech | [malverian]: whatcha mean? |
03:52.44 | h3x | T400P is quad T1 |
03:52.45 | finejava | 'show sip peers' |
03:53.05 | PBXtech | man this irc client i wrote forever ago has lots of colors in it. annoying. have to fix it |
03:53.13 | malverian | PBXtech: I want asterisk to pick up the line after 1 ring (done), and if they press for example... "1234" I want it to ring at the line that is plugged into the phone part of the zaptel modem. |
03:53.38 | malverian | I want to basically just ring the telephone line like would happen if asterisk hadn't picked up at all. |
03:53.43 | PBXtech | thats basic |
03:53.48 | malverian | Yay :) |
03:53.49 | PTG1234 | opus: you know a guy was selling them on ebay cheap |
03:53.51 | malverian | I'm glad it's easy. |
03:54.03 | PTG1234 | two people were |
03:54.07 | opus_ | ptg - yeah.. |
03:54.23 | opus_ | wonder what happened with that |
03:55.04 | malverian | I tried making it dial(Phone/phone0) but it doesn't seem to work! |
03:55.09 | PBXtech | malverian you just have the incoming FXO line answer play a background message and have 1234 dial the FXS ZAP circuit |
03:55.30 | PTG1234 | opus: i am emailing them |
03:56.20 | opus_ | PTG -- i think shentech.com is short for shenzhen china.. |
03:56.31 | malverian | PBXtech: I'm having it play a background message, but what is the correct command to have it go to the zap circuit? |
03:56.35 | Zipper_32 | PBXtech: This is my first day with Asterisk, and I completely understood that message. |
03:56.47 | Zipper_32 | I'm so excited... |
03:56.49 | opus_ | haha |
03:56.50 | Zipper_32 | =) |
03:56.58 | opus_ | what was your question? |
03:57.22 | PBXtech | exten => 1234,1,Dial(Zap/2) |
03:57.29 | malverian | Ah... |
03:57.30 | malverian | Thanks man :) |
03:57.37 | PBXtech | assuming FXS is #2 |
03:58.01 | shmaltz | http://bugs.digium.com/bug_view_page.php?bug_id=0002905 |
03:58.03 | shmaltz | in the above bug will revision 5 have everything from the previous ones as well? |
03:58.11 | PBXtech | ive been into asterisk for a year now and this still excites me :) |
03:58.20 | Zipper_32 | If anyone wants to help me though, I'm trying to figure out what kind of equipment I'll need for a 4 line office with 12-15 extensions. |
03:58.27 | opus_ | Hmmm.. I think I'm going to buy one of these linksys pap's and see if they have a JTAG port |
03:58.35 | h3x | Zipper_32: your # of lines is going down .. heh |
03:58.39 | PBXtech | [Zipper_32]: quad FXO card |
03:58.53 | h3x | opus_: haha what do you need jtag for on that |
03:58.57 | Zipper_32 | h3x: The main location is 9 lines. |
03:59.17 | opus_ | h3x - i hear if you upload the XML to it, it reboots, it goes back to vonage |
03:59.19 | Zipper_32 | h3x: this is a new site. The cement is being poured for the building right now. |
03:59.38 | h3x | Oh that thing |
03:59.39 | opus_ | h3x -- file[] just turned me on to this nice piece of hardware |
03:59.55 | opus_ | <PROTECTED> |
04:00.02 | opus_ | i'm just going to /s/vonage/opus_/g |
04:00.04 | opus_ | :) |
04:00.10 | opus_ | put on a new sticker, tada |
04:00.11 | PBXtech | [opus_]: read this http://lestblood.imagodirt.net/archives/83-Asterisk-on-OpenWRT.html#extended |
04:00.20 | h3x | hah |
04:00.25 | h3x | and get sued for millions |
04:00.46 | opus_ | oh yeah, wait its cisco |
04:00.46 | h3x | PBXtech: one person i talked to running asterisk on openwrt said it crashed a lot |
04:01.00 | Carp1 | Why is Postgres better than MySQL? |
04:01.07 | PBXtech | interesting i thought it was stable |
04:01.17 | h3x | uhoh, here comes the religious RDBMS floodgate from hell |
04:01.17 | opus_ | PBXtech -- I am talking about the linksys with the FXS port : http://www.voip-info.org/wiki-Linksys?page=Linksys&comments_threshold=0&comments_offset=0&comments_sort_mode=commentDate_desc&comments_maxComments=10&comments_parentId=861#threadId937 |
04:01.44 | terrapen | why the hell would you want to run Asterisk under OpenWRT |
04:01.49 | terrapen | OpenWRT runs on junk hardware |
04:01.52 | opus_ | h3x - wait, you can reverse engineer hardware to be compatible with it.. should be legal |
04:01.58 | h3x | terrapen: I would but the problem is theres not enough flash for voicemail etc |
04:02.12 | terrapen | there are many more problems than that |
04:02.19 | h3x | its a 300mhz processor |
04:02.20 | terrapen | it's woefully underpowered |
04:02.26 | h3x | its actually overpowered |
04:02.39 | h3x | it just dosent have enough ram and flash |
04:02.39 | terrapen | not for anything useful |
04:02.53 | opus_ | h3x - for a VPN gateway it is is underpowered.. 4mb/s |
04:02.54 | h3x | well its not like it has a pci slot to stick a t1 card in or anything |
04:03.14 | opus_ | h3x - the older ones have minipci!!! |
04:03.16 | h3x | hah, whos got more than that on their cable or dsl anyway |
04:03.20 | h3x | they do? |
04:03.25 | opus_ | yup |
04:03.42 | opus_ | the real old ones have pcmcia |
04:03.46 | h3x | maybe atacomm should make their dsp iVolution card in a minipci format for that thing |
04:03.47 | h3x | heheehhe |
04:04.10 | opus_ | i think you could build a minipci to pci adapter with something from opencores.org |
04:04.26 | h3x | um i think its just a different socket, same bus |
04:04.40 | malverian | PBXtech: Hmm.. it appears that's not the correct channel, what's the best way to figure out which channel it is? |
04:04.43 | opus_ | however, its out of my r&d budget:) and time. would be cool thou |
04:04.47 | PBXtech | i just want to see linux put on my PSP |
04:04.52 | finejava | as i understand...'sip show peers' will list all the user regardless whether they r logon or not |
04:04.57 | PBXtech | in the CLI type zap show channels |
04:04.58 | h3x | actually |
04:05.01 | malverian | PBXtech: Also, I can't find anywhere in the configuration that defines how many rings to wait before picking up. |
04:05.11 | PBXtech | you should have already configured the channels |
04:05.28 | finejava | but all i need to to list the user with the status 'OK' |
04:05.35 | finejava | anyway we can do that |
04:05.38 | PTG1234 | i couldn't deface my psp with linux |
04:05.40 | malverian | PBXtech: I only have "pseudo" and "1" |
04:06.10 | opus_ | once you load linux on it I'll deface it for you |
04:06.18 | PBXtech | [finejava]: did sip show peers not work? |
04:06.46 | finejava | sip show peers works...but it shows every single peer regardless it's offline or online... |
04:07.14 | PBXtech | sip show registry |
04:07.34 | finejava | but i just need to view the user who is logon...with the fiels status 'OK' |
04:07.36 | terrapen | i would like a box with a >= 667MHz P3-comparable CPU, 512M RAM, totally fanless |
04:07.43 | terrapen | that takes a 12VDC power input |
04:07.49 | Qwell | terrapen: picky |
04:07.53 | terrapen | and comes in a very small box |
04:08.06 | opus_ | terrapen - laptop on ebay with broken screen... $30-$50 |
04:08.08 | PBXtech | malverian is that 1 channel what your trying to send the call to? guess you dont have a FXO and a FXS port eh |
04:08.22 | harryvv | opus what laptop |
04:08.26 | Qwell | opus_: Thats not a bad idea actually |
04:08.29 | terrapen | perhaps |
04:08.33 | malverian | It's a digium wildcard w100p |
04:08.34 | terrapen | no, not a bad idea |
04:08.44 | terrapen | except that most laptops were not made to run 24/7/365 |
04:08.48 | terrapen | heat can be an issue |
04:08.50 | terrapen | and they have fans |
04:08.51 | opus_ | i also offer professional servies:) |
04:08.58 | finejava | sip show registry is to show the peers which uses the sytanx register => |
04:09.30 | PBXtech | if you dont have them register then sip show peers is all that i can think of |
04:09.31 | harryvv | I need a laptop and if its a dell I can repair it. |
04:09.42 | finejava | i just wan a list of user which r successgully logon...and their status fiels is "OK" |
04:09.45 | malverian | PBXtech: I used the instructions from digium website on setting up the zaptel configuration, and it only told me to use one channel (1) with fxsks |
04:10.06 | finejava | is there any other like using manager API or something |
04:10.13 | PBXtech | malverian if you only have 1 card for an incoming line where are you trying to send the call? |
04:10.16 | PBXtech | ip phone? |
04:10.45 | PBXtech | [finejava]: write your own app and use the manager API sure |
04:11.34 | PBXtech | if you just looking for something like an operator console deal, that stuff is out and about |
04:12.24 | malverian | PBXtech: It's a single card, but there are two ports (one from wall, one to normal analog telephone). I was just trying to figure out if it was possible to pass the call on to the phone after asterisk has already answered it. |
04:12.46 | PBXtech | might have 2 physical port but its not for that purpose |
04:12.49 | finejava | but i can't seems to find anything in the manager API which can give me the counter |
04:12.55 | malverian | If I don't have asterisk answer the line, it rings on the normal phone by default. |
04:13.10 | malverian | PBXtech: Ah, so this isn't a possibility then? That's fine, I just figured I'd ask in case. |
04:13.41 | PBXtech | you can have it answer after 2 rings of what not, but if you want it send to a phone you have to have a different card or IP phone to do that |
04:14.32 | PBXtech | finejava manager API is just a CTI style feed you would have to build your own counter. |
04:15.26 | PBXtech | malverian killer answering machine? |
04:15.27 | PBXtech | heh |
04:15.39 | WilliamK | blaaaaaaaaaaaah |
04:15.40 | malverian | PBXtech: Gotcha, and how do I set the number of rings before pickup? |
04:15.54 | PBXtech | do a Wait(3) then Answer |
04:16.01 | malverian | Ah. |
04:16.15 | malverian | I notice it won't pick up at any less than 2 rings. |
04:16.31 | PBXtech | waiting for CallerID info eh |
04:16.51 | robl^ | caller ID comes in between first and second ring |
04:17.05 | malverian | PBXtech: Will it still do that if I disable caller id? |
04:17.23 | shido6 | back |
04:17.27 | shido6 | whats up |
04:17.31 | malverian | PBXtech: I'm basically just messing around on my home analog line because I'm going to be creating a new phone server at work.. it's sort of my playground. |
04:17.35 | PBXtech | should be able to answer if upon ring detection |
04:17.49 | PBXtech | good place to start :) |
04:18.18 | PBXtech | hi shido6 |
04:18.35 | PBXtech | you still at nufone? |
04:19.56 | PBXtech | well im out tata |
04:20.02 | finejava | thx PBXtexh |
04:20.09 | finejava | appreciate ur help |
04:20.37 | *** join/#asterisk Rick_Hunter (~rhunter@03-111.008.popsite.net) |
04:21.49 | *** join/#asterisk ericw (~eric@pcp04966776pcs.benslm01.pa.comcast.net) |
04:21.51 | ericw | hello |
04:22.59 | ericw | trouble with connecting to sipphone - I initially received response 480, "you are not who you say you are".. although I'm now getting circuit-busy errors instead |
04:24.09 | mishehu | ok. lovely. I don't think we have nearly enough information to even attempt to assist you. |
04:24.41 | greg_work | Can anyone see why this: exten => s,1,Noop(${MACRO_CONTEXT}) exten => s,2,GotoIf($[${MACRO_CONTEXT}=macro-rg-group]?5:3) does this: -- Executing NoOp("SIP/219-8925", "macro-rg-group") in new stack -- Executing GotoIf("SIP/219-8925", "macro-rg-group=macro-rg-group?5:3") in new stack -- Goto (macro-dial,s,3) ? |
04:24.50 | ericw | mishehu, well, what info is needed? I have my settings, but I'd prefer to avoid flooding the channel :) |
04:24.51 | Carp1 | I havnt updated asterisk in like 6 months |
04:24.58 | Carp1 | and i'm on 56k |
04:25.07 | Carp1 | and its been updating for over an hour :-\ |
04:26.27 | ericw | sip.conf: type=peer, secret= is set appropriately, fromdomain=proxy01.sipphone.com, callerid= my number, host=proxy01.sipphone.com, quality=no, nat=yes, dtmfmode=inband (also tried rfc2833). reinvite=no, canreinvite=no, context=default, disallow=all, allow=ilbc, allow=gsm, allow=ulaw, allow=alaw, insecure=very |
04:26.52 | *** join/#asterisk jdg (~jdg@CA03F89B.adsl.mana.pf) |
04:27.06 | ericw | externip is set to my internet IP. localnet is set to 10.0.0.0/255.0.0.0 |
04:28.09 | ericw | the machine's IP is 10.0.0.16 and is behind the firewall 10.0.0.1 and is configured on the DMZ port. The WAN IP of that (NAT) firewall is 10.1.0.2 and has an upstream router of 10.1.0.1 and is connected on the DMZ port of that router. |
04:28.19 | ericw | the WAN port of that router is my internet IP. |
04:28.42 | *** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz) |
04:31.02 | mishehu | jbot: pastebin |
04:31.03 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
04:31.27 | mishehu | ericw: that's a good place to dump the stuff you'd like to dump. I use it fairly often. |
04:31.57 | mishehu | quality=no ??? is that a direct copy & paste? |
04:32.12 | *** join/#asterisk ericw (~eric@pcp04966776pcs.benslm01.pa.comcast.net) |
04:32.19 | ericw | got disconnected |
04:32.41 | mishehu | ericw: generally speaking, http://pastebin.ca is a good place to put stuff like you wanted to |
04:32.46 | mishehu | quality=no ??? is that a direct copy & paste? |
04:32.59 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
04:33.22 | ericw | mishehu, I was copying and pasting from examples I found online regarding sipphone |
04:33.35 | ericw | everyone I saw had it |
04:34.30 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
04:35.04 | malverian | Are there any gnome based sip phones? |
04:35.13 | malverian | (gtk based, rather) |
04:35.19 | robl^ | gnophone? |
04:35.27 | malverian | I'd install kphone, but I dread installing qt and the kde libs. |
04:35.28 | ericw | malcolmd, gaimphone, if you can get it working. |
04:35.37 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
04:35.37 | malverian | That works :) |
04:35.56 | mishehu | ericw: you sure they didn't mean "qualify=yes" ? |
04:36.00 | mishehu | err |
04:36.06 | mishehu | "qualify=no" I mean |
04:36.31 | malverian | Ooh.. linphone |
04:36.36 | carlosh | hello every one. Could someone tell how many calls would I be able to connect to a meetme dedicated server? many thanks.. |
04:36.56 | ericw | mishehu, know what..... |
04:37.00 | mishehu | carlosh: however many you had bandwidth and cpu to handle. |
04:37.03 | ericw | *I* have quality=no |
04:37.09 | ericw | *They* have qualify=no |
04:37.15 | shmaltz | how do I patch app dial with this patch: |
04:37.17 | shmaltz | http://bugs.digium.com/bug_view_page.php?bug_id=0002905 |
04:37.18 | shmaltz | How do I apply the .diff file? |
04:37.23 | mishehu | ericw: well, I guess you didn't want quality ;-) |
04:37.41 | ericw | mishehu, is that even a valid directive? :P |
04:37.42 | carlosh | wishehu : say, 120 simultaneous calls on a beefy server ? |
04:37.51 | mishehu | shmaltz: try patch -p0 --dry-run < diff.file |
04:38.01 | mishehu | that will do everything but actually patch it |
04:38.10 | *** join/#asterisk alphaque (~Alphaque@218.111.60.60) |
04:38.14 | ericw | mishehu, changing that didn't fix it, though |
04:38.15 | mishehu | you might need to use 1 or higher instead of 0 to patch. |
04:38.50 | mishehu | ericw: you probably have a different problem. if you could paste the relevant code to pastebin.ca, it'd make it a bit easier to read. |
04:40.45 | shmaltz | mishehu, gives me a msg that 7 out of 7 hunks FAILED |
04:41.41 | shmaltz | it was telling me can't find file blah blah blah, so I gave it the path |
04:42.00 | shmaltz | but it still failed |
04:42.15 | Qwell | I hate seeing the word "hunks" when I patch something...they really should make it "chunks" |
04:42.29 | ericw | http://pastebin.ca/8758 |
04:42.30 | shmaltz | Qwell any idea? |
04:42.36 | Qwell | nope |
04:42.37 | Corydon76-home | Qwell: blowing chunks? |
04:42.37 | malverian | Is there some kind of asterisk command reference? |
04:42.43 | Qwell | Corydon76-home: far better |
04:42.53 | Qwell | "7 out of 7 of your files blew chunks |
04:42.54 | Qwell | " |
04:42.55 | shmaltz | malverian, try the wiki |
04:43.01 | Corydon76-home | Qwell: I like blowing hunks better. |
04:43.11 | robl^ | OH MY! |
04:43.12 | shmaltz | Qwell, so what should I do? |
04:43.13 | Qwell | Corydon76-home: tmi |
04:43.31 | shmaltz | is it something wrong with the diif file? or the way I'm applying it? |
04:43.38 | shmaltz | ~docs |
04:43.39 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
04:43.40 | mishehu | shmaltz: man patch |
04:43.42 | Corydon76-home | robl^: I'm good at it, too |
04:43.44 | Qwell | shmaltz: The way you're applying it probably |
04:44.02 | shmaltz | mishehu, didn't help for me I tried |
04:44.05 | shmaltz | will try again |
04:44.25 | mishehu | shmaltz: does it prompt for files or does it run thru seemingly automatically? |
04:44.26 | iceyp | yone here use voipjet? |
04:44.35 | iceyp | anyone* |
04:44.58 | terrapen | is it me or is Sourceforge's download server always...down? |
04:45.06 | shmaltz | mishehu, iit prompts me for files |
04:45.08 | JerJer[interop] | terrapen: there are a ton of them |
04:45.09 | ericw | sip uses udp, right? |
04:45.18 | terrapen | no, i mean, the selector server |
04:45.24 | terrapen | prdownloads.sourceforge.net |
04:45.25 | shmaltz | am I suppose to be in /usr/src/asterisk or in /usr/src/asterisk/apps? |
04:45.30 | terrapen | http://prdownloads.sourceforge.net/vnc-tight/tightvnc-1.2.9_unixsrc.tar.gz?download |
04:45.32 | shmaltz | and where should teh .diff file be? |
04:45.35 | terrapen | that's not loading for me |
04:45.53 | shmaltz | and how do I run the patch command? |
04:46.01 | Sedorox | http://cogent.dl.sourceforge.net/sourceforge/vnc-tight/tightvnc-1.2.9_unixsrc.tar.gz |
04:46.13 | terrapen | thx |
04:46.16 | Sedorox | yup |
04:46.25 | ericw | nevermind.. I'm forwarding both |
04:46.26 | terrapen | is prdownloads broken? |
04:46.30 | terrapen | or is it just me? |
04:46.36 | Qwell | terrapen: just you I'd say |
04:46.38 | bkw_ | ok who wishes to fund the res_sms.c ? |
04:46.39 | ericw | so I should be good there |
04:46.50 | Qwell | bkw_: I've got $2.50 |
04:46.54 | bkw_ | hehe |
04:47.01 | Sedorox | I've got $0.04 |
04:47.04 | Qwell | sorry, misplaced decimal |
04:47.07 | bkw_ | well i'm actually gonna get my hands on a GSM/GPRS modem soon |
04:47.08 | Qwell | $.250 |
04:47.11 | Sedorox | terrapen: worked for me |
04:47.16 | bkw_ | and whip out res_sms |
04:47.28 | ericw | mishehu, any idea? |
04:47.29 | terrapen | how WIERD |
04:47.30 | Sedorox | hmmm |
04:47.35 | terrapen | it works on my win32 machine |
04:47.39 | bkw_ | and write some ast_sms* api's to queue sms's in and out of the machine |
04:47.45 | terrapen | but Firefox on my MacOSX box fails to load it |
04:48.03 | terrapen | just hangs on "Waiting for prdownloads.sourceforge.net..." |
04:48.04 | Qwell | terrapen: Is it on a roundrobin DNS or something maybe? |
04:48.11 | mishehu | bkw_: with iax2, when the user context has trunk=yes defined, any idea why delayreject would cause it to return INVAL ? |
04:48.15 | terrapen | dunno lemme check that |
04:48.31 | Qwell | terrapen: Just see if it resolves to something different on each machine |
04:48.43 | terrapen | nope |
04:48.48 | mishehu | ericw: It's not exactly how I'd do it (I'd not use the macro) but I don't see anything that sticks out at me as a problem immediately. what ver are you using? |
04:48.53 | terrapen | only 1 A record |
04:49.03 | Qwell | terrapen: weird |
04:49.08 | ericw | mishehu, cvs from a week, maybe 2 weeks ago. |
04:49.21 | Carp1 | Why is Postgres better than MySQL? |
04:49.26 | mishehu | ericw: cvs stable or cvs head? |
04:49.31 | Qwell | Carp1: Who says it is? |
04:49.32 | ericw | I had the problem with the cvs from 2 months ago too.. been struggling with this for a while |
04:49.37 | mishehu | Carp1: because it isn't. |
04:49.50 | iceyp | ne here use voipjet and find problems with them? my calls die after 22 seconds |
04:50.01 | ericw | *default release=cvs |
04:50.08 | ericw | ^-- asterisk.sup |
04:50.10 | Carp1 | It seems to me as if everyone writing apps for Asterisk are either switching from MySQL to it or starting with it completely |
04:50.21 | Sedorox | What would I search for if I wanted to run my own "cell site" like.. if I had a few gsm cell phones, with sims.. and I could register the sims with it..? |
04:50.25 | mishehu | iceyp: I've not made many calls this past week, but I have no problems. |
04:50.35 | ericw | I suppose that is head? |
04:50.44 | terrapen | UGH |
04:50.47 | terrapen | tightvnc uses imake |
04:50.51 | mishehu | ericw: show version will tell you |
04:51.06 | iceyp | mishehu i;ve made 5 test calls and using all servers, westcoast is better for me |
04:51.08 | ericw | Asterisk built by root@warhol on a i686 running Linux |
04:51.09 | ericw | .... |
04:51.19 | iceyp | but after 22 seconds of the call its dead |
04:51.27 | iceyp | all their email addresses bounce too |
04:51.43 | mishehu | iceyp: fastsupport@voipjet.com bounces? |
04:51.51 | iceyp | yep |
04:51.54 | iceyp | same with carriers |
04:52.15 | *** join/#asterisk Inv_arp (junya@adsl-3-237-168.mia.bellsouth.net) |
04:52.29 | ericw | mishehu, btw.. I added a paste of the error I'm getting now in there. I was getting another before. |
04:52.31 | iceyp | anyone here using ser @ asterisk? |
04:52.41 | iceyp | i cant get ser to accept calls from asterisk |
04:53.47 | Sedorox | hmmm |
04:53.49 | Sedorox | guess no |
04:53.50 | Sedorox | not* |
04:54.01 | mishehu | iceyp: I'd have to say the problem is either with westcoast, as I use eastcoast, or it's with you. I just tested, I called more than 22 seconds. |
04:54.25 | iceyp | hmmm, i tried primary east coast server and westcoast server |
04:54.33 | iceyp | my calls are fine with nufone ;/ |
04:54.40 | iceyp | maybe its calls to new zealand |
04:54.48 | mishehu | iceyp: where are you calling to? |
04:54.51 | iceyp | NZ |
04:54.56 | iceyp | my home phone |
04:55.17 | mishehu | iceyp: and do you have the 22 second problem if you call another country? |
04:55.24 | iceyp | havent tried |
04:55.32 | iceyp | u got a number i can dial |
04:55.33 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
04:55.40 | mishehu | ericw: sec |
04:55.59 | iceyp | u dont have to talk just leave it off the hook when i dial u |
04:56.11 | Qwell | iceyp: Want me to msg you a number? |
04:56.40 | greg_work | features.conf mods require a complete restart? |
04:57.21 | mishehu | ericw: I don't see the update. you sure it posted to the same pastebin? |
04:57.29 | ericw | http://pastebin.ca/8759 |
04:57.37 | Qwell | mishehu: it updates the number I believe |
04:57.40 | ericw | "--- error being given (currently) ---" |
04:58.05 | iceyp | i seem to have dialed u fine |
04:58.09 | iceyp | argh |
04:58.37 | shmaltz | how do I patch app dial with this patch: |
04:58.39 | shmaltz | http://bugs.digium.com/bug_view_page.php?bug_id=0002905 |
04:58.40 | shmaltz | How do I apply the .diff file? |
04:58.52 | shmaltz | should I apply just the last revison? |
04:58.59 | shmaltz | or all previous ones as well? |
05:01.02 | wildcard0 | ok...am i just missing it or is there go 'and.gsm' sound file from allison smith? |
05:01.23 | Qwell | /usr/cvsroot/asterisk-sounds/sounds/and.gsm |
05:01.36 | wildcard0 | hmm. why don't i have that? |
05:01.39 | Qwell | There's also vm-and.gsm |
05:03.11 | ericw | mishehu, with a different number (a long distance number, which I would need to pay for), I'm getting - Got SIP response 400 "Bad Request" back from 198.65.166.131 |
05:03.33 | mishehu | ericw: have you tried tcpdumping? |
05:03.37 | mishehu | might be useful. |
05:06.32 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || Who wants SMS in asterisk? TRUE SMS? /msg bkw_ |
05:09.30 | mishehu | alright, time for sleep. |
05:09.42 | ericw | mishehu, ok. night. thank you |
05:10.02 | ericw | btw, I got that tcpdump |
05:10.12 | ericw | http://eric.bwbohh.net/sipphone.log |
05:10.33 | mishehu | ericw: sorry, wife is getting mad |
05:10.34 | mishehu | heh |
05:10.42 | ericw | mishehu, heh.. mine too! |
05:10.55 | ericw | night |
05:13.46 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || res_sms anyone? |
05:13.47 | *** join/#asterisk veesmooth (~veesmooth@bgp505102bgs.verona01.nj.comcast.net) |
05:14.05 | veesmooth | hey guys |
05:14.12 | *** part/#asterisk esandeen (~sandeen@sandeen.net) |
05:14.17 | veesmooth | anyone here |
05:15.17 | `Sauron | Nobody here, move along... |
05:15.37 | veesmooth | he he, thanks for tryin to chase me away |
05:15.47 | veesmooth | got a question if anyone willing to answer |
05:16.23 | shido6 | shoot |
05:17.05 | marlowe | Don't ask to ask.. Just ask. |
05:17.38 | Sedorox | anyone know what keyword/companies I should look for if I want to register sim cards and have my own "Cell site" ? |
05:17.50 | shido6 | ATEUS |
05:17.51 | marlowe | Sedorox: Why do you want to? |
05:17.57 | marlowe | Out of curiosity.. :) |
05:18.11 | *** join/#asterisk rowter (~Drake@201.133.210.80) |
05:19.32 | Sedorox | well I have some old cell phones.. and was kinda hoping to do my own little setup with * to have my own personal cell service going, basicly just to mess around with |
05:20.34 | marlowe | thats cool |
05:20.40 | marlowe | thats beyond me |
05:20.51 | Sedorox | hehe |
05:20.55 | Qwell | Thats an expensive "messing around" :p |
05:21.20 | *** join/#asterisk sunil (~sunil@202.54.37.181) |
05:21.21 | marlowe | Ill buy service... Whats the reception area? Around your house? |
05:21.53 | Sedorox | lol |
05:21.54 | Sedorox | probably |
05:22.01 | Sedorox | since It would run into fcc problems |
05:22.13 | Sedorox | and Qwell.. yes.. most of my 'messing arounds' seem to be |
05:22.35 | Sedorox | shido6: thanks... have you used it before? |
05:23.13 | shido6 | yes |
05:23.17 | shido6 | in aussie land |
05:23.20 | shido6 | and an E1 |
05:23.27 | shido6 | buts its gsm |
05:23.32 | Sedorox | yea |
05:23.34 | Sedorox | cool |
05:23.39 | Sedorox | work well? |
05:24.02 | shido6 | absolutely |
05:24.10 | Sedorox | neat |
05:24.13 | shido6 | what are you looking to do? |
05:24.43 | *** join/#asterisk yxa (~void@203.118.40.42) |
05:24.54 | nDuff | What's the Right Way to dump a bunch of data into the Asterisk DB? (I've got a script w/ "database put" statements, but piping it to "asterisk -r" doesn't seem to Do The Right Thing). |
05:26.05 | Sedorox | just use some old cell phone and make it where I can dial into asterisk and from asterisk... kinda a longer-range cordless phone, just with stuff I have (minus the gateway thingy) |
05:26.42 | nDuff | I suppose "asterisk -rx <command>" would work, but that would mean a separate invocation for every put, right? |
05:26.52 | *** join/#asterisk Fddayan (~fddayan@c-67-191-7-6.hsd1.fl.comcast.net) |
05:27.08 | greg_work | nDuff: use the asterisk manager api |
05:27.20 | Fddayan | sombody knows how to emulate a call with a WAV file in asterisk ? |
05:28.12 | wildcard0 | Fddayan, you want to make a call and then play a wav file to it? |
05:28.52 | marlowe | Keep in mind the asterisk datbase is just version 1 of the berkley db |
05:29.05 | Fddayan | yes ! |
05:29.13 | Silik0n | *yawn* |
05:29.18 | Fddayan | there is any way ? |
05:29.20 | marlowe | Tats my hint |
05:29.22 | marlowe | Fddayan: Of course |
05:29.22 | wildcard0 | Fddayan, create a call file and have it use an extension that has a Playback command in it |
05:29.44 | marlowe | umm |
05:30.29 | Sedorox | hmmm |
05:30.44 | marlowe | Look up the cmd Dial |
05:30.51 | marlowe | It's right there |
05:31.10 | marlowe | A(x): Play an announcement (x.gsm) to the called party. |
05:31.15 | marlowe | Can't be much easier. |
05:31.31 | Sedorox | Ok... heading in |
05:31.35 | Sedorox | Night |
05:31.41 | sunil | hello, can any one help me in configuring asterisk on mfcr2 signalling |
05:31.56 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:36.46 | *** join/#asterisk wiseguy_ (~chivilis@vadyba.vtu.lt) |
05:36.48 | wiseguy_ | helow |
05:37.08 | wiseguy_ | anybody with cisco routers? |
05:37.09 | wiseguy_ | :) |
05:37.16 | wildcard0 | im with 2 of them |
05:37.19 | wildcard0 | we're close |
05:37.30 | *** part/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
05:37.41 | wiseguy_ | wildcard0 |
05:37.49 | wiseguy_ | can show me part of sip.conf |
05:37.57 | wiseguy_ | where cisco peer is described |
05:38.14 | wiseguy_ | ? |
05:38.24 | wildcard0 | oh i dun have asterisk talking to them directly. they're IP routers |
05:38.45 | wiseguy_ | :/ |
05:39.43 | DEEZED | how do you get asterisk to say a number? does it need a gsm file, or does it have it built it to say numbers? |
05:39.55 | wildcard0 | SayNumber |
05:40.11 | wildcard0 | it uses the numbers in /var/lib/asterisk/sounds/digits |
05:40.31 | robl^ | DEEZED: yes. SayNumber will find the sound files it needs and say the the number for you |
05:40.59 | tessier | Can anyone tell me what Vovida does that asterisk doesn't? |
05:41.22 | h3x | tessier: 0 results returned |
05:41.24 | h3x | actually |
05:41.28 | DEEZED | thanks... so SayNumber(1) |
05:41.32 | DEEZED | would say 1> |
05:41.33 | DEEZED | ? |
05:41.40 | robl^ | yes |
05:41.44 | DEEZED | thanks |
05:41.46 | h3x | Vovida what application? |
05:41.49 | robl^ | and 201 would say "two hundred one" |
05:41.59 | DEEZED | sweetness |
05:42.05 | Qwell | SayNumber(304) says "Hello Bob, how is your day?" |
05:42.07 | Qwell | etc |
05:42.13 | robl^ | SayDigits(201) would say "two zero one" |
05:43.51 | h3x | tessier: what are you talking about, VOCAL? |
05:43.55 | wiseguy_ | anyone with cisco 1760 + asterisk? |
05:44.32 | veesmooth | anyone good with programming in astersisk? |
05:44.42 | nDuff | veesmooth, I'm sure someone is. |
05:44.43 | tessier | h3x: Yes, vocal. |
05:44.47 | veesmooth | might need some assitant |
05:44.49 | veesmooth | lol |
05:44.51 | *** join/#asterisk TekGecko (~univ@dhcp-96878f9f.rescomp.arizona.edu) |
05:44.57 | h3x | well i bet it does a better job of h.323 -> sip than asterisk :P |
05:44.59 | veesmooth | with what to do with asterisk as a masters project |
05:45.17 | veesmooth | trying to get some ideas |
05:45.41 | wildcard0 | what area are you studying? |
05:45.46 | veesmooth | computer science |
05:45.49 | wiseguy_ | when i write sip show peers, status is unmonitored |
05:45.56 | wiseguy_ | how to change into monitored? |
05:45.56 | wiseguy_ | :) |
05:45.56 | tessier | Lots of people say they are using ser, asterisk, and vocal. |
05:46.10 | wildcard0 | veesmooth, yes i gathered that...i ment what concentration |
05:46.12 | tessier | I don't see what vocal does that asterisk doesn't. |
05:46.22 | nDuff | veesmooth, maybe you could try for a better echo cancellation algorithm? That's a good computer-sciency subject. |
05:46.35 | wildcard0 | wiseguy_, add qualify=yes |
05:46.47 | wildcard0 | nDuff, that'd be REALLY useful |
05:46.56 | veesmooth | well im not really concentrationg on anything |
05:47.03 | veesmooth | more general with software and networking |
05:47.07 | h3x | vocal and ser have a provisioning gui |
05:47.25 | wildcard0 | maybe work on the more general jitterbuffer? |
05:47.56 | veesmooth | echo cancellation algorithm huh |
05:47.57 | veesmooth | hmmmm |
05:48.04 | veesmooth | a better one |
05:48.19 | veesmooth | is that something you can find more info on the net or anything? |
05:48.55 | wildcard0 | http://cnx.rice.edu/content/m11909/latest/ |
05:49.05 | wildcard0 | http://www.embeddedstar.com/articles/2003/7/article20030720-1.html |
05:49.14 | wildcard0 | http://lcavwww.epfl.ch/~prandoni/dsp/echo/echo.html |
05:49.22 | wildcard0 | and generally |
05:49.27 | veesmooth | all for me wild card? |
05:49.29 | wildcard0 | http://www.google.ca/search?q=echo+cancellation |
05:49.31 | wildcard0 | yes |
05:50.59 | veesmooth | hold on trying to copy these links now |
05:51.32 | veesmooth | but that does sound good for a project right |
05:51.46 | wildcard0 | i think it sounds good, but i'd run it by my advisor :) |
05:51.54 | Zipper_32 | How does one dial into a phone (from a PSTN) which is behind asterisk via an E1 connection? Where is the actual number being routed? |
05:52.04 | wildcard0 | you might choose 2 or 3 and run them all by your advisor |
05:52.32 | veesmooth | so there are more then one then huh |
05:52.40 | wildcard0 | Zipper_32, so you want to go E1 --> asterisk --> ata --> phone? |
05:53.00 | wildcard0 | veesmooth, i meant 2-3 possible topics. but it's your thesis |
05:53.16 | veesmooth | yeah thats what i want to do , im a def put that as one of my topics |
05:53.49 | wildcard0 | veesmooth, it'd be cool to do some research on extending the max latency for a voip call also |
05:53.50 | Zipper_32 | wildcard0: [trying to figure out what 'ata' means... wanna give me a hand? |
05:54.24 | Zipper_32 | wildcard0: I'm just wondering how someone from behind the PSTN can call me if I'm behind an E1 line running asterisk with an IP phone. |
05:54.24 | wildcard0 | Zipper_32, analog telephone adapter. like http://www.sipura.com/products/spa2000.htm for example |
05:55.21 | wildcard0 | they dial the number to the E1 you have plugged into asterisk. in extensions.conf you have something like 'exten => 9995551212,1,Dial(SIP/1)' |
05:55.44 | wildcard0 | where 9995551212 is your DID and SIP/1 is the path to your IP phone |
05:56.20 | Zipper_32 | So the E1 has a number... gotcha. [I didn't know that part, I assumed it was just a broadband channel] |
05:56.35 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
05:56.48 | wildcard0 | oh it CAN be a broadband channel. or it can be a channelized voice connection |
05:56.52 | wildcard0 | it depends how you have it set up |
05:56.57 | *** join/#asterisk SuperMMan (~graphic@edtntnt2-port-216.dial.telus.net) |
05:57.32 | Zipper_32 | Alright, say I have 4 lines, I'm assuming that I want a channelized voice connection to a VOIP provider, right? |
05:57.46 | wildcard0 | yes |
05:57.48 | Zipper_32 | By the way, forgive my ignorance, it's my first day. |
05:57.57 | wildcard0 | lemme find a url |
05:58.07 | Zipper_32 | OoOo, goodie, Thanks. |
05:58.13 | Zipper_32 | I've been reading ever since morning. |
05:58.45 | veesmooth | so to do these research u think it will take a lot to figure these things out |
05:58.51 | veesmooth | especially since im new to astrisk |
05:59.32 | wildcard0 | Zipper_32, this is a good starting place. a t1 is roughly the same as an E1. just an E1 has more bandwidth/channels |
05:59.33 | wildcard0 | http://en.wikipedia.org/wiki/DS1 |
06:01.11 | Zipper_32 | Alright, now I read earlier that you can run 4 simultaneous lines off of a regular DSL connection, so is this really all I would need? |
06:01.13 | wildcard0 | Zipper_32, if you're really starting from scratch to be a voice providor, consider hiring a consultant |
06:01.47 | Zipper_32 | I'm starting from scratch to find out what to setup for a new office location. And if possible, set it up myself. |
06:01.53 | wildcard0 | ah |
06:02.01 | wildcard0 | oh you -need- a voice provider |
06:02.04 | wildcard0 | where are you located? |
06:02.11 | Zipper_32 | Coquitlam BC |
06:02.18 | Zipper_32 | 20min east of Vancouver |
06:02.21 | wildcard0 | oh. hmm. that would be me actually |
06:02.22 | wildcard0 | hehe |
06:02.25 | Zipper_32 | lol! |
06:02.27 | wildcard0 | <-- downtown van |
06:02.36 | Zipper_32 | I'm in White Rock right now |
06:02.44 | wildcard0 | "surrey" :) |
06:02.49 | Zipper_32 | Shh! |
06:02.53 | wildcard0 | hehe |
06:02.54 | Zipper_32 | Don't say that too loud... |
06:03.10 | Zipper_32 | What kind of work do you do? |
06:03.32 | wildcard0 | termination services for small businesses, hotels and commercial highrises |
06:03.39 | wildcard0 | http://www.mxunetworks.com |
06:03.41 | Zipper_32 | That's me. |
06:03.50 | Zipper_32 | me = small business |
06:10.31 | *** join/#asterisk jwitte (~jwitte_su@port-212-202-101-206.static.qsc.de) |
06:12.52 | veesmooth | this asterisk stuff is pretty interesting though |
06:13.01 | veesmooth | just installed it and getting use to it on my linux computer |
06:15.45 | *** join/#asterisk l00p (~l00p@c-67-171-201-105.hsd1.or.comcast.net) |
06:16.26 | l00p | Anyone awake? |
06:16.49 | veesmooth | i am |
06:16.59 | shepherd | NO! |
06:17.01 | shepherd | :D |
06:17.12 | shepherd | dangit |
06:17.17 | shepherd | Day light savings time |
06:17.18 | shepherd | ni ni |
06:17.23 | veesmooth | yup |
06:17.23 | l00p | I've been doing a lot of reading on asterisk and have a question. |
06:17.26 | veesmooth | make it closet to work |
