00:05.29 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
00:05.48 | Shotta30five | St, Petersburg.... Used to be broward county until I started USF |
00:06.26 | Shotta30five | rene: What you think about shootwall firewall for the asterisk box |
00:06.31 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.10) |
00:09.33 | rene- | Shotta30five: i have never used it, people talk good things about it |
00:10.02 | rene- | but first try connecting with no firewall |
00:11.27 | Shotta30five | Will let you know my progress.. |
00:11.41 | nain | Hi |
00:13.31 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
00:14.14 | *** join/#asterisk redcap1 (n=phez@redcap.xs4all.nl) |
00:14.36 | *** join/#asterisk suma (n=suma@222.165.112.215) |
00:15.19 | suma | Where can I get the list of asterisk actions to use in asterisk IVR C Program |
00:18.35 | DoktorGreg | ok, pri question |
00:18.48 | rene- | hi nain |
00:18.52 | DoktorGreg | i figured how to dig into MICS system, the proper menu system |
00:19.05 | nain | I am getting -- Transmitting RFC2833 on payload 101 |
00:19.06 | nain | Ouch ... error while writing audio data: : Broken pipe |
00:19.06 | nain | Segmentation fault |
00:19.08 | DoktorGreg | my PRI line..??? doesnt support d channel |
00:19.16 | Qwell | DoktorGreg: Then it isn't PRI |
00:19.30 | nain | Can any one know why this call failed and asterisk crashed on h323 call, while sip is working fine |
00:19.31 | DoktorGreg | no, in the MICS, it says its PRI |
00:19.41 | DoktorGreg | but for D channel it says... None |
00:19.43 | Qwell | PRI has D channel |
00:19.46 | Qwell | So set one |
00:20.56 | file | if anyone in here has bugs on mantis they want me to look at, or I replied to... well... say your number now or forever hold your peace and quiet! |
00:22.03 | *** part/#asterisk rene- (n=rene-@dsl-201-128-115-107.prod-infinitum.com.mx) |
00:22.37 | drray | is your PRI going to a channel bank first and then into your asterisk box? |
00:22.41 | Qwell | file: close em all! |
00:23.25 | file | Qwell: marvelous idea |
00:24.00 | DoktorGreg | no |
00:24.13 | nain | Any one can let me know what's wrong with my setup |
00:24.21 | DoktorGreg | dirct from the csu/dsu unit from phone company |
00:24.27 | nain | while dialing h323 * crashed with this log |
00:24.28 | nain | -- Transmitting RFC2833 on payload 101 |
00:24.28 | nain | Ouch ... error while writing audio data: : Broken pipe |
00:24.28 | nain | Segmentation fault |
00:24.28 | nain | PK_Server:/etc/asterisk# Warning, flexibel rate not heavily tested! |
00:24.46 | drray | does it work without the d-channel? |
00:24.52 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
00:25.35 | wwalker | If I downloaded a bad config file into a polycom 501, and I now want it to not use the contents of the config, how do I reset the phone |
00:27.14 | wwalker | I'm stuck in a loop where it |
00:27.40 | wwalker | comes back with "config file error" "error is 0x4020" |
00:28.24 | [TK]D-Fender | wwalker : fix the file and reboot |
00:28.35 | file | nooooo don't fix me |
00:28.43 | wwalker | :) |
00:28.52 | Qwell | file: snip, snip |
00:29.19 | wwalker | I want the phone to NOT download an app. I want it to use the APP it has in flash. |
00:29.20 | file | go after Qwell! he's smaller, it won't matter! |
00:29.26 | Qwell | pfft |
00:31.04 | Qwell | blitzrage: where art thou? |
00:31.33 | Qwell | It's April 15th! Why isn't Astricon registration open? :( |
00:31.36 | wwalker | I don't have a sip.ld :( so I gave it a config file without a app_file_path entry |
00:32.16 | file | Qwell: who knows! |
00:35.34 | [TK]D-Fender | wwalker : You think you've screwed up your sip.ld load? |
00:35.36 | Shotta30five | Renee: Having it on the internet work |
00:35.39 | Shotta30five | thanx you |
00:36.14 | Shotta30five | Just got to figure out what is going on with the router |
00:37.21 | wwalker | [TK]D-Fender: no, not yet. I went back and added app_file_path and booted once (with no sip.ld to load) so now it seems to do Ok. |
00:37.46 | wwalker | It gets thru loading application, then loading sip.ld with no errors, but it then reboots |
00:38.28 | wwalker | looks like it made it this time... maybe... |
00:39.48 | [TK]D-Fender | wwalker : Sounds like you should reset your <mac>.cfg file to something normal and get the firmware in the folder where it belongs |
00:40.53 | wwalker | back to don't have the firmware :( |
00:41.07 | [TK]D-Fender | wwalker : Which version were you on? |
00:41.17 | wwalker | 1.6.2.0041 |
00:41.28 | wwalker | I get all the way to the Welcome screen. |
00:41.38 | wwalker | It shows the version. |
00:41.39 | [TK]D-Fender | wwalker : Ok, which version would you LIKE? |
00:41.46 | wwalker | 1.6.2.0041 |
00:42.07 | wwalker | I think sip.ld is fine. it doesn't like something in the config file it pulled down |
00:42.32 | *** join/#asterisk Anexs_and_Mai (n=Dartagna@pc-58-176-104-200.cm.vtr.net) |
00:42.41 | wwalker | How do I get the polycom to push it's config up to the server? |
00:42.47 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
00:43.05 | DoktorGreg | ok now im getting somewhere |
00:43.20 | DoktorGreg | i just made the phones print from * to mics |
00:43.28 | DoktorGreg | er ring |
00:43.40 | DoktorGreg | but |
00:43.50 | DoktorGreg | i cant do it a second time |
00:44.07 | DoktorGreg | cli gives me a all channels busy error |
00:44.42 | [TK]D-Fender | wwalker : typilcally it only updates the locally amde changes in <mac>-phone.cfg |
00:44.54 | [TK]D-Fender | wwalker : As an override to base. |
00:45.32 | wwalker | <PROTECTED> |
00:46.10 | [TK]D-Fender | wwalker : Though you really DON'T want the phone thinking it knows better... thats the POINT of provisioning them. |
00:46.29 | luke-jr_ | Is there a website that explains what "SNFC CNTRL" and similar 'city names' are? |
00:46.30 | wwalker | I agree, can't convince the client |
00:46.45 | [TK]D-Fender | wwalker : Have you tried blunt trauma? ;) |
00:47.10 | wwalker | :) |
00:48.48 | wwalker | back in an hour, dinner calls |
00:57.13 | luke-jr_ | BTW, SellVoIP lets you use both IAX2 and SIP for calls ;) |
00:57.30 | kamileon | whoa i found a FXS module sitting in a drawer! |
00:57.34 | luke-jr_ | which is potentially useful for supporting reinvites between different outgoing services |
00:58.27 | luke-jr_ | Is it possible to set 'use reinvites' in the dialplan? |
01:01.21 | *** join/#asterisk jofre (n=jofre@201.2.192.43) |
01:02.04 | luke-jr_ | in particular, I'd like to disable all reinviting for local destination calls (which are recorded) and reinvite anything goes back out over the net |
01:02.41 | [TK]D-Fender | luke-jr_ : Nope... once a calls path is decided you can't jsut go and grab it back... |
01:04.01 | SplasPood | Question, when using Agents /w AgentCallbackLogin and ackcall=yes, where do you define the announcement played *before* the callee hits '#' |
01:06.20 | [TK]D-Fender | SplasPood : I believe ackcall only applies to "AgentLogin" users |
01:07.37 | [TK]D-Fender | since they're already "on the call" already as a warning. |
01:07.58 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net) |
01:08.01 | *** join/#asterisk hfb (n=hfb@adsl-69-231-83-94.dsl.irvnca.pacbell.net) |
01:08.10 | Qwell | agents are silly |
01:08.20 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
01:08.21 | Qwell | Why would they be used over queue members? |
01:08.31 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
01:08.47 | [TK]D-Fender | Qwell : Useful in cases where you want to trigger events what the agent is called I guess. |
01:09.00 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
01:09.05 | [TK]D-Fender | Qwell : that style was how I was going to implement "screen-pops" for mine |
01:09.23 | *** join/#asterisk websae_ (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
01:09.41 | Qwell | manager can send for queue members too... |
01:10.54 | [TK]D-Fender | Qwell : yeah, but thats an approach that'd require more programming skills. |
01:11.24 | SplasPood | [TK]D-Fender: nope, its the opposite.. it def works.. waits for me to hit '#', its just all slient before that... |
01:11.50 | [TK]D-Fender | SplasPood : that blows... hope you rigged the callerID before passing off the calls o you know what to do :) |
01:12.17 | SplasPood | [TK]D-Fender: After '#' it plays an announcement defining the name of the queue so people know how to react |
01:12.24 | *** join/#asterisk phrog123 (n=francois@ns.menards.ca) |
01:12.29 | SplasPood | I'd like something before the '#' that says "Please press '#' to accept this call" |
01:14.15 | phrog123 | folks, with Asterisk at Home, can you tell me which SIP header on an inbound SIP connection from another asterisk box (not trunked) binds to the exten command in extensions.conf ? |
01:14.33 | Qwell | To: ? |
01:14.55 | phrog123 | I've spent more than 4 hours today trying every possible permutation of DID |
01:15.20 | mitcheloc | phrog123: what for? |
01:15.24 | phrog123 | what I have in my log is that the other server sends the call as s@192.168.0.1 |
01:15.26 | SplasPood | Qwell: Agents allow a given person to sit down at any phone and then take calls.. without just having all phones ring |
01:15.40 | Qwell | SplasPood: like AddQueueMember? |
01:16.07 | phrog123 | I want a rule for trapping that DID so that I can have a different IVR on my SIP account dialing into my * box than from my digium board |
01:16.31 | SplasPood | Qwell: Well I suppose you could put something together with that, but why use that OVER agents? |
01:16.44 | Qwell | SplasPood: Because, Agents are lame :p |
01:16.50 | Qwell | not as flexible, IMO |
01:17.21 | phrog123 | It is as if the DID is not recognized by asterisk, yet the calls comes in and rings my IP phones just fines, but throught the default rule which is the same than a call coming over POTS and ringing my IP phones |
01:17.25 | SplasPood | hrm... what type of flexibility? |
01:17.28 | luke-jr_ | [TK]D-Fender: this is before the path is decided |
01:17.34 | luke-jr_ | eg, before I execute Dial |
01:17.53 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net) |
01:17.54 | phrog123 | Of course, I had to do the normal extensions.conf tweak (wonder why this is not yet enableable through the Web UI |
01:18.02 | [TK]D-Fender | luke-jr_ : perhaps you could set up 2 peer entries, each with the same connection details, only different rules... |
01:18.08 | luke-jr_ | :/ |
01:18.25 | luke-jr_ | hackish |
01:18.33 | phrog123 | anyways, the DID does not work, so I want to trace the SIP flow and figure out what's missing in the SIP header so that I can check with the ISP who's providing me that DID |
01:19.06 | luke-jr_ | phrog123: Ethereal? |
01:19.29 | phrog123 | why? |
01:19.45 | phrog123 | sip debug peer myispaccount spits out what I want |
01:20.26 | mitcheloc | does ethereal cook and clean? |
01:20.47 | Qwell | luke-jr_: Wanna record me a pcap file? :p |
01:20.51 | phrog123 | I just do not know what * binds on exten=>5555551212 = what in the SIP message ??? from: 5555551212@sip.sipprovider.com? |
01:20.56 | phrog123 | Is this the from field? |
01:20.58 | [TK]D-Fender | mitcheloc : You've got to keep current with your plug-ins! |
01:21.04 | Qwell | phrog123: To: |
01:21.06 | macTijn | mitcheloc: yes, and it vacuums too! |
01:21.36 | luke-jr_ | Qwell: no, why? |
01:21.39 | phrog123 | Qwell: to doesn't make sense, the to is the telephone number being dialled, not the DID |
01:21.42 | Qwell | luke-jr_: because I need one, heh |
01:21.46 | luke-jr_ | Qwell: of what? |
01:21.46 | Qwell | phrog123: duh? |
01:21.58 | Qwell | You don't match the number dialed to the From: |
01:22.08 | Qwell | luke-jr_: about 10 seconds of a gsm (or g729) rtp stream |
01:22.20 | phrog123 | ok, assuming this is the case, is it what's at the left of the @ |
01:22.29 | luke-jr_ | Qwell: why can't you do that yourself? O.o |
01:22.40 | phrog123 | does it matter what's right of the @? |
01:22.43 | Qwell | phrog123: exten => _NXXNXXXXXX/5555551212,1,Blah() |
01:22.49 | Qwell | luke-jr_: don't know how :( |
01:22.59 | luke-jr_ | Qwell: ... |
01:23.07 | Qwell | I'm an ethereal newb :p |
01:23.21 | *** join/#asterisk TTT_Travis (n=Travis@bal-broadband2-ws-14.dsl.airstreamcomm.net) |
01:23.32 | luke-jr_ | Qwell: the problem is, I don't know enough about RTP to know that keys/passwords won't be crackable from it =p |
01:23.56 | luke-jr_ | and also that I use ulaw... |
01:24.00 | Qwell | meh :p |
01:24.04 | phrog123 | if the * box on my ISP's side sends me a to: s@x.y.z.w (public IP address), can I assume that this is where the problem is? |
01:24.13 | phrog123 | wtf is s@ ? |
01:24.17 | Qwell | exten s |
01:24.25 | luke-jr_ | phrog123: that means you didn't give them an extension to call |
01:24.27 | Qwell | Then thing you're adding at the end of your register => line |
01:24.31 | TTT_Travis | hi guys, I'm interested in learning about asterisk so I though I'd take the beginners path and install Asterisk@home, I have a Rockwell Voice Modem do you think it will be possible to make this work with Asterisk without too much effort? |
01:24.44 | Qwell | TTT_Travis: As what? |
01:24.50 | Shotta30five | I think i found out why my Asterisk Box was not working with external client |
01:24.51 | Qwell | a modem? sure |
01:25.03 | luke-jr_ | when I register, how can I specify the username the remote side should authenticate with when placing calls? |
01:25.18 | Shotta30five | The router I was using was a Linksys ATA that I wasn't using |
01:25.33 | TTT_Travis | Qwell well I want to connect it to my phoneline in my house and then for example beable to pickup the phone and dial an extension and stuff |
01:25.38 | luke-jr_ | TTT_Travis: a modem will only work for a phone line, not for a phone itself |
01:25.40 | Qwell | TTT_Travis: no |
01:25.44 | Shotta30five | So make a quick Smoothwall Firewall and now it works like a charm |
01:25.48 | Qwell | TTT_Travis: It'll be a modem, thats about it |
01:25.55 | Qwell | unless you can write zaptel fxo drivers for it |
01:26.15 | TTT_Travis | how come with like windows I can dial numbers on it and it will dial it on my phoneline? |
01:26.17 | luke-jr_ | Qwell: don't those exist for Rockwell? |
01:26.32 | Qwell | TTT_Travis: because there are drivers |
01:26.35 | tainted- | anyone know how to speed up boot process with polycom 301s |
01:26.35 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net) |
01:26.47 | luke-jr_ | TTT_Travis: will it transmit voice on windows too? |
01:26.47 | phrog123 | what does it mean if I'm getting an invite from another * box which is s@IP.IP.IP.IP ? |
01:26.59 | TTT_Travis | luke-jr_ not really sure |
01:27.02 | TTT_Travis | I think it would |
01:27.05 | TTT_Travis | since its a voice modem |
01:27.17 | luke-jr_ | what's that mean? |
01:27.24 | luke-jr_ | modem != voice |
01:27.25 | Qwell | it means they can charge $45 for it |
01:27.28 | luke-jr_ | they're kinda exclusive |
01:27.51 | TTT_Travis | it says on the modem that its a Voice/Fax Modem |
01:28.01 | Qwell | It's a winmodem. Nothing more, nothing less |
01:28.06 | phrog123 | if my register line is not a telephone number, but rather a username:passsword@sip.sipprovider.com? Does this mess up asterisk insofar as parsing a DID? |
01:28.08 | TTT_Travis | so its impossible? |
01:28.10 | Qwell | Whatever extra "stuff" it can do, is done in software |
01:28.16 | Qwell | TTT_Travis: no, you just need to write a driver |
01:28.23 | TTT_Travis | so impossible ;) |
01:28.30 | luke-jr_ | TTT_Travis: not impossible if its a winmodem |
01:28.36 | phrog123 | the problem is that this provider, the username is not the 10 digit number, but rather an account name |
01:28.58 | TTT_Travis | luke-jr_ what do you mean by winmodem? |
01:28.58 | luke-jr_ | TTT_Travis: if nobody's written a driver, you can either write one yourself or hire someone to |
01:29.08 | Qwell | phrog123: It will send the DID in the To: header |
01:29.16 | luke-jr_ | TTT_Travis: software modem |
01:29.19 | TTT_Travis | k |
01:29.28 | TTT_Travis | I don't know if it is |
01:29.39 | Qwell | TTT_Travis: Did you buy it after 1999? |
01:29.42 | TTT_Travis | I am new to think kind of stuff |
01:29.50 | TTT_Travis | Qwell I am guessing right around 2000 |
01:30.02 | TTT_Travis | I got this computer from someone that didn't want it |
01:30.07 | TTT_Travis | and that was the card that was in there |
01:30.08 | Qwell | Then there is a fairly high chance that it is |
01:30.18 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
01:30.23 | Qwell | jbot: tell phrog123 pastebin |
01:30.26 | luke-jr_ | channels/chan_modem_aopen.c: * A/Open ITU-56/2 Voice Modem Driver (Rockwell, IS-101, and others) |
01:30.58 | TTT_Travis | fair chance that it is what? |
01:31.03 | Qwell | a winmodem |
01:31.15 | TTT_Travis | luke-jr_ whats that about rockwell? |
01:31.19 | TTT_Travis | it has a rockwell chip on it |
01:31.25 | luke-jr_ | TTT_Travis: if you want to buy a non-winmodem, you'd need to spend extra money and time looking for it |
01:31.41 | luke-jr_ | TTT_Travis: apparently, the chan_modem_aopen module supports Rockwell modems |
01:31.50 | Qwell | chan_modem is dead |
01:32.00 | luke-jr_ | ... |
01:32.20 | luke-jr_ | dead how? |
01:32.20 | Qwell | It won't be there in 1.4 |
01:32.41 | TTT_Travis | so there is a chance it might work? |
01:32.45 | luke-jr_ | what replaces it and provides the same features? |
01:32.57 | Qwell | luke-jr_: chan_zap? |
01:33.08 | luke-jr_ | Qwell: chan_zap supports regular winmodems? |
01:33.12 | Qwell | no |
01:33.29 | luke-jr_ | so chan_modem is being removed because it cuts into Digium's sales? |
01:34.03 | Qwell | I'm pretty sure zap supports other hardware |
01:34.12 | luke-jr_ | what version will TTT_Travis be stuck with if he wants to use chan_modem? |
01:34.19 | Qwell | 1.2 |
01:34.28 | TTT_Travis | will the latest Asterisk@Home work? |
01:34.34 | marcus2 | zap supports intel winmodems |
01:34.37 | luke-jr_ | TTT_Travis: @Home gets no support here |
01:34.45 | TTT_Travis | is there a channel for it? |
01:34.51 | luke-jr_ | see topic |
01:34.59 | luke-jr_ | tho it's a waste of time, IMO |
01:35.07 | TTT_Travis | why? |
01:35.17 | marcus2 | but there has definitely been some questionable behavior in the past with regards to digium natively supporting cards that they dont get profits from |
01:35.18 | luke-jr_ | it's user friendly |
01:35.25 | TTT_Travis | whats so bad about that? |
01:35.37 | luke-jr_ | it's not smart-people friendly |
01:35.46 | marcus2 | one big reason that it sucks is that it is based on centos |
01:36.13 | Qwell | The reason *@~ is crap, is because AMP is crap, as is phpmyadmin |
01:36.49 | Qwell | phrog123: 1) See channel topic. |
01:36.56 | Qwell | phrog123: 2) Unless you're paying me, don't msg me |
01:37.05 | TTT_Travis | the rockwell chip says this: RCV336ACF/SP R6749-21 ? Rockwell 96 9742 B31697-4 MEXICO |
01:37.14 | TTT_Travis | so its a pretty old card |
01:37.17 | TTT_Travis | its not PCI |
01:37.28 | Qwell | TTT_Travis: Should've said it wasn't PCI before |
01:37.30 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
01:37.31 | TTT_Travis | its the long black slots |
01:37.36 | Qwell | It's almost definitely not a winmodem |
01:37.39 | TTT_Travis | k |
01:37.42 | Nugget | the "long black slots". |
01:37.46 | Nugget | oof. |
01:37.46 | TTT_Travis | lol |
01:37.49 | iceyp | hey guys, been some time sicne i setup voicemail... how does one add a new mailbox to the system? i justed added it to voicemail.conf |
01:37.52 | TTT_Travis | they were before my time |
01:38.01 | luke-jr_ | ISA |
01:38.06 | TTT_Travis | yeah |
01:38.06 | Qwell | iceyp: That's all |
01:38.07 | TTT_Travis | thats it |
01:38.16 | iceyp | Qwell mmm, doesnt appear to be working |
01:38.22 | Qwell | Did you reload? |
01:38.27 | iceyp | yea |
01:38.35 | TTT_Travis | so what would I need to use this Chanzap driver? |
01:38.36 | luke-jr_ | but |
01:38.44 | luke-jr_ | if it's not a winmodem, how does it claim 'voice'? |
01:38.46 | Qwell | TTT_Travis: Get real telephony hardware |
01:38.47 | iceyp | <PROTECTED> |
01:38.52 | iceyp | <PROTECTED> |
01:39.10 | TTT_Travis | Qwell I'm working with what I got for now |
01:39.22 | TTT_Travis | but it says Fax and Voice modem card |
01:39.22 | Qwell | TTT_Travis: well, it isn't going to work |
01:39.29 | TTT_Travis | ok |
01:39.37 | TTT_Travis | so what is the cheapest card I can get? |
01:39.38 | iceyp | can i start a mailbox with a 0 on the beginning? |
01:39.45 | phrog123 | if My ISP is giving me a To: <sip:pro01473@sip2.isptel.ca> in the SIP header rather than 5555551212@sip2.isptel.ca, does this mean that the extensions.conf entry should be more like exten => pro01773@,1,SetVar(FROM_DID=pro01473 rather than exten => 8196018096,1,SetVar(FROM_DID=8196018096) |
01:39.47 | Qwell | iceyp: sure |
