| 00:00.15 | De_Mon | the res_ one is in resources |
| 00:00.55 | ice_croft | oh. |
| 00:01.07 | ice_croft | a cant see it in my res folder |
| 00:01.15 | ice_croft | damn |
| 00:02.00 | ice_croft | j |
| 00:02.02 | ice_croft | o |
| 00:02.03 | ice_croft | no |
| 00:02.08 | ice_croft | here it is |
| 00:02.17 | ice_croft | res_smdi's present |
| 00:02.25 | ice_croft | next? |
| 00:03.25 | De_Mon | that zaptel_vldtmf sounds like its external? |
| 00:03.48 | ice_croft | i don't know |
| 00:07.05 | Qwell | ice_croft: tell the BSD folks to update zaptel |
| 00:07.34 | Qwell | zaptel_dtmf is something in (normal) zaptel that was added during the 1.4 cycle that is required to use it |
| 00:08.22 | JayTee52 | Qwell, is 64 bit Asterisk any good in terms of stability, reliability? |
| 00:08.35 | Qwell | 64-bit asterisk? it's still just asterisk |
| 00:08.43 | De_Mon | Qwell so hes got an old version of zaptel right? |
| 00:08.50 | Qwell | De_Mon: sort of |
| 00:09.05 | ice_croft | Qwell> how can i figure, does my zaptel have zaptel_vldtmf ? |
| 00:09.12 | Qwell | zaptel bsd is a port of zaptel. He may have the latest version of it - but it itself is out of date |
| 00:09.20 | De_Mon | Qwell its bsd's fault then? |
| 00:09.32 | Qwell | well, the guy who maintains it |
| 00:09.43 | Qwell | if it didn't add what is required |
| 00:09.49 | De_Mon | oh fun |
| 00:09.50 | Qwell | check config.log |
| 00:09.54 | De_Mon | glad Im not using bsd! |
| 00:10.26 | Qwell | I tried porting something we were working on to that.. it was kind of a pain |
| 00:10.43 | Qwell | uses bsd make and everything... I was so out of place |
| 00:12.02 | ice_croft | so, anyway, what should i do to make it work? |
| 00:12.17 | Qwell | use linux |
| 00:12.50 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:13.13 | ice_croft | that's ultimate. any semimeasures? |
| 00:14.45 | *** join/#asterisk ManxPower (n=manxpowe@44.sub-70-220-237.myvzw.com) |
| 00:16.00 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:16.12 | Nugget | I hate Linux just as much as the next guy, but I gotta admit that for Asterisk you'd be a fool to use anything else. |
| 00:16.15 | De_Mon | contact the bsd zaptel maintainer and pray |
| 00:16.49 | Nugget | even in the absence of zaptel, asterisk on BSD can be a bit of a hemorrhoid. |
| 00:16.53 | *** join/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 00:16.53 | ice_croft | hhe |
| 00:17.11 | Qwell | s/.*on // |
| 00:17.14 | ice_croft | i looked for VLDTMF on sources |
| 00:17.20 | ice_croft | found nothing |
| 00:17.58 | ice_croft | :)) |
| 00:26.13 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:26.24 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 00:26.40 | ice_croft | yo!!! |
| 00:26.43 | ice_croft | ppl! |
| 00:26.50 | ice_croft | it works!! :))))) |
| 00:26.59 | endre | glad to hear that |
| 00:27.13 | ice_croft | it needs ./configure --with-zaptel_vldtmf :)))) |
| 00:27.45 | endre | wtf is VL btw? |
| 00:28.24 | ice_croft | don't know. google says |
| 00:28.57 | ice_croft | http://mtaipe.zonaz.net/wiki/build-asterisk-1.4 |
| 00:29.35 | ice_croft | i didnt meet that in ./configure --help |
| 00:31.13 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 00:31.34 | esaym | Howdy folks |
| 00:31.55 | ice_croft | pretty well |
| 00:32.02 | ice_croft | for now:))))) |
| 00:32.16 | esaym | Whats is a good provider to get a phone number for my asterisk server? |
| 00:38.44 | ice_croft | damn i'm good |
| 00:38.46 | ice_croft | hehe |
| 00:39.32 | ice_croft | and there was no need to portupgrade |
| 00:40.39 | *** join/#asterisk dklima (n=dklima@201.47.19.50.adsl.gvt.net.br) |
| 00:58.59 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-187.pskn.east.verizon.net) |
| 01:01.17 | ectospasm | any of y'all know what the default URL for asterisk-gui configuration is? http://hostname/ and http://hostname/asterisk don't work... |
| 01:01.27 | ectospasm | I just get 404's |
| 01:01.55 | *** part/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 01:03.41 | Iamnach0 | is it not port 8088 ? |
| 01:04.28 | Iamnach0 | take a look at http.conf in your /etc/asterisk directory |
| 01:04.29 | ectospasm | no... it looks like it's port 5038 on this system... testing... |
| 01:05.07 | ectospasm | it's doing something... |
| 01:05.29 | ectospasm | well, it seems to be hanging |
| 01:05.37 | ectospasm | hrmm... |
| 01:14.25 | *** join/#asterisk AntiInit (n=chris@host81-132-176-11.range81-132.btcentralplus.com) |
| 01:14.44 | AntiInit | hi |
| 01:23.00 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 01:32.01 | ManxPower | The proper place to ask AsteriskGUI questions is, astoundingly enough, #Asterisk-GUI |
| 01:32.25 | ManxPower | ~gui |
| 01:32.25 | jbot | gui is probably (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 01:32.27 | ManxPower | ~trixbox |
| 01:32.28 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 01:32.29 | ManxPower | ~amp |
| 01:32.29 | jbot | i heard amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 01:32.30 | ManxPower | ~zeek |
| 01:32.45 | ManxPower | ~asteriskgui |
| 01:36.43 | [TK]D-Fender | ~zeeek |
| 01:36.44 | jbot | [zeeek] someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 01:37.45 | watchy | tk i'll show you the good stuuff baby |
| 01:40.01 | [TK]D-Fender | "Snorting lines of snow.. on a one horse... |
| 01:40.17 | watchy | haha |
| 01:41.36 | [TK]D-Fender | 1 week to cell-phone upgrade.... |
| 01:42.09 | ManxPower | [TK]D-Fender: What are you upgrading to? |
| 01:42.26 | ManxPower | BTW, I finally found a use for the camera on my new phone. |
| 01:42.47 | [TK]D-Fender | ManxPower, Motorolla E815 to HTC Touch (Bell CDMA) |
| 01:42.47 | ManxPower | Held my finger over the lens, took a picture, made it the cell phone screen background. |
| 01:43.24 | [TK]D-Fender | ManxPower, lol. Actually the flash on my E815 has served as a flashlight on a number of occasions :) |
| 01:43.50 | ManxPower | Mine was $40 at Walmart, no contract 8-) |
| 01:44.19 | ManxPower | The package said something like "With a .3 megapixel camera!" |
| 01:45.39 | [TK]D-Fender | ManxPower, CrapTASTIC! |
| 01:46.18 | watchy | haha |
| 01:46.31 | ManxPower | *nod* 640x480 resolution, so that is about .3 megabixel, I guess. |
| 01:46.38 | ManxPower | and megapixel too. |
| 01:46.52 | ManxPower | ~seen Shido6 |
| 01:47.01 | jbot | shido6 <n=shido6@204.126.120.132> was last seen on IRC in channel #asterisk, 2d 8h 17m 52s ago, saying: 'for that phone'. |
| 01:47.03 | *** join/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 01:47.23 | [TK]D-Fender | My current plan : $30 = Unlm eve/wknd @6pm, 250 day, VM,CW,3WC,CID. Getting unlimited mobile browser for $7 more and might get a cheaper plan.... |
| 01:47.31 | [TK]D-Fender | ManxPower, Yup, VGA |
| 01:47.38 | ManxPower | I refuse to do a contract anymore. |
| 01:48.24 | [TK]D-Fender | ManxPower, Whats your monthly usage & expenditue on cell? |
| 01:48.29 | ManxPower | One of my clients pays for my Verizon USB EVDO Rev A "modem" and service. |
| 01:48.54 | ManxPower | [TK]D-Fender: $50 of mins lasts me 3 -4 weeks |
| 01:49.10 | BigCanOfTuna | I'm hoping someone can point me in the right direction. I have asterisk running on my local development machine. What I would like to do is have it bridge a call to my asterisk server at home so that I can call an extension through my home server. My development machine has a dynamic IP and my home machine has a static IP. |
| 01:49.24 | [TK]D-Fender | ManxPower, well... my $30 covers so much more it seems... |
| 01:49.46 | ManxPower | that gives me 300 mins + unlimited mobile to mobile + unlimited nights/weekends. |
| 01:50.10 | ManxPower | Where I live I have a choice of exactly one carrier. |
| 01:50.27 | [TK]D-Fender | BigCanOfTuna, lookup "asterisk dual servers" on the WIKI for some guides |
| 01:50.35 | [TK]D-Fender | ManxPower, Verizon IIRC, no? |
| 01:50.38 | BigCanOfTuna | Thanks! |
| 01:50.53 | ManxPower | They are the only one with EVDO at all, and the only one with voice service where I live. Yes, Verizon. |
| 01:51.01 | [TK]D-Fender | http://www.bell.ca/shopping/PrsShpWls_PrdClpDetail.page?language=en®ion=QC&languageToggle=true&content=/portlets/personal/wireless/product_details.jsp&metaKey=PrsShpWls_Content&wlcs_catalog_item_sku=66393&INT=MOB_SA_Q4_XMAS07_RP_BTN_satsite_BuyNow_HTCTouch |
| 01:51.04 | [TK]D-Fender | thats the phone |
| 01:51.34 | [TK]D-Fender | not sure how much my 2yr HW upgrade discount will come to. Somewhere between 100-200$ I'd bet |
| 01:51.37 | watchy | nice |
| 01:52.31 | [TK]D-Fender | With unlimited internet I'll be happy camper. Google Maps on demand, e-mail pickup of home/work VM's. |
| 01:52.35 | ManxPower | Looks nice. |