06:17.32 | l00p | Can I use DIDs instead of extensions? |
06:18.11 | veesmooth | im a asterisk newbie so im tring to figure out stuff myself |
06:18.18 | shepherd | DID's like voip lines from a provider? |
06:18.26 | *** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net) |
06:19.19 | l00p | We use a Meridian phone system that has direct numbers (555-5555 not 555-5555 x1234) |
06:19.30 | Qwell | l00p: sure |
06:19.40 | l00p | We are talking about going to asterisk, but don't want to change everyone's number |
06:19.51 | l00p | Anywhere I can read about this? |
06:19.52 | shepherd | yeah.. you can do that :) |
06:20.11 | shepherd | well.. |
06:20.12 | shepherd | basically |
06:20.22 | shepherd | you have to associate a DID with an extension |
06:20.27 | carlosh | hello every one. Could someone tell how many calls would I be able to connect to a meetme dedicated server? many thanks.. |
06:20.36 | shepherd | but it works the same |
06:20.47 | l00p | Where do you do the association? |
06:20.53 | Qwell | l00p: extensions.conf |
06:21.07 | shepherd | <PROTECTED> |
06:21.10 | shepherd | yeah.. there |
06:21.16 | l00p | K, that's what I thought. |
06:21.20 | l00p | Thanks. |
06:21.36 | l00p | You guys here often? |
06:21.40 | shepherd | l00p: read the handbook, it's the old way of doing it, but it works nonetheless |
06:21.48 | Qwell | l00p: 24 hours a day |
06:22.06 | l00p | I read the handbook(mostly) |
06:22.18 | l00p | It is good. |
06:22.18 | shepherd | k |
06:22.31 | l00p | Know anyone in Oregon who has a system up and running? |
06:22.32 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
06:23.37 | l00p | I made my first call about two hours ago using kphone. It was pretty cool. |
06:25.44 | munchausen | someone should set up a simple service which calls any two phone numbers with any specified callid |
06:26.12 | munchausen | why? because |
06:26.34 | luke-jr_ | l00p: exten => _555555XXXX,1,Dial(SIP/${EXTEN:6}) if you're lucky ;) |
06:27.05 | shepherd | neat.. a phone that supports iax |
06:27.05 | luke-jr_ | munchausen: you mean dial 2 #s and connect em? |
06:27.06 | shepherd | http://www.virbiage.com/products/lanphones.php |
06:27.30 | luke-jr_ | shepherd: too bad everyone's moving to the new XIAX, eh? =p |
06:27.37 | l00p | luke-jr: thanks |
06:27.37 | shepherd | yeah.. |
06:27.42 | shepherd | xiax is the shit |
06:27.44 | shepherd | :) |
06:28.19 | shepherd | lord.. if xiax was a reality, i would shoot myself |
06:28.21 | shepherd | heh |
06:28.33 | luke-jr_ | why? |
06:28.38 | shepherd | okay.. not really |
06:28.42 | luke-jr_ | lol |
06:28.42 | shepherd | but it's a bad idea |
06:28.45 | shepherd | hehe |
06:29.01 | luke-jr_ | which part? |
06:29.16 | shepherd | oh just the overhead |
06:29.17 | luke-jr_ | native SSL support would be nice |
06:29.32 | shepherd | the cpu required to parse it |
06:29.39 | luke-jr_ | better than when I have to tunnel VoIP over PPP over SSH at least |
06:29.51 | shepherd | yeah |
06:29.54 | *** join/#asterisk veesmooth (~veesmooth@bgp505102bgs.verona01.nj.comcast.net) |
06:29.56 | shepherd | iax3 needs ssl :) |
06:29.57 | luke-jr_ | (when I don't trust a network) |
06:30.15 | shepherd | you could vpn |
06:30.15 | luke-jr_ | does IAX use TCP? |
06:30.18 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
06:30.20 | luke-jr_ | that's what it is |
06:30.20 | shepherd | no.. udp |
06:30.25 | luke-jr_ | PPP over SSH VPN |
06:30.29 | shepherd | oh, lol |
06:30.46 | shepherd | so you see the server as a local ip? |
06:30.51 | luke-jr_ | yes |
06:31.09 | shepherd | i wonder how much compression it would do on ulaw |
06:31.23 | luke-jr_ | SSL compresses? O.o |
06:31.39 | luke-jr_ | I thought it was merely encryption |
06:31.49 | shepherd | yeah.. yeah.. openssl does compression too, i think |
06:31.58 | shepherd | just depends on the setting |
06:32.06 | luke-jr_ | SSH does... dunno about SSL |
06:32.52 | shepherd | ssl is the protocol layer of ssh :) |
06:32.52 | l00p | I don't think ssl normalyl does compression, but I could be wrong. |
06:33.00 | shepherd | i'm pretty sure |
06:33.12 | luke-jr_ | for the encryption, yes |
06:33.26 | luke-jr_ | I would expect SSH to use something else like zlib for compression tho |
06:33.27 | shepherd | like.. sometimes |
06:33.31 | shepherd | when i use sftp |
06:33.39 | shepherd | it downloads websites superfast |
06:33.44 | shepherd | because it's several files |
06:33.50 | luke-jr_ | sftp is SSH, not SSL |
06:33.53 | shepherd | that have a lot of waist |
06:33.57 | shepherd | and are uncompressed |
06:34.21 | luke-jr_ | SSL may be used in SSH, but SSL still != SSH ;) |
06:34.24 | shepherd | so i just always assumed it was compressing all the files together and sending them at once |
06:34.37 | luke-jr_ | I doubt that |
06:34.52 | luke-jr_ | if it did that, it'd need to be at sftp layer |
06:34.54 | luke-jr_ | not even SSH |
06:35.00 | shepherd | nonetheless, i can pull a website down faster with sftp than ftp :) |
06:35.18 | luke-jr_ | i'm lazy when I do stuff like that |
06:35.26 | l00p | Thanks for the help. I'm out. |
06:35.33 | luke-jr_ | I usually do something like tar cjvp files | nc |
06:35.43 | shepherd | yeah.. |
06:35.47 | shepherd | but when you don't have console access |
06:35.54 | shepherd | got to get it somehow, heh |
06:35.58 | luke-jr_ | ... |
06:36.04 | luke-jr_ | sftp requires shell access |
06:36.08 | shepherd | i have a provider that supports sftp |
06:36.17 | shepherd | but no login |
06:36.21 | luke-jr_ | interesting |
06:36.25 | shepherd | it's a windows server |
06:36.27 | shepherd | bah :( |
06:36.29 | luke-jr_ | evil |
06:39.10 | *** join/#asterisk _vic (~riccardo@gw-fi.esaote.com) |
06:39.33 | veesmooth | so guys |
06:39.36 | veesmooth | im a newbie here |
06:39.42 | veesmooth | can i ask some newbie questions lol |
06:39.47 | shepherd | no |
06:39.49 | shepherd | hehe |
06:39.55 | shepherd | sarcasim |
06:39.57 | shepherd | :( |
06:40.02 | veesmooth | ok, i take that as a yes he he he |
06:40.05 | veesmooth | hopefully |
06:40.12 | veesmooth | i just install asterisk on my system |
06:40.14 | veesmooth | and it work great |
06:40.20 | veesmooth | well with the sample install that is |
06:40.32 | veesmooth | but im wondering if there is something else i can do with it |
06:40.42 | luke-jr_ | sample install I got was useless, IIRC |
06:40.56 | veesmooth | i know most likly i cant dial regular phone lines |
06:41.02 | luke-jr_ | why not? |
06:41.08 | veesmooth | just asumming |
06:41.13 | shepherd | yeah you can |
06:41.13 | veesmooth | can you? |
06:41.16 | luke-jr_ | yea |
06:41.17 | veesmooth | hmm |
06:41.21 | veesmooth | thats wild |
06:41.25 | shepherd | do you have any hardware? |
06:41.25 | veesmooth | so through high speed |
06:41.25 | nDuff | shepherd, no, SSH doesn't use SSL/TLS. |
06:41.25 | luke-jr_ | I suggest http://www.voipjet.com/ |
06:41.27 | drumkilla | not on the sample install ... |
06:41.34 | veesmooth | no hardware |
06:41.45 | veesmooth | doing this through highspeed |
06:41.46 | shepherd | voicepulse! |
06:41.52 | shepherd | nduff: okay.. heh |
06:41.54 | luke-jr_ | tho VoipJet is meant for big groups |
06:42.10 | veesmooth | so i need a special hard ware to call regular phones then |
06:42.15 | luke-jr_ | veesmooth: no |
06:42.15 | veesmooth | cant do it through highspeed right |
06:42.32 | luke-jr_ | veesmooth: Just need someone providing VoIP termination |
06:43.03 | shepherd | nope |
06:43.03 | veesmooth | <PROTECTED> |
06:43.03 | luke-jr_ | lots of companies do it |
06:43.10 | shepherd | you can do a total voip solution :) |
06:43.33 | luke-jr_ | veesmooth: VoIP services are generally either termination or origination |
06:43.41 | luke-jr_ | termination is for outgoing POTS calls |
06:43.49 | shepherd | i use voicepulse for my phone number coming in, into a TDM411 |
06:43.51 | luke-jr_ | origination is for incoming POTS calls |
06:43.58 | shepherd | but my fxo isn't hookup |
06:44.10 | malverian | Hmm, I've noticed my Asterisk server doesn't catch keypad presses when I dial from a cellular phone a lot. |
06:44.17 | malverian | Is there anything I can do to tweak this? |
06:44.27 | luke-jr_ | shepherd: heh... my interest in VoIP originally stemmed from the absense of a non-SBC landline solution |
06:44.53 | luke-jr_ | malverian: use a different codec maybe |
06:45.31 | shepherd | malverian: isn't by any chance a sony cell phone? |
06:46.09 | luke-jr_ | veesmooth: Termination is usually charged in cents/minute; origination is usually a flat monthly fee |
06:46.24 | veesmooth | got ya |
06:46.44 | luke-jr_ | For example, I pay $5/mo origination and 1.3c/min termination |
06:47.03 | veesmooth | ok, so lets say i dont have any of those providers and i just have the software |
06:47.08 | veesmooth | is there anything i can do with it |
06:47.10 | veesmooth | just as it is |
06:47.20 | luke-jr_ | you need to configure it to do something |
06:47.35 | veesmooth | hmm, ok |
06:47.35 | luke-jr_ | but you should be able to call anyone else using VoIP in theory |
06:47.47 | luke-jr_ | via some enum service or dundi |
06:47.55 | luke-jr_ | I haven't got those setup myself yet tho |
06:48.06 | veesmooth | ok, so if someone had a voip then i can dial their number |
06:48.12 | veesmooth | i know the sample software let me dial their company |
06:48.17 | luke-jr_ | yes |
06:48.18 | veesmooth | and fool around with them |
06:48.24 | veesmooth | ok cool |
06:48.33 | veesmooth | so what do u do with asterisk |
06:48.40 | luke-jr_ | me? |
06:48.48 | veesmooth | yes, or anyone else |
06:48.49 | veesmooth | just curious |
06:48.50 | luke-jr_ | It's my phone :) |
06:49.02 | luke-jr_ | well, controls the phones in my appt |
06:49.02 | veesmooth | oh |
06:49.12 | luke-jr_ | and handles calls |
06:49.20 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
06:49.24 | veesmooth | hanles call in a smart way |
06:49.25 | malverian | shepherd: It is.. an ericsson |
06:49.30 | *** join/#asterisk Rick_Hunter (~rhunter@03-111.008.popsite.net) |
06:49.39 | luke-jr_ | not really... I just have it doing speeddial mostly now ;) |
06:49.41 | shepherd | Kevin Mitnick uses asterisk to spoof callerids :) |
06:49.47 | luke-jr_ | shepherd: LOL |
06:49.50 | shepherd | malver: intresting |
06:50.10 | shepherd | i had the same problem yesterday, but wasn't sure it was the phone |
06:50.16 | shepherd | or something else |
06:50.41 | veesmooth | reason im asking cause im suppose to do a project with the software for school |
06:50.48 | veesmooth | trying to get some good ideas with this |
06:50.52 | shepherd | i set mine to 555 555 4444 |
06:50.56 | shepherd | coffee talk :) |
06:51.13 | veesmooth | got two decent subjects so far |
06:51.15 | luke-jr_ | veesmooth: could setup something for your dorm to use... assign em extensions and stuff |
06:51.44 | shepherd | you could use asterisk to control your firewall :) |
06:51.51 | shepherd | meaning.. |
06:51.53 | luke-jr_ | shepherd: wtf? |
06:51.58 | shepherd | firewall everything out |
06:52.03 | shepherd | then when you want access to your sever |
06:52.08 | shepherd | call up your asterisk box |
06:52.16 | luke-jr_ | lol |
06:52.22 | shepherd | key in your ip address and update iptables :) |
06:52.42 | luke-jr_ | shepherd: is there actually apps to do that? |
06:52.46 | veesmooth | think the prof wants me to do something software wise |
06:52.47 | shepherd | no |
06:52.53 | shepherd | but you could do it! |
06:52.55 | shepherd | hehe |
06:52.58 | veesmooth | is that something simple? |
06:53.03 | veesmooth | sounds kinda tricky |
06:53.06 | luke-jr_ | shepherd: sounds like it would be fun if I had time |
06:53.12 | shepherd | well.. it's like writting a cgi |
06:53.16 | luke-jr_ | veesmooth: you'd need to code it |
06:53.23 | shepherd | only in asterisk its an agi |
06:53.29 | veesmooth | can u code that in c? |
06:53.33 | shepherd | yes |
06:53.37 | shepherd | c, java, perl |
06:53.40 | shepherd | python |
06:53.44 | shepherd | php |
06:53.45 | *** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net) |
06:53.53 | shepherd | c# even |
06:53.58 | Qwell | eww |
06:54.11 | Qwell | Does it use mono or something? |
06:54.15 | shepherd | yeah |
06:54.29 | shepherd | i think the support for c# agi is kinda sketchy though |
06:54.31 | veesmooth | is that something complicated to do |
06:54.32 | shepherd | i wouldn't trust it |
06:54.50 | shepherd | vees: not really.. i learned agi before i learned extensions.conf ;0 |
06:54.53 | shepherd | ;) |
06:54.56 | luke-jr_ | maybe write a voice bulletin board ;) |
06:55.33 | veesmooth | maybe |
06:55.42 | shepherd | that might be harder to do though |
06:55.45 | veesmooth | im writting down all these ideas |
06:55.51 | veesmooth | so i can look into it |
06:55.57 | veesmooth | ok |
06:55.58 | luke-jr_ | the hard part of a BB would be the UI design |
06:55.58 | shepherd | the firewall idea would be really simple project |
06:56.05 | shepherd | as far as agis go |
06:56.12 | veesmooth | but i think what the prof wants me to do |
06:56.16 | veesmooth | is take the asterisk code |
06:56.17 | veesmooth | itself |
06:56.23 | veesmooth | which i believe is written in c |
06:56.25 | shepherd | yup |
06:56.28 | veesmooth | and do something with it |
06:56.48 | veesmooth | i have to refresh my brain on c |
06:56.59 | veesmooth | lol but i love programming |
06:57.03 | shepherd | well.. asterisk is modular, so you can do a lot without changing the core of asterisk :) |
06:57.22 | veesmooth | oh |
06:57.30 | veesmooth | so in other words i can just add on with no problem |
06:57.36 | shepherd | yeah |
06:57.38 | veesmooth | with out changing the basics though of the program |
06:57.44 | shepherd | if it doesn't already do it |
06:58.04 | shepherd | more than likely, it can, or someone has a program out there that will do whatever you want it to do |
06:58.15 | veesmooth | oh |
06:58.19 | veesmooth | so i can download programs |
06:58.21 | veesmooth | and just add it on |
06:58.30 | veesmooth | just like legos |
06:58.31 | veesmooth | lol |
06:58.34 | veesmooth | add and take away |
06:59.01 | veesmooth | is that what u mean? |
07:00.08 | veesmooth | ok shepherd and luke |
07:00.08 | shepherd | not as simple ;) |
07:00.19 | veesmooth | tell me if this sound like something |
07:00.24 | veesmooth | that is do able |
07:00.30 | veesmooth | i got two subjects so far though |
07:00.36 | veesmooth | that was sugested by some other folks |
07:00.50 | veesmooth | one: better echo cancellation algorithm |
07:00.53 | *** part/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
07:01.04 | veesmooth | two: extending the max latency for a voip |
07:01.14 | veesmooth | do you think that sounds like something? |
07:01.36 | shepherd | heh |
07:01.46 | shepherd | i would stay away from the echo cancellation |
07:01.59 | shepherd | mark said that was the hardest thing he ever had to do in asterisk ever |
07:01.59 | veesmooth | that sound kinda complicated |
07:02.05 | veesmooth | hmm |
07:02.06 | veesmooth | ok |
07:02.09 | veesmooth | i keep that in mind |
07:02.23 | veesmooth | <PROTECTED> |
07:02.24 | shepherd | but the max latency sounds neat |
07:02.28 | veesmooth | oh yeah |
07:02.42 | shepherd | but, most of that would just be configuration changes |
07:02.47 | veesmooth | think that sounds kinda doable |
07:03.06 | shepherd | and you can do the jitter buffer and just talk really slow |
07:03.14 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
07:03.24 | veesmooth | hmm, have u work with c and asterisk before |
07:03.31 | shepherd | no |
07:03.36 | shepherd | i can't really code in c |
07:03.41 | veesmooth | hmm |
07:03.42 | shepherd | most of what i've use was perl and php |
07:04.00 | veesmooth | hmm cause im wondering if it would take a lot to tweak the program |
07:04.03 | veesmooth | using c |
07:04.14 | veesmooth | since i never really went into a program and change stuff |
07:04.22 | veesmooth | i usually create things from scratch if anything |
07:04.29 | shepherd | yeah |
07:04.40 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
07:04.41 | shepherd | now i have tweaked stuff in c |
07:04.49 | veesmooth | think this sounds like a tough project? |
07:04.51 | shepherd | like max threads |
07:04.54 | veesmooth | or something thats doable |
07:05.00 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
07:05.02 | *** join/#asterisk tartar (~tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
07:05.39 | shepherd | you could speed up iax initiations by doing away with linked lists ;) |
07:05.54 | veesmooth | hmm, ok |
07:06.10 | veesmooth | but this project does sound do able though right |
07:06.24 | shepherd | yeah |
07:06.33 | shepherd | but |
07:06.43 | *** join/#asterisk |nix (~inix@202.148.164.48) |
07:06.45 | shepherd | you might want to look over the code first |
07:06.55 | shepherd | and focus on one part |
07:07.01 | veesmooth | yeah true |
07:07.07 | veesmooth | the comments should tell u what it does right |
07:07.18 | shepherd | because, i'm not sure how about pieces affect latency, etc |
07:07.36 | shepherd | how many pieces |
07:07.40 | veesmooth | yeah, well he said im a be doing a lot of hours |
07:07.42 | veesmooth | and research |
07:07.48 | veesmooth | as long as i have an idea |
07:07.52 | veesmooth | and willing to put in some work |
07:07.54 | veesmooth | and time |
07:07.56 | opus_ | does anyone know how to simulate latency? |
07:08.03 | veesmooth | this is mostly in the fall |
07:08.05 | veesmooth | just got to prepare now |
07:08.31 | veesmooth | so this site a lot of people come on and help right |
07:08.41 | veesmooth | so far i got some decent help here |
07:09.02 | shepherd | opus: you could probably throttle back the bandwidth and ping flood yourself :) |
07:09.05 | opus_ | whats up man |
07:09.14 | opus_ | shepherd- not a bad idea |
07:10.39 | Qwell | wtf... |
07:10.57 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
07:10.58 | shepherd | ?? |
07:11.11 | Qwell | my brother is telling me about a hotel he and my dad did a telecom install for... They asked them to give the bathroom in each room a seperate extension |
07:11.28 | Qwell | main room would be 1011, bathroom would be 2011 |
07:12.05 | Himeko | that would be handy |
07:12.11 | Qwell | seems retarded |
07:12.22 | shepherd | why would anyone want that |
07:12.33 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
07:12.34 | Himeko | you could call the bathroom from the room then |
07:12.41 | Himeko | and visa versa |
07:12.52 | shepherd | or! talk through the door |
07:12.54 | shepherd | hehe |
07:12.56 | Qwell | heh |
07:13.16 | Himeko | why, when you got separate extentions |
07:14.55 | Himeko | hey slePP, you still up |
07:15.22 | Qwell | How much could a hotel with 2,000 rooms save by going from a normal PBX to *? |
07:15.43 | Qwell | or, rather... |
07:15.48 | Qwell | going to * instead of a normal PBX |
07:16.00 | Qwell | I didn't really think my question out the first time :p |
07:16.26 | slePP | Himeko: yeh |
07:16.29 | shepherd | the question should be |
07:16.34 | shepherd | can asterisk do 2,000 lines |
07:16.43 | Qwell | shepherd: I'm sure it could, with a few boxes |
07:16.55 | Himeko | slePP i don't think the * box knows it is DST |
07:17.13 | slePP | the box does.. asterisk apparently isn't aware, though |
07:17.20 | Himeko | weird |
07:17.37 | slePP | ergasio root # date |
07:17.37 | slePP | Mon Apr 4 01:17:04 MDT 2005 |
07:18.09 | Himeko | it must checking the rtc itself and doing the the tz or something |
07:18.33 | slePP | no.. asterisk tends to maintain its own clock |
07:18.37 | slePP | actually |
07:19.15 | slePP | what are you figuring the time is wrong from? |
07:19.15 | slePP | caller id? |
07:19.15 | Himeko | ya |
07:19.15 | slePP | that's the pap2 |
07:19.25 | Himeko | it has it's own clock eh |
07:19.27 | slePP | it lacks a concept of DST |
07:19.50 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
07:19.50 | slePP | the pap2 uses ntp to keep in sync, but it doesn't adjust itself |
07:19.50 | *** join/#asterisk eivindtr (~eivind@062016241059.customer.alfanett.no) |
07:19.52 | slePP | on the next profile update it should clear itself up |
07:19.59 | Himeko | i just figured it passed the cid though not generate it from itself |
07:20.20 | slePP | it generates the local CID info from the SIP |
07:20.43 | slePP | but it doesn't take the time from it |
07:21.08 | Himeko | i thought it was set to check the config every 5 min |
07:21.29 | Himeko | or not anymore |
07:21.29 | slePP | ew, no... they have this nasty habit of rebooting when they don't need to. |
07:21.35 | slePP | it checks configs every hour or two or 24 |
07:22.01 | slePP | mind you, you're not on the new provisioning system.. |
07:24.53 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
07:33.44 | carlosh | hello every one. Could someone tell how many calls would I be able to connect to an server dedicated to meetme conferences ? many thanks.. |
07:34.08 | carlosh | ..an * server... |
07:35.25 | Qwell | carlosh: There are alot of variables |
07:35.36 | Qwell | hardware used, codecs, amount of transcoding, etc |
07:35.45 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-28-204.w82-124.abo.wanadoo.fr) |
07:35.45 | Qwell | could be 10, could be 1,000 |
07:35.53 | *** part/#asterisk mbaron (~mbaron@AVelizy-154-1-28-204.w82-124.abo.wanadoo.fr) |
07:36.53 | carlosh | Qwell: say good bandwidth, 2.4GHZ processor or better, 1GBRAM... hyperthreading CPu, the works... |
07:37.06 | Qwell | carlosh: There are still alot of variables involved |
07:37.28 | carlosh | Qwell: how would you measure the usage per 10 users or 100 ? |
07:37.45 | carlosh | Qwell: please ellaborate.. what variables.. ? |
07:37.52 | Qwell | codecs, amount of transcoding, etc |
07:37.53 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
07:38.14 | Qwell | carlosh: Thats like me asking "How fast can my car go if I push the pedal all the way down?" |
07:38.21 | Qwell | without me saying what type of car I have |
07:38.49 | Qwell | Its something you'll have to test yourself, for your specific settings |
07:39.25 | PTG1234 | Qwell: pretty fast in my case :) |
07:39.32 | Qwell | PTG1234: Nobody asked you :p |
07:39.56 | Qwell | How's your coding going? |
07:40.00 | carlosh | Qwell: so, there are not yet benchmarks done i take it.. |
07:40.13 | zoa | yes, where is that new chan_sip ? |
07:40.13 | Qwell | carlosh: I'm sure there are plenty done, but it doesn't work that way |
07:40.18 | PTG1234 | Um well i got distracted this weekend :) |
07:40.20 | zoa | your deadline was last friday wasnt it ? |
07:40.21 | zoa | :) |
07:40.23 | PTG1234 | but should have it done by tommorow |
07:40.29 | Qwell | isn't the chan_sip stuff done? |
07:40.33 | PTG1234 | i need the dnc list as well, working on gettingt hat |
07:40.36 | oej | Tell me more about the chan_osip |
07:40.39 | PTG1234 | well a good chunk |
07:40.51 | carlosh | Qwell: R U one of the programmers? |
07:40.53 | Qwell | PTG1234: Yeah, I was wondering if you were gonna remember the dns, heh |
07:40.54 | Qwell | carlosh: no |
07:41.02 | PTG1234 | the dnc you mean? :) |
07:41.08 | Qwell | dnc, yeah |
07:41.09 | PTG1234 | technically we are exempt |
07:41.09 | Qwell | typo ;p |
07:41.17 | PTG1234 | so i am 50/50 rather i should use it :) |
07:41.20 | PTG1234 | well maybe 75/25 |
07:41.37 | Qwell | I would use it, honestly. I can explain my reasoning in a pm if you'd like to hear it |
07:41.42 | PTG1234 | sure |
07:42.53 | carlosh | Qwell: U using * to handle lots of calls ? |
07:43.11 | Qwell | carlosh: 1 simultaneous call |
07:43.26 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
07:43.40 | carlosh | Qwell: hmm... don't know what to say.. I initially thought (from youranswers) that you had the experience.. |
07:43.54 | zoa | PTG1234, could you discuss your chan_sip changes with oej ? |
07:44.02 | carlosh | I handle more than that.. ;o) no offense.. |
07:44.03 | Qwell | carlosh: I watch alot. |
07:44.19 | Qwell | carlosh: I've simply seen your question asked countless times. |
07:44.20 | PTG1234 | discus how so? :) when i am done there won't be any original code in there :) |
07:44.21 | PTG1234 | more or less |
07:46.35 | carlosh | Qwell: I'd say there should be by now an idea of the footprint (processor time/memory) per call... per one call using a particular protocol and codec.. |
07:46.46 | *** join/#asterisk kensuke (~bryan@rrba-146-111-08.telkomadsl.co.za) |
07:46.53 | Qwell | carlosh: to be fair, you still haven't specified any details about the calls |
07:47.02 | kensuke | morning all ... can anyone reccomend a console based sip phone ? |
07:47.10 | carlosh | SIP/ilbc no transcoding... |
07:47.34 | *** join/#asterisk three55ml (~none@cpe-66-68-98-68.austin.res.rr.com) |
07:47.35 | Qwell | see, thats a bit better |
07:47.43 | PTG1234 | why would you use asterisk to handle alot of calls if you don't need transcoding |
07:47.45 | carlosh | :o) |
07:47.58 | PTG1234 | thats whate ser is for, till ir elease my chan_sip :) |
07:47.59 | three55ml | Anyone with a SIP ATA interested in doing some beta testing? I'll provide a free DID for temporary use. |
07:48.20 | three55ml | Or SIP phones. |
07:48.24 | carlosh | cos you might need it only as an Open source quicker replacement to SER.. maybe.. |
07:48.45 | PTG1234 | ser is open source |
07:48.51 | carlosh | i know |
07:50.43 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
07:50.53 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
07:51.56 | wildcard0 | PTG1234, are you updating chan_sip to behave more like a traditional proxy? |
07:52.10 | carlosh | I think that answers my question (thanks PTG1234) |
07:52.56 | carlosh | However, if I want people to use the meetme functionality.. SER would not apply.. :( |
07:54.55 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
07:55.08 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
07:55.12 | smurfix | Anybody knowledgeable WRT zaptel.conf? I want to declare spans for the third+fourth card (zaphfc) only, but ztcfg doesn't appear to let me do that |
07:57.50 | *** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net) |
08:04.30 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
08:08.22 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
08:08.32 | Mazda-MX5 | hi ~ every body . |
08:08.47 | three55ml | Hey |
08:10.40 | timecop | hm |
08:11.14 | Mazda-MX5 | I have question , what mean the "callid" for sip.conf? |
08:11.19 | Mazda-MX5 | thank you |
08:11.22 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
08:11.23 | Qwell | callerid? |
08:11.31 | Mazda-MX5 | yes , is callerid |
08:11.51 | Qwell | the name and number of the person calling you, or the name and number the person you are calling sees |
08:12.16 | Qwell | Thats by far the most interesting question I've seen. |
08:12.23 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
08:12.59 | Mazda-MX5 | thank you , I see. callerid can same the sip context name ? |
08:13.00 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
08:13.20 | Qwell | huh? |
08:13.43 | three55ml | Mazda-MX5: The callerid can be set to anything you want. It will show up to other SIP clients as whatever you set. |
08:14.03 | three55ml | Mazda-MX5: So yes, it can be the same as the context if you so desire. |
08:14.03 | Qwell | three55ml: That made sense? |
08:14.11 | PTG123 | needs to be a # not a name |
08:14.35 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
08:14.39 | Mazda-MX5 | thank you ! three55ml , and Qwell , thank you very much |
08:14.49 | three55ml | Ideally, yes, it should be the extension :) |
08:15.12 | Mazda-MX5 | I can use ${CALLERIDNUM} to know callerid number , right ? |
08:15.21 | Qwell | Mazda-MX5: yes |
08:15.26 | three55ml | Yes |
08:15.43 | smurfix | Is there a way to get yor password reset in the bugtracker? Want to submit a patch and my old login seems to be ... old. |
08:15.49 | three55ml | But keep in mind that you can easily change the caller ID information based upon what type of channel you're pulling it from. |
08:16.21 | Qwell | smurfix: gotta ask an admin I think |
08:16.25 | nesys | hi folks ... there's someone with * and ccme? I've a call-forward problem from ccme to * via sip trunk |
08:22.47 | oej | smurfix: The asterisk-bugs channel is where you find bug marshals that handle mantis. I'll check if you can change. What's your id? |
08:23.25 | smurfix | oej: Thanks -- found the old email with my password in an unlikely place. :-/ |
08:23.36 | oej | Ok, see you in the bug tracker |
08:23.41 | smurfix | oej: (Depending on cookies for your brain isn't a good idea sometimes ;) |
08:23.43 | Qwell | "unlikely place", email inbox? |
08:24.51 | smurfix | Qwell: no, personal email in box from friends (i.e.. grossly misfiled) |
08:24.57 | Qwell | ahh |
08:24.59 | oej | smurfix: You need to disclaim all patches, regardless if they're trivial |
08:25.59 | smurfix | oej: Gah. With real physical paper, or does Digium accept GPG signatures? |
08:26.03 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
08:27.04 | oej | Real physical paper on fax as far as I know. Sometimes a scan in mail to mark |
08:29.52 | oej | smurfix: Why two bug reports? |
08:30.39 | smurfix | oej: Because my browser crashed sending the first one and I didn't think to check whether it succeeded |
08:31.03 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
08:31.07 | oej | Which one do I close? |
08:32.15 | smurfix | oej: the one whose explanation sounds less clear to you |
08:32.29 | smurfix | the patch is the same |
08:32.40 | oej | smurfix: Closed the first one |
08:32.45 | smurfix | thanks |
08:34.34 | miller7 | Any European here interested in USA numbers? |
08:34.47 | miller7 | I have a new * box so if anyone's interested msg me |
08:36.01 | *** join/#asterisk zilas (~zilas@c-24-30-75-206.hsd1.ga.comcast.net) |
08:36.26 | zilas | hello |
08:36.33 | zilas | anybody still alive? |
08:36.43 | three55ml | zilas: A few people |
08:37.16 | zilas | :) good. I sent 5 hours trying to configure call parking...no luck |
08:37.51 | three55ml | Sorry, I haven't played with it much. |
08:37.58 | Zeeek | zilas what's wrong? |
08:38.05 | Zeeek | what's to configure? |
08:38.07 | *** join/#asterisk |Vulture| (~Vulture@152.238.204.68.cfl.res.rr.com) |
08:38.09 | zilas | its a simple thing |
08:38.16 | zilas | how you place on hold |
08:38.21 | |Vulture| | anyone here use nagios? |
08:38.32 | zilas | you should punch #700? |
08:38.38 | Zeeek | yes |
08:38.47 | Zeeek | when you hit # you should hear "transfer" |
08:39.01 | Zeeek | if you don't, it's because you didn't include a 't' in the dial command |
08:39.34 | zilas | oh this may be the issue |
08:39.59 | Zeeek | could be |
08:40.02 | zilas | ok how should it look like in extension.conf? |
08:40.11 | Zeeek | show application dial |
08:40.26 | Zeeek | that will give you the options |
08:41.03 | Zeeek | the trouble is that using 't' or 'T' will stop you from using the # key if you call an IVR |
08:41.12 | Zeeek | so you have to be careful |
08:41.21 | zilas | I dont have IVR |
08:41.42 | Zeeek | I mean if you call your bank or something |
08:42.02 | zilas | it makes sense |
08:42.05 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
08:42.18 | Zeeek | if you used T in the dial and they say "now hit your pin followed by the # key" the poiund key will not be heard. Instead you'll hear "transfer" |
08:42.44 | *** join/#asterisk scanna (~scannachi@ppp-28-156.25-151.libero.it) |
08:43.45 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
08:44.41 | Zeeek | so did you try a dial with the 't' ? |
08:44.46 | zilas | how it should look exten => 1,1,Dial??? |
08:44.55 | zilas | under [parking]? |
08:45.05 | Zeeek | what does it look like now? |
08:45.19 | zilas | all I have is include parking |
08:45.24 | Zeeek | no not under parking, in whatever context you want to put it in |
08:47.00 | zilas | file:/media/cdrom/bad boys blue - euro hits 2000.mp3 |
08:47.00 | zilas | file:/media/cdrom/Bad Boys Blue - Gimmie, Gimmie Your Lovin'.mp3 |
08:47.07 | zilas | sorry |
08:48.44 | zilas | I have as an example exten => 1,1,Dial(SIP/phone1,20,tr) ???? |
08:48.54 | Zeeek | that's good |
08:49.09 | Zeeek | that will work for parking if you RECEIVE a call |
08:49.23 | zilas | what does here phone1 stand for? |
08:49.35 | Zeeek | the name of your phone |
08:49.54 | zilas | like extension of the phone? |
08:50.10 | Zeeek | if you had a phone called 2002 that would be SIP/2002 |
08:50.27 | Zeeek | it seems to me you should try to read up a little on the dialplan |
08:50.46 | Zeeek | here are a few links: |
08:50.48 | Zeeek | Starter tutorial: |
08:50.48 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
08:50.48 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
08:50.48 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
08:50.48 | Zeeek | THE reference of the moment: |
08:50.49 | Zeeek | http://www.asteriskdocs.org |
08:51.16 | zilas | I know :) but its so puerly documented for dialplan.... I'll try on those urls |
08:51.45 | Zeeek | the dialplan is explained in detail in the asteriskdocs.org book |
08:51.47 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
08:51.47 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
08:51.50 | Zeeek | and here |
08:52.23 | zilas | I read google on all call parking and you was the one who put me on a right direction |
08:53.15 | Zeeek | call parking is so simple that it is complicated! |
08:54.04 | zilas | I figured that :-)... I am reading this good Guide to Asterisk VOIP man its so goofy |
08:54.34 | Zeeek | I think the best thing to read is this: http://www.automated.it/guidetoasterisk.htm |
08:54.43 | Zeeek | it has most of the stuff you want to start wiht |
08:55.08 | zilas | so what you need to park a dialed call? |
08:55.20 | Zeeek | T instead of t |
08:55.30 | Zeeek | you can put both to test rtT |
08:56.24 | ronn | hi guys |
08:57.06 | ronn | i have been trying to use g726 codec on a supura 2000 with asterisk .. but always get incompatible codec error |
08:57.15 | ronn | anyone having the same problem? |
08:57.57 | *** join/#asterisk cced (~wangxinta@222.33.36.198) |
08:58.41 | zilas | zeek: ok if I have 4 phones I write my example 4 times correct? |
08:59.16 | Zeeek | zilas you could do that, yes. Or you could use a macro that does it |
08:59.51 | zilas | thats more complicated already I just need to start with simple anyway THANKS A LOT! |
09:00.54 | Zeeek | yeah, once you get the four working you can look into macros |
09:04.28 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
09:07.44 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
09:12.58 | langals | hi there....wondering if someone could answer a few questions on use of the g729 codec? |
09:13.39 | WilliamK | langals, ask away |
09:13.48 | langals | I am using meetme conferencing. All of my phones will be using g729, so do I need to license g729 on Asterisk? |
09:13.58 | WilliamK | yep |
09:14.22 | langals | Even though there is no transcoding being done? |
09:14.59 | WilliamK | if you're connecting to the asterisk box using g729 and asterisk is having to speak back on g729 then you gotta license it |
09:15.15 | timecop | er |
09:15.19 | *** join/#asterisk kks (~kks@203.115.208.140) |
09:15.22 | timecop | this aserisk flash panel shiz seems to be rather limited |
09:15.31 | timecop | how the hell do I tell it to rename a trunk of SIP buttons with their Caller ID? |
09:16.01 | timecop | its kinda useless to see 10 open sip channels all pointing into a conference and not being able to know who is on that channel |
09:16.05 | langals | William - ok - another question: Is it possible to use g729 for non-commercial purposes without buying licenses? |
09:16.21 | zigman | langals njo |
09:16.23 | zigman | no |
09:16.26 | langals | I know there is a non-commercial version, but I believe this is not as good quality? |
09:16.33 | *** join/#asterisk E818 (anonymous@rrcs-24-199-5-190.west.biz.rr.com) |
09:16.52 | langals | So one would need to buy licenses in order to enable it? |
09:17.18 | timecop | right |
09:17.52 | langals | And then on the client side - does anyone know anything about licensing g729 on windows? |
09:17.59 | ard | As far as I know, this is a software patent. That means it doesn't matter: you need a license to use it, even for non-commercial purposes. |
09:18.42 | ard | Licensing starts at 15k $ :-) |
