| 00:00.09 | rasterix | mostly early asterisk |
| 00:00.16 | jplank | what are you trying to find? |
| 00:00.25 | nhuisman_work | rasterix, I dunno I used the book and figured it out. |
| 00:00.28 | rasterix | i want complex extensions.conf |
| 00:00.40 | rasterix | with real world loose syntax |
| 00:00.41 | nhuisman_work | i've seen some of those on google |
| 00:00.42 | rasterix | etc |
| 00:00.45 | [TK]D-Fender | rasterix: Go look at any FreePBX install :) |
| 00:00.49 | jplank | ROFL!!! |
| 00:00.53 | nhuisman_work | with like 10 sip trunks |
| 00:00.54 | rasterix | fender: :P |
| 00:01.00 | nhuisman_work | and all kinds of rules for each area code |
| 00:01.14 | jplank | I was going to say trixbox if you really want worthless complex extension.conf |
| 00:01.15 | nhuisman_work | i guess they aren't "trunks" |
| 00:01.24 | [TK]D-Fender | rasterix: If you can't even think up a dialplan what are you doing thinking you're qualified to write a GUI to MANAGE one? |
| 00:01.42 | rasterix | fender: i CAN think up a dialplan |
| 00:01.45 | nhuisman_work | is that what he wants to do? |
| 00:01.48 | [TK]D-Fender | rasterix: Then get to it |
| 00:02.08 | rasterix | but i right them according to how i think extensions.conf should be set up |
| 00:02.28 | rasterix | so strangely my editor is designed for them |
| 00:02.39 | rasterix | write* |
| 00:02.44 | rasterix | damn it |
| 00:02.58 | *** join/#asterisk Levonk (n=lk@adsl-76-246-208-119.dsl.lsan03.sbcglobal.net) |
| 00:03.43 | jplank | writing simple gui's to manage extensions.conf isn't really that hard, if you have a specific purpose in mind |
| 00:03.47 | rasterix | fender: i am NOT writing a gui |
| 00:03.59 | rasterix | im not writing a gui |
| 00:04.00 | rasterix | >< |
| 00:04.03 | jplank | lol |
| 00:04.14 | nhuisman_work | rasterix, so what _are_ you doing :P |
| 00:04.23 | rasterix | *collapses* |
| 00:04.26 | nhuisman_work | hehe |
| 00:05.06 | rasterix | i am wasting my time writing an editor for the conf files |
| 00:05.15 | rasterix | but its NOT gui |
| 00:05.16 | nhuisman_work | rasterix, vi... |
| 00:05.22 | jplank | nano |
| 00:05.27 | rasterix | lol thanks guys |
| 00:05.39 | nhuisman_work | jplank, NANO!?!?!? |
| 00:05.43 | jplank | lol |
| 00:05.43 | rasterix | im an idiot... BUT im not so dumb im not aware of vi and nano |
| 00:06.00 | nhuisman_work | we know ;) |
| 00:06.15 | jplank | hey now, I know a lot of people who use nano |
| 00:06.24 | jplank | hell of a lot easier to remember the commands |
| 00:06.36 | nhuisman_work | I know a lot of people who use windows too, your point :P |
| 00:06.45 | jplank | oh, windows, notepad++ |
| 00:06.52 | rasterix | < choses to assume nhuisman meant he knows im not so dumb... not that im an idiot |
| 00:07.02 | rasterix | chooses* |
| 00:07.05 | *** join/#asterisk irieKEN (n=irieken@130.191.205.128) |
| 00:07.11 | jplank | no I think nhuisman_work is now calling me the idiot |
| 00:07.11 | nhuisman_work | yeah |
| 00:07.15 | jplank | :) |
| 00:07.23 | nhuisman_work | jplank, no, I think nano is a lame editor |
| 00:07.29 | nhuisman_work | jplank, that's all |
| 00:07.39 | jplank | its not as powerful as vi, I agree |
| 00:07.44 | jplank | but for a novice, whats the harm? |
| 00:07.49 | rasterix | i love vim so i thought id rewrite it for windows |
| 00:07.55 | rasterix | and limit it to asterisk |
| 00:07.57 | nhuisman_work | rasterix, it exists in windows... |
| 00:07.58 | rasterix | XD |
| 00:07.59 | jplank | I usually recommend nano to new users |
| 00:08.08 | rasterix | so does notepad++ |
| 00:08.18 | nhuisman_work | jplank, yeah it's not quite as ubiquitous as vi |
| 00:08.40 | jplank | I've been using vi for a while, and I still keep a cheat sheet handy |
| 00:08.51 | nhuisman_work | jplank, yeah usually I use very little of what it can really do. |
| 00:09.00 | rasterix | basically i dont understand why people are getting so raged... if i wanna waste my time writing a windows editor why people picking on me :( |
| 00:09.06 | nhuisman_work | hehe |
| 00:09.11 | nhuisman_work | because that's what bored people do. |
| 00:09.13 | jplank | yea |
| 00:09.20 | jplank | but I guess thats your same excuse |
| 00:09.30 | rasterix | exactly! |
| 00:09.32 | jplank | lol |
| 00:10.11 | jplank | are you really writing a asterisk file editor for windows? |
| 00:10.14 | Qwell | rasterix: you realize that once the file is transferred over to the linux box...it'll have to be edited...right? |
| 00:10.15 | nhuisman_work | i guess it makes more sense to me to write a linux or cross platform editor for something that runs on linux |
| 00:10.27 | rasterix | uhoh... this sounds bad |
| 00:10.34 | rasterix | qwell: why? |
| 00:10.38 | rasterix | line terminators? |
| 00:10.43 | rasterix | i have that covered |
| 00:10.50 | nhuisman_work | or just the fact that you have to upload it everytime |
| 00:10.53 | nhuisman_work | you make any changes at all |
| 00:10.59 | jplank | if he's writing it from scratch, he can get around those obstacles |
| 00:11.00 | jaytee | ~101 |
| 00:11.00 | jbot | it has been said that 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 00:11.06 | mchou | Qwell: nah, it wont need to be edited. vi set ff=unix and save :) |
| 00:11.18 | nhuisman_work | if it had a built in uploader |
| 00:11.21 | rasterix | mchou: grrrrr |
| 00:11.21 | nhuisman_work | it might be neat. |
| 00:11.36 | Qwell | mchou: only newbs use vi ;) |
| 00:11.38 | rasterix | look ive started writing it |
| 00:11.46 | rasterix | i dont stop once ive started |
| 00:11.54 | nhuisman_work | that's not what she said |
| 00:12.04 | [TK]D-Fender | rasterix: Yes, we've heard your speeches... |
| 00:12.14 | jplank | wouldn't it be easier as a plugin into notepad++? |
| 00:12.23 | rasterix | fender: and we have hear your whining... move on |
| 00:12.24 | rasterix | :P |
| 00:13.05 | mchou | ok guys, all joking aside, what exactly is the difference between username and fromuser? |
| 00:13.23 | mchou | I need an explanation that makes sense |
| 00:14.05 | rasterix | the point is nobody apart from me has to use my editor (and probably wont) ... |
| 00:14.10 | mchou | all the asterisk docs I've read gloss over this |
| 00:14.17 | rasterix | i still use vim lol |
| 00:14.58 | JT | mchou: one is for peers, one is for users, iirc |
| 00:15.25 | jplank | yea, at least thats what voipinfo makes it sound like |
| 00:15.36 | Unlockgod | hey, im tryinto get the right kernel source fo OpenVZ installed, i run yum install openvz-kernel-rhel5 |
| 00:15.37 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 00:15.37 | rasterix | anyway 1am... sleep time... |
| 00:15.48 | Unlockgod | but it gimes me a differnt version |
| 00:15.53 | nhuisman_work | oh you are running it on a wireless ap? |
| 00:15.57 | jplank | rasterix: are you going to be demoing your gui at astircon? |
| 00:16.09 | jplank | :) |
| 00:16.12 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 00:16.15 | mchou | JT: which one should be used for friends? |
| 00:16.23 | rasterix | jplank: lol its never going to be THAT good |
| 00:16.34 | mchou | JT: or should bothe be used for friends? |
| 00:16.40 | mchou | both* |
| 00:16.44 | jplank | come on, if trixbox is going to have an exhibit, so can you |
| 00:16.53 | rasterix | jplank: and its NOT gui!!!!!!!! <smashes head against wall> |
| 00:17.22 | nhuisman_work | it's a tui |
| 00:17.23 | nhuisman_work | ;) |
| 00:17.35 | rasterix | haha i dont know why i come in here |
| 00:17.38 | rasterix | its like |
| 00:17.53 | nhuisman_work | *inserts words into rasterix's mouth* |
| 00:18.08 | jplank | wait, your writing a application for windows to edit asterisk files? |
| 00:18.12 | rasterix | < avoids the abvious retort to nhuisman |
| 00:18.24 | jplank | isn't that like the definition for gui? |
| 00:18.46 | rasterix | jplank: is it? i thought gui meant GRAPHICAL user interface |
| 00:18.57 | jplank | is it going to be CLI? |
| 00:19.01 | nhuisman_work | yeah but in windows everything is a gui :P |
| 00:19.10 | jplank | my point ----^ |
| 00:19.19 | rasterix | < takes knife and looks at his wrists |
| 00:19.46 | nhuisman_work | rasterix, vertical cuts along the veins... |
| 00:19.51 | jplank | you really want to turn some heads |
| 00:19.52 | rasterix | i know |
| 00:19.57 | rasterix | and get in the bath first |
| 00:20.02 | rasterix | so it doesnt clot |
| 00:20.05 | jplank | write a CLI editor for windows to edit asterisk config files |
| 00:20.06 | TJNII | Down the road, not across the street. Make it count. |
| 00:21.53 | [TK]D-Fender | ~emo |
| 00:21.53 | jbot | /wrists |
| 00:22.07 | nhuisman_work | if I"m never going to have more then one outgoing trunk, is there any reason to have separate contexts for local outgoing and long distance/international outgoing? |
| 00:22.15 | nhuisman_work | [TK]D-Fender, lol |
| 00:22.22 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 00:22.25 | nhuisman_work | clever |
| 00:22.38 | [TK]D-Fender | nhuisman_work: Yes, so you can prevent people for running up your LD costs who have no business dialing LD |
| 00:24.38 | nhuisman_work | [TK]D-Fender, yeah, I don't really see us ever making some phones not work with LD |
| 00:24.46 | nhuisman_work | university staff mostly |
| 00:25.42 | [TK]D-Fender | nhuisman_work: Either way a little separation is a good thing even if no grouping context exclude all the capabilities. It prepares your infrastructure for growth. |
| 00:25.46 | JT | mchou: yeah usually both for friends seems to work |
| 00:25.55 | nhuisman_work | [TK]D-Fender, ya |
| 00:27.01 | nhuisman_work | hmm actually I remember why I had 11 digit dialing in the local outbound |
| 00:27.24 | nhuisman_work | we need to dial 1808xxxxxxxx to get to the other island, but I should put the 808 in there for that rule or it matches everything |
| 00:27.41 | nhuisman_work | as you pointed out |
| 00:29.20 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.20.90) |
| 00:30.58 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 00:31.01 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 00:34.08 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:37.17 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 00:37.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:37.37 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7de723b58cf97cb7) |
| 00:39.32 | *** join/#asterisk UD (n=steve@unaffiliated/underdawg) |
| 00:40.51 | UD | a long time ago i remember a free service that would actually give you a real number on the PSTN for incoming calls. I forget what it was called. Does anyone know anything like this? |
| 00:41.24 | [TK]D-Fender | UD: www.ipkall.com |
| 00:41.30 | UD | yes that was it |
| 00:41.31 | UD | thanks |
| 00:41.58 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 00:43.13 | sergey | Hi. How have info about agent (from queue) who transfer the call ? |
| 00:43.51 | *** join/#asterisk russellb_ (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:43.51 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 00:44.55 | nhuisman_work | starting to get the hang of this gui ;) |
| 00:45.15 | Nugget | can't take that russell character anywhere |
| 00:45.21 | nhuisman_work | lol |
| 00:45.30 | russellb_ | o.O |
| 00:46.19 | outtolunc | hands russellb_ a sharp stick |
| 00:46.36 | *** kick/#asterisk [Guest14104!n=russellb@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 00:49.17 | *** join/#asterisk communique (n=Jordan@sce-178.norlight.net) |
| 00:51.30 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 00:54.33 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:55.34 | nhuisman_work | hmm definitely nicer to delete the sample configs and let the gui create the bare minimum |
| 00:55.46 | nhuisman_work | that was a terrible mess earlier |
| 00:55.47 | communique | has anyone recently had droppped calls on the AEX800 |
| 00:56.36 | communique | I am really having problems with the dell 2970 |
| 00:57.24 | communique | anyone have any suggestions |
| 00:57.24 | Qwell | call support |
| 00:57.31 | *** join/#asterisk tkbeat (n=tk@p54B97723.dip.t-dialin.net) |
| 00:59.02 | communique | been there, done on at least 8 dell 2950, no probs. I am looking for someone who has been there |
| 01:02.55 | *** join/#asterisk MrNaz (n=naz@ppp118-208-176-116.lns10.mel4.internode.on.net) |
| 01:04.15 | UD | heh, what about free outgoing calls? |
| 01:05.45 | UD | [TK]D-Fender: any free outgoing calls to ordinary phone #? |
| 01:06.07 | [TK]D-Fender | UD: nope. |
| 01:06.19 | UD | err i mean do you know of a service that allows this |
| 01:06.37 | *** join/#asterisk dwayne (i=dwayne@76.29.245.9) |
| 01:08.48 | communique | i have great luck with iaxtel |
| 01:08.57 | communique | ten bucks a month |
| 01:09.05 | communique | plenty of free proxies |
| 01:09.12 | communique | pay for what you get |
| 01:13.19 | [TK]D-Fender | Free = 10$/month. News at 11! |
| 01:14.53 | JT | why are people so damn stingey? |
| 01:15.09 | nhuisman_work | anyone here know much about what config files the gui requires to start with? I don't want 68 example files filled with crap |
| 01:15.17 | *** join/#asterisk Levonk (n=lk@adsl-75-62-133-28.dsl.lsan03.sbcglobal.net) |
| 01:15.26 | nhuisman_work | I also asked in asterisknow but I figured i could ask here too in case anyone was lurking. |
| 01:17.50 | UD | i dont mind paying for service i just wanted to test my stuff first |
| 01:17.58 | UD | like without waiting a week |
| 01:21.47 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 01:21.49 | *** join/#asterisk ManxPower (n=manxpowe@187.sub-75-248-47.myvzw.com) |
| 01:22.12 | UD | so with ipkall if i get a certain # can i change it later? |
| 01:24.49 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:27.31 | [TK]D-Fender | UD: "test your stuff"? What does that have to do with an ITSP? |
| 01:33.08 | *** join/#asterisk Levonk (n=lk@adsl-76-246-209-15.dsl.lsan03.sbcglobal.net) |
| 01:50.55 | russellb | communique: clearly you meant something else other than iaxtel ... |
| 01:51.00 | russellb | ... right? |
| 01:54.09 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-247-111-195.dsl.pltn13.sbcglobal.net) |
| 01:56.12 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:56.47 | *** join/#asterisk genji1981 (n=gej@203.152.122.50) |
| 01:58.05 | genji1981 | hiya all! openwrt Kamikaze 7.09 on a Asus Wl-Hdd (125 mhz 4mb flash, 16mb ram) running Asterisk 1.4.something. Fantastic new jitter options.. and everythings fine and smooth. |
| 02:03.21 | genji1981 | okay, i seem to have set it up, so that * gets sent to the call.. not to asterisk, for things like parking calls and transfers. how do i enable such features? additionally.. how can i still get a * sent to a call, with features enabled? |
| 02:03.55 | russellb | configure feature codes in features.conf |
| 02:04.18 | russellb | and then you enable them using options to Dial (core show application Dial), or by setting the DYNAMIC_FEATURES variable (see features.conf.sample) |
| 02:05.34 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 02:06.37 | Yourname | Hi, what's the best and easiest way to implement call forwarding? |
| 02:07.01 | [TK]D-Fender | genji1981: And no decent SIP phone requires you to to DTMF transfers or for parking. |
| 02:07.22 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 02:07.24 | [TK]D-Fender | Yourname: Put a picture of the person they should talk to on your phone and circle it. |
| 02:08.19 | [TK]D-Fender | Yourname: Easiest way = leave it up to the phone. Best depends on what you value and how you want to operate. |
| 02:08.49 | genji1981 | [TK]D-Fender: can you suggest to me, a good free SIP phone, or a good free IAX phone, with transfers and parking, outside of DTMF? |
| 02:09.01 | genji1981 | currently uses SJphone. |
| 02:09.24 | [TK]D-Fender | genji1981: Zoiper. |
| 02:09.29 | genji1981 | searching... |
| 02:09.37 | [TK]D-Fender | genji1981: Ekiga. |
| 02:10.28 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:12.34 | genji1981 | hmm.. zoiper doesn't do attended transfers.. but still nice. |
| 02:12.45 | Yourname | [TK]D-Fender: Can't find the easy way out on the phone. IP 330. Unless you dig deep into the Menu to set it up. |
| 02:13.11 | Yourname | [TK]D-Fender: I wanted it so that the CFWD number is set, and then they hit one button, and it's on.. and the other button and it's off. |
| 02:14.03 | [TK]D-Fender | Yourname: Maybe its a littel more buried on an ip320/330, it isn't on any other model. |
| 02:14.23 | [TK]D-Fender | Yourname: My IP 301, 501, 601 all have a FwD softkey on idle. |
| 02:14.43 | Yourname | Yeah, unfortunately, on the IP330s.. it's not. So sad. :( |
| 02:14.53 | genji1981 | Ekiga doesn't do attended transfers either? |
| 02:15.28 | Yourname | genji1981: Tried eyebeam or xlite? |
| 02:15.45 | genji1981 | hmm.. |
| 02:15.48 | Yourname | [TK]D-Fender: Even activation will not be easy.. |
| 02:16.04 | [TK]D-Fender | Yourname: is on every other model. |
| 02:16.33 | Yourname | [TK]D-Fender: dangit.. that's the reason why I'm looking around for an easy on/off cfwd method. :S |
| 02:16.55 | frogonwheels | Any clues? I'm getting the following error crosscompiling asterisk1.4.21.2 for openwrt |
| 02:16.55 | frogonwheels | -> checking for a sed that does not truncate output... ./configure: line 6396: syntax error near unexpected token `SED,' |
| 02:16.55 | frogonwheels | ./configure: line 6396: ` _AC_PATH_PROG_FEATURE_CHECK(SED, sed gsed,' |
| 02:17.08 | [TK]D-Fender | Yourname: Speed dial entry or maybe remap another key |
| 02:17.54 | Yourname | [TK]D-Fender: That's what I was thinking. Just don't know how to go about it. Wanted to try out both methods actually.. |
| 02:18.30 | [TK]D-Fender | Yourname: Go read the admin guide or play around with your dialplan then. |
| 02:18.55 | Yourname | [TK]D-Fender: There's an admin guide somewhere? |
| 02:19.04 | [TK]D-Fender | Yourname: www.polycom.com |
| 02:19.13 | Yourname | Oh, hah. |
| 02:21.14 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-138-70.neo.res.rr.com) |
| 02:23.26 | Nugget | http://www.besttechie.net/forums/Linux-Humor-t14545.html <-- awesome |
| 02:25.41 | jaytee | hahahaa, I love the Gentoo one! |
| 02:25.47 | *** join/#asterisk Maxous (n=Maxous@76.97.26.56) |
| 02:26.16 | *** part/#asterisk Maxous (n=Maxous@76.97.26.56) |
| 02:26.23 | genji1981 | frogonwheels: hiya! ive got the url for the asterisk 1.4 package, you want it? |
| 02:30.13 | frogonwheels | I've been building it myself. |
| 02:30.50 | frogonwheels | I've got some mods that I"m still trying to get somebody to comit. |
| 02:30.57 | *** join/#asterisk axisys (n=axisys@117.18.224.168) |
| 02:31.08 | *** join/#asterisk Levonk (n=lk@76.237.14.175) |
| 02:31.11 | frogonwheels | ..I think I must have downloaded a new m4 ./configure generator that busted the build |
| 02:31.49 | frogonwheels | I think asterisk configure is using an undocumented macro that has been obsoleted :( |
| 02:32.09 | russellb | asterisk includes the configure script for you |
| 02:32.21 | russellb | you tried to reconfigure it with an incomplete/broken autoconf installation, and broke it |
| 02:32.27 | frogonwheels | russellb: it used to work fine - but suddenly it has broken. |
| 02:32.31 | russellb | so, revert your changes to configure, and try again |
| 02:32.39 | Yourname | Anyway, how do I disable the voicemails that are trying to be sent by email? |
| 02:32.51 | frogonwheels | I didn't make changes to it. |
| 02:32.53 | russellb | Yourname: have you checked voicemail.conf? |
| 02:32.59 | Qwell | it may hosed a couple other files too |
| 02:33.05 | russellb | frogonwheels: you ran "autoconf" or "./bootstrap.sh" which did it ... |
| 02:33.05 | Qwell | may have* |
| 02:33.07 | Yourname | russellb: I'm looking, I don't see a disable type variable anywhere.. |
| 02:33.10 | russellb | you should probably just start over |
| 02:33.25 | russellb | Yourname: it's in there somewhere, i'm too lazy to look |
| 02:33.35 | frogonwheels | russellb: each time openwrt builds it, it starts over. |
| 02:33.40 | Yourname | russellb: Yeah, I found attach=.. but that's set to =no |
| 02:34.31 | Yourname | Uh oh. |
| 02:34.43 | Yourname | Is it sendvoicemail= |
| 02:34.47 | Yourname | err, nah. nvmd |
| 02:36.02 | frogonwheels | ah - I think I've got it. problem is I'm NOT running autoconf and rebuilding it :) |
| 02:37.52 | genji1981 | frogonwheels: if you could try, prehaps see if asterisk-mini could be trimmed down even further.... save cycles somewhere. i can't do conference calls on my wl-hdd.. only one channel at a time. |
| 02:38.12 | frogonwheels | genji1981: is asterisk-min 1.2 ? |
| 02:38.24 | frogonwheels | genji1981: lets shift to openwrt |
| 02:38.27 | genji1981 | frogonwheels: asterisk 1.4 |
| 02:46.46 | *** join/#asterisk Jenna (i=JJ@gateway/tor/x-da10c266e7b248b2) |
| 02:46.55 | *** join/#asterisk Levonk (n=lk@76.246.209.178) |
| 02:48.23 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:50.21 | *** join/#asterisk luxxx77 (n=luxxx77@e179205253.adsl.alicedsl.de) |
| 02:50.46 | UD | does fwd still work? |
| 02:51.29 | jaytee | sure! I can fwd my phone just fine |
| 02:51.38 | UD | doesnt work for me |
| 02:51.44 | UD | maybe its this crappy client im using |
| 02:52.18 | jaytee | oh wait! you probably meant FWD (FreeWorldDialup). |
| 02:52.26 | UD | yea |
| 02:52.29 | Qwell | it's "dead" |
| 02:52.36 | UD | darn |
| 02:52.40 | Qwell | the "free" part is, anyways |
| 02:52.46 | UD | well i signed up |
| 02:52.50 | UD | it gave me a number |
| 02:52.53 | Qwell | it's $30/year now.. |
| 02:53.01 | Qwell | as of...Monday |
| 02:54.26 | UD | as of now, it doesnt work |
| 02:54.29 | UD | :/ |
| 02:54.39 | Qwell | yeah |
| 02:54.56 | jaytee | and you can't dial a pstn number or have someone call your VOIP number from a PSTN phone |
| 02:55.14 | UD | i cant even add my account to sip-communicator |
| 02:55.16 | russellb | wow, i'm sold |
| 02:55.21 | jaytee | lol |
| 02:55.38 | UD | i think the server is down |
| 02:55.50 | jaytee | their website is running |
