| 00:01.40 | jaytee | yep |
| 00:01.40 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 00:01.43 | jaytee | ~sipnat |
| 00:01.43 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:10.54 | shmaltz | anyone here know of a callaccounting/ppbilling system for asterisk that actualy works? |
| 00:14.32 | seb- | help! just installed asterisk 1.4 from source but can't register my softphone to it |
| 00:15.02 | seb- | [TK]D-Fender: you around? |
| 00:15.27 | [TK]D-Fender | seb-: Yup... Dear God... still failing to reg. |
| 00:15.55 | seb- | [TK]D-Fender: i installed 1.6 from source but had to downgrade to 1.4 for appconference which only works w/ 1.4 |
| 00:16.08 | seb- | [TK]D-Fender: appconference is a MeetMe replacement that works in Xen |
| 00:16.16 | *** join/#asterisk danielqb1 (n=danielqb@200.118.167.17) |
| 00:16.45 | seb- | [TK]D-Fender: oh wait...looks like ekiga actually came to life |
| 00:16.51 | [TK]D-Fender | seb-: Mine failed |
| 00:17.03 | seb- | [TK]D-Fender: oh that was YOU! |
| 00:17.06 | [TK]D-Fender | :p |
| 00:17.17 | seb- | [TK]D-Fender: why the @#$#@ doesn't MY Ekiga reach my *? |
| 00:17.17 | [TK]D-Fender | seb-: Denied :-) |
| 00:17.40 | seb- | [TK]D-Fender: i can ssh to this server...i turned off FW..not sure why my client is blocked |
| 00:17.41 | [TK]D-Fender | seb-: Your firewall is fubar'd. Or your ISP. Or your Ekiga settings |
| 00:18.00 | seb- | *sigh* |
| 00:18.34 | Titanous | how do I prevent Dial from playing ringing to a caller? |
| 00:19.06 | [TK]D-Fender | Titanous: Use m() and set an empty MoH class |
| 00:19.39 | [TK]D-Fender | seb-: PM me your new use auth so I can test |
| 00:19.50 | [TK]D-Fender | user* |
| 00:22.17 | Globettrotter | Fender,, you got an answer for this?? |
| 00:22.18 | Globettrotter | response 482 "Loop Detected" back from 0.0.0.0 |
| 00:23.10 | [TK]D-Fender | Globettrotter: Stop talking to yourself. They have 1-size-fits-all-plus-200%-sleeves jackets for people like that. |
| 00:23.32 | Globettrotter | lol |
| 00:23.37 | Globettrotter | understood |
| 00:25.52 | *** part/#asterisk danielqb1 (n=danielqb@200.118.167.17) |
| 00:35.26 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 01:05.27 | *** join/#asterisk jdblack (n=jblack@pool-71-181-243-204.sctnpa.east.verizon.net) |
| 01:08.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:11.40 | VaGoNeTaS | is back from the dead. Gone: 1h 43m 8s |
| 01:12.35 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:13.57 | saxa | hi, short question, is it possible to use asterisk 1.6.1.0 with zaptel 1.4.12.1 ? |
| 01:14.12 | saxa | i have a tdm 410 |
| 01:14.49 | saxa | was trying to build dahdi on my 64bit system but never succeeded to load well the kernel modules |
| 01:15.03 | saxa | as it was always did a kernel panic |
| 01:15.11 | *** join/#asterisk tjz (n=tjz@bb219-75-22-243.singnet.com.sg) |
| 01:15.40 | saxa | so tried now with zaptel 1.4.12.1 and the modules loaded ok, and on my board i see the green lights lit up |
| 01:15.48 | saxa | which before never happened |
| 01:18.20 | infinity1 | saxa: ack. thats a bummer |
| 01:18.29 | infinity1 | i dunno if it works, try it |
| 01:18.56 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-04f0a8413e5b6be8) |
| 01:19.17 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 01:20.34 | saxa | infinity1: hmm, ok, will try it . But in any case would be good if the devs could fix up the kernel panic i get, probably i should do some bug report somewhere :) |
| 01:28.17 | [TK]D-Fender | saxa: No, 1.6+ uses DAHDI |
| 01:28.37 | [TK]D-Fender | saxa: http://www.asterisk.org/downloads |
| 01:28.48 | [TK]D-Fender | saxa: Do pay attention to the rather clear groupings |
| 01:28.57 | [TK]D-Fender | ~dahdi |
| 01:28.57 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 01:31.42 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 01:34.11 | *** join/#asterisk utahsaint_ (n=utahsain@cpe-72-190-16-177.tx.res.rr.com) |
| 01:35.44 | *** join/#asterisk jtodd (n=jtodd@253.sub-75-254-237.myvzw.com) |
| 01:35.44 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:35.47 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:36.05 | saxa | [TK]D-Fender: thx, but when trying dahdi 2.2.0-rc1 i was all the time getting kernel oops |
| 01:36.18 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 01:36.37 | saxa | [TK]D-Fender: so today i tried 1.4.12.1 zaptel and it worked out immediately |
| 01:37.01 | saxa | [TK]D-Fender: i have not tried the dahdi 2.2.0-rc2 yet |
| 01:37.25 | saxa | will try to compile it to see if i can get it to work |
| 01:37.38 | saxa | if not i will just stick with asterisk 1.4 |
| 01:38.29 | [TK]D-Fender | saxa: DAHDI is the renamed REPLACEMENT for Zaptel. Same thing, but is the version required for 1.6+ Its just a question of using the toosl together |
| 01:38.42 | [TK]D-Fender | saxa: 1.6.x works, just follow the instructions |
| 01:38.49 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 01:39.18 | [TK]D-Fender | saxa: And you should not be using RC's like that |
| 01:39.29 | [TK]D-Fender | saxa: use the release versions as listed on the page I linked |
| 01:40.04 | saxa | i just tried the rc after the stable was not working |
| 01:40.45 | saxa | also i´m using now zaptel with asterisk 1.4.24.1 |
| 01:40.54 | saxa | or trying to use it :) |
| 01:41.19 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 01:41.54 | saxa | tried the vrsions on the page you linked before, but i was unable to make the modules load |
| 01:42.25 | saxa | same system, same kernel as here now, but with zaptel 1.4.12.1 loads without complaining anything |
| 01:43.57 | f0ner00t | [TK]D-Fender: Goodevening. |
| 01:44.10 | f0ner00t | SIP Declined error message 603? |
| 01:44.29 | [TK]D-Fender | f0ner00t: pastebin something complete |
| 01:44.40 | f0ner00t | [TK]D-Fender: Yea I know. |
| 01:44.46 | f0ner00t | [TK]D-Fender: Thanks. |
| 01:45.04 | [TK]D-Fender | saxa: well jsut about everyone I know works jsut fine on 1.6.0.9 and the DAHDI release listed there |
| 01:45.31 | [TK]D-Fender | saxa: Maybe if you're running an extrememly bleeding edge kernel there might some kind of issue |