09:18.54 | langals | ard - so it is not like mysql, where it is based on trust that you get a license when you go commercial? |
09:19.11 | ard | mysql is GPL, you buy support. |
09:19.27 | ard | Unless you need some extra features. |
09:20.02 | ard | But digium made a binary codec, and made that $15k deal, so you can buy from them a license for a single channel for $10 or so. |
09:21.32 | WilliamK | langals, no because it's patented technology, and the owner of the patent wants his/her royalties |
09:21.33 | ard | Actually you can make the codec yourself (use source from intel). And if you tell nobody, your ok :-) |
09:22.34 | *** join/#asterisk Qorky (~goaway@202.173.160.18) |
09:22.54 | langals | ard - where do I get that - don't get me wrong, I do mean to license when going commercial, but I want to try it out first to make sure it works with client |
09:24.26 | WilliamK | langals, get a couple licenses then and build on to it later |
09:24.38 | ard | http://lists.digium.com/pipermail/asterisk-users/2004-January/035492.html |
09:24.49 | WilliamK | this is interesting |
09:24.50 | WilliamK | =) |
09:25.00 | ard | the easy way out is really buying that license from digium |
09:25.29 | ard | you can write the codec yourself, but that means your only save in the EU for now... or china |
09:26.21 | langals | I am in South Africa |
09:26.50 | ard | http://www.voip-info.org/wiki-Asterisk+G.729+licensing |
09:26.55 | ard | Hmmmm |
09:27.11 | ard | I dunno about south africa |
09:29.10 | Qorky | anyone have some example configs for connecting two asterisk servers with iax ? |
09:32.10 | Zeeek | Qorky what do you want to achieve specifically? |
09:34.07 | ronn | has anyone managed to use g726 codec on a supura 2000 with asterisk? |
09:38.57 | Qorky | Zeeek. I have 2 offices. wonna be able to talk between them. |
09:39.22 | Qorky | one office is 10XX numbers. the other is 11XX numbers. ideally i want to leave the option of adding a third later. |
09:39.27 | Zeeek | I would say you can just set them up as peers |
09:39.54 | h3x | i think digium's codec does g.729b too |
09:39.58 | Zeeek | and route the calls by the first two digits as you have planned |
09:40.32 | Qorky | yar. makes sence. any config example or howtos around. maybe on the wiki ? |
09:41.39 | Zeeek | I think if you just "pretend it's FWD" it'd work! |
09:41.54 | Zeeek | you know, standard peer entry |
09:41.59 | Zeeek | in iax.conf |
09:42.15 | Zeeek | there was also a switch keyword you might want to look up |
09:43.59 | Qorky | ah yeah ok. let me see. |
09:50.55 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
09:51.30 | *** join/#asterisk tainted- (~tainted@adsl-69-108-96-54.dsl.irvnca.pacbell.net) |
09:56.45 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
09:57.25 | *** join/#asterisk RoyK (~roy@83.223.171.239) |
09:59.49 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
10:06.06 | tainted- | anyone have issues where sip channels don't go away after hangup? |
10:10.29 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
10:11.32 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
10:18.06 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
10:18.09 | *** join/#asterisk sunil (~sunil@202.54.37.181) |
10:18.54 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
10:18.59 | facek_ | anyoen alive? |
10:19.12 | RestLessGemini | yup |
10:22.34 | facek_ | what are the callgroups? |
10:24.16 | |Vulture| | facek_: in zap configs? |
10:24.53 | facek_ | yes |
10:25.05 | facek_ | and how can i connect groups with peers from iax and sip.conf |
10:25.06 | |Vulture| | call groups if you mean in zapata.conf are just groups that are set for a dial out |
10:25.11 | *** join/#asterisk cced (~wangxinta@222.33.36.198) |
10:26.06 | |Vulture| | facek_: not following.. but groups are like instead of Dial(Zap1/555) you can issue Dial(Zap/g1/555) |
10:26.14 | |Vulture| | something like that... I am kinda tired lol |
10:31.22 | *** join/#asterisk brc-tux (~brc-tux@p54A98D54.dip0.t-ipconnect.de) |
10:31.30 | *** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net) |
10:33.20 | lters | how soon will we see the 1.2 release ? |
10:33.46 | lters | can't wait for the new options :) |
10:37.43 | *** join/#asterisk RoyK (~roy@83.223.171.239) |
10:40.25 | *** join/#asterisk _Crash_ (~me@adsl3p183.access.maltanet.net) |
10:40.41 | _Crash_ | Can anyone help me with an MOH issue I have? |
10:45.16 | zoa | _crash_ what is the issue ? |
10:48.29 | _Crash_ | Hi zoa...is there some specific setting to save MP3s for MOH? |
10:52.57 | zoa | yeah |
10:53.05 | zoa | no vbr for a start |
10:53.16 | zoa | try old formats |
10:53.17 | zoa | no vbr |
10:53.20 | zoa | 128kbit |
10:53.23 | zoa | will probably do fine |
10:53.23 | *** part/#asterisk brc-tux (~brc-tux@p54A98D54.dip0.t-ipconnect.de) |
10:53.41 | _Crash_ | 10x.... |
10:53.57 | _Crash_ | Would you know of there is some bug reporting list for AMP? |
10:55.05 | |Vulture| | AMP... has a bug?! *gasp* :P |
10:57.49 | E818 | help! |
10:57.50 | E818 | asterisk: relocation error: /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_cust_config_register |
10:58.06 | pif | hi, if a phone sends back a 302 redirect to an outside number, how can I have * refuse that? |
10:58.10 | E818 | is there a problem with the current cvs version? |
10:58.56 | _Crash_ | What are the viable alternatives to AMP...? |
11:08.36 | three55ml | Anyone know how to enable SIP realtime caching? |
11:09.27 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
11:09.39 | |Vulture| | _Crash_: for a household user AMP is fine, but for any business AMP should not be used, its just sloppy if you know how to do conf files yourself |
11:12.32 | _Crash_ | |Vul|: Yes can edit conf directly but is there "the" business AMP equivalent? |
11:13.02 | facek_ | i am working on that ;] |
11:14.57 | three55ml | _Crash_: I'm working on one, to a degree |
11:19.10 | facek_ | do you have soem interesting dialplan for an example? |
11:20.46 | three55ml | I'm doing more of an end-user interface, I'm splitting off the UI I've developed for a service I'm working on to it's own program. It looks like this: http://www.premierpbx.com/images/screenshots/2.jpg |
11:25.30 | *** join/#asterisk Corydon76-home (twelve@pcp08665860pcs.500ash01.tn.comcast.net) |
11:27.52 | *** join/#asterisk Darwin35 (~Darin@24.3.226.147) |
11:27.59 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
11:30.54 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
11:32.23 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
11:32.47 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net) |
11:33.09 | mithro | hello, if i could produce an 8 port FXS device for under $100 US, do you think it would be a good deal? |
11:33.33 | InfraRed | i'd buy one |
11:34.38 | mithro | it would be a USB based system with the host computer software doing most of the work |
11:34.45 | InfraRed | heh |
11:36.08 | mithro | it would be basically a simple USB PIC with 8 DACs and some DC/Ringing logic |
11:37.01 | mithro | so DTMF detection, voice encoding, etc would all be software |
11:38.37 | *** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au) |
11:38.39 | h3x | Theres 4 port fxs to ethernet already |
11:38.48 | h3x | it costs maybe a bit more but hey at least you dont need a computer |
11:39.07 | Wonka | mithro: do you intend to put the plans open? |
11:39.14 | mithro | yeah but they are all expensive ie > $300 US |
11:39.19 | mithro | Wonka: most probably |
11:39.26 | Wonka | h3x: you need a computer to run * already... |
11:39.30 | h3x | nah ive seen one for less than 200 |
11:39.31 | Wonka | cool |
11:39.34 | h3x | Wonka: not if its a remote location |
11:39.41 | h3x | or service provider model |
11:39.42 | tainted- | three55ml that looks a lot like switchvox |
11:39.46 | mithro | h3x: where? |
11:39.54 | webman | anyone used wcusb with linux 2.6.x ?? I seem to get a segv when running ztcfg ? |
11:39.59 | h3x | well a taiwanese company makes a 4 port |
11:40.01 | h3x | ummm |
11:40.03 | h3x | repotec.com |
11:40.34 | h3x | ive never used it but i have some of their ethernet switches |
11:40.36 | mithro | if we pushed for mass production we could quickly drop the price |
11:40.46 | h3x | and they are of decent quality and cheap |
11:41.35 | tainted- | mithro how much would it be to include the other stuff |
11:41.42 | tainted- | mithro voice encoding etc |
11:41.53 | mithro | tainted-: quite alot |
11:42.08 | tainted- | licensing costs or chip costs |
11:42.13 | mithro | h3x: where can you buy them |
11:42.20 | mithro | tainted-: both |
11:42.35 | h3x | ive got a chinese importer in walnut california that i buy the other stuff form |
11:42.35 | h3x | from |
11:42.41 | h3x | ill have to ask them about that box |
11:42.51 | tainted- | h3x are u in socal? |
11:43.00 | h3x | no im in vegas |
11:43.10 | tainted- | which importer? |
11:43.14 | h3x | pi |
11:44.13 | tainted- | pi? |
11:44.20 | h3x | yes, P.I. thats what its called |
11:44.20 | tainted- | just pi? |
11:44.20 | h3x | heh |
11:44.26 | tainted- | oh |
11:44.59 | h3x | who knows its probably some mandarin acronym for "stupid roundeye" |
11:45.00 | h3x | :) |
11:45.11 | tuxinator_linux | he he |
11:45.13 | Supaplex | lol |
11:45.34 | h3x | I've spent 10 grand with those guys |
11:45.36 | mithro | i could most probably do a single USB FXS device for under $30 US |
11:45.37 | h3x | in the past 8 months or so |
11:45.53 | tainted- | mithro but where do u see the need? |
11:45.56 | h3x | they even carry fiber stuff |
11:46.03 | Supaplex | I'm courios, how do you do the ringing circuit? |
11:46.04 | tainted- | repotec looks pretty cool |
11:46.27 | h3x | thats it, im emailing my sales rep now and asking her if they have these things |
11:46.40 | mithro | i see the need because currently it costs me ~$100 for 2 FXS ports |
11:47.01 | tainted- | why USB |
11:47.02 | h3x | buy Sipuras |
11:47.06 | h3x | they are only $83 for two ports |
11:47.11 | Supaplex | yea |
11:47.19 | tainted- | i'm waiting for someone to embed asterisk in a soekris box or something |
11:47.26 | tainted- | that'd be cool |
11:47.32 | three55ml | tainted-: It's been done |
11:47.46 | tainted- | three55ml your stuff looks like switchvox |
11:47.51 | tainted- | three55ml where? |
11:48.02 | three55ml | There's some info about it on the wiki |
11:48.36 | three55ml | tainted-: I actually wrote it before Switchvox even came out. I think the UI is infiniately more friendly than Switchvox's as well. Two different target audiences I think. |
11:48.40 | h3x | the real question is |
11:48.43 | h3x | *drumroll* |
11:48.48 | h3x | does that repotec REALLY support T.38 |
11:48.56 | mithro | if i could do $100 for 4 FXS ports, it's better then $83 for 2 FXS |
11:49.04 | three55ml | http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20Soekris |
11:49.36 | tainted- | nDuff u can roll your own in a few hours |
11:49.36 | tainted- | oh wait.. hardware phone |
11:50.05 | h3x | mithro: maybe, but the sipura rules. |
11:50.09 | tainted- | no way manu would use openvpn.. they'd prefer something proprietary and sell it as a 'premium feature' |
11:50.20 | h3x | it has a dsp in it |
11:50.24 | h3x | so the echo can actually works right |
11:50.29 | h3x | and it supports a shitload of codecs |
11:50.43 | webman | I think the PA1688 chipset phones support IAX2, and probably could be convinced to support openvpn ... if they can take the CPU encryption penalty |
11:53.40 | cypromis | they could |
11:53.42 | h3x | Repotec's RP-ID162 is interesting also |
11:53.57 | cypromis | but they still miss some iax2 features, which is no wonder with the current state of documentation and specification of the protocol |
11:54.29 | h3x | Repotec's stuff is actually pretty solid for ethernet anyway, the metal casings are heavy duty, and I opened up an ethernet switch and it was a good quality circuit board and broadcom chipset |
11:54.37 | h3x | with heatsinks and everything |
11:55.03 | h3x | I bought a gigE 8 port switch for $80 from california |
11:55.09 | h3x | and thats qty 1 of course |
11:55.21 | h3x | they have some rack mount stuff with 802.1q vlans etc |
11:55.57 | tainted- | three55ml |
11:55.59 | tainted- | http://www.premierpbx.com/products.php |
11:56.11 | tainted- | ever heard of 37signals? |
11:56.12 | tainted- | http://www.37signals.com/ |
11:56.35 | three55ml | Yep :) I'm actually redoing the site to match the look of the internal program. |
11:56.58 | tainted- | which site will the internal program match? |
11:57.31 | three55ml | The main site will match the look of the control panel - http://www.premierpbx.com/images/screenshots/2.jpg |
11:59.38 | h3x | is that supposed to be one of those nortel phones |
11:59.52 | h3x | well |
11:59.53 | h3x | aastra |
12:00.15 | tainted- | that looks a lot better than http://www.switchvox.com/sv?cmd=screenshots&pic_id=8 |
12:01.03 | three55ml | I think sticking with the Asterisk terminology and conventions is way too confusing to end-users. |
12:01.28 | tainted- | i agree |
12:01.31 | three55ml | I'm pondering dropping the term "Call route" all together in lieu of something simpler. |
12:01.35 | tainted- | your landing page it top notch |
12:01.42 | tainted- | very user centric |
12:01.47 | tainted- | s/it/is |
12:01.50 | file[laptop] | an hour late! |
12:02.01 | file[laptop] | stupid daylight savings |
12:02.02 | three55ml | There's also an admin panel on top of it all that does allow for more of a "raw" interface, but nothing the end-user needs to see. |
12:02.18 | three55ml | file[laptop]: Haha, you're a day off. |
12:02.47 | *** part/#asterisk RazaMetaL (razametal@pc.gsalas.manta.telconet.net) |
12:02.52 | file[laptop] | I'll miss my first period, but I can get there for the rest of the day |
12:03.02 | tainted- | first period!? |
12:03.05 | tainted- | how old are you! |
12:03.08 | file[laptop] | 18 |
12:03.15 | file[laptop] | :p |
12:03.16 | tainted- | jesus |
12:03.26 | three55ml | Man, I haven't even gone to bed yet. |
12:03.37 | three55ml | Figured I'd stay up, I have to go downtown and sign some papers at 10 anyways. |
12:03.59 | tainted- | you're going to be wasted in a few hours |
12:04.04 | tainted- | a walking zombie |
12:04.18 | file[laptop] | or I can like, not go to school today |
12:04.30 | file[laptop] | but that's not advised |
12:04.30 | three55ml | Possibly. I'm used to it, I usually play poker 1-2AM until 6-7. Hit up all the drunks. |
12:04.39 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
12:04.45 | *** join/#asterisk SkyTel_DK (~skytel@cpe.atm2-0-1041166.0x50a3c79a.albnxx14.customer.tele.dk) |
12:05.33 | tainted- | ok supermen |
12:05.50 | SkyTel_DK | can anyone help me on pri i get an error on my te410 card on one of the spans that say : Apr 4 14:03:23 NOTICE[-1250526288]: chan_zap.c:7379 pri_dchannel: PRI got event |
12:05.50 | SkyTel_DK | : HDLC Bad FCS (8) on Primary D-channel of span |
12:05.56 | tainted- | i'm off to dream about dialplans and channels in never never land |
12:06.24 | three55ml | Later |
12:06.26 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
12:06.35 | webman | SkyTel_DK: AFAIK, this is 'normal' if it happens rarely (ie, once every few hours/days) |
12:07.02 | SkyTel_DK | no i get it every 5-10 secs |
12:07.48 | SkyTel_DK | could it be bit errors on the e1 side ??? |
12:09.26 | webman | SkyTel_DK: well, I have/had the same problems, though I get it together with other error messages, and frequant dropped calls.... |
12:09.43 | SkyTel_DK | okey |
12:10.01 | webman | currently, I've been led to believe that it is due to a faulty 'timer' chip on the board... |
12:10.27 | webman | but then, I got my board when they were first released, one of the very first ones.... |
12:11.19 | *** join/#asterisk HoppaZ (~HoppaZ@xdsl-81-173-148-174.netcologne.de) |
12:11.23 | webman | hopefully tonight I will talk to digium and see if they'll still replace it for me.... (hope so, those things are expensive) |
12:11.24 | SkyTel_DK | hmmm just that i have it on 2 te410 cards on the same span IF i move the pri from the public side the the other card' |
12:11.25 | *** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
12:11.27 | Mother__ | greetings |
12:11.33 | HoppaZ | hi |
12:11.50 | webman | skytel_dk: are you sure you are taking timing from the right pri/span ? |
12:11.50 | Zeeek | hello Mother__ |
12:12.05 | Mother__ | question: just bought 15 g729 licenses off Digium, but they didn't ask for any MAC address, how does it work afterwards? |
12:12.07 | HoppaZ | got problems with h323 and asterisk... no sound with gnomemeeting |
12:12.08 | Mother__ | hiya Zeeek |
12:12.11 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
12:12.18 | SkyTel_DK | yes i take the timing from the rigth one |
12:12.30 | webman | mother__: just run the register, and it will get the MAC address the first time.... |
12:13.13 | florz | Do these licenses work with tap interfaces, too, BTW? |
12:13.14 | Zeeek | then just don't change the MAC ! |
12:13.19 | webman | skytel_dk: do you get the same problem if you run a crossover on the same card (ie span 1 <=> span 2) with no other PRI connected, and span 1 provides timing to span2 ?? |
12:13.32 | Mother__ | webman: OK, but how does it make sure I'm the licensee? I should stick some reference or serial or something somewhere no? |
12:13.48 | webman | zeek: you can change the MAC address once.... |
12:14.05 | SkyTel_DK | Zeeek if i do a cross over the error folows the e1 kabel |
12:14.08 | webman | mother__: you are emailed a registration key, which you need to give to the register program |
12:14.08 | Mother__ | yes, I read that you can re-register the MAC address once if you have to, but does it work on good faith? |
12:14.12 | Mother__ | AH! |
12:14.19 | Mother__ | webman: OK, that clears it up, thanks |
12:14.24 | SkyTel_DK | i have 2 * with te410 in it |
12:14.35 | SkyTel_DK | it = them |
12:15.02 | webman | mother__: AFAIK, it works automatically the first two times (with different MAC address) then after that you need to contact digium to discuss it... |
12:15.32 | webman | skytel_dk: what do you mean it follows the cable? |
12:16.40 | SkyTel_DK | lets say that cabel 1 is the one with errors on and cabel 2 is error free in the other asterisk with te410 in |
12:17.34 | SkyTel_DK | then if i take the cabel 2 and put it in the 1.st asterisk and the cabel 1 into the 2. |
12:17.40 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
12:18.02 | SkyTel_DK | then the 1.st asterisk is error free but the errors have moved to the 2.nd one |
12:18.34 | webman | skytel_dk: yes, but what I asked is, if you config one asterisk box, so that span 1 is pri_net and span 2 is pri_cpe, and connect them with a crossover cable, do you get this error? |
12:19.11 | SkyTel_DK | ahhhh |
12:19.18 | SkyTel_DK | i have not testet that |
12:19.20 | webman | skytel_dk: anyway, sounds like the error is in the cable, so check your physical cable from the telco to your asterisk, and then ask your telco to test the line |
12:19.36 | SkyTel_DK | yeah |
12:19.37 | *** join/#asterisk FireFoxIXI (~FireFox@floyd.gms.lu) |
12:19.47 | SkyTel_DK | it can be the cabel |
12:22.48 | *** join/#asterisk shodan (~shodan@216.113.99.220) |
12:23.53 | Mother__ | when an error follows a piece of hardware around, it's quite possible that the piece of hardware is the problem... |
12:24.32 | Mother__ | also, if you use a test tool that always reads failures on different used & new equipment, it may be the test tool has failed itself |
12:25.00 | Mother__ | I once saw a good number of VGA cards go to the crapper because they were testing them on the same faulty motherboard |
12:25.05 | Mother__ | dumbasses |
12:25.30 | Zeeek | Mother__ what about when an error follows a user around? :) |
12:25.53 | Mother__ | oh, that happens a lot, then I get my Rhyno and label the person as a (L)user |
12:25.58 | Mother__ | :) |
12:26.15 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
12:26.23 | SkyTel_DK | do any of you have any idea how i can get a dial tone using asterisk and my te410 card i have problems dialing over sea calls do to the fack that when i press 00 on my set to dial overseas the pabx sends the 2 leading 00 to the pri and i get a dial time out |
12:26.27 | Zeeek | have you ever watched frustrated users retype the same command after an error? |
12:26.34 | Mother__ | lol yeah |
12:26.41 | Mother__ | as if typing it nth times will make it work |
12:26.47 | Zeeek | then they type it in all caps to see... |
12:27.34 | Mother__ | SkyTel_DK: is your dialplan OK? I'd check that first - can you dial other numbers OK? |
12:27.44 | SkyTel_DK | all okey |
12:27.56 | Mother__ | so you can dial local numbers? |
12:28.21 | SkyTel_DK | i got it to work if i set overlapdialing to yes |
12:28.44 | SkyTel_DK | but that is not to rigth way to do it |
12:29.12 | *** join/#asterisk devi (~dev@gw.01063telecom.de) |
12:29.18 | devi | hello everyone |
12:30.06 | Mother__ | hmmm found this http://lists.digium.com/pipermail/asterisk-users/2003-July/015351.html |
12:30.54 | SkyTel_DK | the reason it works lokaly is that the pabx does not send any digits to the pri before the dialplan legth is reached but it sends the 00 to the pri only if it is international calles |
12:31.22 | Mother__ | hmmm but why is it doing that? it's gotta be something in the dialplan no? |
12:31.39 | SkyTel_DK | nope not in asterisk |
12:31.53 | SkyTel_DK | its done from the pabx |
12:31.56 | Mother__ | I use two context for international and local calls personally |
12:32.03 | SkyTel_DK | okey |
12:32.03 | Mother__ | AH, you have a sepparate PABX? |
12:32.08 | SkyTel_DK | yes |
12:32.16 | SkyTel_DK | an eads |
12:32.47 | SkyTel_DK | i lead it into the asterisk and then out to the publicexchange |
12:32.55 | Mother__ | righto |
12:32.58 | SkyTel_DK | on 2 spans |
12:33.41 | Mother__ | try to use two contexts with the two dial lengths |
12:33.45 | SkyTel_DK | so i use the asterisk as a pipe |
12:34.13 | Mother__ | in mine I have on which is 00XXXX...... and another for local/national numbers (all begin with 9) |
12:34.48 | Mother__ | it helps if you want to handle things sepparately at some stage, I like to keep things on sepparate contexts |
12:35.26 | SkyTel_DK | okey |
12:35.37 | SkyTel_DK | but in dk its |
12:35.55 | SkyTel_DK | ZXXXXXXX for dk |
12:36.40 | SkyTel_DK | and 00Z. for international |
12:37.19 | SkyTel_DK | but the pabx sends first the 00 then the rest of the digits |
12:37.25 | smurfix | anybody know their way around mISDN? my chan_misdn driver doesn't want to start |
12:38.03 | SkyTel_DK | it its not in overlap mode |
12:38.40 | SkyTel_DK | in overlay mode it sends the hole digit string at once |
12:39.34 | SkyTel_DK | but if for some reason the user sends a wrong number its takes a long time before it times out |
12:39.55 | Zeeek | check this when you have time to waste |
12:39.56 | Zeeek | http://www.msnbc.com/modules/airport_security/screener/default.asp |
12:40.02 | SkyTel_DK | but in te way that it sends first the 00 |
12:40.07 | *** join/#asterisk forkqueue (~sam@spc1-ward2-5-0-cust27.bagu.broadband.ntl.com) |
12:40.36 | Mother__ | OK, how long between it sends the 00 and the rest of the digits? |
12:40.47 | Mother__ | because maybe you'll need to change digit timeout in * |
12:40.48 | *** part/#asterisk FireFoxIXI (~FireFox@floyd.gms.lu) |
12:41.04 | SkyTel_DK | the publicexchange listens to the digits that are sendt 1 by one and test it for wrong nummer rigth away |
12:41.59 | Mother__ | well, that's the exchange's job, but I don't think * will transparently send digits between your PABX and the rest of the world as they are dialled |
12:42.00 | *** join/#asterisk jakepdev (~jakepdev@pool-68-163-51-71.phil.east.verizon.net) |
12:42.10 | Mother__ | I may be wrong there though, don't take it for granted |
12:42.37 | Mother__ | AFAIK when a dial string is complete and it matches an entry in extensions.conf then the extension sequence is executed |
12:43.03 | SkyTel_DK | yeah thats what i mean |
12:43.06 | *** join/#asterisk airios (~andres@66.28.87.10) |
12:43.29 | SkyTel_DK | cant you just put some thing in to halt the execution |
12:44.17 | Mother__ | in which case, if it's invalid, * will still dial it and the telco's exchange will send back a response, either tones or a nice voice telling you to rotate your phone 180 degrees and dial again |
12:44.20 | Mother__ | ;) |
12:44.33 | Zeeek | heh |
12:44.53 | Mother__ | so your PABX is sending the 00 then waiting for something? or it sends the 00 then pauses for x seconds then sends the rest? |
12:45.10 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
12:45.25 | SkyTel_DK | no it waits |
12:46.02 | Mother__ | so it waits for something to come back from the PRI saying "shoot, I'm ready?", then as * doesn't send anything it times out¿ |
12:46.18 | SkyTel_DK | its goes into call mode and waits for the rest of the digits 1 at a time |
12:46.29 | *** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net) |
12:47.00 | Mother__ | then maybe you could increase digit timeout in the * conf |
12:47.41 | Mother__ | I had to increase it on one I have installed as the (L)users took too long to dial... |
12:49.02 | SkyTel_DK | yes but that should not be nessesary since the pub-Xchange dials the b caller 1 digit at a time |
12:49.53 | SkyTel_DK | asterisk sould be able to send the 2 00's and then go into dtmf mode |
12:50.29 | Mother__ | dammit, gotta go |
12:50.31 | SkyTel_DK | but since it sends a "no more digits" |
12:50.33 | Mother__ | bbl |
12:50.38 | SkyTel_DK | cu |
12:51.07 | airios | iaxy question: how i look at the debug information in port 9999 ( when i set debug in the conf )? |
12:51.09 | SkyTel_DK | or end of digit string the pub_xchange hangs up |
12:53.28 | vaewynAFK | SkyTel_DK: You could have a exten => 00,.... and it should do something akin to that |
12:53.59 | SkyTel_DK | coma |
12:54.02 | SkyTel_DK | okey |
12:54.30 | vaewyn | the comma is just the delimeter |
12:55.09 | SkyTel_DK | so it is not the same as . "dot" |
12:55.28 | vaewyn | now bummer is ... if you do that you get all your cdr logs as calls going to '00' |
12:55.56 | vaewyn | sorry... to be more clear it would be exten => 00,1,Dosomething() |
12:56.10 | SkyTel_DK | aahhh |
12:56.30 | vaewyn | but it screws up your CDR on * if you are using that for billing |
12:56.56 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
12:57.15 | SkyTel_DK | that is testet i did 00,1,dial(zap/g2/00) but it then hangs up since there is sendt a end of digits |
12:57.38 | airios | anybody with experience using IAXY modems? |
12:58.05 | h3x | <PROTECTED> |
12:58.20 | bjohnson | SkyTel_DK: sounds like the problem stems from the pabx you're using |
12:58.28 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
13:00.00 | *** part/#asterisk HoppaZ (~HoppaZ@xdsl-81-173-148-174.netcologne.de) |
13:00.54 | SkyTel_DK | bjohnson i dont think so because when i put the pabx direktly into the Pub_Xge it works fine |
13:03.34 | *** join/#asterisk jcims (~jcims@cpe-24-210-60-100.columbus.res.rr.com) |
13:06.32 | dwmw2_gone | SkyTel_DK: that works for me when dialling out via mISDN or CAPI |
13:07.13 | dwmw2_gone | exten => _X!,1,Dial(mISDN/g:extern/${EXTEN}) |
13:10.56 | nesys | there's someone expert on sip that could help me? I've an issue about call-forward between ccme and * |
13:12.03 | *** join/#asterisk shodan (~shodan@216.113.99.223) |
13:15.30 | SkyTel_DK | hmmm |
13:15.40 | SkyTel_DK | what is mISDN |
13:15.40 | shido6 | :) |
13:16.54 | *** join/#asterisk klictel (~klictel@207.107.208.137) |
13:16.58 | klictel | morning all |
13:17.18 | shido6 | yep |
13:17.24 | *** join/#asterisk bsdfreak (ninja@enterthebass.com) |
13:18.17 | *** part/#asterisk jcims (~jcims@cpe-24-210-60-100.columbus.res.rr.com) |
13:20.28 | airios | h3x: yes, IAXY modems? |
13:20.50 | shido6 | ? |
13:20.55 | shido6 | what are you doing airios |
13:20.56 | shido6 | ? |
13:21.27 | airios | the question is how to fetch the debug info when the debug option is given in the config file? |
13:22.44 | airios | i see the pacets bing broadcasted in port 9999, how i parse them/see them? |
13:22.47 | *** join/#asterisk LoRez_ (lorez@lorez.staff.freenode) |
13:23.38 | shido6 | tcpdump, ethereal |
13:23.44 | shido6 | pick your poison |
13:23.59 | *** join/#asterisk malverian (~malverian@adsl-065-005-207-210.sip.gnv.bellsouth.net) |
13:24.45 | airios | yes ... i am using both, but aren't those packets menat for something that can parse them? |
13:25.02 | airios | meant |
13:25.10 | shido6 | what are you going to do with the info? |
13:25.14 | shido6 | did you misconfigure your IAXy |
13:25.15 | shido6 | ? |
13:25.51 | airios | i am trying to see what the modem is doing, as i am not sure if it register with my astersik server |
13:26.02 | shido6 | if the light isnt on |
13:26.04 | shido6 | it didnt |
13:26.33 | airios | which light? |
13:26.34 | shido6 | at the CLI if you do a iax2 show peers |
13:26.42 | shido6 | and dont see your IAXy there |
13:26.46 | shido6 | then the IAXy did not register |
13:26.50 | airios | dont see it with show peers |
13:26.55 | shido6 | then its not registered |
13:27.17 | shido6 | there are 2 lights on the IAXy ( well 4 if you count the link and xmit lights ) |
13:27.17 | airios | mmmm ... so that is the point to see the debug |
13:27.27 | shido6 | 1 of which stays on when the IAXy is registered |
13:27.33 | shido6 | ........nope |
13:27.40 | shido6 | configure your IAXy properly |
13:27.47 | shido6 | show me your iax.conf |
13:27.50 | shido6 | pastebin.ca |
13:28.18 | airios | i saw that in the man, i saw the link lights in the ether , and another light that seems to be ligthing on when the phone is off the hook |
13:28.24 | airios | is there another one>? |
13:28.24 | yaboo | anyone got the soyo n400s fxs unit yet? |
13:28.39 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
13:28.40 | shido6 | what do you have in iax.conf for your IAXY, use pastebin.ca to show us |
13:29.00 | dreamcode | does anyone use musiconhold ? |
13:29.14 | shido6 | whats wrong dreamcode ? |
13:29.19 | webman | anyone used the s100U (USB FXS) on 2.6.x kernel recently ?? |
13:29.32 | shido6 | heh |
13:29.35 | dreamcode | i don't hear any music when puting someone on hold |
13:29.40 | webman | I get a sig segv when running ztcfg ..... |
13:29.42 | airios | it is textual from the iaxy config pdf |
13:29.44 | shido6 | webman you got your hands on an old S100U, luck you |
13:29.53 | shido6 | airios, that doesnt help |
13:29.55 | shido6 | pastebin.ca |
13:30.23 | webman | shido6: well, I bought it a long time ago... I'm just re-setting it up at home (I upgraded to the TDM shortly after they were first released) |
13:30.41 | airios | ok ... dont have it now, will do. THANKS! |
13:30.51 | dreamcode | shido6, i have instaled mpg123 , MP3Player is working , but the MusicOnHold doesn seems to start.. why ? |
13:31.02 | shido6 | ps -aux |
13:31.04 | shido6 | no mpg123? |
13:31.09 | shido6 | wrong mpg123, perhaps |
13:31.09 | airios | one more quick question, there is a binary in the iaxyprov called iaxydebug or something, what is it? |
13:31.27 | shido6 | this is such a problem its even got its own install command now in CVS |
13:32.02 | webman | dreamcode: what is the CLI output when you try to put some1 on hold?? |
13:32.03 | SkyTel_DK | has anyone got the cisco 7970G ipset to work under asterisk ??? |
13:32.16 | shido6 | cd /usr/src/asterisk |
13:32.18 | shido6 | make mpg123 |
13:32.23 | shido6 | make install |
13:32.28 | shido6 | SkyTel_DK |
13:32.33 | shido6 | yes, my 7960 works well |
13:32.38 | shido6 | as do my students and customers |
13:32.44 | shido6 | whats happening with yours? |
13:33.09 | SkyTel_DK | i just got it |
13:33.33 | shido6 | so its got skinny on it? |
13:33.35 | shido6 | :) |
13:34.02 | webman | 7970 I think is skinny only ... |
13:34.16 | shido6 | yes, but 7960s new in the box |
13:34.19 | shido6 | come wit skinny |
13:34.22 | shido6 | +h |
13:34.26 | dreamcode | i don't have nothing on CLI |
13:34.41 | shido6 | go to /etc/asterisk/logger.conf |
13:34.45 | shido6 | and look for "console" |
13:34.50 | shido6 | then at the end ofo the line add , debug |
13:34.55 | *** join/#asterisk lbarth (~lbarth@62.4.65.13) |
13:35.08 | shido6 | and stop and restart asterisk with -vvvvvgcd or /usr/sbin/asterisk -vvvvvgcd |
13:35.16 | shido6 | or wherever you have asterisk installed |
13:35.19 | shido6 | and check again |
13:35.27 | shido6 | but more than likely you have the wrong mpg123 |
13:35.36 | shido6 | get mpg123 installed first |
13:35.56 | shido6 | how many phones do you have SkyTel_DK |
13:35.57 | shido6 | ? |
13:36.55 | SkyTel_DK | 1 |
13:36.58 | bjohnson | webman: the s100u won't work on anything but a 2.4 kernel .. and that is according to digium |
13:37.15 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
13:37.16 | shido6 | would you like me to upgrade your phone and get it working wit asterisk while you watch and ask questions? |
13:37.21 | shido6 | hhh |
13:37.35 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.87) |
13:38.30 | Moc____ | mornuing |
13:40.20 | ariel_ | morning all |
13:43.31 | dreamcode | shido6, but.. if MP3Player command works in extensions.conf , shouldn't musiconhold work too ? |
13:44.00 | shido6 | grrrrrr....... |
13:44.09 | shido6 | what does /etc/asterisk/musiconhold.conf say? |
13:44.11 | shido6 | pastebin.ca |
13:44.19 | *** join/#asterisk markak2 (~twist@ndn-165-134-119.telkomadsl.co.za) |
13:44.23 | markak2 | hi all |
13:46.35 | dreamcode | http://pastebin.ca/8772 |
13:46.37 | markak2 | something strange has happened on my asterisk. for some reason this was working but now just gives an open line. it used to remember the ${EXTEN} and dial the number correctly. this is a part of that que for new calls i was working on. |
13:46.39 | markak2 | [dialnow] |
13:46.39 | markak2 | exten => s,1,ChanIsAvail(Zap/2&Zap/1) |
13:46.39 | markak2 | exten => s,2,Cut(theChannel=AVAILCHAN,,1) |
13:46.40 | markak2 | exten => s,3,Dial(Zap/g1/${EXTEN:1},20,Ttm) |
13:46.40 | markak2 | exten => s,4,Hangup |
13:46.42 | markak2 | exten => s,102,Playback(allbusy) |
13:46.44 | markak2 | exten => s,103,Wait(5) |
13:46.46 | markak2 | exten => s,104,Goto(dialnow,s,1) |
13:46.54 | newl | ahh, shido6, just the person that might have an answer to a perplexing question. Scenario is this, two internal extension X and Y, extension X is diverted, extension Y calls extension X which diverts, CDR for the diversion gets entered for Y, not X. Bug, or feature? If bug, I can file a report. If feature, how to make X take the CDR entry instead of Y? :) |
13:47.00 | shido6 | holy paste |
13:47.02 | newl | markak2: please use pastebin. |
13:47.09 | markak2 | sorry |
13:47.14 | shido6 | dood |
13:47.17 | nvrswork | that guy is out of control |
13:47.17 | shido6 | s? |
13:47.33 | shido6 | what are you cutting off the exten line with the exten:1 |
13:47.36 | shido6 | Ttm ? |
13:47.52 | shido6 | careful as both the called and the calling can xfer if that ever worked which I doubt.. |
13:47.57 | markak2 | supposedly the preceding 9 they dial to get here. |
13:48.07 | shido6 | what is that supposed to do markak2, what are you LOOKING to do ? |
13:48.17 | markak2 | exten => _9.,1,Goto(dialnow,s,1) |
13:48.26 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
13:48.26 | *** mode/#asterisk [+o bkw_] by ChanServ |
13:48.30 | markak2 | we have 2 pstn lines |
13:48.50 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
13:48.53 | markak2 | but too many people using them. if you dial and all lines are busy it notifies you and then connects you when a line is open. |
13:49.19 | shido6 | dreamcode |
13:49.24 | shido6 | dreamcode, http://pastebin.ca/8773 |
13:49.26 | markak2 | but 2 days ago instead of connecting you to the number you dialed it just connects you to an open pstn line. |
13:49.29 | langals | join #freenode |
13:49.41 | *** part/#asterisk shodan (~shodan@216.113.99.223) |
13:49.44 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
13:50.09 | shido6 | what is calling these s extens in dialnow |
13:50.31 | shido6 | oh ok |
13:50.34 | shido6 | exten => _9.,1,Goto(dialnow,s,1) |
13:50.35 | markak2 | exten => _9.,1,Goto(dialnow,s,1) |
13:50.40 | markak2 | yes |
13:51.01 | nesys | switch works only between * servers? |