| 02:57.37 | UD | i have a feeling this is going to be one of those things that takes me like 6 months finding one that actually works |
| 03:00.13 | russellb | TINSTAFVS |
| 03:00.19 | russellb | there is no such thing as a free voip service |
| 03:01.42 | UD | ya i figured the whole site and oreilly book was lying about it being able to be used |
| 03:02.02 | russellb | heh |
| 03:02.07 | russellb | well, it used to be free ... |
| 03:03.15 | jaytee | TINSTAAFLOAHP |
| 03:03.31 | jaytee | there is no such thing as a free lunch or an honest politician |
| 03:03.37 | russellb | ah ... |
| 03:03.45 | JT | Fee World Dialup |
| 03:03.57 | UD | bleh they should put that on their page instead of "hey this is free, click here to get started" and "you're all set. here's your account info" |
| 03:04.00 | russellb | I thought it was Free Loaf of Angry Hornet Poets |
| 03:04.34 | UD | maybe they just want to try to piss off as many customers as possible |
| 03:04.35 | jaytee | ummm, russell? I think you need some sleep :-) |
| 03:04.53 | UD | well potential customers |
| 03:05.07 | russellb | jaytee: probably. |
| 03:05.21 | jeev | wiped his ass a lot after this last dum |
| 03:05.22 | jeev | p |
| 03:06.01 | russellb | wtf |
| 03:06.03 | jaytee | "Welcome to Free World Dialup! For the standard "dry hump" package just click here and bend over! For the Premium Fully Lubed package just click there and have your credit card ready. |
| 03:06.25 | UD | ya basically |
| 03:07.05 | jaytee | jeev, sometimes you make me miss being 12. It was so silly, immature and carefree back then. Just like you are now. |
| 03:07.16 | jeev | isn't it just great? |
| 03:08.11 | dwayne | when you are 12 you can laugh at fart noises. that was fun. |
| 03:08.25 | jeev | yea, you can fart wherever you want |
| 03:08.26 | jaytee | told you once already but I'll tell ya again since I know from trying to help you with * just how slow you are to "get it". I personally don't care about the quality, consistency, frequency or aroma of your craps! |
| 03:08.29 | jeev | nobody can say anything |
| 03:08.37 | frogonwheels | russellb: found the issue - it was a mismatch between toolchain autoconf and installed autoconf... so openwrt build issue - thanks |
| 03:08.52 | russellb | np, i didn't do much, heh |
| 03:09.13 | frogonwheels | nah - but sometimes all it needs is a little nudge :) |
| 03:09.33 | jaytee | so FreePBX is going to do a seminar at Digium? that's a bit odd |
| 03:09.46 | russellb | why is it odd? |
| 03:11.15 | *** join/#asterisk spokra (n=spokra@75-92-217-76.sea.clearwire-dns.net) |
| 03:11.17 | jaytee | because * has it's own gui and freepbx, while using * as it's base is not the same thing. just figured with all the deep love you feel in here for GUIs in general it is just a bit odd. |
| 03:11.45 | russellb | personally, I think the more GUIs the merrier |
| 03:12.01 | russellb | and sure, it's not the same thing, but it's an important part of the asterisk community that can't be ignored |
| 03:12.09 | frogonwheels | russellb: I'm just in the process of making openwrt asterisk web configurable. |
| 03:12.24 | russellb | a lot of people use it, a lot of people like it, so we might as well do what we can to support the effort |
| 03:12.33 | jaytee | russellb, I dare you to go into #asterisk-cli and say that :-) |
| 03:12.52 | frogonwheels | russellb: Exactly. |
| 03:13.10 | jaytee | hey, I think a GUI is actually a necessity for some things. |
| 03:13.25 | russellb | frogonwheels: the asterisk-gui should be perfect for that ... it's all javascript that talks directly to asterisk |
| 03:13.27 | frogonwheels | there are people in openwrt against guis too - but I think they serve a very useful service. |
| 03:13.50 | russellb | at least use it as a base |
| 03:14.05 | frogonwheels | russellb: The problem is that openwrt has its own configuration mechanism. |
| 03:14.28 | russellb | web config you mean? |
| 03:14.34 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:14.34 | russellb | well, to start, you could just run the asterisk web server on a different port |
| 03:14.38 | frogonwheels | russellb: so I wanted a way of using the openwrt configuration mech to provide useful asterisk config |
| 03:14.46 | russellb | gotcha, well sounds good |
| 03:15.02 | jaytee | adding phones and changing user settings like CallerID name and number, VM passwords etc and even provisioning phones if possible but where I have a problem is when a GUI based * derivative changes your dialplan logic or doesn't allow you to modify it to do things other than a strict template design. |
| 03:15.05 | frogonwheels | russellb: well actually the config is a CLI config - but the web interfaces use the cli config to do the stuff |
| 03:15.07 | russellb | whatever makes asterisk more accessible and easier to use gets a +1 from me :) |
| 03:15.20 | *** join/#asterisk saint_ (n=saint@cpe-75-83-219-24.socal.res.rr.com) |
| 03:15.30 | frogonwheels | russellb: exactly .. and s/asterisk/openwrt/ for me as well |
| 03:15.33 | jaytee | russellb, good attitude |
| 03:16.27 | saint_ | Hi all |
| 03:16.38 | frogonwheels | jaytee: I've gone for a the thing you hate - generating a dialplan - but will allow #include to be specified for any other stuff. |
| 03:16.48 | saint_ | Is there a way to have a pbx with users using an external SIP for voicemail ? |
| 03:17.12 | saint_ | I've been trying to do that with no success.. I can see the user's line being overflowed on the SIP , but SIP answer with UNKNOWN USER |
| 03:17.13 | russellb | yes. |
| 03:17.21 | saint_ | and I created the user on the Asterisk, with voicemail |
| 03:17.27 | saint_ | I meant ASTERISK, not SIP |
| 03:17.29 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:18.00 | frogonwheels | saint_: I made scripts to bring the external SIP voicemail - which can send it's voicemail by email - and put it into the local asterisk voicemail box |
| 03:18.26 | *** join/#asterisk mike-ekim (n=mike@24.238.28.248) |
| 03:18.34 | jaytee | I route all unanswered calls to my Exchange Server for Unified Messaging |
| 03:19.08 | *** part/#asterisk Jenna (i=JJ@gateway/tor/x-da10c266e7b248b2) |
| 03:19.08 | russellb | srsly? |
| 03:19.09 | jaytee | but I have to route through sipX as proxy to do udp to tcp transforms |
| 03:19.12 | saint_ | russellb, / frogonwheels .. My users are on an Alcatel PBX.. I have the Asterisk registering on the Alcatel. This works. When I create the Asterisk as external voicemail, the calls are being routed to asterisk, but it goes to the main greeting |
| 03:19.30 | jaytee | until I get 1.6 working with Exchange then I can dump sipX |
| 03:19.45 | russellb | jaytee: I'd be interested how that works for you (with 1.6) |
| 03:19.50 | russellb | so I can help fix it if you have problems |
| 03:19.55 | saint_ | When I try to make it part of Alcatel , like I would do with an AVST voicemail , I can see Alcatel transfering the call to Asterisk, but asterisk does not like it anymore ... I have some traces if someone would not mind looking at them .. |
| 03:19.56 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 03:19.56 | jaytee | saint, you need to use SIPAddHeader with Diversion |
| 03:20.07 | russellb | though I won't have any time to work on it until at least after Astricon ... |
| 03:20.19 | saint_ | jaytee, where is this ? |
| 03:20.30 | jaytee | russellb, I'll take copious notes and I'm going to write a how-to for voip-info.org |
| 03:20.45 | russellb | jaytee: ok, awesome ... keep me updated if you think about it |
| 03:20.49 | jaytee | saint, at my work in Indianapolis |
| 03:23.59 | jaytee | saint, this is the macro I use to dial Exchange UM. http://pastebin.ca/1193312 |
| 03:24.38 | jaytee | the first part of the macro just checks for blocked callerid and sets the null to "Unknown" |
| 03:24.47 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 03:26.11 | jaytee | then it dials the phone, if the phone doesn't answer it adds SIP header info for Diversion and the extension number and then dials my Exchange server. the "trunk-sipx" is a the sipX proxy that registers to * as a peer |
| 03:26.42 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 03:27.45 | jaytee | sipX is setup to route any calls to it from * to 1111 to the Exchange UM mail box number after converting the packets to tcp from udp |
| 03:28.30 | jaytee | calls from Exchange work the same way. I can call the Exchange UM auto attendant and do dial by name voice recognition |
| 03:28.40 | saint_ | holy shit, you need this to have it work correctly !? |
| 03:28.56 | saint_ | I don t understand why it works localy ? |
| 03:28.57 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 03:29.06 | jaytee | to use it with Exchange? yes. until * 1.6 which supports SIP over TCP |
| 03:29.49 | jaytee | 1.4 or earlier only supports SIP over UDP so * can't "talk" directly to Exchange because it only "talks" SIP TCP |
| 03:30.46 | jaytee | but that's just my environment, your needs may be different but the SIPAddHeader piece in my macro is what you need to tell your VM system what number the call was Diverted from. |
| 03:31.58 | saint_ | I am using AsteriskNOW! actually. Does it follow the same releases as Asterisk ? |
| 03:32.32 | jaytee | I believe *NOW is up to 1.4 already |
| 03:33.00 | jaytee | but I haven't used it. I just installed the asterisk-gui on my home system on vanilla * the hard way. |
| 03:35.01 | saint_ | I am VPNing right now in the office. Going to check out what release I have |
| 03:36.09 | *** join/#asterisk h3x (n=Hex@64.192.116.17) |
| 03:36.14 | jaytee | I have to use Putty to SSH into my * boxes at work from my Windows box here because I've yet to find a Cisco VPN client for linux (Ubuntu) that works with our setup. |
| 03:36.52 | saint_ | ok, it's running 1.4.18 |
| 03:38.51 | saint_ | jaytee, , can you look at http://pastebin.ca/1193321 , and tell me what you think ? |
| 03:39.47 | saint_ | so extension 2000 is calling 1000 |
| 03:39.51 | saint_ | both are on 192.168.2.10 |
| 03:40.00 | saint_ | and voicemail is 7000 on asterisk (192.168.0.254) |
| 03:40.51 | saint_ | ha shit |
| 03:40.55 | saint_ | it did not take all the traces |
| 03:40.58 | saint_ | hold on |
| 03:41.24 | saint_ | never mind, it did |
| 03:41.28 | saint_ | you cann see the 404 NOT FOUND |
| 03:43.18 | jaytee | 2000 and 1000 are both on 192.168.2.10. What is that? |
| 03:43.30 | saint_ | it's my alcatel pbx |
| 03:44.10 | jaytee | so those are both SIP phones on the Alcatel? |
| 03:44.15 | saint_ | no |
| 03:44.23 | saint_ | IP phones, |
| 03:44.30 | saint_ | but they can call the asterisk no problem |
| 03:44.38 | saint_ | the only issue i have is when I try to use asterisk as a voicemail |
| 03:44.45 | saint_ | i took the same configuration I have with AVST |
| 03:45.08 | saint_ | where when a user does not answer, calls are fw to AVST with the invite to the called user.. AVST gets to his voicemail right away |
| 03:45.25 | saint_ | with asterisk, if I call directly with a set ON the asterisk to extension 1000, i get to 1000 voicemail |
| 03:45.45 | saint_ | when i call from the alcatel B to the alcatel A , its supposed to overflow on asterisk VM 1000 ... |
| 03:45.52 | scooby2 | is there any way to view global variables? I am trying to figure out why AGENTBYCALLERID_XXX is no longer able to be read. (yes i know its old) |
| 03:45.54 | saint_ | and this is when asterisk answer with user unknown .. |
| 03:46.09 | scooby2 | show globals is not showing anything |
| 03:48.33 | jaytee | saint, so when you use 2000 to dial 1000 you WANT calls to go to VM on *? |
| 03:49.31 | jaytee | or is this from a call placed to 2000 that tries to "roll" to voicemail on the alcatel and gets redirected? |
| 03:56.15 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 03:57.21 | saint_ | jaytee, after my timer, year |
| 03:57.24 | trnzmeta | guys: I am not getting phone not in service response when I'm making sip phone calls |
| 03:57.40 | saint_ | jaytee, 2000 calls 1000 , and instead of overflowing to alcatel voicemail, i try to overflow on asterisk |
| 03:57.41 | trnzmeta | when I call using my mobile, I get the phone... is disconnected |
| 03:57.47 | saint_ | so i created an account 1000 on the asterisk .. |
| 03:58.02 | trnzmeta | all that happens is it rings out using sip phone |
| 03:58.11 | saint_ | trnzmeta, any traces ? |
| 03:58.16 | UD | hmm does ipkall actually work? |
| 03:58.18 | trnzmeta | any settings I should check |
| 03:58.34 | UD | anyone used recently? |
| 03:58.57 | trnzmeta | saint_: come again? |
| 03:59.01 | mchou | UD: Ipkall works great |
| 03:59.15 | UD | all i get is a busy signal |
| 03:59.30 | UD | i tried using my ekiga information |
| 03:59.34 | UD | no go |
| 03:59.40 | jaytee | saint, from the SIP header info you posted it doesn't look like the call is trying to dial 1000 at * but 1000 at alcatel. |
| 03:59.42 | rob0 | I've used and relied on ipkall numbers for a long time. |
| 03:59.46 | mchou | that's cause ipkall doesnt allow outbound calls, only inbound |
| 04:00.11 | mchou | rob0: you ever use grandcentral, by any chance? |
| 04:00.18 | UD | i am calling my # with my cell phone |
| 04:00.37 | mchou | UD: what can we say, your config must be wrong |
| 04:00.50 | rob0 | no, I use asterlink for outbound, ipkall & stanaphone inbound. |
| 04:01.05 | mchou | UD: cause ipkall WORKS |
| 04:01.15 | rob0 | "set sip debug on" is your friend |
| 04:02.04 | jplank | my boss is gearing up for astricon, he's bought a bunch of books |
| 04:02.17 | jplank | and he just emails me this |
| 04:02.20 | jplank | Never in the history of telecommunications has a system so suited to the needs of business been available, at any price! |
| 04:02.42 | UD | ok cool |
| 04:02.55 | UD | i did make a typo |
| 04:04.56 | saint_ | jaytee, so the invite is wrong ? |
| 04:05.37 | jaytee | looks like it. the invite is to 192.168.2.10 |
| 04:06.00 | saint_ | damn... i ll run traces again tomorrow with the AVST sip voicemail |
| 04:06.06 | jaytee | there's an ACK there from s@192.168.0.254, your * box but the 404 was from your alcatel |
| 04:07.23 | saint_ | I can run a tcpdump remotely. Let me do that . i'll use wireshark.. it's easier to read traces for me.. |
| 04:07.54 | jaytee | well, do it tomorrow because I'm going to bed in a couple minutes |
| 04:08.53 | saint_ | where r u located ? |
| 04:09.19 | jaytee | Indianapolis, Indiana, USA |
| 04:09.22 | jaytee | you? |
| 04:09.28 | saint_ | california |
| 04:09.39 | saint_ | @ alcatel :) |
| 04:09.48 | jaytee | ok, kewl! |
| 04:09.53 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 04:10.24 | jaytee | hey! look who's back after getting a whole 3 hours sleep |
| 04:11.48 | jaytee | and speaking of sleep, I'm off to catch some zzzzzz's. |
| 04:11.53 | jaytee | nite folks |
| 04:11.58 | saint_ | nite |
| 04:16.20 | *** join/#asterisk Levonk (n=lk@76.227.117.171) |
| 04:16.52 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 04:17.01 | *** join/#asterisk irieKEN (n=irieken@netblock-68-183-115-231.dslextreme.com) |
| 04:18.34 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 04:18.47 | UD | argh still no good |
| 04:19.56 | UD | ipkall forwards the call but when i try to answer a call with SIP-communicator, I immediately get "Failed" |
| 04:20.31 | UD | any other ports that i need to open besides 5060? |
| 04:21.28 | UD | this is really hard to debug. i'm using ekiga.net which seems to be 100% free, but who knows. i'm thinking i could have done it all right and the service is screwing it up |
| 04:28.31 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 04:41.06 | UD | got it |
| 04:41.17 | UD | it was this buggy alpha software i was using |
| 04:41.24 | UD | neat :) |
| 04:45.11 | *** join/#asterisk astassistant (n=support@h-67-101-6-180.sttnwaho.dynamic.covad.net) |
| 04:54.11 | *** join/#asterisk jsolis (n=jimmy@200.106.43.180) |
| 04:57.34 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 05:03.59 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 05:04.23 | Sargun | nano > vmware |
| 05:20.39 | *** part/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 05:21.28 | *** join/#asterisk genji1981 (n=gej@203.152.122.50) |
| 05:42.09 | *** join/#asterisk UD (n=suck@unaffiliated/underdawg) |
| 05:50.35 | *** join/#asterisk i3inary (n=shaman@cpe-76-88-94-149.san.res.rr.com) |
| 05:54.09 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-07925800d923806d) |
| 06:02.48 | *** join/#asterisk Levonk (n=lk@75.62.137.112) |
| 06:13.07 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 06:14.56 | gnorbert | Hi all. I have a question. If I change the value of a variable from a call file, does that value change in the other call files too? |
| 06:16.17 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-3f2e720fc659330c) |
| 06:20.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:21.41 | tzafrir_laptop | gnorbert, only if it is a global |
| 06:23.16 | gnorbert | tzafrir_laptop: Thank you. |
| 06:34.19 | *** join/#asterisk Maxous (n=Maxous@76.97.26.56) |
| 06:34.32 | *** part/#asterisk Maxous (n=Maxous@76.97.26.56) |
| 06:37.25 | UD | fuck |
| 06:37.39 | drmessano | Maybe later |
| 06:37.43 | UD | i've been trying to get zaptel to work with my x100p all night |
| 06:38.40 | UD | everything i tries just leads to... |
| 06:38.41 | UD | ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) |
| 06:39.04 | UD | the pci card is there |
| 06:39.10 | UD | the modules are there |
| 06:39.13 | UD | loaded |
| 06:39.16 | UD | still, fail |
| 06:41.50 | drmessano | Dunno, at $13 an hour.. your effort has cost you three times as much as the card is worth |
| 06:42.30 | obnauticus | :| |
| 06:42.35 | *** part/#asterisk t0ny-p40 (n=t0ny-p40@67.61.114.85) |
| 06:46.17 | *** join/#asterisk barakuda_work (n=barakuda@aliens4.betex.ru) |
| 06:49.41 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 06:50.42 | tzafrir_laptop | UD, cat /proc/zaptel/* |
| 06:50.56 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 06:50.56 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 06:51.07 | gnorbert | tzafrir_laptop: It looks like it doesn't make change in other call files... Should I make something else, then change the value of the variable with a call file? |
| 06:51.49 | tzafrir_laptop | gnorbert, if you want that, use a global. Normal variables are local to channels |
| 06:52.22 | gnorbert | tzafrir_laptop: Sorry, I pastebin the extensions.conf and the call file, I think it's more simple that way. |
| 06:56.21 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 06:56.52 | UD | Span 1: ZTDUMMY/1 "ZTDUMMY/1 (source: RTC) 1" (MASTER) |
| 06:57.18 | UD | tzafrir_laptop ^^ |
| 06:57.40 | barakuda_work | hi ppl |
| 06:57.53 | UD | hmm i didnt think i needed ztdummy |
| 06:57.57 | barakuda_work | has anyone connected cisco phone 7971 with * by SIP |
| 06:57.58 | barakuda_work | ? |
| 06:58.24 | tzafrir_laptop | UD, rmmod ztdummy; modprobe wcfxo |
| 06:58.42 | tzafrir_laptop | hmmmm, actually: |
| 06:59.03 | tzafrir_laptop | what do you then see in /proc/zaptel/* |
| 06:59.49 | UD | ztdummy doesnt even exist on my system, wcfxo was already loaded |
| 07:00.07 | UD | and i just posted the output from that ^ |
| 07:00.20 | UD | wont change because nothings changed |
| 07:01.40 | gnorbert | tzafrir_laptop: http://pastebin.com/d29f03391 I would like to make stop a repeated sound file, that I made by a call file. |
| 07:01.59 | gnorbert | However it looks like it doesn't change the value of "notagain" |
| 07:02.00 | UD | i actually just did a downgrade to zaptel 1.2 |
| 07:02.23 | UD | and apt-get upgrade ; apt-get update |
| 07:02.30 | tzafrir_laptop | UD, what is the output of: zaptel_hardware |
| 07:02.36 | UD | so now wcfxo doesnt exist either |
| 07:02.38 | UD | :( |
| 07:02.45 | UD | it did before |
| 07:03.09 | tzafrir_laptop | cat /sys/module/zaptel/version |
| 07:04.03 | UD | i get FATAL: Could not load .....modules.dep |
| 07:04.08 | UD | the file's gone |
| 07:04.13 | kaldemar | barakuda_work: http://lists.digium.com/pipermail/asterisk-users/2006-July/158790.html |
| 07:04.14 | UD | i have to now fix other stuff |
| 07:04.22 | UD | but anyway... (sorry i have to manually copy) |
| 07:04.44 | tzafrir_laptop | UD, let's go over things one step at a time, ok? |
| 07:04.54 | tzafrir_laptop | I need clear datapoints |
| 07:04.54 | UD | pci:0000:01:04.0 wcfxo- 1057:5608 Wildcard X100P |
| 07:05.55 | UD | and for output #2 (version) |
| 07:06.06 | UD | SVN-branch-1.4-r4432M |
| 07:06.19 | tzafrir_laptop | now: could you please pastebin the output of: 'modinfo wcfxo' |
| 07:06.20 | UD | i just got the 1.2 debian package. |
| 07:06.44 | UD | my modules.dep is gone |
| 07:06.45 | tzafrir_laptop | have you built zaptel-modules package? m-a a-i zaptel |
| 07:06.47 | UD | i have to relink it |
| 07:06.52 | UD | or see whats going on |
| 07:07.17 | tzafrir_laptop | if you have to mess with modules.dep, you're doing something wrong |
| 07:07.19 | UD | apt-get just screwed me |
| 07:07.22 | UD | trying to fix it |
| 07:07.32 | *** join/#asterisk supa_disko (n=bleh@secure27.lnk.telstra.net) |
| 07:07.37 | tzafrir_laptop | dpkg -l zaptel\* |
| 07:07.40 | UD | hold up a sec lemme sort that out real quick |
| 07:08.07 | tzafrir_laptop | UD, can you please stop that and show the status of your system? |
| 07:08.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:08.26 | UD | 1.2.11.dfsg-1 |
| 07:08.42 | barakuda_work | kaldemar: aaarrghh =(( thanks for pointing |
| 07:08.51 | tzafrir_laptop | just zaptel? what about zaptel-modules ? |
| 07:09.02 | tzafrir_laptop | apt-get install zaptel-source |
| 07:09.06 | tzafrir_laptop | m-a a-i zaptel |
| 07:09.37 | UD | they all just disappeared when i tried the apt-get install zaptel |
| 07:09.55 | UD | also i think i have 2 versions from 2 attempts or something |
| 07:10.29 | kaldemar | barakuda_work: that message is from 2006 though. it was more like a hint to google 'til your fingers bleed. :) |
| 07:10.31 | *** join/#asterisk cbr1000 (n=AnthonyC@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 07:10.39 | cbr1000 | Hello, |