| 01:52.51 | watchy | hey tk: on Answer is there anyway to make it let me dial more then 1 digit? its giving issues if i dial something with more then 1 |
| 01:53.11 | watchy | i like my iphone |
| 01:53.15 | watchy | i just wish it was 3g |
| 01:53.20 | ManxPower | watchy: Answer has nothing to do with anything. |
| 01:53.27 | [TK]D-Fender | watchy, before Q3'08 it will be |
| 01:53.41 | watchy | i hope |
| 01:53.58 | watchy | well i got a basic answer set up running Background(file) |
| 01:54.09 | watchy | but if i hit say 200 102 it doesnt like that |
| 01:54.13 | watchy | i mean 102 |
| 01:54.28 | ManxPower | then your dialplan is at fault. |
| 01:55.19 | watchy | oh |
| 01:55.25 | watchy | i think eye gnos whoi |
| 01:55.49 | watchy | i guess if its not in the same context |
| 01:55.53 | watchy | of course it can dial it |
| 01:56.09 | watchy | cant |
| 01:56.15 | watchy | why must i be such a newbie |
| 01:58.00 | watchy | what do you guys do for custom menus? |
| 01:58.06 | watchy | pay a sexy voiced dude? |
| 01:59.13 | ManxPower | I'll do the prompts or the customer can have someone do the prompts. On systems I admin, most of the prompts can be re-recorded just using the dialplan and dialing an extension |
| 01:59.34 | watchy | custom stuff u wrote? |
| 01:59.43 | ManxPower | *nod* |
| 01:59.51 | watchy | you impress me |
| 02:00.27 | watchy | someday i hope to beable to do that |
| 02:00.29 | ManxPower | It's more of a jumble of code and half finished routines, but it works and the customers can use it. |
| 02:00.33 | [TK]D-Fender | watchy, GotoIf, Record, Set <- big bloody deal :p |
| 02:00.42 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:00.45 | watchy | haha |
| 02:01.06 | [TK]D-Fender | watchy, I have a numbered voice prompt script so that users can review/update all their prompts. |
| 02:01.43 | watchy | do you use a gui to manage your installs? |
| 02:02.01 | AntiInit | hi |
| 02:02.15 | ManxPower | watchy: No need to get vulgar. |
| 02:02.46 | AntiInit | does asterisk provide data as well as voice calls? |
| 02:02.48 | watchy | haha the guy i work with uses the coolest thing in the world |
| 02:02.52 | watchy | its called "freepbx" |
| 02:03.00 | watchy | its so mega awesome |
| 02:03.15 | watchy | i mean "stupid" |
| 02:03.18 | ManxPower | [TK]D-Fender: Do you think he might not be well? |
| 02:03.55 | watchy | Personally, I'm so anti gui on * it pisses my co worker off |
| 02:04.23 | [TK]D-Fender | AntiInit, and what do you mean by "data call"? |
| 02:04.35 | watchy | i think modem usage |
| 02:04.40 | watchy | is what he means? |
| 02:04.53 | watchy | -- Saved useragent "PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036" for peer 211 |
| 02:04.53 | ManxPower | My scripts are designed so you can plug an AGI at the beginning of any extension which just sets channel varliables for the script to use. |
| 02:05.05 | watchy | i really need to upgrade these phones |
| 02:07.25 | ManxPower | I just did a search on "CFL sunlight" and the 2nd returned result was "25watt cfl - Marijuana Growing", and I said to myself "Ah, that makes sense." |
| 02:08.11 | watchy | heh |
| 02:08.19 | watchy | where you from manx? |
| 02:09.15 | AntiInit | [TK]D-Fender: i.e. can i link 2 isdn terminals via asterisk and allow them to transfer data |
| 02:09.23 | AntiInit | i.e. as in dial up modem |
| 02:09.54 | [TK]D-Fender | AntiInit, highly doubt it... |
| 02:10.06 | AntiInit | k, thanks |
| 02:10.11 | ManxPower | I currently reside in Alabama about a 90 min drive SE of Digium. |
| 02:10.15 | *** part/#asterisk AntiInit (n=chris@host81-132-176-11.range81-132.btcentralplus.com) |
| 02:10.42 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:11.11 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) [NETSPLIT VICTIM] |
| 02:13.49 | watchy | Set(TIMEOUT(digit)=timeout) |
| 02:13.54 | watchy | digit = ? |
| 02:14.04 | watchy | the type? |
| 02:14.15 | [TK]D-Fender | watchy, how much time BETWEEN digits |
| 02:14.18 | watchy | n/m |
| 02:14.25 | watchy | i found it at the top of the manpage |
| 02:14.34 | [TK]D-Fender | ~cluebat watchy |
| 02:14.34 | jbot | ACTION pulls out a ClueBat (tm) and thwaps watchy. |
| 02:14.34 | watchy | i don't wanna ask retarded stuff if i can find it yself |
| 02:14.56 | [TK]D-Fender | watchy, justy FYI but... you're failing :) |
| 02:15.02 | watchy | hey i'm trying |
| 02:15.08 | watchy | atleast i'm reading voip-wiki man |
| 02:15.24 | watchy | and making my own IVR by hand instead of using freepbx |
| 02:15.33 | watchy | you should be proud i'm not a trixbox sheep |
| 02:17.10 | watchy | i'm sure you've seen many young asterisk warriors fall to the gui |
| 02:17.29 | *** join/#asterisk jdunck (n=jdunck@adsl-76-246-175-155.dsl.rcsntx.sbcglobal.net) |
| 02:18.41 | [TK]D-Fender | watchy, and HOW long have you been here? |
| 02:18.56 | watchy | on and off a year maybe |
| 02:19.12 | watchy | but I don't deal with phone systems everyday, i rarely do |
| 02:19.29 | watchy | but i just landed a job doing pretty much nothing but * |
| 02:19.57 | watchy | so I will be teaching myself and learning everything I can about * |
| 02:21.36 | watchy | either way I owe u for a few steaks |
| 02:24.23 | *** join/#asterisk [hC] (n=a@66.119.167.162) |
| 02:24.36 | watchy | now that my basic IVR is answering and yelling at me |
| 02:24.36 | [hC] | has anyone created a manager app to sanitize cli output? |
| 02:25.02 | [hC] | i tried just going to verbose 1, and using (Verbose,1,blah) to fit my needs but you dont get progress messages |
| 02:25.03 | watchy | i need to find out what this company wants to do |
| 02:30.54 | *** join/#asterisk pitbossy_ (n=frankjr@12.46.64.130) |
| 02:31.07 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 02:33.53 | watchy | hrm |
| 02:34.01 | watchy | i have a wierd request i have no idea how to do |
| 02:34.30 | watchy | think you can help manx |
| 02:35.47 | watchy | from my ivr I want to dial a # if 400 is dialed |
| 02:35.51 | watchy | thats quiet simple |
| 02:36.32 | *** join/#asterisk salzh (n=salzh@124.77.15.177) |
| 02:36.57 | watchy | i wanna beable to change that # remotely |
| 02:38.25 | watchy | i bet there has to be a way to set a variable in * |
| 02:38.29 | watchy | that stays global |
| 02:38.57 | watchy | i gotta figure out how to set it using a phone |
| 02:41.35 | watchy | how do i set a variable that never goes away ? |
| 02:41.50 | *** join/#asterisk techie (n=techie@adsl-76-214-5-177.dsl.lsan03.sbcglobal.net) |
| 02:42.11 | [hC] | Set(__VARIABLE=something) |
| 02:42.21 | [TK]D-Fender | watchy, Global variables ar the preferred way "show function DB" |
| 02:42.25 | *** join/#asterisk hohum (n=dcorbe@wsip-70-166-81-42.sd.sd.cox.net) |
| 02:42.26 | [TK]D-Fender | [hC], EW! |
| 02:42.38 | [TK]D-Fender | [hC], vars like that don't survive * restarts IIRC |
| 02:42.38 | watchy | ah |
| 02:42.40 | esaym | is http://www.inphonex.com/ good with asterisk? |
| 02:42.45 | [hC] | [tk]: :) |
| 02:42.47 | watchy | but once say the cals over, do they stay in the system? |
| 02:42.50 | esaym | does anyone recommend anything else? |
| 02:43.07 | [hC] | yeah you could use AstDB |
| 02:44.00 | watchy | SetGlobalVar(RemoteSupport=8187838) |
| 02:45.04 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 02:45.06 | [TK]D-Fender | watchy, Use AstDB.... far saner |
| 02:45.56 | watchy | wtf is astdb? |
| 02:46.01 | JT | googleable |
| 02:46.21 | watchy | ah |
| 02:47.35 | watchy | wtf do i need astdb if all i wanna do is store like 1 variable right now |
| 02:47.43 | coppice | googleable could be the answer to almost anything these days |
| 02:48.07 | JT | astdb will survive asterisk restarts |
| 02:48.52 | coppice | sometimes googling my own lost stuff is the easiest way to find it :-\ |
| 02:48.55 | watchy | but just for the sake of argument right now, SetGlobal will survive everything but a restart right? |
| 02:50.18 | watchy | whats the command to see variables in *? |
| 02:54.26 | [TK]D-Fender | watchy, using is take 1 line, just like a var, so big friggen deal. You really should stick with FreePBX, you clearly aren't cut out for this |
| 02:56.32 | *** part/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 03:07.09 | BigCanOfTuna | Would someone be willing to have a look at how I have my IAX.conf files configured for a dual asterisk configuration? I am trying to bridge calls from my laptop to a asterisk server (no need to go the other way)....here are the relevent code snippets: http://pastebin.ca/808994 |
| 03:08.17 | [TK]D-Fender | hostname=dynamic <- not valid |
| 03:08.31 | [TK]D-Fender | and you can't have BOTH be "dynamic" |
| 03:08.40 | [TK]D-Fender | one has to have a FIXED host. |