13:51.06 | markak2 | the only problem is that it seems to lose the dialed number. |
13:52.00 | shido6 | thats actually pretty cool. |
13:52.10 | shido6 | whats the CLI say? |
13:52.26 | dwmw2_gone | SkyTel_DK: mISDN is just another way of using ISDN BRI cards. |
13:52.32 | shido6 | what has changed in the past 2 days? |
13:52.42 | shido6 | what could have changed in the past 2 days? |
13:53.23 | markak2 | strangely nothing i can remember doing |
13:53.59 | markak2 | http://www.pastebin.com/266954 |
13:54.04 | markak2 | all normal just loses the number |
13:54.55 | markak2 | this was a call when a line is open and it connects me immidiately |
13:55.00 | dreamcode | yes, shido6 , still doesn't work,my mpg123 is Version 0.59s-mh4 (2000/Oct/27), and doesn't work, and what is strange.. that i don't have any message on console when i put someone on hold |
13:55.15 | shido6 | ok |
13:55.16 | shido6 | -- Executing Dial("IAX2/2001@2001-6", "Zap/g1/|20|Ttm") in new stack |
13:55.16 | shido6 | <PROTECTED> |
13:55.21 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
13:55.26 | shido6 | what the heck are you "calling" here? |
13:56.25 | markak2 | Zap/g1 is the group for the ZAP channels it should be follwed by the number ${EXTEN:1} |
13:56.34 | shido6 | err |
13:56.38 | shido6 | thats not in the CLI |
13:56.51 | markak2 | i know |
13:57.01 | markak2 | its getting lost somewhere ? |
13:57.23 | markak2 | exten => s,3,Dial(Zap/g1/${EXTEN:1},20,Ttm) this is the line |
13:58.32 | shido6 | zapata.conf shows what group=1 is , right? |
13:59.03 | mishehu | does anybody know why using delayreject=yes in iax.conf would prevent a trunked call from being created? Inbound calls work fine if delayreject=yes and there is no trunk=yes defined for the user, but the instant trunk=yes is defined, after AUTHREQ an INVAL is given... |
13:59.12 | tzanger | mishehu: it's a bug I think |
13:59.15 | tzanger | I had that exact same problem |
13:59.24 | tzanger | it's an OLD bug too |
13:59.29 | bjohnson | markak2: I'm surprised it ever worked .. how is the EXTEN getting to that macro? You're not feeding it as an arg |
13:59.30 | mishehu | tzanger: thats what I think too. and that's what turned out to be my problem yesterday. |
13:59.36 | tzanger | ahh |
13:59.37 | bjohnson | also EXTEN won't survive a goto() |
13:59.50 | tzanger | bjohnson: it won't? |
13:59.56 | tzanger | Goto(Context,${EXTEN},1) |
14:00.15 | mishehu | tzanger: the instant I commend out delayreject, I could make trunked calls again, with plaintext or rsa auth. |
14:00.18 | bjohnson | that is feeding it |
14:00.24 | mishehu | s/commend/comment |
14:00.30 | tzanger | bjohnson: you didn't say that was a bad thing |
14:00.38 | bjohnson | tzanger: he only has an s exten and trying to use the EXTEN variable |
14:00.40 | tzanger | how do you expect exten to survive otherwise? |
14:00.51 | tzanger | wont' work because s *is* the extension |
14:00.53 | bjohnson | that is exactly my point |
14:01.00 | tzanger | bjohnson: ahh |
14:01.04 | tzanger | well now you have some backup. :-) |
14:01.35 | *** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au) |
14:01.38 | shido6 | either EXTEN or... |
14:01.44 | shido6 | use ${ARG1}'s |
14:01.53 | *** part/#asterisk lbarth (~lbarth@62.4.65.13) |
14:02.22 | *** join/#asterisk dalabera (~Dalabera@mail2.pmrtechnologies.com) |
14:06.27 | Aze` | How use txfax ?? |
14:08.18 | Hmmhesays | heh, php has a compiler now, interesting |
14:09.07 | mishehu | a bytecode compiler. |
14:09.08 | bjohnson | markak2: the dial line is ok .. it's the EXTEN that is missing |
14:09.10 | mishehu | never used it. |
14:09.14 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
14:09.36 | bjohnson | markak2: look at the superdal macro on the wiki for an example of how to pass the EXTEN to a macro to dial |
14:09.49 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
14:10.49 | malverian | I'm setting up a outgoing context for dialing a number and using Festival to read a message to the person who picks up the phone. I'm wondering if there is some way that the file in /var/spool/asterisk/outgoing can somehow define the message that should be spoken? |
14:10.51 | *** join/#asterisk MattB2 (~mattb@pcp01068561pcs.andrsn01.tn.comcast.net) |
14:11.05 | malverian | Because as of right now I just have a Festival('......') hardcoded into the context. |
14:11.40 | *** join/#asterisk jpe (~jpe@www.brooklynbassmint.com) |
14:12.00 | Faithful | Hi guys! |
14:12.47 | malverian | I'm not sure if I'm being clear. |
14:12.55 | *** join/#asterisk VirTERM (~virterm@shiva.kanatek.com) |
14:13.01 | VirTERM | morning |
14:13.20 | VirTERM | res_sms what's that? |
14:13.41 | malverian | In less cryptic terms, I want to have asterisk dial a number and use festival to send a dynamic message. |
14:14.49 | forkqueue | malverian: Is the message 'Congratulations, you have won a prize'? |
14:15.17 | ariel_ | does anyone here have the latest firmware for the sayson 480i phone that you can let me have? |
14:15.59 | malverian | forkqueue: No, it's more along the lines of "Warning, the website is down" :-P |
14:16.12 | malverian | forkqueue: I'm replacing voiceshot. And using it in our nagios server. |
14:16.12 | forkqueue | malverian: Heh :) |
14:16.36 | malverian | If something superbad happens, I don't mind if it calls people's cell phones instead of just paging me ;) |
14:17.04 | forkqueue | malverian: That's not a bad idea actually :) |
14:17.28 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
14:17.38 | malverian | forkqueue: It's a damn good idea.. I think I'm going to have to write my own agi though.. |
14:17.43 | malverian | From the way things look. |
14:17.57 | malverian | Unless I want to restart asterisk and edit my extensions.conf every time. |
14:18.02 | malverian | (not an option) :-P |
14:18.19 | forkqueue | malverian: Yeah, I think a custom-written AGI would be necessary, but it needn't be the worlds most complicated script or anything.. |
14:18.46 | malverian | I just need to learn how to write them.. shouldn't be too awfully difficult I imagine. |
14:19.00 | malverian | Going to look at a few of the example ones. |
14:19.29 | forkqueue | Let me know how you get on, it's a good enough idea that I might do it myself :) |
14:19.43 | malverian | Oh, SWEET! |
14:20.07 | malverian | It just pipes it to the program, so you can write it in any language.. perfect. |
14:21.00 | BuckRogers | good morning |
14:21.23 | *** join/#asterisk Slainte (~Slainte@66.55.112.85.ppp.northrock.bm) |
14:21.30 | pif | hi, is there a way to test if a SIP phone is available before ringing it? |
14:21.41 | malverian | Hmm.. |
14:21.53 | bkw_ | <rant> |
14:22.06 | bkw_ | Ok it pisses me off to have to jump thru hoops to buy a product from a company. |
14:22.09 | bkw_ | </rant> |
14:22.47 | tzanger | bkw_: so don't buy from them, and let them know why you're not buying from them |
14:23.10 | bkw_ | the problem is a lot of companies do this |
14:23.12 | bkw_ | its pure bullshit |
14:23.26 | bkw_ | I don't wanna have to contact them.. go to a distributer |
14:23.28 | *** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
14:23.31 | Slainte | I want a user to be prompted to enter a three digit billing code, if they dial a longdistance number. Can someone take a peak at http://pastebin.ca/8774 and see if there is anything obivously wrong with my plan? |
14:23.34 | cypromis | lazy bunny |
14:23.41 | bkw_ | then find they don't have a website that I can buy from.. nor do they return phone calls |
14:24.19 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
14:26.14 | tzanger | ahh |
14:26.16 | tzanger | get it off, get it off |
14:26.52 | tzanger | bkw_: what are you trying to buy? |
14:27.56 | kram | get what off? |
14:27.57 | bjohnson | pif: dial will tell you if it is unavaiable |
14:28.11 | kram | were these starving crazed weasels? |
14:28.13 | bjohnson | a spider |
14:28.14 | tzanger | kram: hahah I was gonna say 'you' but now that's just not right |
14:28.16 | kram | ah |
14:28.34 | kram | tzanger: You're Not Right(TM) |
14:28.48 | tzanger | kram: this is true |
14:29.06 | tzanger | kram: I emailled greg, hopefully he can find some time in your schedule |
14:29.07 | bjohnson | Slainte: no .. nothing obvious. Does the noop show the right info? |
14:29.17 | tzanger | I'd invite you over but I've not got a hot tub |
14:29.27 | tzanger | I have a bathtub I can put hot water in but I don't think it's the same |
14:29.35 | kram | tzanger: hopefully, if you don't hear from him in the next couple of days, let me know and i'll try to get it figured out |
14:29.40 | kram | tzanger: alas, no, it's not :) |
14:29.40 | Slainte | I have not tried it yet, as it is production. I wanted to make sure that it had a high chance of success before putting it in :) |
14:29.41 | tzanger | ok |
14:29.46 | Slainte | my test system is down |
14:30.01 | Slainte | bjohnson, you think I did it the correct way? |
14:30.27 | bjohnson | Slainte: put it in as a test exten so you dial the exten to test it |
14:30.46 | bjohnson | looks ok from the parts I see |
14:31.13 | Slainte | Thanks, if it works I will add it to the Wiki for SetAccount because I could find nothing like it there. |
14:31.19 | bkw_ | tzanger, you wanna know? |
14:31.26 | tzanger | bkw_: yeah |
14:31.34 | tzanger | damn it takes forever to copy 1.2G over USB1 |
14:31.56 | *** part/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
14:33.28 | *** join/#asterisk iq (~iq@70-59-163-239.omah.qwest.net) |
14:35.00 | bjohnson | Slainte: look at the superdial macro in the tips and tricks section |
14:35.57 | Slainte | bjohnson, I looked at it but still could not figure out how to get a user prompted request to enter a code, and then set the variable based on what the user entered |
14:36.35 | malverian | forkqueue: Looks like the easiest way is just using SET VARIABLE to set a message variable and using that in the dialplan itself. |
14:36.44 | *** join/#asterisk Darwin35 (~Darin@24.3.226.147) |
14:36.46 | malverian | forkqueue: Worked like a charm :) |
14:37.33 | shido6 | Slainte |
14:37.34 | forkqueue | malverian: Care to document it on the wiki? I'm sure there are plenty of people that might like to have Nagios phone and tell them what the problem is.. |
14:37.36 | shido6 | dont freak out |
14:37.40 | shido6 | its not difficult |
14:38.18 | shido6 | we've been using nagio since it was called netsaint |
14:38.19 | shido6 | :) |
14:38.22 | shido6 | nagios |
14:38.33 | forkqueue | I used to use SNIPS |
14:38.38 | forkqueue | But nagios is loads nicer |
14:38.55 | shido6 | back when I was at global crossing we used it there, too |
14:38.59 | *** join/#asterisk pluto70 (~god@80.70.179.70) |
14:39.15 | forkqueue | Heh, gblx? |
14:39.22 | forkqueue | Did they go bankrupt? |
14:39.30 | *** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
14:39.58 | shido6 | they filed for protection |
14:40.02 | shido6 | and laid off a ton of workes |
14:40.04 | shido6 | workers |
14:40.07 | shido6 | *grumble* |
14:40.22 | shido6 | then laid off a ton more |
14:40.41 | shido6 | now Im a carrier and GX can kiss my black ass |
14:40.55 | shido6 | at a medium pace |
14:42.56 | malverian | forkqueue: I may actually do that. |
14:43.55 | sivana | can anyone here recommend a good spam smtp proxy? |
14:43.56 | Slainte | shido, didn't XO buy their NorthAmerican assets for chump change? |
14:44.01 | sivana | anti-spam |
14:44.13 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
14:44.44 | *** join/#asterisk marno (~marno@213-182-114-20.teleos-web.de) |
14:44.52 | ariel_ | spamassasin |
14:45.00 | sivana | for windows? |
14:45.47 | sivana | actually.. does that install before the mail server? |
14:46.16 | bjohnson | doesn't MS have one you can buy? |
14:46.22 | sivana | ya.. right |
14:46.45 | bjohnson | we run linux on our mail server and use spamassassin with procmail |
14:47.20 | sivana | don't worry.. I'm teetering on the fence.. OS has its own problems |
14:47.52 | ariel_ | sivana, for windows server I have been using IpSwitch imail it comes with a good spam blocker |
14:48.08 | sivana | I'm using a NoSpamToday system that's suppose to be based on SpamAssassin, it uses tokens? |
14:48.58 | *** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net) |
14:49.08 | ariel_ | I was reading that CamAV has released a version for Windows. Maybe there is a version of spamassasin for windows out there? |
14:50.03 | sivana | ariel_: I'm pretty sure these other folks placed a win32 GUI on the SA engine |
14:50.13 | blitzrage | anyone know what that toolkit is so that you can popup screens from the taskbar in Windows? (like the Gmail email notifier program) |
14:50.15 | malverian | What's the recommended codec to use? |
14:50.21 | focks | 200/200 10.10.10.60 D 255.255.255.255 5060 Unmonitored |
14:50.36 | focks | that shows a valid registered SIP extenstion no? |
14:50.41 | blitzrage | yes |
14:50.43 | ariel_ | blitzrage, if your using IE msn has a free popup blocker |
14:50.51 | nesys | focks try quality=yes :) |
14:51.00 | forkqueue | malverian: That very much depends on your situation :) |
14:51.07 | eKo1 | Man, some of these debug messages are just plain weird. |
14:51.13 | focks | nesys can you tell me why? |
14:51.16 | blitzrage | focks: if you see IP, then prettty much yes. qualify=yes will show you the latency where Unmonitored is |
14:51.19 | malverian | forkqueue: Well, my sony ericsson mobile phone is giving me issues on my pbx. |
14:51.21 | eKo1 | e.g. Oooh, something is weird, backing out |
14:51.33 | focks | blitzrage ahh |
14:51.36 | forkqueue | malverian: If you've got the bandwidth, use ulaw or alaw |
14:51.38 | *** join/#asterisk jpe (~jpe@66.114.77.37) |
14:51.47 | malverian | forkqueue: This is for analog. |
14:52.03 | blitzrage | ariel_: not what I mean. You can install a little program and send info to it to make it popup a cool little window in the taskbar. Look at the gmail notifier program for an example of what I mean. |
14:52.04 | malverian | forkqueue: When I dial in from my cell phone, it doesn't recognize keypad presses correctly sometimes. |
14:52.10 | malverian | But works fine from a land line. |
14:52.19 | blitzrage | malverian: ulaw |
14:52.28 | *** join/#asterisk CoderCR (~creyna@ip68-6-244-85.sd.sd.cox.net) |
14:52.29 | CoderCR | hello all |
14:52.34 | focks | blitzrage i'm still having this pesky Call Not Allowed issue |
14:52.49 | focks | blitzrage just been playing around with *@Home |
14:53.04 | jpe | <PROTECTED> |
14:53.13 | webman | anyone here know anything about the zaptel kernel modules? I'm trying to 'fix' the wcusb driver... |
14:53.35 | jpe | I beleive I have everything set up correctly, and I can see the calls coming in on the console, but get rejected with no authorization |
14:53.49 | jontow | webman; your best bet is to actually ask a question with meat to it so someone will pay attention ;) |
14:54.15 | webman | basically this line fails: if (usb_submit_urb(&p->datawrite[x].urb, GFP_KERNEL)) |
14:54.24 | jpe | I have the inbound contexts set up correctly for each sixTel and voicepulse-in-01, and the dids set up in amp |
14:54.26 | ariel_ | for help with Asterisk@home for the AMP use there is a location here for that. #amportal |
14:54.42 | webman | with -EINVAL which I assume is a) Invalid transfer type specified (or not supported) |
14:54.45 | malverian | blitzrage: Hmm.. is that an option I can set in my zapatta.conf ? |
14:54.50 | jpe | thanks for the #amportal info |
14:54.51 | dalabera | guys on my cdr reports appears IAX2/gw1@xx.xx.xx.163:21 for my inbound calls, isn't suppose to appear only IAX2/gw1 since it's configure in iax.conf? |
14:55.07 | webman | or less likely, but possible: b) Invalid interrupt interval (0$<=$n$<$256) |
14:56.29 | jpe | I don't quite think it is an amp problem as I had it working before before i went to 0.8, it gave the same trouble in 0.6 and I did something simple and stupid to get it working |
14:57.03 | shido6 | try configureing the conf files manually as amp needs a lot of improvement |
14:57.24 | focks | jpe did you run into any access problems with SIP clients? |
14:58.40 | bjohnson | eww .. editing AMP conf files by hand is not for the type of person that typically wants to use AMP |
14:58.44 | jpe | no, everything else is working fine with the system |
14:58.46 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:58.46 | *** mode/#asterisk [+o anthm] by ChanServ |
14:58.57 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released |
14:59.36 | jpe | When I started with *, I did everything by hand and had a nice little thing going, then I came across aah and have been using it |
15:00.26 | jpe | all is working well with zaps and outbound |
15:00.45 | *** join/#asterisk nDuff (~cduffy@64.128.31.220) |
15:00.46 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th |
15:02.03 | Slainte | GotoIf($["${accountcode}" != "???"]?3) Is this valid? Will it require a three digit number? |
15:02.10 | nesys | I've a problem with call forward between cisco ccme and *: |
15:02.16 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
15:02.17 | jpe | the only other proble that I have not tackled yet is the sipura 2000 not reckognizing *xx stuff (*69 etc) , flash hook transfer to call waiting and things like that, that is next on my list |
15:02.21 | nesys | http://www.nesys.it/sipwork.txt is ok |
15:02.30 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
15:02.32 | nesys | http://www.nesys.it/sipdnwork.txt doesn't work :( |
15:04.18 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
15:04.40 | *** join/#asterisk ikey (ikey@220.226.28.82) |
15:04.46 | *** join/#asterisk wmoran (~wmoran@pa-plum-cmts1e-68-68-113-64.pittpa.adelphia.net) |
15:05.06 | wmoran | This is probably an old, tired subject, but ... |
15:05.11 | wmoran | who wants to talk about echo problems? |
15:05.27 | Slainte | wmoran, Ok lets talk because I am having issues |
15:05.41 | wmoran | We got most of ours fixed ... |
15:05.57 | Slainte | good, I have not :) |
15:06.00 | wmoran | The biggest problem we have now is the _other_ end hearing echo ... |
15:06.07 | wmoran | We can't seem to do anything about that. |
15:06.28 | Slainte | Did you have incoming echo issues? |
15:06.29 | nDuff | My system is trying to dial 7 digits in to an international number. Presumably it's not recognizing them correctly. There's a "international_pstn" section in extensions.conf that specifies "exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})", which looks about right -- but it doesn't appear to be referenced from anywhere else. Should it be? Am I on the right path in debugging this? |
15:06.57 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
15:07.19 | ManxPower | nDuff: ${TRUNK} and ${TRUNKMSD} are variables that you have to set in [global] |
15:07.20 | Slainte | NDuff, make sure you have the international_pstn context included in your internal dial plan |
15:07.20 | wmoran | There's a HOWTO on how to properly set your rxgain and txgain ... that helped a LOT |
15:07.54 | wmoran | Of course, if you're using PSTN ... that doesn't help if you're 100% VoIP |
15:08.27 | nDuff | Slainte, where would that be that I should include it? [default] |
15:08.45 | nDuff | s/$/?/ |
15:08.47 | Slainte | nDuff, default is fine, for now. Make sure people dialing in cant then dial back out |
15:08.56 | Slainte | wmoran, we have a PRI |
15:09.47 | *** join/#asterisk loko-w- (~rbrown@host1.vocollect.com) |
15:10.09 | Slainte | so I should not have cross talk |
15:11.00 | Slainte | But certain places we call always have echo, and other places not. |
15:11.10 | tzanger | Slainte: all that PRI means is that you do not GENERATE echo |
15:11.30 | wmoran | Yeah, if my brother calls the office from his house, the echo on his end is almost unbearable |
15:11.34 | Slainte | yes, I know that. Thats what I was referring by "cross talk" |
15:11.40 | wmoran | But he never gets echo when he calls anywhere else |
15:11.44 | wmoran | It's difficult to track down. |
15:12.35 | malverian | I got it set up so if I press 1, I can reschedule the next check of the service for 15 minutes later ;) |
15:13.00 | Slainte | malverian, is that event handling in Nagios? |
15:13.25 | malverian | I'm just using the command queue file directly.. there might be a more "proper" way to do it. |
15:13.40 | Slainte | yeah, but that will work. |
15:13.49 | *** part/#asterisk CoderCR (~creyna@ip68-6-244-85.sd.sd.cox.net) |
15:13.52 | malverian | I know, it -does- work, I just tested it :-P |
15:13.54 | Slainte | I have mine setup with SMS back and forth |
15:14.00 | malverian | Yeah, have had that set up for a while. |
15:14.24 | Slainte | well my daddy can beat up your daddy, and my daddys dead. |
15:14.26 | malverian | I'm doing it via email though.. |
15:14.28 | *** join/#asterisk Corydon-w (cinnamon@vcchgate.vcch01.springfield.tn.us.vcch.net) |
15:14.35 | malverian | So there is probably a better way. |
15:14.56 | malverian | eg.. 3492834@cingular.net (etc) |
15:14.59 | Slainte | so you email to SMS< and the SMS emails back? I have mine setup with direct SMS out, incase the internet or mail gateway goes down |
15:15.03 | malverian | Which is a problem if our mail server goes down ;) |
15:15.37 | malverian | Slainte: Hmm.. how hard is that to set up for direct sms? And do you have to do something special with your phone service provider? |
15:15.39 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
15:16.33 | Slainte | malverian, very easy, I bought a special siemens little box, that takes a sim card, has a long antenna, and it sits like a cell phone. You then plug it into your serial port, |
15:16.51 | malverian | Hmm.. neat, didn't know such devices existed :) |
15:17.04 | Slainte | First one I set up like that was in 2000, |
15:17.17 | Slainte | back in the Netsaint days |
15:17.17 | malverian | $$ ? |
15:17.35 | Slainte | 150 bucks or so. |
15:17.39 | malverian | Wow, not bad. |
15:17.43 | Slainte | its a phone without any screen or number pad |
15:18.04 | malverian | And you operate this little jewel through asterisk or some other software? |
15:18.24 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
15:18.34 | Slainte | nope, through a small app, that listens to the serial port, creates a queue, and then responds. |
15:18.38 | Slainte | very much like an MTA |
15:19.13 | Mimmus | Hi, I'm trying to receive fax by chan_chapi, using latest CVS with all patches but I get a coredump |
15:19.25 | malverian | Slainte: qpage? |
15:19.33 | Slainte | no, but very similar |
15:19.40 | Slainte | I use qpage as well |
15:19.50 | malverian | I use those for my arch pagers. |
15:19.53 | malverian | (qpage) |
15:19.59 | malverian | I just noticed it had SMS support. |
15:22.17 | ChkDigit | Is there a project to add Skype "channels" to asterisk? |
15:22.18 | Wonka | i hope not |
15:22.29 | Wonka | it would encourage people to use that closed-source stuff |
15:22.42 | *** join/#asterisk jaiger (~jaiger@fire.innovationsw.com) |
15:23.02 | Mimmus | any help with Asterisk coredumping? |
15:23.51 | Nugget | dunno, maybe you should ask in #asterisk. |
15:24.11 | Mimmus | :-) |
15:24.35 | Nugget | core dump files are a black art. I'd help if I could, but I have to settle for being unhelpful. |
15:24.36 | *** join/#asterisk h4mm3r` (~h4mm3r@213-140-17-106.fastres.net) |
15:25.04 | Mimmus | Nugget: thanks, asterisk coredumps when receive a fax |
15:25.07 | Nugget | probably not a bad idea to not load any modules you don't need, just to minimize the potential for issues. |
15:25.18 | Nugget | strange |
15:25.33 | Mimmus | Nugget: it dies just after capiAnswerFax |
15:25.34 | Nugget | is that with spandsp or something, or just the mere fact that faxes are talking on a normal channel that does it? |
15:25.37 | Nugget | ah |
15:25.43 | Nugget | no clue, sorry |
15:25.58 | Mimmus | Nugget: (I'm using chan_capi) |
15:28.05 | *** join/#asterisk Corydon76-home (three@pcp08665860pcs.500ash01.tn.comcast.net) |
15:30.09 | *** join/#asterisk umb (~java@adsl203-158-089.mclink.it) |
15:30.11 | umb | hi |
15:30.43 | umb | wow, this channel is much much more popular since I last visited |
15:31.45 | java_ | frankly I have a lame question |
15:32.00 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
15:32.04 | jontow | what the hell.. ZapRAS() ? :o |
15:32.17 | java_ | eheh even worse |
15:32.39 | java_ | how can I download a sip image for a 7905? |
15:32.48 | bublbobl | Hello all, sorry for my question (unrelated to *) is there a newbiez channel on freenode ? :-$ |
15:32.56 | nDuff | Anyone have an international [relative to the US] number I can use for testing purposes? |
15:33.08 | java_ | nDuff: i do |
15:33.33 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
15:34.55 | java_ | anyhow, lame question=no answer, nothing to argue |
15:36.48 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
15:36.51 | Mother_ | greetings |
15:36.54 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
15:36.57 | Mother_ | professor Falken... |
15:37.08 | java_ | these poor 7905s keep belinking |
15:37.33 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
15:37.49 | PBXtech | can you send a CLI command from the extension.conf file? |
15:38.19 | Mother_ | anyone here has successfully configured the RTP ports on a Cisco 7960? |
15:38.30 | Mother_ | no matter what I do they keep going back to default |
15:38.45 | Mother_ | be it configured from the phone or via tftp |
15:39.04 | JunK-W | PBXtech: extension.conf is a text file, how ya want to send a CLI command? |
15:39.05 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
15:39.35 | PBXtech | i just want to destroy a meet-me bridge from the extension.conf thats my goal |
15:40.07 | bjohnson | PBXtech: system() |
15:40.26 | PBXtech | so system(MeetMe kick <confno>) will work? |
15:40.34 | Corydon-w | So write an app to destroy a meetme bridge |
15:40.52 | bjohnson | PBXtech: no .. system runs a system command |
15:41.27 | bjohnson | PBXtech: you could likely do something like asterisk -rx 'MeetMe kick <confno>' |
15:41.41 | PBXtech | yea ok |
15:41.51 | PBXtech | thx |
15:42.37 | Mother_ | so I'm out of luck with this Cisco... |
15:42.39 | Mother_ | dammit |
15:43.10 | *** join/#asterisk enots (dimka@freelsd.net) |
15:43.26 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
15:45.42 | *** join/#asterisk ikey (ikey@220.226.31.180) |
15:46.07 | *** join/#asterisk ckruetze (~nospam@i3ED63FCC.versanet.de) |
15:47.14 | java_ | anyone have a 7905 sip image to pass me? |
15:48.13 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
15:49.27 | *** join/#asterisk fugitivo (~ajf@201.255.105.150) |
15:49.40 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
15:49.59 | *** join/#asterisk klictel (~klictel@207.107.208.137) |
15:50.05 | klictel | hello all |
15:50.20 | *** join/#asterisk dave_mwi (~dave_mwi@harpo.dreamhost.com) |
15:51.18 | *** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl) |
15:51.25 | *** join/#asterisk gpled (~gpled@firewall.fccfurn.com) |
15:51.55 | Lee__ | what's the command line command to show what codec a current channel is using? |
15:52.48 | Lee__ | java_: you have to get one from Cisco with their service contract |
15:53.15 | java_ | ok, this is the official answer, where's the dirty one? ;) |
15:53.39 | gpled | has anyone seen any docs on using asterisk as a deleayed paging system? |
15:53.48 | Hmmhesays | delayed paging? |
15:55.06 | *** join/#asterisk phpkid (~phpkid@adsl-068-153-207-210.sip.bct.bellsouth.net) |
15:55.09 | loko-w- | does anyone know the truth behind the livevoip screw up |
15:55.23 | Mother_ | Lee__: sip show channel x |
15:55.30 | Lee__ | thanks |
15:55.31 | Mother_ | where x is the call ID |
15:55.42 | Mother_ | you can get the call IDs from sip show channels |
15:56.11 | java_ | PSALLOC=early |
15:56.35 | Lee__ | wait, this is for an IAX device |
15:56.44 | Mother_ | iax2 show channels |
15:58.08 | Lee__ | the format field is "unknown" |
15:59.10 | Lee__ | ah, it only shows the codec for a two-way connection, not just dialtone. |
16:00.04 | Lee__ | I was talking to a stranger and they asked me if something was wrong with my phone cause the sounds was dropping out, so I'm thinking I should start tweaking codecs |
16:01.00 | robl^ | there is no codec for dialtone |
16:01.08 | robl^ | dialtone is generated by the phone itself |
16:02.11 | java_ | I have a question about the 7905 sip image, can somebody (willing to help) query me? |
16:02.42 | *** join/#asterisk The_Duke (~the_duke@80.92.64.103) |
16:03.07 | Mimmus | I have a problem with Asterisk coredumping on capiAnswerFax, can somebody help me? |
16:03.33 | The_Duke | hello does someone have some experience with the Junghanns Cards configured in NT/Network modus??? |
16:03.57 | *** join/#asterisk |nix (~inix@cm240.gamma116.maxonline.com.sg) |
16:04.10 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
16:04.33 | The_Duke | I'm missing a setting, I cannot remember it. I have ISDN Phones connected to the Junghans Card, as soon as I go off-hook, i get : == Primary D-Channel on span 3 up for TEI 64 |
16:04.33 | The_Duke | <PROTECTED> |
16:04.56 | The_Duke | how do I make asterisk wait for digits to be dialled??? |
16:05.12 | |nix | guys, did CVS head copy changed voicemail extensions from .WAV to .wav49? |
16:06.49 | webman | The_Duke: change immediate=no in /etc/asterisk/zapata.conf |
16:07.14 | webman | The_Duke: I mean, for that channel, change it to immediate=no ... |
16:07.29 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
16:07.34 | sudhir492 | Hi all |
16:07.42 | cypromis | or put an exten => s,1,digittimeout(5) |
16:07.47 | cypromis | in the context where the phones land in |
16:07.52 | webman | ooops... capi... have no idea... |
16:08.02 | webman | sorry, time for my sleep I guess... |
16:08.21 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
16:08.52 | ManxPower | NOBODY uses immediate=yes |
16:09.03 | greg_work | i have a tdm400p with 3 fxo connected to it .. for some reason, it'll only answer calls on the first port, though I can make calls out from any port.. my zapata.conf is here: http://pastebin.ca/8782 anyone know what would cause this? |
16:09.14 | Mimmus | webman: thank you anyway |
16:09.38 | greg_work | console shows nothing when a call comes in. i know it's ringing because i plugged an analog phone in |
16:09.50 | ManxPower | immediate= should be renamed donotwaitfordigits= |
16:10.30 | greg_work | and it's not the physical line, because if i plug any line into port 1, it answers.. but plugged into 2 or 3, nothing |
16:11.33 | *** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net) |
16:12.48 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
16:13.13 | viLeR | Hi, I am trying to authenticate to me with Free World DialUp, this is my line sip.conf: register => login:passwd@fwd.pulver.com/1020 but asterisk log says: Failed to authenticate on REGISTER |
16:13.22 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
16:13.52 | Zeeek | is your user account 1020? |
16:14.34 | viLeR | Zeek: no 1020 is my local extension. |
16:14.41 | Zeeek | I'm guessing it's not, so replace 1020 with your real one |
16:15.13 | Nugget | FWD doesn't care what your local extension is. |
16:16.55 | viLeR | Failed to authenticate again. |
16:16.59 | |nix | hi all, i've upgraded my copy of asterisk to CVS HEAD 01/04 |
16:17.34 | |nix | after upgrading, my voiemail failed to work, i noticed that upon recording, the voicemail now lives with a extension of .wav49 instead of .WAV like in the past |
16:17.40 | |nix | was anything broke in HEAD / something changed? |
16:17.42 | JunK-W | in France, for year, do u say dix-neuf-soixante-dix-neuf ou mille neuf-cent soixante-dix-neuf? |
16:17.47 | mgth | |nix: its head |
16:17.49 | *** join/#asterisk moy (~kvirc@201.137.229.111) |
16:18.11 | *** part/#asterisk gpled (~gpled@firewall.fccfurn.com) |
16:18.37 | jpe | greg_work: I am a real newbie, but if all the ports are on one card, I think you have to name them 1-1, 1-2, 1-3 etc. I have two single port cards in my set up system and have to name them 1-1 and 2-1 to get them reckognized |
16:18.42 | bkw_ | |nix, gotta patch voicemail now |
16:18.43 | |nix | mgth: thanks for the correction? |
16:18.44 | ManxPower | |nix: Anyone that uses Asterisk CVS and is not on the asterisk-cvs mailing list is an idiot. |
16:18.59 | bkw_ | ManxPower, please stop spewing crap like that. |
16:19.06 | ManxPower | bkw_: that is not crap. |
16:19.10 | bkw_ | it sure is |
16:19.20 | |nix | ManxPower: i might be an idiot, but i didn't install this copy |
16:19.24 | ManxPower | How many changes have been made to CVS-HEAD in the last 40 hours? 50 or so. |
16:19.37 | cypromis | so ? |
16:19.39 | bkw_ | but I know what change did this..and it wasn't in the past 50 hours |
16:19.40 | cypromis | it should still work |
16:19.41 | JunK-W | exacly, how can someone knows all that? |
16:19.41 | vaewyn | anyone that uses CVS and is not on the asterisk-cvs actually has time to develop ;P |
16:19.44 | cypromis | or the changes are nonsense |
16:19.57 | Slainte | does the rxgain and txgain work for a PRI setup? |
16:19.59 | ManxPower | cypromis: So people wine about something broken. A known broken issue. |
16:20.13 | cypromis | so they are right |
16:20.15 | bkw_ | its not broken really... its doing exactly what it should |
16:20.16 | cypromis | if something is broken |
16:20.17 | cypromis | it is wrong |
16:20.18 | cypromis | no ? |
16:20.21 | bkw_ | just not in the way you would expect it |
16:20.24 | bkw_ | :P |
16:20.33 | vaewyn | "undocumented feature" :P |
16:20.41 | cypromis | as in usual feature ? |
16:20.46 | |nix | ManxPower: if it so pleases you, i'm still knew in this asterik thingni as I took over it recently, but i'll take your advise and join asterisk-cvs mailing list |
16:21.01 | |nix | bkw_: thanks for the help, but can i say that its a known bug? |
16:21.11 | |nix | because i'm wondering if its my issue, or an issue with asterisk |
16:21.17 | ManxPower | |nix: asterisk-cvs mailing list will tell you EVERY change made to CVS. |
16:21.29 | cypromis | so ? |
16:21.32 | malverian | Has anyone used Linphone for connecting to Asterisk SIP ? |
16:21.37 | cypromis | does usage of asterisk require to be a c guru ? |
16:21.43 | |Vulture| | no |
16:21.45 | ManxPower | cypromis: Do you are not suprized when, oh, say, they remove Voicemail2. |
16:21.47 | cypromis | your comment looks like it |
16:21.52 | ManxPower | or change the codec processing order |
16:21.54 | |nix | ManxPower: i'll take note of THAT, thank you very much |
16:22.15 | ManxPower | or accidently comit a patch that totally breaks Asterisk (see kpflemming last night) |
16:22.25 | JunK-W | ManxPower: i fully disagree with you, using HEAD doesnt make u an idiot if u dont read -cvs mailing daily. |
16:22.28 | ManxPower | Granted, that happens very seldom, but it does happen. |
16:22.31 | Slainte | cypro,, if you can Read, use patch and make then you should be ok. You better know how to backup aswell |
16:22.35 | bkw_ | I don't recall a patch from lastnight that totally broke asterisk |
16:22.44 | zoa | me neither |
16:22.48 | zoa | did he break something ? |
16:22.53 | zoa | FLEMMIMG! :P |
16:23.12 | vaewyn | Ok... that's it /// KP duty for him |
16:23.19 | ManxPower | Modified Files: |
16:23.19 | ManxPower | frame.h |
16:23.19 | ManxPower | Log Message: |
16:23.19 | ManxPower | fix breakage from slin endianness commit earlier today (sorry :-() |
16:23.40 | zoa | ah, thats only for some people |
16:23.43 | zoa | with sparcs or so |
16:23.48 | bkw_ | that didn't break asterisk |
16:23.53 | ManxPower | now I don't know if that totally broke everything, but it does seem pretty serious and you would NEVER know about it if you were not on the mailing list. |
16:24.10 | bkw_ | it didn't break it... it just wasn't quite right. |
16:24.19 | ManxPower | You would also not know about the bug in ztcfg that was fixed this morning |
16:24.22 | vaewyn | ManxPower: I am glad you have all the time in the world to read email... :} cause none of us do |
16:24.30 | zoa | but he still deserves spanky spanky! |
16:24.38 | ManxPower | vaewyn: Dude, it takes me 5 mins per day to read the asterisk-cvs mailing list. |
16:24.57 | ManxPower | Unless you are using a web based e-mail interface, but you are far beyond my help if that is the case. |