| 07:11.00 | cbr1000 | I connected two astersik via iax2 (i work with asterisk 1.4.21.2) |
| 07:11.17 | tzafrir_laptop | UD, please pastebin the *full* output of: dpkg -l zaptel\* |
| 07:11.46 | *** join/#asterisk Levonk (n=lk@75.62.140.2) |
| 07:11.47 | barakuda_work | kaldemar: i googling more than a week extensivly :-) but not much info about succesful installation for this model =(( |
| 07:11.51 | cbr1000 | so it's work fine. but is it possible to use the BLF to know if extension is busy? |
| 07:11.55 | tzafrir_laptop | UD, is this a new installation or an existing system? |
| 07:12.22 | UD | its pretty old, i did it but it was a while ago |
| 07:13.19 | *** join/#asterisk _trine (n=trine@38.100.90.3) |
| 07:13.49 | UD | ok i did a depmod |
| 07:14.24 | kaldemar | barakuda_work: there's always chan_skinny to try if you can use the SCCP firmware. |
| 07:14.39 | UD | and now modprobe wcfxo which failed... i'm going to load up x11 on the box so i dont have to copy it manually |
| 07:15.19 | barakuda_work | kaldemar: this was my 1st step =) latest skinny firmware with the simpliest basic config doesnt seem to work right =( |
| 07:17.48 | gnorbert | Do I make global variables by writing the variable name and the value in the [globals] section in extensions.conf? If yes, then what could be the cause that it doesn't work? (That I pastebined, I copy the address again of course, if it's needed.) |
| 07:18.48 | mchou | ~/book |
| 07:18.51 | mchou | ~book |
| 07:18.52 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 07:19.20 | cbr1000 | To resume, is it possible to "use" BLF via IAX2 channels? |
| 07:20.01 | kaldemar | gnorbert: you're using the variables wrong in your dialplan. |
| 07:20.27 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 07:20.40 | gnorbert | kaldemar: What did I wrong? |
| 07:20.41 | *** join/#asterisk ToTo (n=ToTo@209.8.41.139) |
| 07:21.02 | kaldemar | gnorbert: in the playbacks, the variables should be inside ${}, otherwire it's just text. |
| 07:21.05 | gnorbert | +do |
| 07:21.09 | *** join/#asterisk UD (n=steve@unaffiliated/underdawg) |
| 07:21.44 | UD | tzafrir_laptop: http://pastebin.com/m5dede32e |
| 07:21.50 | kaldemar | gnorbert: in the gotoif's, you must have an expression to evaluate, not just a variable with or without brackets. two different kind of failures with those. |
| 07:21.52 | gnorbert | kaldemar: That's not the point, it's only at comments. |
| 07:22.50 | tzafrir_laptop | UD, so how have you build the zaptel modules? |
| 07:23.03 | tzafrir_laptop | try just: m-a a-i zaptel |
| 07:23.09 | gnorbert | kaldemar: Sorry, just I left it in as comments, because maybe I'll need later something similar to this. |
| 07:23.12 | kaldemar | gnorbert: NOT like this: exten => 99,2,Gotoif(${notagain}?99,4:99,3), but like this: exten => 99,2,Gotoif($[${notagain}]?99,4:99,3) |
| 07:23.27 | UD | i compiled them, it have me the ioctl error so i used apt-get, and now i probably have bits and pieces of both i guess |
| 07:23.30 | gnorbert | Hmm... |
| 07:23.32 | kaldemar | gnorbert: anyway, you're using GotoIf wrong, fix it. |
| 07:23.40 | UD | should have used checkinstall more often |
| 07:23.44 | gnorbert | kaldemar: Thanks, I try it now... |
| 07:23.49 | tzafrir_laptop | UD, compiled them from? |
| 07:24.03 | UD | the latest copy from svn |
| 07:24.20 | tzafrir_laptop | that version is not compatible |
| 07:25.21 | UD | The following NEW packages will be installed: build-essential |
| 07:25.42 | UD | is that all it should get? |
| 07:26.14 | tzafrir_laptop | build-essential is a dummy package to make sure you have all the basic packages needed to build stuff |
| 07:28.48 | gnorbert | kaldemar: I changed this, but I guess something can be still wrong, because it still doesn't work. |
| 07:29.56 | kaldemar | gnorbert: paste your extensions.conf again and a cli output of a failed call. |
| 07:30.23 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 07:30.32 | kaldemar | gnorbert: and write what you're expecting it to do in the paste. |
| 07:36.28 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 07:36.39 | gnorbert | kaldemar: I can't really paste a wrong call, because asterisk runs a a sound file continously and it means that writes a lot of line... http://pastebin.com/dfc61c5f |
| 07:37.53 | kaldemar | gnorbert: you don't have to paste an infinite number of lines, the first loop is enough. |
| 07:39.04 | UD | i tried to modprobe usb_uhci and got. FATAL: Module usb_uhci not found. |
| 07:39.09 | UD | is that a thing of the past? |
| 07:41.27 | kaldemar | gnorbert: where are you trying to set notagain to 1? your dialplan has no such line. |
| 07:41.49 | UD | i just updated my kernel and linux-headers |
| 07:41.51 | gnorbert | Set:notagain=1 |
| 07:42.02 | gnorbert | kaldemar: I have a line like that. ^ |
| 07:42.31 | UD | and the linux-source package too |
| 07:42.45 | UD | im thinking maybe start over again |
| 07:42.46 | kaldemar | gnorbert: not in your paste. if you're not showing me relevat parts of your dialplan, i can't help you. |
| 07:43.14 | gnorbert | kaldemar: Sorry, I copied it manually from a script... |
| 07:43.16 | kaldemar | gnorbert: taking a wild guess, you're probably setting a channel variable to 1, not a global one. |
| 07:43.23 | gnorbert | Sec, I see it. |
| 07:43.40 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 07:43.42 | gnorbert | kaldemar: It's there. |
| 07:43.46 | kaldemar | use Set(GLOBAL(var)=value) in the other channel. |
| 07:43.48 | gnorbert | The second line from the button. |
| 07:44.12 | gnorbert | *bottom of the page... :) |
| 07:44.31 | kaldemar | second line from the bottom of your paste is "exten => o,1,Hangup" |
| 07:44.49 | gnorbert | http://pastebin.com/d29f03391? |
| 07:44.59 | gnorbert | I mean http://pastebin.com/d29f03391 ? |
| 07:45.28 | kaldemar | you gave me a different paste earlier... |
| 07:45.58 | UD | tzafrir_laptop: any suggestions? i still have the error |
| 07:46.02 | gnorbert | kaldemar: I've already closed that page, reloaded it from the link in xchat... |
| 07:46.10 | gnorbert | From my previous message. :S |
| 07:47.43 | tzafrir_laptop | you don't need linux-source . Only linux-headers-`uname -r` . And m-a should fetch it for you, IIRC |
| 07:48.48 | tzafrir_laptop | that said, I maintain a set of newer packages, with -modules built at: deb http://updates.xorcom.com/rapid etch main |
| 07:49.06 | kaldemar | gnorbert: when are you showing me a failed call? |
| 07:49.36 | UD | well i figured that (and originally built it without linux-source on my system), but figured i'd need it eventually |
| 07:49.58 | gnorbert | kaldemar: Working on it, in a minute... :) |
| 07:51.38 | *** join/#asterisk lesouvage (n=lesouvag@82.199.132.58) |
| 07:52.03 | UD | could you give me a link to the actual directory? |
| 07:53.13 | gnorbert | kaldemar: That's, how the cli output looks like, when it plays a loop sound at me: http://pastebin.com/d56428eaa |
| 07:53.17 | UD | i dont see it |
| 07:54.00 | *** join/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 07:54.10 | bootc | morning folks |
| 07:54.27 | gnorbert | good morning bootc. |
| 07:55.16 | bootc | In 1.4.22 with chan_dahdi and libpri you can set "inbanddisconnect" to get the inband messages from the telco |
| 07:55.44 | bootc | this is great on outbound calls but on inbound it also plays them, and thus you get voicemails that are too long and get dropped, etc... |
| 07:56.06 | kaldemar | gnorbert: btw, looks like you're always trying to set notagain to 1 so your dialplan logic is meaningless. you could as well just playback track and hang up. |
| 07:56.18 | bootc | what can I do to control this a bit? I don't mind hacking the source a little if I need to |
| 07:57.04 | kaldemar | gnorbert: removing the notagain definition from globals might help. |
| 07:57.22 | kaldemar | it might be canceling your set in the callfile. |
| 07:57.25 | gnorbert | kaldemar: I should loop a sound... (That works now...) And I should be able to stop it also. |
| 07:57.47 | kaldemar | if you set the variable in the callfile, it will never loop. |
| 07:58.18 | kaldemar | set the variable in the other end in your dialplan when you're done looping. |
| 07:58.21 | gnorbert | It loops, I can hear it. :) |
| 07:59.10 | gnorbert | And I can also give it whether to loop or not.. |
| 08:00.18 | *** join/#asterisk angryuser (n=aster@78.116.106.127) |
| 08:01.44 | gnorbert | kaldemar: Thanks, now it works. |
| 08:02.19 | angryuser | good day |
| 08:03.48 | *** join/#asterisk genji1981 (n=gej@203.152.122.50) |
| 08:07.17 | angryuser | i have 2 outgoint peers for the same local calls, whright now if peer1 is used i call with peer2 (set grop & group count) i wold like to use peer1 & peer2 one after another, any ideas ? |
| 08:08.38 | angryuser | hm i think i should use some var in astdb and add a value and second time change that value |
| 08:09.19 | dandre | hello, |
| 08:10.10 | dandre | is there any possibility to use goto to skip next priority in the dialplan w/o user labels? |
| 08:10.50 | *** join/#asterisk Levonk (n=lk@adsl-75-62-142-148.dsl.lsan03.sbcglobal.net) |
| 08:10.55 | dandre | ie |
| 08:10.55 | dandre | exten => s,n ,GotoIf(...?n+2) |
| 08:12.40 | gnorbert | dandre: Yes, I guess it also looks like a working sample...^ |
| 08:19.16 | UD | hey tzafrir_laptop i think i bought a junk card maybe |
| 08:20.19 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:22.58 | tzafrir_laptop | UD, if you're beginning with Debian now, it's p[robably best to try Lenny. By the time you get to deploy your production system, it will probably become Stable |
| 08:23.26 | UD | ya i guess |
| 08:23.44 | UD | thats what i need just something that works i want to rent another server anyway with better connection |
| 08:27.23 | angryuser | tzafrir_laptop, what so good in lenny ? |
| 08:30.20 | tzafrir_laptop | first off, newer kernel, newer stuff |
| 08:30.25 | tzafrir_laptop | newer everything |
| 08:31.13 | angryuser | tzafrir_laptop, that's nice in general, but any points related to * ? |
| 08:31.40 | kaldemar | dandre: the n's convert to numbers when the dialplan is read. |
| 08:32.19 | dandre | so I must use labels in that case |
| 08:32.25 | tzafrir_laptop | interesting thing I read on LWN yesterday: it seems that as of kernel 2.6.25 scheduling policy RR will not let a process use 100% of the time |
| 08:33.05 | tzafrir_laptop | So if Asterisk freaks out and has -p it should theoretically still leave out 5% CPU time available. And the canary would not kick in |
| 08:33.15 | tzafrir_laptop | Didn't get to test that |
| 08:33.53 | UD | is lenny like an asterisk specific thing? |
| 08:34.14 | tzafrir_laptop | Lenny is the upcoming Debian release |
| 08:34.19 | kaldemar | dandre: you can use numbers too, but they won't work if you add a new n-line between the goto line and the target. labels are better in that sense. |
| 08:34.23 | tzafrir_laptop | nothing Asterisk-specific |
| 08:34.23 | UD | it'd kinda be nice to have lightweight since im running on old crap hardware |
| 08:34.43 | Maliuta | tzafrir_laptop: really? I thought he was a Simpsons character :P |
| 08:34.46 | tzafrir_laptop | What hardware? |
| 08:35.02 | tzafrir_laptop | Debian => Toy Story |
| 08:35.18 | Maliuta | tzafrir_laptop: I go back to bo, no need to tell me that |
| 08:35.25 | UD | its a pentium 4 system but it has issues |
| 08:35.39 | UD | i think the ram is going bad maybe the mobo |
| 08:35.44 | UD | sometimes the ram doesnt always work |
| 08:35.52 | Maliuta | tzafrir_laptop: before apt, before glibc :) |
| 08:35.53 | UD | does goofy things and lags a lot |
| 08:38.25 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 08:42.31 | phix | :D |
| 08:54.54 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 08:56.57 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:13.42 | EmleyMoor | I remember seeing a patch a long time ago, aimed mainly at Australia, to enable dring detection and caller ID at the same time. I am looking for a patch that would do the same in the UK and think that may be my starting point, but I can't find it. |
| 09:19.12 | *** join/#asterisk Levonk (n=lk@adsl-75-62-143-15.dsl.lsan03.sbcglobal.net) |
| 09:19.26 | *** join/#asterisk aksyn (n=aksyn@93-97-50-116.zone5.bethere.co.uk) |
| 09:23.40 | *** join/#asterisk axisys (n=axisys@117.18.224.162) |
| 09:30.17 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 09:34.21 | *** join/#asterisk cbr1000 (n=AnthonyC@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 09:35.31 | cbr1000 | hello, How can I use see BLF status on other asterisk. I connected 2 asterisk via iax channel and I want see the blf status on other ipbx. How can I do? |
| 09:35.34 | cbr1000 | thanks |
| 09:39.57 | *** join/#asterisk HumpBack (n=sdcdcssc@gentoo/developer/HumpBack) |
| 09:40.16 | HumpBack | Hello all |
| 09:41.09 | HumpBack | I'm trying to use the odbc connector. isql -v asterisk-connector reports a valid connection |
| 09:41.23 | HumpBack | I configured that dsn into res_odbc.conf |
| 09:41.37 | HumpBack | but the cli command odbc show reports nothing |
| 09:41.52 | HumpBack | How can I see more infor on the possible error in the connection? |
| 09:41.54 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 09:47.02 | *** join/#asterisk Levonk (n=lk@adsl-75-62-128-117.dsl.lsan03.sbcglobal.net) |
| 09:53.08 | HumpBack | ok found the issue |
| 09:53.44 | HumpBack | the asterisk book says to use a limit => 0 in the configuration of the odbc->porstgre connection |
| 09:53.50 | HumpBack | that was the problem |
| 09:54.32 | angryuser | cbr1000, i think you cant share BLF upon many servers, this feature will be implemented in 1.6 |
| 09:54.40 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 09:56.08 | cbr1000 | angryuser: ok and in 1.4.21, this feature is not available? how can I do to resolv my problem? |
| 09:56.23 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:57.03 | *** join/#asterisk ric0h (i=Miranda@phoenix.arminco.com) |
| 09:59.13 | angryuser | cbr1000, maybe use some kind of manager to connect remotely ? or flash panel ? |
| 10:01.24 | cbr1000 | angryuser: ok. so there is another solution to connect 2 astersik together (without iax channel) allowing to use BLF? |
| 10:02.48 | angryuser | cbr1000, no flashpanel let you know the status of line/peer on remote server, |
| 10:03.05 | angryuser | cbr1000, connect you servers as you want |
| 10:03.10 | angryuser | your* |
| 10:03.31 | cbr1000 | ok |
| 10:07.51 | angryuser | cbr1000, there is a tool on asteriskguru site you can the status too, easy to delploy |
| 10:08.35 | cbr1000 | I will see |
| 10:08.37 | cbr1000 | :) |
| 10:09.47 | cbr1000 | angryuser: do you know the name of the tool? |
| 10:11.09 | angryuser | cbr1000, go to their site, it is down for a moment |
| 10:11.29 | angryuser | cbr1000, it's operato panel |
| 10:11.38 | angryuser | operator |
| 10:12.24 | cbr1000 | ok |
| 10:12.27 | cbr1000 | thanks |
| 10:13.52 | cbr1000 | angryuser: but with my hard phone, I can't use this tool to know the status of others phones? |
| 10:14.07 | angryuser | cbr1000, no |
| 10:25.40 | cbr1000 | :( |
| 10:32.56 | HumpBack | any idea on where to do bug reports on the "Asterisk The Future of Telephony" book? |
| 10:35.44 | kaldemar | HumpBack: http://oreilly.com/catalog/9780596510480/errata/ |
| 10:35.48 | *** join/#asterisk jareq (n=jarek@81.15.166.2) |
| 10:36.04 | HumpBack | kaldemar: thanks |
| 10:37.17 | *** join/#asterisk lesouvage (n=lesouvag@82.199.132.58) |
| 10:39.07 | *** join/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
| 10:40.30 | codestr0m | I'm having a brainfart. . maybe someone knows the link to the voice recognition software that was funded by darpa and sorta recently released.. there was some initial tests with telephony.. iirc asterisk specifically |
| 10:40.57 | lesouvage | I just forget how to dial from the cli. Thought it was something like "console dial(sip/300,40,t). Can somebody help me out? |
| 10:43.57 | codestr0m | found it http://cmusphinx.sourceforge.net/html/cmusphinx.php |
| 10:44.00 | *** join/#asterisk jblack (n=jblack@pool-71-181-244-67.sctnpa.east.verizon.net) |
| 10:44.31 | kaldemar | lesouvage: console dial exten[@context], without context it dials in default |
| 10:51.22 | lesouvage | kaldemar: it doesn't work. |
| 10:54.46 | kaldemar | lesouvage: yes it does. do you have chan_oss or chan_alsa loaded? |
| 10:55.12 | *** part/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
| 11:02.16 | kaldemar | lesouvage: or did you mean originate command in cli? |
| 11:10.51 | zamba | i have a problem with dmtf tones not getting "transferred" from one asterisk server to another |
| 11:10.54 | zamba | they are connected using sip |
| 11:11.15 | zamba | and i've set dmtfmode=rfc2833 for the corresponding entries in sip.conf |
| 11:11.46 | zamba | dmtf works just fine when i try a sip client connected directly to either, but when "transferring" calls from one to the other, dmtf fails |
| 11:20.08 | genji1981 | zamba: use a client that does transferring without dmtf. |
| 11:20.43 | genji1981 | zamba: SJphone, Ekiga, Zoiper |
| 11:20.47 | *** join/#asterisk Levonk (n=lk@75.62.138.76) |
| 11:21.19 | zamba | genji1981: i've tried ekiga |
| 11:21.21 | EmleyMoor | I sincerely hope you're also spelling it dtmf |
| 11:21.37 | zamba | EmleyMoor: hehe, i am :) |
| 11:22.09 | zamba | genji1981: tried both from a landline phone, x-lite and ekiga |
| 11:22.57 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 11:23.12 | genji1981 | zamba: then try SJphone. works for me, attended transfer (dial first number, then click CONF, second number, right click the first call... go to transfer, click the second call). Blind transfer, you just click the trans button. |
| 11:23.26 | genji1981 | zamba: in other words.. no dialing numbers. |
| 11:23.37 | zamba | genji1981: hm.. i'm not sure i follow you now |
| 11:24.01 | kaldemar | zamba: what did you mean by "transfer" in the first place? |
| 11:24.13 | zamba | genji1981: the point here is that only one of the * boxes have a connection to the "outside" telephone world.. |
| 11:24.47 | kaldemar | pasting your sip.conf would be helpful. |
| 11:24.56 | genji1981 | zamba: Yes.... and? as long as your outside sip source can handle more than one outgoing connection, you'll be fine. |
| 11:24.59 | zamba | so to reach extensions on *2 from pstn i have to go through the first one |
| 11:25.45 | zamba | so i have given the internal (*2) an extension on *1 |
| 11:25.54 | zamba | one sec.. pasting stuff in pastebin |
| 11:26.22 | genji1981 | zamba; dont forget to anonymise your username and password. |
| 11:26.27 | zamba | sure |
| 11:30.25 | genji1981 | zamba: what would also help, would be a more concise explaination of your definition of transfer. Whos the person transfering the call? a sip client, or a pstn? |
| 11:30.55 | *** join/#asterisk MegaIRC (n=MegaIRC@mail.marcopolo.co.za) |
| 11:31.02 | MegaIRC | hi there |
| 11:31.03 | zamba | http://pastebin.com/m5b85760e |
| 11:31.10 | zamba | see the pastebin, and you'll understand |
| 11:31.14 | zamba | hopefully |
| 11:31.38 | MegaIRC | is it possible to customize the voicemail |
| 11:31.56 | zamba | MegaIRC: in every aspect thinkable :) |
| 11:32.12 | MegaIRC | do you do it in the voicemail.conf? |
| 11:32.37 | MegaIRC | because what i want to do is |
| 11:33.00 | MegaIRC | thecaller goes to the extensions voicemail and the voicemail says |
| 11:33.44 | MegaIRC | press 1 to listen to listen to a message or 2 or 3 etc and there after they can leave a message aswell |
| 11:34.00 | MegaIRC | i just need an idea on how to approch such a task |
| 11:34.29 | EmleyMoor | MegaIRC: Do you mean you actually want to change the menu options? |
| 11:34.50 | MegaIRC | yes something like that |
| 11:35.12 | zamba | kaldemar and/or genji1981: what do you think? |
| 11:36.09 | genji1981 | zamba: still don't get ya. |
| 11:36.15 | zamba | genji1981: what do you mean? |
| 11:36.28 | zamba | genji1981: there you see what i mean by "transfer".. it's basically a dial |
| 11:36.51 | zamba | line 24 |
| 11:37.02 | genji1981 | zamba:oh... okay. just chain dialing.. |
| 11:37.29 | zamba | if that's what it's called, then yeah :) |
| 11:38.07 | MegaIRC | EmleyMoor, ? is it possible |
| 11:38.16 | genji1981 | ive got no idea. |
| 11:39.22 | MegaIRC | does any 238 people maybe know |
| 11:39.46 | MegaIRC | :) |
| 11:40.47 | kaldemar | zamba: are you 100% sure that your call matches the peer definition you have there? do you have some other dtmfmode defined under general? |
| 11:41.00 | genji1981 | goes to bed. night all |
| 11:41.52 | zamba | kaldemar: the call goes to the right asterisk, since it answers and then outputs: " -- Accepting a maximum of 4 digits." |
| 11:41.57 | kaldemar | MegaIRC: use the source, luke. :) |
| 11:41.58 | zamba | on *2 |
| 11:42.29 | kaldemar | zamba: yes it goes to the right asterisk, but does it use the right peer definition? |
| 11:42.31 | zamba | kaldemar: that's the only dtmfmode defined in the whole sip.conf, yeah |
| 11:42.35 | kaldemar | two separate things. |
| 11:43.22 | kaldemar | ok, the mode is rfc2833 by default, so it _shouldn't_ be a mode issue. |
| 11:44.10 | zamba | hm, ok |
| 11:44.49 | kaldemar | do you have a noop after the read to determine whether NUMBER has anything in it? |
| 11:45.22 | zamba | i'll try adding that, one sec |
| 11:47.36 | kaldemar | also, put the same test in *1 and dial in from pstn to test it. |
| 11:48.32 | zamba | already done that.. that works |
| 11:50.48 | zamba | <PROTECTED> |
| 11:50.48 | zamba | <PROTECTED> |
| 11:51.00 | zamba | this is from pstn to *1 |
| 11:51.04 | zamba | over zap channel |
| 11:51.19 | zamba | could it be a timing issue? the same stuff about zaptel timers? |
| 11:53.47 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
| 11:54.53 | kaldemar | zamba: don't know. have you tried different modes between *'s? |
| 11:57.06 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.176) |
| 12:00.42 | zamba | kaldemar: dtmf modes, you mean? |