| 03:08.58 | [TK]D-Fender | the one that registers should have host=[ip or FQDN] |
| 03:09.10 | BigCanOfTuna | [TK]D-Fender: OK. so is the 'peer' the Server, or my laptop? |
| 03:09.39 | [TK]D-Fender | BigCanOfTuna, yes, your [peer] should be "host=[ip or FQDN]" |
| 03:09.54 | [TK]D-Fender | BigCanOfTuna, [serverb] looks ok |
| 03:11.07 | *** join/#asterisk MmixX (n=mmixx@202.58.249.5) |
| 03:12.56 | BigCanOfTuna | [TK]D-Fender: So on my laptop (the user) needs to register with the server, so in the laptop's iax.conf, I register => username:password@server.name ? |
| 03:13.03 | *** join/#asterisk mistermocha (n=chef@adsl-75-22-54-73.dsl.irvnca.sbcglobal.net) |
| 03:13.15 | mistermocha | ahoy mateys |
| 03:13.21 | mistermocha | anyone in the house? |
| 03:14.11 | [TK]D-Fender | BigCanOfTuna, your dynaim one acts like a client and registeres, and has a peer that has a FIXED host. the server is jsut a dynamic friend |
| 03:15.07 | mistermocha | what may cause a IAX POKE to come in on the wrong port? |
| 03:16.35 | mistermocha | this is driving me bananas |
| 03:17.53 | BigCanOfTuna | [TK]D-Fender: OK, so I have made some changes, but what I am seeing is on my user (laptop): Registration of 'xyzuser' rejected: 'Registration Refused' from: '192.168.2.21' and on the server, I get: No registration for peer 'xyzuser' (from 192.168.2.20)...what am I missing? |
| 03:18.31 | [TK]D-Fender | BigCanOfTuna, pastebin your new configs |
| 03:20.08 | mistermocha | ... |
| 03:21.31 | BigCanOfTuna | [TK]D-Fender: Thanks for your help! : http://pastebin.ca/809000 |
| 03:21.48 | *** join/#asterisk troy- (n=troy@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 03:22.56 | [TK]D-Fender | BigCanOfTuna, under |
| 03:22.56 | [TK]D-Fender | [macbuntu] host=dynamic |
| 03:23.04 | [TK]D-Fender | Ditch -> auth=plaintext |
| 03:23.42 | BigCanOfTuna | done. |
| 03:25.28 | BigCanOfTuna | http://pastebin.ca/809000 |
| 03:25.35 | BigCanOfTuna | errr.....No registration for peer 'macbuntu' |
| 03:28.19 | [TK]D-Fender | BigCanOfTuna, NEW pastbion please. |
| 03:28.42 | BigCanOfTuna | [TK]D-Fender: Sure, one second....I must have the register messed up. |
| 03:30.29 | BigCanOfTuna | [TK]D-Fender: http://pastebin.ca/809006 |
| 03:32.09 | [TK]D-Fender | under your peer - fromuser=macbuntu |
| 03:33.44 | BigCanOfTuna | [TK]D-Fender: This is probably where I am getting confused..... when you say 'under your peer' , you mean on my server's IAX.conf, right? |
| 03:34.23 | [TK]D-Fender | # |
| 03:34.23 | [TK]D-Fender | [anassina] is you PEER |
| 03:34.32 | [TK]D-Fender | type=peer <- |
| 03:34.57 | [TK]D-Fender | username=macbuntu <- remove from [macbuntu] |
| 03:35.19 | *** join/#asterisk leprasmurf (n=tforbes@ool-4576a090.dyn.optonline.net) |
| 03:36.24 | leprasmurf | hello all |
| 03:37.19 | leprasmurf | is there somewhere I can go or someone I can talk to about setting up asterisk and see what I can do? |
| 03:43.46 | [TK]D-Fender | leprasmurf, can you try to rephrase that a little. |
| 03:46.59 | leprasmurf | basically, I have a combination cable modem / VOIP interface (MTA?). |
| 03:47.08 | *** join/#asterisk ming_zym (n=ming_zym@124.14.236.56) |
| 03:47.41 | leprasmurf | I'd like to be able to setup VOIP clients for all the standard reasons (remote use, advanced mythtv functions, etc...) |
| 03:48.29 | [TK]D-Fender | leprasmurf, basically you need your Cable modem to keep out of the way because odds are you're locked out of it. |
| 03:48.35 | leprasmurf | my company controls my modem, and I'm wondering if I would be able to setup asterisk to relay from the internet to my cable modem via the network |
| 03:49.09 | [TK]D-Fender | leprasmurf, Your cable modem is a dead issue. not really usable. |
| 03:49.46 | leprasmurf | [TK]D-Fender: ok, but would I be able to relay audio/voice through a modem or something like that? |
| 03:50.19 | [TK]D-Fender | leprasmurf, what do you mean by "RELAY" and "MODEM"? |
| 03:51.16 | leprasmurf | [TK]D-Fender: strict definitions, relay = take input and send to output; modem = communication device |
| 03:51.50 | leprasmurf | [TK]D-Fender: I'm pretty much just trying to understand my options |
| 03:51.53 | [TK]D-Fender | leprasmurf, ok, your "cable has nothing to do with anything here. ok? It is 100% useless. |
| 03:52.03 | [TK]D-Fender | leprasmurf, So fine, you have * to start with. what now? |
| 03:52.17 | [TK]D-Fender | "cable modem"* |
| 03:53.33 | leprasmurf | I'm sorry, I was saying it screwed up, modem != cable modem; modem = pci card, rj-11 connectors, standard telephone device |
| 03:54.47 | [TK]D-Fender | leprasmurf, ok, that is useless too. |
| 03:54.56 | leprasmurf | [TK]D-Fender: oic, ok |
| 03:55.52 | leprasmurf | [TK]D-Fender: I did an nmap scan of my cable modem, and noticed "1720/tcp filtered H.323/Q.931" is that consistent with the cable modem being useless? |
| 03:57.46 | [TK]D-Fender | leprasmurf, Good sign. Means it isn't using SIP. That should mean it'll keep out of the way |
| 03:58.14 | leprasmurf | [TK]D-Fender: even if it is filtered? |
| 03:58.23 | [TK]D-Fender | leprasmurf, H.323 doesn't matter |
| 03:58.50 | leprasmurf | [TK]D-Fender: cool, is there a way to test functionality before I attempt to setup a fulltime box? |
| 04:00.30 | [TK]D-Fender | leprasmurf, just install * and a softphone on anothe PC |
| 04:01.01 | leprasmurf | [TK]D-Fender: and just point * towards my cable modem on that port? |
| 04:01.55 | [TK]D-Fender | leprasmurf, no, this has nothing to do with your cable modem |
| 04:02.41 | leprasmurf | [TK]D-Fender: oh, maybe I'm just thinking to far into this |
| 04:03.56 | leprasmurf | [TK]D-Fender: I don't want to flood the channel with newb questions. what would be the best place to gather info? asterisk.org? |
| 04:04.11 | [TK]D-Fender | ~book |
| 04:04.12 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 04:04.14 | [TK]D-Fender | ^^^ |
| 04:04.25 | leprasmurf | [TK]D-Fender: tyvm |
| 04:04.29 | [TK]D-Fender | np |
| 04:30.52 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 04:43.25 | *** join/#asterisk pitbossy_ (n=frankjr@12.46.64.130) |
| 04:44.14 | watchy | hey tk |
| 04:44.31 | watchy | whats a asterisk command that will store dialed digits to a variablr |
| 04:50.59 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 04:57.26 | De_Mon | watchy Set() ? |
| 04:57.40 | De_Mon | watchy or maybe you want show application Read |
| 04:57.53 | watchy | what i want is to Dial say 5000 |
| 04:58.05 | watchy | and beable to set a variable using the keypad |
| 04:58.11 | watchy | just #s |
| 05:03.51 | De_Mon | read will read dtmf keys into a variable |
| 05:04.41 | watchy | i kinda ghettoed it |
| 05:04.45 | watchy | till i know the correct way |
| 05:05.22 | watchy | look at my code its retarded but it works |
| 05:05.32 | watchy | its a bad ideai think to do this |
| 05:06.10 | watchy | exten => _NXXXXXX,1,Set(TIMEOUT(digit)=5) |
| 05:06.10 | watchy | exten => _NXXXXXX,2,Set(TIMEOUT(response)=10) |
| 05:06.10 | watchy | exten => _NXXXXXX,3,SetGlobalVar(RemoteSupport=${EXTEN}) |
| 05:06.14 | watchy | but it does what i want |
| 05:09.47 | watchy | is the Read variable global? |
| 05:11.01 | *** join/#asterisk thinko (i=jdoe6alp@smaug.rackdragon.com) |
| 05:17.02 | Corydon76-dig | Set(GLOBAL(RemoteSupport)=${EXTEN}) is preferred now |
| 05:17.59 | watchy | oh |
| 05:18.00 | watchy | exten => 93,1,Background(welcome) |
| 05:18.00 | watchy | exten => 93,2,Read(RemoteSupport) |
| 05:18.00 | watchy | exten => 93,3,SetGlobalVar(RemoteSupport=${EXTEN}) |
| 05:18.06 | watchy | thats my solution |
| 05:18.16 | watchy | is it ok? |
| 05:18.27 | Corydon76-dig | Does it work? |
| 05:18.37 | watchy | amazingly yes |
| 05:18.50 | Corydon76-dig | Note that EXTEN is always 93 in that situation |
| 05:19.08 | watchy | yea |
| 05:19.34 | watchy | oh shit |
| 05:19.39 | watchy | i meant to change that |
| 05:19.40 | watchy | hold onm |
| 05:19.46 | Corydon76-dig | BTW, you can also do: Read(GLOBAL(RemoteSupport)) |
| 05:20.01 | watchy | oh |
| 05:20.04 | watchy | really? |
| 05:20.09 | Corydon76-dig | Yep |
| 05:20.11 | watchy | well damn then that gets rid of one set |
| 05:20.22 | watchy | and line |
| 05:21.03 | Corydon76-dig | I think that's what you wanted, right? |
| 05:21.09 | watchy | very much so |
| 05:21.13 | watchy | Simple but very efective |
| 05:21.32 | watchy | exten => 93,4,SayDigits(${RemoteSupport}) |
| 05:21.38 | watchy | how come that didnt work? |
| 05:22.37 | Corydon76-dig | One of the neat things about the GLOBAL dp is that now, you can access global variables, even if a channel variable has the same name |
| 05:22.58 | watchy | hrm actually |
| 05:23.01 | watchy | it doesnt work |