16:25.19 | vaewyn | ManxPower: yes... but you also say that we all 'have' to read the -users... and -dev...and.... ... etc... which... by your ideals would be like 4-5 hours of reading a day |
16:25.21 | ManxPower | vaewyn: I don't READ the list, I glance at the comment to see if it might apply to me. |
16:25.24 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
16:25.32 | ManxPower | vaewyn: Naw, -dev is optional. |
16:25.43 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
16:25.45 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
16:25.45 | bkw_ | ManxPower, we can agree to disagree on this. Running head is fine. |
16:25.46 | vaewyn | and -users is a @#$@#$stream :} |
16:26.05 | bkw_ | -users is hopeless |
16:26.06 | ManxPower | bkw_: I agree that running CVS-HEAD is OK in some situations. |
16:26.17 | ManxPower | vaewyn: I guess my procmailrc filters are working well then. 8-) |
16:26.19 | PBXtech | h,1,System(asterisk -rx 'meetme kick ${UNI} 1') why isnt the ' passing? |
16:26.22 | vaewyn | Now... if someone is to lazy to check the archives... then they need to be bashed upside the head |
16:26.32 | ManxPower | I'm down to about 50-100 messages per day on -users |
16:26.39 | bkw_ | PBXtech, full path |
16:26.49 | ManxPower | vaewyn: the problem is that google doesn't index often enough |
16:26.59 | |nix | bkw_: any help on the voicemail issue? |
16:27.01 | PBXtech | that will make the ' pass? |
16:27.05 | vaewyn | Who needs google... browse the threads |
16:27.16 | ManxPower | bkw_: I think you should be on the asterisk-cvs mailing list if you are using 1.0.x CVS too. |
16:27.27 | |nix | i've checked the cvs-head mailing list, only 1 voicemail.c entry |
16:27.29 | vaewyn | There are these things called 'Subjects' that generally tell you what the email is about :P |
16:28.26 | ManxPower | |nix: check for codec or format changes too |
16:28.56 | |nix | ManxPower: Thanks. I'll take note of it |
16:29.24 | PBXtech | bkw still didnt pass the ' |
16:29.50 | ManxPower | UGH! I just CANNOT remember how to correctly do ARP spoofing. |
16:30.15 | forkqueue | ManxPower: man ifconfig |
16:30.20 | ManxPower | <PROTECTED> |
16:30.28 | pif | is there a way to set an ALERT_INFO in a queue ? I try setvar(ALERT_INFO=blah) before Queue(myqueue) to no effect. |
16:30.31 | ManxPower | forkqueue: that should do it, but doesn't seem to |
16:31.05 | mutilator | a cashiers check is guaranteed money right? |
16:31.15 | vaewyn | in the US yes |
16:31.32 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
16:31.38 | mutilator | k cause i got this email from a guy who wants to buy my car, sounds kinda fishy tho |
16:33.31 | ManxPower | vaewyn: NOT always |
16:33.34 | *** join/#asterisk gpled (~gpled@firewall.fccfurn.com) |
16:33.49 | ManxPower | it's guarnteed if it's not a FAKE cashiers check. |
16:34.00 | mutilator | heh yea |
16:34.16 | ManxPower | One of my vendors got ripped off by giving the UPS guy a fake cashiers check for a COD order. |
16:34.21 | ManxPower | ..er.. |
16:34.26 | PBXtech | " worked |
16:34.27 | ManxPower | One of my vendors got ripped off by someone giving the UPS guy a fake cashiers check for a COD order. |
16:34.48 | ManxPower | UPS would not cover their loss, their insurance company would not cover their loss. |
16:34.55 | ManxPower | I think it was about US$5,000 |
16:35.07 | mutilator | http://pastebin.ca/8784 |
16:35.14 | mutilator | this would be a bit more than 5k |
16:35.23 | DEEZED | hey guys.. I downloaded the asterisk sounds from asterisk.org and im getting an error when it plays it: Apr 4 12:33:41 WARNING[4699]: app_playback.c:90 playback_exec: ast_streamfile failed on IAX2/sixTel@sixTel-3 for im-sorry-unable-to-connect-to-eng.gsm |
16:35.36 | DEEZED | any idea why im getting this? |
16:36.09 | ManxPower | mutilator: can you call the bank that issued the check to confirm it. |
16:36.39 | ManxPower | DEEZED: don't put the file extension on Playback or Background |
16:36.46 | mutilator | i just got the email this morning, havn't even done anything about it yet |
16:37.02 | DEEZED | lol |
16:37.06 | DEEZED | oops |
16:37.09 | DEEZED | thanks |
16:38.20 | jpe | money orders or cashiers checks are easily forged. I would have him give you the check number and issuing vendor to call and make sure its good before accepting it. |
16:38.45 | jontow | woo, got the new PRI to the voicemail server up :) |
16:38.53 | BuckRogers | Hey has anyone configure wake up calling , i just recently did and the voice menu works i can set my time check it delete it but it will not execute, ? |
16:39.52 | mutilator | yea, |
16:40.00 | mutilator | thats what i just emailed him jpe |
16:40.19 | mutilator | sucks the only hit on my car in 2 months is fraud |
16:40.20 | mutilator | heh |
16:41.12 | jpe | mutilator: that is a scam, scam ,scam. |
16:41.22 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
16:41.32 | jpe | I just looked at the link, you wont hear back |
16:41.37 | munchausen | wallpaper the closest college with ads |
16:41.41 | cjk | hi, is there any way to define the callerid for iax2 like for sip? |
16:42.02 | ManxPower | cjk: Yes. The SAME way. |
16:42.22 | cjk | ManxPower, thats what i did. logically does not seem to work and thats strange |
16:43.05 | cjk | ManxPower, i have *1 connectiong to *2 i set the caller id on *2. can *1 overwrite this setting |
16:43.13 | ManxPower | cjk: then it's not patching |
16:43.25 | shido6 | Im going to smack someone |
16:43.26 | jpe | mutilator: how that scam works is this, they send you a cashiers check, you deposit it, if you have the funds to back it, it clears right away. You take the " balance" and sent it off, Western Union or some other untracable route. 3-4 days later when the check comes back bad, you get his for the full amount. A laser printer and MICR toner is all you need to make forges |
16:43.30 | shido6 | spandsp and r2 |
16:43.38 | ManxPower | cjk: explain it better to me. |
16:43.51 | mutilator | ya |
16:43.56 | mutilator | thats what i figure |
16:44.35 | mutilator | and as i don't think i've ever had a balance more than $100 in my account he's sol anyway |
16:44.36 | Gand_DJ | I get scams also.. don't worry. Had a couple on my business site.. hate those. |
16:44.51 | cjk | ManxPower, you should see *2 as the main server. *1 is a server connected to a pabx using bri's. i define the caller_id on the main server but it is somehow not applied |
16:45.14 | mutilator | so.. anyone want a 2002 intrepid ;) |
16:45.14 | ManxPower | cjk: You use * as part of your extension |
16:45.22 | mutilator | has in dash dvd player :O |
16:45.29 | mutilator | i know lot of you are in michigan heh |
16:45.59 | cjk | ManxPower, not really 2 bris in and 2 bris out |
16:46.06 | cjk | so its between the telco and the pabx |
16:46.21 | ManxPower | cjk: Perhaps you could type the word Asterisk. |
16:46.33 | ManxPower | cjk: So where the hell are your extensions? |
16:46.34 | malverian | Apr 4 12:46:03 NOTICE[6286]: Registration from '<sip:user@some.host.com>;tag=3564100787' failed for 'XXX.XXX.XXX.XXX' |
16:46.45 | *** part/#asterisk gpled (~gpled@firewall.fccfurn.com) |
16:46.46 | malverian | I keep getting this error.. and the logs dont' provide any more useful information than that :-/ |
16:46.54 | ManxPower | malcolmd: you don't have a [user] entry in sip.conf or the password is wrong. |
16:47.08 | cjk | ManxPower, the pabx sends everything to the small asterisk gates which does only this exten => _X.,1,Dial(IAX2/dcluxpabx@voipgate/${EXTEN},60) |
16:47.19 | |nix | ManxPower: i tried upgrading to the latest CVS and its still not working. you mentioned checking codecs, can you give me moreadvise to point me in the correct direction? |
16:47.32 | ManxPower | cjk: What callerid are you getting on the far side. |
16:47.41 | cjk | nothing |
16:47.53 | ManxPower | |nix: That was just an EXAMPLE of a significant change in CVS in the past. |
16:48.02 | mutilator | ;) |
16:48.12 | ManxPower | cjk: What device are you using to make the call? |
16:48.21 | malverian | ManxPower: I have [user] with username=user and secret=thepassword |
16:48.40 | |nix | darn |
16:48.44 | |nix | ok then |
16:48.44 | |nix | thanks |
16:48.46 | ManxPower | malverian: What is the type= |
16:49.04 | cjk | ManxPower, an pabx system phone |
16:49.20 | *** join/#asterisk _Sam-- (~sam@207.245.79.253) |
16:49.32 | ManxPower | cjk: and you have a callerid= set for the channel? |
16:49.55 | Faithful | How do I get asterisk to autoload zaphfc.ko ? |
16:50.12 | cjk | ManxPower, no i just do it in the friends part of my iax.conf and sip.conf |
16:51.28 | *** part/#asterisk java_ (~java@adsl203-158-089.mclink.it) |
16:51.34 | ManxPower | Well, if anyone has questions about Asterisk 1.0.x stable I'll be in, oddly enough, #asterisk-stable |
16:51.38 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
16:53.13 | *** join/#asterisk focks (~chatzilla@nsc66.147.95-93.newsouth.net) |
16:54.40 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
16:54.57 | tzafrir | Faithful, asterisk does not load kernel modules. What distro do you use? |
16:55.00 | Zeeek | JunK-W Mille neuf cent soixante etc |
16:55.28 | tzafrir | basically you need to run 'modprobe zaphfc' sometime in the boot process |
16:56.42 | Gand_DJ | What voip service do you guys use for calling out? I've browsed through a couple companies offerings so far. |
16:59.05 | vaewyn | nufone... they rock... |
16:59.16 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
16:59.28 | Gand_DJ | I haven't had a chance to readup on them. Appears the servers are being upgraded still |
16:59.42 | JunK-W | Zeeek: thx |
16:59.46 | Zeeek | np |
16:59.48 | robl^ | Nufone isn't accepting new customers at the moment |
16:59.56 | Zeeek | I was ont he phone forever so I didn't see the quest |
16:59.56 | langals | hi there...I am using asterisk meetme and trying to get better quality from the gsm codec |
17:00.08 | vaewyn | They are having growing pains :P |
17:00.09 | jsharp | I've used nufone, voipamericas, and voicepulse. |
17:00.20 | Zeeek | Junk-W Dix-Neuf Cent cinquante is also correct |
17:00.21 | langals | When I use the gsm codec with another conferencing server the quality seems to be quite a bit better |
17:00.52 | Gand_DJ | Right now I'm kinda eye-ing Broadvoice... since you can call upto 35 countries. |
17:00.53 | JunK-W | Zeeek: in quebec, we're not saying Dix-Neuf Cent cinquante anymore, only old ppl saying that. |
17:00.55 | langals | Does anyone have any idea how I could try and improve the quality? Would frame rate have an effect? |
17:00.57 | Gand_DJ | also allows asterisk. |
17:01.06 | Zeeek | Junk-W I think it's rare here too |
17:01.18 | JunK-W | so u opt for my patch? |
17:01.18 | Zeeek | anyway now we're in Deux Mille |
17:01.37 | JunK-W | yes, but if u want to say the birthdate of someone for example. |
17:02.03 | Zeeek | someone over 5? Ya, mille neuf cent soixante-neuf |
17:02.54 | JunK-W | fine. |
17:03.00 | nestAr | gah |
17:03.29 | JunK-W | i've remark some problem with other options in the ast_say_date_with_format_fr aren't correct. |
17:05.14 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
17:05.56 | Zeeek | I don't use the French stuff |
17:06.18 | Zeeek | (sounds) |
17:06.31 | Zeeek | In fact I wasn't using the indications for a long tiùe |
17:06.35 | Zeeek | time |
17:08.41 | DEEZED | Im trying to script my IVR. I want to add time between prompts so the user can have time to press what they need. I tried Wait(1) but it seems like the wait command doesn't allow a response. What is the correct command i should use? |
17:09.13 | jakepdev | DEEZED - use ResponseTimeout |
17:09.27 | Hmmhesays | or you could put silence in |
17:09.33 | nestAr | DEEZED: you can do a Backround(silence/1) |
17:09.39 | DEEZED | yeah couldn't find a silence sound |
17:09.49 | nestAr | there's a silence directory |
17:09.58 | nestAr | /var/lib/asterisk/sounds/silence |
17:10.02 | DEEZED | i put the command and it wasn't there... |
17:10.05 | DEEZED | ill check again |
17:10.08 | DEEZED | thanks guys |
17:10.13 | nestAr | no problem |
17:10.14 | nestAr | :D |
17:10.32 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
17:11.17 | DEEZED | silence: No such file or directory |
17:11.28 | moy | hi guys...... im having troubles to receive the digits from a Grandstream Phone when i use Background(), im using INFO |
17:11.31 | DEEZED | odd |
17:11.37 | moy | does any one knows what could be? |
17:11.53 | Hmmhesays | change to rfc2833? |
17:12.00 | vaewyn | moy: done an 'Answer' ? :} |
17:12.05 | vaewyn | no... info works |
17:12.32 | moy | ive already tryed rfc2833...... the funny thing is that i use the same phone in other PBX, and it works |
17:12.53 | Zeeek | there is something unusual about GS + codecs + DTMF |
17:12.57 | Zeeek | what codec? |
17:13.05 | Zeeek | iLBC by any chance? |
17:13.08 | moy | sip debug says that is using Ilbc |
17:13.10 | moy | yep |
17:13.13 | vaewyn | moy: make sure you have done an Answer otherwise unpredictable things happen |
17:13.14 | Zeeek | I think that was it! |
17:13.27 | *** join/#asterisk imagmo (~imagmo@c-24-20-249-117.hsd1.or.comcast.net) |
17:13.32 | Zeeek | I had that problem - I believe it was with iLBC |
17:13.57 | moy | yeawyn: you mean doing an Answer() before Background() ??? |
17:14.07 | moy | Zeeek: so what codec do you recommend? |
17:14.30 | vaewyn | moy: yep |
17:14.34 | Zeeek | Depends but what I see right now is that I'm using RFC at the moment |
17:14.44 | Zeeek | I wan't that thrilled with iLBC |
17:15.01 | Zeeek | (er... RFC for DTMF obviously) |
17:15.23 | vaewyn | ilbc is great for packet loss problems |
17:15.34 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
17:15.55 | moy | ok, mmm let me see, i will add the answer and try may be again with rfc... thanks |
17:15.59 | *** join/#asterisk infra (~infra@216-251-177-106.ips.cpinternet.com) |
17:15.59 | moy | :) |
17:16.36 | infra | hello all; I need help with IAXTEL!!! |
17:16.53 | Zeeek | moy it depends on the GS firmware too |
17:17.03 | blitzrage | IAXtel doesn't work right, unless they've fixed it. |
17:17.05 | Zeeek | there was an issue early on |
17:17.13 | blitzrage | use Free World Dialup |
17:17.20 | blitzrage | it has IAX connections |
17:17.27 | infra | I am limited to GSM codec...can't use FWD |
17:17.40 | blitzrage | it only used ulaw? |
17:17.44 | blitzrage | uses* |
17:17.45 | infra | indeed |
17:17.48 | blitzrage | how odd |
17:17.56 | Lee__ | iaxtel won't register here. i'm using voicepulse. |
17:17.58 | infra | so IAXTEL really is down? |
17:18.10 | infra | yeah, I can only register for about 20 seconds |
17:18.15 | blitzrage | yes, I found it unusable, but that was a few months ago, but I don't imagine its any better... |
17:18.44 | Lee__ | I think it's popularity outgrew it's capacity |
17:18.55 | infra | when registered, if I make a call I get: Max retries exceeded to host 69.73.19.178 |
17:19.10 | Zeeek | Lee__ we use FWD to test iax |
17:19.24 | Zeeek | too bad iaxtel hasn't been working for ages |
17:19.27 | Lee__ | pay the $11 for voicepulse connect. it's working good and you can talk to the PSTN |
17:19.42 | Zeeek | I have vp connect with no monthly |
17:19.48 | infra | does it use GSM or iLBC or G.729? |
17:20.05 | Lee__ | Zeeek: how'd you get that? |
17:20.11 | Zeeek | no DID |
17:20.15 | Zeeek | just outgoing |
17:20.26 | Lee__ | ah. we need the DIDs for the old skoolers :) |
17:20.37 | Zeeek | heh |
17:20.44 | Zeeek | I'm using nufone for DID |
17:21.02 | Lee__ | is that a commercial service? |
17:21.19 | Zeeek | debatable :) |
17:21.31 | Zeeek | depends entirely on the definition of "commercial" |
17:21.31 | vaewyn | moy: which firmware you using on that GS phone? |
17:21.53 | vaewyn | moy: cause 1.0.5.22 lists "Fixed we do not use RFC2833 to send DTMF when the incoming SDP contains iLBC and the immediate next "a" line is not the fmtp line for iLBC" |
17:21.54 | moy | vaewyn..... let me see |
17:22.03 | Lee__ | how would you define "commercal"? |
17:22.08 | jakepdev | commercial - usually dealing with commerce - money |
17:22.14 | *** join/#asterisk boch (~as24@200.59.172.98) |
17:22.32 | Zeeek | well, if you mean "available to the general public" they mostly are |
17:22.48 | vaewyn | moy: not exactly our issue... but a good hint they were messing around in there |
17:22.54 | vaewyn | s/our/your |
17:23.29 | Zeeek | but I keep hearing they aren't opening new accounts |
17:23.52 | moy | yaewyn, im using the version 1.0.0.7 ......... i haver tryed using Answer, no results yet, im trying to change the codec now |
17:23.55 | Lee__ | that's what the web page says |
17:23.55 | Zeeek | Lee__ what about voipjet? |
17:24.04 | mgth | voipjet sucks |
17:24.06 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
17:24.11 | Zeeek | I've had good luck with them |
17:24.30 | Lee__ | looked at them too. Eventually we'll become our own origination/termination but I imagine that's hard. |
17:24.42 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
17:24.44 | Zeeek | voipjet is so cheap I can call my cell in France cheaper than I can from a phone in France! |
17:24.56 | vaewyn | I am glad JerJer had the guts to stop accepting customers until they can get to handle the influx... that takes big nuts |
17:25.20 | Zeeek | inspite of the "bad rep" they sometimes get, those boys rock |
17:25.42 | Lee__ | I'm looking for a good backup o/t service |
17:25.55 | Zeeek | let's face it, no one gives good cust service of the kind you got...ummmm last century :) |
17:26.00 | vaewyn | Nufone gets my vote cause they are rock stable |
17:26.11 | Zeeek | cellphone providers are horrible, |
17:26.18 | vaewyn | cells are abysmal |
17:26.25 | Zeeek | no way to get anything fixed with cell providers |
17:26.38 | Zeeek | I refuse to engage in a contract with any of em |
17:27.03 | vaewyn | The best cell i can get in these parts is Nextel... and egads... dealing with them is still >< close to and amputation |
17:27.15 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
17:27.30 | Zeeek | In this country internet access is so cheap it's impossible to get service for anything |
17:27.51 | Zeeek | it's now like $18/mo for 8Meg/256 |
17:27.59 | Lee__ | !! where? |
17:28.03 | Zeeek | France |
17:28.10 | Lee__ | word! |
17:28.17 | Zeeek | and they just rolled out theis "20meg" (16 actual) |
17:28.26 | DEEZED | what is the best IAX service? pay per min allowing multiple connections? |
17:28.37 | DEEZED | 256 upload isn't that hot |
17:28.41 | Zeeek | depends so much on how you use it |
17:28.48 | vaewyn | DEEZED: IMO Nufone... but they arn't taking new customers at the moment |
17:29.05 | Zeeek | VP connect isn't bad tho |
17:29.20 | mgth | vp is expensive |
17:29.28 | Lee__ | The pay per miniute has a large wow factor if you are selling it to someone else. One DID, unlimited lines :) |
17:29.30 | Zeeek | yes, and voipjet is cheap |
17:29.40 | vaewyn | VP gives me issues about 2 hours/week |
17:29.46 | Zeeek | oh? |
17:30.01 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-170.dsl.scarlet.be) |
17:30.03 | vaewyn | I think it is a link between us and not actually them... but still... not acceptable |
17:30.11 | Zeeek | true I don't spend the day calling on any of the services I use, I just rotate them around for testing |
17:31.04 | mgth | Lee__ who is one did unlimited lines |
17:31.11 | mgth | does that mean unlimied imcomming calls? |
17:31.14 | Zeeek | I hooked up a toll free from nufone->asterisk->outgoing SIP provider and it worked great they tell me |
17:31.14 | Lee__ | yeah |
17:32.11 | mgth | lee___ nufone? |
17:32.21 | Lee__ | voicepulse |
17:32.50 | DEEZED | i have iax.cc and it lets me do that... but I need access to 1800 numbers and i had one that didn't work with them |
17:32.53 | nestAr | anyone having trouble building CVS-HEAD? It crashes at app_queue |
17:34.34 | bjohnson | mgth: unlimited is a funny word in voip .. it rarely means unlimited |
17:34.46 | cypromis | limited unlimited |
17:34.49 | cypromis | like unstable stable |
17:34.53 | DEEZED | lol |
17:34.59 | bjohnson | mgth: also, you should compare to per minute services which oftenn come out to be cheaper |
17:35.08 | jsharp | Sure, its unlimited. You can use it for an unlimited time, just as long as you pay 2 cents a minute. |
17:35.17 | Zeeek | cypromis what the latest on farfon? |
17:35.27 | moy | Zeek, yaewyn: still wrong my digit reception in Background, just one last question, do yoy thing that the fact that im running in a chroot enviroment could have something to do? because exactly the same config in my other PBX is doing it well |
17:35.28 | DEEZED | yeah per minute owns.. especially when you can handle multiple calls at once |
17:35.33 | bjohnson | uh oh .. he poked his head out |
17:36.25 | Zeeek | moy no idea |
17:36.47 | Lee__ | the limit is your bandwidth not how many "phone lines" you have |
17:37.04 | bjohnson | Lee__: also depends on the service |
17:37.09 | moy | ok, many thanks anyway, im going to move more config stuff to see what happens |
17:37.13 | bjohnson | some limit to one at a time |
17:37.38 | bjohnson | most limit to 4 to 6 at a time |
17:37.43 | Lee__ | well that's kind of lame. |
17:37.57 | bjohnson | start your own .. shown them how it should be done |
17:38.09 | Lee__ | that's the plan :) |
17:39.19 | Lee__ | don't hold your breath |
17:39.35 | Gand_DJ | How does one get setup though for offering free unlimited calls to certain places. |
17:39.46 | *** join/#asterisk brianj (~brian@guardian.pmt.org) |
17:40.09 | brianj | Gents, in the cvs head a lot of caller id was changed for the * manager, on a state ringing why does it show my callerid as my extensions caller id? |
17:40.23 | bjohnson | Gand_DJ: let us all know when you find out |
17:40.36 | *** join/#asterisk FirstSword (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
17:40.40 | Gand_DJ | :P |
17:40.41 | Gand_DJ | lol |
17:41.17 | bjohnson | you offering your pstn for our use or looking to use someone else's for your own use? |
17:42.24 | Gand_DJ | What I was meaning was, vonage & most places offer customers free calling to usa & canada.. but broadvoice offers free calling to 35 countries |
17:42.47 | Slainte | Gand_DJ, the business modelling is very complex for setups like that |
17:43.22 | Zeeek | you mean tollfree? |
17:43.43 | Slainte | You have to understand how they make their money |
17:43.47 | Slainte | nothing is ever free |
17:45.32 | *** join/#asterisk convey (~chatzilla@63.115.106.66) |
17:47.34 | *** join/#asterisk jtodd (~jtodd@207.230.254.134) |
17:47.50 | PBXtech | my IAX channel is hanging up immediatly after the Background command |
17:48.25 | PTG123 | GRand: you make your money by counting on people won't ever use enough to bust the amount of money it costs you |
17:48.30 | *** join/#asterisk devel (~devel@wiggum.digitalcoven.com) |
17:48.42 | *** join/#asterisk jtodd (~jtodd@207.230.254.134) |
17:48.42 | PTG123 | if it costs you 1penny a minute, and you sell $25 plans, then they won't exceed 2500minutes |
17:49.05 | PTG123 | and if you see someone exceeding it you bust them saying, you are not using this line for general use.. it must be business use so upgrade |
17:50.09 | jaiger | PTG123, nothing is free? next you'll tell me there is no easter bunny!! |
17:50.36 | PBXtech | does anyone know if the Background command is messed up in todays HEAD? |
17:51.18 | PBXtech | ResponseTimeout doesnt seem to work |
17:53.24 | bjohnson | Gand_DJ: last I checked you had to pay for that service .. not sure if it qualifies for free if you are paying for it |
17:53.55 | convey | Has anyone used Asterisk realtime with MySql 4.1? |
17:54.17 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
17:54.23 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
17:54.57 | Slainte | convey, I am using 4.0.23 |
17:55.36 | Slainte | does the rxgain and txgain work for a PRI setup? |
17:55.45 | PBXtech | yes |
17:56.04 | convey | Slainte: I am also using 4.0.23. I tried 4.1 and had authentication errors due to the password hashing in 4.1 |
17:56.11 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
17:56.18 | Slainte | phuket, I have no idea why I have not tried it yet |
17:56.30 | Slainte | convey; why do you need to upgrade :) |
17:56.49 | convey | Slainte: my developer ;) |
17:57.02 | Slainte | convey; why dont you null the passwords before you upgrade |
17:57.37 | jontow | heheh, my music on hold has 'crosby, stills, nash & young - carry on' stuck in my bosses head |
17:57.38 | jontow | ;) |
17:57.48 | harryvv | Is there a way to know how far a companies or goverment 1800 number will reach to in a state or region by looking it up online? |
17:58.18 | brianj | Gents, in the cvs head a lot of caller id was changed for the * manager, on a state ringing why does it show my callerid as my extensions caller id???? |
17:58.20 | brianj | :P |
17:58.20 | convey | Slainte: the problem is in the client. Mysql is looking for a shashed password and the Asteriskk client does not support it. I have been trying to turn off the password hashing with very little suck. |
17:58.49 | convey | slainte: sorry bout the type-o's :) |
17:58.50 | jontow | convey; simple solution.. you need more suck.. :) |
17:58.53 | harryvv | better suck harder :) |
17:58.54 | convey | LOL |
17:59.06 | Slainte | damit you guys beat me too it |
17:59.11 | jontow | (very much walked into it..) |
17:59.16 | vaewyn | best of suck to you |
17:59.32 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211) |
17:59.51 | emrah | Anyone can please help me with the Astcc application? I'm having a problem with the timeout before it says "The number is not answering". |
17:59.55 | Slainte | convey: What I did before I had the sql compiled in, is had a wrapper script populate the database from the csv file |
17:59.56 | SPoon_TSX | hello everyone, just wondering do you know if I can unregister a SIP peers from the CLI? |
18:01.08 | Slainte | convey: Change management for a system I set up for a swiss bank. |
18:01.20 | Slainte | thats why I needed the workaround |
18:01.33 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
18:01.33 | convey | slainte: maybe I can try something like that. I am afraid it is in the database authentication at the SQL driver level. |
18:01.54 | Slainte | convey, do the Perl mod's support 4.1? |
18:02.02 | Slainte | you can use perl. |
18:02.32 | bjohnson | harryvv: just get people here to call it |
18:03.08 | InfraRed | how cheap? |
18:03.12 | InfraRed | you buying a bank ? |
18:03.25 | Slainte | emmmm, something like that yea. |
18:03.28 | bjohnson | cheap as in .. don't want to pay or cheap as in .. low service fees |
18:03.34 | Slainte | dont want to pay |
18:03.48 | InfraRed | uk banks are free |
18:03.49 | bjohnson | it's habitual |
18:03.56 | InfraRed | most of them anywya |
18:04.05 | bjohnson | they are used to money flow being one way |
18:05.04 | *** join/#asterisk ChulJin (~chuljin@65.211.236.166) |
18:05.36 | ChulJin | Good morning Gentlemen! |
18:06.57 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
18:07.46 | ChulJin | is there a current universal favourite among low-price (~$100) hardware SIP phones? My co. is seeking to replace our BT101's with something better (as now fewer of them work than don't) |
18:08.41 | *** join/#asterisk michael_t (~michael_t@c-24-20-234-51.hsd1.or.comcast.net) |
18:09.16 | Gand_DJ | We use Lucent phones at my work, but don't know the cost per phone |
18:09.40 | nestAr | the sipuras are in that price range, i think |
18:10.09 | nestAr | http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38735287552.htm |
18:11.13 | malverian | What do you guys use to listen to .gsm files? |
18:11.29 | malverian | They sound REALLY crappy on my Alsa sound card when I do tcat some.gsm > /dev/sound/dsp |
18:11.44 | malverian | Is there a better way/utility? |
18:11.59 | jontow | sox? :) |
18:12.01 | ChulJin | right, Sipura is what I'm currently recommending to them. |
18:12.26 | jontow | sox file.gsm -t wav file.wav ; audacity file.wav |
18:12.48 | ChulJin | we're already a fan of their ATA's, of which we have several, and with which we've had not a single problem. |
18:14.31 | malverian | Ah.. okay |
18:14.43 | jontow | i wrote a script for it.. 'gsmplay' ;) |
18:15.17 | jontow | checks for existance of the file, automatically changes the name and converts to wav, then runs audacity on it and when i exit audacity, asks if i want to remove the .wav (leaving the .gsm) |
18:15.21 | jontow | will be glad to provide it if you'd like |
18:16.32 | malverian | Sure. |
18:16.45 | malverian | Out of curiosity, why play it in audacity? |
18:16.58 | malverian | Instead of a command line one. |
18:17.37 | jontow | i wanted the quick potential of removing bits and pieces as necessary |
18:17.54 | jontow | it is very easy to change it to use a command line player instead of audacity though. |
18:18.44 | *** join/#asterisk bannerman (~bannerman@c-24-20-88-59.hsd1.wa.comcast.net) |
18:18.54 | malverian | I'm trying.. |
18:19.00 | jontow | http://web.slic.com/~jontow/gsmplay.txt |
18:19.06 | malverian | sox /var/lib/asterisk/sound/foo.gsm -t wav /dev/dsp |
18:19.11 | malverian | And I get static. |
18:19.34 | vaewyn | If you have sox try 'play /var/lib/asterisk/sound/foo.gsm' |
18:19.34 | _Sam-- | sox winwave.wav -r 8000 -c 1 linwave.gsm |
18:19.56 | _Sam-- | sorry |
18:19.59 | bannerman | I asked last week, but I lost my notes. ulaw sounds good, but with varying network conditions tends to degrade and get junky. Anyone have a recommendation for the right codec to use with moderately poor QoS? |
18:20.18 | zoa | just use quicktime |
18:20.20 | bannerman | I don't mind paying for a commercial codec if that's the right way to go. |
18:20.35 | malverian | vaewyn: That works :-P |
18:20.43 | vaewyn | bannerman: The less traffic the better... so smaller codecs do better... and for really bad packet loss ilbc is the nicest |
18:20.50 | vaewyn | malverian: good good :} |
18:21.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
18:21.34 | bannerman | I don't think I have really bad packet loss .. maybe I do. I have a full T1 for a fairly small office. I only have trouble with ulaw on rare occasions. |
18:25.15 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
18:26.06 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
18:26.13 | hardwire | hey smart people |
18:26.21 | MattH | lol |
18:26.34 | hardwire | I need a T1 going into my office w/ DID's coming in. |
18:26.37 | MattH | hey... anyone here using broadvoice... what's the limit on the number of incoming and outgoing calls? |
18:26.46 | hardwire | where you can have as many as you want until the t1 is funn |
18:26.47 | hardwire | err |
18:26.50 | hardwire | until the span is full |
18:27.00 | hardwire | of different DID's mapped to your span |
18:27.06 | hardwire | what exactly should I be asking for? |
18:27.58 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
18:29.48 | *** join/#asterisk jpe (~jpe@66.114.77.37) |
18:31.45 | Slainte | hardwire, how many users? |
18:32.02 | hardwire | need a full span coming in |
18:32.09 | hardwire | going to only have 8 numbers |
18:32.14 | bjohnson | MattH: read their terms of service |
18:32.32 | bjohnson | hardwire: numbers don't matter .. how many concurrent calls |
18:32.38 | hardwire | a full span |
18:32.41 | Slainte | so whats your question, if you know you need 8 numbers and a full span |
18:32.42 | bjohnson | so 23 |
18:32.46 | hardwire | yup |
18:32.55 | bjohnson | so ask for a PRI |
18:33.05 | bjohnson | se what they have to offer |
18:33.08 | *** join/#asterisk DougCoker (~dcoker@adsl-67-37-176-252.dsl.chcgil.ameritech.net) |
18:33.19 | hardwire | cool |
18:33.34 | hardwire | Slainte: what the terms to use would be :) |
18:33.39 | hardwire | if its exactly that.. then I am all set |
18:33.44 | Slainte | ah,,, yes a PRI |
18:33.55 | bjohnson | hardwire: it's more a question of what they have available |
18:34.03 | hardwire | esp here |
18:34.05 | hardwire | in alaskatown |
18:34.14 | bjohnson | hardwire: also .. if you want internet access through it you'll have to discuss that with them too |
18:34.21 | hardwire | I don't |
18:34.36 | hardwire | we have a 2Mbps sync DSL for the office. |
18:34.41 | hardwire | from that same company |
18:34.44 | bjohnson | also ask about installation schedule |
18:35.03 | *** join/#asterisk marno (~marno@213-182-126-48.teleos-web.de) |
18:35.11 | marno | hi |
18:35.14 | hardwire | I asked about lead time on it |
18:35.30 | DougCoker | Anyone successful in making Audiocodes MP-104 work with asterisk ? |
18:36.02 | marno | after some time my asterisk uses 100% cpu. is there a command to see what is going on? |
18:37.03 | blitzrage | top |
18:37.06 | MattH | bjohnson: I did read them.. I see nothing limiting it |
18:37.38 | bjohnson | last I read them they had a clause saying more than one concurrent call would be billed at some posted rate |
18:38.24 | bjohnson | MattH: in any event .. they would know more about their current service offering than we would. To date that info has been pretty scarce |
18:38.31 | *** part/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
18:38.42 | Gand_DJ | bjohnson, I recall that being true if multiple devices are registered at once for 1 number |
18:38.50 | Gand_DJ | They are slow for answering questions |
18:38.54 | *** join/#asterisk Cadu20 (~Cadu83@200-215-114-219.fnsce7001.e.brasiltelecom.net.br) |
18:38.56 | Gand_DJ | I emailed this friday, and it's now monday |
18:39.09 | hardwire | behold the power of a weekend. |
18:39.13 | Gand_DJ | they are open 24/7 they say. |
18:39.15 | Cadu20 | Hi there, anybody could explain me better what the "ignorepat" command does? |
18:40.04 | marno | blitzrage, how can top tell me, what asterisk is doing??? |
18:40.05 | nvrswork | anyone have much success running a NAT'd asterisk server behind a linksys WRT54G router connecting to SIP termination services? im getting what I think are sip connection timeouts. |
18:40.29 | Gand_DJ | nvrswork I did it. seemed ok. |
18:40.54 | Gand_DJ | I also ran it through dmz just in case. seemed ok also that way |
18:41.23 | Qwell | Cadu20: If the first digit(s) you dial are the same as ignorepat, you'll keep your dialtone |
18:41.35 | Qwell | Cadu20: Ever been somewhere that required "9" to dial out? |
18:41.59 | Gand_DJ | I had my server linking to FWD |
18:42.17 | marno | no idea? |
18:42.38 | Cadu20 | Qwell, ah ok!... thank you very much. |
18:42.58 | emrah | Anyone can help me with the timeout in Astcc? (The number is not answering) I want to incrise the time before I get it |
18:43.02 | Cadu20 | Qwell, so, there is no really use for it... just to make the user "think" it is in the old system...? |
18:43.15 | Qwell | Cadu20: no, there is plenty of use for it |
18:43.29 | Qwell | When somebody dials 9, they expect a dialtone still |
18:43.38 | hardwire | ok |
18:43.40 | hardwire | they were cool |
18:43.47 | hardwire | PRI will be installed with a few temp DID's for testing |
18:43.48 | Qwell | (if you require it for dialing out, that is) |
18:43.59 | hardwire | then POTS lines will be moved over the the PRI when the system is ready to rumble. |
18:44.04 | Qwell | hardwire: They're giving you a test PRI? |
18:44.28 | hardwire | I wouldn't call it that.. I am buying a PRI.. |
18:44.31 | Cadu20 | Qwell, right.. i got it... thank you very much! |
18:44.32 | Qwell | oh |
18:44.41 | hardwire | its my money during the test :) |
18:45.00 | hardwire | and I will have signed a contract to use it for a certain minimum length of time under the terms of service. |
18:45.01 | hardwire | so yhe. |
18:45.02 | hardwire | heh |
18:45.38 | bjohnson | MattH: http://voxilla.com/voxstory71-nested-order0-threshold0.html .. in addition to monitoring usage patterns to look for suspect activity, which many providers do, BroadVoice will also charge the end-user 3.9 cents per minute if more than one outbound call is active using the same set of SIP credentials (except in the case of a three-way call). |
18:45.56 | bjohnson | another limit on "unlimited" |
18:46.48 | Gand_DJ | So much for using an * server, and giving people in the house extensions for their softphones |
18:47.24 | bjohnson | cadu20: there is indeed a use/need for dialing 9 for an outside line |