| 01:46.19 | f0ner00t | [TK]D-Fender: http://paste.cluenet.org/1996 |
| 01:47.01 | f0ner00t | \ |
| 01:47.42 | [TK]D-Fender | f0ner00t: Dialplan please |
| 01:50.50 | *** join/#asterisk AJayMN (i=AJayWisc@24-159-234-93.dhcp.mdsn.wi.charter.com) |
| 01:50.57 | xheliox | [TK]D-Fender: You're always wanting such mundane details. ;) |
| 01:51.08 | AJayMN | Can Asterisk do video as well as Skype video is? |
| 01:53.19 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 01:53.38 | f0ner00t | [TK]D-Fender: Where would dialplan be located what file? |
| 01:53.56 | [TK]D-Fender | ........... |
| 01:54.01 | [TK]D-Fender | f0ner00t: FFS extensions.conf |
| 01:54.52 | f0ner00t | I put extensions.conf I thought in their. |
| 01:55.20 | tjz | hey guys |
| 01:55.26 | f0ner00t | Sorry my bad. |
| 01:55.29 | f0ner00t | [TK]D-Fender: http://paste.cluenet.org/1995 |
| 01:56.27 | [TK]D-Fender | f0ner00t: -- Executing [1745@incoming:1] Goto("SIP/66.54.140.46-09eb7fd0", "Directory,2000,1") in new stack -- Goto (Directory,2000,1) |
| 01:56.42 | f0ner00t | [TK]D-Fender: What is wrong with that statment? |
| 01:56.44 | [TK]D-Fender | f0ner00t: Your destination doesn't exist. You basicaly just dumped them off a cliff. |
| 01:57.01 | f0ner00t | The directory is setup as ext 2000. |
| 01:57.05 | tjz | isit possible to make free call to landline from my asterisk server? |
| 01:57.16 | f0ner00t | I'm trying to understnad. I'm not being diffcult. |
| 01:57.23 | [TK]D-Fender | f0ner00t: Look at your own pastebin. No. It. Isn't. |
| 01:59.10 | f0ner00t | [TK]D-Fender: It was I corrected that earlier. |
| 01:59.11 | f0ner00t | Sorry. |
| 02:00.04 | f0ner00t | Is Directory the wrong app? |
| 02:00.26 | tjz | hmm.. |
| 02:01.00 | VaGoNeTaS | =) |
| 02:01.18 | [TK]D-Fender | f0ner00t: exten = 9999,1,Directory(default|default|ef) |
| 02:01.22 | f0ner00t | [TK]D-Fender: Sorry I had it set right I correctred that erlier. |
| 02:01.23 | f0ner00t | Yea |
| 02:01.29 | f0ner00t | Its - Executing [1745@incoming:1] Goto("SIP/66.54.140.46-0926cfb0", "directory,2000,1") in new stack -- Goto (directory,2000,1) |
| 02:01.32 | [TK]D-Fender | f0ner00t: Does that look like ***2000*** to YOU? |
| 02:01.43 | f0ner00t | I corrected it. |
| 02:01.47 | f0ner00t | 9999 is vm |
| 02:01.50 | f0ner00t | So I made changes. |
| 02:01.59 | [TK]D-Fender | f0ner00t: Show me something new then |
| 02:02.19 | [TK]D-Fender | f0ner00t: All I see is stuff that deserves an error |
| 02:02.31 | tjz | <- noob |
| 02:02.45 | f0ner00t | What else besides the 2000 do you see that deserves an error? |
| 02:02.55 | [TK]D-Fender | tjz: Who do you think is out there providing free service for you? |
| 02:02.56 | VaGoNeTaS | gus |
| 02:03.27 | tjz | even harder question to ask.. |
| 02:03.29 | f0ner00t | [TK]D-Fender: His parents LOL! |
| 02:03.31 | tjz | answer* |
| 02:03.57 | [TK]D-Fender | f0ner00t: Your dialplan as shown is broken. Fix it and if you still have an error provide a new pastebin |
| 02:03.58 | tjz | foneroot, i am fine with everything you say except talking about parent, mother etc |
| 02:04.08 | f0ner00t | [TK]D-Fender: Ok. |
| 02:04.27 | f0ner00t | tjz: I did not say anything mean about your parents. |
| 02:04.45 | tjz | don't even mention it? |
| 02:04.47 | f0ner00t | tjz: But thats the only way your going to get free that is all I was saying. |
| 02:04.55 | f0ner00t | tjz: I was playing / joking. |
| 02:05.02 | f0ner00t | tjz: Sorry if you took it offensive. |
| 02:05.45 | tjz | np |
| 02:07.33 | f0ner00t | [TK]D-Fender: http://paste.cluenet.org/1997 |
| 02:07.56 | f0ner00t | tjz: It was meant to be funny. Vonage if you buy there softphone service will allow you to hook their softphone service up to it. |
| 02:08.44 | tjz | cool |
| 02:08.47 | tjz | will check it out |
| 02:08.49 | tjz | thx, foneroot |
| 02:08.50 | [TK]D-Fender | f0ner00t: You can't have just an "n" priority you HAVE to have a "1" FIRST |
| 02:08.53 | tjz | =) |
| 02:09.42 | xheliox | [TK]D-Fender: Again, a stickler for details. |
| 02:10.00 | f0ner00t | exten => 1745,1,goto(directory,2000,1) |
| 02:10.07 | f0ner00t | Isn't that a priority? |
| 02:11.08 | [TK]D-Fender | f0ner00t: READ IT AGAIN |
| 02:11.21 | f0ner00t | [TK]D-Fender: Read the docs again'? |
| 02:11.38 | [TK]D-Fender | f0ner00t: exten = 2000,n,Directory(default|default|ef) <- your DESTINATION doesn['t have a "1" and is BROKEN |
| 02:11.54 | f0ner00t | [TK]D-Fender: Got it. |
| 02:11.59 | f0ner00t | It has to match. |
| 02:12.09 | [TK]D-Fender | f0ner00t: I swear I could remove your brain and make a rack-mount reverb usint out of it sometimes... |
| 02:12.16 | VaGoNeTaS | i work in Vonage |
| 02:12.16 | [TK]D-Fender | unit* |
| 02:12.50 | VaGoNeTaS | i use to* |
| 02:13.10 | f0ner00t | [TK]D-Fender: Why thank you. |
| 02:13.14 | joobie | TK |
| 02:13.16 | joobie | the man himself |
| 02:13.25 | f0ner00t | Well that didn't work but we will continue tommorrow maybe when I got time. |
| 02:13.26 | joobie | thanks for the heads up on the phone |
| 02:13.28 | f0ner00t | Have a good night. |
| 02:13.31 | joobie | didnt know it was in the menu system |
| 02:15.15 | f0ner00t | [TK]D-Fender: Something else is messed up.. I'll beback tommorrow thank you for your help. |
| 02:15.40 | f0ner00t | [TK]D-Fender: So it should be exten = 2000,1,Directory(default|default|ef) |
| 02:17.00 | [TK]D-Fender | f0ner00t: For at least that line, yes |
| 02:17.24 | [TK]D-Fender | f0ner00t: Of course THAT is completely broken anyways |
| 02:17.38 | [TK]D-Fender | f0ner00t: You have no functional [default] context |
| 02:18.36 | f0ner00t | Okay. |
| 02:18.38 | [TK]D-Fender | f0ner00t: Oh, and another thing.. still using illegal '|' delimiters |
| 02:18.49 | f0ner00t | I'll look at my [default] tommorrow. |
| 02:19.00 | f0ner00t | The gui did it I swear! |
| 02:19.07 | f0ner00t | LOL. |
| 02:19.09 | [TK]D-Fender | f0ner00t: Its pretty much completely busted fro the look of things. |
| 02:19.17 | f0ner00t | beback tommorrow. |
| 02:19.21 | f0ner00t | Thanks [TK]D-Fender. |
| 02:19.25 | [TK]D-Fender | f0ner00t: And the GUI is a continually broken POS |