| 05:23.05 | Corydon76-dig | watchy: because you eliminated step 3, perhaps? |
| 05:23.27 | watchy | i have to elimanate #2 |
| 05:23.35 | Corydon76-dig | This is why we use priority n, to automatically reorder priorities |
| 05:23.41 | watchy | cant use a background before read |
| 05:23.57 | Corydon76-dig | Read takes multiple arguments |
| 05:23.59 | watchy | i can but it seems if i dial anything while background is running |
| 05:24.09 | watchy | it trys to dial an extension |
| 05:24.11 | Corydon76-dig | one of which can be a prompt |
| 05:24.19 | Corydon76-dig | Read(varname,prompt) |
| 05:24.23 | watchy | yea |
| 05:24.54 | Corydon76-dig | So you don't need Background |
| 05:25.56 | watchy | Read(variable[|filename][|maxdigits][|option][|attempts][|timeout]) |
| 05:25.56 | *** join/#asterisk metfan2007 (n=metfan20@189.180.217.155) |
| 05:25.56 | Corydon76-dig | Read(GLOBAL(RemoteSupport),welcome) |
| 05:28.58 | watchy | you %100 sure |
| 05:29.05 | watchy | about READ(GLOBAL)? |
| 05:29.19 | Corydon76-dig | I can check the code |
| 05:29.29 | watchy | I know Set(Global stuff is fine but it dont seem to be a read(global) |
| 05:29.33 | metfan2007 | hi all!! I just installed today a new TDM2400 card with FXO modules connected to PSTN, in a new Proliant server and a IP phone, the issue is that during a call a ugly and loud sound starts inmmediatly in the IP party side, and in the FXO side it sounds fine... any idea?? I have recorded the sounds if you want to check it |
| 05:29.42 | watchy | i dunno alot but it didnt seem to store it correctly |
| 05:29.57 | watchy | exten => 93,1,Read(Global(RemoteSupport),welcome) |
| 05:29.57 | watchy | exten => 93,2,SayDigits(${RemoteSupport}) |
| 05:30.04 | watchy | it played the incorrect digits |
| 05:30.07 | Corydon76-dig | Are you using 1.2 or 1.4? |
| 05:30.12 | watchy | 1.2 |
| 05:30.18 | Corydon76-dig | There's the problem |
| 05:30.19 | watchy | oh hell |
| 05:30.22 | watchy | thats why i bet |
| 05:30.31 | Corydon76-dig | plus GLOBAL is all caps |
| 05:30.43 | Corydon76-dig | all dialplan functions are case-sensitive |
| 05:30.49 | watchy | so i need to put in a Set(Global? |
| 05:30.55 | watchy | to make it global |
| 05:31.10 | Corydon76-dig | No, you'll need to use SetGlobalVar |
| 05:31.20 | watchy | oh |
| 05:31.22 | Corydon76-dig | We've made improvements to make things easier in 1.4 |
| 05:31.29 | watchy | thats what i meant |
| 05:31.45 | watchy | well i'llbe moving this box to 1.4 but not right now |
| 05:31.50 | Corydon76-dig | The GLOBAL dp is not in 1.2 |
| 05:31.56 | WilliamK | does anyone know if weighted random was added to queues as a strategy? I remember seeing it a while back in some notes |
| 05:33.41 | *** join/#asterisk chesty (n=chesty@unconcerned.org) |
| 05:36.01 | watchy | i think i got it cory |
| 05:36.50 | watchy | <PROTECTED> |
| 05:36.51 | watchy | hmm |
| 05:37.24 | watchy | exten => 93,2,SetGlobalVar(RemoteSupport=${NewRemoteSupport}) |
| 05:39.14 | watchy | <PROTECTED> |
| 05:39.15 | watchy | <PROTECTED> |
| 05:39.15 | watchy | <PROTECTED> |
| 05:39.15 | watchy | <PROTECTED> |
| 05:39.17 | watchy | awee yea |
| 05:45.20 | watchy | its so simple |
| 05:45.26 | watchy | but i feel i did something useful |
| 05:47.01 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 05:47.50 | watchy | thank you Corydon76-dig |
| 05:47.58 | Corydon76-dig | <PROTECTED> |
| 05:49.07 | watchy | now the remote support team can change the # every monday without me having to do it manually |
| 05:49.13 | watchy | it saves everyone time |
| 05:59.51 | De_Mon | hehe |
| 06:00.42 | De_Mon | http://www.vgcats.com/comics/extras/stillalive.php |
| 06:00.47 | De_Mon | its sooo cute! |
| 06:15.41 | watchy | hahaha |
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| 07:14.42 | blitzrage | evening all |
| 07:16.14 | [TK]D-Fender | How are you gentlemen!! |
| 07:22.48 | coppice | http://www.youtube.com/watch?v=5xPood_vRUQ&feature=related |
| 07:32.04 | [TK]D-Fender | coppice, lol |
| 07:32.16 | [TK]D-Fender | coppice, I love Deep Purple... |
| 07:32.34 | coppice | Huh? that song is Kansas |
| 07:33.16 | [TK]D-Fender | coppice, Was it? Covered by so many.... |
| 07:33.43 | coppice | http://www.youtube.com/watch?v=CB17uWuBrL0 |
| 07:39.16 | coppice | I love the way the violinist walks around looking like an idiot, because there is no violin part in the song. :-) |
| 07:40.34 | [TK]D-Fender | I liked the violin in Whitesnake's "Still of the Night" :) |
| 07:43.25 | blitzrage | I am well |
| 07:43.31 | blitzrage | well intoxicated.... |
| 07:47.26 | coppice | [TK]D-Fender: its hard to find material by some of the best rock violinists. Michael Dreyfuss of McKendree Spring was pretty amazing to listen to. If you try Googling for him now, all you really find is his later career in medical research. |
| 07:48.28 | [TK]D-Fender | coppice, years wasted trying to save humanity! Oh the waste! |
| 07:49.27 | coppice | yeah, well lots of people do that, but not many can play with enough emotion to get an instrumental banned from US radio :-) |
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| 07:57.07 | nephfl | anyone here know how to pass a variable from an html form to a bash cgi script? |
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| 08:21.09 | coppice | RFCs are a pain. Why can't they have revision 2, 3, etc as they update stuff? Why do they insist on fragmenting a single functional entity across so many RFCs? |
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| 12:00.03 | joelsolanki | Good Morning All |
| 12:00.12 | joelsolanki | Problem in retrieving voicemail |
| 12:00.14 | joelsolanki | http://pastebin.com/m3ebf110d |
| 12:00.19 | joelsolanki | plz check |
| 12:00.27 | joelsolanki | it is context=default problem but not able to sort it out |
| 12:00.31 | joelsolanki | any hints plz |
| 12:01.33 | Psychobilly | im having some prbs with jitter buffer on sip channels with asterisk 1.4.15, when i set it to adaptive (jbimpl=adaptive) in sip.conf i have no voice in my calls, all works fine when i switch back to fixed buffer |
| 12:02.37 | Psychobilly | in there some problem with adaptive mode? do i have to change something in my configs? |
| 12:10.43 | Psychobilly | and somehting else, any tips/suggestions for astrisk and T38? as i read asterisk cannot terminate T38, is there a way to make it work? maybe using asterisk in combination with something else |
| 12:13.25 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 12:14.54 | coppice | you could pass the T.38 through to t38modem or Callweaver |
| 12:15.49 | Psychobilly | i found some infos about t38modem, does it work with sip? |
| 12:16.07 | coppice | recent versions do, if used in the right way |
| 12:16.14 | Psychobilly | aha |
| 12:16.49 | ussrback | how can i limit number of participants in meetme? |
| 12:17.59 | Psychobilly | the prob is that im working on hardware with very limited resources, ram and disk space are limited, can t38modem be easily installed and operated in such an eviroment? |
| 12:18.29 | Psychobilly | i m talking about arm cpu, 32mb ram and a small flash disk |
| 12:19.00 | coppice | i doubt it will fit there. it needs the whole opal system to provide its SIP stack |
| 12:19.08 | Psychobilly | ah |
| 12:20.26 | Psychobilly | so lets think about plan B :p switching to callwaver, maybe its not the right room to ask but anyone had any expirience with it? |
| 12:21.21 | Psychobilly | i know its a fork of asterisk 1.2 and it supports t38 termination, but nothing more |
| 12:24.25 | coppice | callweaver is doing T.38 termination on small ARM and MIPS machines. It doesn't do T.38 gateway on those right now, as it uses too much floating point. as more of the modem code has fixed point compile options, Callweaver should be able to do T.38 gateway operation on those boxes too. |
| 12:26.43 | *** join/#asterisk bjweeks (n=bjweeks@unaffiliated/bjweeks) |
| 12:27.15 | Psychobilly | can it be considered as a viable replacement of asterisk? (suppoting sip calls and some other stuff not only t38) |
| 12:27.56 | coppice | that depends mostly on whether you need some very specific feature which asterisk has and callweaver doesn't |
| 12:28.21 | Psychobilly | i care most about stability etc |
| 12:28.45 | Psychobilly | can it be considered stable? i think its still in rc stage |
| 12:29.01 | coppice | its generally more stable than asterisk |
| 12:29.16 | Psychobilly | hm thast good |
| 12:29.57 | Psychobilly | i ll give it a try |
| 12:30.03 | Psychobilly | i have some channel drivers to port back to callwaver then and start testing |
| 12:30.13 | Psychobilly | thanks a lot for the info coppice |