| 12:01.35 | kaldemar | yes |
| 12:03.50 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-088-012.pools.arcor-ip.net) |
| 12:04.23 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:04.23 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:10.29 | *** join/#asterisk cbr1000 (n=AnthonyC@LAubervilliers-153-52-32-150.w217-128.abo.wanadoo.fr) |
| 12:19.59 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:23.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:27.22 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:28.46 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:28.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:28.59 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:28.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:29.55 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 12:30.43 | scooby2 | In queue_log I get a ENTERQUEUE when someone calls, but if they hang up immediately nothing else is added. Shouldnt something like an ABANDON? |
| 12:34.59 | [TK]D-Fender | scooby2: Yes |
| 12:35.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:39.08 | scooby2 | [TK]D-Fender: any idea why Asterisk would not do that? It seems random. Seeing it on 1.2.15 and 1.4.21.2 |
| 12:40.29 | [TK]D-Fender | scooby2: No idea. Make sure you aren't locking the file while testing as well and try it a few more times. |
| 12:41.49 | scooby2 | thats an idea |
| 12:42.23 | scooby2 | i wonder if this new version of the queuemetrics loader is locking it |
| 12:44.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:48.43 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:54.32 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 12:55.30 | zamba | [TK]D-Fender: can you help me figuring out my problem with dtmf digits not getting "transferred" from one asterisk to another? (see http://pastebin.com/m5b85760e for details) |
| 12:56.05 | zamba | there's no additional configuration in either sip.conf about dtmfmode |
| 12:56.33 | [TK]D-Fender | zamba: pastebin a failed cal |
| 12:57.52 | zamba | it's basically just like this: |
| 12:57.53 | zamba | <PROTECTED> |
| 12:57.53 | zamba | <PROTECTED> |
| 12:59.32 | [TK]D-Fender | zamba: Pastebuin the ENTIRE bloody call, and stop maksing things. |
| 13:00.27 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:01.41 | zamba | [TK]D-Fender: http://pastebin.com/m195e980e |
| 13:01.46 | lmadsen | [TK]D-Fender: stop being so belligerent |
| 13:02.33 | [TK]D-Fender | lmadsen: I ask for stuff and people do everything in their power to do everything except that. Whats the point of "helping" under these conditions? |
| 13:03.15 | [TK]D-Fender | zamba: please PB the exact peers masking only the host & password |
| 13:03.27 | *** join/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
| 13:03.37 | lmadsen | [TK]D-Fender: then don't help them. But I'm tired of looking in this room, and every single time I do, it's you being, well frankly, a dick |
| 13:05.13 | zamba | [TK]D-Fender: you just need the two peer declarations from sip.conf, yeah? |
| 13:05.18 | [TK]D-Fender | lmadsen: Maybe I should retire from helping those that must not want to be helped. |
| 13:05.23 | scooby2 | [TK]D-Fender: Any way to always make Asterisk pick up the call that has been waiting longest or is it always going to get the call that is ringing? |
| 13:05.52 | zamba | [TK]D-Fender: http://pastebin.com/m1e0464be |
| 13:06.13 | [TK]D-Fender | zamba: Every section relevent to this call, and the output from both sides. everything leading into it. Additionally something proving the remote device works local to their server as it is |
| 13:06.15 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
| 13:06.23 | Blackvel | hi guys. do you sleep even? for example [TK]D-Fender seems to be in this channel everytime I am here :) |
| 13:06.51 | [TK]D-Fender | scooby2: I don't follow you at all... |
| 13:07.19 | scooby2 | two calls in the queue. one for a minute and the other for 10 seconds |
| 13:07.19 | lmadsen | [TK]D-Fender: you can certainly do it in a more diplomatic fashion than the way you approach things. There are very often people who either aren't native english speakers, or perhaps don't exactly know what you're asking for. You can either re-ask them politely for what you're looking for, or you can just be silent if they continue to do it. But there is no need to keep being so belligerent with the majority of people you a |
| 13:07.19 | lmadsen | re trying to help. It comes across as extremely egotistical. |
| 13:07.31 | scooby2 | its random which gets answered first |
| 13:07.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.53 | lmadsen | scooby2: are you talking about a queue? |
| 13:07.57 | scooby2 | yes |
| 13:08.19 | beek | lmadsen: If you want the non-dick version of [TK]D-Fender all you need to is pay him for a consulting gig. If not, you get the dick version -- and arguably the best free support available. |
| 13:08.41 | lmadsen | scooby2: there are different methods of call delivery. Most of them would send the longest waiting call to the agent. |
| 13:08.47 | Odd_Bloke | What's the difference between sipregs and sippeers? |
| 13:09.49 | [TK]D-Fender | lmadsen: Holding back information thats requested and my having to waste my time repeating myself dosn't make me "egotistical". |
| 13:10.33 | [TK]D-Fender | beek: No, I hand out tons of help here on a regular basis. Those who are open to being helped provide the things we request of them and lead happy productive lives. |
| 13:10.49 | scooby2 | lmadsen: any tips what I should be looking for? I am not seeing anything in the queue.conf docs |
| 13:10.55 | [TK]D-Fender | Guess I should drop helping people the second I don't get what i clearly ask for to help them with. |
| 13:11.23 | lmadsen | scooby2: I guess you didn't look for 'strategy' then |
| 13:11.41 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:11.54 | Odd_Bloke | I ask because I'm looking at the source code for chan_sip.c and I can't see a great deal of difference in the way they are used. |
| 13:12.06 | *** join/#asterisk tinkerghost (n=eric@thelocalphonecompany.net) |
| 13:12.11 | lmadsen | [TK]D-Fender: well, since you're already asking them again, but in a less than ideal tone, then just ask again without the extra flavour, or that's what the 'up' button is for |
| 13:12.12 | beek | [TK]D-Fender: No argument from me. I've lurked for a few months and I'm amazed by the amount of time and assistance you provide. |
| 13:12.13 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 13:12.48 | scooby2 | lmadsen: that must be a side effect of using ringall |
| 13:13.26 | lmadsen | scooby2: shouldn't be -- the longest waiting person in the queue will ring all agents |
| 13:14.02 | [TK]D-Fender | scooby2: ringall should be calling out on behalf of the persona waiting around the longest (weighted by priority) |
| 13:14.20 | scooby2 | lmadsen: it does not seem to. It seems to take turns |
| 13:14.26 | lmadsen | beek: it's not about the amount of assistance and help provided -- it's about being more diplomatic |
| 13:14.34 | scooby2 | [TK]D-Fender: thats what I would hope for |
| 13:14.39 | tinkerghost | I am getting a horrible echo using Kiax(iax) with my Asterisk Server. Using Twinkle(sip) doesn't show that problem, but I can't use SIP from my remote location (firewall issues). IAX connections through FWD don't seem to have the echo.Any suggestions? |
| 13:14.45 | *** join/#asterisk mirrorcolor (n=mirrorco@unaffiliated/mirrorcolor) |
| 13:15.11 | lmadsen | tinkerghost: try using zoiper instead -- could be the softphone |
| 13:15.52 | [TK]D-Fender | scooby2: a thought for implementation of ringall : best when you're pretty sure people aren't going to line-up and not get answered slow. Otherwise an auto-fill RRMEMORY style method would probably get your callers answered faster with less repeat ringing. |
| 13:16.06 | zamba | [TK]D-Fender: is that more what you're after? pastebin in msg |
| 13:16.36 | scooby2 | [TK]D-Fender: thanks |
| 13:17.02 | *** join/#asterisk Levonk (n=lk@adsl-76-227-119-41.dsl.lsan03.sbcglobal.net) |
| 13:18.46 | [TK]D-Fender | scooby2: Can you please show that complete call from both sides as requested. |
| 13:22.59 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:22.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:23.34 | jaytee | mornin russellb |
| 13:23.54 | jaytee | mornin [TK]D-Fender |
| 13:23.56 | russellb | g'mornin' |
| 13:25.12 | Blackvel | how does asterisk match incoming calls to 2 sip peers, when the hosts / registered patton gateway is on the SAME ip? type=user does matching by username property and type=peer matches on ip and port? It always jumps the 2nd peer (last in sip.conf). http://pastebin.com/m25c10c6d |
| 13:26.13 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:26.37 | lmadsen | Blackvel: that is how it works -- reorder the peers |
| 13:26.50 | lmadsen | Blackvel: last peer in sip.conf will match first |
| 13:26.56 | Blackvel | i'd have on problem to use type=user config and matching by username. I just seem not to understand where username comes from. I can't it when I do sip debug (its no in the sip invite. where is it???) |
| 13:27.12 | Blackvel | lmadsen: the idea was to use two different incoming channels |
| 13:27.17 | [TK]D-Fender | jaytee: Mornin' |
| 13:27.23 | Blackvel | err 2 different incoming contexts |
| 13:27.46 | lmadsen | Blackvel: you should see it in the From header I believe, assuming the other end is sending you the username there |
| 13:28.02 | Blackvel | maybe the patton box doesnt do that |
| 13:28.18 | lmadsen | quite possible -- you might only be able to match on IP |
| 13:28.24 | Blackvel | i just can see from:public-pstn-number@ip and to:mypstnnumber@ip |
| 13:29.00 | Blackvel | and matching by username works when config is type=user AND the username is sent even if host=dynamic (when the patton gw registered on asterisk for the peer)? |
| 13:29.19 | [TK]D-Fender | Blackvel: IIRC your use of "insecure=port,invite" precludes the need to match by name therefor only leaving the IP & port to match by |
| 13:29.22 | Blackvel | lmadsen: is it only a matching on IP or is it IP+port? |
| 13:29.33 | lmadsen | Blackvel: [TK]D-Fender is right |
| 13:29.36 | ph8 | Hey all - should it be possible to make two phones ring from the same incoming number until one of them picks up? I've got the book in front of me but can't find a section on it |
| 13:29.48 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:29.53 | [TK]D-Fender | Blackvel: And thus the source of a problem. |
| 13:29.57 | Blackvel | yes my config is a mess right now |
| 13:30.02 | lmadsen | ph8: Dial(SIP/phone1&SIP/phone2) |
| 13:30.18 | Blackvel | insecure=very and insecure=port,invite is the same? |
| 13:30.18 | ph8 | that's all? |
| 13:30.20 | ph8 | that's fit :p |
| 13:30.23 | lmadsen | indeed |
| 13:30.26 | ph8 | thx |
| 13:30.27 | zamba | [TK]D-Fender: was my debugging information still too vague? |
| 13:30.30 | ph8 | voicemail's next :) |
| 13:30.34 | [TK]D-Fender | Blackvel: Yes, and you should not be doing either. |
| 13:30.35 | ph8 | I was just reading about call parking lots |
| 13:30.53 | ph8 | where john on 103 puts mary into parking lot slot 701 and then tells george on 104 that he can call 701 to get here |
| 13:31.01 | ph8 | why didn't john just transfer mary to george? |
| 13:31.21 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:31.26 | Blackvel | guys, when I leave out insecure=ports,invite then its not working *somehow* as the authentication between patton 4636 and * seem not to be working (therefore I added insecure). but of course seems to put me back into "false peer matching" |
| 13:31.30 | [TK]D-Fender | ph8: In case George isn't at his desk but will be momentarily |
| 13:32.08 | [TK]D-Fender | ph8: So you can yell out into the office "someone waiting for you on lot 701" and wherever he is he'll pick it up |
| 13:32.19 | ph8 | I see |
| 13:32.32 | ph8 | From my question above, i want to ring one SIP phone and ring my mobile on IAX2 |
| 13:32.51 | Blackvel | if matching by name does not work in my config with patton, shall I use type=peer and host=IP and use two different ports for the both peers? will it fix it? can't I leave insecure=invite then? |
| 13:32.52 | ph8 | is there a way for me to save my IAX2 details so they're not duplicated everywhere? - Can i do like OUTBOUND1/numberhere |
| 13:32.52 | lmadsen | ph8: sometimes George doesn't have his own extension to transfer to -- or maybe he wanders around the warehouse, so he can use any phone to pick it up |
| 13:32.58 | ph8 | makes sense |
| 13:33.17 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 13:33.18 | [TK]D-Fender | ph8: That can be problematic. You cell might hit VM and be considered "answered" and thus deprive you of being able to answer it on SIP |
| 13:33.18 | lmadsen | ph8: see global variables |
| 13:33.57 | [TK]D-Fender | Blackvel: "insecure=no" <- if you don't do this then it will not look for the unique name you gave each port. |
| 13:35.16 | Blackvel | unique username? |
| 13:35.23 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-17-25.revip2.asianet.co.th) |
| 13:35.25 | [TK]D-Fender | Blackvel: Exactly |
| 13:35.44 | Blackvel | and if patton doesn't send it (somehow I seem to have maybe register or authentication problems) |
| 13:36.08 | lmadsen | Blackvel: then as you said, you have authentication problems -- because the other end must not be sending you the username you are expecting |
| 13:36.14 | [TK]D-Fender | Blackvel: "insecure" = don't bother authenticating. You are bypassing it with this choice. |
| 13:36.29 | [TK]D-Fender | Blackvel: just try without it. |
| 13:36.32 | lmadsen | Blackvel: registrations will have nothing to do with your authentication -- registration is only used to tell the other end where you are located |
| 13:36.40 | Blackvel | looks to me that maybe I have to use it |
| 13:36.42 | Blackvel | lmadsen: i see |
| 13:36.52 | Blackvel | its a whole mess as I said :) |
| 13:37.02 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 13:37.13 | ph8 | [TK]D-Fender: VM? |
| 13:37.14 | Blackvel | insecure=invite bypasses host+port matching always? |
| 13:37.19 | ph8 | lmadsen: thx |
| 13:37.21 | [TK]D-Fender | ph8: Voicemail |
| 13:37.42 | ph8 | good point |
| 13:37.43 | [TK]D-Fender | Blackvel: No, it bypasses asking WHO is calling. |
| 13:37.46 | ph8 | very good point |
| 13:38.02 | Blackvel | isn't there any option like peer1 host=ip, port=5060, auth off, peer2 host=ip, port=5062, auth off |
| 13:38.07 | ph8 | i might turn voicemail on ringout off on my cell |
| 13:38.09 | [TK]D-Fender | ph8: Any kind of multi-dail involving PSTN is a tricky thing |
| 13:38.11 | ph8 | not the most useful thing ever |
| 13:38.32 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 13:38.37 | [TK]D-Fender | Blackvel: No, really not a good idea to think of it either. |
| 13:38.50 | [TK]D-Fender | Blackvel: What don't you just stop right now and change it and see what happens. |
| 13:38.57 | Blackvel | so insecure=no MUST check for username. what happens when this /%&" patton gw doesn't send a username on INVITE? (only on register) |
| 13:39.15 | Blackvel | I will [TK]D-Fender of course |
| 13:39.40 | Blackvel | must I use two different ports or can it be default 5060 for correct matching? |
| 13:40.08 | [TK]D-Fender | Blackvel: depends on how it processes calls. Just try it one step at a time and stop worrying about it. |
| 13:42.19 | Blackvel | for sip debug: whats the packet called to include the username? INVITE and from header? |
| 13:43.10 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
| 13:44.07 | *** join/#asterisk moy (n=moy@189.169.91.147) |
| 13:44.18 | [TK]D-Fender | Blackvel: look at the whole convo invluding proxy auth |
| 13:44.30 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 13:47.01 | Blackvel | is there somewhere on the internet a big picture how sip works? |
| 13:47.16 | Blackvel | like what astersk sends to patton and what patton sends to asterisk (for invite, username and auth)? |
| 13:50.29 | Blackvel | Found peer 'patton_ntba' |
| 13:50.41 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) |
| 13:50.51 | Blackvel | Sending to 192.168.1.4 : 5062 - SIP/2.0 407 Proxy Authentication Required |
| 13:51.06 | Blackvel | no package is sent back to asterisk with any username or something like that |
| 13:51.50 | Blackvel | ah sip error on patton |
| 13:52.04 | Blackvel | <PROTECTED> |
| 13:52.18 | Blackvel | well of course not |
| 13:52.24 | Blackvel | its only a pstn number nothing more |
| 13:52.28 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 13:53.29 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-181-171.dsl.stlsmo.sbcglobal.net) |
| 13:54.15 | LemensTS | anyone do paging on speakerphones with polycom phones? |
| 13:56.10 | [TK]D-Fender | LemensTS: Yup. |
| 13:56.48 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 13:57.15 | ph8 | is the [internal] dial plan special, or just an example used by the book? |
| 13:57.25 | ph8 | so could i use [companyname-internal] to the same effect? |
| 13:58.37 | LemensTS | TK: Where do you suggest I look up information on it, and how to do it? The cmd page in the cli? |
| 13:58.59 | Blackvel | [TK]D-Fender I can remember that ManxPower said something like that he heard or knows that sip gateways do not support outbound authentication (even its configurable in the software) |
| 13:59.12 | [TK]D-Fender | LemensTS: Yup. |
| 13:59.13 | *** join/#asterisk mog (n=mog@nat/digium/x-c7adc5f58c3463cf) |
| 13:59.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:59.40 | [TK]D-Fender | Blackvel: I would not say that of all. None that I have worked with required open-ended calls. |
| 14:00.00 | [TK]D-Fender | ph8: jsut an example |
| 14:00.07 | ph8 | cool |
| 14:00.31 | [TK]D-Fender | ph8: On special ones are [globals] and 1 other |
| 14:00.35 | [TK]D-Fender | only* |
| 14:00.48 | LemensTS | TK: Do you know if all the polycoms support this? |
| 14:01.19 | [TK]D-Fender | LemensTS: yup |
| 14:02.23 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 14:02.44 | LemensTS | TK: awesome thanks for the help, I will test this |
| 14:03.01 | ph8 | i've got 666 going to VoiceMailMain() |
| 14:03.01 | *** join/#asterisk Drasha (n=Drasha@faster11.westcom.cz) |
| 14:03.09 | ph8 | and it misses the first word |
| 14:03.09 | *** part/#asterisk Drasha (n=Drasha@faster11.westcom.cz) |
| 14:03.28 | ph8 | something like 'plebian mail, mailbox <pause>..... <password>......login incorrect......<mailbox> |
| 14:03.30 | ph8 | what am i missing? |
| 14:03.49 | *** join/#asterisk kalel008 (n=kalel008@mail.marcopolo.co.za) |
| 14:04.06 | kalel008 | hi |
| 14:04.10 | Blackvel | [TK]D-Fender: interesting. I am too new to those gateways. but its seems to be one of the echo free solution.... |
| 14:04.31 | kalel008 | is it possible to edit the voicemail menus and if yes were do i do that |
| 14:05.21 | Blackvel | [TK]D-Fender: where should patton but in the username (sip invite)? from? its nowhere. from/to/via only has the external pstn numbers. no username. |
| 14:05.51 | *** join/#asterisk [netman] (n=netman@20.Red-88-25-137.staticIP.rima-tde.net) |
| 14:06.05 | Blackvel | therefore the patton gw logs: NO AUTHENTICATION CREDENTIALS FOUND. User: sip:calling-pstn-number@192.168.1.4:5062, realm: asterisk |
| 14:06.27 | [netman] | hi all |
| 14:06.36 | Blackvel | so username is missing |
| 14:07.07 | kalel008 | any ideas |
| 14:07.53 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:08.04 | [netman] | I need a simple prepaid system, anybody could suggest one? |
| 14:08.06 | dandre | hello, |
| 14:09.47 | *** join/#asterisk Levonk (n=lk@adsl-76-230-110-52.dsl.lsan03.sbcglobal.net) |
| 14:10.53 | dandre | I have registered my asterisk to two accounts on the same provider (account1 and account2). If I receive calls from account1 all is fine but I can't properly receive calls from account2. All is shown as if the call comes from account1. |
| 14:11.49 | Blackvel | dandre: problem of matching probably |
| 14:11.54 | Blackvel | any insecure settings? :) |
| 14:12.03 | Blackvel | I have the same problem |
| 14:12.21 | dandre | yes I had to set insecure to something |
| 14:12.26 | dandre | ... |
| 14:12.51 | Blackvel | what settings? |
| 14:12.59 | Blackvel | very? |
| 14:13.03 | dandre | I have this: |
| 14:13.03 | dandre | insecure = port,invite |
| 14:13.21 | Blackvel | do you use any port settings for both peers? |
| 14:13.31 | dandre | no |
| 14:14.03 | Blackvel | then it matches by ip |
| 14:14.21 | Blackvel | and if both (same provider) have the same ip it looks to be a problem |
| 14:14.28 | dandre | ah ok that's the reason |
| 14:14.34 | Blackvel | http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer (there are some dicussions on that) |
| 14:14.59 | Blackvel | cant you do insecure=no like [TK]D-Fender suggests? |
| 14:15.15 | Blackvel | then it authenticates/chooses by username if type=user |
| 14:15.38 | phpboy | how do I check what PRI card is in the asterisk server? |
| 14:15.39 | ph8 | why do phones come with pc ethernet ports? is there some popular pc software i don't know about that will integrate with my grandstream? |
| 14:15.40 | Blackvel | maybe also for type=peer (not sure about that. voip-info documents username by type=user |
| 14:15.41 | phpboy | model, etc.. |
| 14:15.50 | Blackvel | lspsi? |
| 14:15.54 | Blackvel | lspci |
| 14:18.13 | Blackvel | WOWIEEEEEEEEEEEE! |
| 14:18.15 | Blackvel | its working |
| 14:18.17 | Blackvel | insecure=invite |
| 14:18.23 | Blackvel | and i am done |
| 14:19.30 | dandre | is there a Can I use type = user or is it strongly discouraged? |
| 14:19.54 | Blackvel | there seem to be so many changes in 1.2 and 1.4 |
| 14:20.05 | Blackvel | voip-info discourages user. but i wouldn't care |
| 14:20.18 | *** join/#asterisk hatoon (n=musis@189.71.108.19) |
| 14:20.22 | lmadsen | has been using 1.4 successfully since 1.4.0 :) |