18:48.08 | vaewyn | bjohnson: yeah... screwing your employer over with RSI lawsuits :} |
18:48.12 | bjohnson | Gand_DJ: use a per minute service |
18:48.21 | bjohnson | vaewyn: ? |
18:48.41 | vaewyn | carpal tunnel from hitting the extra digit all the time |
18:48.52 | Gand_DJ | bjohnson, I'm going to setup the asterisk@home system, and give family members in the house an extension each.. and install a softphone on each pc. |
18:48.59 | bjohnson | vaewyn: if you use a 9 .. you can make a pattern that doesn't require a timeout on the dial plan = faster execution = same as ending with '#' |
18:49.11 | Gand_DJ | was going to use broadvoice because of the 35 country free calling |
18:49.16 | bjohnson | (btw .. '#' is also an extra digit |
18:49.17 | bjohnson | ) |
18:49.33 | bjohnson | Gand_DJ: IT ISN"T FREE!!!! |
18:49.51 | nvrswork | Gand_DJ, Linking to FWD? |
18:49.52 | vaewyn | bjohnson: Umm... you can also plan your internal extensions to not use local prefixes and voila... same effect :P |
18:50.04 | nvrswork | oh FreeWorldDialup |
18:50.06 | Gand_DJ | Fine..... I was going to get the $24.95 plan that includes calling to 35 countries for no extra charge |
18:50.07 | nvrswork | hmmm |
18:50.28 | *** join/#asterisk mcukstorm (~mcukstorm@neo.matrix-lan.net) |
18:50.40 | jaiger | bjohnson, let hom get the bill and see how free it is |
18:50.45 | nvrswork | I have my server in DMZ as well., |
18:50.56 | bjohnson | Gand_DJ: $24.95 / 0.02 = 1200 minutes of calling if you go with a per minute provider |
18:50.58 | vaewyn | '9' prefix is only for those that can't choose extensions correctly |
18:51.11 | mcukstorm | Hi, does anyone know where the wcfxs module has dissapeared to? i just grabbed the cvs and it isnt in there :$ |
18:51.31 | bjohnson | vaewyn: I'd love to see that .. a dial plan that doesn't rely on timeout for internal OR external calls |
18:51.41 | Gand_DJ | 0.02 is probably for USA/Canada only though right? or anywhere on earth? |
18:51.54 | nvrswork | the problem i keep getting is, when I connect to my SIP termination service, it will register then people can call the number for a couple minutes and it will reach me. then a little later and it will just ring once then give busy signal |
18:52.02 | vaewyn | bjohnson: Just don't use extensions that overlay your local dialing prefixes... and you are set |
18:52.07 | jsharp | On an IAX call, if the called system doesn't get frames for a certain time, it drops the call, yes? Does it send back a "BUSY" indication to say that it did? |
18:52.12 | nvrswork | sort of like the sip connection is timing out |
18:52.15 | bjohnson | Gand_DJ: you will have to do you're own shopping. I noticed on the livevoip international rates that many countries are available for 0.02 |
18:52.19 | Gand_DJ | My mom has a friend in australia she likes to talk to, I know a couple people in malaysia & UK |
18:52.39 | bjohnson | vaewyn: it isn't that simple |
18:53.07 | *** join/#asterisk obelix-o (~fabio@200-138-246-242.fnsce7006.dsl.brasiltelecom.net.br) |
18:53.08 | vaewyn | bjohnson: yes it is... because for any non local prefixes people must dial '1' first... :} |
18:53.13 | bjohnson | vaewyn: give me an example of an internal extension that doesn't overlap an outbound number |
18:53.18 | hardwire | bigjohnson :) |
18:53.18 | obelix-o | hi guys |
18:53.40 | vaewyn | bjohnson: 6103 :} |
18:53.47 | obelix-o | there are a open g729 codec for asterisk? |
18:54.08 | hardwire | no |
18:54.20 | vaewyn | Yes... not legal in the US of A though |
18:54.28 | vaewyn | not even for 'testing' |
18:54.33 | vaewyn | so don';t ask :} |
18:54.36 | Qwell | Its not "open" then |
18:54.44 | hardwire | sure it is.. |
18:54.46 | hardwire | in Antartica |
18:54.47 | vaewyn | it is in every other country |
18:54.50 | marno | after some time my asterisk uses 100% cpu. is there a command to see what is going on? |
18:55.00 | Qwell | "open" or "hacked and legal"? |
18:55.04 | Qwell | massive difference |
18:55.08 | hardwire | marno: did you change nicks? |
18:55.16 | bjohnson | vaewyn: what pattern do you use for outgoing local |
18:55.19 | vaewyn | Qwell: it is open.. the code is available... you just can't use it in the US |
18:55.29 | hardwire | marno: you should probably check out strace, and asterisk in debug mode |
18:55.35 | DougCoker | Anyone successful in making Audiocodes MP gateways working with asterisk ? |
18:55.38 | obelix-o | i have tried to install g729-041103.diff |
18:55.42 | hardwire | there is no one specific command to find out the one thing asterisk is doing to your system |
18:55.48 | obelix-o | on l_ipp_ia32_itanium_em64_eval_p_4_1_2_ev05 |
18:55.52 | *** join/#asterisk kmest (~kmest@adsl-158-39-198.asm.bellsouth.net) |
18:55.55 | vaewyn | bjohnson: local prefixes and then 1 prefix for long distance... 0prefix for international...etc... |
18:56.07 | obelix-o | but can't run |
18:56.11 | dmabe | I just got an IAXy and provisioned it, but I can't get it to make calls or be called. It gets a fast busy when dialing out. Everyone is busy congested when trying to call it. |
18:56.13 | marno | hardwire, change nicks? |
18:56.16 | obelix-o | this lib works on i386 ? |
18:56.16 | hardwire | nm |
18:56.16 | dmabe | any ideas? |
18:56.26 | hardwire | I thoguht somebody else just asked that question verbatim a few minutes ago |
18:56.35 | bjohnson | vaewyn: like exten => _NXXXXXX,1, ? |
18:56.47 | Gand_DJ | Qwell: http://www.readytechnology.co.uk/open/g729/ |
18:56.57 | Gand_DJ | if you arn't in usa |
18:57.23 | Qwell | I wouldn't use it anyways |
18:57.25 | vaewyn | bjohnson: nope... local defined prefixes... ie 208XXXX,1,... 471XXXX,1,... etc... |
18:57.29 | obelix-o | Gand_DJ i have tried but can't install this |
18:57.33 | bjohnson | vaewyn: how many do you have listed? |
18:57.38 | marno | hardwire, there is nothing on the interface, this seems to be "inside" of the asterisk |
18:57.42 | vaewyn | bjohnson: 41 |
18:57.44 | bjohnson | vaewyn: most people wouldn't bother |
18:57.44 | jontow | marno; "gdb" |
18:57.49 | marno | hardwire, that was me, with the same nick |
18:58.03 | jontow | thats the one command to tell you exactly what is happening inside asterisk at the moment. |
18:58.04 | vaewyn | bjohnson: no... but I prefer to make it hard on the admins and easyon the users... 9 is stupid :P |
18:58.26 | hardwire | marno: run asterisk on the console in full debug |
18:58.38 | vaewyn | bjohnson: and really.. around here locals are only added once a year if that |
18:58.42 | marno | jontow, what is gdb? |
18:59.03 | bjohnson | vaewyn: you should point that out when people are asking about it .. most people would not appreciate the extra config involved in not using '9' |
18:59.07 | marno | hardwire, it is debuglevel 9 but there is nothing interesting |
18:59.10 | vaewyn | bjohnson: and they can still dial them 1(area)+local till we add it |
18:59.32 | hardwire | marno: sorry.. |
18:59.41 | jontow | oh dear.. yeah, good luck |
18:59.44 | hardwire | turn on channel debugging then if you think a channel is doing something bad |
18:59.47 | jontow | gdb is the GNU Debugger |
19:00.18 | jontow | it allows for a specifically intensive point of view over a running process (ANY running process) on a linux/bsd/etc system |
19:00.19 | bjohnson | vaewyn: what do they get if the dial plan doesn't match? |
19:00.29 | jontow | however.. you also very much have to know how to use it properly to get anything useful from it.. |
19:00.34 | vaewyn | bjohnson: Other (simpler for admin... harder for user) is require 1+area+local for everything :} |
19:01.13 | vaewyn | bjohnson: "sorry no such extension" for 4 digits... "unable to understand the number you suppliied" for 5+ |
19:01.25 | bjohnson | vaewyn: someone here was trying to use agi to do a dynamic lookup of local prefixes |
19:01.51 | marno | hardwire, there is no active channel |
19:02.04 | hardwire | rm -rf / |
19:02.05 | hardwire | :) |
19:02.10 | vaewyn | bjohnson: is doable... but stupid... that stuff changes so little... |
19:02.14 | hardwire | your machine is pocessed. |
19:02.20 | hardwire | pocessed.. I don't think thats a word |
19:02.26 | hardwire | posessed. |
19:02.38 | vaewyn | bjohnson: even in DC area we only add 1 or 2 new prefixes per month |
19:02.47 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
19:02.48 | vaewyn | and that is slowing down again |
19:02.58 | Qwell | hardcoded prefixes and extensions? |
19:03.09 | vaewyn | hardcoded local prefixes |
19:03.10 | Qwell | Must be one (pointlessly) massive dialplan |
19:03.29 | jontow | hmm |
19:03.31 | vaewyn | no... it has a point... everyone dials normally... and it works fast |
19:03.33 | hardwire | vaewyn: it would be nice to have a list of all prefexes for certain dialing areas |
19:03.44 | hardwire | and the dialing area code.. if one even exists.. for that area |
19:03.47 | Gand_DJ | Has anyone tried using sipura fxo boxes to work on asterisk (instead of using digium cards) |
19:03.51 | Gand_DJ | I'm looking at http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-34702223616.htm |
19:04.02 | hardwire | Gand_DJ: I have |
19:04.10 | vaewyn | hardwire: SBC provides a TXT file of that for us... one of the few things I like about them... comes with our call rating data |
19:04.15 | Qwell | vaewyn: I can 7 digit, or 11 digit dial, with no wait |
19:04.25 | Qwell | and no hardcoded prefixes |
19:04.47 | vaewyn | Qwell: yes... but this is 4digit 7digit and 11digit |
19:04.55 | hardwire | vaewyn: we have SBC up here |
19:04.56 | vaewyn | much more difficult :P |
19:04.59 | BuckRogers | Hardwire, sbc does the same for us |
19:05.05 | hardwire | well crap |
19:05.09 | hardwire | I want one |
19:05.09 | vaewyn | hardwire: talk to your rep... they can give it |
19:05.15 | *** join/#asterisk Jearil (~Jearil@216-224-56-213.client.dsl.net) |
19:05.17 | hardwire | well thats the weird thing |
19:05.23 | Qwell | vaewyn: _XXXX ? |
19:05.26 | hardwire | in alaska things are very different |
19:05.27 | Gand_DJ | I want to setup the houseline to be answered by asterisk.. and then someone can either use voip to call out internationally (if they can authenticate for using that option) or just leave a voice msg in the proper voicemail box for the right family member |
19:05.33 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
19:05.42 | Gand_DJ | Thinking of using asterisk@home setup |
19:05.42 | ChulJin | Gand_DJ: I'm not prepared to configure it for you (see the wiki :P), but I can tell you I've found that the SPA-3000 works great with * |
19:05.43 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
19:05.45 | hardwire | I don't have an SBC rep.. because none of my lines come from SBC.. everything goes through ACS.. |
19:05.46 | vaewyn | Qwell: yep... is why I have to hardcode locals |
19:05.48 | hardwire | and they know crap. |
19:05.54 | Qwell | eh? |
19:05.57 | SPoon_TSX | Sorry everyone, does anyone know how can I force to unregister a SIP client from CLI on asterisk? |
19:06.07 | hardwire | Gand_DJ: thats what I have set up at home |
19:06.19 | hardwire | use the SPA-3000 if you wanna use your standard lines. |
19:06.29 | Qwell | hmm, I see... |
19:06.30 | vaewyn | Qwell: You either have to hardcode the local prefixes or use 9 for outside dialling... or wait for timout... that is the options |
19:06.32 | hardwire | I have the SPA-3000 answering.. and then redistributing the FXS to the rest of th ehouse |
19:06.59 | _SMP_ | Hi folks, does anyone know whether there are debian packages for AMP out there? |
19:07.03 | vaewyn | and I'll take 41 extra dialplan lines over having to dial '9' first anyday :P |
19:07.06 | *** join/#asterisk lliquibop1234 (~lliquibop@dataflo-office-rtr.dataflo.net) |
19:07.47 | Gand_DJ | hardwire: so if someone calls your home... the phone plugged into the FXS port will ring until asterisk picks up the line? |
19:08.09 | hardwire | you scare me |
19:08.25 | hardwire | thats not how it works at all if you set it up right |
19:08.31 | hardwire | every single incoming call goes to asterisk |
19:08.36 | hardwire | asterisk decides to ring the fxs |
19:08.48 | Qwell | vaewyn: what happens when your area adds a new 603 prefix? (assuming your extensions are 6000-6050, or something) |
19:09.29 | Gand_DJ | hardwire.. ah so the fxs port would be setup like an extension.. and is someone wants to reach extension xxx, then asterisk rings that line (instead of a softphone that would be setup for another extension) |
19:10.59 | vaewyn | Qwell: They are not... prefixes in our area are assigned as 2XX and 4XX until those run out... and then they are slated for 3XX and 5XX |
19:11.33 | jontow | man.. my old house.. we only had a single prefix :P |
19:11.42 | vaewyn | and plus... not all 4XX and 2XX patterns are valid... remember they can't use them all :P |
19:11.43 | jontow | the entire 40mile^2 area ;) |
19:11.52 | Qwell | shit...phone company should have let you 4 digit dial your neighbors then :p |
19:11.54 | jontow | actually, bigger than that now that i think about it |
19:11.58 | jontow | no kidding |
19:12.42 | ChulJin | gand: that's right. |
19:13.04 | ChulJin | gand: if set up as hardwire did. |
19:13.05 | Gand_DJ | ChulJin: sounds better then using a digium card |
19:13.37 | Sedorox | bbiab.. g/f needs help installing something |
19:13.37 | hardwire | ChulJin: send me an email |
19:13.42 | Gand_DJ | how else could it be setup? (guess you could not use fxs port at all) |
19:13.42 | hardwire | hardwire at bogomip dot com |
19:13.45 | hardwire | same with you gand |
19:13.49 | ChulJin | hardwire: about what? |
19:13.56 | hardwire | so I know who to notify when I publish my really simple config |
19:14.00 | emrah | Please, anyone have experience with Astcc? |
19:14.05 | hardwire | the SIPuras are the PITA |
19:14.31 | ChulJin | hardwire: I've had no trouble...mine's already configured and working fine. |
19:14.39 | hardwire | oh |
19:14.45 | hardwire | well then you help him :) |
19:14.58 | hardwire | heh |
19:15.14 | hardwire | all non CID calls go directly to voicemail |
19:15.22 | hardwire | simple |
19:15.24 | *** join/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net) |
19:15.31 | hardwire | I want to use PrivacyManager() at some point |
19:15.47 | ChulJin | hardwire: mine's a bit different than yours... |
19:16.05 | ChulJin | I use the 'spouse-friendly' style setup I found [somewhere]...I think on the wiki |
19:16.05 | gtigene | What version of Debian Linux should I use for a new * system? |
19:16.07 | hardwire | I would assume so. |
19:16.14 | hardwire | gtigene: I use testing |
19:16.29 | ChulJin | calls on the fxo ring through to the fxs |
19:16.54 | ChulJin | (* never answers the fxo) |
19:16.57 | gtigene | harwire: any particular reason? |
19:17.05 | Gand_DJ | hardware & ChulJin, you figure the best (reliable) and cheapest fxo device to use for linking * to a houseline is the sipura 1000+,2000+,3000+ cards |
19:17.07 | *** join/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net) |
19:17.20 | Qwell | Gand_DJ: the 1000 and 2000 aren't FXOs |
19:17.22 | ChulJin | calls from * ring through with a different ringtone... |
19:17.39 | Qwell | can an analog phone have a different ringtone? ;/ |
19:18.04 | hardwire | Gand_DJ: what about the X100P? |
19:18.08 | ChulJin | qwell: with the sipura spa's, you can set alert_info |
19:18.15 | ChulJin | gand_dj: 3000 only. |
19:18.22 | Qwell | ChulJin: I've got a tdm |
19:18.27 | Gand_DJ | ok. Isn't the X100P really expensive? |
19:18.39 | hardwire | do I like like pricegrabber.com ? |
19:18.42 | Qwell | Gand_DJ: Its either expensive, or a non-reliable clone |
19:18.44 | jontow | $99? |
19:18.51 | jontow | yeah, i've got a crappy clone.. |
19:18.58 | jontow | i paid $40 for 3 of them |
19:19.03 | jontow | (notice the giant price gap) |
19:19.05 | ChulJin | 1000 and 2000 are the best (reliable), but not cheapest, way to burn up 1000 and 2000 when ringing voltage curses through their FXS-only ports :P |
19:19.19 | jontow | and i have volume problems like crazy on them all |
19:19.27 | Gand_DJ | Sipura card isn't that expensive compared to X100P I don't think |
19:19.33 | *** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com) |
19:19.34 | Qwell | meh, just go voip only :p |
19:19.38 | Qwell | drop the fxo idea |
19:20.01 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
19:20.22 | johnnyb | What is the "family" parameter of DBput and DBget for? Is it just a namespace issue? |
19:23.23 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
19:25.54 | bjohnson | vaewyn: I documented 6 ways to combine local calling and internal extensions at http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Planning |
19:26.04 | Gand_DJ | I would drop the fxo, but if someone calls in the houseline, I want to setup * to allow the user to get ahold of the proper family member (provided they have their softphone activated on the pc), ring the FXS port (for general answer), or goto voicemail |
19:27.12 | DougCoker | Anyone successful in making Audiocodes gateways work with asterisk ? |
19:27.21 | bjohnson | Gand_DJ: sipura aren't cards .. they are standalone ATAs that plug into your LAN |
19:27.39 | Gand_DJ | yeah |
19:27.56 | Gand_DJ | saw them at atacomm or whatever the site was someone mentioned earlier |
19:28.01 | bjohnson | and if you want to tie a telco line into * .. you HAVE to have a fxo |
19:28.13 | Gand_DJ | the sipura 3000 has fxo |
19:28.16 | bjohnson | Gand_DJ: all sorts of places have the Sipuras |
19:28.23 | bjohnson | yes .. I have 3 SPA 3ks |
19:28.42 | bjohnson | btw .. the X100P is like $20 |
19:28.51 | bjohnson | the SPA 3k are about $100 |
19:28.55 | Gand_DJ | the clone / intel voice modem? |
19:29.00 | bjohnson | yes |
19:30.20 | bjohnson | hardwire: I started to set up a privacy manager .. but then saw that getting callerid from telco was going to cost extra $10 / month |
19:30.41 | hardwire | bjohnson: hah |
19:30.53 | bjohnson | ChulJin: why don't you have the fxo calls go to *? |
19:30.57 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
19:30.59 | Qwell | bjohnson: Thats one of the reason I dropped Verizon |
19:31.11 | *** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
19:31.57 | bjohnson | ChulJin: I guess you've also answered the question about analog phones having different rings since you do that |
19:32.00 | DEEZED | after recording a file with the Record function, Is there an automated way to attach the gsm file to an extension? |
19:32.08 | bjohnson | DEEZED: no |
19:32.39 | DEEZED | but it can be done right? with a macro? |
19:32.57 | bjohnson | only if you always want to do exactly the same thing with each |
19:33.00 | bjohnson | most people do not |
19:33.37 | bjohnson | most people find doing by hand much faster and flexible than writing a macro that will only do what they want SOME of the time |
19:34.02 | DEEZED | my dad has a Virtual PBX with a company... he types in a random extension then presses a *7 and it lets him record |
19:34.15 | DEEZED | then when you call that extension it plays the file back |
19:34.46 | bjohnson | not very useful .. is it |
19:35.22 | DEEZED | well if i could do that i would be farther along then i am now |
19:35.42 | bjohnson | farther along to what? recording ramdom sound files? |
19:36.25 | bjohnson | provide us with an example purpose for such a thing |
19:36.54 | vaewyn | it's like a phone post-it-note... cool idea... but man... I can't see much use for it :P |
19:37.01 | bjohnson | also .. wouldn't he end up potentially overwriting valid extensions with those "random" extensions to play back sound files? |
19:37.19 | bjohnson | vaewyn: he could use voicemail for that |
19:37.41 | DEEZED | ok. My dad is a realtor. He has a 1800 number and he creates a pin number for each house that he is selling.. When his client calls in, they can type in the pin and hear about that house |
19:37.56 | vaewyn | bjohnson: but voicemail requires authnetication... this is nothing but placeholders |
19:38.13 | DEEZED | my dad can record his pins by typing in the new pin, then a command which will ask for a password, and then allow him to record the pin |
19:38.25 | DEEZED | thus creating a new extension that plays the sound file he recorded |
19:38.38 | bjohnson | so make an exten to record a sound file .. then make a bash script or something to edit the extensions.conf file to make an exten to that |
19:38.52 | vaewyn | fairly easy to do... but I would turn it around and make him dial *7 first... and then record |
19:38.59 | bjohnson | don't forget to check that you're not overwriting another extension |
19:39.18 | bjohnson | and he'll need a way to delete them too I guess |
19:39.25 | *** join/#asterisk DannyF (~dannyf@h27n3c1o848.bredband.skanova.com) |
19:39.26 | DEEZED | oh ok... so this would be more of a linux feature then an asterisk feature |
19:39.28 | vaewyn | bjohnson: or just make them all in one range and never touch the extensions.conf... use an AGI to drive it |
19:39.32 | DEEZED | to modify the extensions file |
19:39.41 | bjohnson | DEEZED: combination of both |
19:39.47 | DEEZED | ic.. |
19:40.02 | bjohnson | DEEZED: or agi which allows whatever programming language you want |
19:40.10 | vaewyn | hmm.. wait... don't even need a program... js |
19:40.15 | DannyF | evening folks |
19:40.22 | Qwell | You could do all that in the dialplan... |
19:40.39 | Qwell | You can force Record to give a specific filename, can't you? |
19:40.57 | vaewyn | Yep... trivial |
19:41.07 | Qwell | So, no agi or anything needed really |
19:41.21 | bjohnson | the recording isn;t the problem .. it's the making of a new extension to play it back that needs something that can edit text files and reload the dialplan |
19:41.27 | DEEZED | yeah.. but the idea is to be able to create extensions on the fly |
19:41.33 | Qwell | no dialplan reload needed |
19:41.53 | vaewyn | hahaha... 2 liner :} |
19:41.54 | vaewyn | exten => XXXXX,1,Play(${EXTEN}) |
19:41.54 | vaewyn | exten => *7XXXXX,1,Record(${EXTEN}) |
19:41.55 | Qwell | When an extension(pin) is dialed, check the existance of ${EXTEN}.wav |
19:42.07 | Qwell | and yeah, if it exists...do that |
19:42.10 | vaewyn | and a 'i' to catch wrong ones |
19:42.28 | Drel | Does the 'Asterisk Community' recommend Polycom IP 500 phones in general? There seems to be some mixed opinions on voip-info.org, so I thought I'd run this by #asterisk. :-) The phones would be used in a small-office setting, but there'd be very limited tolerance for misbehavior/need for resets/call completion failure. |
19:42.40 | bjohnson | ahh .. just play it and if it doesn't exist .. you get nada |
19:42.42 | vaewyn | ok... so 3 lines |
19:42.44 | vaewyn | exten => XXXXX,1,Play(${EXTEN}) |
19:42.45 | vaewyn | exten => *7XXXXX,1,Record(${EXTEN}) |
19:42.45 | vaewyn | exten => i,1,Play(no-such-home) |
19:42.53 | DEEZED | wow thanks |
19:42.54 | Qwell | vaewyn: yeah, something like that |
19:42.54 | bjohnson | what about overwriting existing files? |
19:42.55 | vaewyn | Drel: They rock |
19:43.02 | Qwell | bjohnson: fuck em :p |
19:43.03 | DEEZED | thats going to help me |
19:43.34 | bjohnson | make sure he doesn't type a pin for a file he wants to keep |
19:43.36 | Drel | vaewyn: Are they at the "fire and forget" stage, where you can set them up, plug them in, and unplug them a year later for an office move, without doing anything but making and receiving calls in between? |
19:44.17 | vaewyn | Drel: pretty much |
19:45.18 | vaewyn | Drel: we run a couple call centers with them and they are drop and forget |
19:45.40 | *** join/#asterisk _chad (~Chad@c-24-6-142-55.hsd1.ca.comcast.net) |
19:45.59 | shido6 | im going to smack Mr spandsp |
19:46.00 | _chad | anyone know any good PSTN termination in california? pref bay area? |
19:46.13 | _chad | and on the cheap side |
19:46.36 | Drel | vaewyn: Do you mind my asking what the general call center setup is? These would be used on a 10/100 Mbps ethernet network, all connected to a 10/100 switch. |
19:46.50 | Drel | DHCP assigned via a Netgear firewall. |
19:47.01 | shido6 | how many simultaneous calls, Drel? |
19:47.16 | vaewyn | Drel: 10/100 Cisco... 35 phones per |
19:47.52 | *** join/#asterisk ikey (ikey@220.226.16.105) |
19:48.20 | Drel | vaewyn: Do you use static ip allocation or dhcp? |
19:48.40 | emrah | Please, anyone can help me with Asterisk Calling Card application? |
19:49.06 | vaewyn | Drel: ultra long lease dhcp |
19:49.07 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
19:50.06 | *** join/#asterisk bah (048830696@AC9AEC59.ipt.aol.com) |
19:50.15 | PBXtech | if i only get audio in 1 side of the convo, is that a firewall prob? only happens to 1 carrier |
19:50.32 | Drel | shido6: per phone or on the LAN? For the former, two, I guess, with one call on hold, for the latter, maybe 8-10 max? |
19:51.55 | emrah | Why no one answer? |
19:52.43 | harryvv | emray check your msg |
19:52.49 | Drel | emrah: Why not ask your specific question instead? |
19:53.36 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
19:53.56 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
19:54.22 | emrah | I've asked it before, sorry |
19:54.27 | emrah | My question is: |
19:54.49 | emrah | How is it possible to incrise the time before the timeout comes with Astcc, when we hear "The number is not ansering" |
19:54.54 | harryvv | emrah did you get my message? |
19:55.13 | malverian | How can you give on hold music while asterisk is trying to contact a channel? |
19:55.15 | emrah | harryvv: nop :S I will search it |
19:55.57 | malverian | Eg, while the line is ringing, or is this not possible? |
19:56.45 | emrah | YES |
19:56.57 | emrah | sorry for the capslock |
19:56.58 | hardwire | :) |
19:57.09 | _asr | any polycom ip500 users in the house? |
19:57.11 | hardwire | making a very old school *@home install :) |
19:57.18 | *** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com) |
19:57.25 | Drel | _asr: You must be my echo ;) |
19:57.28 | bjohnson | malverian: after * has answered the call .. you can use 'm' for the dial command |
19:57.40 | harryvv | hardwire thats cool |
19:57.53 | hardwire | it owuld be nice .. you know .. |
19:57.54 | emrah | I'm lost with this all messages. Anyone have the anser of my quesiton |
19:57.58 | hardwire | too bad I can't uinterrupt |
19:58.12 | emrah | the answer of my question* |
19:58.15 | _chad | man my latency makes voip useless from here.. ugh |
19:58.21 | _chad | anyone know of good termination in california? |
19:58.23 | hardwire | _chad: where are you? |
19:58.28 | _chad | hardwire, bay area |
19:58.32 | bjohnson | I "think" that is possible .. have you looked at the examples using CONSOLE= |
19:58.35 | hardwire | whats the latency |
19:58.40 | hardwire | I have do deal with 600+ ms |
19:58.43 | *** join/#asterisk truescot (~na@a62-216-27-177.adsl.cistron.nl) |
19:58.47 | _chad | hardwire, jesus, how many hops? |
19:58.52 | hardwire | one |
19:58.52 | bjohnson | _chad: some voip providers have west coast servers |
19:59.13 | bjohnson | I was told that was why my voipjet lagged excessively .. they're west coast |
19:59.27 | bjohnson | hardwire: sat? |
19:59.29 | _chad | hardwire, I have 15 hops to my colo (about 80ms average), and then another 13 from the colo to nufone (60ms average) |
19:59.30 | hardwire | yeh |
19:59.33 | malverian | bjohnson: Sample dialplan? |
19:59.34 | _chad | bj, awesome i'll take a look |
19:59.54 | hardwire | _chad: are you doing nufone to nufone calling? |
19:59.57 | _chad | hardwire, how do you accomidate that? the jitter buffer? literally every call i make/rcv *crunches* out after 5 minutes |
20:00.03 | emrah | Anyone please can anser my question? |
20:00.06 | _chad | hardwire, nufone to pstn |
20:00.14 | hardwire | its weird you are having problems. |
20:00.16 | _chad | hardwire, using them to dial out and rcv calls |
20:00.31 | bjohnson | malverian: exten=>s,1,Answer() |
20:00.36 | hardwire | thats a little further away |
20:00.36 | _chad | hardwire, I have not been able to complete one real call for more than a minute since i got in with them |
20:00.48 | _chad | hardwire, wheres your * box located? |
20:00.54 | hardwire | in Anchorage. |
20:00.59 | bjohnson | exten=>s,2,dial(sip/1000,60,m) |
20:01.05 | malverian | Ah... ,m |
20:01.10 | malverian | Automatically does on hold music? |
20:01.19 | _chad | hardwire, the only thing I can think of is that I'm connecting via this crappy cable modem to my colo in texas, which is then going to them in michigan |
20:01.26 | bjohnson | malverian: I think so |
20:01.30 | _chad | hardwire, how do you handle that amount of latency on your box? |
20:01.38 | hardwire | _chad: yeh.. crappy ass routing could be an issue :) |
20:01.48 | emrah | Anyone have an idea for an interesting provider for pstn calls? Voipjet have bad routing to Switzerland mobile |
20:01.53 | Drel | does anyone here use wireless bridges -> ip phones on desktops in combination with 802.11g wireless lan? |
20:02.03 | jsharp | long latency isn't going to kill a link. Massive changes in latency will though. |
20:02.13 | _chad | bj, you don't know the switch voipjet uses so i can traceroute it do you? |
20:02.13 | hardwire | I have low jitter on our VSAT |
20:02.15 | emrah | Very big latency too |
20:02.22 | hardwire | but over starband I have a shrinking jitter buffer set up |
20:02.28 | bjohnson | emrah: if you are in Switzerland .. try looking for a Swiss based voip provider |
20:02.29 | hardwire | but really.. you shouldn't need anything drastick |
20:02.33 | _chad | hmm |
20:02.43 | hardwire | _chad: why all the hoops and ladders? |
20:02.50 | _chad | any ideas for me, I did enable the jitter correction.. what else might i take a look at |
20:02.53 | emrah | bjohnson: Switerzland based VoIP company are very expensive |
20:03.00 | _chad | hardwire, how do you mean? :) |
20:03.06 | hardwire | _chad: you are in the bay area |
20:03.07 | bjohnson | _chad: 216.118.117.46 |
20:03.10 | _chad | hard, yah |
20:03.14 | _chad | bj, thanks |
20:03.17 | hardwire | and you are doing somethign in texas too |
20:03.23 | hardwire | which then goes to nufone |
20:03.30 | _chad | i'm in cali, my colo running * is in texas, then to nufone yeah |
20:03.52 | hardwire | ok |
20:03.58 | bjohnson | _chad: I get 45ms but it goes > 2000 multiple times a day when my other voip providers do not |
20:03.59 | hardwire | why are you doing that? |
20:04.36 | bjohnson | _chad: surely you have a local isp |
20:04.52 | _chad | hardwire, originally just ended up that way.. we have an ev1server box out there for our web stuff.. stuck * on there when I started geeking with it... ended up going w/ nufone because of the rates and the reccomendations i read on the mailing list |
20:05.07 | hardwire | ok |
20:05.14 | _chad | bj, yeah i connect locally via a comcast cable modem, then to my * box in texas, then nufone |
20:05.20 | bjohnson | ahh .. move * to a more sane server |
20:05.32 | hardwire | _chad: or have a (*) box at your house |
20:05.37 | hardwire | or a sip proxy that will do what you want |
20:05.38 | bjohnson | exactly |
20:05.49 | hardwire | but |
20:05.59 | hardwire | if you want all incoming to go to your (*) box in texas |
20:06.02 | hardwire | that could be an issue |
20:06.14 | hardwire | I bet sip "REDIRECT" can help you a lot |
20:06.20 | _chad | HMM |
20:06.23 | hardwire | HMM! |
20:06.31 | bjohnson | HMS |
20:06.33 | hardwire | _chad: what about you calling non nufoners |
20:06.35 | hardwire | off of your colo |
20:06.37 | _chad | that essentially snaps the call free of my * box and shoots it right to my sip phone yah? |
20:06.44 | hardwire | say.. have somebody else register w/ your account and call them |
20:06.48 | hardwire | and see if its just as shitty |
20:06.57 | hardwire | how well do echo tests turn out |
20:06.58 | bjohnson | _chad: no .. run your own * server |
20:07.00 | _chad | hardwire, i've only had calls w/ non nufoners |
20:07.08 | *** join/#asterisk pfn (500@netblock-66-245-252-239.dslextreme.com) |
20:07.15 | _chad | bj, ahh it redirects from the texas colo * box to another local * box then? |
20:07.25 | hardwire | redirects the session |
20:07.33 | hardwire | I dunno how the auth works on things like that however |
20:07.44 | hardwire | but I would run my own (*) server |
20:07.50 | hardwire | at your house |
20:07.55 | _chad | http://www.pastebin.com/267115 |
20:08.03 | _chad | lol didnt realize that pastebin was only for php code |
20:08.06 | _chad | but you get the idea |
20:08.17 | hardwire | its not |
20:08.19 | _chad | the first traceroute is to nufone from my local machine, the 2nd is to voipjet |
20:08.45 | hardwire | it might have something to do with your box rebelling against the name you gave it. |
20:08.51 | hardwire | thats my first hunch |
20:08.55 | hardwire | gost in the machine syndrome. |
20:08.57 | hardwire | ghost :) |
20:09.01 | _chad | hahahaha |
20:09.04 | _chad | dr frankendoodle? |
20:09.05 | _chad | lol |
20:09.16 | hardwire | then again I am on a machine I called bastard |
20:09.24 | hardwire | our routers name is fauker |
20:09.44 | hardwire | all named after planes :) |
20:09.47 | hardwire | but we chose the bad ones. |
20:09.53 | _chad | lol |
20:10.06 | _chad | grabbing a traceroute to my server |
20:10.23 | hardwire | I don't think hops is your issue |
20:10.23 | hardwire | heh |
20:10.41 | hardwire | mm |
20:10.42 | hardwire | beer |
20:10.44 | _chad | http://www.pastebin.com/267116 |
20:10.48 | _chad | no, you don't think so? |
20:11.02 | _chad | that last paste is from my local machine to my colo, and then from my colo to nufone |
20:11.17 | _chad | hardwire, whats your best guess? |
20:11.25 | hardwire | GITM |
20:11.39 | hardwire | traceroute from (*) to 209.193.36.94 |
20:11.43 | hardwire | and find out your latency |
20:11.52 | hardwire | thats me in anchorage |
20:12.29 | hardwire | or you aren't paying enough so they just enjoy plugging/unplugging you at random |
20:12.30 | _chad | http://www.pastebin.com/267118 |
20:12.43 | _chad | lol probably both |
20:12.56 | hardwire | I would take your phone |
20:13.00 | hardwire | and connect it directly to nufone |
20:13.02 | hardwire | and try to call |
20:13.02 | PBXtech | is there some command that will play a random .gsm file ? |
20:13.22 | _chad | hardwire good idea, let me give it a go |
20:13.34 | bjohnson | _chad: forget the Texas * server .. go directly from voip provider to your house |
20:13.59 | _chad | bj, what about inbound calls though? |
20:14.12 | _chad | would still need to go through the great state of texas :D |
20:14.14 | bjohnson | _chad: where do you need it to ring? |
20:14.27 | _chad | home office, but it has to hit the pbx first and whatnot |
20:14.35 | bjohnson | _chad: put a * server as close to that location as possible |
20:14.38 | _chad | menu's prompts, etc |
20:14.46 | _chad | bj, better to have it closer to me than the termination? |
20:14.47 | bjohnson | why does it need to hit the texas pbx at all? |
20:15.13 | bjohnson | _chad: wouldn't matter if it were a straight network path .. but it isn't |
20:15.25 | bjohnson | _chad: minimize the ms |
20:15.37 | bjohnson | _chad: run the ivr from home if low load |
20:16.18 | _chad | maybe i have a fundamental flaw with my thinking... i thought it had to go from the texas pstn to my texas colo "welcome to such and such company, press 1 for sales, 2 for support, 3 for burgers", they hit "1", then it gets routed from texas through my cable modem in cali and my sip phone here rings |
20:16.53 | bjohnson | you have a texas pstn or a texas voip did? |
20:18.31 | truescot | can anyone tell me what wireless phones are best with asterisk? |
20:18.33 | harryvv | after watching supersizeme I dont like them as much now :) |
20:18.53 | _chad | bj :) |
20:19.04 | jsharp | _chad: You can get a closer voip provider that can probably get you texas DIDs. |
20:19.09 | bjohnson | _chad: try running a traceroute directly to nufone |
20:19.21 | _chad | bjohnson, from my local machine right? |
20:19.25 | Blissex | _chad: there may be indeed a fundamental flaw with your thinking... |
20:19.25 | bjohnson | yes |
20:19.42 | *** join/#asterisk ACiDV (~joel@iteckGW.infoteck.qc.ca) |
20:19.44 | _chad | blissex, lol hopefully you guys can help me get this axe out of my forehead :) |
20:19.50 | bjohnson | you have 76 ms from home to Texas .. cut that down or even route around it if possible |