| 12:31.04 | bjweeks | the real battle will be if CW can keep up with asterisk, which at this point it doesn't seem to be |
| 12:31.28 | coppice | well, asterisk isn't exactly moving very fast |
| 12:31.43 | Psychobilly | i have no expirience with cw at all |
| 12:32.28 | bjweeks | asterisk has more "pieces of flair" than CW, if that is useful in a PBX is debatable ;) |
| 12:32.49 | coppice | do you mean more woolly bits? |
| 12:33.33 | bjweeks | not sure what you mean by that |
| 12:33.39 | Psychobilly | and btw anyone has any idea why i have this prob with jitter buffer in sip that i described earlier? |
| 12:40.50 | coppice | I think its time there was a proper free T.37 implementation :-\ |
| 12:42.25 | ussrback | where should i put language=de to use german prompts? in extension.conf ? |
| 12:45.51 | dklima | zapata.conf |
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| 13:00.40 | ussrback | i have added language=de in my sip.conf but it still uses english files. how can i fix this? |
| 13:02.30 | Psychobilly | sip reload in asterisk console and then sip show perr foo to be sure that its language seting is set to de |
| 13:03.05 | Psychobilly | s/perr/peer |
| 13:03.16 | Psychobilly | and mak sure you have installed the de sound files |
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| 13:06.15 | ussrback | yes i have installed it |
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| 13:38.22 | mvanbaak | anyone here that can give some info about jabber/gtalk support in asterisk ? |
| 13:38.29 | mvanbaak | I have a connection to talk.google.com |
| 13:38.32 | mvanbaak | and added 2 buddies |
| 13:38.55 | mvanbaak | is it possible to set my asterisk account to busy when one of my phones is offhook ? |
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| 13:57.28 | mvanbaak | ok, JabberSend is working |
| 13:57.32 | bjweeks | mvanbaak: I think their is a func that does it, let me check |
| 13:58.58 | mvanbaak | bjweeks: ok |
| 13:59.03 | mvanbaak | I cant find it |
| 14:00.32 | bjweeks | mvanbaak: JabberStatus() |
| 14:01.13 | mvanbaak | you can use that to set status ? |
| 14:01.28 | mvanbaak | the documentation only states it will read the status of a contact |
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| 14:52.29 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 15:08.39 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:08.55 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:08.55 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:09.44 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 15:12.21 | ussrback | hi all |
| 15:14.50 | ussrback | how can i limit number of conference participants in meetme? |
| 15:18.08 | DarKnesS_WolF | i have a fast question .. i'm trunking 2 asterisk server one of them is NATed and no port fwd ... so i'm trying to trunk them togither so mak incoming and outgoing to each of them |
| 15:18.40 | DarKnesS_WolF | and i did sometning like this registe => username:password@peername-since-idon't-have-real-ip |
| 15:18.43 | DarKnesS_WolF | any idea? |
| 15:18.45 | DarKnesS_WolF | but it didn't work |
| 15:19.34 | DarKnesS_WolF | but i get this error Rejected connect attempt from nated-server-ip-address, who was trying to reach '401@' |
| 15:20.07 | blitzrage | ussrback: use GROUP() and GROUP_COUNT() in the dialplan |
| 15:20.58 | blitzrage | DarKnesS_WolF: uhhh.... peername? that's wrong -- registration goes to the far end to tell it where you are |
| 15:21.35 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:22.33 | blitzrage | DarKnesS_WolF: what is it you're trying to do? |
| 15:22.43 | blitzrage | and what do you *expect* to happen? |
| 15:25.47 | ussrback | 10xs |
| 15:26.04 | blitzrage | 42zw |
| 15:27.03 | *** join/#asterisk shinao1 (n=shinao1@196.207.1.30) |
| 15:27.35 | DarKnesS_WolF | blitzrage: i want to trunk 2 asterisk servers one of them behind NAT |
| 15:27.40 | DarKnesS_WolF | so i don't have the real IP |
| 15:28.43 | blitzrage | that means you need to send the registration to the external IP of the other box, and have the NAT device on that remote box forward port 5060 to the asterisk server |
| 15:29.55 | blitzrage | and you'll need nat=yes in the [general] section -- I'm not positive that the internal IP address the remote box will respond from will be substituted for the external IP address though.... register might not be that smart |
| 15:30.32 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 15:31.52 | DarKnesS_WolF | blitzrage: i'm using IAX |
| 15:32.16 | *** join/#asterisk manopulus (n=andrius@cable-9-236.cgates.lt) |
| 15:32.18 | blitzrage | s/5060/4569/ -- everything else still applies |
| 15:32.31 | DarKnesS_WolF | blitzrage: i don't have access to the nat |
| 15:32.36 | blitzrage | you're screwed |
| 15:32.53 | blitzrage | how is the REGISTER packet supposed to get into the NAT'd asterisk? |
| 15:33.26 | DarKnesS_WolF | blitzrage:i have 2 asteriks server |
| 15:33.29 | DarKnesS_WolF | 1 with public ip |
| 15:33.31 | blitzrage | I know |
| 15:33.32 | DarKnesS_WolF | 1 is behind nat |
| 15:33.45 | blitzrage | and you're trying to get the public server to register with the NAT'd box? |
| 15:33.50 | DarKnesS_WolF | i were able to register the one beind nat with the public ip one |
| 15:33.55 | blitzrage | of course you can |
| 15:33.57 | blitzrage | that makes sense |
| 15:33.59 | manopulus | hi, can someone help me with G argument in dial command? upon answer, it forward both parties to context, problem, that both parties are not bridged. i calling from phone 201 to 200, and my context is exten => s,1,NoOp(), exten => s,2,Wait(200), so, answering phone (200) in wait and calling phone (201) - still ringing. how i can bridge calls? |
| 15:34.00 | DarKnesS_WolF | can't i call the nated one using the same ports |
| 15:34.07 | DarKnesS_WolF | blitzrage: ok k how :D? |
| 15:34.56 | blitzrage | DarKnesS_WolF: you don't register the public box with the NAT'd box. You hard code the IP of the public server into the peer object on the NAT'd server with host=xx.xx.xx.xx |
| 15:35.03 | blitzrage | or host=hostname.tld |
| 15:35.06 | DarKnesS_WolF | now i want both able to call ... eachothers but now the alls works from the nated one to the public one .. when the public one try to call it get rejected no authincat fild |
| 15:35.32 | DarKnesS_WolF | blitzrage: ok this is done that is how i can call the ublic server then? |
| 15:35.54 | blitzrage | Dial(IAX2/peer/number_i_want) |
| 15:36.03 | DarKnesS_WolF | blitzrage: this is from the nated |
| 15:36.05 | DarKnesS_WolF | and it works |
| 15:36.08 | blitzrage | right |
| 15:36.14 | DarKnesS_WolF | now i want to reach the nated from teh public |
| 15:36.30 | DarKnesS_WolF | so i do something like dial(IAX2/natedpeer/number) |
| 15:36.33 | blitzrage | same thing -- only the NAT'd box can register to the public server though |
| 15:36.45 | DarKnesS_WolF | blitzrage: ok when i do so i got error |
| 15:36.56 | DarKnesS_WolF | but i get this error Rejected connect attempt from nated-server-ip-address, who was trying to reach '401@' |
| 15:37.20 | blitzrage | I think you have the contexts setup wrong |
| 15:37.20 | DarKnesS_WolF | sorry s/nated-server-ip-address/public-server-ip-address |
| 15:37.33 | DarKnesS_WolF | this error on the nated box |
| 15:37.59 | blitzrage | show the exact Dial() line from the public server |
| 15:38.16 | blitzrage | it should not be an IP address or hostname |
| 15:38.35 | blitzrage | it should be the peer object that the NAT'd box registers to (host=dynamic) |
| 15:38.53 | DarKnesS_WolF | exten => 796101,1,Dial(IAX2/ev-imblb/500) |
| 15:41.10 | blitzrage | and what does the [peer] object look like on the NATd box |
| 15:42.10 | DarKnesS_WolF | on the nated ? |
| 15:42.17 | DarKnesS_WolF | host=XXXXXX domainname |
| 15:44.50 | DarKnesS_WolF | blitzrage: so ? |
| 15:44.50 | blitzrage | thats not the whole definition |
| 15:45.10 | DarKnesS_WolF | u want it all ? |
| 15:45.19 | DarKnesS_WolF | i'm using users.conf cuz it's generated with asterisk-gui |
| 15:45.31 | DarKnesS_WolF | what pattern ur looking for ? |
| 15:46.21 | blitzrage | context= |
| 15:46.34 | blitzrage | and does the context specified match a context in the dialplan? |
| 15:47.15 | DarKnesS_WolF | yes it dose |
| 15:47.18 | DarKnesS_WolF | DID_TRUNK_2 |
| 15:47.21 | DarKnesS_WolF | and it do exists |
| 15:47.22 | DarKnesS_WolF | i checked this |
| 15:47.32 | blitzrage | and that object is labeled as: [ev-imblb] and have username=ev-imblb ? |
| 15:48.20 | DarKnesS_WolF | yep |
| 15:48.48 | blitzrage | not sure -- it still looks to me like its trying to get into a context that doesn't exist |
| 15:48.57 | blitzrage | I don't use IAX2... i just use SIP |
| 15:49.04 | DarKnesS_WolF | i did this issue before :( |