| 02:19.43 | f0ner00t | [TK]D-Fender: That sucks. |
| 02:19.51 | f0ner00t | Okay I gotta run have a good night thanks for your help. |
| 02:19.57 | [TK]D-Fender | f0ner00t: Oh, actually I'm not 100% sure on the [default] bit depending how users.conf gets parsed |
| 02:20.09 | [TK]D-Fender | Silly twit :p |
| 02:20.35 | [TK]D-Fender | Almost had a working standalone system WEEKS ago and went down the GUI route. What a sad investment of our time |
| 02:26.03 | *** join/#asterisk crazyx__ (i=c4ced777@gateway/web/ajax/mibbit.com/x-bf5a0d98837a484f) |
| 02:28.00 | crazyx__ | hello everybody. please I need some help about url sending from asterisk to the softphone with the queue command - s,n,Queue(queuename,http://url.com) |
| 02:28.29 | crazyx__ | i'm trying to get it working with zoiper which support this url features |
| 02:28.45 | crazyx__ | but nothing ... |
| 02:29.16 | crazyx__ | some advice about syntax or about using sip or iax ? or may be another softphone -eyebeam etc...- supporting this features? |
| 02:30.56 | joobie | TK |
| 02:31.24 | joobie | I'm going to write a small c# app that sits on a users desktop and somehow displays the number of users that are in a queue on the asterisk system |
| 02:31.26 | crazyx__ | ? |
| 02:31.55 | [TK]D-Fender | joobie: Go for it |
| 02:31.58 | joobie | someone in here suggested the asterisk AMI.. which can grab this info.. but the problem is you have to send it an Action to get the info back |
| 02:32.17 | [TK]D-Fender | joobie: Can't get an answer if you don't get a question. |
| 02:32.18 | joobie | there doesn't appear to be an event that will just display to show the queue length |
| 02:32.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:32.33 | [TK]D-Fender | joobie: Keep looking |
| 02:32.39 | joobie | is there another way you can think of to achieve this, where asterisk will notify you? |
| 02:33.04 | joobie | TK, i went through all the events.. the only one that shows it is 'QueueStatus'... but that requires the Action to be sent |
| 02:33.06 | [TK]D-Fender | joobie: Maybe you shouldn't sit around expecting to be notified and instead ASK <- |
| 02:33.16 | [TK]D-Fender | joobie: So SEND IT |
| 02:33.18 | crazyx__ | please no one can help me about the url sending feature in asterisk queue command using zoiper softphone ? |
| 02:33.30 | [TK]D-Fender | crazyx__: Look at how that phone is actually being called |
| 02:33.31 | joobie | AMI has the ability to notify on certain events, just not that |
| 02:33.44 | joobie | TK, the problem with sending is to make it "realtime" .. i need to send the Action every second |
| 02:33.52 | joobie | which is a waste of i/o .. bandwidth.. etc etc |
| 02:33.58 | joobie | i'm trying to save the planet here and not waste i/o |
| 02:34.14 | [TK]D-Fender | joobie: Feel free to mod app_queue.c |
| 02:34.24 | joobie | is there another interface that can be used for this sorta stuff? I was thinking I can even use AGI.. but that's just getting gay |
| 02:34.44 | joobie | yea I had a look at app_queue.c .. it's pretty straight forward I think.. but is that my only option? |
| 02:34.47 | [TK]D-Fender | joobie: AGI certainly has absolutely nothing to do with this |
| 02:34.49 | joobie | apart from AGI |
| 02:34.53 | joobie | sec phone |
| 02:34.57 | [TK]D-Fender | NO AGI |
| 02:36.24 | joobie | back |
| 02:36.26 | joobie | AGI can do it |
| 02:36.46 | joobie | i'm scripting something anyway to tack into the AMI interface.. can just script osmething that tacks onto AGI script |
| 02:36.49 | joobie | but, it's gay. |
| 02:37.03 | joobie | i was just hoping there was a simple way to achieve it |
| 02:37.44 | joobie | TK.. i've never .. ever.. moded the src for app_queue or a specific asterisk module.. if i amend it, can i just recompile that one module and overwrite the old? |
| 02:37.46 | crazyx__ | TK how that softphone is called? i just want to get this action : queue["Myqueue", http://url.com/) working... i tried many softphone |sip/iax] but it isn't work and i'm sure that Zoiper supports this option. May be i'm missing something. or may be i didn't understand what u mean by "look at how phone is called".. |
| 02:38.30 | [TK]D-Fender | joobie: And what does AGI offer to break down queue stats? |
| 02:38.51 | [TK]D-Fender | joobie: AGI implies you even have a CALL trying to do tracking. |
| 02:38.58 | [TK]D-Fender | joobie: That alone is retarded |
| 02:39.03 | joobie | AGI offers endless posiblities, but they are gay |
| 02:39.05 | *** join/#asterisk timeshell__ (n=chatzill@206.248.136.108) |
| 02:39.12 | joobie | dialplan can write to astdb everytime someone joins the queue |
| 02:39.14 | joobie | and removes from the queue |
| 02:39.16 | [TK]D-Fender | crazyx__: No, tahts the caller falling into Queue. Look at the MEMBERS |
| 02:39.20 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a847b4091eb914f7) |
| 02:39.24 | [TK]D-Fender | joobie: No, it doesn't |
| 02:39.25 | joobie | and agi can just read the astdb |
| 02:39.38 | joobie | it doesn't, but it can |
| 02:39.48 | [TK]D-Fender | joobie: And that isn't a very safe way to try to track thing |
| 02:39.55 | joobie | i know.. can fall out of sync |
| 02:39.58 | [TK]D-Fender | joobie: thats is actually beyond ugly |
| 02:40.01 | joobie | would need somehting to do a sanity check |
| 02:40.05 | joobie | agreed:) |
| 02:40.10 | joobie | which is why ive been looking at AMI |
| 02:40.12 | [TK]D-Fender | joobie: No... I'm SURE you're insane. |
| 02:40.15 | joobie | but AMI is limiting.. |
| 02:40.33 | [TK]D-Fender | joobie: No, AMI can do a lot more. |
| 02:40.33 | joobie | TK, the sanity check would keep me sane |
| 02:40.35 | xheliox | [TK]D-Fender: Well, he called it gay, so how can you argue with such well reasoned thought? |
| 02:40.58 | [TK]D-Fender | xheliox: His mind is too little to be let out alone :) |
| 02:41.00 | xheliox | Anyone who calls something gay that doesn't work to their liking is clearly of superior logic and insight than yourself. |
| 02:41.20 | [TK]D-Fender | joobie: Go sniff another line... that's some good shit you're on :) |