20:19.58 | _chad | bj, http://www.pastebin.com/267115 |
20:20.11 | _chad | first set is to nufone from local, 2nd is voipjet |
20:20.28 | terrapen | i wish nufone sold texas DIDs |
20:20.30 | ACiDV | Hmmm, weird IAX problem... one way audio.. 2 servers linked in IAX, if I have a sip phone on each server calling other, I have one way sound, if both phone are on the same server, all work ok |
20:20.34 | jpe | hello all, Im still here trying to get dids working with aah 0.8. been at it all day to no avail |
20:20.34 | terrapen | IAX.cc was a total wash for us |
20:20.35 | Blissex | _chad: in theory, hopefully the caller and your home phone get connected directly, if REINVITE is possible |
20:21.13 | bjohnson | if you truely have a Texas pstn connection .. then you will want to have a texas * .. and route calls from that to home .. but you may want to route outgoing from hoem directly to voip provider and incoming from voip provider should go to wherever is faster for end destination of most calls (also depends on call volume) |
20:21.50 | _chad | bj, hmm |
20:22.40 | Blissex | _chad: the problem with your sewtup is that PSTN does not do REINVITE... :_0 |
20:23.09 | _chad | bj, my pstn termination service is in michigan (via a 1800#), if I understand you correctly I should ditch them, setup a pstn closer to my colo in texas... or ditch my colo in texas and setup a local machine or a closer machine running * and keep the pstn in michigan? |
20:23.31 | Blissex | _chad: depends on what you want to do! |
20:23.34 | _chad | :D |
20:23.50 | _chad | blissex, low volume office use (2-5 lines max) |
20:23.59 | _chad | blissex, without my phones crapping out :( |
20:24.00 | bjohnson | _chad: I thought you said you had a Texas pstn |
20:24.02 | Blissex | _chad: if you want your calls to end up on your SIP phone at home, put Asterisk and PSTN termination at home. |
20:24.21 | _chad | bjohnson, * box is at a server in texas, i use nufone in michigan for the termination |
20:24.32 | Blissex | _chad: that's a mad mad setup... |
20:24.40 | _chad | blissex :D hahaha :D |
20:24.43 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
20:24.52 | _chad | this is my first attempt, at least i'm learning :D |
20:24.53 | _chad | lol |
20:25.02 | bjohnson | _chad: btw .. your home isp is killing your home use .. looks like you either go through AT&T or Verizon and both are adding about 50ms to your routing |
20:25.07 | PBXtech | how does the Random command work? |
20:25.17 | Blissex | _chad: and if PSTN terminates at Nufone, then Asterisk in TX should REINVITE Nufone to your SIP phone at home. |
20:25.43 | _chad | bjohnson, yeah this is through comcast cable modem.. i've tried traceroutes on other dsl providers and gotten about the same unfortunately |
20:26.20 | bjohnson | _chad: you're going to have trouble with voip to/from your house until your get faster traceroutes |
20:26.23 | _chad | blissex, reinvite would essentially take the texas * out of the picture and connect the call directly from nufone to me correcT? |
20:26.37 | bjohnson | _chad: it looks like the bottleneck is somewhere around LA |
20:26.40 | Blissex | _chad: yes, ideally, if Nufone and your home SIP phone are compatible. |
20:26.51 | *** join/#asterisk khaled (~khaled@CPE000625627f09-CM001225d88b2e.cpe.net.cable.rogers.com) |
20:26.56 | *** part/#asterisk khaled (~khaled@CPE000625627f09-CM001225d88b2e.cpe.net.cable.rogers.com) |
20:27.05 | Blissex | _chad: but if the problem if your home line as bjohnson suspects, you can't do anything. |
20:27.25 | _chad | okay so a reinvite if the problem is here would be like polishing the brass on the titanic yah? |
20:27.26 | bjohnson | well .. he can save 50ms by not routing through texas first |
20:27.31 | Blissex | _chad: do you have the same problem with outgoing calls as with incoming ones? |
20:27.37 | _chad | yep |
20:27.40 | bjohnson | but that won't be enough .. he still has 80ms + |
20:27.55 | tzanger | what's wrong with 80ms latency |
20:27.58 | mutilator | 80ms is fine.. |
20:28.08 | _chad | blissex, but both incoming and outgoing are routed through my * and then to nufone |
20:28.19 | _chad | tz/multilater, I have about 30 hops of it |
20:28.23 | Blissex | _chad: as in, routed or resolved? |
20:28.31 | mutilator | uhm.. |
20:28.38 | mutilator | 30x80ms? |
20:28.48 | Blissex | _chad: routed as in the data travels to TX and then to to Nufone? |
20:28.50 | tzanger | 2400ms is a bit much yes |
20:28.53 | mutilator | you live in somolia with a satellite phone? |
20:28.55 | _chad | blissex, its my understandig its routed yeah |
20:28.57 | bjohnson | sorry .. wrong numbers .. he can end up with 78 ms if he drops the connection to texas which makes the trip to nufone add up to 143ms |
20:29.03 | _chad | multilator hahaha |
20:29.05 | tzanger | 143ms isn't much |
20:29.09 | tzanger | it's noticeable but acceptable IMO |
20:29.18 | tzanger | you're gonna have around that with a jitter buffer anyway |
20:29.23 | mutilator | if i get >100ms over wireless it breaks pretty bad |
20:29.25 | bjohnson | look at his pastebins |
20:29.28 | mutilator | 80ms is tolerable |
20:29.30 | tzanger | uh |
20:29.33 | tzanger | latency doesn't cause breakups |
20:29.35 | tzanger | jitter does |
20:29.44 | _chad | hmm |
20:29.46 | mutilator | hence i say wireless |
20:29.56 | mutilator | where both latency and jitter are affected |
20:30.08 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
20:30.22 | *** join/#asterisk clive- (~pirch@rrba-146-99-95.telkomadsl.co.za) |
20:30.35 | Blissex | _chad: why don't you try with some other provider? You might find one much nearer to you in network terms... |
20:30.53 | _chad | blissex, in regards to the colo or the termination (or both? ) :) |
20:30.55 | tzanger | mutilator: but you're giving latency numbers |
20:31.01 | Blissex | _chad: and then make sure you can do REINVITE, so you dont have the leg to TX. |
20:31.01 | mutilator | yes |
20:31.02 | tzanger | which do NOT affect breaking |
20:31.21 | Blissex | _chad: well, _ideally_ both. |
20:31.30 | mutilator | for ME it's a very good estimate of breaking |
20:31.32 | Drel | We currently have two analog telephone lines. I am putting together an Asterisk box, and looking at the Digium Wildcard TDM400P to bring those two analog phone lines into the Asterisk system (using 2 FXO modules). We also have 4 Polycom analog telephones that are currently connected to the analog phone lines. How many FXS modules would I need to support these four telephones? Would one... |
20:31.34 | Drel | ...be sufficient, or would I need four? Could I disconnect our internal telephone wiring from the PSTN, connect the incoming lines to the FXO modules, and run a cord from the FXS module to the internal telephone wiring with all four phones plugged in? Sorry for the lengthy question! |
20:31.40 | mutilator | i have 100 or so customers on our wireless network across northern michigan and thats what i've come to find |
20:32.05 | tzanger | Drel: you need an FXS port for each analogue phone, just like you need an FXO for each POTS line |
20:32.19 | Blissex | _chad: if you have setup things so that REINVITE works, then it is more important for the provider to be ''near'' your home, because all your Asterisk instance then is just tell Nufone to connect to your home phone. |
20:32.30 | nestAr | anyone build CVS-HEAD today? |
20:32.45 | mutilator | the latency on wifi will determine a lot about your 'jitter' atleast for me it does |
20:32.49 | _chad | blissex, I see, and then for outbound calling I would just connect to nufone directly? |
20:32.53 | harryvv | mutilator how are you doing |
20:32.56 | tzanger | mutilator: odd but ok, I have not specifically tested it |
20:32.57 | Drel | tzanger: Darn. That sounds like a wiring nightmare! I guess ATAs are in order instead of FXS modules. |
20:33.00 | nvrswork | tzanger, if you connected 2 phones to one FXS module, would it blow? |
20:33.03 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
20:33.04 | nvrswork | blow the card |
20:33.05 | Blissex | _chad: if you want, or you get it handled by Asterisk too. |
20:33.09 | harryvv | tzanger by any chance is that releated to udp checksum error? I had a string of them one and causes a 15 second delay in a connecting into the states during a conversation. |
20:33.12 | tzanger | nvrswork: no, but they will have the same extension |
20:33.16 | mutilator | harryvv.. fine and you? |
20:33.20 | _chad | blissex, and it could reinvite in the same way and connect me directly to nufone? |
20:33.21 | tzanger | harryvv: is what related to that |
20:33.26 | Qwell | nvrswork: it also depends on the REN value, if it can even do that |
20:33.32 | harryvv | jitter and breakup |
20:33.32 | tzanger | Drel: 6 lines is a wiring nightmare? |
20:33.34 | Blissex | _chad: again, if Asterisk handles only call setup, and not data transmission, where ti is does not matter a lot. |
20:33.35 | nvrswork | tzanger, how many phones can one FXS support in parallel |
20:33.42 | tzanger | nvrswork: depends on the REN of each phone |
20:33.56 | iceyp | hi guys, could someone on fwd, please test my connection on 265744 , i've recently change the port on my asterisk to 5070 and want to ensure it's still working, i've also changed the context |
20:34.05 | Qwell | iceyp: one sec |
20:34.08 | mutilator | avg of 5 |
20:34.11 | iceyp | thanks qwell |
20:34.17 | PBXtech | how does the Random command work? |
20:34.19 | tzanger | mutilator: I am guessing that latency has ntohing to do with it but rather when you get higher latency links it's because the wirless network is more congested, leading to higher latency due to more collisions and collision avoidance |
20:34.25 | Blissex | _chad: well, yes as to connect directly. Asterisk is a fully symmetrical (even too symmetrical) switch. It does not know which end is the pprovider and which end if the home phone... |
20:34.27 | harryvv | mutilator do you have windows and would care to test a voip connection to my server with xlite? |
20:34.28 | nvrswork | REN values? |
20:34.31 | Qwell | iceyp: 1 ring - fast busy |
20:34.34 | bjohnson | _chad: if you can cut Texas out of the routing you can save 50ms on your calls to your house. You an always route outgoing directly from your house sip phones .. and would be a good test for call quality if you hosted your * locally there |
20:34.38 | _chad | blissex, ahh good stuff |
20:34.46 | Drel | tzanger: Well, a bad headache anyway :) We're going to have to run ethernet anyway, I think (we're currently using 802.11g to all desktops, so our current wiring (or lack thereof) makes any wiring addition look unpleasant. |
20:34.55 | tzanger | ahhh |
20:35.08 | iceyp | qwell: mmm, let me lookie |
20:35.09 | mutilator | yea tzanger, thats whats going on |
20:35.16 | Qwell | iceyp: I had verbose off, lemme try again, see what it says on my end |
20:35.18 | mutilator | but it's pretty much a direct correlation on wireless |
20:35.24 | _chad | bj,bliss, good deal |
20:35.26 | mutilator | whereas dsl it's not |
20:35.35 | Blissex | _chad: all Asterisk does is given an extension, find the address for that extension, and then tells the requesting endpoint to setup the call to that address, ideally. |
20:35.35 | *** join/#asterisk ariel_ (~Ariel@ip67-93-229-222.z229-93-67.customer.algx.net) |
20:35.36 | _chad | bj/bliss: would upping my jitter buffer help some as well? |
20:35.37 | Qwell | iceyp: yeah, it hits FWD, then dies at you |
20:35.40 | Drel | This is an old bldg w/ no ceiling panels (we have 17' ceilings instead) and an open layout, so running wiring isn't easy. |
20:35.49 | tzanger | mutilator: I suppose... distance would give you more latency without more CSMA action :-) but it'd be long links |
20:35.52 | _chad | blissex, i see.. and take itself out of the picture |
20:36.00 | tzanger | seventeen foot cielings... NICE |
20:36.06 | tzanger | echo cho ho o... |
20:36.06 | mutilator | we've got links up to 15 miles on some customers |
20:36.06 | harryvv | I need somone to test a voice connection to a remote client. |
20:36.13 | mutilator | most run about 5 |
20:36.15 | Blissex | _chad: it will help if you are getting breakups, but it will make delay worse... |
20:36.30 | iceyp | Qwell can you try again plz |
20:36.37 | bjohnson | Drel: I thought I saw someone mentrion REN .. you only need one fxs per phone if you want them to act as independant extensions |
20:36.41 | iceyp | oops |
20:36.42 | iceyp | Apr 5 08:36:17 WARNING[99164]: chan_iax2.c:4300 iax2_register: Host 'iax2.fwdnet.net/01' not found at line 12 |
20:36.43 | mutilator | some backbone links run 20+ miles |
20:36.44 | iceyp | 1 sec |
20:36.47 | Qwell | iceyp: no go |
20:36.48 | _chad | http://www.pastebin.com/267135 w/in the norm? |
20:37.32 | DannyF | anyone played with FastSMS btw? |
20:37.32 | bjohnson | _chad: jitterbuffer adds delay to the beginning of the call usually |
20:37.37 | Drel | bjohnson: having the legacy analog phones we've invested in relegated as "extra"/floating phones that share a single extension would probably be acceptable. |
20:37.48 | tzanger | bjohnson: it adds delay thoughout the call, that's its purpose |
20:37.53 | Blissex | _chad: yes, if Asterisk cannot take itself out of the picture, because the calling and called endpoints cannot connect directly, then it will switch to transparent mode, in which it will store-and-forward data from endpoint to another, over two connection isntead of one. |
20:37.58 | facek_ | hm |
20:38.00 | *** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net) |
20:38.04 | Drel | bjohnson: In that case, do you see any problem with having 4 phones on a single FXS? |
20:38.19 | tzanger | Drel: they will act as one extension... is that what you want? |
20:38.32 | _chad | Ah, bliss- so if I understand you.. the jitter buffer is not really even a factor once the reinvite is made? |
20:38.42 | *** join/#asterisk RubyT (~joeblow@lan.mobilcom.net) |
20:38.57 | Drel | tzanger: It's not ideal, but if it saves a few hundred dollars on either scrapping the analog sets or investing in ATAs, it might be acceptable. |
20:39.02 | Blissex | _chad: the jitter buffer in _Asterisk_ is no longer an issue _if_ the reinvitation succeeds. |
20:39.14 | tzanger | Drel: well you can't have it both ways, so it's your choice. :-) |
20:39.21 | bjohnson | Drel: depends on the ata and how much voltage the phone draw from the line. I have 6 powered phones and 2 non-powered phones on one SPA 3k fxs port |
20:39.46 | Blissex | _chad: but the endpoints themselves will usually have buffering too, for the same reasons why Asterisk has buffering too. |
20:39.57 | _chad | i see |
20:40.05 | Drel | bjohnson: The analog sets are Polycom SE-220 powered sets, we'd probably go with a Digium TDM400P |
20:40.25 | jpe | anyone have any insight as to why I cant get dids entered through amp to work via aah 0.8? |
20:40.31 | bjohnson | Drel: you could always do a mix .. often not all the extensions need to be independant (think home system) but in business .. you usually want them to be independant (how does it look if on with a customer and someone else picks up to order pizza) |
20:40.41 | *** join/#asterisk klictel (~klictel@207.107.208.137) |
20:40.56 | _chad | changing my cisco right now to dial out directly |
20:41.31 | klictel | hi all |
20:41.43 | iceyp | Why would i get this when trying to receive a call from FWD |
20:41.44 | iceyp | Apr 5 08:41:21 NOTICE[99654]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121 |
20:41.45 | Blissex | _chad: Asterisk also has buffering because it has _three_ modes of operation: one is resolve the call and then reinvite the endpoints to connect directly to each other. Another is transparent mode, where it passes data between the endpopints. The third is when it acts an endpoint itself, that is terminates call in itself, like when it does voicemail. |
20:41.53 | iceyp | 65.39.205.121:4569 265744 202.150.105.150:4569 60 Registered |
20:41.56 | mutilator | bjohnson: least you know that there is pizza on it's way :P |
20:41.56 | Drel | bjohnson: Well, as long as the line light was illuminated showing the extension to be in use, I don't think it would be an issue, it wouldn't be any different from our current situation. The phones are two line, they show which line is in use, and if one is in use, they pick up the other line when you pick up the handset. What I'd probably do is buy two FXS modules and hook those up to the... |
20:41.58 | Drel | ...internal phone wiring so that each module handled a line on the analog sets. |
20:42.03 | Drel | mutilator: hehe. |
20:42.12 | facek_ | what i need to allow my peers connect by by h323? |
20:42.37 | Blissex | facek_: it is a FAQ... Check the Wiki. |
20:42.39 | jakepdev | facek - you need JerJer |
20:43.14 | facek_ | jakepdev right ;] |
20:43.42 | RubyT | Is there a way to have a TDM400 FXO port detect if a working phone line is connected to it? |
20:43.46 | iceyp | Apr 5 08:43:15 NOTICE[99654]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121 <--- Why do i receive this error when trying to receive a call via FWD? |
20:43.46 | *** join/#asterisk platcd (Atropine@S0106000f3d37a96d.va.shawcable.net) |
20:43.56 | _chad | bliss/bj/etc its amazing how much you guys know about this stuff |
20:44.27 | hardwire | its because they have no women! |
20:44.28 | hardwire | :) |
20:44.33 | mutilator | ooo disssss |
20:44.38 | hardwire | I would be so insanely smart if a woman wasn't stealing all that energy |
20:44.40 | bjohnson | Drel: or .. with an internet connection, and fxs per phone, and some dialplan routing you could give each phone access to what would like like it' own independant line for outgoing calls. In the end .. you decide what works best for you .. ATA's come for as little as $35 USD per fxs port |
20:45.02 | _chad | hardwire, haha :D |
20:45.26 | Qwell | hardwire: I wouldn't be able to find my socks. :p |
20:45.32 | Blissex | _chad: <hardwire> may me more right than you think... Freud and sublimation :-). |
20:45.33 | hardwire | haha |
20:45.53 | platcd | good day all. I just starting reading about Asterisk, Digium hardware etc, and I must say I am stoked to build a PBX for my small home office. What do I need hardware wise. I simply require 2 unique lines such that one operator can be talking while another call can come in... Are there boards with simple POTS interfaces ? What do I need to get started ? |
20:46.07 | Drel | bjohnson: Good to know. Basically, I need to come up with some suggestions and pricepoints for my boss to consider, so it's good to know what the alternatives are! |
20:46.11 | bjohnson | Drel: of course you could also stick regular cordless phones on a fxs |
20:46.23 | sudhir492 | facek_: I have recently compiled h323 with CVS head. openh323 1.17.1, pwlib 1.8.1. Works well so far |
20:46.32 | Blissex | Freud said that sublimated horniness is the very motor of civilization. Or at least geekyness I say :-) |
20:46.40 | _chad | So all I need to is punch in my nufone register information (from iax.conf) into my cisco 7940 sip preferences (as the proxy and whatnot) and i would be connecting directly yah? |
20:46.55 | tzanger | _chad: your 7940 can do IAX2? |
20:47.07 | iceyp | Qwell you there? |
20:47.10 | Qwell | yeah |
20:47.12 | Qwell | try again? |
20:47.13 | Blissex | _chad: if you want, but I suppose it's better for you to resolve calls via your Asterisk anyhow. |
20:47.18 | iceyp | you using fwd via iax? |
20:47.20 | bjohnson | _chad: register tells the only end what IP address to send incoming calls to |
20:47.21 | Qwell | yeah |
20:47.27 | iceyp | can i see your iax.conf entry |
20:47.37 | iceyp | something is broken with mine |
20:47.54 | iceyp | i'm rejecting the calls |
20:47.55 | iceyp | Apr 5 08:46:59 NOTICE[99654]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121 |
20:47.57 | Drel | platcd: I'd recommend a Digitum TDM400P with two FXO modules to handle your incoming POTS lines. Keep in mind I have very little real world experience with Asterisk so far, though :) |
20:48.00 | iceyp | dont know why |
20:48.04 | bjohnson | _chad: I think you want to leave texas as your incoming * server until you do some testing on your home connection |
20:48.10 | Blissex | _chad: so on outgoing you can access all your menus/stored numbers/whatever logic you got. Phones are a lot dumber than Asterisk. |
20:48.42 | Qwell | iceyp: http://pastebin.ca/8803 |
20:48.43 | _chad | whew lot to learn |
20:48.47 | Blissex | _chad: also on incoming you want Asterisk to do voicemail etc. for you. |
20:48.58 | _chad | blissex, yeah exactly |
20:49.09 | _chad | blissex, reinvite dosn't take those elements out of the picture does it? |
20:49.20 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
20:50.01 | Blissex | _chad: no, reinvitation only handles the data transfer bit. All those elements are about call initiation etc. |
20:50.06 | bjohnson | if you decide your home connection can handle enough calls ok .. you can move your * server there and forget the texas location .. but as far as * is concerned, I think you should consider getting a colo closer to you or your voip provider |
20:50.30 | _chad | bj, definitely.. seems like i have just too many bends in the pipes |
20:50.36 | iceyp | Qwell you dont have the host iax2.fwdnet.net & user / pass? |
20:50.53 | Blissex | _chad: however bjohnson has a good point if your phone can do IAX2 and SIP, make sure it uses the one that your provider uses. |
20:50.54 | iceyp | do you provide your user/pass in the dial? |
20:51.00 | Qwell | iceyp: yeah |
20:51.44 | iceyp | can you try call me now 265744 |
20:51.44 | AgiNamu | yes, make sure you use the right protocol. that' |
20:51.48 | AgiNamu | is some good advice :) |
20:51.56 | _chad | blissex, aiee another concept entirely :) |
20:51.58 | AgiNamu | I had someone flash a phone to IAX2, then send me a SIP debug log and ask why it didn't work |
20:52.14 | clive- | AgiNamu what phone is this? |
20:52.18 | Qwell | iceyp: yeah, that worked |
20:52.18 | AgiNamu | anyone here have experience with PortaOne?\ |
20:52.22 | _chad | I don't believe this cisco 7940 supports IAX2 does it? I know I went through a grip to get it converted to SIp |
20:52.22 | AgiNamu | clive- PA168 |
20:52.27 | Qwell | "Please enter the extension of the user you wish to connect to" |
20:52.30 | bjohnson | nope .. just PortaPotties |
20:52.43 | neopher | got probs with x100p, I did load properly and i still get Apr 4 16:25:00 WARNING[3942]: Ignoring switchtype |
20:52.43 | AgiNamu | PA168 is the only phone I know of in mass production that supports IAX |
20:52.58 | bjohnson | _chad: no cisco phones currently support iax2 |
20:53.03 | bjohnson | none |
20:53.06 | AgiNamu | PA168 is a chip made by a Chinese company that's used a LOT |
20:53.07 | clive- | AgiNamu, I have th pa168 also....just doesnt do native transfer...yet |
20:53.20 | AgiNamu | clive- no. I might implement it some day |
20:53.24 | _chad | bj, k |
20:53.25 | AgiNamu | if they dont do it for v1.43 |
20:53.39 | AgiNamu | I added some more support to their 1.42 IAX firmwar |
20:53.48 | AgiNamu | and i will probably work on it a bit more soon |
20:54.08 | clive- | AgiNamu, are you the one doing the iax firmware? |
20:54.08 | _chad | bj/bliss, so 1) check to see if reinvite gives me the results I want.. if not 2) setup a local * box and 3) start trying closer pstn boxes |
20:54.13 | AgiNamu | i dont know what all is involved in native transfer. i'll have to read chan_iax2 a bit more |
20:54.14 | malverian | neopher: I get that as well, but everything still works fine. |
20:54.20 | _chad | that the gist of it? |
20:54.24 | AgiNamu | clive- no, they do it |
20:54.29 | AgiNamu | but I wrote some patches for them, to help |
20:54.31 | iceyp | Qwell thanks |
20:54.37 | AgiNamu | cause i needed, for example, POKE/PONG support |
20:54.40 | AgiNamu | so I wrote it, sent it to them |
20:54.45 | neopher | malverian: i am not able to make or receive calls via zap |
20:54.46 | AgiNamu | and they're including it in 1.43 |
20:55.04 | AgiNamu | clive- do you know much about IAX2 , so as to write firmware? if so, i'll send you the source and you can work on it |
20:55.11 | malverian | neopher: Probably an unrelated issue then. |
20:55.12 | jaiger | AgiNamu, they sent you their source code? |
20:55.16 | AgiNamu | yes |
20:55.36 | clive- | AgiNamu, I wish I had the time and knowledge to do it all myself, but I cant:( |
20:56.09 | AgiNamu | SteveK told me that their iax is based on an old libiax |
20:56.13 | AgiNamu | but with every line of code replaced |
20:56.33 | malverian | neopher: Do you have wcfxo linux module installed? |
20:56.43 | facek_ | what is the best way to divide outgoing calls by prefix to specifi channel to make call ? |
20:56.48 | clive- | AgiNamu, I am surprized no-one from Digium has partnered with atcom to create a good iax product...oh well, just have to wait for my native bridge transfer to happen |
20:56.53 | facek_ | and when channel is must other some other channel |
20:57.07 | AgiNamu | clive- when you say native transfer |
20:57.15 | AgiNamu | you refer to performing a transfer? or some kind of bridge? |
20:57.33 | AgiNamu | cause native bridging works fine. its just that the transfer button doesnt do anything |
20:58.05 | clive- | AgiNamu, native bridging,,,doesnt work for me, Woody said its the phone thats not transferring |
20:58.24 | AgiNamu | yea there is no transfer support |
20:58.35 | AgiNamu | but when i place a call, * natively bridges it to my gateway no problem. |
20:58.59 | clive- | AgiNamu what is your setup that allows the bridge?... |
20:59.12 | AgiNamu | PA168 -> Asterisk -> Asterisk -> Telica |
20:59.16 | AgiNamu | ( -> DS3) |
20:59.34 | hardwire | ooh |
20:59.36 | AgiNamu | my asterisk machine bridges just fine. |
20:59.40 | clive- | and the first asterisk transfers out of the loop |
20:59.54 | AgiNamu | no, no transfers. just native bridging |
20:59.59 | AgiNamu | i dont want that any ways |
21:00.02 | AgiNamu | fucks with my CDRs |
21:00.15 | AgiNamu | and im worried about the NAT issues (which is why im using IAX in the first place) |
21:00.46 | AgiNamu | maybe im misunderstanding. i see native bridging as when asterisk can just passthrough the frames |
21:00.48 | neopher | being that i am using kernal 2.6.9-5, do i need to do make linux26 instead of make |
21:01.07 | AgiNamu | and transfer when it actually can reconnect to another server |
21:02.40 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
21:02.55 | hajekd | is it possible to pass argument to agi script? |
21:03.33 | bkw_ | yes |
21:03.35 | bkw_ | go read the notes |
21:03.52 | bkw_ | [E|Dead]AGI(command|args): |
21:04.06 | bkw_ | and next time say Hi before you burst in to ask questions please |
21:04.51 | Qwell | Hi! |
21:05.09 | rvhi | any one knows how to extrace 'username' in 'from' header of a sip packet? |
21:05.23 | rvhi | e.g. From: <sip:123@test.com> |
21:05.35 | rvhi | i'd like to see 123 in one variable |
21:06.08 | rvhi | oops, |
21:06.10 | rvhi | Hi! |
21:06.51 | bkw_ | *SMACK* |
21:06.52 | bkw_ | hehe |
21:07.05 | *** part/#asterisk platcd (Atropine@S0106000f3d37a96d.va.shawcable.net) |
21:07.17 | bkw_ | I think thats already done for you |
21:07.22 | Blissex | rvhi: I think that's documented. |
21:07.28 | *** join/#asterisk KristinG (~KristinG@muppet.geekgirls.us) |
21:07.31 | jakepdev | ~docs |
21:07.33 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
21:07.33 | Blissex | rvhi: it is a prese3t variable. |
21:07.39 | bkw_ | rvhi, you have ${SIPURI} |
21:07.46 | KristinG | good afternoon |
21:08.02 | bkw_ | Blissex, thats not what he asked |
21:08.02 | Beirdo | ~beer |
21:08.03 | jbot | it has been said that beer is ummm, ummm good!, or good for you! |
21:08.06 | *** join/#asterisk AmigaMan (~fsck@cpe-68-203-212-229.rgv.res.rr.com) |
21:08.09 | Beirdo | good afternoon, KristinG |
21:08.26 | KristinG | can anyone tell me if there are any open source voice mail systems that would work with asterisk? |
21:08.28 | Beirdo | it has one built in |
21:08.32 | MikeJ[Laptop] | ~guatamala |
21:08.33 | jbot | guatamala is, like, small central america country jointly owned by the USA government and the Illuminati. |
21:08.40 | KristinG | comedian is lacking in a number of ways |
21:08.44 | bkw_ | KristinG, dude |
21:08.48 | bkw_ | asterisk has one already |
21:08.50 | Beirdo | well, it's open source :) |
21:08.50 | Blissex | Beirdo: she did not ask that, she asked if anyone could tell her... |
21:09.05 | Blissex | KristinG: the answer to your question is "yes". :-) |
21:09.14 | jakepdev | what you don't like you can change |
21:09.15 | Beirdo | Blissex: be nice :) |
21:09.16 | bkw_ | its app_voicemail.c |
21:09.19 | KristinG | I'm not a dude but yes, i am awarrrrrrrrrrrrrrre of the native app |
21:09.24 | rvhi | is sipusr only in head? |
21:09.38 | bkw_ | KristinG, YAY another girl in here |
21:09.44 | MikeJ[Laptop] | another? |
21:09.50 | bkw_ | Katty and KristinG |
21:09.52 | KristinG | there are features that my booooooooooooooooooooooss would like that asterisk doesnt offer |
21:09.55 | Beirdo | KristinG: "dude" is often used generically, but from your name, I was fairly sure you aren't a dude :) |
21:10.04 | bkw_ | KristinG, what is that? |
21:10.04 | Beirdo | so add them |
21:10.09 | KristinG | thank you :) |
21:10.11 | Beirdo | the source is right there :) |
21:10.18 | eKo1 | You should have used dudett |
21:10.22 | file[laptop] | cause it's easy once you know how it's done |
21:10.22 | jakepdev | ette |
21:10.23 | Blissex | KristinG: thats very unlikely, but... What is one such feature? |
21:10.25 | bkw_ | first lets hear what the PHB wants |
21:10.26 | file[laptop] | you can't stop now it's already begun |
21:10.39 | Beirdo | oh good point. |
21:10.44 | KristinG | playing back your recorded grettings |
21:10.47 | facek_ | anyone had create a good dialplan using AGI? or made in AGI without extensions.conf (only ro run some scripts by agi) |
21:10.51 | bkw_ | KristinG, thats there already |
21:10.57 | Qwell | NEXT!!! |
21:11.00 | Qwell | :p |
21:11.06 | KristinG | hold on |
21:11.07 | bkw_ | if not its a few lines of code to add it |
21:11.07 | KristinG | brb |
21:11.15 | bkw_ | but I'm sure it is |
21:11.21 | file[laptop] | bkw_: back to work you slacker! |
21:11.26 | bkw_ | WORK? |
21:11.28 | bkw_ | call 996 you hoe |
21:11.33 | file[laptop] | ;) |
21:11.38 | file[laptop] | nah |
21:11.40 | *** join/#asterisk objectivelogic (~tinyiko@196.46.67.175) |
21:11.42 | bkw_ | YES YOU WILL |
21:11.44 | jpe | calling from a cell into the syste, |
21:11.44 | bkw_ | and you WILL NOW |
21:11.52 | file[laptop] | I'm quite tired and am currently being in a semi-concious state on my bed |
21:11.53 | jpe | calling from a cell into the system |
21:11.56 | file[laptop] | THINKING OF YOU DEAR! |
21:11.56 | MikeJ[Laptop] | hoe... now that's not very nice... is that the way to make friends bkw_? |
21:11.58 | file[laptop] | :p |
21:11.58 | MikeJ[Laptop] | :D |
21:12.55 | Beirdo | stupid ebay arsehole didn't have the manual |
21:13.26 | rvhi | i looked at http://voip-info.org/wiki-Asterisk+variables, can't find the username field |
21:14.15 | luke-jr_ | Does any LC routing exist that does not require SQL? |
21:14.41 | Darwin35 | who did what to who |
21:14.47 | Blissex | rvhi: but you probably have found something from which you can extract it... |
21:15.01 | iceyp | should asterisk be able to handle a call in this way.... FWD -> Asterisk -> Ser -> Voip Phone ? |
21:15.14 | iceyp | My problem is when the call connects between asterisk and Ser, it dies on first ring |
21:15.22 | luke-jr_ | iceyp: ...why? |
21:15.32 | luke-jr_ | are * and SER on the same box? |
21:15.40 | KristinG | ok back |
21:15.52 | iceyp | luke-jr_ yes |
21:16.12 | KristinG | playback-options: rewind/pause/foward through a message |
21:16.18 | iceyp | diff ports obviously, 5060 for ser & 5070 for * |
21:16.19 | luke-jr_ | iceyp: I'd check to make sure SER is configured for a non-default port |
21:16.29 | KristinG | timestamp of rmessage |
21:16.45 | facek_ | can i use ChanIsAvail from AGI scripts? |
21:17.05 | luke-jr_ | iceyp: * dying on first ring sounds like its looping to me |
21:17.17 | KristinG | and notification, ie via, email, pager or remote phone |
21:20.02 | objectivelogic | need help getting my asterisk to terminate to POTS |
21:20.15 | Qwell | KristinG: It does that too |
21:20.32 | objectivelogic | no ringing,just says Spawn extension (from-sip, 2719, 1) exited non-zero on 'SIP/tinyiko-9d1d' |
21:20.32 | objectivelogic | <PROTECTED> |
21:20.32 | objectivelogic | <PROTECTED> |
21:20.32 | objectivelogic | <PROTECTED> |
21:20.41 | FirstSword | anyone heard of Sangoma's card? |
21:20.49 | file[laptop] | objectivelogic: ${EXTEN} not $EXTEN |
21:20.51 | Qwell | objectivelogic: ${EXTEN{ |
21:20.55 | Qwell | yeah, what he said |
21:21.11 | objectivelogic | ok, let me try that |
21:21.57 | KristinG | ok guess i have to pick the source apart then |
21:21.57 | AmigaMan | has anyone setup h.323 with g729? |
21:22.21 | KristinG | just to confirm though, there are no other packages that will work with it? |
21:23.16 | iceyp | ok, can someone do me a favour and dial 64273040757@max.fast.co.nz |
21:23.33 | Blissex | KristinG: no need usually to look at the source... |
21:23.46 | *** join/#asterisk Zipper_32 (~none@s207-6-25-182.bc.hsia.telus.net) |
21:23.54 | Blissex | KristinG: there are quite a few examples of even some obscure features online. |
21:24.14 | KristinG | oh? i'd love a link or two then |
21:24.51 | Blissex | KristinG: for example, from one such example I think I discovered that there can be two sections with the same name in 'sip.conf', one with 'type=peer' and one with 'type=user'. Amazing. |
21:25.03 | Blissex | KristinG: as usual, there is the Wiki... |
21:25.04 | Nivex | ~wiki |
21:25.38 | Blissex | KristinG: but if you do a clever Google search you can find a number of fairly sophisticated sample 'extensions.conf' that people have posted on the web. |
21:26.22 | Blissex | KristinG: by «clever Google search» i mean one that contains a few keywords that are likely to be present only in an 'extensions.conf' file. |
21:27.42 | KristinG | ok thanks |
21:27.49 | KristinG | will look into it |
21:27.58 | KristinG | appreciate the info |
21:28.04 | Blissex | KristinG: for example "static=yes", "exten", "voicemail", "playback", "gotoif" |
21:29.03 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-247.modem.logical.net) |
21:29.19 | Carp1 | I just updated Asterisk last night. Last update before that was like 6 months ago. |
21:29.28 | Carp1 | I'm now getting errors trying to load chan_zap |
21:29.39 | Carp1 | and setup_zap |
21:29.39 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
21:29.39 | iceyp | can someone please ring 64273040757@max.fast.co.nz and tell me if they hear ringing or if it goes dead |
21:29.53 | `Sauron | I tried to call it with my cell phone |
21:29.54 | eKo1 | eh, did you zapata? |
21:29.57 | `Sauron | but it complained about the @ |
21:29.57 | `Sauron | ;) |
21:30.05 | eKo1 | Or zaptel or whatever |
21:30.23 | objectivelogic | thanks guys that helped |
21:30.34 | Blissex | KristinG: for example try this: http://www.google.com/search?num=100&as_q=exten+exten+exten+voicemail+playback+gotoif |
21:31.06 | Nivex | www.voip-info.org |
21:31.11 | Nivex | all hail the wiki :) |
21:31.23 | Carp1 | any one know anything about my problamo? |
21:31.31 | Blissex | Nivex: Google is giant Wiki :-) |
21:32.01 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) |
21:32.11 | file[laptop] | colddddddd |
21:32.37 | eKo1 | Carp1: I told you, upgrade zaptel. |
21:34.38 | Carp1 | I did. |
21:35.15 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
21:36.30 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
21:37.49 | eKo1 | What kind of error is it? |
21:38.02 | *** join/#asterisk RaYmAn-Bx (user@x1-6-00-11-80-c3-8a-d7.k441.webspeed.dk) |
21:38.02 | eKo1 | Better yet, post the error somewhere. |
21:38.16 | Carp1 | Just a second |
21:39.49 | Carp1 | www.pastebin.ca right? |
21:40.23 | eKo1 | Whatever, as long as you don't flood this channel. |
21:40.46 | Carp1 | http://pastebin.ca/8806 |
21:42.30 | eKo1 | What does ztcfg say? |
21:46.22 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