| 15:49.08 | DarKnesS_WolF | but both were asterisk boxs |
| 15:49.13 | DarKnesS_WolF | i 'm getting crazy for hours now ! |
| 15:49.20 | blitzrage | when you fixed it you should have documented what you did :) |
| 15:49.36 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:49.41 | DarKnesS_WolF | blitzrage: yes :-s |
| 15:49.45 | DarKnesS_WolF | i suck in docs :-s |
| 15:49.51 | DarKnesS_WolF | everything in my head and i do forget :( |
| 15:54.46 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 15:54.55 | *** join/#asterisk Psychobilly (n=fuzz@194.219.45.122) |
| 15:55.09 | MrTelephone | hey do any of you see a problem setting the sip distroy time to 35 seconds? |
| 16:02.11 | *** join/#asterisk etfonhomey (n=chatzill@mobile-166-214-199-083.mycingular.net) |
| 16:02.47 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:03.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:10.28 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 16:13.38 | DarKnesS_WolF | blitzrage: got it to work it was a bug with this crazy GUI |
| 16:13.51 | blitzrage | yet another reason not to use a GUI |
| 16:14.09 | blitzrage | MrTelephone: it'll take a long time for the connection to time out I guess |
| 17:58.48 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:58.48 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.15 (2007/11/29), *-Addons 1.4.5 (2007/12/1), Zaptel 1.4.7 (2007/11/27), Libpri 1.4.2 (2007/10/16) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn/mantis -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 17:58.49 | Psychobilly | yes sorry, by saying 'numbers' i mean the extension numbers in sip.conf eg. [210001] |
| 18:01.15 | *** join/#asterisk ManxPower (n=manxpowe@252.sub-75-202-218.myvzw.com) |
| 18:02.46 | Nugget | I still don't understand the question. |
| 18:02.48 | [TK]D-Fender | Psychobilly, that is NOT an extension. An extension is a number you dial. and [] is the username |
| 18:03.36 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 18:04.15 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 18:04.17 | Psychobilly | oh i see, anyway in my configs i use the same username and dial number, is there a way to have 2 dirrerent numbers for the same username? |
| 18:05.58 | [TK]D-Fender | Psychobilly, you are still throwing "number" around carelessly. What are you really trying to do with these account(s)? |
| 18:09.37 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:09.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 18:09.39 | *** join/#asterisk cjk (n=cjk@d90-129-18-139.cust.tele2.lu) |
| 18:11.00 | cjk | hi, when i do core show hints i only see unavailable and idle even though my phone is ringing? |
| 18:11.21 | cjk | any idea? |
| 18:13.48 | ManxPower | Psychobilly: start out by not making device userids to be numbers. |
| 18:13.56 | ManxPower | We use the MAC address as the SIP userif. |
| 18:14.04 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 18:14.36 | ManxPower | this will help you realize that making the SIP userID a number and an extension be the same number is generally a bad idea, |
| 18:14.42 | [TK]D-Fender | ManxPower, I'm starting to think he jsut wants 2 extens in the dialplan do dial the same sip DEVICE.... |
| 18:15.08 | ManxPower | [TK]D-Fender: I blame it on the schools. They don't teach people how to ask a technical question! |
| 18:15.34 | Psychobilly | ok my english sucks i admit it :P |
| 18:15.37 | [TK]D-Fender | ManxPower, for him, s/marklar/number/ |
| 18:15.57 | De_Mon | Psychobilly its not your english its your termonology userid is not a number. you just made it one |
| 18:16.00 | ManxPower | Psychobilly: non-english speakers get some wiggle room |
| 18:17.03 | Psychobilly | wiggle? |
| 18:17.06 | [TK]D-Fender | Psychobilly, Your english is fine, your lack of clarity and abuse of the word "number" as though it has 1 single meaning isn't, |
| 18:17.25 | [TK]D-Fender | Psychobilly, So you want to be able to DIAL 2 different #'s on a phone to ring the same SIP user? |
| 18:19.20 | Psychobilly | [TK]D-Fender i know how to do this in extensions.conf, i was aksing if it is possible in sip.conf to assing 2 different phone muners in a user, but i guess i got it all wrong |
| 18:20.03 | [TK]D-Fender | Psychobilly, aer you talking about the CALLER-ID number people see when you call them? |
| 18:20.09 | blitzrage | it seems one of the hardest things for people to get around is that extension numbers don't need to (and shouldn't) be mapped directly to a device. If you abstract the numbering scheme from the device scheme, and then you abstract the users from both of those, then you end up with much greater flexibility |
| 18:20.22 | ManxPower | Psychobilly: sip.conf does not have the concept of "extension" or "number" |
| 18:20.30 | Psychobilly | ok i got it now |
| 18:20.38 | ManxPower | blitzrage: that and contexts |
| 18:20.47 | blitzrage | ManxPower: indeed |
| 18:21.16 | blitzrage | contexts seem to be a hard thing to "get" until someone understands it enough to visualize them in their head |
| 18:21.48 | blitzrage | possibly a good thing |
| 18:21.49 | Psychobilly | its what blitzrage said, i was using extension numbers as usernmanes in sip.conf, and i got confused |
| 18:21.52 | *** join/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 18:21.53 | [TK]D-Fender | Psychobilly, So got something you can clarify for us? And don't use the word "number" in your answer! |
| 18:22.03 | yassine | good evening everyone |
| 18:22.07 | blitzrage | morning! |
| 18:22.09 | Psychobilly | heh |
| 18:22.45 | [TK]D-Fender | Psychobilly, exten => 123,1,Dial(SIP/400) <- dial 123 to ring SIP device [400] |
| 18:22.55 | Psychobilly | [TK]D-Fender i htik i got the answer i was looking for, i was wrong in the first place about the use of sip.conf |
| 18:22.55 | [TK]D-Fender | Psychobilly, wash-rinse-repeat |
| 18:22.56 | yassine | i'm doing my first steps with asterisk and would like know where i can turn debugging to see what is happening when a sip user try to login |
| 18:23.12 | [TK]D-Fender | yassine, "sip debug" |
| 18:23.15 | yassine | and suggestions or reading are welcome! |
| 18:23.24 | blitzrage | jbot: tell yassine about book |
| 18:23.25 | yassine | [TK]D-Fender: thanks |
| 18:23.32 | [TK]D-Fender | yassine, "sip debug peer [IP or device in sip.conf]" |
| 18:23.41 | [TK]D-Fender | yassine, and "sip no debug" to stop. |
| 18:25.19 | yassine | okay thanks [TK]D-Fender |
| 18:40.25 | *** join/#asterisk reallost1 (n=reallost@72.169.24.231) |
| 18:41.06 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:42.52 | riddlebox | is this correct exten => _NXXXXXXX,1,Dial(Zap/1/1{EXTEN})? |
| 18:43.02 | riddlebox | if I wanted to insert a 1 before the other 10 digits? |
| 18:43.11 | reallost1 | looks right |
| 18:43.39 | riddlebox | ok cool thanks |
| 18:44.15 | riddlebox | my sip phones have call logs, which I can dial from but they only have 10 digits in them |
| 18:44.30 | [TK]D-Fender | riddlebox, Yes... if your exten actually TOOK 10 digits :) |
| 18:44.46 | riddlebox | ahh crap |
| 18:44.51 | reallost1 | you only have 8 digits |
| 18:45.31 | [TK]D-Fender | riddlebox, for NANPA you'd want - exten => _NXXNXXXXXX,1,Dial(ZAP/1/1${EXTEN}) |
| 18:45.38 | riddlebox | for some reason I was thinking _N counted as two I guess |
| 18:46.29 | [TK]D-Fender | riddlebox, Bringing you up to *9*. Strike two :p |
| 18:46.42 | riddlebox | dang it |
| 18:46.52 | riddlebox | one more and I am out of questions for the day right |
| 18:46.53 | [TK]D-Fender | ~cluebat riddlebox |
| 18:46.53 | jbot | ACTION pulls out a ClueBat (tm) and thwaps riddlebox. |
| 18:48.49 | reallost1 | I have an interesting situation where I've got 1 sip trunk sending 2 sets of numbers that have need different DTMF modes. |
| 18:49.36 | [TK]D-Fender | reallost1, Make 2 different peer entries for that account then |
| 18:50.08 | reallost1 | That is why I tried, I even have them using different passwords. |
| 18:50.25 | riddlebox | [TK]D-Fender, it is not working for some reason, and I am using verbose 9 and I dont even see the call try to go out |
| 18:51.15 | [TK]D-Fender | riddlebox, pastebin your dialplan fast |
| 18:51.25 | [TK]D-Fender | ~pb |
| 18:51.25 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:53.13 | reallost1 | Fender, this is what I have in my sip.conf http://pastebin.com/d1efc1640 |
| 18:53.52 | riddlebox | [TK]D-Fender, http://pastebin.ca/809283 |
| 18:54.24 | [TK]D-Fender | reallost1, those are 2 differnt usernames |
| 18:54.56 | [TK]D-Fender | reallost1, and your 2nd doesn't have a "host" line |
| 18:55.31 | reallost1 | [TK]D-Fender, oops, I clipped the host=dynamic on the 2nd one. |