| 02:41.41 | joobie | AMI is limiting in the sense (and this is by no way bagging the developers whom do a shitload of mad stuff already), it would be great if you could specify which events you want to alert on.. rather than having only a set few of events.. and another subset of actions which cant be used as events |
| 02:41.49 | joobie | TK, want sum?:P |
| 02:42.07 | crazyx__ | TK yeah the inconming call is sended to the queue, the users, once loggued are added [ asterisk -rx queue add member SIP/... to Queue ] . What i'm trying to do is using the URL parameters on queue command -instead of Sendurl- |
| 02:42.15 | [TK]D-Fender | joobie: But oh no! We only have a limited set of DIALPLAN apps! The dialplan is gay too! |
| 02:42.29 | [TK]D-Fender | joobie: And sip.conf?! D I even have to start? GAY!!!!!!!! |
| 02:42.39 | [TK]D-Fender | joobie: .... ok... are you done now? |
| 02:42.47 | [TK]D-Fender | joobie: AMI works perfectly fine for this |
| 02:42.50 | joobie | ok I get your point. |
| 02:42.57 | eppigy | HOW DARE YOU |
| 02:42.58 | joobie | I think i used the wrong words |
| 02:43.02 | joobie | :) |
| 02:43.10 | joobie | It's gay, relative to what im trying to achieve |
| 02:43.14 | [TK]D-Fender | joobie: You are being a neurotic twit, and I WILL call you on it. Each and every time. |
| 02:43.14 | joobie | that's better .. |
| 02:43.14 | crazyx__ | to get an URL opened once the call is answered . |
| 02:43.21 | joobie | fair enuf |
| 02:43.38 | [TK]D-Fender | joobie: "The only constant factor in all your dysfunctional relationships is YOU" <- |
| 02:43.40 | *** join/#asterisk AJayMN (i=AJayWisc@24-159-234-93.dhcp.mdsn.wi.charter.com) |
| 02:43.54 | AJayMN | does Asterisknow support H.264 or do you have to add it? |
| 02:44.10 | joobie | anyway.. AMI DOESN'T work perfectly fine for this.. perfectly fine would be configuring the AMI to be able to trigger an event on any queue size change.. rather than having to send a QueueStatus each second and test if it's changed |
| 02:44.45 | *** join/#asterisk t3hrealadamd (n=t3hreala@c-24-3-152-246.hsd1.pa.comcast.net) |
| 02:45.19 | [TK]D-Fender | joobie: Polling works. YOU don't like having to send an event. TOO DAMN BAD. Get over it |
| 02:45.57 | [TK]D-Fender | joobie: And you don't need to sent one every second. |
| 02:46.21 | joobie | for this interested.. http://www.russellbryant.net/blog/2008/06/19/how-to-write-an-asterisk-module-part-1/ makes writing asterisk modules a piece of cake |
| 02:46.30 | joobie | .. just the basics on that link |
| 02:47.01 | joobie | TK.. i'm trying to achieve realtime monitoring.. not 5s delay monitoring |
| 02:47.52 | joobie | AMI has the ability to do its own events, wihtout requiring an action.. so it sounds like it's just a feature addon. I'll have a look at the src on the weekend and see if it's hard to extend out to queuestatus |
| 02:48.19 | [TK]D-Fender | joobie: There are other ways to reduce polling. |
| 02:48.41 | joobie | how? |
| 02:48.50 | [TK]D-Fender | joobie: Go think for a little bit. |
| 02:48.50 | dkdkd | hi, so i'm reading this SIP/NAT intro: http://www.voipuser.org/forum_topic_7295.html and I have a couple questions |
| 02:48.56 | [TK]D-Fender | ~sipnat |
| 02:48.57 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:48.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 02:49.02 | joobie | I've thought |
| 02:49.05 | joobie | thought all yesterday |
| 02:49.07 | joobie | then it hit me |
| 02:49.10 | joobie | I should ask TK |
| 02:49.13 | [TK]D-Fender | hits joobie |
| 02:49.17 | joobie | heh |
| 02:49.26 | dkdkd | it sounds to me like the Via and Contact header problem is handled by the SIP protocol itself, which does the re-write. |
| 02:49.42 | crazyx__ | TK any advice plz? |
| 02:49.48 | [TK]D-Fender | joobie: Whenever a channel is created or torn down * spits out an event. Poll on THAT |
| 02:50.08 | [TK]D-Fender | crazyx__: I'm not seeing you show me what members you use and how. |
| 02:50.14 | dkdkd | in asterisk with SIP debug on, i see this: Reliably Transmitting (no NAT) to 67.108.9.165:5060: |
| 02:50.28 | dkdkd | what is up with the (no NAT) part of the log message? |
| 02:50.39 | [TK]D-Fender | dkdkd: pastebin the SIP debug of a failed call |
| 02:50.41 | [TK]D-Fender | ~pb |
| 02:50.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 02:50.42 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 02:51.11 | [TK]D-Fender | dkdkd: because as configured * is responding to that IP trusting that the source is right |
| 02:51.24 | [TK]D-Fender | dkdkd: Go read the guide I linked |
| 02:51.33 | dkdkd | http://pastebin.com/m663a97fd |
| 02:51.53 | dkdkd | that is an OPTIONS that is sent out that I never get a response to |
| 02:52.00 | dkdkd | this is behind NAT |
| 02:52.04 | [TK]D-Fender | dkdkd: Contact: <sip:Unknown@10.16.70.16> <-- indeed your setup is wrong |
| 02:52.21 | [TK]D-Fender | dkdkd: You are telling them a Class-C address. |
| 02:52.23 | dkdkd | yes, i know that is an internal non-routable ip |
| 02:52.29 | [TK]D-Fender | dkdkd: Read the guide and fix your configs |
| 02:52.58 | dkdkd | but the other primer i just linked says that Via/Contact info is re-written on the receiving end by the source IP if the src IP does not match what is in the SIP header |
| 02:53.20 | [TK]D-Fender | dkdkd: do not assume what the RECEIVER doesn |
| 02:53.24 | [TK]D-Fender | -n |
| 02:53.30 | [TK]D-Fender | dkdkd: YOU are sending it wrong. |
| 02:53.50 | [TK]D-Fender | dkdkd: Go read the guide and fix your configs |
| 02:54.36 | joobie | ahh |
| 02:54.38 | joobie | I see what you mean TK |
| 02:54.48 | crazyx__ | queues.conf : [general] ... persistantmembers=yes ... [myqueue] ... context default .... // extensions.conf [from-DID] exten => s,1,Queue("myqueue", http://91.2XX.XX.XX8/client.php) // login.php : system("asterisk -rx queue add member IAX2/$user . to $queue"); .... login.php works fine, queue also (member received calls...) just the URL features not work - |
| 02:55.02 | joobie | so use a 30s poll for example to get the actual queue, and alter it via the channel status until the next poll |
| 02:55.06 | joobie | good idea |