01:40.00 | luke-jr_ | TTT_Travis: you could get a Linksys PAP2-NA for about $60 |
01:40.06 | TTT_Travis | ummm |
01:40.10 | iceyp | 099742910 => 1111,Barry SIP Phone,barry@unix.co.nz |
01:40.12 | TTT_Travis | yeah thats way more then I want to spend |
01:40.21 | Qwell | TTT_Travis: telephony isn't cheap |
01:40.25 | TTT_Travis | http://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=4300 |
01:40.29 | Qwell | Just because the software is free, doesn't mean hardware is |
01:40.33 | TTT_Travis | that says |
01:40.34 | TTT_Travis | "Laptops4me.com sells the same cards for US $7.70 each, which is hard to beat. " |
01:40.43 | TTT_Travis | but when I click on the link its dead |
01:41.03 | iceyp | ahh i think its the context i using :) |
01:41.23 | luke-jr_ | Qwell: also, chan_zap supports Intel because they're the same as Digium hardware, IIRC |
01:41.28 | TTT_Travis | how about X100P card from Digium |
01:41.35 | luke-jr_ | and it only supports intel when you hack it |
01:41.39 | Qwell | TTT_Travis: Digium hasn't sold those in a long time |
01:41.57 | iceyp | ok, all fixed |
01:42.07 | TTT_Travis | Qwell but they do work with Asterisk |
01:42.09 | TTT_Travis | correct? |
01:42.21 | TTT_Travis | yep |
01:42.40 | TTT_Travis | and they're like $15 |
01:44.50 | phrog123 | so folks, if account name is not a telephone number, will exten => accountname,1,... work? |
01:44.54 | iceyp | I have an issue from my cisco 7912 that when i make a call to my remote asterisk box for time, audio, voicemail or anything, it gets through and has no sound till i press something |
01:45.56 | Qwell | iceyp: skinny? |
01:45.58 | luke-jr_ | TTT_Travis: http://www.laptops4me.com/product_info.php/modem/all-56k-modems/p/i-bis-v-92-pci-intel-chip-voice-modem/cPath/176_239/products_id/5232 |
01:46.03 | iceyp | umm sip client |
01:46.09 | iceyp | i beleive |
01:46.26 | iceyp | if i make a call to an external number, like pdsn shes sweet |
01:46.31 | luke-jr_ | TTT_Travis: though I'd give the Rockwell chan_modem a try first anyway |
01:46.32 | TTT_Travis | luke-jr_ that doesn't work |
01:46.35 | TTT_Travis | k |
01:46.36 | TTT_Travis | I will |
01:46.40 | luke-jr_ | TTT_Travis: what doesn't work? |
01:47.06 | TTT_Travis | that link |
01:47.09 | luke-jr_ | no? |
01:47.12 | luke-jr_ | does for me |
01:47.27 | TTT_Travis | now it does |
01:47.28 | iceyp | Qwell yeah its sip based |
01:47.34 | TTT_Travis | so that card will work? |
01:47.44 | luke-jr_ | TTT_Travis: looks like it |
01:47.49 | luke-jr_ | but no guarantees |
01:48.07 | TTT_Travis | yeah |
01:48.13 | TTT_Travis | from what I see the intel based ones do |
01:48.25 | luke-jr_ | it has an intel chipset, note |
01:48.33 | *** join/#asterisk op3r (i=op3r@210.4.31.234) |
01:48.34 | TTT_Travis | they show up as generic clones to the X100P |
01:48.36 | luke-jr_ | looks like it's merely rebranded |
01:48.42 | TTT_Travis | well if my rockwell one fails then I will get that one |
01:48.43 | op3r | does anyone know the pricing of aheeva? |
01:48.53 | TTT_Travis | and hey if that one doesn't work I'm only out $10 |
01:49.24 | Qwell | and if it does work, you'll have a poorly performing PBX |
01:49.33 | Qwell | and WE'LL have to support you |
01:49.57 | iceyp | Qwell it would appear i get no voice until the pbx hears something, if i wisper or blow into the handset / speakerphone then voice comes through |
01:50.05 | TTT_Travis | what do you mean poorly performing? |
01:50.12 | TTT_Travis | will it just be bad quality? |
01:50.15 | Qwell | TTT_Travis: cheap, as in $10 |
01:50.19 | Qwell | It's a crap card |
01:50.32 | TTT_Travis | just so I can hear whats coming |
01:50.37 | luke-jr_ | X100P is a crap card |
01:50.37 | TTT_Travis | this is just for learning purposes |
01:50.38 | luke-jr_ | ? |
01:50.49 | Qwell | luke-jr_: umm...yes |
01:50.55 | op3r | Qwell: do you know the pricing for Aheeva? |
01:51.04 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
01:51.04 | luke-jr_ | what makes it crap? |
01:51.05 | Qwell | TTT_Travis: So, you're using *@~ to "learn" also? |
01:51.16 | TTT_Travis | I just want to play around |
01:51.34 | TTT_Travis | and according to what I've read Asterisk@home is the easiest way to start |
01:51.38 | Qwell | luke-jr_: the hardware? |
01:51.41 | luke-jr_ | TTT_Travis: learn how to use real Asterisk, not some dumb UI ;) |
01:52.17 | Qwell | iceyp: You must have the silence threshold thing |
01:52.27 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
01:52.56 | iceyp | Qwell on the phone it's self? |
01:52.56 | TTT_Travis | meh |
01:52.58 | iceyp | let me look |
01:53.01 | TTT_Travis | I'll start with the ui |
01:53.05 | TTT_Travis | then move onto the commandline |
01:53.27 | Qwell | iceyp: in voicemail.conf |
01:53.28 | [TK]D-Fender | TTT_Travis : It will NOT teach you * at all, just how to fill in a couple of stupid blank lines on a web-form. |
01:53.54 | iceyp | mmmm |
01:53.55 | TTT_Travis | if it works thats enough |
01:53.56 | De_Mon | [TK]D-Fender well, to be fair. You can create the dialplan and then SEE how to do it manually. |
01:54.06 | iceyp | silencethreshold=128 |
01:54.08 | mitcheloc | [TK]D-Fender: be nice now ;) |
01:54.16 | Qwell | De_Mon: You've obviously never seen an AMP config |
01:54.19 | TTT_Travis | this way I can still digg around the config files if I need too |
01:54.20 | Qwell | It's complete shit |
01:54.30 | op3r | I learned * by installing it and messing it up and editing confs on a production servers |
01:54.32 | op3r | :( |
01:54.32 | [TK]D-Fender | TTT_Travis : Yeah, in all likelyhood it'll work, just forget about learning anything of value or being able to truely control your PBX. |
01:54.35 | Qwell | takes about 50 lines to do anything at all |
01:54.40 | op3r | and I am still 0 clue about it thougj |
01:55.00 | iceyp | Qwell how can i turn it off |
01:55.02 | De_Mon | Qwell wow.. ok nevermind |
01:55.15 | Qwell | De_Mon: It's like 5 layers deep in pointless macros, etc |
01:55.17 | [TK]D-Fender | De_Mon : Yeah, and spend HOW long picking apart the AGI's, etc? And how is it that you'd even understand what it all means without oing a lot of it yourself OUTSIDE of AMP anyways? |
01:55.45 | Qwell | yeah, dumb ass AGIs too |
01:55.57 | mitcheloc | hey now, i'm working on a gui, they aren't *all* bad! |
01:56.09 | Qwell | mitcheloc: Yes they are. Even the one I'm writing :p |
01:56.14 | TTT_Travis | so how hard is chan_zap to setup? |
01:56.18 | mitcheloc | noo! mine isn't! |
01:56.19 | TTT_Travis | I just install zaptel and ? |
01:56.22 | [TK]D-Fender | Qwell : yup.... KNOWELDGABLE * users get lost in there... its like spaghetti code with meatballs and way too much parmesan. |
01:56.26 | Qwell | TTT_Travis: Eithout hardware? impossible |
01:56.34 | Qwell | Without* |
01:56.39 | [TK]D-Fender | TTT_Travis : Get compatible hardware first |
01:56.47 | iceyp | Qwell how can i turn the silence thing off? |
01:56.49 | iceyp | set it to 0? |
01:56.53 | [TK]D-Fender | TTT_Travis : They Rockwell wonder of yours is USELESS to *. |
01:56.55 | Qwell | iceyp: no, that's fine as it is |
01:56.59 | [TK]D-Fender | that* |
01:57.13 | Qwell | iceyp: minimum length? |
01:57.17 | iceyp | Qwell why do i have to blow or make a sound into the mic before it talks to me |
01:57.19 | iceyp | Qwell 10 |
01:57.24 | Qwell | That's why. :) |
01:57.34 | Qwell | It removes the first part, if it's 100% silence |
01:57.45 | iceyp | maxsilence=10 |
01:57.46 | iceyp | ? |
01:57.52 | Qwell | no |
01:58.01 | Qwell | So, your 2 seconds of blowing, after 8 seconds of silence, is only counted as 2 seconds |
01:58.14 | Qwell | I think that was how it worked |
01:58.27 | Qwell | or maxsilence |
01:58.27 | iceyp | i can wait 20 seconds with nothing till i make any sound into the handset then it talks to me |
01:58.46 | Qwell | minmessage |
01:58.54 | Qwell | minmessage and/or maxsilence can affect it |
01:59.15 | iceyp | mmm |
01:59.42 | iceyp | guess i just need to tell people that when they call the pbx for anything, just to make some sort of sound into the handset? |
01:59.50 | Qwell | or turn off those options |
02:00.01 | Qwell | and, no, those are only for voicemail |
02:00.33 | op3r | does anyone have any idea of the pricing of Aheeva? |
02:00.39 | iceyp | ;maxgreet=60got them all ;commented |
02:03.04 | TTT_Travis | [TK]D-Fender why is it useless? |
02:05.04 | luke-jr_ | TTT_Travis: he's assuming chan_modem won't work with it, which is probably true |
02:05.12 | luke-jr_ | but like I said, I'd give it a chance just in case |
02:05.18 | TTT_Travis | yeah |
02:05.24 | TTT_Travis | if not I'll get the $10 one |
02:05.29 | TTT_Travis | I just need something that works |
02:05.36 | TTT_Travis | don't care how crappy the quality is |
02:05.39 | *** join/#asterisk rowter (n=Silver@201.138.157.112) |
02:06.04 | rowter | anyone has heard of dr nolazco corpus files for sphinx for spanish recognition? |
02:06.29 | iceyp | thanks for your help guys |
02:06.30 | iceyp | cya |
02:07.28 | Qwell | TTT_Travis: I never said the quality was crappy |
02:07.36 | *** join/#asterisk esculapio_ (n=ESCulapi@142stb68.codetel.net.do) |
02:07.39 | Qwell | I said the card is. Meaning, it'll BARELY work, if at all |
02:07.57 | Qwell | Don't count on having any sort of hangup detection. |
02:08.09 | Qwell | So, anticipate 24 hour long voicemail messages |
02:12.28 | *** join/#asterisk synaptic (i=synaptic@68.62.176.196) |
02:13.14 | hinckc | TTT_Travis: if you want to make a "science project", just do sip only, and get a sip service provider for 20 bucks a month. then when your frankenstein is ready, just forward your regular # to the SIP #. |
02:20.05 | luke-jr_ | hinckc: if I understand correctly, he wants other phones to be on the line too |
02:20.57 | [TK]D-Fender | TTT_Travis : What do you want out of *? Then we'll see about what best help you get there. |
02:21.43 | *** join/#asterisk coppice (n=chatzill@37.162.17.210.dyn.pacific.net.hk) |
02:24.51 | TTT_Travis | [TK]D-Fender just something to fill my spring break boredom |
02:24.57 | TTT_Travis | like maybe making a voice prompt menu |
02:25.41 | [TK]D-Fender | TTT_Travis : For your existing line? |
02:25.56 | TTT_Travis | possibly? |
02:26.03 | [TK]D-Fender | TTT_Travis : What would use as phones on your system? |
02:26.03 | TTT_Travis | I really haven't looked into it much |
02:26.17 | TTT_Travis | can you just pick up a phone and dial an extension? |
02:26.52 | [TK]D-Fender | TTT_Travis : With what is the question. You can't jsut plug a phone into that modem of yours and it'd be almost miraculous if it was uable for even your line. |
02:27.23 | [TK]D-Fender | TTT_Travis : You'd need a special interface in order to use your analog phone with *. Either ana ATA or special PCI card. |
02:27.23 | TTT_Travis | k |
02:27.24 | [TK]D-Fender | s/ana/ATA |
02:27.28 | [TK]D-Fender | s/ana/ATA/ |
02:27.33 | [TK]D-Fender | grrr |
02:28.02 | [TK]D-Fender | close enough. either way an ATA will run you about $70 (the best way to go), and the PCI solution $130 +/- (I think) |
02:29.52 | TTT_Travis | maybe I will just use it as sip phone over my network |
02:29.57 | TTT_Travis | or something |
02:30.39 | hinckc | you can get a cheap (but functional) sip phone for ~50 bucks. |
02:30.56 | TTT_Travis | did I say sip? |
02:30.58 | TTT_Travis | I mean't soft |
02:31.12 | hinckc | oh, sure... even cheaper |
02:32.05 | *** join/#asterisk suma (n=suma@222.165.112.215) |
02:32.46 | hinckc | it just doesn't really feel like a PBX until a _phone_ rings... :) |
02:33.35 | TTT_Travis | I wonder what my school uses for a PBX |
02:33.45 | TTT_Travis | its not asterisk since they've had it for a long time |
02:34.11 | mitcheloc | replace it! |
02:37.25 | *** join/#asterisk inv_Arp (i=junya@adsl-11-225-195.mia.bellsouth.net) |
02:39.02 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
02:39.28 | wwalker | OK, I can call my Polycom501 from Asterisk (Dial 510) and it rings and gets sidetone. But it won't dial out. It doesn't send anything to asterisk (sip debug show nothing) |
02:39.52 | wwalker | It's registered (sip show peers show address and 68 ms). What am I missing? |
02:41.19 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
02:42.48 | wwalker | OK, I see that I'm getting sip stuff now. "Failed to authenticate user "SoundPoint IP"" I set the user for the line to 510...??? |
02:53.13 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
02:54.36 | MoutaPT | Any one could explain me or guess, why using Xlite if i'm in a call and choose music on hold ant then just click to hangup, Xlite presents the call like finish and it stills runing on asterisk?? The only way to finish this was soft hangup ?? |
02:55.37 | DoktorGreg | oh i think i found it... |
02:55.56 | DoktorGreg | I was mapping pri channels onto a pseudo channel |
02:56.10 | DoktorGreg | ??? |
02:56.19 | DoktorGreg | how do i tell if i have a pseudo channel? |
02:57.04 | MoutaPT | Any one expert with PRI? any way to hang up a call automatically if the called party hangs the call? |
02:57.54 | DoktorGreg | I am working with pri.... But it sounds like you have more connectivity than i do right now... |
02:58.05 | MoutaPT | i'm using E1 |
02:58.20 | DoktorGreg | im on t1 |
02:58.28 | DoktorGreg | no fair you get more channels! |
02:58.35 | MoutaPT | everything is fine, only problem is some of my users seem to not hangup correctly the calls! |
02:58.37 | [TK]D-Fender | wwalker : bad user:pass |
02:58.49 | MoutaPT | i need to finnish the calls if the called party hangs |
02:59.12 | [TK]D-Fender | MoutaPT : typically the call IS terminated immediately. |
02:59.31 | [TK]D-Fender | MoutaPT : Thats the point of digital signalling, so you don't have to "guess" |
02:59.39 | MoutaPT | i have * behind a legacy PBX |
02:59.46 | MoutaPT | and i've been tracing |
02:59.49 | wwalker | [TK]D-Fender: thx |
02:59.51 | MoutaPT | the operator sends |
02:59.58 | MoutaPT | PROGRESS message |
03:00.03 | MoutaPT | Cause Code =16 |
03:00.13 | MoutaPT | and asterisk answers with cause code =98 |
03:01.02 | MoutaPT | ISDN says that cause code= 98 is This cause is sent when the equipment sending this cause has received a message which includes the information elements not recognized because the information element identifier is not define or it is defined but not implemented by the equipment sending the cause. However, the information element is not required for the equipment sending the cause to process the message. |
03:01.56 | MoutaPT | i got a technician with equipment to watch the ISDN packets between asterisk and old PBX... |
03:02.04 | MoutaPT | any one could help me? |
03:02.51 | *** part/#asterisk apple (i=appleboy@about/cooking/nakedchef/apple/tarts) |
03:05.19 | MoutaPT | DoktorGreg any troubles with your PRI? |
03:06.15 | MoutaPT | my mistake cause code=98 is : |
03:06.18 | MoutaPT | This cause indicates that the message received is not compatible with the call state or the message type is non-existent or not implemented. |
03:06.58 | DoktorGreg | i have some sort of pseudo channel driver loaded |
03:07.08 | MoutaPT | that one is Ztdummy |
03:07.12 | DoktorGreg | and its messing with how the channels are mapped by zaptel |
03:07.15 | MoutaPT | are you with kernel 2.6? |
03:07.23 | DoktorGreg | 2.4 |
03:07.28 | MoutaPT | edit your zaptel.conf |
03:07.37 | MoutaPT | have you done it ? |
03:07.48 | DoktorGreg | its not being loaded in zaptel |
03:08.07 | MoutaPT | which card you have? |
03:08.14 | DoktorGreg | 205 |
03:08.23 | MoutaPT | digium? |
03:08.24 | DoktorGreg | xorcom rapid distro |
03:08.26 | DoktorGreg | yah |
03:08.35 | MoutaPT | digium? |
03:08.38 | *** join/#asterisk websae2k (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
03:08.47 | DoktorGreg | digium card, the 205 |
03:08.59 | DoktorGreg | 2 pri [ports at 5 volts |
03:09.01 | MoutaPT | ok, i've only experience with TE110P |
03:09.27 | MoutaPT | but i can tell you that for me worked fine editing zaptel.conf |
03:09.37 | MoutaPT | ztcfgv -vvvv |
03:09.38 | DoktorGreg | editing editing... |
03:09.56 | MoutaPT | modprobe |
03:10.05 | MoutaPT | genzaptelconf |
03:10.16 | MoutaPT | have you done all this stuff? |
03:11.33 | MoutaPT | Any one here has tried Mitel Hardphone with Asterisk? |
03:11.49 | DoktorGreg | ive done all sorts of stuff |
03:11.52 | Qwell | MoutaPT: Is it SIP? |
03:11.57 | DoktorGreg | thats the thing with asterisk |
03:12.02 | DoktorGreg | er linux |
03:12.09 | DoktorGreg | there is always more stuff to try |
03:12.13 | MoutaPT | yes |
03:12.13 | Qwell | DoktorGreg: You have to know what you're doing, to config stuff? :P |
03:12.19 | Qwell | MoutaPT: Then it should work fine |
03:12.38 | MoutaPT | i will try it on monday:) |
03:13.04 | MoutaPT | can u tell me why with xlite i seem to loose control of call if i put it on hold? |
03:13.22 | MoutaPT | then i won't get control of the call any more... |
03:13.23 | Qwell | because they want you to use the pro version |
03:13.47 | Qwell | I think they introduce random bugs, and stupid config options to the free version |
03:13.49 | *** join/#asterisk rickb|server (n=none@cpe-71-66-110-248.neo.res.rr.com) |
03:13.53 | rickb|server | Hello. :) |
03:13.54 | MoutaPT | r u sure? i need to explain my boss... |
03:14.03 | Qwell | well, I'm paranoid |
03:14.13 | MoutaPT | he though we could get it working with softphone |
03:14.28 | rickb|server | I am really new to PBX and asterisk, is there a way that someone could help me out with just a few things? |
03:14.29 | MoutaPT | i've 70 users... and the problems are arriving... |
03:14.37 | Qwell | softphone for 70 users? |
03:14.38 | Qwell | ugh |
03:14.43 | MoutaPT | i think hardphone would help me |
03:14.45 | MoutaPT | a lot |
03:14.50 | Qwell | rickb|server: Only if we know what those things are - ie; ask a question |
03:14.51 | MoutaPT | i get all the emails.... |
03:14.57 | MoutaPT | complaining... |
03:14.59 | MoutaPT | and so on... |
03:15.01 | rickb|server | Well.. Ok |
03:15.06 | MoutaPT | hard to manage... 70 users |
03:15.11 | Qwell | MoutaPT: not really |
03:15.25 | MoutaPT | sjphone seems to get critical error |
03:15.45 | MoutaPT | and xlite seems to loose control of the call if u click to music on hold... |
03:16.00 | rickb|server | I have a POTS line in my house.. So I run the PSTN line from my house into the server running Asterisk, then that Asterisk Box, (if setup properly) will pickup those calls? And direct them to whereever I set? |
03:16.02 | MoutaPT | Qwell not really what? |
03:16.07 | Qwell | not really hard |
03:16.27 | Qwell | 70 users should be a snap |
03:16.37 | MoutaPT | how do u explain this calls... |
03:16.44 | MoutaPT | like zombie |
03:16.45 | MoutaPT | calls |
03:16.58 | Qwell | xlite sucks :p |
03:17.03 | MoutaPT | for a mistake users click music on hold then click to hangup |
03:17.10 | MoutaPT | xlite presents no call running |
03:17.17 | MoutaPT | and they are running |
03:17.22 | MoutaPT | i see it on CLI |
03:17.37 | MoutaPT | imagine you have 70 outbound calls like this |
03:17.44 | MoutaPT | you are paying!!!! |
03:17.57 | MoutaPT | which SIPsoftphone do u recommend me? |
03:19.36 | rickb|server | I am using the Rapid Xorcom version of the Asterisk. :) It is nice from what I can tell.. |
03:20.10 | Qwell | 70 calls? I wouldn't USE a softphone |
03:20.18 | Qwell | If the company can't afford hardware, they can't afford my services... |
03:20.27 | MoutaPT | Ok now i understand you! |
03:20.40 | Qwell | and if they're just too cheap...screw them |
03:20.48 | Qwell | softphones do not a production PBX make |
03:21.01 | MoutaPT | you r right , i though u were saying to me 70 users with softphone were peanuts |
03:21.10 | Qwell | it is |
03:21.16 | rickb|server | I am running the newest version of the Rapid, I tried to use the Asterisk Flash Operator Panel, It prompts for a password, I don't know it and consulted google for 2 hours to find the default. Any ideas? |
03:21.33 | MoutaPT | passw0rd |
03:21.40 | Qwell | rickb|server: maybe "asterisk"? |
03:21.52 | Qwell | or "password" |
03:21.55 | Qwell | or a blank string |
03:22.04 | MoutaPT | Qwell is peanuts with sofphone??? |
03:22.12 | MoutaPT | which softphone? |
03:22.23 | mitcheloc | rickb|server: why not ask xorcom? |
03:22.26 | Qwell | MoutaPT: doesn't really matter, because it's stupid to do |
03:24.02 | MoutaPT | it depends... if you an IT company every one with its own workstation... |
03:24.13 | MoutaPT | they thing it is good idea... |
03:24.17 | MoutaPT | think |
03:24.20 | Qwell | well, it isn't |
03:24.38 | MoutaPT | yeah i got it now :( unfortunately... |
03:25.05 | MoutaPT | IAX trunks working perfectly, fax to email too |
03:25.15 | MoutaPT | only problem are the softphones. |
03:25.15 | rickb|server | yeah.. I tried. |
03:25.30 | rickb|server | I am just setting up a third party program for admining |