| 18:55.44 | [TK]D-Fender | reallost1, And those are users, meaning incoming. Does that mean the people sending you calls withh be choosing equally different peers on their side? |
| 18:56.02 | reallost1 | yes |
| 18:57.02 | reallost1 | its the same host ip, but we are trying to get 2 different DTMF types depending on the originating set of numbers. |
| 18:57.05 | reallost1 | its incoming only. |
| 18:57.31 | [TK]D-Fender | reallost1, What set of numbers? What exactly is sending you those calls? |
| 18:57.58 | [TK]D-Fender | riddlebox, exten => _NXXNXXXXXXX,1,Dial(Zap/1/1${EXTEN}) <-- you still can't count |
| 18:57.59 | reallost1 | I'm running asterisk 1.4.15 and they are running asterisk 1.2.x |
| 18:58.12 | [TK]D-Fender | riddlebox, notice its the sale LENGTH as your 11-digit dial |
| 18:58.28 | [TK]D-Fender | reallost1, and they are using 2 different peers on their end? |
| 18:58.33 | reallost1 | yes |
| 18:58.49 | [TK]D-Fender | reallost1, basically they should just 1 1 format and * will convert them over anyways... |
| 18:59.21 | *** join/#asterisk z00m00 (n=Miranda@myhome.oplot.com) |
| 18:59.31 | z00m00 | Hello everyone! |
| 19:00.01 | reallost1 | Hmm... That is what we tried initally. But I had to specify some of the numbers as inband or dtmf wouldn't be recognized. |
| 19:00.10 | reallost1 | but that broke the rest of the numbers. |
| 19:00.25 | reallost1 | I even tried auto and info, but no dice. |
| 19:00.42 | [TK]D-Fender | ok, I've gotta jet, back later-ish |
| 19:00.48 | reallost1 | thanks |
| 19:01.41 | z00m00 | Who can help me with research. I need get a statistic from Asterisk. When I can get a help with this question ? |
| 19:02.09 | [TK]D-Fender | MurderCount++ |
| 19:02.32 | z00m00 | MurderCount-- |
| 19:04.08 | z00m00 | Чем умничать лучше бы помог |
| 19:04.13 | z00m00 | сучара |
| 19:04.40 | riddlebox | whoa my dogs have settled down, after 4 hours of non stop wrestling |
| 19:07.48 | cjk | why does pickupchan exists when there is no call? couldnt it go to another priority and continue a diall process? |
| 19:11.49 | *** join/#asterisk bjweeks (n=bjweeks@unaffiliated/bjweeks) |
| 19:12.04 | yassine | is zap show channels command supported in asterisk 1.4.13 ? |
| 19:12.41 | styelz | yes |
| 19:12.48 | yassine | i get no such command when i try it, but since this is self compiled install im not sure if its my fault or the command is no more supported |
| 19:13.20 | styelz | got chan_zap loaded? |
| 19:13.20 | yassine | styelz: are there any alternatives for it? |
| 19:13.34 | styelz | not that i know of |
| 19:14.28 | styelz | zap command wont be there if you dont have zaptel drivers installed |
| 19:14.36 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 19:14.36 | styelz | maybe |
| 19:15.40 | yassine | styelz: i compile zaptel package and installed it, but did not notified that i need the insmod chan_zap |
| 19:17.26 | nestAr | yassine: yes, it's in the documentation |
| 19:17.48 | styelz | sounds like you need to compile asterisk with chan_zap |
| 19:17.54 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 19:18.03 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 19:26.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:28.28 | *** join/#asterisk saftsack (n=saftsack@193.218.17.99) |
| 19:33.04 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 19:33.04 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:48.18 | [TK]D-Fender | yassine, If you'd just compiled zaptel you have to REDO asterisk afterwards so it picke it up |
| 19:49.07 | yassine | [TK]D-Fender: I#m actuall about to recompile everything and this time ( i made sure that i have invoked the make menuselect) |
| 19:50.26 | yassine | [TK]D-Fender: sorry i did not get what you mean by REDO ? (recompile) ? |
| 19:57.13 | styelz | he means, recompile asterisk, after setting up zaptel |
| 19:59.14 | yassine | styelz: im about to right can i have differnt language support ? |
| 19:59.34 | yassine | should i select that now or i can do this later (after the compiling) |
| 20:00.38 | *** join/#asterisk rabelais (n=blank@hpolaris.Stanford.EDU) |
| 20:01.00 | styelz | not sure what you mean |
| 20:01.12 | styelz | the audio files ? |
| 20:01.15 | yassine | yes |
| 20:01.31 | styelz | you can do that after. i dont know whats out there though |
| 20:01.47 | [TK]D-Fender | yassine, Follow this order : libpri, zaptel, Asterisk, Addons |
| 20:02.05 | yassine | omg i have to redo |
| 20:02.14 | yassine | i have started with zaptel |
| 20:02.48 | styelz | yassine: check out http://www.voip-info.org/wiki/view/Asterisk+sound+files+international maybe |
| 20:03.27 | yassine | [TK]D-Fender: in the oreilly book suggeted by blitzrage they started with zaptel |
| 20:04.50 | yassine | styelz: thanks that will make things a lil bit clearer |
| 20:07.09 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:07.54 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 20:08.17 | JayTee52 | is anyone here running * on 64 bit linux? |
| 20:09.11 | yassine | [TK]D-Fender: do you think i should restart the whole process or i can go with the order suggetd in the book : http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:11.10 | blitzrage | JayTee52: yep |
| 20:11.36 | [TK]D-Fender | yassine, restart it all, what the heck, 5 minutes well spent |
| 20:11.46 | JayTee52 | I'm going to be using Digium's TDM04B cards and a TE210P card. Do you see any problems there? |
| 20:12.15 | yassine | [TK]D-Fender: okay thanks (restarting) if you mind i would really like to understand why? |
| 20:12.19 | JayTee52 | I've read a few posts about 64 bit causing problems with the zaptel driver timing |
| 20:13.54 | JayTee52 | I've been thrust into a nightmare scenario (for me anyways) with a Dell PowerEdge 2950 that has an ATI Radeon graphics chip that acts flaky running RHEL 5 64bit. I wanted 32 bit but they shipped it with 64 installed. |
| 20:14.12 | JayTee52 | and they don't seem to have as much driver support for 32 bit devices. |
| 20:14.18 | blitzrage | ok... |
| 20:15.00 | blitzrage | I just installed CentOS 5 32bit on my 2950s |
| 20:15.22 | JayTee52 | do they have the ATI graphics chip or the older Intel chips? |
| 20:15.27 | blitzrage | no idea |
| 20:15.36 | blitzrage | box is probably almost a year old now |
| 20:15.51 | JayTee52 | they're probably the Intel 915 chipset then |
| 20:16.29 | JayTee52 | I had a devil of a time getting a res higher than 800x600 on RHEL. Odd that RHEL 5 loads Gnome desktop by default. |
| 20:16.46 | blitzrage | why the heck are you running X windows on an Asterisk box? |
| 20:16.54 | JayTee52 | Ubuntu Server doesn't use a GUI, does CentOS 5? |
| 20:16.57 | blitzrage | I figure it's a server -- it should never have a GUI |
| 20:17.07 | blitzrage | you just said you were changing your screen resolution |
| 20:17.07 | JayTee52 | blitzrage, I tend to agree |
| 20:17.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:17.32 | JayTee52 | yes, I was. Because this OS came preinstalled and came with the Gnome desktop installed. |
| 20:17.39 | blitzrage | JayTee52: ya... by default -- just unselect the button that installs it |
| 20:17.56 | blitzrage | anything preinstalled always gets wiped by me |
| 20:18.06 | blitzrage | first thing I do when I get a new computer is format the HD |
| 20:18.39 | ManxPower | The problems with graphics on an Asterisk server is that graphics drivers lock interrupts for so long you are likely to miss data coming in from Zaptel and you might cause excessive jitter on VoIP connections. |
| 20:18.56 | JayTee52 | I may just scrub the Red Hat install and use CentOS 5 if I can get the PERC RAID controller to work with the SAS drives so I can have RAID 1 at the hardware level. |
| 20:19.40 | JayTee52 | If I'm gonna use a gui with * I'd prefer it be remote web based, not on the host itself. |
| 20:19.50 | blitzrage | right |
| 20:20.05 | blitzrage | CentOS 5 --> minimal install (everything unchecked) |
| 20:20.10 | blitzrage | yum install <apps_I_need> |
| 20:20.17 | blitzrage | svn co asterisk |
| 20:20.20 | blitzrage | install |
| 20:20.24 | ManxPower | JayTee52: you realize here that we feed GUI people to alligators with a taste for GUI people, right? |
| 20:20.39 | blitzrage | ya... we're hard core :) |
| 20:20.51 | JayTee52 | I have a good walkthrough for * using CentOS. It says to do libpri first, then zaptel, then asterisk. Does that sound right to you? |
| 20:21.02 | blitzrage | people in here are like a gamer clan |
| 20:21.10 | blitzrage | yep |
| 20:22.01 | JayTee52 | I've already got * running on an Ubuntu 6.06 server with no gui. I have no problem with it but my retarded "Windows only" boss hates it. He just loves the pricetag of * |