| 02:55.13 | crazyx__ | don't know if i'm doing mistake on asterisk or something with softphone or something else |
| 02:55.28 | crazyx__ | thanks for helping me TK |
| 02:55.52 | joobie | TK, do you have any referneces for compiling a single asterisk module? I might give the src a go on the weekend, but im finding people are saying u need to recompile asterisk for it to do the module |
| 02:55.53 | crazyx__ | i'm using zoiper which support receving url on answer calls... |
| 02:56.07 | joobie | which i want to avoid.. rather just recompile app_queue.so and overwrite the existing module |
| 02:57.13 | [TK]D-Fender | crazyx__: I don't know that the client supports it mind you |
| 02:57.33 | joobie | scratch that TK.. someone on http://www.russellbryant.net/blog/2008/06/19/how-to-write-an-asterisk-module-part-1/ in the comments provided a process to do this |
| 02:57.36 | joobie | cheers for the help / ideas |
| 02:58.27 | [TK]D-Fender | joobie: And FYI I monitor 2 queues, 4 agents, and 2 VM boxes all live on my CSR Polycom phones screen, no desktop UI required. |
| 02:58.39 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 02:58.44 | [TK]D-Fender | joobie: Been there, done that |
| 02:59.02 | [TK]D-Fender | crazyDoes it work if you do it direct via Dial? |
| 02:59.10 | *** join/#asterisk crazyx__ (i=c4ced777@gateway/web/ajax/mibbit.com/x-d40e9aad223df41c) |
| 02:59.30 | crazyx__ | TK$ sorry was disconnet |
| 02:59.34 | crazyx__ | ct |
| 02:59.43 | joobie | TK, i got the fuken microbrowser all setup and ready to go.. wrote a PHP interface to AMI so the microbrowser just polls that.. but the freaken client wants it on the desktop now instead of the phone.... ergh |
| 02:59.57 | joobie | was interesting tho looking into the microbrowser - new territory for me |
| 02:59.57 | crazyx__ | so any advice may be - plz plz plz - |
| 03:00.31 | joobie | got an annoying quirk in the polycom 320's... whenever images refresh on the screen, the LCD goes black for the full size of the image, and then removes the pixels as per the image |
| 03:00.50 | joobie | so you see this really crappy refresh happening on the screen.. only seems to happen with images, text just renders as per normal |
| 03:01.20 | [TK]D-Fender | crazyx__: Does it work if you do it direct via Dial? |
| 03:01.48 | [TK]D-Fender | joobie: I run mine on IP 600's |
| 03:02.24 | crazyx__ | i dont try it yet |
| 03:02.25 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 03:02.32 | crazyx__ | give me 2 mn |
| 03:02.33 | crazyx__ | i'll see |
| 03:04.05 | *** join/#asterisk blkry (n=blkry@24-241-119-223.dhcp.gnvl.sc.charter.com) |
| 03:06.49 | Qwell | anybody know where 931 areacode is? |
| 03:07.46 | joobie | 600's have a sweet sized display |
| 03:08.20 | [TK]D-Fender | Qwell: TN |
| 03:09.00 | crazyx__ | TK no it doesn't work with Dial |
| 03:09.06 | crazyx__ | but it's work with MozIax |
| 03:09.17 | crazyx__ | i'll try for queue also |
| 03:09.24 | crazyx__ | i think only moziax can do it ... |
| 03:09.35 | [TK]D-Fender | crazyx__: Or only Zoiper CAN'T |
| 03:10.05 | [TK]D-Fender | slings his BlameThrower on and begins to spray furiously |
| 03:12.26 | crazyx__ | i tried zoiper free and biz, xlite, eyebeam, and some others same result |
| 03:13.01 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 03:16.13 | crazyx__ | grrr ... moziax get the url and it's working... so the problems is coming for softphone |
| 03:16.34 | crazyx__ | TK i'll search for a solution |
| 03:16.57 | crazyx__ | moziax is too limited - few features... - |
| 03:17.19 | crazyx__ | maybe someone else know a softphone supporting the URL sending on call answer |
| 03:17.23 | crazyx__ | ? no? |
| 03:19.09 | *** join/#asterisk securevoip (n=hardcore@c-76-123-20-170.hsd1.va.comcast.net) |
| 03:20.06 | *** part/#asterisk securevoip (n=hardcore@c-76-123-20-170.hsd1.va.comcast.net) |
| 03:20.32 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 03:41.18 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:41.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:43.33 | *** join/#asterisk TXTrader (n=kvirc@72.183.119.173) |
| 03:49.54 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 03:56.00 | *** join/#asterisk ectospasm (i=ectospas@i.hate.your.vhosts.shellium.org) |
| 03:57.28 | ectospasm | I'm trying to set up a scenario in our lab, and I'm trying to get one system (called lab2) to signal a hangup with a polarity reversal (to lab1). The call are progressing nicely, but lab2 always signals hangup with a battery drop. |
| 03:57.48 | ectospasm | Is there any way force lab2 to signal hangup by a polarity reversal? |
| 03:58.00 | ectospasm | lab1(FXO) -> lab2(FXS) |
| 03:58.57 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:00.45 | *** join/#asterisk canadait (n=canadait@S01060011950c0261.lb.shawcable.net) |
| 04:01.24 | canadait | hi...quick question about MOH...i have a call queue with MOH defined and I have uploaded an MP3 but it won't play in the queue |
| 04:02.40 | [TK]D-Fender | canadait: Have you installed asterisk-addons? |
| 04:02.42 | ectospasm | canadait: try a different format instead of MP3 |
| 04:02.50 | canadait | ok |
| 04:02.53 | ectospasm | mp3's generally sound like crap anyway |
| 04:03.04 | canadait | oh ok like wave or something |
| 04:03.06 | [TK]D-Fender | ectospasm: hardly. MP3 > PSTN |
| 04:03.23 | ectospasm | [TK]D-Fender: yeah, but MP3 usually sounds like crap on PSTN... |
| 04:03.44 | [TK]D-Fender | ectospasm: Shouldn't sound worse than anything else. |
| 04:04.05 | *** join/#asterisk shinao1 (n=shinao1@41.219.245.142) |
| 04:04.13 | [TK]D-Fender | ectospasm: and that should be reprased "PSTN sounds like crap, MP3 just happens to be going on at that moment" |
| 04:04.15 | canadait | isn't there samples included..can't i just use those? |
| 04:04.33 | [TK]D-Fender | canadait: Sure, but answer my question. |
| 04:04.47 | ectospasm | you should have the freeplay MOH files by default. |
| 04:06.34 | canadait | frankly and sadly I don't know if I have add ons ..i did do a full mod admin update |
| 04:13.59 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 04:17.28 | [TK]D-Fender | canadait: "full mod admin update"? Huh? |