03:25.46 | Qwell | rickb|server: passw0rd, like MoutaPT said |
03:25.55 | rickb|server | I tried that to. :) |
03:25.59 | rickb|server | It didn't work |
03:26.10 | rickb|server | I tried all the configs to see if there were entries to.. |
03:26.12 | rickb|server | :( |
03:26.23 | MoutaPT | check your .conf files |
03:26.35 | MoutaPT | somewhere probably u will find it |
03:26.40 | rickb|server | Yeah |
03:27.21 | MoutaPT | any one could say me if i can by BRI connection to my operator and use a PRI card? |
03:27.38 | Qwell | MoutaPT: no |
03:27.38 | MoutaPT | it's ISDN so...? |
03:27.43 | rickb|server | The default password for asterisk its self, how do you set it? or is it allready set? |
03:27.52 | Qwell | rickb|server: There is no password for asterisk |
03:27.57 | rickb|server | ok. :) |
03:28.18 | MoutaPT | Qwell why not do u know? did u try it? |
03:28.34 | Qwell | MoutaPT: It won't work |
03:29.03 | MoutaPT | ok |
03:29.13 | Qwell | besides, a BRI card would be cheaper |
03:29.22 | MoutaPT | it is not so much.. |
03:29.31 | MoutaPT | TE110P is around 500Euros |
03:29.38 | MoutaPT | and BRI could be 400 |
03:29.39 | drray | govarion |
03:29.41 | Qwell | That's sad |
03:29.43 | drray | has a cheaper tor |
03:30.19 | MoutaPT | which hardphones u ser Qwell? |
03:30.24 | Qwell | ser? |
03:31.45 | MoutaPT | ? |
03:31.50 | MoutaPT | sorry |
03:31.51 | Qwell | exactly |
03:32.04 | Qwell | plz2be using proper english |
03:32.15 | MoutaPT | which hardphones do you use? |
03:32.21 | Qwell | much better |
03:32.23 | Qwell | Cisco |
03:33.14 | MoutaPT | ok, most of the people i talk they advice me cisco |
03:33.23 | websae2k | cisco for speaker phone :) |
03:33.31 | drray | once it's working the 7960 is a damn fine phone |
03:33.31 | SplasPood | polycom for speaker phone |
03:33.38 | brookshire | polycom designed cisco's speaker phone ;) |
03:33.39 | websae2k | grandstream works quite nicely, polycom even better, and then cisco |
03:33.39 | Qwell | It's the same speaker phone :p |
03:33.42 | [TK]D-Fender | Polycom is typically a cheaper and equal choice. |
03:33.53 | websae2k | i think 7960 has great speaker phone |
03:33.57 | websae2k | yes exactly |
03:34.09 | SplasPood | I just bought a few Grandstream BT-101s... great value for $48/ea |
03:34.20 | drray | if the polycom lets you dowload firmware without paying, tehn I'd go that way |
03:34.30 | SplasPood | drray: previous revision, yea |
03:34.35 | SplasPood | not the latest, but one behind |
03:34.49 | SplasPood | your vendor should be able to hook you up /w the latest firmwarez tho |
03:35.03 | MoutaPT | easy one question, could you tell me what for is the dialparties.agi ? |
03:35.11 | rickb|server | What is a good free gui web based administration program to use? |
03:35.18 | Qwell | MoutaPT: To dial...parties |
03:35.23 | Qwell | rickb|server: no such thing |
03:35.25 | drray | rick - winvi |
03:35.40 | rickb|server | i'l check it out |
03:35.57 | MoutaPT | Qwell, but wouldn't be better to just use a simple macro to dial ? |
03:36.04 | Qwell | MoutaPT: I don't know |
03:36.17 | [TK]D-Fender | MoutaPT : Thats an AMP/FreePBX script. Not the sort of thing to speak of here... |
03:36.26 | Qwell | ugh |
03:36.28 | Qwell | figures |
03:37.33 | MoutaPT | ok, Does any one could answer me how to make asterisk to hang up a call when the called party hangs, i'm using PRI. |
03:38.08 | VoIPMasta | MoutaPT: it should hang up by itself |
03:38.36 | MoutaPT | I have PSTN---OLDPBX----Asterisk , and it doesn't happens... |
03:38.52 | VoIPMasta | maybe something is wrong in your oldpbx |
03:38.56 | MoutaPT | if i dial a local extension on OLDPBX, it happens |
03:39.02 | VoIPMasta | why don't you plug your pri directly into asterisk? |
03:39.05 | MoutaPT | but if i dial to outside world... |
03:39.27 | VoIPMasta | my best guess would it be that your old pbx isn't handling the call signaling correctly |
03:39.49 | MoutaPT | my users just start listenning busytones... |
03:40.06 | VoIPMasta | yup, there's a signaling problem in your oldpbx |
03:40.07 | MoutaPT | but the call stills there... |
03:40.30 | MoutaPT | i got there a technician watching this... |
03:40.54 | MoutaPT | and i found the asterisk receives PROGRESS , CAUSE CODE=16 |
03:41.00 | MoutaPT | when the called party hangs |
03:41.14 | MoutaPT | then asterisk answers with cAUSe CODE=98 |
03:41.18 | MoutaPT | This cause indicates that the message received is not compatible with the call state or the message type is non-existent or not implemented. |
03:41.41 | MoutaPT | any test i can make with the local extensions |
03:41.44 | MoutaPT | on old pbx |
03:41.59 | MoutaPT | to show my boss that oldpbx is the problem? |
03:42.30 | Qwell | remove it from the loop |
03:43.02 | brookshire | so why does asterisk support the jpeg codec? |
03:43.04 | brookshire | heh |
03:43.15 | Qwell | brookshire: pr0n for the 7970, duh |
03:43.22 | mitcheloc | it's good a compressing audio? |
03:43.37 | Qwell | pr0n IVR! |
03:43.39 | Qwell | omg, brb |
03:43.51 | brookshire | push one for pr0n |
03:43.54 | coppice | right. a picture paints a thousand words, so JPEG compression is excellent |
03:43.54 | mitcheloc | lol wash your hands before you come back |
03:43.54 | brookshire | push two for pr0n |
03:44.01 | brookshire | push three for pr0n |
03:44.04 | Qwell | brookshire: no, no, no |
03:44.09 | Qwell | brookshire: Seen the jukebox AGI? |
03:44.13 | brookshire | no |
03:44.18 | Qwell | it's in trunk |
03:44.28 | Qwell | jukebox menu thing |
03:44.37 | brookshire | ahh.. for mp3? |
03:44.40 | Qwell | yeah |
03:44.49 | brookshire | that's so pre-stable 1.0 |
03:44.51 | mitcheloc | *wonders how the jukebox agi got in* |
03:44.52 | Qwell | that could be so easily modified for chan_skinny, pr0n.agi :D |
03:45.07 | Qwell | mitcheloc: because people tested it, and it worked well |
03:45.39 | brookshire | so like you can change the photos on a cisco phone? |
03:45.47 | mitcheloc | when you get your pr0n make sure to use ulaw to get the best experience |
03:45.59 | file | pr0n? WHERE |
03:45.59 | Qwell | brookshire: I don't know if the protocol can |
03:46.00 | brookshire | g722! |
03:46.09 | MoutaPT | Qwell do you know which message i should receive in Asterisk when the called party hangs, with zap intense debug? |
03:46.11 | mitcheloc | lol file |
03:46.20 | Qwell | file: qwell.com/pr0n/ |
03:46.29 | file | Qwell: is that... Qwell pr0n? |
03:46.31 | Qwell | wget WILL crash if you try to mirror it |
03:46.35 | Qwell | umm...mebbe |
03:46.36 | coppice | are the women in ulaw pr0n fatter than in other pr0n? |
03:46.46 | Qwell | coppice: less compression, is all |
03:46.53 | brookshire | :( |
03:46.53 | Qwell | They actually look better...no lossage |
03:46.55 | brookshire | you lie |
03:47.05 | Qwell | g729 pr0n is ...ugh |
03:47.06 | file | brookshire: Damn Best! |
03:47.14 | Qwell | lpc10 is like...yeah |
03:47.19 | brookshire | file: where did you get Damn Best? |
03:47.25 | file | brookshire: somewhere. |
03:47.29 | brookshire | hehe |
03:47.31 | brookshire | rar! |
03:47.38 | file | it MAY have appeared on my account |
03:47.55 | Qwell | "Damn Best"? |
03:48.07 | brookshire | yes.. Damn Best! |
03:48.13 | file | it's the damn best! |
03:48.14 | Qwell | which are? |
03:48.50 | file | exactly! |
03:51.18 | file | Qwell: nub! |
03:51.22 | Qwell | :( |
03:51.30 | *** join/#asterisk bmg505 (n=leon@dsl-146-14-214.telkomadsl.co.za) |
03:53.35 | *** join/#asterisk isamar (n=isamar@202.95.220.92) |
03:54.11 | isamar | hi folks |
03:59.04 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
03:59.07 | wwalker | I've got two IP501's. one will authenticate, the other fails. I've set and reset them over and over. the sip.conf paragraph for one is a copy of the other with the [sip_name] changed. So I can't see how one works and the other doesn't |
03:59.12 | wwalker | Any ideas? |
04:00.09 | alephcom | I have a favor to ask... Would somebody be willing to try calling 14035387846 and see if you can get through? |
04:00.36 | alephcom | It should hit a recording. |
04:01.13 | wwalker | "connected to the server" |
04:01.22 | Qwell | servICE |
04:01.37 | Qwell | "hoi, you are being connected to the service" |
04:01.41 | drray | you are being connected to the service" |
04:01.46 | drray | is that festival? |
04:01.52 | wwalker | alephcom: I don't think it works :( |
04:02.02 | alephcom | Thanks a bunch everyone... This is weird. |
04:02.10 | alephcom | wwalker, you couldn't get through? |
04:02.28 | alephcom | drray: Gotta love my voice. :-P That's me :-( |
04:02.30 | wwalker | just joking. I heard server instead of service as many corrected me |
04:02.41 | drray | that's you>? |
04:02.51 | drray | what codec are you using? |
04:02.58 | Qwell | lpc10? |
04:03.05 | wwalker | Allison makes nice recordings.... |
04:03.39 | alephcom | trashy microphone on a bad day. I know I have bought all the recording from her. For some reason this one hasn't been replaced yet. |
04:03.54 | file | QWELL |
04:05.04 | file | awwww |
04:05.22 | Qwell | <3 |
04:06.04 | Qwell | drray: perv |
04:10.40 | isamar | today my 1.2.6 astersik stopped responding SIP connections... suddenly.. |
04:12.18 | VoIPMasta | a firewall maybe? |
04:12.23 | drray | did you do a yum updated? |
04:12.26 | drray | -d |
04:12.33 | Qwell | yum -d updated? |
04:13.08 | Qwell | oh, yum upated |
04:13.15 | VoIPMasta | yum? |
04:13.21 | VoIPMasta | Is that another "automated" installer? |
04:13.22 | drray | did you upgrade your kernel |
04:13.29 | drray | or something on your box |
04:14.30 | *** join/#asterisk synaptic (i=synaptic@68.62.176.196) |
04:14.32 | isamar | yum ? |
04:14.47 | isamar | not firewall.. |
04:14.48 | Qwell | I prefer apt and synaptic |
04:14.49 | isamar | public IP |
04:14.59 | synaptic | lol |
04:15.03 | VoIPMasta | isamar: maybe your ISP added a new firewall? |
04:15.15 | isamar | not first time with 1.2.x ... :-( |
04:15.17 | VoIPMasta | isamar: or something else is opening port 5060 |
04:15.26 | isamar | everything ws ok.. except sip :-( |
04:15.29 | wwalker | I've got two IP501's. one will authenticate, the other fails. I've set and reset them over and over. the sip.conf paragraph for one is a copy of the other with the [sip_name] changed. So I can't see how one works and the other doesn't |
04:15.33 | wwalker | Any ideas? |
04:15.47 | VoIPMasta | Qwell: apt? as in debian? |
04:15.57 | Qwell | VoIPMasta: sure |
04:16.02 | VoIPMasta | wwalker: what's in your logs? |
04:16.17 | VoIPMasta | Qwell: I prefer make, gmake, configure |
04:16.32 | Qwell | make, then gmake, THEN configure? |
04:16.39 | VoIPMasta | Qwell: not in the right order |
04:16.50 | VoIPMasta | Qwell: as a matter of fact you can't use make and gmake in the same compile |
04:16.52 | file | Qwell: come on baby I want to party, come on right to the dance floor |
04:16.54 | Qwell | indeed you can't |
04:16.58 | Qwell | file: umm |
04:17.03 | Qwell | you scare me |
04:17.10 | file | excellent |
04:19.44 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
04:20.40 | wwalker | VoIPMasta: Apr 16 00:13:45 NOTICE[3423]: chan_sip.c:10879 handle_request_register: Registration from '<sip:511@192.168.50.1>' failed for '192.168.50.200' - Username/auth name mismatch |
04:20.49 | SplasPood | Hrm.. is it possible to have the grandstream bt-101 hang up the call when the remote party hangs up when on speakerphone? Mine starts giving a busy tone |
04:21.38 | VoIPMasta | wwalker: Username mismatch |
04:21.49 | VoIPMasta | wwalker: double check your username and password settings |
04:21.53 | Qwell | SplasPood: what is on the other end? |
04:22.31 | VoIPMasta | SplasPood: I don't think it's possible |
04:22.49 | wwalker | the "username" is the part between the []'s in sip.conf, right? |
04:23.00 | VoIPMasta | wwalker: remove any type of encryption, use cleartext usernames and passwords stored in sip.conf (don't use mysql to set usernames) |
04:23.30 | wwalker | plaintext in sip.conf. |
04:23.37 | VoIPMasta | wwalker: you can also set it using username=yourusername |
04:24.48 | file | beeeeep no you can't.... |
04:25.02 | VoIPMasta | file: you can't what? |
04:25.02 | file | whatever is in the context name, ie between [ and ] is what is the username... |
04:25.30 | VoIPMasta | file: AFAIK it can be overrided using username= in sip.conf |
04:25.38 | wwalker | http://rafb.net/paste/results/A2nlZz59.html 510 works, 512 works, 511 doesn't... |
04:25.40 | file | that's used for something different |
04:28.11 | VoIPMasta | wwalker: your sip.conf looks ok\ |
04:28.38 | VoIPMasta | wwalker: check your settings in your IP501 |
04:30.27 | wwalker | Which IP501 setting is the password for SIP? Which setting is the username? I think that is Auth under Line 1 |
04:34.10 | SplasPood | Qwell: Asterisk... |
04:34.18 | SplasPood | VoIPMasta: So this is a known... bug/feature? |
04:34.18 | VoIPMasta | wwalker: don't know, I don't have an IP501 |
04:34.27 | Qwell | SplasPood: and on the other end of that? |
04:34.28 | wwalker | thx |
04:34.37 | SplasPood | Qwell: nothing.. local dialplan app stuff.. |
04:34.40 | VoIPMasta | SplasPood: most speakerphone phones won't hang up until you press the hangup button |
04:34.53 | SplasPood | polycom will |
04:35.02 | SplasPood | as will cisco |
04:35.44 | VoIPMasta | yup, but Grandstream's chipsets are usually less feature-packed |
04:35.55 | SplasPood | haha... "feature" |
04:36.07 | SplasPood | I don't consider this a feature, but for $48, i can deal :) |
04:36.34 | VoIPMasta | I said "less feature-packed" what I mean is that there are less features in a Grandstream chipset |
04:36.54 | SplasPood | Yes there are, but hanging up the phone when it gets a hangup indication isn't much to ask. |
04:38.54 | drray | my budgetone was worth the $70 i paid for it 2 years ago |
04:39.10 | drray | I'd not deploy them in an office though |
04:39.27 | SplasPood | yea we bought a few to throw in people's homes |
04:39.39 | SplasPood | prolly end up going for IP301s in the long run |
04:39.48 | drray | just don't power cycle it during a reboot |
04:40.00 | SplasPood | hrm? |
04:40.20 | drray | it'll lose its mind |
04:42.03 | SplasPood | so like.. i plug it in... then while it's booting, unplug/replug it? |
04:42.25 | drray | well, I was rebooting it from the webpage |
04:43.53 | SplasPood | when you say it'll lose it's mind do you mean.. it'll forget it's config.. or it'll brick itself.. |
04:45.05 | drray | mine bricked itself, I had to set a subnet up to mimic the one it was looking for to load a new firmware on it |
04:45.12 | drray | it kept clicking every 10 seconds |
04:45.29 | VoIPMasta | what you mean is to not unplug it while upgrading, not rebooting |
04:45.43 | drray | mine locked up during rebooting |
04:45.47 | VoIPMasta | most devices will have problems if you cycle the power while they're having the firmware updated |
04:45.51 | drray | not an upgrade |
04:45.56 | VoIPMasta | mmm odd |
04:46.11 | drray | I was able to revive it |
04:47.23 | drray | I hit reboot in the budgetones web gui, and while it was clicking to restart I unplugged it |
04:47.29 | drray | I don't know what possed me to do it |
04:48.57 | *** join/#asterisk achandra (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
04:49.15 | VoIPMasta | I wish there was a local grandstream dealer here, but there aren't any |
04:49.30 | Qwell | grandstream is pretty bad too |
04:49.38 | VoIPMasta | Qwell: but pretty cheap |
04:49.50 | VoIPMasta | Qwell: it can be the perfect solution for lame residential users |
04:49.54 | drray | I'd buy an iaxy before I bought another grandstream |
04:49.57 | Qwell | I was actually supposed to email one of the grandstream guys... |
04:54.00 | *** join/#asterisk op3r (i=op3r@210.4.31.234) |
04:54.09 | op3r | anyone using aheeva here? |
04:55.16 | achandra | Hello, had a few questions about * in a LB setup. Can multiple * boxes, using say SRV records function with a central PGSQL boxes ( assume active, active setup for LB). |
04:56.34 | *** part/#asterisk TTT_Travis (n=Travis@bal-broadband2-ws-14.dsl.airstreamcomm.net) |
04:56.38 | Qwell | achandra: LB? |
04:56.51 | achandra | load balanced |
04:57.22 | achandra | ; ) |
05:17.01 | *** join/#asterisk Plecebo (n=x@216-160-101-90.tukw.qwest.net) |
05:18.42 | *** join/#asterisk suma (n=suma@222.165.112.215) |
05:18.54 | suma | i'm havng problem with iax registration |
05:19.01 | suma | can anybody please help |
05:20.25 | suma | <PROTECTED> |
05:22.18 | suma | this is in my iax.conf file |
05:22.20 | suma | [kans] |
05:22.20 | suma | type=user |
05:22.20 | suma | context=kanscontext |
05:22.53 | brookshire | type=user is for outgoing lines.. is that what you want there? |
05:23.32 | suma | thanks a million |
05:24.07 | file | er |
05:24.17 | brookshire | heh |
05:24.19 | brookshire | file! |
05:24.26 | file | user is incoming, as they use your system |
05:24.35 | file | peer is outgoing, as you peer with another system |
05:24.44 | brookshire | oh now you've got me all confused |
05:24.54 | file | good. |
05:25.00 | brookshire | A user makes calls. The following would be needed in iax.conf on the user machine to identify (authenticate) itself to the peer before the peer will take the call. |
05:25.09 | tainted- | why not type=incoming, type=outgoing, type=bidirectional |
05:25.26 | brookshire | oooo! lets just make everything 'friend' |
05:25.34 | Qwell | Let's kill the user |
05:25.49 | Qwell | type=whateveritdoesntmatteranymore |
05:25.50 | tainted- | type = friend, type = aquaintance, type = onenighter, type = voipwanker |
05:26.04 | Qwell | type=bootycall |
05:26.15 | tainted- | i need more of those users |
05:26.25 | Qwell | more, as in 1? |
05:26.40 | tainted- | one is the new more |
05:28.36 | Qwell | type=user ~= fxs? |
05:28.48 | Qwell | type=peer ~= fxo? |
05:30.30 | brookshire | oh yeah.. don't get me started about the fxs fxo thing |
05:31.00 | tainted- | yea talk about confusicating |
05:31.51 | tainted- | get started brookshire get started |
05:36.24 | Shotta30five | Anyone know abou unlocked a VOIP device |
05:36.38 | Shotta30five | unlocking |
05:36.46 | Qwell | Shotta30five: You don't |
05:38.55 | achandra | Are you reffering to say unlocking a Fleebay special sipura that has been "locked" and you want to ulock it? |
05:39.31 | Qwell | No, he got a free vonage ATA, and wants to unlock it |
05:39.51 | drray | can't you pay vonage $5 to unlock it? |
05:39.52 | file[laptop] | Qwell: are you locked? |
05:40.02 | Qwell | file[laptop]: depends where |
05:40.16 | file[laptop] | eep |
05:40.24 | Qwell | sicko, not like that |
05:41.01 | file[laptop] | Qwell: you're sooooooo cool |
05:41.07 | Qwell | I so know |
05:41.09 | dlynes | btw...why not just have one context that defines outgoing and incoming, so you don't need two separate contexts? |
05:41.21 | dlynes | Most of the info is duplicated, anyways |
05:41.22 | Qwell | dlynes: because that would be silly |
05:41.37 | Qwell | two seperate contexts is a VERY good thing |
05:42.09 | dlynes | yes, it makes sense, conceptually |
05:42.37 | dlynes | but if you don't need the type= line, what's to differentiate the two contexts? |
05:42.50 | achandra | this cleary explained...because I myself re-read it here - page 78 of Asterisk: The future of Telephony...about contexts and why they are seperate... |
05:43.38 | dlynes | that book isn't terribly useful for the most part, either |
05:43.43 | achandra | I suppose you could argue and be convincing that seperate contexts enforce security. |
05:43.47 | brookshire | friend is bad! |
05:43.51 | brookshire | i learned this today |
05:44.04 | dlynes | Yeah...friends are bad; enemies are good! |
05:44.14 | achandra | wtf?? |
05:44.34 | achandra | i thought we were discussing use of contexts..anyhow. |
05:44.37 | file[laptop] | brookshire: am I your friend?!? |
05:44.47 | brookshire | file: no.. |
05:44.51 | file[laptop] | good |
05:44.52 | file[laptop] | I don't want to be |
05:44.53 | achandra | lol |
05:44.58 | dlynes | lmao |
05:45.45 | dlynes | achandra: I think the different contexts are just for conceptualizing direction of traffic |
05:45.52 | dlynes | achandra: i.e. for humans |
05:46.08 | tainted- | omg freshly baked poundcake + ice cream = crazy delicious |
05:46.10 | Qwell | contexts help IMMENSELY with security |
05:46.17 | Qwell | tainted-: poundcake linux?! |
05:46.19 | dlynes | how so? |
05:47.02 | tainted- | apt-get consume poundcake |
05:47.06 | dlynes | And how's that gonna help you get another free pstn proxy? |
05:47.18 | file[laptop] | I'm watching William Shatner... rap... |
05:47.22 | file[laptop] | I feel violated |
05:47.23 | Qwell | dlynes: Because you obviously haven't read README.security |