| 14:20.37 | lmadsen | although I've been lucky that I've been able to get many bugs I found fixed very quickly, and lately, I don't really find any |
| 14:20.46 | ph8 | any idea how i can debug my voicemail? i've managed to ring and leave a message in 101@company - but my phone can't login at the 'mailbox' and 'password' prompt, presumably i enter 101 and my PIN (1234) |
| 14:20.47 | ph8 | ?? |
| 14:21.09 | Blackvel | lmadsen: latest 1.4.30.2 segfaults with patton box. wanted to use, but cant. i am on 1.2.x latest now |
| 14:21.16 | lmadsen | ph8: be sure when you're using Voicemail() that you have Voicemail(101@company) as otherwise you are logging into default |
| 14:21.28 | ph8 | yeh i do :/ |
| 14:21.32 | ph8 | i might pastebin my config? |
| 14:21.33 | Blackvel | dandre: i am using for both [peer_sectinos] type=friend |
| 14:21.43 | Blackvel | not even type=user |
| 14:21.45 | ph8 | it's basically what the book says |
| 14:21.51 | Blackvel | trying with type=peer NOW |
| 14:22.17 | lmadsen | Blackvel: have you filed a bug with a backtrace after reproducing the issue and having the DONT_OPTIMIZE flag turned on? |
| 14:22.41 | lmadsen | type=friend is a shortcut to having both a user and a peer |
| 14:22.46 | lmadsen | asterisk will try to match on user first, and if not found, will try to match on IP (peer) |
| 14:22.52 | [netman] | there are a lot of billing & prepaid applications on voip-info.org... I would like to hear the experience of anybody who has tested any |
| 14:23.52 | phpboy | Blackvel: thanks :D |
| 14:23.54 | Blackvel | lmadsen: not yet for bug. but I will :) |
| 14:24.14 | Blackvel | dandre: ok. type=peer works for both too |
| 14:24.28 | lmadsen | Blackvel: just come find me in #asterisk-bugs if you need any help as I'm a bug marshall |
| 14:24.55 | Blackvel | i didn't set port for any of them. but you have to know, that the patton defines two gateways (for two peers). one sits on 5060, the other on 5062. |
| 14:25.03 | lmadsen | I'd rather you file the bug report with all the right information the first time :) |
| 14:25.10 | Qwell | throws dry spaghetti at lmadsen, and runs off |
| 14:25.19 | lmadsen | cooks the spaghetti |
| 14:25.22 | lmadsen | free food |
| 14:25.41 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 14:26.05 | Blackvel | it may happen that this is the reason that it works without peer port configuration (one peer registers to ip:5060, the other to ip:5062). i am happy that invite works in that scenario. insecure=invite,port (very) not as [TK]D-Fender pointed out |
| 14:26.23 | Blackvel | dandre: so try insecure=invite :) |
| 14:28.02 | Blackvel | lmadsen: prolly i do it on weekends |
| 14:29.17 | Blackvel | lmadsen: have this gdb backtrace: http://pastebin.com/m730413a7 is that enough? :) |
| 14:30.10 | Blackvel | whats the benefit of type=friend over type=peer in 1.2.x? |
| 14:30.20 | lmadsen | Blackvel: that looks good -- looks like you already have DONT_OPTIMIZE enabled. Also do 'bt full' in addition, then output that via 'tee' to a file so you can attach the file to the bug (don't post pastebin links to the bug tracker) |
| 14:30.41 | lmadsen | Blackvel: no benefit other than that peer is only matched by IP, not username |
| 14:30.42 | dandre | ok I'll try |
| 14:30.56 | Blackvel | lmadsen: ip + port |
| 14:31.11 | lmadsen | right :) |
| 14:31.32 | Blackvel | lmadsen: i am just thinking about (what port) as the 2 peers have no special port settings. only patton uses sipgw1 5060 and sipgw2 5062 |
| 14:31.37 | Blackvel | so its the destination port? |
| 14:31.47 | Blackvel | and then there is peer port=5062 setting |
| 14:32.06 | Blackvel | does that influence matching by ip+port? |
| 14:32.11 | lmadsen | makes sense |
| 14:32.42 | Blackvel | as i told my config is a mess (at least my brain) |
| 14:32.47 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 14:32.52 | *** join/#asterisk drcode (n=user1@85.65.12.156.dynamic.barak-online.net) |
| 14:32.54 | drcode | hi all |
| 14:33.12 | Blackvel | because I really tested setting [peer2] to port=5062. and then patton sipgw had destination 192.168.1.2:5062 |
| 14:33.13 | drcode | Is there possible to use astrikes with fxo and analog modem? |
| 14:33.23 | Blackvel | and it was not working .... maybe I should retry |
| 14:33.38 | Blackvel | lmadsen: do you know of any good reason where there HAS TO BE a port=5062 for a peer? |
| 14:33.49 | lmadsen | drcode: nope -- only special modems that have the right chipset |
| 14:34.04 | Blackvel | wishes to make [TK]D-Fender happy and get patton to do user authentication for invite with insecure=no :) |
| 14:34.11 | lmadsen | Blackvel: not really... I suppose only when the other end point expects that to be the port |
| 14:34.23 | lmadsen | ~x100p |
| 14:34.23 | jbot | extra, extra, read all about it, x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 14:34.33 | ph8 | hey all - this is my voicemail config -> http://pastebin.com/m15b2875b <- can anyone tell me why it's not working, i can leave a message, and my hardphone registers it (red light flashes) but when i dial 666 and enter 101 at the mailbox prompt, then 1234 i get 'login incorrect' |
| 14:34.34 | drcode | there are modem that can answer to FXO and ppp ? |
| 14:34.43 | drcode | pstn? |
| 14:35.04 | Blackvel | lmadsen: and [peer2] type=peer port=5062 really opens socket, so patten can contact asterisk to ip:5062 instead of ip:5060 (general)? |
| 14:35.08 | lmadsen | drcode: some with a certain motorola chipset, but they don't work very good at all |
| 14:35.21 | drcode | also I have t100p can it answer to voice and fax , with iaxmodem? |
| 14:35.36 | drcode | x100p sorry |
| 14:35.46 | lmadsen | drcode: technically, yes it should |
| 14:35.54 | drcode | and data/ |
| 14:35.57 | lmadsen | but it won't be all that reliable |
| 14:36.05 | drcode | I understand it can be use only 9600 in data? |
| 14:36.22 | *** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net) |
| 14:37.35 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 14:38.20 | Kobaz | okay so i have an aastra phone, i place a call, hit conference, place another call, his conference, now all three channels are linked, i hang up the originator of the conference, and all channels hang up |
| 14:38.52 | Kobaz | now, this is only on one system... if i do the same thing over here, the two legs are bridged when the middle hangs up, and those two can still talk... i'm not sure what the difference is |
| 14:39.02 | Kobaz | i dont know if it's an aastra setting or an asterisk setting |
| 14:40.27 | *** join/#asterisk ManxPower (n=manxpowe@245.sub-70-220-211.myvzw.com) |
| 14:40.38 | dandre | How can I see wheather a sip endpoint is registered as peer or friend? |
| 14:42.54 | ManxPower | Is there a way to remove all the sip registration info from Asterisk's cache? |
| 14:43.11 | ph8 | dandre: 'asterisk -r' then 'show sip peers' |
| 14:43.13 | ManxPower | (i.e. Asterisk still knows the IPs of it's registered peers even after a reboot) |
| 14:43.18 | Blackvel | hey ManxPower: i have my solution. insecure=invite. |
| 14:43.22 | ph8 | dandre: Sorry 'sip show peers' |
| 14:43.30 | ManxPower | Blackvel: that is not a solution, that is a hack,. |
| 14:43.32 | dandre | yes |
| 14:43.45 | Blackvel | ManxPower: didn'T you tell me yesterday that patton sip gateway does not do outbound authentication for some reason? |
| 14:43.52 | russellb | ManxPower: clear it out of the astdb |
| 14:43.56 | dandre | where should it show friend? |
| 14:44.00 | Kobaz | ManxPower: oooo you're herer |
| 14:44.07 | Kobaz | you got in just after my question |
| 14:44.10 | Blackvel | tried with [TK]D-Fender insecure=no and its not working |
| 14:44.11 | ManxPower | russellb: what's the family/key or whatever it's called? |
| 14:44.20 | russellb | ManxPower: don't remember, but i'll look real quick |
| 14:44.21 | Kobaz | i have an aastra phone, i place a call, hit conference, place another call, his conference, now all three channels are linked, i hang up the originator of the conference, and all channels hang up |
| 14:44.24 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:44.24 | ManxPower | Blackvel: It may be your only option, but it is still an ugly hack. |
| 14:44.33 | Blackvel | there is the username missing. it doesn't know for what user it should do authentication |
| 14:44.33 | russellb | ManxPower: SIP/Registry |
| 14:44.46 | Blackvel | no sip paket has the username in it |
| 14:44.47 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 14:45.00 | Blackvel | dunno what to do. I will send email to patton directly about it |
| 14:45.45 | ManxPower | russellb: can I just kill the entire SIP tree? |
| 14:45.49 | *** join/#asterisk drcode (n=user1@85.65.12.156.dynamic.barak-online.net) |
| 14:45.49 | Blackvel | ManxPower: to be honest. i only need to have a working solution right now. its always time for improvements (if they work in general) :) |
| 14:45.59 | russellb | ManxPower: so something like ... *CLI> database deltree SIP Registry |
| 14:46.21 | ManxPower | russellb: *nod* What OTHER keys are in SIP/? |
| 14:46.41 | russellb | none in the asterisk 1.4 chan_sip i'm looking at :) |
| 14:46.52 | Kobaz | noone has any idea? :( |
| 14:46.57 | russellb | so killing all of SIP is fine, too |
| 14:48.12 | ManxPower | russellb: the IPs still show in sip show peers |
| 14:48.12 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 14:48.24 | russellb | you killed the tree and restarted? |
| 14:48.35 | ManxPower | russellb: Just did, that cleared it. |
| 14:48.41 | russellb | ah, gotcha |
| 14:49.45 | *** join/#asterisk esaym-acer (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 14:50.05 | ph8 | hey all - this is my voicemail config -> http://pastebin.com/m15b2875b <- can anyone tell me why it's not working, i can leave a message, and my hardphone registers it (red light flashes) but when i dial 666 and enter 101 at the mailbox prompt, then 1234 i get 'login incorrect' |
| 14:50.16 | *** join/#asterisk seanmh (i=seanmh@216.31.101.121) |
| 14:50.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:51.28 | *** join/#asterisk scampbell (n=scampbel@mail.contplastics.com) |
| 14:51.48 | lowtek | Yes, your context is mine-internal where your voicemail context is just 'mine'. |
| 14:51.58 | lowtek | Try 666,1,VoiceMailMain(|mine) |
| 14:52.09 | Kobaz | blarg |
| 14:52.13 | lowtek | And of course, the mark of the beast thing will probably have you rotting in hell ... |
| 14:52.16 | Kobaz | my conferencing is all borken |
| 14:52.20 | Kobaz | borken i say! |
| 14:52.32 | esaym-acer | I have 2 asterisk boxes linked together with iax. One box works fine always, the other one will loose the iax connection after a few days and will require a restart. The logs display iax2 no route to destination. The host= parm in iax.conf is a domain name so I am wondering if the asterisk dns stops working or something. The other box is set up the same and it never has any problems though |
| 14:52.56 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 14:53.08 | ManxPower | esaym-acer: use an IP address and see what happens. Asterisk SUCKS when it comes to DNS |
| 14:53.14 | Kobaz | ManxPower: haha |
| 14:53.18 | Kobaz | ManxPower: yeah |
| 14:53.37 | Kobaz | esaym-acer: 'iax2 reload', should fix that |
| 14:53.38 | *** join/#asterisk spokra (n=spokra@75-92-217-76.sea.clearwire-dns.net) |
| 14:53.45 | Kobaz | ManxPower: i do an iax2 reload once an hour to fix any dns problems |
| 14:54.09 | Kobaz | ^--> esaym-ace |
| 14:54.15 | esaym-acer | ManxPower: yes I think IP address fixes it, but still, the other box has a domain name for a host and it has never had a problem? |
| 14:54.52 | ManxPower | esaym-acer: if DNS goes down for even 1 second bad things happen if you use hostnames in asterisk |
| 14:55.10 | ManxPower | also make sure /etc/hosts contains IP and name for all interfaces on the system |
| 14:55.11 | esaym-acer | Kobaz: how did you automate iax2 reload? |
| 14:55.15 | Kobaz | esaym-acer: cron |
| 14:55.32 | esaym-acer | hmm |
| 14:55.35 | Kobaz | ManxPower: if the box doesn't have a default gateway, asterisk will barf as well |
| 14:55.42 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 14:55.48 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 14:55.50 | esaym-acer | no it has that |
| 14:55.57 | lowtek | esaym-acer: You could do a cron job every so often that does an asterisk -rx "iax2 reload" or whatever |
| 14:55.58 | ManxPower | Kobaz: that is because without a default gateway DNS won't usually work |
| 14:56.03 | Kobaz | anyways, anyone wanna help with my phone conferencing problem |
| 14:56.12 | Kobaz | ManxPower: probably... but! |
| 14:56.43 | Kobaz | ManxPower: it doesn't seem to do with dns when you lack a default gw... it will fail on creating a sip channel without a default gw, even if you have an ip as the remote host |
| 14:56.44 | *** join/#asterisk boch (n=fran@customer191-9.iplannetworks.net) |
| 14:57.10 | esaym-acer | Kobaz: so in cron: asterisk -rx iax2 reload ? |
| 14:57.23 | Kobaz | esaym-acer: pretty much... |
| 14:57.30 | Kobaz | except you need quotes |
| 14:57.36 | Kobaz | asterisk -rx "iax2 reload" |
| 14:57.38 | esaym-acer | ok, didn't know I could inject cli like that |
| 14:57.42 | esaym-acer | oh |
| 14:57.43 | *** part/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
| 14:57.53 | esaym-acer | thanks |
| 14:57.57 | esaym-acer | rofl |
| 14:58.02 | ManxPower | esaym-acer: there are many things you don't know |
| 14:58.06 | Kobaz | heh |
| 14:58.10 | Kobaz | pokes ManxPower |
| 14:58.27 | lowtek | MCCAIN/MILF '09 |
| 14:58.54 | ph8 | lowtek: Same issue unfortunately :/ |
| 14:58.58 | esaym-acer | ManxPower: yes of course, I am nothing but poor farmer and my land under big drought |
| 14:59.12 | lowtek | ph8: did you reload your configs? |
| 14:59.24 | Kobaz | ManxPower: any idea about my conference issue? |
| 14:59.35 | ph8 | lowtek: restarted asterisk yeh |
| 14:59.37 | Kobaz | ManxPower: i know i'm killing you, but i've been beating my head against a wall |
| 15:00.03 | lowtek | Just make your voicemail context match the context you're calling from. |
| 15:00.11 | ManxPower | Kobaz: I am working on restoring 15 circuits for my customers affected by Gustav. I am here to have MY questions answered today. |
| 15:00.17 | Kobaz | ManxPower: fine fine :P |
| 15:00.33 | [TK]D-Fender | Kobaz: What you have describe for "conference" is a 3-way call. |
| 15:00.37 | Kobaz | yeah |
| 15:01.03 | ph8 | lowtek: so in voicemail.conf i'll use mine-voicemail and in extensions.conf i'll call it mine-voicemail as well (which takes 666 and calls voicemailmain()) |
| 15:01.04 | lowtek | ManxPower: No, it doesn't make it longer ... Yes, you will go blind ... And Yes, it actually does matter |
| 15:01.17 | [TK]D-Fender | Kobaz: On Aastra when you are the one bridging the parties and you hangup, it disconnects them. Thats jsut the way it work. If you want tthem to keep talking you'd have to transfer each of them to a Meetme or something. |
| 15:01.30 | Kobaz | [TK]D-Fender: not with the aastra i have here |
| 15:01.32 | lowtek | ph8: Yep. Also take a look at the docs for voicemailmain() |
| 15:01.55 | [TK]D-Fender | Kobaz: Now Polycom is so much nicer in that it forces a bridge between these separate channels when you hang up. |
| 15:01.56 | Kobaz | [TK]D-Fender: on the same model aastra, on a different asterisk box, it disconnects |
| 15:01.58 | ph8 | still get login incorrect :/ |
| 15:02.04 | [TK]D-Fender | Kobaz: Polycom call handling > ALL |
| 15:02.08 | Kobaz | yeah |
| 15:02.11 | lowtek | ph8: repost your config |
| 15:02.20 | *** join/#asterisk angryuser (n=angryuse@245.220.20.81.dynamic.adsl.abo.nordnet.fr) |
| 15:02.30 | angryuser | ~book |
| 15:02.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:02.32 | ph8 | thx |
| 15:02.36 | Kobaz | but the weird thing, is *this* aastra causes the bridging after hangup |
| 15:03.24 | *** join/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
| 15:03.49 | Kobaz | why i'm not sure, same model, same asterisk version, same aastra configs, mostly the same asterisk configs |
| 15:04.03 | Kobaz | i'm trying to figure out what's different... maybe it's the firmware version |
| 15:05.24 | lowtek | ph8: voicemailmain(${EXT}@whatevervoicemailcontextmailboxexistsin) |
| 15:05.30 | *** join/#asterisk bmg505 (n=leon@196-209-78-72-tbnb-esr-2.dynamic.isadsl.co.za) |
| 15:05.34 | Kobaz | [TK]D-Fender: it's wiggity |
| 15:06.05 | ManxPower | ph8: if you do not specify a context when running Voicemailmain then it will not work. put the voicemail context in it. |
| 15:06.19 | [TK]D-Fender | Kobaz: Aastra firmware isn't just flakey, thhhheeeerrrr GGGRRRREEEAATT! |
| 15:06.20 | ManxPower | in previous versions it would default to "default", but not anymore. |
| 15:06.35 | ph8 | ah ok - that's not what the book said |
| 15:06.37 | ph8 | ahh |
| 15:06.45 | Kobaz | [TK]D-Fender: haha |
| 15:06.49 | ManxPower | ph8: the book is for the CVS version between 1.2 and 1.4 |
| 15:07.03 | Kobaz | ooooh |
| 15:07.10 | Kobaz | [TK]D-Fender: there's newer firmware on the other phone |
| 15:07.16 | Kobaz | [TK]D-Fender: the phone that doesn't behave |
| 15:07.20 | [TK]D-Fender | ph8: "CORE SHOW APPLICATION [WHATVER]" <- THIS IS THE FIRST PLACE YOU SHOULD LOOK EVERY TIME. |
| 15:07.24 | [TK]D-Fender | darn caps. |
| 15:07.25 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:07.43 | jeev | wow, fender is yellin |
| 15:07.47 | [TK]D-Fender | Kobaz: So now you have only the latest, most fashionable issues :) |
| 15:07.53 | Kobaz | [TK]D-Fender: yeah :) |
| 15:08.02 | ph8 | ${EXT} <--- is what's dialed right? So if i'm using 666 to access my 101 mailbox, it won't work? |
| 15:08.40 | [TK]D-Fender | ph8: pass it the number you want to pass it. You should know what that is already. if you want to prompt for the box, then don't pass it anything. |
| 15:08.41 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 15:08.42 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:08.52 | [TK]D-Fender | ph8>${EXT} <--- is what's dialed right? <- no. |
| 15:09.07 | ManxPower | ph8: No. That was a test to see if you understand what is supposed to be there. |
| 15:09.17 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-befe7c5b8d767fc4) |
| 15:09.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:09.18 | ph8 | i thought it was an alias for ${EXTEN} |
| 15:09.22 | ManxPower | either that or lowtek is an idiot. |
| 15:09.23 | ph8 | that i didn' tknow about |
| 15:09.34 | ManxPower | EXT is NOT an alias for EXTEN |
| 15:09.35 | ph8 | so i have: exten => 666,1,VoiceMailMain(101@mine-voicemail) |
| 15:09.54 | [TK]D-Fender | ph8: pastebin it all.... |
| 15:09.54 | ManxPower | ph8: and you have a [mine-voicemail] section in voicemail.conf and you did a reload? |
| 15:10.20 | ph8 | and in voicemail.conf [mine-voicemail] 101 => 1234,Henri,henri@dom.co.uk,henri-mobile@dom.co.uk |
| 15:10.25 | ph8 | i'm restarting asterisk between all changes |
| 15:10.27 | ph8 | it's not a live system |
| 15:10.33 | ph8 | well not a production system |
| 15:10.44 | ph8 | now when i dial 666 it skips the mailbox prompt and goes straight to password |
| 15:10.49 | ph8 | but when i enter 1234 it says login incorrect |
| 15:10.56 | [TK]D-Fender | ph8: because you are telling it which box. |
| 15:11.01 | ph8 | could it be something with how the numbers are being entered? can i check what voicemail sees? |
| 15:11.02 | ManxPower | ph8: as is documented in "core show application voicemailmain" |
| 15:11.03 | ph8 | yeh i get that |
| 15:11.18 | [TK]D-Fender | ph8: "restarting" Asterisk how? |
| 15:11.18 | ManxPower | ph8: I suspect you really have a DTMF issue. |
| 15:11.44 | ph8 | [TK]D-Fender: /etc/init.d/asterisk restart (Ubuntu 8.04) |
| 15:11.54 | ph8 | ManxPower: Serverside? |
| 15:11.54 | scooby2 | not to bug you again but is there anything here that would be causing people in the queue to get pushed to the back? maybe the retry=5? http://pastebin.com/m2c698f10 |
| 15:12.05 | [TK]D-Fender | ph8: Ok, please pastebin your voicemail.conf & complete CLI output of the entire attempt |
| 15:12.08 | sacitec | good morning, i'm working with server HP Proliant ML110G4 with asterisk 1.4. I will start working with g729 audio codec from digium. On the download page there are several versions for x86 32 bits processor. I have a 'pentium 912 Dual Core 1.86', what's the best choice for my server ? i586 or i686 ? |
| 15:12.41 | Qwell | sacitec: i686 will definitely work, but it won't be the most performant. try a few |
| 15:12.49 | ManxPower | ph8: pbx-1*CLI> core show application voicemailmain |
| 15:13.02 | ManxPower | try it you'll like it. first hit is free |
| 15:13.13 | ph8 | [TK]D-Fender: heh |
| 15:13.17 | sacitec | so, what do you suggest, i586 then ? |
| 15:13.19 | *** join/#asterisk Levonk (n=lk@adsl-76-237-15-59.dsl.lsan03.sbcglobal.net) |
| 15:13.21 | ph8 | i'll try skipping pass |
| 15:13.25 | Qwell | sacitec: no, that's going backwards.. |
| 15:13.33 | [TK]D-Fender | ph8: Don't. |
| 15:13.39 | scooby2 | this is killing me. One person in the queue for 5 minutes, one in the queue for 3 minutes, and the person that just dialed in got answered. |
| 15:13.56 | ph8 | :o ok? |
| 15:14.59 | Kobaz | [TK]D-Fender: oh |
| 15:15.10 | Kobaz | [TK]D-Fender: i should make sure of what's going on before i get too deep |
| 15:15.25 | ManxPower | ph8: Did you install from a package or from source? |
| 15:15.25 | Kobaz | [TK]D-Fender: since the buttons on this phone aren't labeled... i was using transfer and not conference |
| 15:15.44 | Kobaz | [TK]D-Fender: you're very correct as usual... three way calling works like that on aastra |
| 15:15.47 | Kobaz | :P |
| 15:15.51 | ph8 | ManxPower: package |
| 15:15.53 | [TK]D-Fender | Kobaz: Grab a ClueBat and do it yourself... I'm too tired... |
| 15:16.01 | ManxPower | ph8: you poor, poor thing. |
| 15:16.06 | Kobaz | [TK]D-Fender: no no, i'm saying i figured it out |
| 15:16.14 | ManxPower | best of luck with your problem. |