| 04:18.23 | canadait | yes..ran>>module admin and updated all the modules to the latest |
| 04:19.22 | [TK]D-Fender | canadait: Asterisk doesn't have an admin module. |
| 04:19.42 | canadait | ok well i am mistaken then sorry |
| 04:20.40 | [TK]D-Fender | canadait: What exactly have you installed? |
| 04:21.18 | canadait | asterisknow |
| 04:21.48 | [TK]D-Fender | canadait: well go to * CLI and pastebin the output of "module show like format" |
| 04:21.50 | [TK]D-Fender | ~pb |
| 04:21.50 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 04:21.52 | [TK]D-Fender | ^^^^^^^^^^^^^^^6 |
| 04:22.50 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:22.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:25.28 | canadait | http://pastebin.com/m1922e3a6 |
| 04:26.45 | [TK]D-Fender | canadait: You clearly do not have fomat_mp3.so which is part of asterisk-addons (a separate package) |
| 04:26.52 | [TK]D-Fender | canadait: Thus cannot decode MP3 files |
| 04:27.00 | [TK]D-Fender | canadait: Install it and you will be able to |
| 04:27.16 | canadait | ok thanks I will take a look |
| 04:29.06 | *** join/#asterisk jtodd (n=jtodd@253.sub-75-254-237.myvzw.com) |
| 04:29.06 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 04:39.27 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:41.20 | *** join/#asterisk voxter (n=voxter@76.77.91.252) |
| 04:49.13 | *** join/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 04:51.40 | AndyML | if anyone has any experience with sangoma/wanpipe and dahdi... when I run wancfg_dahdi I get FATAL: Error inserting wanpipe (/lib/modules/2.6.18-128.1.6.el5/kernel/drivers/net/wan/wanpipe.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 04:52.45 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 04:53.11 | tzafrir_laptop | AndyML, take a look at the kernel logs |
| 04:53.19 | tzafrir_laptop | dmesg | tail |
| 04:53.27 | AndyML | sure. just a moment. |
| 04:53.38 | tzafrir_laptop | maybe you rebuilt dahdi after you built wanpipe? |
| 04:54.41 | AndyML | nah. this is a fresh install. I've run Setup dahdi several times since i built dahdi |
| 04:54.53 | AndyML | http://pastie.org/463464 |
| 04:56.11 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 05:03.10 | TXTrader | AndyML: did you try depmod -a after the install? |
| 05:03.28 | AndyML | i did. turns out my kernel was out of sync with my kernel-devel version |
| 05:03.39 | AndyML | then my dahdi modules were out of sync with my kernel once i upgraded it. |
| 05:03.41 | AndyML | what a zoo |
| 05:06.47 | AndyML | of course now dahdi isn't loading the module right. |
| 05:06.47 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 05:07.04 | [TK]D-Fender | checkout time, later all |
| 05:07.10 | AndyML | ni ni |
| 05:08.08 | AndyML | wanconfig: WAN device wanpipe1 driver load failed !! |
| 05:08.08 | AndyML | <PROTECTED> |
| 05:08.08 | AndyML | <PROTECTED> |
| 05:08.49 | AndyML | TXTrader: any ideas? ^ |
| 05:09.16 | TXTrader | AndyML: not right off, haven't used the zaptel stuff in a few years |
| 05:09.46 | AndyML | everyone swears by these sangoma cards but i don't think i've ever seen them work perfectly... |
| 05:10.05 | TXTrader | doesn't really sound like the card's fault in this case |
| 05:10.18 | AndyML | oh i'm sure it isn't the card's fault... |
| 05:10.28 | AndyML | but if it was a tdm400p it'd be configured by now... |
| 05:10.39 | AndyML | more to do with previous experience than anything else |
| 05:11.05 | TXTrader | what's dmesg say now? |
| 05:11.54 | AndyML | i bet the power cable isn't connected |
| 05:12.03 | AndyML | http://pastie.org/463481 |
| 05:12.07 | AndyML | no modules found |
| 05:12.32 | AndyML | ERROR: wanpipe1: No FXO/FXS modules are found! |
| 05:12.45 | TXTrader | right - so might be the card after all ;) |
| 05:12.54 | AndyML | heh |
| 05:28.31 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d56aea9eb41147d9) |
| 05:31.52 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 05:46.44 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 05:55.55 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 06:01.11 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:02.24 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:19.30 | *** join/#asterisk cyd777_wrk (n=jbihari@bihari.eiszk.pte.hu) |
| 06:20.04 | cyd777_wrk | hi guys |
| 06:21.39 | cyd777_wrk | I'm very newbie in asterisk and I have a question |
| 06:21.56 | *** join/#asterisk mattwj2002 (n=matt@c-71-63-163-89.hsd1.mn.comcast.net) |
| 06:22.57 | cyd777_wrk | I have a an asterisk on a debian lenny to which a connect with a softwarephone. this asterisk connect to an avaya pbx with a h323 trunk. If there is incoming call I can answer with softwarephone. My question is how can I route all outgoing call to the trunk to avaya? |
| 06:23.31 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 06:25.40 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 06:26.34 | ltd_wk | Is there any reason why exten => *98,x,yyyy wouldn't match in a context? |
| 06:32.18 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 06:35.18 | *** join/#asterisk jeffspeff (n=jeffspef@c-98-240-118-231.hsd1.ky.comcast.net) |
| 06:36.27 | jeffspeff | can somebody help... i'm getting the following error. I recompiled from scratch and didn't fix... [Apr 30 01:34:12] WARNING[22731]: func_strings.c:652 acf_strftime: C function strftime() output nothing?!! |
| 06:36.49 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 06:36.52 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 06:38.55 | jeffspeff | <PROTECTED> |
| 06:39.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:41.06 | jeffspeff | hey, is anybody online? |
| 06:43.57 | mattwj2002 | yup |
| 06:44.31 | mattwj2002 | no idea though |
| 06:44.54 | jeffspeff | ok, |
| 06:45.30 | jeffspeff | i get that error when trying to make an outbound call |
| 06:45.45 | mattwj2002 | did it compile fine? |
| 06:45.59 | mattwj2002 | or did you get some errors? |
| 06:46.08 | jeffspeff | yeah, no probs... it was working for a bit, then just started doing that |