05:47.29 | achandra | on contexts again...quoting... " One of the most important use of contexts is to enforce security. By using contexts xorrectly, you give certain callers access to features...If you dont design the dialplan carefully you may indaventrly allow others to fradulently use your system.....se SECURITY" |
05:47.37 | tainted- | dlynes by forcing intruders to be so innundated with extra text that they run away with despair |
05:48.24 | achandra | anyhow thats on page 79 ;) |
05:48.26 | tainted- | dlynes he hax0rs your skinny session |
05:48.26 | dlynes | And? You think I send all iax and sip traffic into the same incoming context? |
05:48.47 | Qwell | 100 things going into one incoming context != insecure |
05:48.59 | Qwell | incoming AND outgoing in the same context == stupid |
05:49.06 | brookshire | yes |
05:49.09 | brookshire | i agree |
05:49.16 | Qwell | and covered in README.security |
05:49.17 | tainted- | i like to use [incoming] + _. matching.. |
05:49.23 | tainted- | it's really cool |
05:49.27 | dlynes | I thought we were talking about sip/iax contexts, not dialplan contexts? |
05:49.27 | brookshire | qwell: and don't forget about the headaches |
05:49.31 | file[laptop] | let's put our outbound PSTN calling into our incoming context... yes |
05:50.01 | achandra | plus imagine having a complex dialplan with multiple macros, and definitions, what a friking nightamare keeping it all straight...and imagine changing say one or two things and having crap blow up...yuck. |
05:50.13 | Qwell | public IPs on a DS3, with a simple password, is also insecure :D |
05:50.19 | Qwell | file[laptop]: eh, eh?! ^^ |
05:50.22 | dlynes | I wasn't even talking about the context= line in iax/sip.conf |
05:50.50 | file[laptop] | The Running Man is on, like omg |
05:50.52 | achandra | give us an example...maybe a good place to start?? |
05:50.53 | file[laptop] | what an old movie! |
05:50.59 | Qwell | file[laptop]: does he really run? |
05:51.07 | dlynes | An outgoing iax/sip context afaik doesn't even use the context= line, does it? |
05:51.15 | file[laptop] | Qwell: yes |
05:51.21 | achandra | file[laptop]: The asternator? lol |
05:51.35 | file[laptop] | dlynes: you can send a context over IAX2 actually |
05:51.40 | Qwell | Does he also run...for governer?! |
05:51.46 | Qwell | wow, I botched that word |
05:51.49 | Qwell | governor? |
05:52.01 | dlynes | file[laptop]: but that's only if you're including the remote iax switch into your dialplan context, right? |
05:52.12 | achandra | its okay you can account the mis-spelling to his lack of funding for education. |
05:52.23 | dlynes | file[laptop]: which, in theory is not a sound concept from a security standpoint, to begin with |
05:52.35 | file[laptop] | no, you can send it when you dial as well |
05:53.16 | dlynes | file[laptop]: Yeah...I never did understand that concept...why you want the outside world to know what one of your contexts is |
05:54.04 | dlynes | It should route you into the context you've defined in your iax.conf file for that user; they shouldn't be able to choose which context to go into |
05:54.29 | file[laptop] | the music on this movie is so... yeah |
05:54.38 | achandra | dlynes: assuming the complexity, a simple context might be figured out by simply calling the number, and running through scenarios of key presses. No? |
05:55.30 | dlynes | achandra: Ok....where are you going with this? |
05:55.55 | suma | when i make a call to asterisk it answers and playback a sound file and i could not hear that |
05:56.02 | suma | the call is iax2 |
05:56.05 | achandra | well...in essence you program the contexts based on what you want the user to say experience...so in essence the user DOES know the context. |
05:56.18 | achandra | if he/she goes through the presses.. |
05:56.23 | dlynes | They don't know the name of that context, though |
05:56.40 | achandra | right..but the function yes... |
05:56.48 | dlynes | correct |
05:57.03 | dlynes | so why should they be allowed to specify which context to use? |
05:57.20 | *** join/#asterisk cced (n=dev2003@222.33.36.205) |
05:57.21 | dlynes | The context should be chosen, based upon which username and password they have chosen |
05:58.00 | achandra | which that function exists in * today.. a little confused...go on though.. |
05:58.42 | suma | when i make an iax2 call, i could not hear the audio, asterisk says it is playing ? ! can anyone please help me |
05:58.44 | dlynes | Well, if you're able to specify a dialplan context outside of what's defined in your iax context, you can theoretically invoke someone else's context |
05:58.57 | dlynes | and thus get access to a dialplan you wouldn't normally be able to use |
05:59.22 | cced | :) |
05:59.22 | dlynes | i.e. possibly dialing long distance if say normally you're restricted to local pstn calls |
05:59.35 | achandra | by seperating them though..you could have them access international calls or not etc. but that is based on seperate contexts to keep some users from doing one thing or another.. |
05:59.44 | achandra | woops you beat me to it. |
05:59.47 | dlynes | But if you don't allow the remote user to specify which context to use, they don't have that ability |
06:00.21 | dlynes | Which brings us back to my original question, why does iax allow you to specify the context in the dial command? |
06:01.03 | achandra | i just see it in reverse...the contexts are rules by which a user can do one thing or another...and the user abides by those set rules... sure as a usere youd be able to understand those rules...(contexts in this case), but your bound by them. |
06:01.05 | dlynes | Or in the switch command for that matter? I realize that's the whole concept of the switch command is to share dialplans, but it seems to me that is insecure in and of itself |
06:02.20 | dlynes | Is there a way to lock down asterisk so that the remote caller is not allowed to specify which context to use? i.e. it'll just use the context that's specified in the 'context=' line of iax.conf? |
06:02.46 | dlynes | Otherwise, what's the point of even having the context= line? |
06:04.12 | achandra | design issue...i guess...define the dial plan differently?...im not sure..unless we start talking about "what" you are intending on designing. |
06:04.35 | kamileon | what voip provider is suggested for use with * with local DIDs |
06:04.39 | dlynes | I'm not intending on designing anything |
06:04.40 | achandra | what do you want the system to do and how do you want the user to interact with the system. |
06:05.10 | dlynes | Qwell was just saying i'm running a sip bot, but I don't see any way to lock down the context issue, either |
06:05.53 | achandra | dlynes: I will say this...your making me think :) |
06:06.06 | achandra | on a friking saturday too. |
06:06.13 | dlynes | Well, wouldn't that seem like a pretty basic thing? |
06:06.28 | dlynes | Allowing the system admin to lock out any requests for an alternate context? |
06:06.51 | dlynes | I've never understood why iax even allows that |
06:06.54 | cced | : |
06:07.55 | dlynes | Sure, it makes it more flexible, but at the risk of making it less secure |
06:08.32 | dlynes | It's like Microsoft Windows...Windows allows you to do a lot of things as a regular user that UNIX doesn't, but it also makes it less secure |
06:09.28 | dlynes | I guess you must be on PDT, too |
06:09.43 | dlynes | Everywhere else on the continent, it's Sunday :) |
06:13.01 | suma | yes |
06:13.07 | achandra | yep |
06:13.27 | suma | i could not hear audio in an iax2 call |
06:13.37 | suma | asterisk says it is playing the gsm file |
06:13.44 | suma | it is the demo which comes with asterisk |
06:14.09 | achandra | behind nat or.. ?? |
06:14.50 | suma | I forwarded my port 4569 to my PC |
06:15.00 | suma | i mean the router port |
06:15.06 | suma | where the client is |
06:15.19 | suma | shall i show you the ethereal output ? |
06:16.00 | achandra | not sure if im entirely qulaified to solve your issue but someone may know whats up |
06:16.47 | suma | server is through DMZ |
06:16.55 | suma | and my client is through port forwarding |
06:17.01 | suma | will there be a NAT problem |
06:18.28 | suma | 46.534929 222.165.112.215 -> 192.168.1.13 IAX2 IAX HANGUP, source call# 16216, timestamp 990ms |
06:18.29 | suma | <PROTECTED> |
06:18.44 | achandra | so you have the * in the DMZ and then you are connecting to external ip of firewall or router which routes to internally natted * box? |
06:19.34 | achandra | what is in your iax.conf file? |
06:19.48 | suma | it is the default conf file |
06:19.55 | suma | i added new user with that |
06:20.13 | suma | [kans] |
06:20.13 | suma | type=friend |
06:20.13 | suma | context=default |
06:20.13 | suma | host=dynamic |
06:20.32 | suma | default is the default demo that comes with asterisk |
06:20.53 | achandra | the sample file....okay |
06:21.05 | suma | yes |
06:21.51 | suma | when i say ifconfig, it says 192.168.0.12 and 192.168.0.13 respectively in linux |
06:23.12 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
06:23.19 | iceyp | whats the variable for digit input? |
06:23.27 | iceyp | i.e. if i want to do SayDigit($input) |
06:25.23 | suma | i could not find the application saydigit |
06:25.30 | suma | does it comes with asterisk ? |
06:25.55 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
06:25.58 | X-Rob | suma, try 'show application say' and then push TAB |
06:26.02 | X-Rob | (in the asterisk console) |
06:26.10 | X-Rob | or just 'show applications' |
06:26.12 | *** join/#asterisk MaddieBoi (n=MaddieBo@210-84-15-248.dyn.iinet.net.au) |
06:27.21 | achandra | im not sure exaclty but do you need to define qualify=no, and port=4569 in the definition which you posted explicitly? |
06:27.39 | suma | yes, there is SayDigits but not SayDigit |
06:27.56 | achandra | as well as context=from-internal ( at least thats in my case). |
06:28.55 | suma | port=4569 is in global definition of iax.conf |
06:29.13 | achandra | okay |
06:29.34 | achandra | also defined the externip ? |
06:30.13 | suma | externip in the general or in the user context ? |
06:30.20 | achandra | in general |
06:31.15 | suma | no, just now mentioning it |
06:31.29 | achandra | ahhhh |
06:33.55 | achandra | so.. externip = aaa.bbb.ccc.ddd ; localnet = 192.168.0.0./255.255.255.0; bind addr: 0.0.0.0, etc. |
06:34.18 | achandra | or whatever the internal network is.. |
06:34.51 | achandra | then you must recompile the kernel....just kidding... :) |
06:35.03 | cced2 | who is familar with H323? |
06:35.48 | achandra | cced2: your going to use for media streaming of video ? cool. |
06:36.47 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
06:36.53 | suma | no luck |
06:38.33 | suma | I also get these messages in the meantime |
06:38.34 | suma | update_registry: Restricting registration for peer 'kans' |
06:38.54 | suma | to '60' seconds |
06:40.59 | achandra | do you have a delayreject =yes in general as well? |
06:42.19 | suma | just now i did |
06:42.24 | suma | and restarted asterisk |
06:42.30 | achandra | okay |
06:43.24 | iceyp | achandra whats the cariable to remember my input digits? |
06:43.34 | iceyp | or actually it's just my dialed number |
06:43.36 | iceyp | i should be ok |
06:44.04 | suma | Apr 16 07:42:57 NOTICE[8311]: chan_iax2.c:5692 update_registry: Restricting registration for peer 'kans' to 60 seconds (requested 1200) |
06:44.11 | suma | still this message comes |
06:44.47 | achandra | this is what my general section has - |
06:45.44 | *** join/#asterisk ptblank (n=MURDER1@68.233.145.253) |
06:47.03 | achandra | bindport = 4569 , externip = aaa.bbb.ccc.ddd , local net = 192.168.0.0 / 255.255.255.0 , bindaddr = 0.0.0.0 delayreject = yes disallow=all allow=g729,allow=ilbc, allow=ulaw, allow=alaw. allow=gsm jitterbuffer = yes dropcount =1 |
06:50.09 | achandra | suma: did that help or?? |
06:50.58 | Shaun2222 | is their a uniq id assigned to every call that comes in on asterisk, hopfully one that can be read using ${ID} |
07:02.54 | dlynes | suma: This might seem like a stupid question, but are 192.168.0.12 and 192.168.0.13 both on the same machine? |
07:04.14 | websae | hrm |
07:11.21 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
07:12.20 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
07:12.30 | *** join/#asterisk mkl1525 (n=daniel@pD9533E9E.dip0.t-ipconnect.de) |
07:16.20 | esculapio_ | hola quien habla espanol |
07:17.03 | cced2 | :) . |
07:26.09 | *** join/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
07:27.45 | dlynes | Shaun2222: ${UNIQUEID} |
07:28.32 | dlynes | Shaun2222: Try checking out http://www.voip-info.org/wiki/view/Asterisk+variables |
07:34.29 | esculapio_ | hola quien habla espanol |
07:34.34 | esculapio_ | ? |
07:35.49 | cced2 | why irc often offline? |
07:38.47 | *** join/#asterisk thx2000 (n=the@adsl-66-51-192-221.dslextreme.com) |
07:44.38 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
07:44.55 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-63-118.w86-217.abo.wanadoo.fr) |
07:52.04 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
07:56.17 | *** join/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net) |
07:56.39 | *** join/#asterisk jj1 (n=jj0@c-67-166-96-37.hsd1.ut.comcast.net) |
07:56.48 | thx2000 | Does anyone know of a way to get sendmail to use the /etc/hosts file instead of looking up the host record from the dns server? |
07:59.14 | stoffell_h | thx2000, uhm, hostfile is used 'before' using dns server, but i don't see the *-relation here ;) |
07:59.27 | thx2000 | sendmail seems to bypass it |
07:59.51 | thx2000 | hang on, i might have a way to fix it, and it'll save a lot of explaining :) |
08:00.46 | tecnico | checked /etc/nsswitch.conf ?? make sure files is before dns |
08:05.36 | thx2000 | doesn't look like that file exists in osx |
08:06.14 | thx2000 | Is anyone still here to listen to me ramble about my problem? :P |
08:07.23 | tecnico | nsswitch is part of glibc , weird that you don't have it... |
08:07.38 | dlynes | thx2000: /etc/host.conf: line 1: order hosts,bind line 2: multi on |
08:07.54 | *** part/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net) |
08:08.08 | dlynes | host.conf afaik, is part of bind |
08:08.24 | tecnico | is sendmail bound to 127.0.0.1 as well ? or just your external IP ? |
08:08.28 | jj1 | to grab syslog messages from a pap2 to a special file anybody know the syslog.conf line? eg pap2.* /var/log/pap2.log? |
08:08.53 | thx2000 | im not sure |
08:09.14 | thx2000 | dlynes, that file doesn't exist, so just create a new one w/ those 2 lines? |
08:09.15 | tecnico | jj1: in syslog-ng you can create a reg. expr. filter.. if that's an option for you. |
08:09.37 | dlynes | jj1: there's no service defined for pap2 |
08:09.48 | dlynes | thx2000: Yes, you can create the file; ymmv |
08:10.00 | jj1 | k, what do most people do? throw syslog messages into mysql somehow? |
08:10.11 | dlynes | thx2000: it all depends on whether or not your bsdsockets actually makes use of that file |
08:10.13 | *** join/#asterisk NirS (n=NirS@62.90.49.98) |
08:10.22 | NirS | hello everybody |
08:10.29 | thx2000 | does sendmail cache mx records? |
08:10.39 | dlynes | thx2000: no..your dns cache does |
08:10.44 | jj1 | whats better SER or OpenSer? |
08:11.29 | dlynes | thx2000: your dns server will have that cache |
08:11.53 | NirS | Say, anyone ever tried working with /n with Manager originate method to Local channel ? |
08:12.01 | thx2000 | what if i created a new zone on my local dns server that should completely bypass the old record? |
08:12.12 | thx2000 | local meaning the dns server i use for my local network |
08:13.05 | dlynes | thx2000: why would you? |
08:13.35 | thx2000 | because it didn't look like sendmail was paying any attention to my hosts file |
08:13.36 | dlynes | thx2000: Why not just override your dns server with /etc/hosts? |
08:13.53 | thx2000 | i tried that and it was still tryin to send to the external ip |
08:13.57 | dlynes | thx2000: try commenting out the entries in your /etc/resolv.conf file |
08:14.19 | dlynes | thx2000: or renaming your /etc/resolv.conf file so sendmail can't find it |
08:14.24 | dlynes | thx2000: and then restart sendmail |
08:15.00 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
08:15.05 | dlynes | thx2000: What operating system are you running on? OSX? |
08:15.21 | thx2000 | yea, 10.4.4 i think |
08:15.48 | dlynes | yeah...i'm not totally familiar with it, but I do remember it's lacking a few files |
08:15.53 | thx2000 | Basically, im just tryin to get sendmail to use my mail server's local IP instead of the external one since my firewall wont accept loopback connections |
08:16.25 | dlynes | Ok, and what's that got to do with your nameserver or hosts, or anything else for that matter? |
08:16.45 | thx2000 | well the dns for the the domain im trying to send mail to is hosted on netsol |
08:16.49 | dlynes | Sendmail is your local mail server, right? |
08:17.16 | thx2000 | so when sendmail goes out to look for the ip for my mailserver it resolves to my external ip |
08:17.23 | dlynes | Or are you trying to use a smart host that's on your local network? |
08:17.53 | thx2000 | I'm trying to send the message to my kerio mailserver, on another box |
08:18.02 | dlynes | oh...nvm...it sounds like you haven't even configured sendmail yet |
08:18.10 | dlynes | It sounds like you're using the default configuration |
08:18.19 | thx2000 | probably |
08:18.22 | thx2000 | :P |
08:18.35 | dlynes | You need to define a smarthost in your /etc/mail/sendmail.cf or /etc/sendmail.cf file |
08:18.46 | dlynes | go to www.sendmail.org to learn how to set up a smart host |
08:19.00 | dlynes | And configure your smart host by ip address, not by host address |
08:19.07 | dlynes | that way it forces it to go to the lan address |
08:19.29 | thx2000 | neither of those files exist :/ |
08:19.52 | dlynes | find / -type f -name sendmail.cf |
08:21.40 | thx2000 | now that i've created a record in my local dns server, to point the mx record to the local ip of my mailserver, shouldn't that bypass it from finding the old record at all though? |
08:22.51 | dlynes | Well, that's not the proper way of doing it, for one |
08:22.59 | dlynes | And two, did you remove the old entry? |
08:23.24 | dlynes | In order to solve a problem properly, it usually helps if you understand why you're having a problem |
08:24.02 | dlynes | The smarthost option in sendmail was designed specifically for your problem in mind, where all mail gets sent to one specific mail server |
08:24.06 | thx2000 | I'm just frusterated and all the help i find has absolutely nothing to do with osx |
08:24.16 | thx2000 | so at this point anything that'll make this thing work im completely fine with |
08:24.21 | dlynes | Your problem is a sendmail problem, not an osx problem |
08:24.40 | dlynes | Sendmail doesn't look at /etc/hosts |
08:24.56 | thx2000 | correct |
08:24.57 | dlynes | Reason being is that it needs to do a reverse lookup |
08:25.03 | dlynes | And it can't do that with a hosts file |
08:25.42 | dlynes | It does the reverse lookup to make sure nobody's trying to use your mailserver as a spambot |
08:26.02 | dlynes | If you choose to configure sendmail to do that |
08:26.18 | thx2000 | ok, i understand that now |
08:26.23 | dlynes | but regardless, it still behaves that way whether you want that functionality or not |
08:26.34 | thx2000 | the search for sendmail.cf came up w/ nothing |
08:26.39 | dlynes | sendmail has a lot of power |
08:26.49 | dlynes | but with that power comes a lot of bulk |
08:27.06 | thx2000 | understood |
08:27.17 | dlynes | Do you have an /etc/mail directory? |
08:27.25 | thx2000 | no |
08:27.36 | dlynes | Ok, next question...is sendmail in your process list? |
08:27.46 | dlynes | I get the feeling your mailserver might not be sendmail |
08:28.02 | dlynes | It might be some other mailserver masquerading as sendmail |
08:28.18 | thx2000 | thats very possible |
08:28.21 | thx2000 | postfix exists |
08:28.31 | dlynes | Postfix is in your process list? |
08:29.15 | dlynes | The reason I suspect you're not using sendmail is because of the lack of existence of a sendmail.cf file |
08:29.28 | dlynes | That file is necessary for sendmail to start up |
08:29.57 | thx2000 | makes sense...i just figured it was there because it would respond through the terminal |
08:30.04 | thx2000 | i can do postfix start |
08:30.06 | dlynes | How so? |
08:30.13 | dlynes | I don't want you to do postfix start |
08:30.22 | dlynes | I want to find out what mail server you've got running, currently |
08:30.29 | dlynes | try this: |
08:30.48 | dlynes | ps auxffww | grep -E "postfix|sendmail|qmail|smail" |
08:31.09 | dlynes | Did you get any results? |
08:32.14 | thx2000 | well postfix, but i had started it before u told me not to :P |
08:32.43 | dlynes | Ok, so postfix is the only thing running in taht list? |