| 15:16.31 | Odd_Bloke | I'm currently trying to get Asterisk Realtime working for sippeers and sipusers (Asterisk 1.4). I've set the database up, and inserted a row. Everything seems to look as it should: http://paste.pocoo.org/show/84363/ but I'm not seeing it in 'sip show peers'. |
| 15:16.33 | [TK]D-Fender | Kobaz: I know. |
| 15:16.37 | ph8 | you can't seriously be implying source is better than a nice package? |
| 15:16.38 | Odd_Bloke | Any ideas on how to go about debugging this? |
| 15:16.44 | Kobaz | so i need to write some custom crap here... need to transfer the channels to a meetme |
| 15:17.02 | [TK]D-Fender | ph8: We wouldn't.... if there were such a thing as a "nice package" |
| 15:17.09 | ManxPower | ph8: no implication required. It is a FACT that installing from source is the only way we can really help you. |
| 15:17.43 | ManxPower | ph8: packages are great for software that does not change much. Asterisk, SpamAssassin, ClamAV, are all software that change too much to really install from package. |
| 15:17.43 | ph8 | because source is bleeding edge? |
| 15:17.47 | [TK]D-Fender | ph8: Well not the only way, though that might minimize surprises. You could start by actually providing what we requested. |
| 15:17.54 | ManxPower | In any case you need to talk to the person that built the package for support. |
| 15:17.57 | ph8 | getting there, i was just trying other settings |
| 15:18.12 | ph8 | to provide debug output, should i be setting verbose=10 and debug=3 |
| 15:18.12 | [TK]D-Fender | ph8: No, because packages are assembled by less than expert people using older versions. |
| 15:18.24 | ManxPower | ph8: no, because almost everyone that builds packages for Asterisk other than tzafrir_laptop is a complete and total moron. |
| 15:18.29 | [TK]D-Fender | ph8: sounds good to start |
| 15:18.48 | ph8 | you do miss out on the latest :/ i'm on 1.4.17 (with some ubuntu patchwork i think) |
| 15:19.10 | ManxPower | ph8: there have been like 400 bug fixes between your version and the current 1.4.22.2 |
| 15:19.43 | ManxPower | As I said, best of luck with that. I'm going to go back to doing things that are not futile. |
| 15:20.03 | ph8 | but is one of those bugs something stopping voicemail from working? |
| 15:20.07 | ph8 | doubtful i would guess |
| 15:20.13 | ph8 | anyway just putting paste together |
| 15:21.05 | Kobaz | okay, back to umm... work |
| 15:22.08 | ph8 | DTMF does appear to be the issue fender |
| 15:22.18 | ph8 | -- Incorrect password '' for user '101' (context = mine-voicemail) |
| 15:22.21 | ph8 | when i definitely dialed 1234 |
| 15:22.41 | dandre | how can I unregister a sip peer? |
| 15:22.41 | Kobaz | ph8: relaxdtmf=yes |
| 15:22.51 | dandre | from cli interface? |
| 15:22.52 | ManxPower | Kobaz: that NEVER works |
| 15:22.57 | ph8 | Kobaz: in the extensions context? |
| 15:22.57 | Kobaz | ManxPower: works for me |
| 15:23.07 | ManxPower | Kobaz: try removing it and see. |
| 15:23.07 | Kobaz | ph8: in the zap cfg |
| 15:23.10 | Kobaz | ManxPower: heh |
| 15:23.13 | ph8 | zap cfg? |
| 15:23.18 | ManxPower | Kobaz: he's not using zap |
| 15:23.21 | Kobaz | i dunno, i jumped in late |
| 15:23.26 | ManxPower | hence the "never works" comment |
| 15:23.31 | ph8 | ok then:p |
| 15:23.51 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:23.57 | Kobaz | ManxPower: all i know is i added it to my zap config, and all of a suddon i was able to get dtmf over analog, when it completely didn't work previously |
| 15:24.01 | [TK]D-Fender | well I see my time is pasted here.... moving on.... |
| 15:24.06 | [TK]D-Fender | wasted even... |
| 15:24.08 | Kobaz | [TK]D-Fender: aww |
| 15:24.17 | ph8 | so do you think this is a phone or asterisk issue? |
| 15:24.22 | Kobaz | [TK]D-Fender: i'll bake you some cookies if you stay |
| 15:24.43 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:25.04 | [TK]D-Fender | Kobaz: Thats a diet no-no. I'm going strict on this and have lost 15 bounds since the start or Aug. |
| 15:25.09 | [TK]D-Fender | pounds* |
| 15:25.22 | [TK]D-Fender | Kobaz: @ 193 now and aiming for 185. |
| 15:25.25 | ManxPower | Kobaz: you will also get doubled digits |
| 15:25.48 | [TK]D-Fender | ManxPower: For a lesbian that'd be considered "well-hung" ;) |
| 15:25.49 | Kobaz | ManxPower: *shrug*, it works very well for me |
| 15:26.20 | Kobaz | [TK]D-Fender: ah... i'm 5'4 120... |
| 15:26.35 | Kobaz | your unusualy fit geek |
| 15:26.40 | [TK]D-Fender | Kobaz: Keep baking, and stop giving them away :) |
| 15:26.55 | [TK]D-Fender | Kobaz: your stats are good... for a girl :) |
| 15:26.58 | Kobaz | haha |
| 15:27.03 | ManxPower | Eating right, exercising, etc do not actually make you live longer. It just SEEMS longer. |
| 15:27.27 | Kobaz | my ex girlfriend is 5'0 and like 109 |
| 15:27.39 | Kobaz | now she had good stats |
| 15:27.43 | ph8 | :-) |
| 15:28.54 | [TK]D-Fender | ManxPower: Being fit tends to get you out enjoying life more. Quality > quantity. Thats why I give LIFE threats now ;) |
| 15:29.28 | Odd_Bloke | Any pointers on my Realtime issues? |
| 15:29.49 | [TK]D-Fender | Odd_Bloke: I don't believe "sip show peers" will show you whats in realtime. |
| 15:31.10 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:32.52 | ph8 | [TK]D-Fender: Am i written off for using packages then? Is there anywhere i can lookup this DTMF problem? |
| 15:33.36 | [TK]D-Fender | ph8: You keep going on and never showing or telling us anything of value. We can't help you. |
| 15:33.38 | brodiem | Odd_Bloke: sip show peers will only list realtime peers that are registered and if rtcachefriends=yes |
| 15:34.28 | dandre | Blackvel: I had to put insecure = invite because I had this error: |
| 15:34.29 | dandre | [Sep 4 17:30:22] WARNING[26213]: chan_sip.c:8385 check_auth: username mismatch, have <OVH_BOXNET>, digest has <anonymous> |
| 15:35.07 | *** join/#asterisk pdc_pdb (n=ccbonill@201.228.24.214) |
| 15:35.34 | ManxPower | Results 1 - 8 of 8 from lists.digium.com for "check_auth: username mismatch" AND "too lazy to use google" |
| 15:36.00 | ManxPower | Results 1 - 10 of about 174 for "check_auth: username mismatch" AND "too lazy to use google" |
| 15:36.03 | ph8 | [TK]D-Fender: I was chatting while i put together the log paste, but i thought in that password '' -- incorrect line i'd found the problem? |
| 15:36.06 | ph8 | do you still want the logs? |
| 15:37.53 | *** join/#asterisk Drasha (n=Drasha@faster11.westcom.cz) |
| 15:44.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:47.05 | ph8 | :-( |
| 15:48.33 | *** join/#asterisk krdian (i=krdian@killer.radom.net) |
| 15:48.41 | ManxPower | ph8: nobody here is paid to provide people with support -- we are all volunteers |
| 15:48.43 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 15:48.45 | Assid | hi |
| 15:49.13 | Assid | im trying to do faxing through asterisk.. and a multitech gateway and finally a fax machine |
| 15:49.17 | ph8 | I know, i'm sorry if i've somehow come off as forceful or a timewaster, it wasn't my intention :/ |
| 15:49.38 | Assid | one of the errors i receive is: Unknown RTP codec 105 |
| 15:52.36 | *** join/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 15:52.38 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 15:52.52 | *** part/#asterisk Maxous (n=Maxous@74.7.13.242) |
| 15:53.45 | ManxPower | ph8: You came off as a package user -- that is a waste of time. |
| 15:54.02 | ph8 | i see, that's a shame |
| 15:54.17 | *** part/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 15:54.19 | ph8 | having done a bit of research the ubuntu packages are based on the debian packages which tzafrir_laptop, whom you mentioned, works on - so at least they should be good outdated packages |
| 15:54.38 | ManxPower | ph8: *nod* If you compiled from source the issue might be fixed by now. |
| 15:54.44 | lowtek | Assid: What codecs? |
| 15:54.46 | ph8 | i think it's the phone anyway, just looking for other dtmf options |
| 15:54.56 | ManxPower | but the REAL problem is that you are using such an old Asterisk. |
| 15:55.13 | Assid | ulaw over the did provider - and i kept the peer at allow=all |
| 15:55.32 | lowtek | Assid: Are you trying to fax over the internet? |
| 15:55.39 | Assid | yeah |
| 15:55.44 | x86 | ManxPower: you might get a kick out of this: http://forums.joerogan.net/showthread.php?t=74612 |
| 15:55.47 | ph8 | ah, asterisk was adjusted for effectively bad phone supplier dtmf signals? |
| 15:55.53 | Assid | but i have a multitech device (gateway) over the other end |
| 15:56.05 | lowtek | Assid: don't ... you may be able to get a couple working faxes here and there but overall you won't have any reliability |
| 15:56.16 | ManxPower | ph8: not at all. you should be doing out of band DTMF which has had major changes thru most of 1.4.x |
| 15:56.26 | lowtek | Assid: This issue has been fought and fought and the outcome is always the same, it isn't going to work very well |
| 15:56.29 | ph8 | i think that's what i've just setup :p |
| 15:56.42 | Assid | hrmm |
| 15:56.49 | ManxPower | But I'm thru with discussing it. You have your answer, I'm not going to justify who I help and who I do not help. |
| 15:57.55 | lowtek | ph8: What are you trying to accomplish overall? Setup a SOHO pbx? |
| 15:58.10 | Assid | whats RTP codec 102 anwyays? |
| 15:58.23 | ManxPower | Assid: your fingers hurt so much you can't use google? |
| 15:58.28 | lowtek | Assid: faxing protocols don't have much tolerance for jitter and no concept of packet loss |
| 15:59.14 | lowtek | Assid: among a dozen other reasons why it won't work well |
| 15:59.23 | Assid | hrmm |
| 15:59.31 | lowtek | Assid: Your only hope is t.38 or just use electronic faxing |
| 15:59.31 | ManxPower | lowtek: you will never convince Assid it's a bad idea. |
| 15:59.41 | lowtek | ManxPower: I know, can only try |
| 15:59.48 | Assid | ManxPower: i know its a bad idea.. the guy doesnt wanna listen |
| 16:00.02 | lowtek | ManxPower: I just hate seeing folks spinning their wheels |
| 16:00.03 | Assid | im working on soemthing for a friend/client of mine |
| 16:00.06 | *** join/#asterisk brettnem (n=brettnem@user-387oe35.cable.mindspring.com) |
| 16:00.10 | ManxPower | lowtek: no reason to even try. |
| 16:00.28 | lowtek | Assid: Tell him it ain't gonna work per the guys that actually write the freakin' software?! |
| 16:00.57 | Assid | lowtek: he has access to the web interface of this multitech crap.. and it says it does fax relay over t.38 |
| 16:00.59 | [TK]D-Fender | ManxPower>but the REAL problem is that you are using such an old Asterisk. <- nope. |
| 16:01.36 | ManxPower | [TK]D-Fender: I disagree. |
| 16:01.49 | lowtek | Assid: Asterisk 1.4 provides t.38 pass-through only |
| 16:02.03 | [TK]D-Fender | ManxPower: I'm not saying that it can't be a factor, but the real problem is much more acute. |
| 16:02.16 | lowtek | Assid: which means (i believe) that the next hop or in your case the destination fax device must support t.38 |
| 16:03.10 | lowtek | Assid: which still boils down to your inability to fax over the internet |
| 16:03.28 | *** join/#asterisk max_______ (n=mochoa@chi-pat.cashnetusa.com) |
| 16:03.48 | ManxPower | [TK]D-Fender: It's just a waste of time trying to help in his situation. |
| 16:04.04 | [TK]D-Fender | ManxPower: That is my assessment as well. |
| 16:04.07 | brodiem | or hand off to another SIP provider that is able to gateway T38 to PSTN, and cross your fingers.. |
| 16:05.45 | lowtek | brodiem: Are there any such providers? |
| 16:06.26 | brodiem | lowtek: yeah there's a couple listed on the wiki. I've had success with gafachi, but not success amounting to reliable faxing |
| 16:06.28 | ph8 | lowtek: SOHO? |
| 16:06.41 | lowtek | ph8: small office / home office |
| 16:07.02 | lowtek | brodiem: Ahh.. and that's all that counts really as far as the end-user.. |
| 16:07.05 | ph8 | essentially, i'm using a VPS in a DC to run one line for my small company 0845 |
| 16:07.53 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 16:08.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:08.30 | apocn | Hello all, is anyone experienced with AMD? |
| 16:08.39 | EmleyMoor | How long a delay do we expect before buildable zaptel-source returns to Debian Lenny? |
| 16:09.04 | lowtek | EmleyMoor: DAHDI? |
| 16:09.14 | ManxPower | EmleyMoor: Go ask the person that builds the packages! |
| 16:09.20 | apocn | Answering Machine Detect |
| 16:09.28 | EmleyMoor | lowtek: That won't come to lenny - but will to sid |
| 16:12.34 | Assid | hrmm.. hrmm ok |
| 16:12.37 | Assid | he cooled down |
| 16:12.39 | dandre | how can I know in which release this bug has been fixed : http://bugs.digium.com/view.php?id=9044 |
| 16:12.50 | Assid | said if i can get it working.. better. if not.. fine.. no choice |
| 16:13.21 | Assid | now i need a good origination location.. unlimited incoming but channel limit will be fine |
| 16:13.33 | Assid | should support number portability |
| 16:14.08 | ManxPower | dandre: try asking on #asterisk-dev |
| 16:15.11 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:15.58 | brodiem | Assid: |
| 16:16.00 | brodiem | ~itsplist |
| 16:16.37 | brodiem | ~itsp |
| 16:16.37 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:16.55 | Blackvel | anyone with patton isdn bri gateway 4636 experience here? how can I get to do outbound authentication, so I do not have to set insecure=invite? |
| 16:17.16 | Blackvel | can I get it to do... |
| 16:17.42 | ManxPower | Blackvel: I doubt you can. |
| 16:19.29 | Blackvel | must be something worse it: identity patton_user authentication outbound authenticate 1 authentication-service asterisk_auth_1 username patton_user |
| 16:20.10 | Blackvel | its just searching for the wrong user. in the sip packets its never patton_user but the from is <pstn-number>@ip:port :) |
| 16:21.35 | ManxPower | Blackvel: It sounds like you need to CALL their support |
| 16:21.51 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 16:25.00 | [TK]D-Fender | dandre: Closed : 2008-02-25 17:19 <- look through the changelog by date. |
| 16:25.25 | dandre | ok |
| 16:26.35 | dandre | should aparently fixed in 1.4.22 |
| 16:27.49 | [TK]D-Fender | dandre: I'm sure long before then |
| 16:32.58 | *** join/#asterisk sgala (n=sgala@ip-140-98.sn3.eutelia.it) |
| 16:33.24 | sgala | hi, I have a problem with MOH. I think that is a timing problem. I have a dell poweredge server with debian etch with 2.6.18 and no digium hardware. so i need to use ztdummy. When i try to using it the kernel complain with the error "rtc: lost some interrupts at 1024Hz.". do you have clean ideas to fix this problem? |
| 16:34.21 | [TK]D-Fender | sgala: You need to have your kernel compiled with a 1000hz timer, not 1024 |
| 16:34.28 | ManxPower | sgala: you do NOT need ztdummy for MoH |
| 16:34.47 | sgala | [intra]lanman: no. i have the debian default |
| 16:36.15 | ManxPower | sgala: it was not a question. |
| 16:36.15 | sgala | ManxPower: are you sure? I'm experiencing this problem with MOH. when the call is terminated into a PSTN line or GSM line, the MOH or the "ring" is very very choppy... via voip to voip (ulaw/alaw) is perfect... |
| 16:36.48 | ManxPower | sgala: I am sure. zaptel timer *might* help make things less choppy on very underpowered machines. |
| 16:36.57 | tinkerghost | Thank you to the people who made suggestions on how to fix my IAX client echo problem. Turned out to be a HW problem on the box - no mike is installed but for some reason it was routing the speaker output through the mike input. Muted & unmuted the mike & all is good. |
| 16:37.06 | ManxPower | but the real problem is exactly what [TK]D-Fender said. |
| 16:37.50 | ManxPower | recompile your kernel with the required kernel HZ |
| 16:38.03 | sgala | ManxPower: ok. where i can find the my current timer? CONFIG_HZ_1000=y , this? |
| 16:38.46 | ManxPower | sgala: I have no idea. I do not do Linux Kernel Support. |
| 16:39.05 | ManxPower | perhaps #linux or #yourdistro is a better place for those kind of kernel problems? |
| 16:40.00 | sgala | ManxPower: ok thanks for the suggestion. i'll try. thanks, bye |
| 16:40.04 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:42.48 | tzafrir_laptop | hmm... folks with Debian Etch should also considering the etchandahalf 2.6.24 kernel |
| 16:45.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:46.34 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 16:46.49 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:46.58 | *** join/#asterisk VaNNi (n=VaNNi___@38.104.76.90) |
| 16:50.38 | scooby2 | ahh, i understand my FIFO issue now. if an agent is in two queues and each queue has someone waiting then it will not always be the person who was waiting longest that gets answered. If all the calls are in one queue, then fifo applies. |
| 16:54.43 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-135-160.dsl.teksavvy.com) |
| 16:59.26 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 17:03.09 | l2trace99 | anyone using astmanproxy ? |
| 17:04.05 | dandre | I haven't found on google any solution for the 'check_auth: username mismatch' |
| 17:04.09 | dandre | problem |
| 17:04.31 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.08 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 17:11.13 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 17:13.17 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:13.47 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:16.20 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:17.28 | *** join/#asterisk Zion800 (n=Zion@38.98.23.250) |
| 17:18.50 | *** join/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
| 17:18.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:19.10 | Zion800 | Has anyone had any success with the Kevin Mitnick method of unmasking inbound callier-id? |
| 17:19.46 | Qwell | Zion800: call your provider and ask them if they violate the RFCs like his "method" requires |
| 17:20.04 | Zion800 | haha |
| 17:20.26 | Qwell | his "hack" is a joke, and he should be ashamed of himself |
| 17:20.35 | lmadsen | he's out of practice |
| 17:20.43 | Zion800 | well, i did notice that my provider sends in the CID of a blocked number through the RPID |
| 17:20.48 | Zion800 | i use Bandwidth.com |
| 17:20.55 | Qwell | there you go then |
| 17:21.45 | Zion800 | well, i was able to take the CID out of the RPID and set it as the caller-id, but the phone still shows as blocked |
| 17:22.19 | _ShrikE | Zion800: was ther also a privacy header? |
| 17:22.24 | Zion800 | ya |
| 17:22.29 | _ShrikE | set that to none |
| 17:22.38 | Zion800 | isnt it read-only? |
| 17:24.31 | Zion800 | do i do SipAddHeader(Privacy: none)? |
| 17:24.48 | Zion800 | oops |
| 17:25.08 | Zion800 | do i do SipAddHeader(Remote-Party-ID: privacy=none)? |
| 17:25.40 | _ShrikE | try the first one.. SipAddHeader(Privacy: none) |
| 17:25.59 | [TK]D-Fender | Zion800: "core show application setcallerpres" |
| 17:26.57 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:26.57 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:27.22 | Zion800 | ah...ok...so I would do SetCallerPres(allowed_not_screened) |
| 17:27.52 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:28.50 | Zion800 | worked like a charm! |
| 17:29.32 | [TK]D-Fender | Zion800: You're welcome. |
| 17:30.06 | [TK]D-Fender | Zion800: I'd do that and compare the CID/RPID for the call and override with a prefix on all calls originating from that peer. |
| 17:30.34 | *** join/#asterisk hi365_m (n=hi365@213.151.50.174) |
| 17:42.55 | rob0 | I have a dumpster-grade telephone from a "Lucent Partner" system. Looks like regular RJ11, but no dial tone from my Sipura ATA. Should I return it to the dumpster from which it came? |
| 17:43.52 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 17:45.14 | [TK]D-Fender | rob0: that'd be a digital handset. Useless unless paire with an expensive gateway. |
| 17:45.52 | Zion800 | [TK]D-Fender: I altered the CID Name to make it apparent. What would the advantage of adding a prefix be? |
| 17:46.13 | Qwell | Zion800: only to make it apparent |
| 17:46.21 | rob0 | thanks |
| 17:46.30 | Zion800 | oh...ok :-) |
| 17:47.04 | Zion800 | thanks for all your help... |
| 17:48.22 | [TK]D-Fender | Zion800: indeed... to make it apparent :) I'd use "P:" or similar. "*" if the char doesn't cause problems. |
| 17:51.33 | *** join/#asterisk KuJaX (n=nick@ip70-190-194-221.ph.ph.cox.net) |
| 17:52.22 | KuJaX | I cannot seem to get one phone (Cisco 7940) to connect for multiple extensions. I am positive that I have the .CNF file configured properly, but i think there may be a setting within a config file of Asterisk that may keep from one IP address having multiple extensions? |
| 17:52.47 | Knightfal | Im trying to set up a SIP trunk with a nortel cs1000 I have in my sip conf [Seibels2] |
| 17:52.47 | Knightfal | type=peer |
| 17:52.47 | Knightfal | host=66.162.7.27 Im getting -- Called Seibels2/4200 |
| 17:52.47 | Knightfal | but it never places the call also when they try to call I see nothing in the CLI and on thier end they are getting just a busy signal...anyone have any ideas |
| 17:53.50 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 17:54.32 | ManxPower | Knightfal: Oh, it's placing the call, the nortel is just not responding, you can see this in sip debug |
| 17:55.14 | ManxPower | Also it should be 4200@Seibels2 NOT Seibels2/4200 incase you have it wrong in your dialplan |
| 17:56.56 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 17:57.00 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 17:57.02 | Knightfal | oh |
| 17:57.11 | [TK]D-Fender | ManxPower: Either is valid |
| 17:57.13 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:57.18 | ManxPower | [TK]D-Fender: not in my experience |
| 17:57.24 | raz | hum. how do i enable my sip phone to dial outbound over the sip connection? any good tutorials? |
| 17:57.26 | Knightfal | what about from the nortel to * |
| 17:57.42 | ManxPower | Knightfal: call your nortel SIP expert. |
| 17:58.09 | [TK]D-Fender | Knightfal: Go look at your actual SIP debug. There is nothing worth saying without that. |