| 06:46.30 | mattwj2002 | did you make any changes ? |
| 06:47.40 | tzafrir_laptop | AndyML, just in case I missed it: all of those symbols are from dahdi.ko |
| 06:48.22 | tzafrir_laptop | Either somebody insmod-ed wanpipe directly or it was built vs. a different version of DAHDI |
| 06:48.30 | tzafrir_laptop | lsmod | grep ^dahdi |
| 06:50.13 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 06:52.53 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 06:52.58 | troy- | sup SwK |
| 06:53.07 | SwK | so did digium shut the office for the rest of the week yet? |
| 06:53.48 | SwK | troy-, wondering how stupid people are going to get around huntsville since they have 2 cases of H1N1/SwineFlu here |
| 06:54.12 | troy- | hah |
| 06:55.07 | troy- | swineflu is for pigs |
| 06:57.17 | Corydon76-dig | Two suspected, but unconfirmed cases |
| 06:57.31 | SwK | confirmed now |
| 06:57.56 | Corydon76-dig | as of when? |
| 06:58.03 | SwK | 10p news |
| 06:58.29 | Corydon76-dig | Last I heard, they were still waiting for the CDC's lab results |
| 06:59.14 | SwK | local news said they confirmed it at 10... but then again.. hah who knows... they closed all the schools and pretty much anything where kids might go |
| 06:59.38 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 07:00.35 | Corydon76-dig | Heh, the news article is misleading. |
| 07:01.15 | Corydon76-dig | It says that the probable cases are confirmed, which means, yes, there are probable cases, and they aren't just a rumor |
| 07:01.35 | Corydon76-dig | but it's not confirmed that the probable cases are, in fact, swine flu |
| 07:01.52 | Corydon76-dig | http://www.whnt.com/news/whnt-two-swine-flu-cases,0,852076.story |
| 07:03.40 | SwK | heh |
| 07:03.43 | SwK | crazy news people |
| 07:03.50 | SwK | i was only 1/2 paying attenion to it anyway |
| 07:05.50 | Corydon76-dig | For that matter, TN now has one "probable" case |
| 07:07.09 | Corydon76-dig | The TN news story has more information on the test: http://www.wrcbtv.com/Global/story.asp?S=10275590 |
| 07:07.22 | Corydon76-dig | 95% accurate, but the CDC must confirm |
| 07:07.22 | SwK | 95% hah |
| 07:08.08 | SwK | Corydon76-dig, you still in nash? |
| 07:08.13 | Corydon76-dig | Yep |
| 07:28.04 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 07:29.24 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 07:33.55 | *** join/#asterisk jeffspeff (n=jeffspef@c-98-240-118-231.hsd1.ky.comcast.net) |
| 07:35.06 | jeffspeff | i've erased everything again, updated my kernel and recompiled... still getting this weird error... WARNING[18140]: func_strings.c:652 acf_strftime: C function strftime() output nothing?!! any help would be great |
| 07:37.53 | *** join/#asterisk porche (n=kursad@88.239.77.171) |
| 07:37.58 | porche | hi all |
| 07:38.50 | porche | is there a way to understand the called number type, such as mobile, landline, voip? |
| 07:39.35 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:41.36 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 07:43.24 | jeffspeff | i've erased everything again, updated my kernel and recompiled... still getting this weird error... WARNING[18140]: func_strings.c:652 acf_strftime: C function strftime() output nothing?!! any help would be great |
| 07:43.27 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 07:47.51 | porche | :) |
| 07:47.56 | porche | no one alive i think |
| 07:48.31 | tzafrir_laptop | notes that jeffspeff is alive |
| 07:48.34 | tzafrir_laptop | hides |
| 07:49.12 | tzafrir_laptop | jeffspeff, strftime is libc and not much kernel |
| 07:49.30 | tzafrir_laptop | Maybe it simply got an empty format string? |
| 07:50.42 | jeffspeff | what do you mean by an empty format string? like codec? |
| 07:51.51 | jeffspeff | tzafrir_laptop, or maybe something in my dialplan? |
| 07:52.36 | porche | :) |
| 07:53.00 | porche | my questions is, is there a way to detect a called number type, as mobile, landline, voip? |
| 07:55.56 | tzafrir_laptop | jeffspeff, please provide a trace from running this with verbosity level 3 (or higher) |
| 07:57.41 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:57.58 | tzafrir_laptop | called number: normally this would be the extension number |
| 07:59.26 | jeffspeff | tzafrir_laptop, http://pastebin.ca/1408271 |
| 08:00.34 | tzafrir_laptop | hmm... I thought this was from STRFTIME(), but aparantly it isn't |
| 08:00.47 | tzafrir_laptop | is too lazy to look at the code right now |
| 08:00.52 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:01.08 | jeffspeff | any suggestions? |
| 08:07.59 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 08:11.52 | *** join/#asterisk kippi (n=chriso@83-244-164-130.cust-83.exponential-e.net) |
| 08:11.53 | kippi | hey |
| 08:12.01 | jeffspeff | hi |
| 08:12.09 | *** join/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 08:12.28 | SwK | porche, well that all depends on where the caller is... (ie what country)... but in the US there is ANI-II if you can get those digits or you have to resolve the number via lerg and then you really wont be able to tell the differences between landline and voip |
| 08:12.30 | kippi | in 1.6 I have seen a application called TestServer, what is this for? are there any docs on this? |
| 08:13.28 | jeffspeff | SwK, any ideas on my issue? |
| 08:13.47 | tzafrir_laptop | kippi, it has been (almost) unchanged since at least 1.0, actually |
| 08:13.55 | kippi | oh ok |
| 08:14.00 | tzafrir_laptop | there's TestCleint and TestServer |
| 08:14.08 | tzafrir_laptop | TestCleint calls TestServer |
| 08:14.45 | tzafrir_laptop | it's a simple test for the line. Will fail if it fails to connect or audio is really lousy |
| 08:14.55 | SwK | jeffspeff, nope... never seen that |
| 08:15.05 | jeffspeff | k |
| 08:15.34 | jeffspeff | tzafrir_laptop, do you think that format and re-install would fix it? the error just started out of the blue it seems |
| 08:16.06 | tzafrir_laptop | jeffspeff, no. "format" is there as in "a format string" |
| 08:16.15 | tzafrir_laptop | man 3 strftime |
| 08:16.17 | porche | swk, sorry just saw |