08:33.03 | thx2000 | root 238 0.0 0.1 27356 744 ?? Ss 1:28AM 0:00.07 /usr/libexec/postfix/master |
08:33.03 | thx2000 | postfix 239 0.0 0.1 27380 756 ?? S 1:28AM 0:00.03 pickup -l -t fifo -u |
08:33.03 | thx2000 | root 251 0.0 0.0 27820 4 p0 R+ 1:31AM 0:00.00 grep -E postfix|sendmail|qmail|smail |
08:33.08 | thx2000 | thats the full output |
08:33.11 | dlynes | There ya go |
08:33.19 | dlynes | postfix is your culprit then, not sendmail |
08:33.29 | dlynes | I know...you started it after I asked you not to |
08:33.40 | dlynes | but i suspect that's the only mail server you have installed, too |
08:33.41 | thx2000 | *before* |
08:33.47 | dlynes | or before :) |
08:33.49 | thx2000 | hehe |
08:34.00 | dlynes | Anyways...I can't help you with postfix, though |
08:34.04 | thx2000 | doh |
08:34.05 | dlynes | I know sendmail quite well |
08:34.10 | dlynes | But I don't know postfix at all |
08:34.33 | dlynes | But, I suspect it probably has an option for a smarthost as well |
08:34.38 | thx2000 | a heck of alot more than i know im sure :P |
08:34.41 | thx2000 | thanx for the help though |
08:34.50 | thx2000 | a shove in any direction is welcome at this point |
08:35.25 | dlynes | try http://freshmeat.net/projects/postfix for more info |
08:35.46 | dlynes | It should have a link to the homepage for postfix there |
08:35.59 | dlynes | On the postfix home page, you can probably find more info about setting up a smarthost |
08:39.20 | glm2k | thx2000: you might find this guide quite straightforward to follow. it is however a mysql/postfix install |
08:39.22 | glm2k | http://flurdy.com/docs/postfix/ |
08:39.41 | *** join/#asterisk heka (n=heka@82.114.68.124) |
08:39.47 | glm2k | just skip the section you don't need |
08:39.52 | glm2k | er, sections |
08:40.04 | thx2000 | cool, thanx |
08:41.47 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
08:42.03 | franck | Hi all |
08:42.37 | franck | When you compile zapata drivers it gives you several option for echo canceller, which one to use? |
08:42.46 | franck | I have a wctdm24xxp card |
08:42.51 | dlynes | franck: Are you using trunk? |
08:43.12 | franck | dlynes, all are FXO interfaces |
08:43.23 | dlynes | franck: If you're using trunk, you can use the MG2 echo canceller |
08:43.40 | dlynes | It's the best one, but I don't think it's available in zaptel 1.2.5 |
08:43.41 | franck | to call outside, inside is all sip phones |
08:44.46 | dlynes | have you tried MG2? |
08:45.04 | franck | this is the one I have compiled |
08:45.06 | franck | in |
08:45.20 | heka | Can I apply the ast_jb-1.2.0.patch3 to asterisk version 1.2.7.1 ? |
08:45.33 | franck | but I get in my asterisk log, cannot set echo cancelelr on channel ... |
08:45.52 | dlynes | you mean zaptel trunk, franck? |
08:46.30 | franck | <PROTECTED> |
08:46.48 | franck | this is the error message I get each time an outside line is used |
08:47.31 | franck | either to call out or receive a call |
08:48.00 | franck | I have Zap/g0 with Zap/1 to Zap/8 |
08:48.01 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
08:48.46 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
08:49.05 | dlynes | Yeah...sorry, dude...no idea what's wrong |
08:49.25 | franck | dlynes: ok thx for trying |
08:49.43 | dlynes | But then again, I'm using a PRI on one asterisk box, and x100p's on everything else |
08:52.10 | franck | I think the wctdm24xxp is a little bit new... |
08:52.41 | dlynes | it's the guy with 4 modules; each module has 6 fxs ports, or 6 fxo ports |
08:52.48 | dlynes | erm |
08:52.58 | dlynes | 6 modules, each with 4 fxs ports, or 4 fxo ports i mean |
08:53.15 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
08:53.24 | dlynes | right? |
08:54.24 | franck | yes |
08:57.38 | franck | you have 24 ports in group of 4 fxo or fxs |
08:57.47 | franck | I have 8fxo total on this card |
09:06.19 | dlynes | yeah...there's a couple people i've seen on here using that card |
09:21.11 | franck | dlynes: yes it puzzles me... this error message |
09:21.34 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
09:21.43 | franck | and especially where I am, I neve r know if it is the card or the other telcos who have a lot of echo |
09:27.50 | thx2000 | relayhost= ... thats all i needed |
09:31.01 | heka | anybody can help me about jitter buffer for sip? |
09:34.15 | RoyK | heka: using the patch from mantis? |
09:34.27 | RoyK | slav's stuff? |
09:35.43 | heka | RoyK: yes! |
09:36.36 | heka | I have try to use it with cdr_mysql but because of memory leak in earlier versions I couldn`t use it |
09:37.08 | heka | but in later versions of asterisk memory leak has been fixed so my question is: Can I apply that patch to version 1.2.7.1? |
09:38.27 | RoyK | I have a newer patch |
09:38.41 | heka | can you share it with me please? |
09:38.45 | RoyK | sure |
09:38.50 | RoyK | but the leak is still there |
09:39.31 | heka | RoyK: that`s bad bacause it dosent give me a chance to use it with any cdr gennerator |
09:39.41 | RoyK | why? |
09:40.12 | RoyK | after four days uptime on 1.2.6 asterisk eats something like 400 megs, so it's kinda ok. just need to restart it every now and then |
09:40.20 | heka | because while trying to put the data to database it get segfaulted |
09:40.23 | RoyK | have there been any leak fixes in 1.2.7? |
09:40.30 | RoyK | that's not the leak |
09:40.52 | RoyK | heka: what db? |
09:40.52 | RoyK | segfault != leak |
09:41.03 | heka | RoyK: I think that some fixes has bean done in 1.2.7 |
09:41.08 | RoyK | make a backtrace and report the crash in mantis |
09:41.24 | heka | RoyK: but it looks to me from the backtrace that it is the result of the leak |
09:41.37 | RoyK | obviously lots of fixes has been done in 1.2.7, but the leak is in the jitterbuffer code |
09:42.09 | heka | RoyK: let me get the newer patch and try with that please! |
09:42.53 | heka | are you thinking about oej`s svn version? |
09:43.13 | RoyK | no |
09:43.30 | RoyK | i had slav logged on to a test server to setup 1.2.6 with the jb |
09:43.53 | RoyK | we paid for this, and we're experiencing problems |
09:44.03 | RoyK | heka: email address? |
09:44.15 | heka | may I pm? |
09:45.19 | RoyK | sent |
09:45.40 | heka | thanks! |
09:45.50 | heka | have you patch the 1.2.6? |
09:46.01 | RoyK | wot_ |
09:46.02 | RoyK | ? |
09:46.04 | CMike | *yawn* morning all... |
09:46.09 | RoyK | CMike: morning |
09:46.16 | CMike | Hiyas RoyK ... allt väl ? :) |
09:46.21 | heka | have you apply this patch to the 1.2.6 version of asterisk or what version? |
09:46.39 | heka | CMike: morning! |
09:46.51 | CMike | morning.. |
09:47.02 | RoyK | CMike: alt vel :) |
09:47.07 | heka | RoyK: got it! let me give a try. |
09:47.26 | RoyK | heka: please do report that segfault anyway |
09:48.05 | heka | RoyK: I`ll do as soon as Im done with this patch. what do you think about trying to apply this patch against 1.2.7.1? |
09:48.05 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
09:48.14 | RoyK | should work |
09:48.19 | heka | ok |
09:48.35 | RoyK | heka: btw, I use cdr_mysql in production and have been for almost two years |
09:49.08 | CMike | me too :) |
09:49.34 | CMike | maybe I should upgrade.. my servers.. |
09:49.56 | heka | RoyK: both with jitter buffer patch? |
09:51.07 | CMike | wtf.. hm .. a big pigeon just sat down om my sat.dish .. my picture disapeard.. *mumble* |
09:51.08 | CMike | BRB |
09:52.41 | *** join/#asterisk robin_sz (n=nospam@adsl.redpoint.org.uk) |
09:52.46 | robin_sz | meep? |
09:53.22 | robin_sz | and .. as expected, its worse than the previous version :( |
09:53.40 | robin_sz | going backwards and fast :( |
09:56.36 | RoyK | heka: with or without - doesn't matter |
09:57.57 | robin_sz | I wonder if there would be any interest in making Grnadstream GXP2000s into something useful? |
09:58.03 | robin_sz | like ... plant pots? |
09:59.14 | *** part/#asterisk heka (n=heka@82.114.68.124) |
10:12.55 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
10:19.16 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
10:34.09 | *** join/#asterisk heka (n=heka@82.114.68.124) |
10:34.19 | heka | Hi |
10:34.22 | heka | RoyK: ? |
10:35.13 | RoyK | ka-ding |
10:35.43 | heka | the same hapens with the new patch |
10:35.49 | RoyK | what happens? |
10:35.51 | heka | do you have time to look at the backtrace? |
10:36.02 | RoyK | what happens? |
10:36.09 | heka | http://pastebin.com/662839 |
10:36.09 | RoyK | coredump? where? |
10:36.51 | heka | this is with another cdr application |
10:37.03 | heka | not cdr_mysql |
10:37.14 | RoyK | app_prepaid_call.c |
10:37.17 | RoyK | pastebin the dialplan as well |
10:38.17 | heka | dial plan is simple! Im calling the app_prepaid_call wich handles the dial plan. puts the initial data in cdr table of database |
10:38.24 | heka | like call id, start time etc etc |
10:38.35 | heka | and the completes it after the call is finished |
10:38.37 | *** join/#asterisk ToTo (n=ToTo@host110-142.pool874.interbusiness.it) |
10:40.08 | RoyK | perhaps the same bug as this |
10:40.08 | RoyK | http://bugs.digium.com/view.php?id=6846 |
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10:40.22 | RoyK | hm |
10:40.30 | RoyK | no |
10:40.30 | RoyK | not really |
10:40.44 | RoyK | heka: but pb the config and let me see if i can reproduce it |
10:41.08 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
10:41.30 | FuriousGeorge | exten => s,n(repeat),setvar(COUNT = $[${COUNT} + 1]) |
10:41.38 | FuriousGeorge | this is constantly evaluating to the same thing for me |
10:42.38 | RoyK | s/setvar/set/ |
10:42.45 | FuriousGeorge | i tried both ways |
10:42.47 | RoyK | but that shoulnd't really matter |
10:42.54 | FuriousGeorge | and what i meant to say was that it keeps evaluating to two |
10:43.01 | heka | RoyK: wich configs should I pb? |
10:43.03 | RoyK | what asterisk version? |
10:43.11 | RoyK | heka: all relevant |
10:43.44 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
10:45.02 | RoyK | also |
10:45.02 | RoyK | heka: i need a url to the prepaid app you're using |
10:45.25 | FuriousGeorge | http://pastebin.ca/49496 |
10:45.33 | FuriousGeorge | can anyone tell me why im not getting out of that loop |
10:45.46 | FuriousGeorge | i know its b/c my counter is not incrememnting but i dont see why it shouldnt |
10:47.15 | RoyK | FuriousGeorge: what version of asterisk? |
10:47.19 | FuriousGeorge | 1.2.6 |
10:47.54 | RoyK | k |
10:48.15 | FuriousGeorge | its showing the counter being increased from 1 to 2 but it doesnt get any farther, and i dont see any cli output for the background and noop there |
10:48.19 | FuriousGeorge | so im real confused |
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10:50.40 | *** join/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) |
10:50.48 | snitt | Mw3: respect |
10:51.13 | RoyK | FuriousGeorge: the while statement is wrong |
10:51.35 | FuriousGeorge | which statement? the counter increment? |
10:51.49 | FuriousGeorge | oh sorry |
10:51.57 | FuriousGeorge | i read the "whole statement" |
10:52.24 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
10:52.32 | RoyK | it should be |
10:52.32 | RoyK | exten => s,n,while($[ ${COUNT} < 7 ]) |
10:52.41 | puzzled | morning |
10:52.50 | FuriousGeorge | oh yeah |
10:53.06 | RoyK | puzzled: morning |
10:54.41 | RoyK | FuriousGeorge: and... |
10:54.48 | RoyK | FuriousGeorge: try this |
10:54.50 | RoyK | FuriousGeorge: exten => s,n,set(COUNT=$[ ${COUNT} + 1 ]) |
10:54.56 | RoyK | that works for me |
10:55.17 | RoyK | seems like asterisk doesn't handle spacing too well around the '=' |
10:57.13 | puzzled | and wasn't there the opposite too, where you have to use spacing or it won't work? |
10:57.24 | wasim | oui |
10:58.00 | RoyK | puzzled: i beleive there was, in the $[ asdf ] thing, but iirc that's fixed |
10:58.23 | puzzled | ah ok |
10:58.28 | FuriousGeorge | RoyK: thanks the counter is working now |
10:58.45 | FuriousGeorge | i would have never figured out that spacing thing |
10:59.15 | RoyK | don't use spaces around = |
10:59.17 | RoyK | use spaces after [ and before ] |
10:59.46 | RoyK | and between elements between [ and ] |
11:02.36 | wasim | ofcourse this may all change without notice |
11:02.42 | puzzled | imho that should be sooo fixes to take spaces or not |
11:02.48 | puzzled | wasim: indeed |
11:04.19 | Rawplayer | vrolijk paasfeest allemaal |
11:04.26 | FuriousGeorge | call me crazy but my counter is getting up way past counter < 7 and the loop is STILL not exiting |
11:04.33 | RoyK | wasim: what? asterisk is _stable_ and _productional_ and doesn't _change_ unless there's a _major_ upgrade |
11:04.49 | RoyK | _and_it_is_the_best_pbx_in_the_word_and_so_on_ |
11:04.56 | Rawplayer | en dikke eieren |
11:05.16 | RoyK | Rawplayer: alt vel? trenger du hjelp? medisiner? |
11:06.18 | Rawplayer | lol |
11:06.23 | FuriousGeorge | my dialplan says this: exten => s,n,while($[${COUNT}<7]) |
11:06.25 | Rawplayer | valium |
11:06.26 | FuriousGeorge | and the CLI says |
11:06.35 | RoyK | FuriousGeorge: http://pastebin.com/662870 this one works for me |
11:06.35 | FuriousGeorge | -- Executing While("Zap/2-1", "1") in new stack |
11:07.54 | RoyK | wasim: and STABLE doesn't mean it doesn't crash, but that new features aren't added.... |
11:09.04 | FuriousGeorge | RoyK: shouldnt the CLI say something else besides - Executing While("Zap/2-1", "1") in new stack |
11:09.09 | RoyK | no |
11:09.16 | FuriousGeorge | considering my while has the > operator and a #7 in there |
11:09.28 | RoyK | the $[ bladdi ] is evaluated and the CLI returns the evaluated result |
11:09.56 | FuriousGeorge | ok, but we just got the counter working, so shouldnt that 1 at least change to a 2 |
11:09.58 | FuriousGeorge | and so on |
11:10.09 | RoyK | no |
11:10.18 | FuriousGeorge | ok |
11:10.34 | FuriousGeorge | so i'm gonna assume this part is working and go on to fix the next part :) |
11:10.38 | FuriousGeorge | tomorrow |
11:10.38 | RoyK | try NoOp( $[ 1 > 2 ] ) and NoOp( $[ 1 > 1 ] ) |
11:10.48 | RoyK | the result returned from $[ asdf ] is a bool |
11:11.01 | FuriousGeorge | i see what you mean |
11:13.17 | puzzled | Rawplayer: same to you :) |
11:13.36 | Rawplayer | :D |
11:13.44 | isamar | anybody using SER? |
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11:14.30 | *** part/#asterisk popvoxdave (n=popvoxda@c-68-49-123-118.hsd1.md.comcast.net) |
11:16.10 | RoyK | no! |
11:16.55 | puzzled | that harry guy is :) |
11:21.40 | FuriousGeorge | jesus, im confused by this new n(label) priority. apparently if im still in extension s then goto(label) works, but if im not in that extension goto(s,label) doesnt |
11:21.54 | FuriousGeorge | same context though |
11:22.15 | RoyK | don't |
11:22.15 | RoyK | just use goto(label) |
11:22.37 | FuriousGeorge | im told it doesnt exist |
11:22.46 | RoyK | pb dialplan again :) |
11:23.05 | RoyK | hm |
11:23.15 | FuriousGeorge | hold on, this is another part |
11:24.57 | RoyK | i think labels are local only |
11:24.57 | RoyK | local to the context |
11:27.00 | FuriousGeorge | http://pastebin.ca/49497 im always in the same context, but now its a different extension |
11:27.17 | FuriousGeorge | what a mess this has become :) |
11:28.05 | FuriousGeorge | im sure there are other mistakes but im just confused by the lables right now |
11:28.47 | wasim | FuriousGeorge: well said |
11:29.07 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
11:29.11 | FuriousGeorge | wasim: which part |
11:29.24 | RoyK | er |
11:29.25 | RoyK | exten => _${ROOMATE[${LASTCOUNT}]},1,random(${MATEWEIGHT[${LASTCOUNT}]},${ROOMATE[${LASTCOUNT}]},3) |
11:29.59 | RoyK | can you use a variable like in the search part of an extension? |
11:30.09 | wasim | 16:27 < FuriousGeorge> what a mess this has become :) |
11:30.11 | FuriousGeorge | i was wondering the same thing |
11:30.25 | FuriousGeorge | i guess you can because its complaining about the goto |
11:30.49 | FuriousGeorge | wasim: if you can think of a better way to give 5 variables a relative weight and pick one based on that, im all ears |
11:31.02 | wasim | no, no, i meant you were talking about * |
11:31.04 | RoyK | what is a roo-mate? :) |
11:31.36 | RoyK | kangaroo-mate |
11:31.38 | RoyK | :) |
11:31.42 | FuriousGeorge | wasim: well tbh, the spaces thing with my = before did piss me off a bit :) |
11:32.01 | RoyK | FuriousGeorge: welcome, to the real world... |
11:33.12 | FuriousGeorge | any idea about that label though, royk? i really would like to undsertand wtf is happening with that |
11:36.35 | isamar | anyone using SER? |
11:36.53 | isamar | what is the most common use of ser+asterisk ? |
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11:37.15 | FuriousGeorge | isamar: as a sip gateway |
11:37.27 | FuriousGeorge | for handling many many sip user registrations |
11:37.33 | FuriousGeorge | routiong, etc |
11:37.35 | FuriousGeorge | or so i hear |
11:38.02 | X-Rob | oooh |
11:38.08 | X-Rob | new series of dr who has started |
11:38.14 | X-Rob | mininova is your friend |
11:38.33 | FuriousGeorge | on hbo |
11:39.10 | FuriousGeorge | RoyK: actually forget that label, im not gonna eork on this anymore till i get some sleep. if i didnt know any better i'd say its almost working |
11:39.46 | FuriousGeorge | i jsut wanna know if there is some glaring error that is gonna force me to start over :) |
11:41.44 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
11:42.03 | BladeRunner05 | Happy easter to all |
11:42.35 | FuriousGeorge | same 2 u |
11:43.20 | BladeRunner05 | I need to use my eicon diva server bri to asterisk 1.2.7.1, I have installed chan_capi-com.0.6.5 but when I start asterisk it say: chan_capi.c:4577 cc_init_capi: CAPI not installed, CAPI disabled! |
12:03.01 | puzzled | BladeRunner05: what does capiinfo say? |
12:09.12 | *** join/#asterisk Garaan (n=jfleisch@user-142h64a.cable.mindspring.com) |
12:09.21 | Garaan | Good morning |
12:09.59 | Garaan | Anyone not AFK? |
12:16.09 | Garaan | Is anyone on? |
12:17.51 | tzafrir | ~ask |
12:17.52 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a quesiton first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
12:18.31 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
12:19.52 | snitt | :) |
12:21.13 | Garaan | I am currently trying to configure zaptel-1.0.10 with an X100P clone and a TDM400P with 1 FXS module installed. The reason for the old version is that is what is currently in place at my work location. I am having issues making the cards initialize at boot, as the /etc/init.d/zaptel script sees the error from wcfxo loading and not seeing the fxs port and quits. Any suggestions? |
12:23.40 | Garaan | Link for zaptel.conf and error from /etc/init.d/zaptel http://pastebin.ca/49500 |
12:26.06 | Mw3 | hi. how can i use an auth username with @ in it in sip.conf (register =>)? is there any way to escape the @ ? |
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13:03.27 | cced2 | :_) |
13:04.03 | cced2 | ? |
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13:26.24 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-88-154-249-174.red.bezeqint.net) |
13:26.34 | PoWeRKiLL | hi |
13:27.22 | PoWeRKiLL | I have a strange thing with my sipura when I'm in a call and call a call wiating I can see the caller id but when there is no call I can't see the caller id any idea ? |
13:40.23 | *** join/#asterisk coppice (n=chatzill@208.197.17.210.dyn.pacific.net.hk) |
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13:43.09 | cced2 | <cced2> IN chan_zap.c start_pri() pri->fds[i] = open("/dev/zap/channel", O_RDWR, 0600); |
13:43.10 | cced2 | <cced2> use /dev/zap/channel as dchannel? |
13:43.34 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
13:46.48 | cced2 | :) |
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14:00.20 | WeeZyyy | Good Morning Everyone |
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14:13.56 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
14:15.09 | n0cturnal_ | is it possible to have any calls from exten x go out one route, but any call from exten z go out another ? |
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14:37.35 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
14:38.12 | MoutaPT | hi, is there any simple tip to make CDR of all calls that go to my voicemail if i'm busy or unavailable? |
14:38.48 | tzanger | what does that have to do with CDR? |
14:39.23 | tzanger | Dial() with option j will jump to n+101 if you can't be contacted, and you can use that to go to VoiceMail()if you're busy |
14:39.37 | tzanger | and add a timeout to Dial and n+1 to go to voicemail if you're unavailable (i..e the Dial() times out) |
14:40.22 | MoutaPT | yes but i want to make record that this call went to voicemail, and i didn't pickup that call |
14:41.34 | MoutaPT | then i would be able to measure how many calls i receive per month , and how many of them go to the voicemail, because i didn't pick it up |