21:46.32 | *** join/#asterisk hohum (corbe@snoop.burghcom.com) |
21:46.35 | hohum | hey |
21:46.43 | hohum | I have a very SIP specific question |
21:47.29 | hohum | a B2BUA is supposed to take one call, terminate it on the gateway, initiate another call and proxy the communication between the two call legs, correct? |
21:47.38 | hohum | so |
21:47.46 | hohum | I have a SIP proxy that acts as a B2BUA |
21:48.58 | hohum | it takes an INVITE from another sip proxy from one UA and sends it back to the same gateway where another UA is registered |
21:49.11 | facek_ | how to allow calling people to make a transfer to another consultant? |
21:49.35 | Qwell | hmm |
21:49.39 | hohum | I noticed that my proxy server stamps Via: into the headers |
21:49.48 | Qwell | facek_: You just gave me an odd random thought |
21:49.53 | hohum | with a unique TAG line |
21:50.12 | hohum | however the Call ID is the same |
21:50.16 | facek_ | Qwell what? |
21:50.17 | Qwell | When you call a tech support place, before the person answers, "If at any time, this rep says anything retarded, please press 1. The call will be logged, and you will be transfered to another rep" |
21:50.27 | Qwell | :p |
21:50.29 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
21:50.32 | hohum | and Asterisk that sent the original invite responds with a 482 Loop Detect |
21:50.34 | facek_ | Qwell right |
21:50.39 | hohum | is that because the Call ID didn't change? |
21:50.43 | file[laptop] | hohum: yes |
21:50.43 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
21:50.51 | file[laptop] | hohum: a call id distinctly identifies a call |
21:50.56 | Qwell | file[laptop]: Help me get ISPs to implement that, would ya? |
21:51.02 | hohum | and if so is that Asterisk's fault for not respecting the TAG line or my B2BUA's fault for not changing the Call ID? |
21:51.14 | hohum | file: a call? or a call leg? |
21:51.15 | file[laptop] | hohum: call id should change because it's a different call |
21:51.20 | file[laptop] | a call leg |
21:51.23 | hohum | okay |
21:51.41 | file[laptop] | a B2BUA acts as if there are two different calls in progress, as a bridge between the two... Back to Back User Agent |
21:51.51 | malverian | So what's all the hubub about IAXcom ? |
21:51.52 | file[laptop] | it's a user agent for both... so the Call ID should be different |
21:51.53 | hohum | this is all a massive excersize in debugging a call forwarding problem from my soft switch |
21:51.58 | facek_ | how to allow calling peolpe to make a transfer |
21:52.01 | facek_ | o configiue mgsp.conf |
21:52.04 | malverian | iaxtel rather |
21:52.05 | facek_ | some extensions? |
21:52.07 | file[laptop] | hohum: sounds like it's acting as a proxy |
21:52.47 | jpe | <PROTECTED> |
21:52.56 | jpe | i wish i could. i have been stuck all day trying to get did to work |
21:53.22 | jpe | sorry as you can guess I am new to this irc thing, my bad |
21:54.00 | hohum | I'm trying to see if my Cisco AS5300 behaves in a similar manner |
21:54.37 | malverian | Here's a better question.. how useful is FWD ? |
21:54.42 | malverian | And how hard is it to set up? |
21:54.43 | *** join/#asterisk three55ml (~none@cpe-66-68-98-68.austin.res.rr.com) |
21:54.54 | Qwell | fwd is good, and really easy to setup |
21:55.05 | Qwell | the sample configs show how to connect to fwd even |
21:55.28 | three55ml | Anyone with a SIP phone or ATA interested in testing out a VoIP service please PM me. I'll give you a DID to use during testing. |
21:55.28 | facek_ | how to allow calling peolple to make transfers? |
21:55.32 | malverian | Qwell: What numbers can you dial? |
21:55.43 | eKo1 | jpe: How did you get here anyways? |
21:55.44 | Qwell | malcolmd: other fwd users ;p |
21:55.49 | Qwell | erm |
21:55.51 | malverian | Qwell: Figured.. that kinda sucks :) |
21:55.52 | Qwell | yeah... |
21:56.21 | eKo1 | three55ml: DID from where? |
21:56.32 | three55ml | eKo1: US or 800 |
22:00.45 | *** join/#asterisk Kyrin (~gostlund@d198-53-224-65.abhsia.telus.net) |
22:04.03 | eKo1 | What area code? |
22:06.54 | Kyrin | Anyone here get Asterisk working under Gentoo? |
22:07.02 | Sedorox | yes |
22:07.05 | facek_ | Kyrin yes |
22:07.16 | Kyrin | facek_: my ebuild keeps breaking |
22:07.22 | *** join/#asterisk PTG123 (~PTG123@66.213.239.122) |
22:07.33 | facek_ | Kyrin i installed from scratch |
22:07.57 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
22:07.59 | Kyrin | Ah, haven't tried that, kinda the purpose behind Gentoo, isn't it? |
22:08.09 | Kyrin | I mean, it's doing the "build from scratch" for me |
22:08.32 | Sedorox | I did a ebuild |
22:08.44 | Kyrin | Sedorox: No issues I take it? |
22:09.02 | Kyrin | [ebuild N ] net-misc/asterisk-0.9.0 +alsa +apache2 -doc -gtk -mmx +mysql -nopri -nozaptel 0 kB |
22:09.17 | Sedorox | not for me |
22:09.19 | Sedorox | wow |
22:09.20 | Sedorox | dude... |
22:09.22 | Kyrin | that's my flags, what should I have? |
22:09.22 | Sedorox | emerge sync |
22:09.25 | Sedorox | then run that again |
22:09.37 | Sedorox | thats a REALLLY old version |
22:09.45 | Kyrin | was sync'd yesterday, what version should it be? |
22:09.52 | Kyrin | ther 1.0.7 is masked by x86 |
22:09.54 | Supaplex | I should write a xmms control to have it pause while the phone is ringing, or while I'm on the phone |
22:09.59 | Kyrin | should I just accept keywords that? |
22:10.28 | Sedorox | yes |
22:10.39 | Kyrin | Oh, what are those last two options 'nopri' and 'nozaptel' do I need those? |
22:10.49 | Sedorox | ACCEPT_KEYWORDS="~x86" emerge asterisk -p |
22:10.59 | Sedorox | no.. that installs the pri and zaptel stuff |
22:11.01 | Sedorox | its fine |
22:11.17 | Sedorox | or whatever it is to add the keywords.. just for that build |
22:11.21 | Kyrin | k, 'cause my error messages said something about pri so I wasn't sure |
22:11.33 | Sedorox | well I'm sure there are new versions |
22:12.08 | Supaplex | Can I get some comments on link2voip and the overall service/expecations etc? |
22:12.31 | Sedorox | they SUCK |
22:12.39 | Kyrin | [ebuild N ] net-misc/asterisk-1.0.7 +alsa -bri -debug -doc -gtk -hardened -mmx +mysql -postgres -pri -resperl -speex (-uclibc) -vmdbmysql -vmdbpostgres -zaptel 0 kB |
22:12.41 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
22:12.43 | Sedorox | me abd Beirdo have waited over a month for a toll free number |
22:12.47 | Sedorox | we have YET to get it |
22:12.53 | Sedorox | and I signed up Feb. 25 |
22:12.56 | Supaplex | sucky :( |
22:12.59 | Sedorox | I do not recommended them at all |
22:13.06 | Supaplex | yea, I need a 800 number |
22:13.08 | Beirdo | I have one |
22:13.17 | Beirdo | but not through those assmasters |
22:13.20 | Sedorox | Beirdo: tell him the company that we ended up going with |
22:13.23 | Sedorox | 'cause I forget |
22:13.23 | Sedorox | :-p |
22:13.27 | Beirdo | thinktel.ca |
22:13.40 | Supaplex | I already have vonage, but I wanna ditch their linksys-ata and use asterisk with sipura or something |
22:13.41 | Sedorox | :) |
22:13.46 | Sedorox | heh |
22:14.11 | Supaplex | some day we'll learn how to fix that issue :) |
22:14.31 | Sedorox | lol |
22:14.50 | Kyrin | hrm... 2.4.28 kernel |
22:19.04 | Zipper_32 | Inside a small office, is it recommended to have a dedicated ethernet connection running to each desk for VOIP, or should it share a standard data cable? |
22:19.21 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
22:19.35 | ManxPower | I'm having a really bad day. What newbie can I abuse? |
22:19.44 | Kyrin | OOh, pick me |
22:20.02 | _Vile | abuse bkw, he likes it |
22:20.07 | Kyrin | I know very little 'bout VoIP, just what I've read on the 'net, and lots of questions |
22:20.07 | Supaplex | thinktel.ca is in 0 wiki articles ;/ |
22:20.10 | Zipper_32 | ManxPowe,r you can abuse my last question... |
22:20.52 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
22:20.56 | Kyrin | What would be the recommended down/up bandwidth for 12 lines to an office? |
22:21.30 | *** join/#asterisk dalabera (~Dalabera@mail2.pmrtechnologies.com) |
22:21.41 | hardwire | Kyrin: python makes a nice console calculator |
22:21.45 | *** join/#asterisk zilas (~1@c-24-30-75-206.hsd1.ga.comcast.net) |
22:22.11 | zilas | hello |
22:22.20 | harryvv | betwen my router and somone elses end router we have ports 5060,5061,10001-10010 open for a sip client to server connection but still cannot hear each other. Getting this error in cli. WARNING[9601]: chan_sip.c:787 retrans_pkt: Maximum retries exceeded on call 4AFFF90F-2C0D-456E-A91D-EB178C3B0100@192.168.1.100 for seqno 24565 (Non-critical Response) |
22:22.41 | *** join/#asterisk montoya (montoya@200.195.87.110) |
22:23.04 | Kyrin | hardwire: so does echo $[x*12], but I don't know the numbers I ought to put for 'x' |
22:23.06 | harryvv | hi zilas |
22:23.30 | hardwire | Kyrin: voip-info.org has the specs |
22:23.39 | Kyrin | hardwire: thx, will read that now |
22:23.46 | zilas | for call parking exten => a,b,(SIP/phone1,20,tr) what does here a and be stand for? |
22:24.03 | Kyrin | omfg |
22:24.09 | hardwire | omfg? |
22:24.11 | Kyrin | how come I never found this one before |
22:24.16 | *** join/#asterisk ChulJin1 (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
22:24.21 | hardwire | Kyrin: because you are a crazy person |
22:24.24 | Kyrin | this site, this wiki |
22:24.25 | Kyrin | wow |
22:24.42 | harryvv | any of you link a remote sip phone to your asterisk box? |
22:24.55 | *** join/#asterisk juice (~juice@mo-69-68-108-44.dyn.sprint-hsd.net) |
22:25.03 | Kyrin | It's like never I saw it, dunno why, reading now |
22:25.14 | ChulJin1 | harryvv: all over the world, yeah |
22:26.29 | zilas | why this call parking so purely documented I can't make it work :( can someone share some tips with me ples... |
22:28.19 | AgiNamu | anyone care to explain how IAX2 transfers work? |
22:28.30 | AgiNamu | I'm looking at a netcap and it is... wierd. |
22:28.54 | Kyrin | hardwire: So, if I'm reading this right... if i have a 1mbit upload, that might be enough for 12 lines if I don't use a the uncompressed PCM codec |
22:28.55 | rowter | am trying to work the app_directory, exten => 30,1,Directory(office,f) it does not 1 => 123,Test,test@tes.com from voicemail mmh.. |
22:28.56 | AgiNamu | I have Asterisk1 calling Asterisk2 which calls Asterisk3. 2 should transfer out, and thats what I see. |
22:29.11 | AgiNamu | but from the tcpdump, it seems as if Asterisk1 initiates the transfer :\ |
22:30.05 | *** join/#asterisk Hogie (~daniel@alpha.dfwservers.net) |
22:30.18 | eKo1 | That info. is classified I'm afraid. |
22:30.37 | *** join/#asterisk captrb (~crozierm@64.65.134.42) |
22:31.44 | AgiNamu | eKo1, judging from chan_iax2 ... yea :P |
22:31.45 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
22:32.34 | eKo1 | Ask kram, he's the iax2 wizard of the north. |
22:32.58 | eKo1 | I rest my case. |
22:33.04 | *** join/#asterisk goobster (goobster@c-67-168-105-166.hsd1.wa.comcast.net) |
22:33.11 | file[laptop] | AgiNamu: I have a doc around here on the procedure |
22:33.15 | *** join/#asterisk xkev (kevin@orbit.xmission.com) |
22:33.59 | *** join/#asterisk chfn (~adolfo@c906ff56.virtua.com.br) |
22:34.02 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
22:34.24 | Kyrin | How easily can several different numbers be routed through and managed (transfers, holds, conference calls, etc) one asterisk system? |
22:35.03 | AgiNamu | file... mgg@atrevido.net :) |
22:35.16 | file[laptop] | AgiNamu: doesn't mean I'll find it |
22:35.16 | AgiNamu | lol ok |
22:35.20 | AgiNamu | install MSN desktop search. |
22:35.51 | Supaplex | s/MSN/google/ |
22:35.53 | Supaplex | :P |
22:36.05 | *** join/#asterisk loko (rbrown@67.171.66.213) |
22:36.06 | eKo1 | how about using grep instead |
22:36.47 | chfn | hi guys. |
22:37.55 | chfn | Can I do this with asterisk? a script will run a 'tool' of asterisk then asterisk will call this number to play a mp3 message. |
22:38.00 | chfn | will I need to code? |
22:38.49 | chfn | a script of asterisk will be runned to call some number then will play a mp3 message. Can i Do this ? |
22:38.58 | Nivex | chfn: http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out |
22:39.01 | loko | Darwin35, you around? |
22:39.07 | eKo1 | chfn: yes |
22:39.21 | chfn | eKo1, Nivex so will I be able to remember people of something ? |
22:39.23 | eKo1 | Thank you for participating. |
22:39.35 | Nivex | chfn: using /var/spool/asterisk/outgoing, code will be minimal |
22:39.49 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
22:40.21 | elriah | Hi guys, on a production asterisk system that services 25-250 phones, running bare-bones linux and no X windows at all, would 256MB of RAM be enough? |
22:40.25 | *** join/#asterisk bjohnson (~bjohnson@ip202-181.tor.istop.com) |
22:40.36 | chfn | Nivex, i love you man! :) sorry for telling you that but I(stupid guy) told a owner(company) that asterisk probably does that. asterisk will be everything talking about PBX. |
22:40.47 | chfn | Nivex, cool project. I like it. |
22:41.05 | chfn | Nivex, whats the language of source code of asterisk ? |
22:41.22 | cypromis | c |
22:41.24 | dalabera | elriah, I think so since * it's cpu eater.... |
22:41.28 | AgiNamu | C# |
22:41.31 | Pinhole | chfn, English |
22:41.31 | cypromis | lol |
22:41.46 | chfn | Pinhole, lol |
22:41.49 | AgiNamu | it was gonna be x86 |
22:41.50 | Kyrin | How easily can several different numbers be routed through and managed (transfers, holds, conference calls, etc) one asterisk system? Or is this something that I have to worry about coordinating with my VoIP provider, someone like Vonage? |
22:41.53 | AgiNamu | but that wasn't as portable. |
22:41.59 | AgiNamu | See, C is just a hack for those who can't think in machine code. |
22:42.04 | Pinhole | actually, it might be Canadian |
22:42.14 | AgiNamu | just as C++ is a hack for people who can't think in objects for themselves. |
22:42.29 | chfn | i always help gnu software with bugs, correction... I will use asterisk and i will report if I find! :) |
22:42.31 | AgiNamu | this whole "high level", "productivity" stuff is a load of crap. |
22:42.37 | *** join/#asterisk fugitivo (~ajf@201.255.105.150) |
22:43.32 | AgiNamu | Kyrin, it can be easy or hard |
22:43.37 | AgiNamu | depending on what you are actually doing. |
22:44.05 | AgiNamu | I dont see any way for asterisk to natively scale. |
22:44.21 | AgiNamu | if you only have a couple hundred clients, itll be fine |
22:44.31 | AgiNamu | apart from that, then you gotta start getting creative. |
22:44.32 | eKo1 | * was made in C because it was first made to work on Linux and C is the most supported language on that platform. |
22:45.04 | AgiNamu | i know. im making sarcastic jokes based off of a certain checkin comment i read |
22:45.14 | eKo1 | AgiNamu: Define 'natively scale'. |
22:45.18 | AgiNamu | that said "C++ is just a crutch for people who can't think in objects for themselves" |
22:45.25 | AgiNamu | I mean, there's no built-in clustering system in asterisk |
22:45.29 | Pinhole | well, if Linux was written correctly, (asm), we wouldn't have problems being forced to use C. ;) |
22:45.31 | Darwin35 | ? |
22:45.37 | chfn | IAX is for comunicating with another IAX right ? |
22:45.45 | Darwin35 | someone call me |
22:45.49 | AgiNamu | chgn, IAX is a protocl, like SIP, H323, etc. |
22:45.53 | *** join/#asterisk E818 (anonymous@rrcs-24-199-5-190.west.biz.rr.com) |
22:45.54 | AgiNamu | IAX is just a good protocol thats easy to use |
22:45.57 | eKo1 | AgiNamu: Give it time. |
22:45.57 | AgiNamu | unlike SIP, H323, etc. |
22:46.09 | AgiNamu | eKo1, sure, in time. meanwhile, today, there's no cluster capability |
22:46.19 | AgiNamu | BUT |
22:46.22 | eKo1 | Pinhole: Parts of Linux are in ASM. |
22:46.24 | AgiNamu | it's not that hard to add your own in |
22:46.34 | captrb | computers are just a crutch for people who can't manipulate electrons properly |
22:46.37 | AgiNamu | lol |
22:47.04 | AgiNamu | im writing my backend in C# and going to have asterisk use SOAP calls to figure out how ot route calls |
22:47.11 | Pinhole | electrons are just a crutch for people that can't manipulate strings properly. |
22:47.23 | captrb | ha |
22:47.26 | AgiNamu | strings? |
22:47.35 | tzanger | AgiNamu: google for string theory |
22:47.49 | AgiNamu | I got as far as a bit of quantum mechanics in high school chemistry |
22:47.51 | AgiNamu | AP |
22:47.55 | AgiNamu | and then i dropped out |
22:48.39 | eKo1 | AgiNamu: Why SOAP? That seems like overkill and will not scale well in my opinion. |
22:48.48 | tzanger | SOAP scales well |
22:48.50 | Kyrin | AgiNamu: I need like ten different numbers starting with 220 to go through the same system that is also handling two 990 numbers |
22:48.56 | AgiNamu | because SOAP is an easy way for me to do cross-platform communications |
22:49.02 | AgiNamu | it's fast, development wise |
22:49.07 | AgiNamu | and the performance kicks ass, in my tests |
22:49.07 | chfn | whats the cheapest hardware for 4-port analog + voice + fax ? i mean whats the price of a VFX/41JCT-LS ? |
22:49.13 | chfn | does someone knows an e-store to buy a VFX/41JCT-LS intel card ? |
22:49.20 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
22:49.22 | tzanger | chfn: why don' tyou contact your distributor and find out |
22:49.25 | AgiNamu | www.atrevido.net : I ran a test using gSOAP on linux, calling ASP.NET on Windows. ASP.NET service queries a DB and returns an answer. |
22:49.26 | E818 | what's the best way to implement a wake-up call? AGI? |
22:49.30 | eKo1 | AgiNamu: Only if you have a good environment to work in. |
22:49.37 | AgiNamu | testing against my DESKTOP, I could do 470 requests/sec |
22:49.40 | tzanger | E818: no. callfiles |
22:49.40 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
22:49.45 | AgiNamu | at which point it became CPU bound. |
22:49.52 | AgiNamu | a single P4 |
22:49.55 | chfn | tzanger, because I live in a FUCKING country called brazil! i have a friend in NY that could buy for me but i need prices |
22:50.09 | tzanger | chfn: so ask him |
22:50.11 | AgiNamu | eKo1, Visual Studio 2005 |
22:50.14 | Kyrin | heh |
22:50.15 | AgiNamu | that's a kick ass environment to work in |
22:50.17 | tzanger | mind you I don't think I'd mind living in a fucking country |
22:50.20 | AgiNamu | thats where I do all my devs |
22:50.25 | AgiNamu | dev. even for asterisk. |
22:50.31 | chfn | tzanger, :( if I call intel from brasil they will say: ARE YOU nutz? whats VFX/41JCT-LS? |
22:50.33 | AgiNamu | just plug in GCC into VS2005 |
22:50.35 | eKo1 | Hmm...I use vim and gcc. |
22:50.41 | eKo1 | That's it. |
22:50.45 | chfn | tzanger, stupid people in call center |
22:50.47 | AgiNamu | chfn, i know how it feels. |
22:50.53 | elriah | notepad.exe |
22:50.55 | AgiNamu | i live in guatemala |
22:51.00 | tzanger | vim |
22:51.05 | elriah | heh |
22:51.08 | AgiNamu | elriah, notepad doesnt handle large files well |
22:51.13 | eKo1 | So how much did you pay for VS2K5? |
22:51.21 | AgiNamu | nothing... |
22:51.28 | eKo1 | warez? |
22:51.29 | elriah | Oh, I was kidding. I use vim on *nix, textpad on windows. |
22:51.30 | Kyrin | chfn: http://shopper.cnet.com/Dialogic_VFX_41JCT_LS_voice_fax_board/4014-3004_9-30387014.html? |
22:51.31 | AgiNamu | I was an MS MVP (didnt get reawarded this year) |
22:51.40 | AgiNamu | and they gave me an MSDN universal suib. |
22:51.46 | AgiNamu | at any rate, the express version of VS2005 is gonna be $49 |
22:51.51 | AgiNamu | unless you are a student. then it's $5. |
22:52.04 | chfn | AgiNamu, :| |
22:52.18 | eKo1 | I remember paying about $300 for VS6. |
22:52.20 | AgiNamu | and MS SQL 2005 Express is free. |
22:52.25 | eKo1 | What a rip off. |
22:52.35 | chfn | Kyrin, thanks. |
22:52.43 | AgiNamu | eKo1, yea, Professional is like $1000, and Team versions are $2500, and the ripoff is Team Suite, which is $11K |
22:52.54 | eKo1 | But back then, I) was a M$ junkie. |
22:52.54 | Kyrin | chfn: np |
22:52.55 | AgiNamu | BUT, Visual Studio team system is EXTREMELY powerful. and it's cheap, compared to Rational, etc. |
22:53.04 | chfn | whats the cheapest pci card for 4 ports ? |
22:53.17 | AgiNamu | chfn analog? |
22:53.19 | Pinhole | $11k is cheaper than a programmer for a year. If it save enough time, it is justified. |
22:53.21 | chfn | AgiNamu, yes |
22:53.24 | elriah | If you want to plug in gcc, just get C#.NET which comes with visual studio ide for $89.00. |
22:53.26 | AgiNamu | pinhole, damn straight |
22:53.33 | AgiNamu | elriah, but C# doesnt allow gcc |
22:53.38 | AgiNamu | you need VC++ |
22:53.43 | elriah | Ahh.. |
22:53.46 | eKo1 | Just use mono. |
22:53.47 | elriah | k, sc |
22:53.54 | AgiNamu | eKo1, mono isn't an IDE :P |
22:54.05 | chfn | AgiNamu, analog. |
22:54.10 | eKo1 | I was talking about c#. |
22:54.10 | AgiNamu | TDM400 |
22:54.21 | AgiNamu | it's only $350 or so for 4 FXO |
22:54.31 | eKo1 | Analog sucks. Go digital. |
22:54.34 | Kyrin | don't suppose any VoIP providers have irc...? |
22:54.48 | chfn | AgiNamu, cnet i could not find prices for TDM400 |
22:54.54 | AgiNamu | www.digium.com |
22:55.08 | zilas | call parking sucks! |
22:55.17 | chfn | AgiNamu, thanks! |
22:55.29 | eKo1 | Say AgiNamu, do they have ISDN in Guatemala? |
22:55.51 | chfn | AgiNamu, hmm asterisk and digium! fine!! :) |
22:56.05 | eKo1 | Or are the people still stuck with pulse tone phones. |
22:56.23 | AgiNamu | eko, yea |
22:56.29 | eKo1 | How much? |
22:56.34 | AgiNamu | I can get an E1 too |
22:56.36 | eKo1 | For a BRI and a PRI. |
22:56.37 | AgiNamu | for like $400 a month |
22:56.48 | chfn | AgiNamu, love you! :) i will buy soon. probably in the next semester i will buy some. |
22:56.49 | eKo1 | How much for an E1? |
22:56.52 | AgiNamu | good luck |
22:56.55 | AgiNamu | like $400 a month |
22:56.59 | AgiNamu | up to $900 |
22:57.00 | AgiNamu | depending |
22:57.08 | AgiNamu | including like 50K local minutes (@ 2cents US a minute) |
22:57.19 | AgiNamu | i almost bought one to do bypass |
22:57.20 | eKo1 | Say I wanted PRI over E1? |
22:57.25 | AgiNamu | right |
22:57.29 | AgiNamu | a few hundred bucks a month |
22:57.31 | chfn | What type of computer should i have to use more VoIP and less analog lines? but i think i probably will use 10 VoIP lines. |
22:57.41 | AgiNamu | chfn, anything really |
22:57.43 | eKo1 | Interesting. |
22:57.50 | AgiNamu | even a 1.5GHz machine can handle 10 lines. |
22:58.03 | eKo1 | I figured it would be mad expensive in that underdeveloped banana country. |
22:58.11 | AgiNamu | lol, yea, well, PRI isn't that bad. |
22:58.14 | AgiNamu | Getting internet access is |
22:58.23 | AgiNamu | 2MBps line was like $1000 a month |
22:58.26 | chfn | AgiNamu, fuck :( i need to travel to visit a company of a friend that is impl. VoIP in theier ISP. |
22:58.28 | AgiNamu | 512K ADSL is $230 |
22:58.37 | AgiNamu | chfn, its not hard |
22:58.39 | chfn | AgiNamu, i dont know. Can i handle how much calls with a VoIP line |
22:58.42 | AgiNamu | chfn, i didnt know shit a few months ago |
22:58.52 | AgiNamu | With a VoIP line? i dont understand. |
22:59.16 | chfn | AgiNamu, with a VoIP phone. Can I handle a call and anwser another. How much can i do this ? |
22:59.27 | AgiNamu | that depends on the phone |
22:59.27 | chfn | AgiNamu, it depends of the phone? |
22:59.30 | AgiNamu | yep |
22:59.36 | AgiNamu | some phones can handle multiple voice channels |
22:59.37 | AgiNamu | some cant. |
22:59.42 | chfn | AgiNamu, i already saw cisco and tested cisco VoIP phones |
22:59.44 | AgiNamu | some can conference 2 calls together. |
22:59.49 | AgiNamu | Cisco phones are very nice from what I hear. |
22:59.54 | AgiNamu | I only use PA168, cause I want IAX2 support. |
22:59.59 | chfn | AgiNamu, what country do you live ? |
23:00.05 | AgiNamu | guatemala |
23:00.09 | chfn | AgiNamu, brasil :D |
23:00.26 | chfn | AgiNamu, these FSF projects is getting a lot of users from these kind of countries. |
23:00.35 | AgiNamu | brasil.... didnt the presidente just mandate that ALL companies that get government funding must use free software? |
23:00.45 | AgiNamu | Sounds like a realllly dumb thing to say.... I Can't wait for the integration nightmares. |
23:00.59 | eKo1 | chfn: 'thse kind of countries'¿ |
23:01.00 | AgiNamu | chfn, yea, they are getting a lot of support, but for the wrong reasons. |
23:01.09 | chfn | eKo1, i mean LDEC |
23:01.16 | chfn | eKo1, i know ? |
23:01.32 | AgiNamu | here, people sya "OH! OpenOffice is free! screw MS" |
23:01.34 | captrb | does anybody have any tips on troubleshooting a pri line? |
23:01.40 | chfn | AgiNamu, for wrong reasons but... i like FSF and its getting a lot of users... so its good |
23:01.49 | AgiNamu | and then they realise that oh, OpenOffice ain't even NEAR the quality of Office. |
23:01.50 | chfn | AgiNamu, i know your thought |
23:02.03 | AgiNamu | they just dont think past the up front licensing fee. |
23:02.05 | captrb | my telco says that they can't detect signal from digium/asterisk |
23:02.11 | eKo1 | Yeah, but most users don't even use half the features of Office so... |
23:02.19 | AgiNamu | eKo1, that's a faulty argument |
23:02.29 | AgiNamu | most users might not use many features. But the features they DO use is different for each user. |
23:02.30 | chfn | AgiNamu, do you have msn? icq? i would like to test voip with you. lol! :D to talk |
23:02.41 | captrb | so they can't loop up the line |
23:02.50 | AgiNamu | and the quality of the feature is huge. For instance, CJK support |
23:03.05 | AgiNamu | On XP, Word 2003... CJK support kicks fucking ass. On Linux, Gnome.... OUCH |
23:03.09 | AgiNamu | even the IME is a piece of shit |
23:03.27 | AgiNamu | but in Office... it detects if i forget to change modes (say, Hangul > Roman) and changes for me. |
23:03.35 | chfn | AgiNamu, do you have msn? icq? i would like to test voip with you. lol! :D to talk |
23:03.42 | AgiNamu | that's a SMALL FEATURE.... but they have tons of small features like that. |
23:03.47 | AgiNamu | chgfn, mgg@atrevido.net |
23:03.50 | AgiNamu | msn |
23:04.00 | chfn | AgiNamu, some kind of topic is not for this channel. :| be carefull the 'masters' of this channel probably does not like |
23:04.10 | chfn | AgiNamu, thanks. i will add now |
23:04.21 | AgiNamu | well, the only time I've been kicked is when I discussed cracking certain stuff. |
23:04.38 | AgiNamu | FSF doesn't extend to reverse engineering :) |
23:06.39 | eKo1 | I need Visio on Linux. |
23:06.46 | fugitivo | eKo1: use kivio |
23:07.08 | Kyrin | Anyone here use VoIP in Canada and can recommend a provider to use with Asterisk? |
23:07.08 | eKo1 | Kivio and Dia don't come close to Visio. |
23:07.28 | fugitivo | what else do you need to make a diagram? |
23:07.38 | AgiNamu | Kyrin: If you want someone in canada, ask Sivana |
23:08.19 | Kyrin | sivana: Pls /msg me whtn you get time, I would like a recommendation as to which VoIP provider I should look at |
23:08.41 | eKo1 | fugitivo: Eh, ERD diagrams for example. |
23:08.50 | Kyrin | AgiNamu: Thx, I was looking at Link2VoIP 'cause it supports IAX, but I dunno any others |
23:09.01 | shido6 | use xetricom |
23:09.06 | AgiNamu | there's a lot. goto the wiki |
23:09.13 | shido6 | xetricom.net's page sucks but they offer voip in canada |
23:09.17 | AgiNamu | I do know that sivana has a local PRI in ontario |
23:09.21 | shido6 | 416 |
23:09.23 | shido6 | 647 |
23:09.26 | mishehu | bah. |
23:09.27 | shido6 | and 8xx |
23:09.47 | shido6 | where do you need the PRI? |
23:09.49 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
23:10.22 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
23:11.19 | Drel | Can anyone recommend a firewall/router with QoS that works well with Asterisk/VoIP in general? It would also need to support PPTP/VPN passthrough. |
23:11.25 | AgiNamu | iptables |
23:11.33 | fugitivo | pf |
23:11.48 | Drel | I'm currently using netgear fvs-318 but it lacks QoS. |
23:11.50 | hardwire | whats the diff inbetween PRI And DSS? |
23:11.53 | dmccollum | IPCOP |
23:11.58 | hardwire | you lie! |
23:12.35 | loko | cisco 831 |
23:12.48 | *** join/#asterisk bjohnson (~bjohnson@ip202-181.tor.istop.com) |
23:13.33 | Drel | AgiNamu: Are you suggesting iptables + tc? |
23:13.38 | AgiNamu | tc? |
23:14.24 | blitzrage | tc is for QoS I believe. |
23:14.30 | RaYmAn-Bx | traffic control or similar. From linux iproute2 |
23:14.32 | Drel | tc/lartc: http://lartc.org/ |
23:14.41 | blitzrage | ahhhh yes. Traffic Control. |
23:14.50 | Drel | As far as I know, iptables has no support for QoS. |
23:14.56 | blitzrage | iptables rocks, but the FreeBSD filter system looks nicer |
23:15.04 | RaYmAn-Bx | Drel: you can approximate it though :P |
23:15.31 | blitzrage | Drel: it doesn't, you're right (afaik as well). But there are tricks you can do to kind of simulate it. tc is much nicer though. |
23:15.34 | RaYmAn-Bx | i.e. you can limit certain connections by dropping packets when they exceed a certain speed |
23:15.56 | Drel | Well, I need a firewall/router with QoS that works well with Asterisk, supports PPTP/VPN passthrough, and doesn't require much maintenance. IE, no patching of kernel to get QoS, I don't want to have to recompile everytime a security update comes out. |
23:17.58 | RaYmAn-Bx | 2.6 has good QoS support builtin..but obviously it has to be enabled in the kernel. |
23:18.13 | Drel | FVS-338 looks like it might not be a bad solution, does anyone have any experience with it? The FVS-318 I'm currently using seems to have bugs. It stops forwarding traffic every month or two and requires a restart. So, I'm a little hesitant to go with Netgear again, but... I like the idea of low-maintenance hardware where I might have to update the firmware once in a while. |
23:18.22 | Drel | RaYmAn-Bx: Do any distributions ship with that enabled? |
23:18.38 | RaYmAn-Bx | I always compile my own kernels so i don't know |
23:19.07 | tzanger | Drel: just get a sokeris or WARP platform and throw on a CF card. same lack of maintenance |
23:19.32 | Drel | tzanger: I'm using a sokeris platform for wireless here right now, actually,... Hmm. |
23:19.35 | hardwire | WRAP |
23:20.27 | tzanger | er yes WRAP I always screw that up |
23:20.27 | hardwire | heh |
23:20.27 | hardwire | I use WRAP + * |
23:20.27 | hardwire | * two tri-mode cards |
23:20.27 | hardwire | err |
23:20.27 | hardwire | + |
23:20.27 | tzanger | what's the geode like for codec conversions |
23:20.27 | hardwire | I always screw that up :) |
23:20.27 | hardwire | tzanger: I don't even bother. |
23:20.28 | hardwire | GSM all the way. |
23:20.28 | Drel | hardwire: Got a link for that? |
23:20.33 | tzanger | hardwire: :-) |
23:20.41 | tzanger | I would love to get a miniPCI ADSL modem for those things |
23:20.44 | Drel | hardwire: WRAP that is. |
23:20.53 | hardwire | mini-box.com |
23:21.06 | hardwire | I talk with the pcengines guy that makes them all the time |
23:21.20 | tzanger | hardwire: nice. get a MiniPCI ADSL modem then |
23:21.22 | hardwire | I am so happy there is a quick-n-dirty way of controlling the three leds on the front of it now |
23:21.28 | tzanger | yup |
23:21.30 | hardwire | tzafrir: that would be nice |
23:21.36 | hardwire | I have never seen a mini-pci ADSL card |
23:21.49 | tzanger | I use the Sangoma S518 ADSL cards... I can flood my uplink and maintain call quality |
23:21.57 | hardwire | I think the FCC won't like it either |
23:22.08 | tzanger | hardwire: why not? |
23:22.12 | hardwire | hmm |
23:22.15 | hardwire | nm |
23:22.18 | hardwire | Sangoma eh |
23:22.20 | tzanger | there are tiny tiny modems (my IBM T30 has one) |
23:22.21 | hardwire | works well w/ linux? |
23:22.27 | tzanger | hardwire: works *excellent* with linux |
23:22.50 | tzanger | it's just a Globespan chipset OEM'd by Sangoma |
23:22.50 | tzanger | but their drivers |
23:22.50 | tzanger | it just works |
23:24.48 | Beirdo | gah |
23:24.53 | Beirdo | globespan |
23:24.59 | tzanger | the funny part is |
23:25.20 | tzanger | their T1 card and their ADSL card don't coexist happily. Digium's T100P and the S518 work just fine |
23:25.47 | TechDawg | Anyone have any experience with the Intel IA92 cards for FXOs? |
23:26.57 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
23:28.15 | TechDawg | Guess that would be no. |
23:29.04 | Supaplex | mmm sangoma gimmie |
23:29.20 | *** part/#asterisk moy (~kvirc@201.137.229.111) |
23:29.46 | hardwire | hmm |
23:29.54 | hardwire | anybody doing DATA + Voice over PRI? |
23:30.05 | hardwire | thats just syncppp on one span right? |
23:31.23 | *** join/#asterisk implicit (~implicit@lgb-cust-66.18.140.106.mpowercom.net) |
23:31.44 | captrb | hardwire: trying, but not succeeding |
23:31.53 | hardwire | captrb: heh! |
23:32.12 | hardwire | I know some companies offer dynamic voice line allocation for PRI lines |
23:32.21 | captrb | actually, data works, but the telco can't loop up the pri |
23:32.21 | hardwire | just wondering how that can be done w/ two t100p |
23:32.31 | captrb | dynamic? |
23:32.37 | hardwire | yeh |
23:32.50 | hardwire | ICG in Colorado offers a data/voice combo and their own router |
23:32.51 | captrb | i saw an example of dynamic pri somewhere... |
23:33.06 | hardwire | kinda neat |
23:33.08 | RoyK | ~seen coppice |
23:33.11 | jbot | coppice <~chatzilla@227.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 7d 14h 25m 29s ago, saying: 'hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest'. |
23:33.38 | captrb | we have fixed channels, but basically that. from eschelon (craptastic service) |
23:33.43 | hardwire | is that all over the D channel how that works? |
23:33.57 | hardwire | and how does the PPP session grow |
23:34.01 | hardwire | unless its multilink ppp |
23:34.03 | hardwire | one per channel |
23:34.09 | zilas | please, can somebody help me with a simple thing for call parking |
23:34.19 | hardwire | zilas: I am lost w/ call parking |
23:35.53 | zilas | I cant make it to work, I am giving up already... can that a phone doesn't support it? |
23:37.10 | tzanger | zilas: what phone |
23:39.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) [NETSPLIT VICTIM] |
23:39.45 | *** join/#asterisk drinker (~lawrence@adsl1.kennedysgroup.com) [NETSPLIT VICTIM] |
23:39.45 | *** join/#asterisk memic (~memic@chicago089.server4free.de) [NETSPLIT VICTIM] |
23:39.45 | *** join/#asterisk Elshar (~Elshar@216.110.205.68) [NETSPLIT VICTIM] |
23:39.45 | *** join/#asterisk kore (kore@mindwipe.org) [NETSPLIT VICTIM] |
23:40.26 | *** join/#asterisk Elshar (~Elshar@216.110.205.68) |
23:41.12 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
23:41.58 | Supaplex | tzanger: you know. that one phone. ;) |
23:42.03 | tzanger | heh |
23:42.34 | *** join/#asterisk jason357 (~m00@67.159.26.120) |
23:43.57 | *** part/#asterisk DougNaka (~Doug@207.225.223.187) |
23:44.01 | *** join/#asterisk DougNaka (~Doug@207.225.223.187) |
23:45.01 | zilas | tzanger: Siemens HiNet LP 5200 |
23:45.11 | tzanger | regular analog phone or what |
23:45.16 | zilas | sorry 5200 |
23:45.20 | zilas | 5100 |
23:45.23 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
23:45.25 | zilas | sip |
23:46.17 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
23:46.34 | shmaltz | has anybody tried applying this patch? |
23:46.35 | shmaltz | http://bugs.digium.com/bug_view_page.php?bug_id=0002905 |
23:46.59 | jason357 | I want a system where people can dial my 800 number, enter an extension and hear a recording with tracking based on extension dialed, most entension will point to a single announcment message |
23:46.59 | shmaltz | it fails on 1.0.4, 1.0.5, 1.0.6, 1.0.7 |
23:47.22 | jason357 | my question is, without running my own * server for this, who provides such a thing for a pre minute or monthly fee? |
23:47.22 | tzanger | shmaltz: probably because it's for HEAD |
23:47.42 | shmaltz | tzanger, but I want it on stable, no way to that? |
23:49.36 | tzanger | shmaltz: sure, figur eout what it does and rework it :-) |
23:49.46 | shmaltz | ;0 |
23:49.54 | shmaltz | ;( |
23:50.56 | brycec | brc_, you alive? |
23:50.56 | jason357 | is there a directory of asterisk users selling services on their host? |
23:51.03 | brc_ | yessir |
23:52.14 | brc_ | brycec, ? |
23:54.45 | reallost1 | Anyone here use BackgroundDetect? |
23:56.20 | *** part/#asterisk Zipper_32 (~none@s207-6-25-182.bc.hsia.telus.net) |
23:57.26 | zilas | I don't believe I was so stupid, wow |
23:57.36 | zilas | figured out... |