| 17:58.15 | ManxPower | raz: your question is both so simple and so complex nobody is going to answer it. Perhaps you should read the Asterisk Book first? |
| 17:58.23 | ManxPower | ~book |
| 17:58.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:58.26 | raz | ManxPower, yup i know sorry |
| 17:58.40 | ManxPower | don't be sorry, be reading the book |
| 17:58.48 | [TK]D-Fender | raz: And you should google up guides for setting up the peer for your provider |
| 18:03.14 | raz | [TK]D-Fender, i have inbound SIP working already. my approach now is to try to make a "catch-all" extension in the context of my sip phone that will Dial() out. |
| 18:03.25 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 18:03.42 | raz | just wondering what the "catch-all" pattern is.. found 's' and '_' .. but umm default? |
| 18:04.22 | ManxPower | um, no "s" is NOT a catch-all pattern. Stop talking. Start reading. |
| 18:04.43 | raz | same to you. i didn't say it is. |
| 18:05.06 | ManxPower | building a catchall extension is a bad idea and can open up massive security hole in your system. |
| 18:05.15 | ManxPower | Using pattern matches is, of course, a good idea. |
| 18:05.43 | raz | well, i simply want to enable outbound calling. i could use _7 or so but that feels wierd, numbers can be of any length |
| 18:05.49 | ManxPower | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@yourprovider) |
| 18:05.57 | ManxPower | this should be covered in the book. |
| 18:05.59 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 18:06.08 | raz | oh thx! |
| 18:06.59 | raz | hmm |
| 18:07.05 | raz | that requires the user to dial "1" first? |
| 18:07.28 | jaytee | other than myself and maybe 4 other people max in here, has anyone else actually read the book? or do they just say they did. |
| 18:07.49 | lowtek | I have the original yellow paperback somewhere .... |
| 18:07.50 | raz | only some chapters |
| 18:08.38 | *** join/#asterisk Greek-Boy (n=email@41.221.43.250) |
| 18:08.39 | [TK]D-Fender | raz: Make your own patterns to look like whatever you want and dialout with your provider manipulating what they dialed however you want |
| 18:09.56 | raz | yup |
| 18:10.01 | ManxPower | [TK]D-Fender: I put raz on /ignore. |
| 18:10.04 | raz | i think something like _XXXX. will do for my purpose :) |
| 18:10.23 | raz | this channel is really hostile at times |
| 18:11.14 | [TK]D-Fender | raz: That is so untrue. |
| 18:11.29 | [TK]D-Fender | raz: Its more of a constant state of being :) |
| 18:11.36 | raz | hehe |
| 18:11.39 | ManxPower | mostly because he's talking and not reading. |
| 18:11.42 | jaytee | [TK]D-Fender, do you know if the SPA3102 supports dialing out of the FXO port without having to enter a pin or authenticate? |
| 18:11.56 | [TK]D-Fender | jaytee: Indeed it does. |
| 18:12.15 | [TK]D-Fender | jaytee: It can be made to be smart or dumb depending on your needs |
| 18:12.49 | lowtek | lol http://www.youtube.com/watch?v=2JYRl96nROA |
| 18:12.58 | jaytee | excellent, because I tested the Valcom stuff using a TDM04B in my * test box and it worked fine but I need an FXO for each unit that's "portable" and the 3102 would be ideal since it also has an FXS I can hang an analog phone on. |
| 18:13.52 | *** join/#asterisk Levonk (n=lk@adsl-75-62-136-206.dsl.lsan03.sbcglobal.net) |
| 18:14.24 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:14.37 | jaytee | some of our outbuildings are just too nasty an environment to put a nice Polycom in, especially our elephant holding and lion/baboon areas. I had a Cisco switch that some dumbass mounted port side up on the wall and unused ports were filled with cockroach eggs. |
| 18:15.21 | raz | lowtek, the real deal: http://subt1.net/2008/08/13/roflcopter/ |
| 18:16.09 | jaytee | lmadsen, Adam West is going to be at a scifi/comic convention in Niles, OH this month. Dont know if he's got the Batmobile with him or not. |
| 18:17.34 | *** join/#asterisk rdgr (n=rich@82.32.1.139) |
| 18:17.50 | *** join/#asterisk dawalama (n=dawalama@wsip-68-105-250-226.sd.sd.cox.net) |
| 18:18.05 | lowtek | raz: Yea, I fly those. I have a TREX 600, 700, an Avant Aurora and a Synergy N9. |
| 18:18.13 | lmadsen | jaytee: put in a crappy grandstream then :) |
| 18:18.13 | seanbright | adam weeeest, adam weeeeeest |
| 18:18.14 | dawalama | Hello call .. |
| 18:18.18 | dawalama | I am having a silly problem |
| 18:18.25 | dawalama | I want to look at the debug log |
| 18:18.30 | dawalama | but I don't know where to find it |
| 18:18.36 | dawalama | res_config_pgsql.c: Postgresql RealTime: Failed to query database. Check debug for more info. |
| 18:18.49 | raz | hmm now i get, when i try to make an outbound call: "Received response: "Forbidden" from '"fony" <sip:2000@192.168.254.254>" .. how to fix? :\ |
| 18:19.01 | jaytee | lmadsen, I'm already using 1 Handytone-286 for (ugh!) fax to one building. Don't want to buy anymore GS crap |
| 18:19.02 | raz | lowtek, cool! it's amazing what these little beasts can do |
| 18:19.11 | lowtek | raz: Very fun hobby. |
| 18:19.19 | ManxPower | dawalama: generally whatever is set for debug in /etc/asterisk/logger.conf |
| 18:19.32 | lmadsen | jaytee: put the phone inside a plexiglass case with a door? |
| 18:19.35 | jaytee | audio quality on the FXS ports using the same analog phone is of a much higher quality than using a Handytone. |
| 18:19.44 | jaytee | lmadsen, hmmm that might work. |
| 18:19.47 | lmadsen | or that'd work too :) |
| 18:20.11 | *** join/#asterisk matt_keys (i=smatt@usaregs.com) |
| 18:20.25 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 18:20.41 | matt_keys | can someone identify this warning for me? |
| 18:20.43 | matt_keys | WARNING[12890]: chan_zap.c:6810 ss_thread: CallerID returned with error on channel 'Zap/1-1' |
| 18:20.53 | jaytee | ever see a motherboard covered with salt film? Our dolphin pavillion is probably our harshest environment for computer equipment. Salt water tanks, humidity and condensation wreak havoc. |
| 18:21.08 | ManxPower | matt_keys: it means Callerid did not work. |
| 18:21.18 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 18:21.26 | matt_keys | yeah i figured that, but where lies the problem? in chan_zap.c or elsewhere |
| 18:21.37 | jaytee | in your zapata.conf most likely |
| 18:22.05 | dawalama | Thanks a lot ManxPower |
| 18:22.08 | dawalama | :-) |
| 18:23.24 | zeeesh | is there any PAP2/SPA2000 etc support channel available ? |
| 18:23.50 | dawalama | in /etc/odbc.ini , does protocol represent the databse version ? |
| 18:24.03 | matt_keys | jaytee: usecallerid=yes callerid=asreceived hidecallerid=no ... is there any particular setting this usually applies to? |
| 18:24.22 | ManxPower | matt_keys: the error message does not indicate in any way shape or form what is wrong. |
| 18:24.37 | ManxPower | Generally with callerid errors you should play with your rxgain |
| 18:24.58 | matt_keys | no other way for it to be more descriptive? |
| 18:25.13 | jaytee | this is analog, not PRI? |
| 18:25.28 | matt_keys | yeah, two lines coming into fxo (tdm400) |
| 18:25.29 | ManxPower | matt_keys: it does not KNOW what went wrong. |
| 18:25.43 | ManxPower | All Asterisk knows is that what it received for callerid was not valid |
| 18:25.57 | ManxPower | not the text, the actual FSK CLID burst. |
| 18:27.19 | matt_keys | ManxPower : i'm assuming your rxgain suggestion means that it's too hot coming in? |
| 18:27.24 | matt_keys | or too weak? |
| 18:27.30 | jaytee | probably too hot |
| 18:27.41 | ManxPower | matt_keys: Yes. |
| 18:27.56 | ManxPower | Chances are it is either too hot or too weak |
| 18:28.16 | ManxPower | be sure to test and fax machines connected to asterisk after you mess with gains |
| 18:28.33 | ManxPower | messing with gains can also cause echo |
| 18:28.53 | matt_keys | the customer had a bad echo problem, i recompiled zap with oslec and that cured it up |
| 18:29.00 | jaytee | might want to run fxotune after adjusting the gain too |
| 18:30.25 | matt_keys | thanks for the tips fellas, i'll go read about rxgain etc.. any suggestions on how much to raise/lower at a time? |
| 18:30.37 | matt_keys | or should it be in the docs |
| 18:30.56 | *** join/#asterisk bradleyprice86 (n=knoppix@fw.datafax.net) |
| 18:31.28 | matt_keys | the customer is at a remote facility in another state so it'd be kinda hard for me to get any readings outside the box itself |
| 18:31.33 | ManxPower | matt_keys: there are no docs, it TOTALLY DEPENDS ON YOUR LINE |
| 18:31.56 | matt_keys | ManxPower : oic |
| 18:33.06 | scooby2 | anyone know what "Hungup, howlong is 1, autologoff is 0" means? |
| 18:33.34 | dawalama | When I do "sip show peers" .. nothing shows up .. but I know I have the values stored in the database |
| 18:33.48 | dawalama | And tailing at the log, no sql command gets executed |
| 18:33.53 | dawalama | is this the right behavior ? |
| 18:33.59 | [TK]D-Fender | dalIIRC "sip show peers" does not look at the DB |
| 18:34.05 | dawalama | hmm .. |
| 18:34.11 | matt_keys | ManxPower and jaytee thanks for the help |
| 18:34.12 | dawalama | how do I make it work with the database |
| 18:34.36 | [TK]D-Fender | dawalama: there is probably a realtime CLI command. Go type "help" and look at the list |
| 18:35.53 | dawalama | Thanks [TK]D |
| 18:38.09 | dawalama | So, looks like the asterisk realtime is doing something really funny .. |
| 18:38.22 | dawalama | I looks for sip_buddies for name='102' |
| 18:38.24 | dawalama | finds it |
| 18:38.39 | dawalama | then it looks for '102-werqaasdfa' |
| 18:38.44 | dawalama | which will never be found |
| 18:38.53 | dawalama | because I don't see it trying to update or insert that information anywhere |
| 18:39.21 | dawalama | Does this have anything to do with me not being able to use the sip registration to make a call |
| 18:39.28 | ManxPower | dawalama: ENTER is not a punctuation character. |
| 18:40.11 | dawalama | lol .. sorry about that |
| 18:42.28 | Kobaz | heh |
| 18:43.24 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 18:45.49 | *** join/#asterisk Grufft3ch (n=A@140.239.216.61) |
| 18:46.57 | Grufft3ch | Hey guys -- Quick Question, I'm sure i'm missing a setting somewhere. I've got a Asterisk PBX behind a Cisco ASA5505 (I've allowed UDP, TCP, IP, and ICMP, as well as turning off Inspect-SIP) |
| 18:47.21 | Grufft3ch | For some reason when a Polycom IP330 is remote -- it registers without any issues |
| 18:47.55 | Grufft3ch | However, when i place a call, the call goes through, the called-phone rings, but, No sound goes through |
| 18:48.25 | ManxPower | ~nat |
| 18:48.26 | jbot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:48.27 | ManxPower | ~sipnat |
| 18:48.28 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:48.38 | ManxPower | There you go, Grufft3ch |
| 18:48.47 | Grufft3ch | Thanks Manx, lemme go read ^,^ |
| 18:49.44 | Kobaz | can you set a channel variable in the ami |
| 18:52.03 | lowtek | No shit, a ticket I just got: "We're trying to get one connected to another and cant find it please let us know" |
| 18:53.38 | DarylVOIP | Sounds like the tickets my Tier 1 gets from customers. |
| 18:53.38 | Grufft3ch | ManxPower > from what i read this is a Server-side issue : How does a softphone, such a xLite then work fine through NAT? |
| 18:53.50 | dawalama | So, is there a trick to have SIP register using Asterisk Realtime |
| 18:54.03 | DarylVOIP | "account 00000 sound not come to phone and me. plz fix." |
| 18:54.09 | [TK]D-Fender | Kobaz: Yes |
| 18:54.28 | [TK]D-Fender | Grufft3ch: Its all in the guide. Read it. |
| 18:59.07 | [TK]D-Fender | *b00m* |
| 18:59.09 | *** join/#asterisk bradleyprice86 (n=knoppix@fw.datafax.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk matt_keys (i=smatt@usaregs.com) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk _nn (n=nn@unaffiliated/nn) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk krdian (i=krdian@killer.radom.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk ManxPower (n=manxpowe@245.sub-70-220-211.myvzw.com) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk [netman] (n=netman@20.Red-88-25-137.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk daniev (n=ganbarim@190.144.60.154) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk moy (n=moy@189.169.91.147) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-088-012.pools.arcor-ip.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk jblack (n=jblack@pool-71-181-244-67.sctnpa.east.verizon.net) [NETSPLIT VICTIM] |
| 18:59.09 | *** join/#asterisk _trine (n=trine@38.100.90.3) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk saint_ (n=saint@cpe-75-83-219-24.socal.res.rr.com) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk bkw_ (n=bkw_@freeswitch/developer/bkw) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk SpaethCo (n=SpaethCo@cache.spaethco.com) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk nhuisman_work (n=nhuisman@aeko.ifa.hawaii.edu) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk hardwire (n=hardwire@rdbk-12639.mtaonline.net) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk thing1 (n=Dwayne@64.42.227.97) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk felipex (n=felipex@213-140-21-233.fastres.net) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) [NETSPLIT VICTIM] |
| 18:59.10 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk x86 (n=x86@p3m/member/x86) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk Alton2 (n=alton@72.48.111.234) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk latouche (n=Latouche@maisel-gw.enst-bretagne.fr) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk errr (n=errr@fedora/errr) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk davidc (n=david@netman1.us.sargasso.net) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk jetlagmk2 (i=jetlag@70.104.82.57) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) [NETSPLIT VICTIM] |
| 18:59.11 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk eharris (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk zoid_99 (n=chris@router.asteriasgi.com) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk brodiem (n=brodiem@e2.72.1243.static.theplanet.com) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk JT (n=j@unaffiliated/jt) [NETSPLIT VICTIM] |
| 18:59.12 | *** join/#asterisk bronx (i=[U2FsdGV@pallas.utanet.fi) [NETSPLIT VICTIM] |
| 18:59.12 | bkruse | yay splits |
| 18:59.16 | [TK]D-Fender | *b00m* |
| 18:59.17 | scooby2 | anyone using QueueMetrics? |
| 18:59.17 | Kobaz | ooo |
| 19:00.37 | Kobaz | scooby2: queuemetrics is a bloated sack |
| 19:00.38 | scooby2 | have anything better? |
| 19:00.38 | Kobaz | no |
| 19:00.38 | scooby2 | lol |
| 19:00.38 | Blackvel | ManxPower: good point for the support calling. but talking and explaining in English won't make it easier for me :) |
| 19:00.38 | Kobaz | heh |
| 19:00.39 | scooby2 | Its qloader.pl is corrupting my queue_log table for some reason. I have to repair the table every 15 minutes or so. |
| 19:00.39 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 19:00.39 | scooby2 | I'm sure its a "new feature" |
| 19:01.22 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 19:02.17 | Grufft3ch | if i make a change to sip.conf -- can i just reload that, or do i actually need to restart the entire Asterisk process? |
| 19:02.44 | Blackvel | reload chan_sip.so |
| 19:03.12 | Blackvel | reload does that too + extensions + iax and does not restart neither |
| 19:03.33 | Blackvel | guess reload does more than just those three ;) |
| 19:04.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:06.13 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:06.53 | scooby2 | does asterisk freak if it cannot talk to mysql for cdr or should I be able to restart MySQL real quick? |
| 19:07.09 | ManxPower | there are a few options that might not be changed on a sip reload. qualify= is one of them |
| 19:07.29 | ManxPower | scooby2: it's supposed to buffer, I don't know which version that applies to. |
| 19:07.59 | scooby2 | ManxPower: ok, i better wait then. thx |
| 19:08.22 | jaytee | [TK]D-Fender, check this out. Valcom has SIP based paging equipment that's pretty slick and has been tested and supported on *. http://voip.valcom.com/interoperability.htm |
| 19:09.30 | [TK]D-Fender | jaytee: Cool. |
| 19:10.04 | ManxPower | jaytee: price? |
| 19:10.05 | jaytee | CDW wants 900 bucks for Valcom's SIP Paging Gateway. it supports up to 256 zones and 65000 speakers. |
| 19:10.11 | ManxPower | ah. |
| 19:10.28 | Blackvel | time for break and freetime. |
| 19:10.37 | Blackvel | [TK]D-Fender. works with insecure=invite now :) |
| 19:10.40 | jaytee | speakers sold separate of course but it'll work with SIP based PA speaker or analog speakers. |
| 19:11.02 | Blackvel | cu all |
| 19:12.02 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 19:12.02 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 19:12.17 | *** join/#asterisk spokra (n=spokra@75-92-217-76.sea.clearwire-dns.net) |
| 19:16.45 | [TK]D-Fender | jaytee: Sounds like a great deal |
| 19:17.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:17.59 | jaytee | [TK]D-Fender, yeah, I'm crunching the numbers and trying to find prices for SIP speakers. If they aren't more than 30 bucks over the analog ones then it'll be cheaper overall to go with this unit instead of multiple single page units using FXO adapters. |
| 19:23.03 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088810630.dsl.bell.ca) |
| 19:23.22 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 19:26.35 | *** join/#asterisk mog (n=mog@nat/digium/x-ae9c3e7751578d56) |
| 19:26.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:30.11 | jaytee | damn, the SIP PA speakers are way too expensive compared to the analog. gonna have to stick with the original strategy. |
| 19:30.59 | jaytee | 512 bucks for a SIP 5W PA horn vs 87 bucks for an amplified 15W analog PA horn. |
| 19:33.11 | l2trace99 | does anyone have a suggestion for a management interface proxy that can handle multiple servers ? |
| 19:34.22 | ManxPower | jaytee: silly human! "SIP" adds $300 to any product. |
| 19:34.36 | ManxPower | l2trace99: for some reason I thought astmanproxy did |
| 19:34.51 | jaytee | ManxPower, seems that way sometimes |
| 19:35.49 | l2trace99 | it does but from what I see you have to specify the server in the manager interface which kinda defeats the purpose |
| 19:36.19 | l2trace99 | unless I am missing it |
| 19:36.25 | jaytee | although a Polycom 550 is just as "feature rich" as a Nortel Meridian 3904 and I can get one of the former from CDW for 200 bucks less than what my Nortel VAR wants for a new 3904 |
| 19:37.00 | l2trace99 | the astmanproxy mailing list is setup to invite only |
| 19:44.39 | jaytee | anyone ever program a "hotline" feature where you pick up a specific phone and it automatically dials a preprogrammed extension? |
| 19:45.31 | *** join/#asterisk bradleyprice86 (n=knoppix@fw.datafax.net) |
| 19:47.24 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 19:48.57 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 19:50.08 | [TK]D-Fender | jaytee: Easily doable on many sip phones including SPA's |
| 19:51.47 | jaytee | [TK]D-Fender, yeah, I was just looking through the docs on the SPA2102 I have because I got hits for that when I googled. Couldn't find anything in the docs for the Polycom 330's but I wouldn't want to use one for this purpose, it's a waste of a nice phone. |
| 19:55.34 | [TK]D-Fender | jaytee: Yup |
| 19:56.26 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:56.36 | jaytee | [TK]D-Fender, was talking with a guy from Montreal earlier today @ Nuance. We got talking about all the bike only lanes up there it made me think of you commuting. |
| 19:56.42 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 19:57.05 | JerJer | ugh - have a real time problem... prolly just my stupidity since i have avoided realtime so much |
| 19:57.25 | lesouvage | I had a brand new linksys ata with two phone ports to setup a hotline but I couldn't fined the hotline option in the web interface of the ata. Is it there under an other name or are there different types of ATA fro Linksys? |
| 19:57.27 | [TK]D-Fender | jaytee: None of which are involved in my commute :) I take a shortcut through the woods to get from home to work. good 20 min trip |
| 19:57.46 | jaytee | mountain bike? |
| 19:58.35 | jaytee | I've seen you do the commute in less than 20. 15 maybe from logout to login |
| 19:59.45 | JerJer | i have extconfig setup with extensions => mysql,asterisk,extensions_conf - but asterisk isn't finding realtime extensions |
| 19:59.55 | jaytee | [TK]D-Fender, but at least they have dedicated lanes up there, down here they want to run you off the road. Heck, if your car looks like it gets more than 10 miles to the gallon you get dirty looks like you're "Unamerican" |
| 20:00.44 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.2) |
| 20:00.45 | JerJer | if i do realtime load extensions exten 1234 - asterisk says it loads the extension, but still won't use it upon incoming call. Do i need the switch statement in there? |
| 20:01.05 | [TK]D-Fender | jaytee: 20 min is a safe #. 16 if I biked like a psycho and not on my wooded trip (unwise path) |
| 20:01.33 | [TK]D-Fender | JerJer: Yes |
| 20:01.39 | _ShrikE | JerJer: you need the switch statement in that context to use real time if the extension in not found in the flat file |
| 20:01.48 | JerJer | ahh ok |
| 20:01.54 | JerJer | since this is 'dynamic' version? |
| 20:01.55 | [TK]D-Fender | JerJer: you still need extensions.conf to set up the contexts and determine which ones refer to realtime |
| 20:02.01 | JerJer | or same for the static way |
| 20:02.01 | jaytee | [TK]D-Fender, I just remember being in here on one particular day and you logged out and then back in again and it was around 16 minutes and I thought "Damn, he's quick!" |
| 20:02.25 | [TK]D-Fender | jaytee: I doubt it was 16, but I'll take it as a compliment :) |
| 20:04.07 | *** join/#asterisk Great_Anta_Baka (i=c4249986@gateway/web/ajax/mibbit.com/x-2221cc7e4f35441c) |