14:41.52 | coppice | why do some people believe 64kbps is some magic number that creates perfect phone calls? :-) |
14:44.45 | tzanger | coppice: :-) |
14:45.20 | tzanger | MoutaPT: you can alter the user field with Set(CDR(...)) before hitting the VoiceMail() app |
14:45.27 | tzanger | show function CDR will give you details |
14:45.36 | tzanger | 64kbps is teh winz |
14:45.39 | MoutaPT | thanks tzanger! |
14:46.31 | coppice | tzanger: I think the same clowns listen to MP3s at 64kbps and never notice how much better than a phone call they sound :-) |
14:48.23 | tzanger | meh. that's just because we can't convince the telcos to do CBR MP3s or OGGs in the trunks |
14:56.58 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
14:57.09 | websae | Happy Easter everyone! |
15:00.48 | Damin | Happe Easter... |
15:02.11 | robin_sz | easter would be better if my GXP2000 was working :( |
15:02.53 | robin_sz | wheres helix and/or thetatag when you want someone to whinge at? huh? |
15:06.34 | MoutaPT | exten => s-BUSY,2,Set(CDR(dst)=Voicemail_busy) |
15:06.34 | MoutaPT | exten => s-BUSY,3,Voicemail(b${ARG1}@${VMCONTEXT}) |
15:06.47 | MoutaPT | is this enough to get CDR of a call that went to voicemail? |
15:07.02 | MoutaPT | i couldn't get it yet.. |
15:11.05 | BladeRunner05 | puzzled: r u there ? |
15:15.03 | BladeRunner05 | I need to use my eicon diva server bri to asterisk 1.2.7.1, I have installed chan_capi-com.0.6.5 but when I start asterisk it say: chan_capi.c:4577 cc_init_capi: CAPI not installed, CAPI disabled! |
15:15.21 | BladeRunner05 | mits on debian kernel 2.6.15 |
15:16.30 | puzzled | BladeRunner05: check the output of capiinfo |
15:18.00 | BladeRunner05 | capiinfo say: capi not installed - No such device or address (6) |
15:19.11 | puzzled | BladeRunner05: you have to load the capi modules first, then start asterisk with chan_capi-cm |
15:20.34 | puzzled | BladeRunner05: here's what I have in /etc/rc.local http://pastebin.com/663188 |
15:21.51 | BladeRunner05 | ok, take a look puzzled |
15:24.24 | BladeRunner05 | puzzled: ok now the driver are loaded correctly, I also copy the firmware in /usr/share/eicon ... |
15:24.59 | BladeRunner05 | can I see a copy of your capi.conf file ? |
15:25.10 | puzzled | BladeRunner05: no :) |
15:25.15 | puzzled | it's mine! |
15:26.30 | puzzled | BladeRunner05: there is nothing special to see in my capi.conf. The author has it commented very well. just read it thoroughly |
15:27.19 | puzzled | BladeRunner05: I only added the msn number and changed the callgroup and pickupgroup |
15:27.32 | puzzled | like that it works fine for me |
15:27.57 | BladeRunner05 | kk |
15:31.35 | tecnico | anyone knows of a AGI or any type of script or way to let iax2 client users to know who else is "online" (registered) , without using Jabber/aim/etc. ? Maybe a webpage ? |
15:31.46 | *** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) |
15:37.06 | x86 | tecnico: dont know of one existing, but i could write you some web software to do it if you wanted to hire me |
15:37.48 | tecnico | looks like there was someone already with an app (http://www.sineapps.com/news.php?rssid=342), but the link is dead... |
15:38.11 | tecnico | tnx. x86 , but this is just personal.. not for profit.. wouldn't make sense for me to pay anyone |
15:38.25 | x86 | tecnico: maybe you could trade something then? |
15:39.37 | tecnico | I could tell you where to maybe get a free asterisk enabled VPS account. (ztdummy/etc enabled) |
15:40.21 | x86 | CPS? |
15:40.22 | x86 | err |
15:40.23 | x86 | VPS? |
15:40.36 | tecnico | virtual private server... you get your own root... arround 5GB |
15:40.45 | x86 | free? |
15:41.17 | tecnico | maybe.... can't guarantee |
15:41.35 | BladeRunner05 | puzzled: r u using debian ? |
15:41.54 | puzzled | BladeRunner05: nope, FC4/5 and CentOS43 |
15:42.01 | BladeRunner05 | kk |
15:44.07 | RoyK | hm. nice sensors output http://pastebin.ca/49514 i wonder what they were smoking. |
15:46.18 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
15:47.45 | RoyK | hehe |
16:00.13 | *** join/#asterisk backblue (n=moo@87-196-79-141.net.novis.pt) |
16:02.07 | Corydon76-home | Lucas_Fernando: please turn off your auto-away announce. |
16:04.46 | *** join/#asterisk azzie (i=az@cpe-24-168-17-173.si.res.rr.com) |
16:09.56 | dlynes | tecnico: try flash operator panel |
16:10.20 | tecnico | tnx dlynes |
16:10.42 | dlynes | tecnico: it's overkill for what you want, but it does what you want, too |
16:11.09 | BladeRunner05 | I'm getting error compiling mpg123 on debian kernel 2.6.15 i486, I try make linux, make linux-486, make linux-pentium, make generic but always get errors |
16:11.18 | tecnico | dlynes: tnx...I remember it from Astricon2004 but never used it.. |
16:11.35 | dlynes | tecnico: Yeah...it's pretty straightforward to set up |
16:11.47 | robin_sz | bladerunner: which mpg123 .. the standard one or the one that comes with *? |
16:12.22 | ManxPower | robin_sz, they are the same |
16:12.23 | BladeRunner05 | I try both .r and .s get it from http://www.mpg123.de/mpg123/mpg123-pre0.59s.tar.gz |
16:12.39 | ManxPower | Well s prolly isn't going to work with Asteirsk. |
16:12.47 | ManxPower | Where did you paste the error message? |
16:12.59 | robin_sz | ManxPower: thought the * one was tweked a bit ... no? |
16:12.59 | dlynes | BladeRunner05: I wouldn't even suggest using mpg123...it's got a really bad bug in it, where the process loses contact with asterisk |
16:13.10 | BladeRunner05 | but .r version won't compile |
16:13.13 | dlynes | BladeRunner05: madplay plays nicer with asterisk |
16:13.30 | dlynes | BladeRunner05: and asterisk 1.2 also supports mpg natively for moh; you don't need an external player any more |
16:13.41 | robin_sz | kewl |
16:13.56 | BladeRunner05 | really ? I have compiled 1.2.7.1 and I don't need mpg123 ? |
16:14.08 | robin_sz | seems not |
16:14.10 | dlynes | BladeRunner05: nope...use mode=files |
16:14.27 | RoyK | native moh starts the music for every new call |
16:14.31 | RoyK | i dislike that |
16:14.36 | robin_sz | now ... fix my GXP2000 for me ;) |
16:14.40 | RoyK | so i use a sox wrapper instead.... |
16:14.42 | dlynes | RoyK: so use madplay then :) |
16:14.45 | BladeRunner05 | dylnes: where I have to use it ? |
16:15.04 | dlynes | RoyK: I just hate mpg123 because when it loses contact with asterisk, it hogs up 20% of the cpu |
16:15.33 | dlynes | Lucas_Fernando: lose your autoaway |
16:16.13 | dlynes | BladeRunner05: if you take a look at your sample musiconhold.conf file, it gives you an example for mode=files |
16:16.30 | dlynes | BladeRunner05: So just copy that info into your [default] musiconhold class |
16:16.40 | BladeRunner05 | ok |
16:16.44 | RoyK | dlynes: they all do that |
16:16.56 | dlynes | RoyK: all what do what? |
16:17.32 | dlynes | RoyK: all external players hog up 20% of your cpu? |
16:17.58 | dlynes | RoyK: I've never had madplay do that; only mpg123 |
16:18.04 | RoyK | sox does it |
16:18.23 | RoyK | my soxwrapper spawns a sox process and that process hangs |
16:18.28 | dlynes | what does sox have to do with mpg123? |
16:18.29 | RoyK | same with mpg123 |
16:18.52 | ManxPower | ribnope |
16:18.54 | dlynes | or madplay for that matter? |
16:19.00 | ManxPower | robin_sz, nope |
16:19.02 | RoyK | it doesn't matter what software. the problem is that children of the called process aren't killed |
16:19.16 | dlynes | RoyK: Whether they're killed or not |
16:19.18 | RoyK | but if madplay doesn't spawn new children, it should work well |
16:19.30 | dlynes | RoyK: I've never had an issue with madplay using up 20% of the cpu |
16:19.39 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
16:19.42 | RoyK | modplay doesn't spawn a child, then, does it? |
16:19.44 | dlynes | RoyK: I've never had an issue with more than one process of madplay running, either |
16:19.59 | RoyK | or perhaps it kills the child if it receives a sigterm |
16:20.14 | dlynes | RoyK: Wouldn't that be the way it's supposed to run? |
16:20.22 | dlynes | RoyK: i.e. play nice with the OS? |
16:21.35 | dlynes | RoyK: to me, any process that hogs up 20% of the cpu for no good reason is a security risk |
16:21.54 | RoyK | it is |
16:22.02 | dlynes | RoyK: Especially when it causes my main process not to respond in a timely fashion (asterisk) |
16:22.29 | dlynes | I haven't used mpg123 for quite some time now |
16:22.49 | dlynes | Whether it starts from the beginning or not is of little concern to me, when it's doing stuff like that |
16:22.49 | RoyK | asterisk calls soxwrapper, soxwrapper calls sox, asterisk stops and soxwrapper is killed, sox goes on. same story with mpg123 |
16:23.18 | dlynes | Then it's a bug in mpg123 and soxwrapper |
16:24.03 | dlynes | I never used it for long enough to find out why it was doing it |
16:24.12 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
16:24.19 | dlynes | As soon as I found out it was doing it, I started looking for alternatives |
16:24.33 | dlynes | That's when I found out about madplay on the musiconhold part of the asterisk wiki |
16:25.06 | dlynes | Well..it was actually the musiconhold/faxing part of the asterisk wiki |
16:26.05 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
16:26.24 | dlynes | http://www.bartroos.com/asterisk/ <-- Asterisk + ISDN HFC_PCI + Music-on-hold + Soft fax HOWTO |
16:27.22 | dlynes | You can also get libmadplay and write your own music-on-hold module |
16:28.27 | dlynes | But, if you like sox, soxwrapper probably wouldn't be that hard to fix |
16:31.17 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net) |
16:31.38 | wwalker | What about a patch to asterisk that starts the external moh player at nice 20? then even if it goes into runaway it should always relinquish CPU to asterisk when asterisk wants it? |
16:36.52 | wasim | any body use astpp? |
16:43.03 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
16:43.51 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
16:43.58 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
16:47.33 | dlynes | wwalker: that's not a fix though...that's an ignore the problem and hope it goes away philosophy |
16:48.01 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
16:48.19 | *** join/#asterisk Dream_WEaver (i=philw@S01060040f41fb6ad.ed.shawcable.net) |
16:49.42 | wwalker | I miscommunicated. The soxwrapper problem should be fixed, but many things can make the child go wild, therefore, I'd always want any MoH child to be niced. |
16:50.07 | Hmmhesays | hey hey, you you, get off of my cloud |
16:51.27 | wwalker | I first saw vi runaways over 15 years ago. I thought they had them fixed for years. I've recently been seeing runaway vi/vim processes again. someone introduced a new bug. Somewhere someone is reading from the tty and not catching that it's gone...ina tight loop, 99%cpu.... |
16:51.59 | wwalker | So, I expect that sox*, mpg123, and libmad derived works all have that same probable failure. |
16:54.02 | Hmmhesays | this is rock n roll radio, come on lets rock n roll with the ramones |
16:55.50 | MoutaPT | how do i enable call waiting for every SIP user i create? any tip? |
16:56.45 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
17:02.54 | dlynes | wwalker: ah....but libmad doesn't have that failure, afaik...it doesn't have any child processes...only the main process |
17:03.05 | dlynes | s/libmad/madplay |
17:05.36 | *** join/#asterisk Araluccl0 (n=ciccio@adsl-ull-1-4.46-151.net24.it) |
17:13.17 | dlynes | bbl\ |
17:18.14 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
17:18.45 | *** join/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
17:18.49 | SkramX | I have a very nice pipe for our company's servers but how do you all do QoS? |
17:18.58 | BladeRunner05 | what does it means: Music class default requested but no musiconhold loaded. |
17:28.15 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
17:30.29 | *** part/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
17:32.19 | *** join/#asterisk Tili (n=Tili@219.137.200.72) |
17:35.39 | Qwell | Lucas_Fernando: stop that |
17:35.43 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
17:37.12 | wasim | oui, non hable espanol, capice? |
17:37.58 | BladeRunner05 | what does it means: Music class default requested but no musiconhold loaded. |
17:48.06 | *** join/#asterisk talljon84 (n=jonathan@68-185-182-59.dhcp.mdsn.wi.charter.com) |
17:48.41 | *** mode/#asterisk [+b %Lucas_Fernando!*@*] by file |
17:49.16 | talljon84 | Hi all-- I have a VM box that has a blank message in it that the user can not delete. It always says they have one new message in the mailbox but there is no recorded message. How do I clear that? |
17:49.17 | Hmmhesays | you gots no music |
17:49.27 | Hmmhesays | delete it out of the users folder |
17:50.29 | talljon84 | will Asterisk recreate it on it's own? |
17:50.40 | Hmmhesays | delete the voicemail file itself |
17:50.44 | Hmmhesays | not the whole folder |
17:50.47 | *** join/#asterisk rene- (n=rene-@dsl-201-128-115-107.prod-infinitum.com.mx) |
17:50.52 | talljon84 | ok. |
17:50.56 | rene- | ~seen Qwell[] |
17:51.07 | jbot | qwell[] <i=north@unaffiliated/qwell> was last seen on IRC in channel #asterisk, 1d 20h 25m 15s ago, saying: 'VoIPMasta: anything that gets added to the bug, will get emailed to me'. |
17:51.07 | Qwell | ? |
17:51.09 | rene- | heh |
17:51.25 | rene- | Qwell: what is the name of the realtime family for agents |
17:51.43 | Qwell | I don't know. Agent? |
17:52.07 | Hmmhesays | Qwell, make me some coffee |
17:52.18 | talljon84 | Hmmmhesays: awesome. that worked. thanks a ton |
17:52.25 | Hmmhesays | np |
17:53.04 | rene- | I downloaded trunk 1.2 from svn, i am going to try that |
17:53.46 | talljon84 | Ok, now next problem: I have a lawyer who has a legal research program he uses. It's only method of update is through a dialup connection that IT must dial. Faxing works fine for me over SIP; however, using the same SIP ATA, the computer won't dialup. Is there any extra configuration anyone knows of to make a dialup connection work across SIP? (using a Sipura ATA) |
17:54.29 | BladeRunner05 | Problem: with asterisk 1.2.7.1 in modules.conf I have load res_musiconhold.so, in musiconhold.conf I have enabled the default, but when I make a call and press hold button asterisk say Music class default requested but no musiconhold loaded, what this mean ? what I have to do ? |
17:54.36 | Hmmhesays | gotta look at the error messages the program spits back |
17:54.46 | *** join/#asterisk jofre (n=jofre@200.215.42.23) |
17:55.56 | nettie | Hey guys anyone know what's the latest firmware for the grandstream handytone 286 please? Mine is Program-- 1.0.7.19 Bootloader-- 1.0.8.9 HTML-- 1.0.7.18 VOC-- 1.0.0.10 and I'm having stability issues during heavy T.38 faxing. Any idea please? |
17:57.58 | *** join/#asterisk MstlyHrmls (n=mh@66.193.14.132) |
17:58.02 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
18:03.53 | Qwell | rene-: realtime agents won't be in 1.2 |
18:04.07 | *** join/#asterisk IceManRISK (n=kart@201.10.99.247) |
18:05.16 | rene- | i tried to configure it the same way as rt queues and that failed, i am looking at the patch, is dated 11/05 i wonder if i can apply it to SVN trunk 1.2... |
18:05.44 | IceManRISK | anyone here uses asterisk with a2billing ? |
18:13.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-92-37.red.bezeqint.net) |
18:16.01 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
18:16.42 | wiseguy_ | sorry, is it possible to do something (extensions.conf) after call ends? |
18:17.24 | rene- | wiseguy_: just add a priority after your dial or hangup statement |
18:17.40 | *** join/#asterisk heka (n=heka@82.114.68.124) |
18:17.44 | rene- | are you receving or making this call? |
18:18.28 | wiseguy_ | i have something like that |
18:18.30 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
18:18.44 | *** join/#asterisk lyroy (n=toor@modemcable146.87-83-70.mc.videotron.ca) |
18:19.03 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
18:19.07 | Assid | heylo |
18:19.11 | wiseguy_ | http://pastebin.ca/49525 |
18:19.33 | Assid | anyone got a changelog available from 1.2.7 all the way down to 1.2.4 |
18:19.35 | wiseguy_ | but system command is not executed after call ends.. |
18:20.14 | lyroy | If I have more than 1 sip device, and when I receive a call I want all my sip device to ring, after someone answer the call, the other device should not ring... how it's possible with asterisk? |
18:20.47 | Assid | lyroy: just use the dial cmd |
18:22.44 | rene- | wiseguy: in your example system gets executed both when the call finished and also when the call cant be connected |
18:22.59 | rene- | e.g. no answer |
18:23.00 | Assid | Dial(SIP/blah&SIP/user2&SIP/user3) |
18:23.38 | Assid | err.. anyone jhave a changelog ? |
18:23.49 | rene- | wiseguy_: if this is not what you want you can take a look at defining an h extension for your incoming context |
18:24.03 | rene- | wiseguy_: http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension |
18:24.11 | lyroy | alrigth thx assid |
18:25.57 | wiseguy_ | rene-: ghem, the problem is the command is not executed in this case |
18:26.38 | wiseguy_ | :/ |
18:26.40 | rene- | well that is because when the other party cant connect, your priority will jump to +101 |
18:27.16 | rene- | http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities |
18:27.36 | rene- | so http://pastebin.ca/49527 |
18:30.15 | wiseguy_ | it does not work |
18:30.16 | wiseguy_ | :/ |
18:31.06 | rene- | is the dial command connecting the call? |
18:31.26 | wiseguy_ | yes, the dial command |
18:31.29 | wiseguy_ | goes okaj |
18:32.36 | wiseguy_ | rene-: any other ideas? |
18:32.52 | rene- | well if the dial goes ok, then priority 103 doesnt get executed but 3 should |
18:33.11 | rene- | re add the priority 3 above priority 103 |
18:33.53 | cybergypsy | wiseguy - have you tried the g option ? |
18:34.01 | cybergypsy | wiseguy - i have a similar problem |
18:34.18 | wiseguy_ | g option? |
18:34.33 | cybergypsy | g: When the called party hangs up, exit to execute more commands in the current context. |
18:34.54 | wiseguy_ | cybergypsy: can you paste an example? |
18:35.00 | wiseguy_ | to pastebin.ca |
18:35.44 | cybergypsy | exten => _0044808XXXXXXX,n,Dial(IAX2/iaxfwd/*${EXTEN:2},60,hHg) ; UK 0808 Numbers - Free |
18:35.52 | *** join/#asterisk _42 (n=r2d2@nvader.sh.nu) |
18:35.53 | PakiPenguin | did anyone try configuring a TE110P with mitel? |
18:36.20 | _42 | Anybody have any experience in getting asterisk working with Sunrocket's voip? www.sunrocket.com |
18:36.22 | rene- | PakiPanguin, they work great both in R2 and ISDN modes |
18:37.27 | wiseguy_ | cybergypsy: it doesn't work for me |
18:37.55 | _42 | Nevermind, Sunrocket blocks access to their SIP servers |
18:38.38 | PakiPenguin | rene-, can you share your zaptel configuration please? |
18:38.57 | PakiPenguin | i have a mitel 3300 right now , and the the e1 card is in red alarm constantly |
18:39.05 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
18:39.25 | rene- | well if its in red then the problem is with your cable i think |
18:39.48 | rene- | i no longer have access to configs, are you using ISDN? |
18:40.17 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
18:40.31 | rene- | does zttool shows your card as being up? |
18:40.58 | PakiPenguin | yes |
18:41.03 | PakiPenguin | for the ISDN |
18:41.18 | PakiPenguin | Alarms Span â |
18:41.18 | PakiPenguin | <PROTECTED> |
18:41.27 | PakiPenguin | and this is the zttool output |
18:41.28 | rene- | what type of cable are you using to link the mitel and asterisk? |
18:41.44 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
18:43.34 | PakiPenguin | a sec |
18:45.03 | wiseguy_ | rene-: have you got an working example? I need to execute that command when call ends not depending on the reason of end |
18:49.20 | talljon84 | I'm trying to setup outbound SIP calling on * but even if I DMZ the server, it always fails when the remote end trys to answer the phone. It will ring, but when they answer, it fails. Any ideas? |
18:49.46 | rene- | talljon84: codecs? |
18:50.16 | talljon84 | the trunk is alowing g729 or ulaw |
18:50.29 | rene- | use of stun in the remote end? |
18:50.47 | rene- | better |
18:50.51 | rene- | use qualify |
18:50.59 | rene- | qualify=yes |
18:51.02 | rene- | nat=yes |
18:51.31 | talljon84 | on the SIP entry or the general sectioN? |
18:51.38 | rene- | on the sip entry |
18:52.11 | talljon84 | ok. lemme try |
18:52.11 | rene- | of the remote peer |
18:53.02 | talljon84 | ok, now it will connect (yay!) and I can transmit, but the remote phone can't send audio back (like it's on mute) |
18:53.24 | rene- | they cant talk back to you |
18:53.30 | talljon84 | right |
18:53.52 | rene- | what device? |