| 20:23.07 | ManxPower | JayTee52: jer |
| 20:23.18 | JayTee52 | and I'm not a distro zealot. I've installed a VM appliance that uses CentOS 5 and sipX and it's fine. |
| 20:23.22 | ManxPower | JayTee52: Then he will hate any GUI unless it works just like Windows. |
| 20:24.00 | ManxPower | Happy Yule, blitzrage |
| 20:24.15 | JayTee52 | ManxPower, yep, he's a wicked MS bigot. Anything else is heresy. He's so stupid he thinks you should use Dr Watson to debug problems in XP or Win2K3. |
| 20:24.31 | JayTee52 | talk about a dinosaur |
| 20:24.54 | blitzrage | ManxPower: danke! |
| 20:25.21 | JayTee52 | Merry Xmas, blitzrage and thanks! |
| 20:25.48 | ManxPower | Asterisk needs to come with a fake GUI. It would look great, but not actually DO anything. Call it "beta" if users complain. |
| 20:25.56 | JayTee52 | hahaha |
| 20:26.36 | JayTee52 | "I tried to think but nothing happened." - Curly's voice from the 3 Stooges. |
| 20:33.10 | *** join/#asterisk nirz (n=nir@bzq-79-179-99-65.red.bezeqint.net) |
| 20:45.44 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 20:47.09 | esaym | Can a person make a receive voip calls from random wifi hotspots with the wireless voip phones like the UTStarcom F1000G? Or can phones like this only be used on configured subnets or local lans? |
| 20:51.02 | [TK]D-Fender | esaym, Depending on how the networking at that spot works it could be fine |
| 21:02.26 | *** join/#asterisk PDani (n=pdani@dsl51B6FDAC.fixip.t-online.hu) |
| 21:02.28 | PDani | hi |
| 21:03.01 | PDani | approximately when will it be possible to use SIP instant messiging with asterisk? |
| 21:06.13 | [TK]D-Fender | PDani, Feel free to code it. If you don't see it in the tracker.... |
| 21:09.21 | *** join/#asterisk RoyK (n=roy@ip-17-23-149-91.dialup.ice.no) |
| 21:10.05 | rabelais | I have a call quality problem that is specific to my sipura ata going through my asterisk server, call quality with softphones is great, and call quality directly to the sipura device (IP to IP calls) is great....the bad call quality through my sipura device has very very jittery and broken outgoing audio....how would I fix this? |
| 21:10.53 | ManxPower | rabelais: change the rtp packet size on the SIPura to 0.20 instead of 0.30 |
| 21:11.40 | rabelais | ok, I will try that, thank you |
| 21:12.08 | rabelais | do you mean from 0.030 to 0.020? |
| 21:12.20 | rabelais | currently it is set to 0.030 |
| 21:12.37 | ManxPower | correct. |
| 21:12.45 | ManxPower | Asterisk does not work with 30ms packet sizes. |
| 21:13.00 | rabelais | wow |
| 21:13.23 | rabelais | I have had it wrong all this time? |
| 21:13.41 | ManxPower | rabelais: virtually everything out there except for modern SIPura devices default to 20ms audio packets |
| 21:14.02 | rabelais | I've been using it that way for five/six years |
| 21:14.03 | rabelais | sigh |
| 21:14.24 | ManxPower | rabelais: that would be the cause of the audio problems when audio is routed thru Asterisk. |
| 21:14.43 | rabelais | ManxPower, thank you |
| 21:14.59 | rabelais | it totally works |
| 21:15.01 | rabelais | hehe |
| 21:15.11 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 21:15.16 | hrmphh | any hylafax/faxmail gurus here? |
| 21:15.17 | yassine | asterisk is running (asterisk 29993 0.3 0.8 22276 8644 pts/2 Sl 22:10 0:00 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 21:15.18 | yassine | <PROTECTED> |
| 21:16.49 | ManxPower | yassine: Did you build from source? |
| 21:17.33 | rabelais | ManxPower, thank you again...it just works! they're not complaining anymore, hehe |
| 21:17.36 | yassine | ManxPower: yes |
| 21:18.52 | yassine | ManxPower: any idea? |
| 21:19.06 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 21:21.17 | [TK]D-Fender | yassine, make sure that "asterisk" has rigths to create the PID file in your /var/run folder <---- |
| 21:21.41 | yassine | okay let me try again |
| 21:22.32 | yassine | [TK]D-Fender: can i configure asterisk to write its asterisk.ctl in /var/run/asterisk/ directory? |
| 21:22.45 | yassine | i have a directory there that i have chown to asterisk:asterisk |
| 21:23.37 | yassine | okay done |
| 21:23.39 | yassine | thanks |
| 21:29.04 | *** join/#asterisk dlynes (n=dlynes@s209-121-50-177.bc.hsia.telus.net) |
| 21:29.30 | dlynes | Asterisk has a function called 'volume' now? |
| 21:29.39 | dlynes | I'm just curious which version of asterisk it's in? |
| 21:29.53 | dlynes | Is it in trunk? |
| 21:30.10 | [TK]D-Fender | yassine, All god now? |
| 21:30.22 | [TK]D-Fender | dlynes, Yes. |
| 21:30.26 | [TK]D-Fender | good* |
| 21:31.39 | dlynes | [TK]D-Fender: do you happen to know if it's droppable into 1.4? |
| 21:34.00 | [TK]D-Fender | dlynes, IIRC 1.6's revamped channels allow you to splice calls up 100x better that we can now and insert DSP-like processes into streams, etc. Volume is a very natural extension of this. And no, there is no way I can' picture this being backported |
| 21:35.00 | dlynes | [TK]D-Fender: ah |
| 21:35.21 | dlynes | [TK]D-Fender: and i guess it's not going to be making it to release stage any time soon either, from the looks of russell's blog... |
| 21:35.35 | [TK]D-Fender | dlynes, link plz |
| 21:36.34 | dlynes | [TK]D-Fender: http://russellbryant.net/blog/?p=19 and http://russellbryant.net/blog/?p=15 |
| 21:41.45 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 21:43.06 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 21:55.24 | jer | 1.6 sounds like it'll be fun |
| 21:55.31 | jer | so... when is it going to be released? =] |
| 21:57.07 | *** join/#asterisk duki (n=duki@85.27.58.159) |
| 21:59.02 | mvanbaak | *burp* |
| 21:59.05 | mvanbaak | oops |
| 22:03.40 | *** join/#asterisk etfonhomey (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
| 22:06.46 | dlynes | jer: if i had to venture a guess, I'd figure on 2009 |
| 22:07.15 | De_Mon | dlynes nahh it'll be in 2008 for sure |
| 22:07.34 | dlynes | De_Mon: why do you say that? |
| 22:08.15 | De_Mon | dlynes because 2009 is too far away and I refuse to accept that answer? |
| 22:08.21 | dlynes | heh |
| 22:08.39 | dlynes | well, russell is wanting more testing going into it before it gets released this time around |
| 22:08.51 | dlynes | which is why i'm suspecting 2009, not 2008 |
| 22:09.06 | De_Mon | has 1.6 gone into feature freze yet? |
| 22:09.14 | dlynes | nope |
| 22:09.21 | De_Mon | oh damn |
| 22:10.09 | De_Mon | I backported app_bridge and have func_odbc, so I think I can live... |
| 22:17.21 | mvanbaak | meh, my neighbour is here |
| 22:17.35 | mvanbaak | normally I listen to Cradle of Filth and stuff like that |
| 22:18.06 | mvanbaak | right now I have 'Faithless - Insomnia (Moody Mix)' on full power |
| 22:18.07 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:18.20 | mvanbaak | whopping 1200 watt blowing away my walls |
| 22:18.42 | mvanbaak | neighbour and wife totally loosing their mind |
| 22:18.45 | mvanbaak | ugh |
| 22:19.13 | mvanbaak | freaking song is 11 minutes |
| 22:21.47 | *** join/#asterisk cy3o3 (n=cy@is.trapped.in.themetaverse.org) |
| 22:21.49 | cy3o3 | fappity |
| 22:22.30 | yassine | is there a way to check an AMI user manaually via telnet ? |
| 22:22.30 | Nugget | telnet is eeeeeeevil! |
| 22:26.07 | *** join/#asterisk fukz (n=basti@p5B063F20.dip.t-dialin.net) |
| 22:32.01 | dlynes | telnet |
| 22:36.21 | *** join/#asterisk Darthclue (n=root@li13-84.members.linode.com) |
| 22:36.42 | Darthclue | hey everyone, hows it going? quick question... |
| 22:37.07 | Darthclue | when a call is packet2packet bridged does it ever return to the dialplan? if not, is it tracked via any type of cdr? |
| 22:37.32 | *** join/#asterisk hohum_ (n=dcorbe@wsip-70-166-81-42.sd.sd.cox.net) |
| 22:53.19 | mvanbaak | aaaaaaaaaaaah |
| 22:53.23 | mvanbaak | finally |
| 22:53.28 | mvanbaak | cradle of filth |
| 22:59.20 | mvanbaak | anyone here ? |
| 22:59.27 | mvanbaak | or is it weekend++ |
| 23:00.56 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 23:02.59 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 23:03.44 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 23:05.31 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 23:09.45 | *** join/#asterisk MindTheGap (n=MindTheG@BHE201062172083.res-com.wayinternet.com.br) |
| 23:38.21 | *** join/#asterisk RoyK (n=roy@ip-146-11-149-91.dialup.ice.no) |
| 23:39.43 | JunK-Y | yassine: yes, just login thru AMI with this user? |
| 23:39.49 | JunK-Y | (via telnet) |
| 23:45.52 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 23:50.20 | yassine | JunK-Y: i found it that it was my configuration that was missing the right secret value using this: |
| 23:50.40 | yassine | http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Login |