| 20:04.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:04.58 | *** join/#asterisk jackson__ (n=jackson@96-42-208-31.dhcp.roch.mn.charter.com) |
| 20:05.00 | JerJer | ugh - does BE have the realtime switch??? |
| 20:05.12 | lmadsen | it may not... |
| 20:05.26 | JerJer | i don't see a pbx_realtime |
| 20:05.32 | JerJer | .so |
| 20:05.48 | lmadsen | hrmmm |
| 20:06.01 | JerJer | c.1.18.1 |
| 20:06.05 | JerJer | er |
| 20:06.06 | JerJer | 8.1 |
| 20:06.07 | lmadsen | I don't use it, so I suppose I may not be missing it |
| 20:07.56 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:08.54 | Qwell | JerJer: it should be there |
| 20:09.16 | JerJer | i'm not seeing it :( |
| 20:09.38 | JerJer | is there like a config situation for BE? i have never installed it |
| 20:09.58 | JerJer | just trying to solve this d00d's problemo |
| 20:12.00 | *** part/#asterisk JenniferAkemi (n=akemi@206-248-135-160.dsl.teksavvy.com) |
| 20:14.10 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 20:14.27 | *** join/#asterisk rasterix (n=IceChat7@86.157.15.94) |
| 20:14.57 | rasterix | evening |
| 20:15.29 | rasterix | can include => exist within the global and general contexts? |
| 20:15.40 | Qwell | ~tias |
| 20:15.41 | jbot | Try it and see! |
| 20:16.24 | rasterix | qwell: im at home i only have asterisk at work... i had it at home but it was leading to a divorce :( |
| 20:16.47 | *** join/#asterisk esaym-acer (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:17.32 | rasterix | i guess you dont know the answer :P JOKE |
| 20:23.03 | unpaidbill | do you have to tell ReceiveFAX to use T38 when receiving/transmitting a fax or does it automatically? i see the variable for faxmode but it only reports on the method used, not sets it |
| 20:26.33 | unpaidbill | oh nm, it looks like it automatically detects based on the debug log.. i guess i'll find out soon enough |
| 20:32.47 | *** join/#asterisk iXombie (n=OhHai@64-240-254-51.segurosdirect.net) |
| 20:32.54 | iXombie | Hello |
| 20:33.42 | *** part/#asterisk gr0mit (n=tim@81.187.32.146) |
| 20:36.37 | Knightfal | Ok so Im back on my issue with the nortel cs1000.... It looks like the calls are placed and then I get an all circuits busy after a few min |
| 20:36.52 | Knightfal | http://pastebin.com/mdfba910 <------here is the call and sip debgger :) |
| 20:37.02 | Knightfal | Anyone see what the heck I may be doing wrong |
| 20:37.44 | Knightfal | FYI the Nortel also gets a reorder when they try to send a call to me |
| 20:37.53 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:37.55 | l2trace99 | jerjer: extensions.conf => odbc,asterisk,ast_config |
| 20:38.54 | l2trace99 | it is file.conf => driver,database[,table] |
| 20:39.37 | JerJer | they want dynamic |
| 20:39.39 | l2trace99 | and thats not realtime |
| 20:40.00 | JerJer | i figured it out.... someone had deleted pbx_realtime.so from this install |
| 20:40.16 | Qwell | JerJer: well, why would they do something silly like that? |
| 20:40.17 | l2trace99 | what's a little reload between friends |
| 20:40.35 | JerJer | Qwell: to run with a minimal asterisk footprint |
| 20:40.56 | JerJer | l2trace99: i agree - but they want dynamic, so they get dynamic :) |
| 20:41.40 | Qwell | JerJer: noload => ... |
| 20:41.47 | Qwell | or just don't set autoload |
| 20:42.08 | JerJer | that seems overly complex |
| 20:42.20 | JerJer | when if the module doesn't exist at all, it won't get loaded |
| 20:42.29 | Qwell | so is ignoring your package manager and removing files it installs... |
| 20:42.33 | Qwell | an upgrade would restore the file |
| 20:43.02 | JerJer | when i do my router based installs of asterisk, its hardcore small ... like 15 modules loaded |
| 20:43.27 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:46.01 | anonymouz666 | and it makes a big difference? |
| 20:47.58 | JerJer | prolly not |
| 20:48.18 | JerJer | but why have 60 modules burning ram when only 15 get used? |
| 20:48.40 | JerJer | <PROTECTED> |
| 20:49.19 | JerJer | and like why would i want chan_mgcp listening when I don't have mcgp phones?? |
| 20:49.27 | JerJer | or chan_skinny for that matter |
| 20:49.35 | anonymouz666 | indeed |
| 20:49.41 | JerJer | just one more potential hole for someone to jump thru |
| 20:49.53 | l2trace99 | only if it is loaded |
| 20:50.05 | l2trace99 | and it shouldn't be in mem if it is not loaded |
| 20:50.14 | JerJer | its gonna be on any install that has done make samples |
| 20:50.21 | JerJer | which is a shit load of systems |
| 20:50.25 | l2trace99 | then you have the option to support the devices |
| 20:50.38 | JerJer | then add that support when you know you need it |
| 20:50.41 | anonymouz666 | JerJer: the module the most pisses me off is codec_zap |
| 20:50.55 | anonymouz666 | you don't have the damn hardware and it loads |
| 20:51.19 | JerJer | yeah - i don't even compile that shit |
| 20:52.04 | anonymouz666 | haha |
| 20:52.12 | Qwell | anonymouz666: without loading, how is it going to know whether it needs to load? |
| 20:52.52 | phix | :D |
| 20:53.07 | JerJer | my theory is, it shouldn't even attempt to load anything unless the system engineer (ME) requires it for a specific purpose |
| 20:53.15 | Qwell | JerJer: autoload=no |
| 20:53.25 | JerJer | make menuconfig - F7 |
| 20:53.27 | JerJer | :) |
| 20:53.54 | JerJer | Qwell: problem with autoload is you gotta figure out what shit has to come first (or used to anyways) |
| 20:53.55 | anonymouz666 | another thing.... when you make install zaptel... it always try to get a firmware file through wget |
| 20:54.08 | JerJer | anonymouz666: same with sound files |
| 20:54.14 | Qwell | how else is it going to get it? |
| 20:54.18 | JerJer | in asterisk tree |
| 20:54.27 | Qwell | you'd rather have the Asterisk tree be 3GB? |
| 20:54.47 | JerJer | didn't say it was a good or bad thing - just pointing it out |
| 20:55.37 | *** join/#asterisk spokra (n=spokra@74-61-17-236.sea.clearwire-dns.net) |
| 20:58.09 | phix | :D |
| 21:01.12 | Corydon76-dig | JerJer: some of the modules now do self-resolution |
| 21:01.46 | Corydon76-dig | e.g. app_stack now loads res_agi |
| 21:02.16 | Corydon76-dig | or if res_agi isn't built, app_stack will turn part of its functionality off |
| 21:02.37 | JerJer | great |
| 21:03.08 | Corydon76-dig | app_dial does the same thing for chan_local |
| 21:04.14 | anonymouz666 | Corydon76-dig: I am fighting with write_mysql_logger() that works on 32-bit and crashes on 64-bit... do you already have any issues with 64-bit stuff? is there a usual reason for that? |
| 21:04.44 | angryuser | anonymouz666: you can disable it into it's configuration meny |
| 21:04.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:05.14 | angryuser | i the firmware download |
| 21:05.15 | Corydon76-dig | I'm not familiar with write_mysql_logger |
| 21:05.36 | anonymouz666 | this is my own shit that ast_queue_log uses write_mysql_logger to put everything into a database... it looks like something that version 1.6.0 already has |
| 21:06.07 | Corydon76-dig | Most likely, you have one of your parameters the wrong bit-width |
| 21:06.18 | iXombie | Is there a way to add a Ring Group to the Directory? |
| 21:07.02 | Corydon76-dig | iXombie: sure, just create an extension in your dialplan that matches up to an entry in voicemail.conf |
| 21:07.35 | iXombie | Ahha! Thanks! |
| 21:07.35 | Corydon76-dig | Nothing says that an extension must only ring a single channel |
| 21:08.15 | Corydon76-dig | That is one of the reasons why there is a separation between the two concepts of extensions and channels |
| 21:08.59 | iXombie | i see |
| 21:13.07 | rasterix | can include => exist within the general and global contexts? |
| 21:13.57 | rasterix | i know #include can exist anywhere... |
| 21:14.05 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:14.09 | Qwell | ~tias |
| 21:14.09 | jbot | Try it and see! |
| 21:14.10 | rasterix | within any conf ile |
| 21:14.21 | anonymouz666 | [TK]D-Fender is back! |
| 21:14.28 | anonymouz666 | from home now |
| 21:14.39 | rasterix | qwell: lol i was hoping corydon might answer |
| 21:15.14 | Qwell | You've been asking the same question for 2 days, after having been given an answer each time... |
| 21:15.38 | rasterix | not true i asked it for the first time today |
| 21:15.49 | rasterix | about an hour ago |
| 21:16.11 | Qwell | sure about that? |
| 21:16.12 | Qwell | ~logs |
| 21:16.13 | jbot | All conversations are logged to http://ibot.rikers.org/channel, where "channel" is replaced by the URL-encoded channel name, such as %23freenode for #freenode. Lines starting with spaces are not logged. |
| 21:16.16 | rasterix | yup |
| 21:16.43 | anonymouz666 | asterix -r |
| 21:16.47 | anonymouz666 | is rasterix |
| 21:17.16 | rasterix | anonymouz666 < fender |
| 21:17.25 | rasterix | fender < rasterix |
| 21:18.44 | anonymouz666 | wtf |
| 21:20.03 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 21:21.11 | rasterix | rasterix = reference to a popular uk cartoon character... its not spelt "asterisk" |
| 21:21.39 | *** join/#asterisk tvirus (i=TheVirus@c-68-54-165-28.hsd1.md.comcast.net) |
| 21:22.06 | tvirus | I have a recording for a voice menu, but I have to wait until it finishes playing before it accepts tones, any way around that? |
| 21:22.07 | [TK]D-Fender | rasterix: that'd be "Asterix" and French, not UK |
| 21:22.22 | [TK]D-Fender | tvirus: Use Background, not Playback |
| 21:22.41 | rasterix | either way its not a claim to being some kind of asterisk guru |
| 21:22.49 | rasterix | if thats whats upsetting people |
| 21:23.08 | Assid | anyone know a good termination provider similar to rapidvox.. good pricing and quality? |
| 21:23.11 | tvirus | Ah |
| 21:23.14 | tvirus | Thanks :) |
| 21:23.15 | Assid | i need to setup some alternatives |
| 21:23.16 | [TK]D-Fender | rasterix: I'm sure its not your nick. |
| 21:23.22 | Assid | tried and tested recommendations |
| 21:23.47 | tvirus | You're the best, [TK]D-Fender |
| 21:24.13 | rasterix | fender: *shrug* |
| 21:25.15 | angryuser | im off |
| 21:26.39 | *** join/#asterisk Maxous (n=Maxous@76.97.26.56) |
| 21:28.20 | *** part/#asterisk Maxous (n=Maxous@76.97.26.56) |
| 21:30.36 | *** part/#asterisk zerohalo (n=zeroHalo@75.150.77.161) |
| 21:33.18 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 21:36.24 | nhuisman_work | does anyone know if there are any web guis or scripts setup for automatic provisioning of cisco phones? |
| 21:36.44 | *** join/#asterisk jip|werk (n=asdf@h66-38-133-13.gtconnect.net) |
| 21:37.22 | rasterix | < cant afford a cisco phone so doesnt care |
| 21:37.57 | nhuisman_work | we already had them from our silly ccm setup. |
| 21:38.02 | nhuisman_work | all my new phones are going to be polycom |
| 21:38.19 | [TK]D-Fender | nhuisman_work: Sounds easy enough to make. I don't any free ones off-hand. I know trixbox bundles one for polycom, not sure if it supports others as well |
| 21:38.27 | rasterix | nhuisman did u try the linksys phones? |
| 21:38.28 | nhuisman_work | yeah digium is all about polycom too |
| 21:38.50 | *** join/#asterisk Lanh (i=lanh@unaffiliated/lanh) |
| 21:39.06 | nhuisman_work | there is a framework there for other phone types but the actual config generation is apparently done in res_phoneprov.so |
| 21:39.14 | nhuisman_work | which I don't believe is open source |
| 21:39.21 | Qwell | nhuisman_work: it's in trunk/1.6 |
| 21:39.27 | nhuisman_work | Qwell, oh is it. |
| 21:39.28 | [TK]D-Fender | nhuisman_work: Still pretty easy to make a config builder. |
| 21:39.31 | Qwell | pretty sure |
| 21:39.47 | Qwell | yeah |
| 21:39.49 | nhuisman_work | [TK]D-Fender, Sure just checking to see if something already exists first |
| 21:39.50 | Lanh | Hi guys, quick question, does anybody know of anywhere I can get a cheap hosted Asterisk box pre-installed, I am shopping around at the moment? |
| 21:40.44 | Lanh | Actually hold that thought, I have to go get some food before the shop shuts, I shall be back |
| 21:40.48 | nhuisman_work | Lanh, I think lots of companies do that. |
| 21:42.08 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 21:46.37 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:48.21 | nhuisman_work | Qwell, do you know how that res_phoneprov.so works? Is that for asterisknow and the gui only? |
| 21:48.33 | nhuisman_work | or is there documentation in the source file? |
| 21:50.15 | *** join/#asterisk Levonk (n=lk@adsl-76-243-65-122.dsl.lsan03.sbcglobal.net) |
| 21:55.13 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
| 21:57.24 | nhuisman_work | this is a silly question but what is the default codec cisco phones use? |
| 21:57.52 | nhuisman_work | g.711 ? |
| 21:59.59 | *** join/#asterisk devhen (n=devhen@70-58-99-185.slkc.qwest.net) |
| 22:01.52 | *** join/#asterisk spokra (n=spokra@74-61-17-236.sea.clearwire-dns.net) |
| 22:07.11 | Lanh | Sorry that I hit and run earlier guys, that's not usually my style but the time crept up on me suddenly |
| 22:07.29 | Lanh | Do you know of any around the $10-15/mo mark nhuisman_work? |
| 22:07.44 | nhuisman_work | Lanh, how many users? |
| 22:07.59 | Lanh | no more than 5 |
| 22:11.43 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:14.09 | mchou | Lanh: what are you talking about? ITSPs? |
| 22:15.21 | Lanh | I'm talking about something like what Lylix do, only with the initial installation done |
| 22:15.49 | mchou | Lanh: ok. no info on that one, sorry |
| 22:16.42 | Lanh | it seems Lylix want an additional $20/mo for the life of your account simply for installing trixbox/asterisk, at least that's how it appears to me anyway, other than that the packages seem identical |
| 22:17.28 | bijit | does anyone know where I can see the logs for e1? On a system. |
| 22:18.02 | nhuisman_work | e1 ? |
| 22:18.09 | nhuisman_work | like european t1 ? |
| 22:18.21 | mchou | Lanh: I thought they all virtually charge around $30/mo, more or less |
| 22:19.20 | bijit | yes |
| 22:19.23 | nhuisman_work | bijit, not sure what type of entries are you looking for? |
| 22:19.26 | Lanh | they do? might be time to get another small VPS and learn the installation myself if that's the case |
| 22:19.53 | bijit | I am trying to see the logs for the E1 errors. |
| 22:20.20 | *** join/#asterisk blepsoaf (n=pbaker@ool-43562836.dyn.optonline.net) |
| 22:20.50 | blepsoaf | is DAHDI only available via SVN? |
| 22:21.25 | blepsoaf | when doing menuselect for 1.6 I notice that somethings like app_meetme depend on DAHDI |
| 22:23.08 | TJNII | Dahdi is zaptel. The name was chenged because of a trademark issue. |
| 22:23.18 | nhuisman_work | bijit, I would try grep -i chan_zap /var/log/asterisk/messages |
| 22:24.02 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 22:24.22 | blepsoaf | TJNII: yeah I understand that, just a bit confused as to why its not being released on the asterisk.org downloads page |
| 22:24.50 | nhuisman_work | bijit, I have messages => notice,warning,error,verbose in my logger.conf |
| 22:24.56 | nhuisman_work | so the console messages goto my logfile |
| 22:25.39 | bijit | nhuisman_work: ty I will try that. |
| 22:26.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:33.10 | TJNII | blepsoaf: Will the zaptel package on the downloads page work? |
| 22:33.48 | blepsoaf | TJNII: of course it does, but it doesnt resolve the dependencies for the asterisk 1.6 download on that page... |
| 22:36.35 | *** join/#asterisk genji1981 (n=gej@203.152.122.50) |
| 22:38.13 | blepsoaf | nevermind found it @ http://downloads.digium.com/pub/telephony/dahdi-linux |
| 22:39.23 | Lanh | Asterisknow is the all in one package with the Gui right? |
| 22:42.17 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 22:44.34 | *** join/#asterisk pud (n=jkatka@sea01-v502-nat.marchex.com) |
| 22:46.27 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 22:46.27 | lmadsen | Lanh: yes |
| 22:46.39 | Lanh | cool, thanks lmadsen |
| 22:51.29 | bkruse | Lanh: I would recommend updating the GUI once you have installed it, however |
| 22:51.36 | bkruse | (http://asterisknow.org/install-related) |
| 22:55.28 | *** join/#asterisk route_ (i=skarecro@onix.kaimenassociates.com) |
| 22:56.40 | route_ | Hi there, I'm running PBX In A Flash, and I have 1 remote phone that's not getting any audio at all. I have ports 5060 and 10000-20000 forwarded directly to the phone's IP, and still no sound at all. Can somebody help me try to troubleshoot this phone? Other remote phones work flawlessly. |
| 23:01.21 | Lanh | Seems easy enough on the instructions :) |
| 23:05.25 | scooby2 | is this still the proper way to set GROUP in 1.4.x? Set(GROUP()=SALES) |
| 23:05.33 | seanbright | route_: what is pbx in a flash? |
| 23:06.07 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 23:07.16 | irieKen | Does anyone know what "Got SIP response 400 "Bad Request" back from xxxxx " would indicate?:D |
| 23:09.15 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:09.26 | *** join/#asterisk EI5GTB-macbook (n=EI5GTB@212.2.180.40) |
| 23:13.11 | scooby2 | why wont zapscan work |
| 23:13.34 | irieKen | No idea. |
| 23:15.56 | rasterix | if you set a variable in the [globals] context like myvar= blah, is the whitespace included in the variable value? |
| 23:16.12 | rasterix | ugh |
| 23:16.36 | rasterix | <PROTECTED> |
| 23:18.00 | irieKen | rasterix: Are you responding to me? |
| 23:18.12 | rasterix | no |
| 23:18.16 | irieKen | k |
| 23:21.37 | *** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-e3bb9ebcab8dc2bf) |
| 23:21.53 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-138-70.neo.res.rr.com) |
| 23:23.38 | TJNII | rasterix: AFAIK the answer is yes |
| 23:24.35 | rasterix | because older syntax pstn => intendedvalue |
| 23:25.09 | rasterix | nearly always includes whitespace following the assignment operator which is not intended (i think) to be part of the variable value |
| 23:28.28 | *** join/#asterisk UD (i=OHNOUDID@unaffiliated/underdawg) |
| 23:30.46 | esaym-acer | how to stop asterisk from spamming up my dns server with srv lookups? |
| 23:31.12 | UD | anyone try to load up that xorcom-rapid livecd? |
| 23:33.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:33.39 | UD | It starts to load for me then I get "/bin/sh: can't access tty; job control turned off" then "(initramfs)" prompt |
| 23:33.55 | *** join/#asterisk keith4_ (n=kbe2@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 23:34.07 | keith4_ | what's that decent ATA that's not too expensive? a linksys maybe? |
| 23:34.21 | keith4_ | is it the PAP2? |
| 23:34.53 | drmessano | yeah |
| 23:35.12 | keith4_ | ok |
| 23:35.53 | nhuisman_work | yeah I use those now |
| 23:35.53 | nhuisman_work | work fine |
| 23:36.35 | keith4_ | why are there "unlocked" PAP2-NA devices on ebay? |
| 23:37.00 | JT | why not? |
| 23:37.07 | keith4_ | what could possibly be "locked" about it? |
| 23:37.48 | unpaidbill | do you have to do something different in 1.6 to originate a fax with SendFAX? i am getting 'We could NOT get the channel lock for SIP/server1-09928920' 'SIP transaction failed: ...' |
| 23:38.03 | unpaidbill | but dialing with a SIP client appears to work fine |
| 23:39.12 | JT | keith4_: all the ones from ITSPs are locked |
| 23:39.18 | JT | keith4_: as they are subsidised |
| 23:39.23 | keith4_ | ugh |
| 23:39.30 | JT | so you don't just get an ATA less than cost price |
| 23:39.34 | JT | then change provider |
| 23:39.44 | JT | as that would make them lose money |
| 23:40.17 | keith4_ | true |
| 23:40.28 | keith4_ | does anyone make an FXO ATA? |
| 23:41.14 | _ShrikE | keith4: audiocodes makes a few fxo models that work pretty good IMO. |
| 23:43.19 | drmessano | Linksys does |
| 23:43.23 | drmessano | SPA-3102 |
| 23:43.27 | drmessano | Works great |
| 23:43.37 | drmessano | 1 FXS, 1 FXO |
| 23:44.41 | keith4_ | excellent |
| 23:46.16 | esaym-acer | n/m srvlookup = no was missing on a box... |
| 23:46.37 | *** join/#asterisk DarrenCT (n=darren@adsl-065-007-146-085.sip.asm.bellsouth.net) |
| 23:46.52 | DarrenCT | is there a way to use Skype with Asterisk? |
| 23:47.16 | keith4_ | there are several. none of them are great |
| 23:47.40 | keith4_ | google "asterisk skype" |
| 23:47.46 | DarrenCT | I want to use my Skype account throughout the house. Is this the way to do it? |
| 23:48.15 | DarrenCT | ... searching |
| 23:48.32 | rob0 | Hey I saw a Linksys Vonage ATA on the discount table at walmart, is it possible to unlock those? |
| 23:49.31 | rasterix | to unlock walmart? |
| 23:49.38 | rasterix | i think u will get arrested |
| 23:49.49 | nhuisman_work | weird how the gui version of asterisk doesn't actually use extensions.conf |
| 23:49.57 | nhuisman_work | just users.conf and then it generates the dialplan on the fly |
| 23:50.09 | lmadsen | nhuisman_work: uh... ya... |
| 23:50.34 | keith4_ | try #asterisk-gui |
| 23:50.43 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:50.44 | nhuisman_work | lmadsen, maybe there is a doc I should have read that tells you this? |
| 23:51.21 | nhuisman_work | don't tell me that's in the book |
| 23:51.22 | nhuisman_work | laugh |
| 23:52.00 | rasterix | i came in here the other day after my gf ditched me and they r like wanna know why... its in the book |
| 23:52.04 | nhuisman_work | hmm doesn't apear to be |
| 23:52.30 | nhuisman_work | ah wait there is one sentence that implies it I guess |
| 23:53.13 | nhuisman_work | the confusing part to me is that it doesn't ONLY use the users.conf, conference extensions, voicemail extensions, etc are in the extensions.conf |
| 23:54.23 | rasterix | *sniggers* |
| 23:54.37 | keith4_ | wait. the SPA3102 is a router? ew |
| 23:54.54 | keith4_ | wtf does that even mean? |
| 23:56.11 | nhuisman_work | probably that it can also route ip traffic |
| 23:57.46 | lanning | it works as a broadband home router/firewall |
| 23:59.09 | nhuisman_work | which is useful if you don't already have one, but yeah. |
| 23:59.44 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 23:59.56 | pcrane | does anyone know if there's a fonailty IRC channel? |