18:54.12 | Qwell | is the * server behind NAT also? |
18:54.14 | talljon84 | the remote end is a cellphone on the PSTN.. the local device is a Cisco 7960 runing SIP firmware |
18:54.20 | Qwell | of course it is... |
18:54.26 | Qwell | You need externip and localnet |
18:55.01 | talljon84 | Qwell: my IP here is dynamic so I can't set externip |
18:55.10 | Qwell | then use externhost |
18:55.13 | rene- | get a dyndns account |
18:58.16 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
18:59.31 | talljon84 | Qwell: I set externip to see if it would resolve the problem and it does. Thanks a ton for that. I know this is outside of the scope of this group but, are you aware of anyway to configure a remote BIND server to accept an update for it? Example: host.domain.com wants to update it's static IP address as itgets it to ns1.domain.com which is running BIND9. Is this even possible? |
18:59.54 | talljon84 | *it's new dynamic IP address |
19:01.35 | Qwell | I think that'd part of the point of rndc |
19:02.21 | rene- | talljon84: use externrefresh and externhost with a dyndns acct, supported in asterisk 1.2+ |
19:03.12 | talljon84 | rene- Ok, thanks. |
19:03.42 | talljon84 | One last Q: Any preference for a compressed codec? g723, g726 or g729? |
19:04.42 | rene- | g729 is your friend if bandwidth is limited but you have to purchase licenses for your * |
19:05.48 | talljon84 | rene-: I'm not as concerned about bandwidth usage in this deployment. thoughts on g723 or g726? |
19:06.00 | rene- | havent really used them |
19:06.26 | rene- | if you are on lan use g711, best audio quality |
19:06.39 | rene- | g729 is not as crisp but well is 8:1 compresion |
19:06.44 | Qwell | g723 isn't supported in asterisk |
19:07.18 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
19:07.34 | talljon84 | I'm using g711 for the LAN, but I'd like to compress it a little for traversal across the net. |
19:07.43 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
19:07.51 | Qwell | so use g729 |
19:08.22 | CMike | hm.. anybody if I easily can save the useragent for a client in realtime ? |
19:09.09 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
19:10.51 | *** join/#asterisk Dovid (n=Dovid@85-250-190-83.bb.netvision.net.il) |
19:15.03 | rene- | Qwell: the patch applied almost cleanly but it still isnt working, i have already loading agents from database via include files so i guess i will wait for 1.4 for that |
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19:16.04 | Dovid | Happy easter everyon e |
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19:42.41 | mitcheloc | ditto Dovid |
19:42.48 | Dovid | hehe |
19:42.52 | Dovid | this room is busy |
19:43.02 | Dovid | i am jewish so i got nothin to celebrate |
19:43.08 | mitcheloc | everyone is getting stuffed or preparing to ;) |
19:43.44 | mitcheloc | i think even non-religious people take the opportunity to spend time with family, so it's not necessarily only for religious people |
19:44.35 | Dovid | guess so |
19:44.41 | Dovid | i am here. tryin to get some work done |
19:44.51 | *** join/#asterisk DeV-rAd (n=jesse@fl-69-69-130-197.sta.sprint-hsd.net) |
19:45.16 | Dovid | lol |
19:45.23 | Dovid | yes tzafir it is Pesach |
19:45.44 | mitcheloc | english please =P |
19:45.57 | Dovid | Pesach is hebrew for Passover |
19:46.14 | mitcheloc | ahh |
19:46.20 | file[laptop] | it's rabbit day |
19:46.26 | file[laptop] | KILL HIM! |
19:46.28 | *** join/#asterisk delacko (n=delacko@lns01-1076.dsl.iskon.hr) |
19:46.37 | Dovid | lol |
19:46.54 | DeV-rAd | Need Some help with asterisk@home get the error 404 on out dial |
19:47.29 | mtgh | All asterisk people, I need some C help with a non asterisk thing, but it should be quick, is there anyone who can help me? |
19:47.58 | mitcheloc | DeV-rAd: from the channel topic - "asterisk@home users should join #freepbx for support" |
19:48.13 | DeV-rAd | ok thank you |
19:48.21 | mitcheloc | np |
19:49.24 | tzafrir_laptop | mtgh, tried #c or something similar? |
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19:49.51 | *** part/#asterisk DeV-rAd (n=jesse@fl-69-69-130-197.sta.sprint-hsd.net) |
19:56.48 | talljon84 | Does anyone know of a way to get MOH (using madplay) to get streaming audio as a source (such as shoutcast)? |
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20:04.23 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
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20:08.33 | dlynes | talljon84: I'm using g726 on some installs...it's pretty good |
20:08.47 | dlynes | talljon84: afaik, asterisk still only supports g726-32, however |
20:22.30 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
20:32.28 | blitzrage | Qwell: I am here! |
20:32.36 | file[laptop] | blitzrage: OH NOES |
20:32.44 | blitzrage | :-O |
20:33.30 | file[laptop] | blitzrage: LJAM |
20:34.01 | *** join/#asterisk redcap1 (n=phez@redcap.xs4all.nl) |
20:35.06 | blitzrage | oh no you didn't |
20:35.24 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
20:39.48 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
20:47.54 | *** join/#asterisk Sedorox (n=penbra67@smartserv/cna/Sedorox) |
21:23.42 | Dovid | so am i |
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21:26.41 | robin_sz | poxy new software ... worse than before |
21:26.45 | Sedorox | 8 |
21:26.47 | Sedorox | er |
21:27.08 | Sedorox | well I'll bbl |
21:28.47 | robin_sz | http://www.atcom.cn/En_products_At320EE.html <=== are these any good? |
21:30.24 | tecnico | anyone knows what module provides "ast_park_call" ?? when trying to load chan_sip, I get "undefined symbol: ast_park_call" |
21:30.30 | file[laptop] | res_features |
21:30.35 | tecnico | tnx. file |
21:30.45 | Dovid | it looks cheap |
21:30.55 | robin_sz | true .. it does |
21:31.00 | robin_sz | but it is cheap ... |
21:31.09 | robin_sz | the question is: does it work OK :) |
21:31.27 | robin_sz | voip phones are WAY overpriced |
21:31.40 | robin_sz | they cost no more than POTS phones to make |
21:31.45 | *** join/#asterisk mwright1night (n=mwright1@203-214-48-213.dyn.iinet.net.au) |
21:31.53 | robin_sz | and sell for 30 times the rate ... things have to change |
21:32.54 | robin_sz | well, 10 times the rate then |
21:33.47 | Dovid | eh |
21:33.53 | Dovid | i like the spa-841 |
21:34.16 | robin_sz | price? |
21:34.42 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
21:34.53 | Dovid | i think 85 |
21:35.20 | Dovid | voipsupply.com |
21:35.25 | Dovid | or ebay |
21:35.45 | DoktorGreg | actually, if you compair voip phones to key system phones, their prices are quite favoriable |
21:35.52 | *** join/#asterisk apardo (n=apardo@87.218.45.213) |
21:36.07 | file[laptop] | VoIP phones have to be smarter then you think, and have more then a regular ol' POTS phone |
21:36.12 | DoktorGreg | or even a nice analog wireless phone... |
21:36.30 | DoktorGreg | can easily pay 150 for a nice analog wireless phone |
21:36.41 | tecnico | I have the spa-841... I wouldn't recommend it.. |
21:37.05 | Dovid | i like it for a basic phone |
21:37.11 | Dovid | polycoms r the best |
21:37.11 | tecnico | speakerphone is terrible.. |
21:37.24 | Dovid | yes, but for a basic phone its good |
21:39.21 | DoktorGreg | man last night i got stuck using a 4x cd burner |
21:39.22 | *** join/#asterisk CukX (n=cuk@nu.cuk.nu) |
21:39.40 | DoktorGreg | this 32x cd burner i have at home is much nicer |
21:39.43 | CukX | can anyone enlighten me, please ? |
21:39.50 | CukX | I have ISDN HFC-S card and wich drivers to use with it ? |
21:40.56 | CukX | no, really, I don't have a clue on all that... mISDN, isdn4linux, zaptel, .... |
21:42.44 | tecnico | any hints on what's causing this? : "chan_sip.c:9633 handle_response_register: Got 200 OK on REGISTER that isn't a register" |
21:44.22 | robin_sz | file[laptop]: no, you misunderstand .. the phoen might have to be "smarter" ... BUT that doesnt mean it costs more in bulk ... infact, the ethernet if is probably cheaper to if than a POTS interface |
21:44.42 | blitzrage | <PROTECTED> |
21:44.43 | robin_sz | the smartness is in the code ... code it once and you;re done |
21:46.01 | mwright1night | robin_sz: I think at the cisco buying millions of them level, they both cost a few cents |
21:46.11 | robin_sz | right |
21:46.35 | robin_sz | but a cisco 7940g still sells for like $350 |
21:46.44 | Dovid | get a polycom |
21:47.01 | mwright1night | chinese (near) slave labour gives us lots of cheap plastic things... uh oh, we need oil for plastic stuff, and china needs oil for it.. but but but |
21:47.01 | robin_sz | they should come down to the <$100level in the not too far off |
21:47.14 | Dovid | hehe |
21:47.14 | mwright1night | we want the oil for us, but we want to buy the plastic chinese things |
21:47.16 | Dovid | u wish |
21:47.17 | mwright1night | what do we do |
21:47.18 | Dovid | u get what u pay for |
21:47.41 | mwright1night | The cisco's will come down to that pricing point you think? |
21:47.50 | robin_sz | sure ... well |
21:47.54 | DoktorGreg | the phones with the big color displays??? |
21:47.55 | Dovid | nope |
21:47.56 | robin_sz | not as cisco |
21:48.11 | DoktorGreg | the discount cicsos will be the sipuras |
21:48.15 | robin_sz | but when Walmart have them rolling off the shelves |
21:48.31 | Dovid | i wish |
21:48.35 | Dovid | that will be the day |
21:48.37 | robin_sz | and they are in 50% of broadband-equipped homes ... |
21:48.47 | robin_sz | I say, thats .. what .. 18 months away |
21:49.00 | DoktorGreg | in 5 years the POTS service providers will be closing up ship |
21:49.03 | DoktorGreg | shop |
21:49.07 | robin_sz | agreed |
21:49.26 | DoktorGreg | and you will be able to buy a no frills voip phone for $30 |
21:49.27 | robin_sz | already BT who had been fighting voip takeup by businesses |
21:49.39 | robin_sz | are now offereign it as an add on on business broadband |
21:49.52 | robin_sz | thats a sure sign its gone mainstream |
21:50.06 | mwright1night | someone has to do complicated switching |
21:50.11 | Dovid | i wish |
21:50.18 | DoktorGreg | comcast is about ready to go zero $ install on their voip service |
21:50.19 | mwright1night | directory for all the number allocations |
21:50.22 | robin_sz | and now it will happen very fast ... because the under pinnin technolgy (adsl) is already widely available and installed |
21:50.36 | mwright1night | someone has to maintain the last mile copper |
21:50.41 | robin_sz | yeap |
21:50.45 | Dovid | some people will still not get the internet |
21:50.49 | mwright1night | in Australia where I'm from this is very expensive |
21:50.49 | robin_sz | thats broadband providers ... |
21:51.18 | mwright1night | our 49% state owned Telecommunications provider owns it |
21:51.29 | robin_sz | it will just "happen" .. you'll pay your $30 for the line and broadband |
21:51.31 | DoktorGreg | I think the cable IP backbone makes more sense than the pots telco backbone |
21:51.38 | robin_sz | and another $10 for your mothly calls |
21:51.45 | Dovid | only time will tell |
21:51.57 | DoktorGreg | with phone you need pair of wires between you and co |
21:51.59 | robin_sz | i think it wont be long now ... its already happening |
21:52.00 | *** join/#asterisk Lino` (n=Lino@i577BDF8D.versanet.de) |
21:52.14 | robin_sz | anyway ... |
21:52.18 | mwright1night | I pay 89.95 + 33 to my broadband provider atm and get 1x POTS line 1x VOIP line and 1x 40GB 8AM --> 12Midnight, 1x 40GB 12midnight --> 8AM |
21:52.24 | DoktorGreg | with cable you need a piece of coax between you and drop... |
21:52.48 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
21:52.59 | mwright1night | exchange rate is .70 |
21:53.02 | robin_sz | in the UK, it generally about £24/month for 512K adsl, no limit |
21:53.08 | mwright1night | so that's a hell of a lot more expensive |
21:53.25 | mitcheloc | what is a "1x 50GB 8AM"?? |
21:53.27 | mwright1night | well this is 19000/1024 |
21:53.30 | mitcheloc | *40 |
21:53.42 | mwright1night | 40GB quota per month |
21:53.45 | DoktorGreg | Im on comcast business about $160 to get 1.5Mb up and 8Mb down |
21:53.48 | mwright1night | from Midnight to 8AM |
21:53.51 | mitcheloc | oh, ouch, where are you from? |
21:53.53 | mwright1night | and 40 GB for peak |
21:54.10 | mitcheloc | i use that in a week =P |
21:54.22 | mwright1night | Australia, we have the worlds most sparsely populated contenent |
21:54.25 | robin_sz | sigh . if nly I hadnt "upgraded" this GXP200 I could seel the fscker on ebay :( |
21:54.29 | mwright1night | so do I |
21:54.43 | robin_sz | Australia? |
21:54.43 | mwright1night | that's why I'm about to be throttled in 6mins to 64k |
21:54.52 | mwright1night | yep |
21:54.57 | robin_sz | Ive always wanted to go to Australia ... |
21:55.14 | robin_sz | but I don't have a criminal record :( |
21:55.15 | mwright1night | expensive broadband.. don't come (hehe) |
21:55.32 | mwright1night | well our immigration dept won't let you in if you don't have a criminal record |
21:55.35 | mwright1night | I'm a kiwi |
21:55.36 | Dovid | lol |
21:55.39 | Dovid | u need one to get in ? |
21:55.43 | mwright1night | I live in aus |
21:55.53 | mwright1night | 5 mins to Shaping (throttling) |
21:56.00 | robin_sz | Dovid: traditionally, from the UK yes ... |
21:56.00 | DoktorGreg | do you guys have on demand cable down under yet? |
21:56.05 | Dovid | lol |
21:56.08 | Dovid | good to know |
21:56.15 | robin_sz | Dovid: stealing a sheep is the most popular route I think |
21:56.20 | Dovid | lol |
21:56.29 | Dovid | so if i am caught wit something i jus tmove there ? |
21:56.44 | mitcheloc | stop giving me more reasons to stay in california! |
21:56.46 | mwright1night | I think stealing a loaf of bread |
21:56.52 | mwright1night | or not having a home to go to |
21:56.57 | Dovid | LA ROCKS |
21:57.00 | Dovid | i am thinkin of movin there |
21:57.06 | mwright1night | doktorgreg: what do you mean by ondemand cable? |
21:57.10 | robin_sz | mwright1night: heh, indeed :) |
21:57.30 | mitcheloc | lol, i'll trade Dovid ;) |
21:57.43 | Dovid | hehe |
21:57.46 | Dovid | to where i am ? |
21:57.57 | mitcheloc | mmm...good point, maybe not |
21:58.02 | Dovid | lol |
21:58.05 | Dovid | its nice here |
21:58.36 | Dovid | not too hard to get a gun |
21:58.37 | Dovid | lol |
21:58.41 | Dovid | and maybe kill some one |
21:58.43 | robin_sz | mwright1night: at one point, they offered £400 cash and a free boat ticket ... that was in the 30s I think |
21:58.44 | Dovid | :) |
21:58.51 | Dovid | lol |
21:58.58 | Dovid | damn. to get rid of the criminals ?> |
21:59.13 | mitcheloc | *creepy* |
21:59.14 | robin_sz | no, criminals just got sent anyway |
21:59.21 | mwright1night | robin_sz |
21:59.24 | mitcheloc | heh i just remembered i need to do my taxes! |
21:59.28 | Dovid | so do i |
21:59.32 | mwright1night | yep that is how all the Italians came |
21:59.36 | robin_sz | heh |
21:59.36 | mwright1night | and Greeks, and slavs |
21:59.37 | Dovid | thanks for reminding me |
21:59.41 | mwright1night | they did really well |
21:59.43 | mitcheloc | your welcome |
22:00.04 | Dovid | calling accountant now |
22:00.06 | Dovid | ... |
22:00.08 | DoktorGreg | wow i havent looked at a proper linux distro in a while debial is really really nice |
22:00.09 | mwright1night | 1 minutes until capping |
22:00.10 | mwright1night | doh |
22:00.27 | mitcheloc | quick send mwright pr0n in non-compressed format! |
22:00.29 | mwright1night | doktorgreg: check out ubuntu dapper drake, due to be released june 1 |
22:00.50 | robin_sz | DoktorGreg: ignore them ... stick with debian. the one true way |
22:00.59 | mwright1night | I have paused my torrents waiting for the last 20mb of my asterisk@home 2.8 to come in |
22:01.16 | DoktorGreg | whats not to like about debian? |
22:01.17 | Dovid | nah |
22:01.18 | mwright1night | 2:53 to go @ 246KB/sec |
22:01.20 | Dovid | CENT OS RULES |
22:01.21 | Dovid | lol |
22:01.32 | mwright1night | 30MB to cover |
22:01.34 | robin_sz | DoktorGreg: nothing .. its great. |
22:01.36 | mwright1night | in 0 secs |
22:01.46 | Dovid | i get 7.0 KB where i colo |
22:01.52 | DoktorGreg | it found all my hardware?!! |
22:01.58 | robin_sz | exactly |
22:02.02 | DoktorGreg | how can I complain about that? |
22:02.03 | mwright1night | and you're in a big broadband country |
22:02.05 | mitcheloc | I get 600+KB/sec where I colo ;) |
22:02.13 | DoktorGreg | omg i just converted distros |
22:02.14 | Dovid | oops |
22:02.15 | robin_sz | and new stuff is just an "apt-get install" away |
22:02.17 | Dovid | not KB |
22:02.18 | mitcheloc | i can get a full linux distro in less then 12 minutes |
22:02.18 | Dovid | MB |
22:02.20 | DoktorGreg | I was slackware forever type |
22:02.31 | mwright1night | it musn't be 8AM yet, my isp hasn't throttled me, 1min 18 to go |
22:02.49 | Dovid | i recently converted |
22:02.50 | mwright1night | I am a Fedora / RHEL person, however ubuntu rocks |
22:02.54 | Dovid | i am now CentOS |
22:03.02 | robin_sz | Iused to be RH, |
22:03.05 | DoktorGreg | but slackware is too old school now days |
22:03.08 | robin_sz | but debian is MUCH better |
22:03.15 | mitcheloc | the fedora people are awesome for including mono in fc5 ;) |
22:03.18 | Dovid | what do u like with debian ? |
22:03.25 | mwright1night | I think after I Dapper Drake comes out, I will switch my www.ltsp.org / FreeNX based terminal server to Ubuntu Dapper Drake |
22:03.27 | Dovid | i dont know y anyone uses fc |
22:03.33 | Dovid | if u have cent os |
22:03.34 | mitcheloc | i do |
22:03.36 | robin_sz | fedora tries hard, but has WAY too short a lifecycle for servers |
22:03.43 | mwright1night | why is mono good |
22:03.46 | Dovid | thats y i use cent os |
22:03.46 | mwright1night | it is a slow pig |
22:03.47 | DoktorGreg | so far ive looked at asterisk on debian in rapid asterisk distro |
22:04.05 | mitcheloc | i prefer to program on it, and there is nothing wrong with that! |
22:04.08 | DoktorGreg | used apt-get to get some thigns |
22:04.20 | robin_sz | apt-get roxxors |
22:04.20 | mwright1night | I like ubuntu for desktop |
22:04.21 | DoktorGreg | was like, "OMG PONIES!!!" |
22:04.31 | robin_sz | yeah |
22:04.39 | robin_sz | pony drop! |
22:04.40 | mwright1night | ok m ISO came through |
22:04.44 | mwright1night | and I still don't feel shaped yet |
22:04.55 | mitcheloc | pink slashdot hahahaa |
22:05.00 | DoktorGreg | was having some problems getting the asterisk 1.2 branch compile |
22:05.16 | DoktorGreg | so decided to start from a clean debian install and dump the rapid thing |
22:05.50 | DoktorGreg | am experiencing apt-get based install right now |
22:05.59 | DoktorGreg | and so far I am like |
22:06.04 | DoktorGreg | OMG PONIES! |
22:06.47 | DoktorGreg | OMG Ponies, fastest meme ever |
22:07.21 | Dovid | ponies ? |
22:07.32 | DoktorGreg | april first edition of slashdot this year |
22:07.49 | DoktorGreg | go look at cuitest website ever article |
22:07.53 | mwright1night | I have been a computer junkie this weekend and got a bad back |
22:08.01 | mwright1night | do you wall have really nice workstations with high quality chair? |
22:08.32 | mitcheloc | i had a decent chair but i broke it =/ |
22:10.35 | robin_sz | ponies was something a long long time ago on #london.pm |
22:12.01 | Dovid | can u elaborate ? |
22:13.15 | DoktorGreg | when you buy an office chair now days spend the extra money and make sure you get one with a steel truck |
22:13.36 | DoktorGreg | the chairs at the office super store almost never have steel trucks |
22:13.51 | DoktorGreg | you have to go to office furniture store |
22:14.01 | *** join/#asterisk Greek-B0y (n=fusion@193.220.93.162) |
22:15.27 | *** join/#asterisk lacym (n=lacym@h-68-164-20-5.hstqtx02.covad.net) |
22:18.50 | *** join/#asterisk AsteriskAlbania (i=Asterisk@80.91.113.253) |
22:19.21 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
22:19.26 | Qwell | Albania? |
22:19.34 | AsteriskAlbania | yes |
22:20.44 | robin_sz | http://mightymake.en.alibaba.com/product/50108544/50491128/VoIP_Phones/VoIP_SIP_IAX2_H_323_MGCP_Phone.html |
22:20.51 | robin_sz | ^^ interesing price ... |
22:21.13 | robin_sz | less than £30 each on ebay, new |
22:21.53 | Greek-B0y | anyone played with wi-fi phones? |
22:22.02 | Qwell | I see no price on that site |
22:22.21 | *** join/#asterisk heka (n=heka@82.114.68.124) |
22:24.01 | talljon84 | All of a sudden, my * is considering every extension to be busy. Extensions with VM are sent directly there and those without are shown as busy. Help! Any ideas why this would be happening? |
22:24.26 | robin_sz | Greek-B0y: yeah, I have a few Zyxel wifi phones |
22:24.40 | robin_sz | Qwell: search on ebay.co.uk |
22:24.45 | file | talljon84: look on the asterisk console and see what it says... |
22:24.54 | robin_sz | Qwell: sorry less than £40 /// 39.99 |
22:25.17 | Dovid | tall is ur internet down ? |
22:25.22 | Dovid | can u make calls on the lan ? |
22:25.35 | [av]bani | robin_sz: how are they? |
22:25.45 | robin_sz | [av]bani: crap! |
22:25.46 | talljon84 | i can access thing such as VM; however, any call to an extension gets busy. |
22:25.47 | Greek-B0y | robin_sz, what do u think of the zyxel ones? any good? i've read negative reviews about them |
22:25.57 | robin_sz | Greek-B0y: they are crap |
22:26.52 | robin_sz | short battery life, crap networking (more like not-working!), crap menus/gui, only 10% chance of it sitting in the charger and charging etc etc etc etc |
22:27.17 | robin_sz | inshort, its a bit like a portable GXP2000 :) |
22:27.46 | DoktorGreg | what POE switch do yall recommend? |
22:28.11 | Qwell | DoktorGreg: anything, as long as it also has poe on the wifi |
22:28.17 | [TK]D-Fender | DoktorGreg : D-Link DES-1526 |
22:28.22 | mitcheloc | oooh, that would be cool |
22:28.24 | robin_sz | POE-knee! |
22:28.53 | talljon84 | I'm using aoh which is using an agi script to make dialing. It seems to be exiting the script with "no extensions to dial" and sends it to vm.. Nothing has been changed with the AGI script though so I don't understand why it suddenly stopped working. |
22:29.15 | [TK]D-Fender | talljon84 : Please read the channel topic.... |
22:29.16 | Qwell | talljon84: type /topic |
22:29.49 | Greek-B0y | robin_sz: i was hoping to get my hands on something workable. I guess I'll have to go for the more expensive linksys wi-fi phone |
22:30.03 | Qwell | Greek-B0y: those seem real nice |
22:30.17 | Greek-B0y | yeah |
22:30.21 | Qwell | physically anyhow |
22:30.26 | robin_sz | Greek-B0y: or a DECT phone plugged into a iaxy adapter! |
22:30.39 | [TK]D-Fender | Greek-B0y : Check your reviews first. If its for a fixed SITE, then I'd suggest an ATA+Cordless phone still. |
22:30.52 | Greek-B0y | maybe i should just wait until they release a cellphone with built-in wifi and sip client |
22:31.04 | [TK]D-Fender | Greek-B0y : They're out there already... |
22:31.08 | robin_sz | theres a dutch thing, that does 10 dect phones to SIP or H323 |
22:31.18 | robin_sz | a sorta DECt base for sip |
22:31.24 | MoutaPT | Asterisk in a DMZ, Xlite registers is just fine and the calls are ok, but when i click to hangup the call it takes about 3 or 5 seconds to get the call hanged in the Xlite.. I've checked ASterisk CLI and the call is hanged correctly! any tip?how should i debug this? |
22:31.45 | mitcheloc | *hung |
22:31.53 | robin_sz | I have one somewhere .... I failed to get my DECT phoens to register with it and lost interest .. its under the desk or behind the bin or something |
22:32.34 | robin_sz | other people have had good success with them .. if only I could remeber what it was called |
22:32.53 | Greek-B0y | so i take it iax2 is still the recommended protocol even for hard phones |
22:33.00 | *** join/#asterisk fugitivo (n=fugitivo@201.255.177.88) |
22:33.08 | Qwell | Greek-B0y: Only if the hardphone runs asterisk |
22:33.13 | *** join/#asterisk oej (n=oej@tcn003124.tcn-catv.ne.jp) |
22:33.14 | Qwell | ~iax |
22:33.15 | jbot | hmm... iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
22:33.26 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
22:33.31 | mitcheloc | MoutaPT: i suggest buying eyebeam or get another softphone, lots of problems are fixed in their paid version, counterpath secretly leaves us in the dark on that |
22:34.11 | Qwell | mitcheloc: They're quite clear that bugs aren't often fixed in xlite :p |
22:34.16 | MoutaPT | i've this at home working fine, but at home * is running in my lanno |
22:34.21 | MoutaPT | Lan no dmz |
22:34.32 | Greek-B0y | and what codec is best for lan? ulaw? |
22:34.42 | MoutaPT | tomorrow i'm trying a MITEL hardphone |
22:34.53 | MoutaPT | could this be better with hardphone? i hope... |
22:34.57 | mitcheloc | yea, well it sucks, i was trying out xlite and the audio quality sucked, switched in eyebeam and it worked like a charm....i think it's a conspiracy! |
22:35.29 | Qwell | mitcheloc: glad to see I'm not the only paranoid one |
22:37.22 | MoutaPT | i think i will try one eyebeam to be sure that is not my mistake... i'm getting paranoid too |
22:38.43 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-69-235.telkomadsl.co.za) |
22:46.10 | CukX | wich driver to use for HFC-S cards, please... |
22:46.48 | puzzled | CukX: bristuff patch, mISDN or vISDN |
22:47.48 | puzzled | CukX: and if you decide to use bristuff I read on the mailing list that adding the florz patch improves it |
22:49.45 | *** part/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
22:49.51 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
22:50.31 | CukX | puzzled are there any manuals on that ? |
22:50.55 | puzzled | CukX: no idea, don't use them |
22:51.08 | CukX | i am from freebsd world, but anyway... i have installed debian 3.1 with 2.4.27 kernel... |
22:51.18 | CukX | puzzled should i have 2.6 for mISDN ? |
22:51.56 | puzzled | CukX: I would prefer 2.6 over 2.4 in general. no idea if you need 2.6 for either bristuff, mISDN or vISDN |
22:52.50 | CukX | i didn't found any relative cheap hw GW's for ISDN |
22:53.10 | dlynes | I take it bri is still quite cheap in Europe? |
22:53.21 | dlynes | I don't know of anyone using bri in na |
22:53.30 | puzzled | dlynes: depends on what you call cheap. I think it's still a ripoff |
22:53.42 | CukX | oh, and... one of our ISP offer VoIP, SIP based... can I "make" asterisk, that send traffic to their SIP server and GW ? |
22:53.56 | dlynes | CukX: yes, and h323 |
22:54.21 | puzzled | CukX: search eBay for an Eicon Diva *Server* card and get chan_capi. I have had the best experience with that so far |
22:54.26 | dlynes | puzzled: ah...so why do people use it instead of dsl? |
22:54.34 | CukX | and have fancy SCCP cisco phone, hihi :) |
22:55.12 | puzzled | dlynes: I don't know anyone who uses ISDN instead of DSL. I know people that have downgrade their ISDN line to analog and get DSL on it |
22:55.42 | dlynes | puzzled: so why is isdn still used then? |
22:55.57 | dlynes | puzzled: is it cheaper than dsl in europe? |
22:56.02 | puzzled | dlynes: it's big in businesses |
22:56.23 | dlynes | puzzled: yeah...not here...everyone uses dsl or cable |
22:56.25 | CukX | |
22:56.25 | CukX | |
22:56.25 | CukX | Eicon DIVA Server BRI-2M PCI aktive ISDN Karte |
22:56.46 | CukX | and better than hack with passive HFC-S cards ? and had troubles with echo cancelation, etc, etc ? |
22:57.05 | puzzled | CukX: yup that's the one. You can also search for AVM Fritz! card to use with chan_capi but I no experience with that |
22:57.42 | dlynes | btw...anyone know how the hdlc code in the oej branch will help with pri's? |
22:57.53 | puzzled | dlynes: it's the smart thing to do money wise. prolly cable even more now they added phone services to it over here and you no longer need a pots line with the added cost |
22:57.59 | CukX | dlynes you don't wanna know... our Telco has bought ISDN, 10 years ago, and they offered DSL only over ISDN... because they wanted to cover expenses in ISDN |
22:58.36 | puzzled | CukX: typical traditional old telco monopoly |
22:58.51 | CukX | puzzled yep |
22:59.07 | CukX | and now... they offer "change" to VoIP phones.. |
22:59.32 | CukX | and they take all ISDN stuff away and put one Sagem with VoIP and analog POTS |
22:59.36 | puzzled | dlynes: I would think that any hdlc stuff lives in libpri but am not sure |
23:00.00 | puzzled | CukX: so then you are stuck with your ISDN phones and have to buy new analog ones :) |
23:00.02 | dlynes | puzzled: but how is hdlc related to pri? i'm not sure what it is |
23:00.10 | CukX | puzzled sort of... |
23:00.11 | puzzled | dlynes: low level data framing |
23:00.28 | dlynes | puzzled: I just know with libpri-trunk and zaptel-trunk that i'm getting messages on my screen about hdlc frames getting dropped now |
23:00.48 | CukX | puzzled or clean dust from old Panasonic phones, wich we didn't throw away |
23:01.06 | puzzled | dlynes: testing of trunk is good but if you need it for production I would have a look at the latest 1.2 branch |
23:01.16 | puzzled | CukX: good for you :) |
23:01.37 | CukX | puzzled so you recomend Diva over HFC-S cards ? |
23:01.40 | dlynes | puzzled: that's just it...latest 1.2 branch wasn't stable for me...trunk is |
23:02.03 | dlynes | puzzled: last release that was stable was libpri for zaptel 1.0.9.2 |
23:02.05 | CukX | puzzled what about a FXO interface ? Sipura ? |
23:02.21 | puzzled | CukX: yeah cause they have echo cancellation on board. If you can find the latest/newer 2.0 revision on eBay than it has even more goodies on board |
23:02.52 | CukX | puzzled how do I know, that it's ver 2 ? |
23:03.07 | puzzled | dlynes: well if you had issues/bugs I can only recommend to file them on bugs.digium.com |
23:03.08 | dlynes | puzzled: libpri 1.2.2 and zaptel 1.2.5 randomly hang my pri on me...sometimes as often as once a day, other times as often as once a week |
23:03.26 | *** part/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-28-100.houston.res.rr.com) |
23:03.40 | puzzled | CukX: the sipura spa-xxxx series are quite popluar. I have a couple here but there are still sitting in their box |
23:03.45 | dlynes | the hanging issue isn't happening with zapte-trunk and libpri-trunk |
23:04.16 | puzzled | CukX: version 2 has a different product id. Search the eicon website for it. Iirc it is noted on the datasheet |
23:04.36 | puzzled | CukX: if they don't mention it on eBay than it's prolly a version 1.0 which works fine for me |
23:04.54 | puzzled | dlynes: hanging pri sucks. did you get core dumps or anything else? |
23:05.48 | dlynes | puzzled: nope...asterisk was still running...just giving everyone a busy signal that tried calling into the pri |
23:06.08 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
23:06.09 | CukX | puzzled can I do all that Asterisk on dual 333Mhz PII ? max 4 concurent connections ? |
23:06.09 | *** join/#asterisk hodrige (n=Hodrige@ip68-98-172-123.dc.dc.cox.net) |
23:06.21 | puzzled | dlynes: even if you get zaptel/libpri from trunk going reliably be aware that asterisk trunk is in serious flux at the moment |
23:06.40 | dlynes | puzzled: i'm not updating it every day |
23:06.42 | hodrige | Hi |
23:06.54 | dlynes | puzzled: i just downloaded it once, it's stable, and i'm not updating it again |
23:07.00 | puzzled | dlynes: did you try upping the filedescriptor count etc. the ulimit stuff? |
23:07.10 | dlynes | puzzled: my policy is, if it ain't broke, don't fix it |
23:07.13 | hodrige | anyone was able to setup DISA on AAH 2.8 |
23:07.36 | dlynes | puzzled: you mean ulimit -c? |
23:07.46 | puzzled | CukX: I have a PII-350 that can do 2 ISDN channels simultaneously without a problem. Just don't do any transcoding |
23:07.52 | dlynes | puzzled: I always have it set at unlimited |
23:08.01 | puzzled | dlynes: ah ok, that's good |
23:08.20 | dlynes | and filedescriptor count is an msdos/windows thing |
23:08.28 | dlynes | and even then, it's a throwback to cp/m-80 |
23:08.42 | puzzled | dlynes: ulimit -n is fd count iirc |
23:09.04 | dlynes | 1024 |
23:09.20 | puzzled | lemme check waht I usually throw it to |
23:09.27 | CukX | puzzled do you remember, your Diva card is low-profile, or full sire ? |
23:09.29 | CukX | size |
23:09.38 | puzzled | full height |
23:09.47 | CukX | so probably 2.0 is low... |
23:09.48 | puzzled | they all are iirc |
23:09.54 | puzzled | the 1.0 ones |
23:10.13 | dlynes | nice to see ulimit doesn't have a manpage :) |
23:10.24 | AsteriskAlbania | any one has experience TE110P with QUINTUM ? |
23:10.50 | *** join/#asterisk hinckc (n=hinckc@c-68-45-24-192.hsd1.nj.comcast.net) |
23:11.34 | puzzled | dlynes: I use ulimit -n 8192 |
23:12.13 | dlynes | puzzled: what does that do, and why do i need it? |
23:12.19 | *** join/#asterisk mwright1nigh1 (n=mwright1@203-214-48-213.dyn.iinet.net.au) |
23:12.31 | mwright1nigh1 | what's the default username password for centos asterisk@home |
23:12.37 | mwright1nigh1 | I am at a login prompt now |
23:12.40 | puzzled | dlynes: if you open a lot of files e.g. on an IVR platform |
23:12.48 | bkw_ | mwright1nigh1, RUDE.. don't bother saying Hi |
23:12.53 | bkw_ | just bust right in and ask questions |
23:12.58 | dlynes | mwright1nigh1: check the topic |
23:13.01 | puzzled | mwright1nigh1: browse the manual on the A@H website |
23:13.04 | bkw_ | mwright1nigh1, I'm sure its like in the docs |
23:13.05 | dlynes | mwright1nigh1: try #freepbx |
23:13.18 | SplasPood | MWRIGHT STANDING BY! |
23:13.27 | SplasPood | He's poised at the prompt |
23:13.32 | SplasPood | hanging on your every word! |
23:13.35 | timscott | haha. |
23:13.57 | dlynes | puzzled: the pri machine is just acting as a softswitch, that plays the odd music on hold file and records the odd voicemail, but that's it |
23:13.58 | mwright1nigh1 | what is the diff between freepbx and @home |
23:14.02 | mitcheloc | lol mwright1nigh1 got jumped |
23:14.03 | dlynes | puzzled: no big usage of files on there |
23:14.27 | dlynes | mwright1nigh1: freepbx is the site that produces AMP...@home uses AMP |
23:14.35 | puzzled | dlynes: did you check for weird stuff like NMI error messages if it's a Dell or HP. Is the card on it's own interrupt |
23:14.52 | dlynes | puzzled: nah...nothing weird like that |
23:15.15 | dlynes | puzzled: The biggest problem with that damned machine |
23:15.25 | mwright1nigh1 | bummer root password password isn't working |
23:15.28 | mwright1nigh1 | handbook isn't much good |
23:15.31 | dlynes | puzzled: is that it's the only rackmount machine we've got that has that particular type of pci slot |
23:15.38 | *** join/#asterisk Araluccl0 (n=ciccio@adsl-ull-1-4.46-151.net24.it) |
23:15.44 | puzzled | dlynes: all I can say is shoot an email to the list but have it well documented, e.g. compile zaptel and libpri with all debug options and log the stuff |
23:15.49 | dlynes | puzzled: so i can't even take the pri card out of there and throw it into another machine |
23:16.14 | puzzled | dlynes: that doesn't make it any easier :/ |
23:16.39 | dlynes | puzzled: well, if i run into that problem with libpri-trunk and zaptel-trunk, i'll try that...but like i said...i'm not running into the problem with the version of trunk i'm using now |
23:17.00 | puzzled | dlynes: ok, hope that the asterisk version you got works too |
23:17.15 | dlynes | puzzled: using asterisk 1.2.5 on that machine, I think |
23:17.35 | dlynes | asterisk 1.2.6 |
23:17.47 | puzzled | dlynes: afaik you can't mix zaptel/libpri from trunk with asterisk from 1.2 branch |
23:17.57 | dlynes | I did, and it works |
23:17.58 | dlynes | :) |
23:18.00 | puzzled | maybe that's been "fixed" but it used to be like that a while back |
23:18.07 | puzzled | lucky you :) |
23:18.21 | file | depends how much we change things |
23:18.30 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
23:18.35 | dlynes | what i did was compile libpri-trunk, install that |
23:18.37 | file | or rather |
23:18.41 | file | mog_home! |
23:18.46 | dlynes | then compiled zaptel-trunk, installed that |
23:18.47 | mog_home | file! |
23:18.57 | dlynes | and hten compiled asterisk-1.2.6, compiled that, and installed it |
23:19.07 | dlynes | erm...fix the second compile :) |
23:19.11 | AsteriskAlbania | dlynes: do you have any idea if TE110P works with quintum CMS960 ? |
23:19.16 | puzzled | dlynes: wrong order. you need to remove all old header/libs. than compile/install zaptel then compile/install libpri |
23:19.19 | dlynes | AsteriskAlbania: no idea |
23:19.41 | dlynes | puzzled: i compiled into a slackware binary package |
23:19.49 | dlynes | puzzled: and installed the slackware binary package |
23:19.54 | dlynes | puzzled: i.e. using upgradepkg |
23:20.01 | dlynes | puzzled: it autoremoves everything |
23:20.17 | puzzled | dlynes: sure as long as you compile/install in the right order zaptel -> libpri -> asterisk |
23:20.49 | dlynes | puzzled: oops...when i said the order i used...that wasn't the order i used |
23:21.03 | dlynes | i actually did do zaptel->libpri->asterisk |
23:21.20 | dlynes | because obviously libpri depends on zaptel |
23:21.37 | puzzled | yup |
23:21.46 | mitcheloc | i thought libpri was first =X |
23:21.47 | file | libpri doesn't depend on zaptel... |
23:21.53 | dlynes | using 2.6.15.5 on there, too, with low latency optimizations |
23:22.33 | dlynes | file: well, i wasn't sure if it did or not, but compiling zaptel first and then libpri should work regardless of whether libpri depends on zaptel, or not :) |
23:22.49 | file | depends |
23:23.08 | puzzled | file: so the CFLAGS+=-I../zaptel are in the libpri Makefile for fun? :) |
23:23.15 | dlynes | didn't you say it wasn't dependant? :) |
23:24.37 | file | some of the utility programs use zaptel |
23:24.43 | file | but libpri itself, does not |
23:24.54 | puzzled | ok |
23:25.19 | *** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net) |
23:25.23 | file | chan_zap is what uses libpri |
23:25.45 | dlynes | correct |
23:26.10 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
23:26.22 | *** join/#asterisk AsteriskAlbania (i=Asterisk@80.91.113.253) |
23:29.45 | DoktorGreg | dammit! |
23:29.56 | DoktorGreg | i just found the problem i was having |
23:30.04 | DoktorGreg | save the trouble |
23:30.11 | DoktorGreg | if you want to do PRI |
23:30.27 | DoktorGreg | DO NOT USE BRIStuff branch |
23:30.40 | dlynes | i don't |
23:31.00 | DoktorGreg | I was sharing the solution to my problem |
23:31.15 | puzzled | DoktorGreg: if you had taken a look at what the patch touches you would have stayed well away from it :) |
23:31.36 | dlynes | i haven't used bristuff since 1.0.9 |
23:31.36 | DoktorGreg | I was using the rapid asterisk distro |
23:31.54 | dlynes | didn't even think there was a patch for it for 1.2 |
23:32.22 | DoktorGreg | was not patch, I assumed that i was using a stable build |
23:32.55 | DoktorGreg | Im gonna post this on every forum i can find |
23:33.02 | file | lol |
23:34.34 | Rawplayer | dont use it then |
23:34.45 | *** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM] |
23:35.24 | *** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM] |
23:35.30 | key2 | someone knows a lil bit about capi ? |
23:35.51 | puzzled | key2: a bit but only in relation to Eicon Diva Server cards |
23:36.22 | key2 | puzzled: it's with avmc4 |
23:36.44 | puzzled | key2: no idea but there is a page on voip-info.org dedicated to making it work with asterisk/capi |
23:36.55 | key2 | I can't take more than 2 lines at the time, if the 2 lines of the first port are busy, then it doesnt take the line from the second line |
23:37.27 | *** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM] |
23:38.18 | key2 | puzzled: something looks wrong here : http://pastebin.com/664015 |
23:40.57 | puzzled | key2: looks ok. maybe try to change [AVMC4.2] to [AVMC42] so loose the dot. alternatively have a look on www.chan-capi.org |
23:41.15 | key2 | puzzled: tryed that |
23:42.03 | puzzled | key2: yeah I see there not a lot there yet. Tried the chan-capi mailing list yet? |
23:43.22 | key2 | nop |
23:43.34 | *** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM] |
23:43.35 | key2 | it's just weird |
23:43.49 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
23:44.08 | key2 | puzzled: here is what it says: http://pastebin.com/664025 |
23:44.29 | key2 | == Everyone is busy/congested at this time (1:0/1/0)] |
23:45.28 | *** join/#asterisk IceManRISK (n=kart@200-181-208-180.mganm7001.dsl.brasiltelecom.net.br) |
23:46.12 | puzzled | key2: if you already have 2 incoming calls on that port than that does not surprise me |
23:46.46 | *** join/#asterisk MrCraig (n=Craig@bb-87-82-12-210.ukonline.co.uk) |
23:46.49 | MrCraig | hi |
23:46.53 | *** join/#asterisk IceManRISK (n=kart@200-181-208-180.mganm7001.dsl.brasiltelecom.net.br) |
23:46.59 | dlynes | hihi |
23:47.00 | key2 | puzzle: AVCM4 is 2 lines and AVCM42 is two other line |
23:47.03 | key2 | in the same group |
23:47.25 | key2 | puzzled: so basically, If I even use one line of AVCM4 and one of AVCM42, i can't dial from any of those anymore |
23:47.44 | MrCraig | I wants to be able to make or recieve calls using my fax modem - I've been told to try asterisk, so my first question is do I have the right software? |
23:47.48 | puzzled | key2: sorry, no idea. I would try the chan-capi mailing list |
23:49.27 | CukX | stupid starting question... i'm getting messages, registration from "11" <sip:11@ASTERISK_IP> failed for PHONE_IP |
23:49.49 | CukX | what to enter ? into extensions.conf something, right ? |
23:50.18 | puzzled | sip.conf |
23:52.58 | Lino` | hmmm |
23:53.35 | Lino` | key2: german isdn or what? |
23:55.28 | Lino` | ok |
23:55.31 | Lino` | most probably french ISDN |