| 08:16.27 | porche | is the answer ANI II? |
| 08:16.32 | tzafrir_laptop | which is a lousy shorthand for STRing Format TIME |
| 08:17.05 | SwK | porche, possibly... |
| 08:17.19 | SwK | porche, like i said it depends on where you are talking about... not all countries do it the same |
| 08:17.34 | porche | i met the same info, only reasonable one ani II |
| 08:17.37 | kippi | tzafrir_laptop: would it allow me to busy out ISDN line? |
| 08:17.37 | SwK | numbering plans etc are region or country specific |
| 08:17.49 | jeffspeff | i mean format as in erase hdd and start over... hopefully erasing what ever the hell is causing this... it was working properly on this machine at one time |
| 08:17.51 | porche | can place a box anywhere, but have to find a way |
| 08:18.07 | porche | swk, ani II is only available to zap channels I think |
| 08:18.10 | *** join/#asterisk dr_gogeta86 (n=fisgro@81-208-88-100.ip.fastwebnet.it) |
| 08:18.13 | tzafrir_laptop | kippi, it's an application as any |
| 08:18.29 | tzafrir_laptop | Though the TestServer listens with some timeout IIRC |
| 08:24.27 | jeffspeff | tzafrir_laptop, would it have anything to do with Using SIP RTP CoS mark 5 |
| 08:26.16 | SwK | porche, why would it only be available on zap channels? thats like saying you can only get RDNIS off of a zap channel |
| 08:28.14 | SwK | porche, 1) there is an ANI-II field that may contain that information, 2) it may not be applicable to the region of the world in question 3) your upstream provider may not even support/translate it properly 4) depending on what region of the world you are in, you might be able to check the organization responsible for numbering resources and see what they have to offer... |
| 08:29.06 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 08:30.31 | SwK | I can not speak for other areas of the world, but int he US ANI-II may or may not be available depending on what carrier you use and then you can just look at a prefix and see what/where a number is (well you can to an extent) but you need a copy of the LERG to resolve what a specific range is... its not like in Europe where 447 is always mobile and 4420 is wireline or possibly voip in london |
| 08:30.47 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:31.24 | SwK | a perfect example is 1901 is Western Tennessee geographically but inside that NPA, 1901853 is wireline and 1901857 is wireless... |
| 08:31.55 | SwK | and you need to depend on ANI-II to see if the wireline is a payphone or not |
| 08:32.02 | jeffspeff | most of memphis area is 901 |
| 08:32.21 | SwK | mostly |
| 08:32.43 | jeffspeff | is southaven/hornlake area still 601? |
| 08:32.48 | SwK | jeffspeff, i'm originally from memphis |
| 08:32.55 | SwK | nope... thats 662 now |
| 08:33.09 | jeffspeff | 662, that's right... been a while |
| 08:33.09 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 08:33.34 | jeffspeff | lived in hornlake for 10 yrs before moved to ky |
| 08:33.58 | SwK | jeffspeff, yeah where? |
| 08:34.03 | SwK | in HL that is? |
| 08:34.15 | jeffspeff | let me think... |
| 08:34.15 | SwK | jeffspeff, i grad from hlhs |
| 08:34.31 | SwK | west mem is no long 501 either... its 732 now |
| 08:34.39 | jeffspeff | wow |
| 08:34.56 | SwK | 662 and 732 have been around for nearly a decade |
| 08:35.12 | jeffspeff | what year did you grad? |
| 08:35.16 | SwK | 91 |
| 08:35.24 | NMR_1122 | Hi everyone, |
| 08:35.24 | NMR_1122 | Is there a way I can send the "alert-info" sip header to certain phones but not all of them? |
| 08:35.49 | porche | got it |
| 08:35.58 | porche | so first step is ANI-II |
| 08:35.58 | ectospasm | NMR_1122: you could have the phones that need to and those that don't in different dialplan contexts |
| 08:36.44 | ectospasm | ...or you could use dialplan logic to determine what type of phone you're sending to, perhaps a DB lookup of some sort. |
| 08:36.52 | SwK | and its a n i - eye eye ... not ani-"two" |
| 08:37.12 | jeffspeff | SwK, go down goodman rd., stop light used to be at stateline rd.; turn right go down about 3/4 mile (right by kingston west neighbor hood); turn left on Rolling Oaks Dr.; first house on right, on top of hill |
| 08:37.37 | NMR_1122 | I'm trying to make them both ring at the same time, in the same dial line.... Ones a cell phone, though a voip trunk, and If i set the header, it won't dial, it gets refused by the provider. The rest are local network phones and respond correctly to the double ring command. |
| 08:37.42 | SwK | yep I know where that is... i lived on the west side... |
| 08:37.54 | jeffspeff | kingston west? |
| 08:38.11 | SwK | jeffspeff, westside of HL... cant remember the damned street name hah |
| 08:38.24 | SwK | but goodman and stateline run parallel... :P |
| 08:38.29 | jeffspeff | i'm amazed it came back to me |
| 08:38.31 | ectospasm | NMR_1122: I guess it's all SIP? |
| 08:38.34 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
| 08:38.35 | SwK | you mean 51 I think |
| 08:38.43 | NMR_1122 | yeah |
| 08:39.03 | jeffspeff | the road they built the new (at the time) HLMS |
| 08:39.22 | ectospasm | NMR_1122: if your voip provider could do IAX2, the SIPAddHeader application would have no effect IIRC. |
| 08:39.29 | SwK | actually you dont have to send alert-into based on context... just set a channel var for the phones that need it and then use that to do the sipaddheader |
| 08:39.38 | ectospasm | ...at least for that section of the Dial call |
| 08:39.52 | porche | hmms |
| 08:40.04 | porche | swk do you know of any provider supports ani II |
| 08:40.17 | SwK | porche, what area of the world |
| 08:40.40 | SwK | most in the US can.. you just have to ask them for it |
| 08:40.40 | jeffspeff | ectospasm, http://pastebin.ca/1408291 what is causing the func_strings.c:652 acf_strftime: C function strftime() output nothing?!! |
| 08:41.23 | ectospasm | jeffspeff: I have no clue. What version of Asterisk? |
| 08:41.47 | jeffspeff | ectospasm, Asterisk 1.6.0.9 |
| 08:42.10 | SwK | jeffgus, is it causing a problem? |
| 08:42.16 | ectospasm | what other symptoms are you having? Or is it just that error? |
| 08:42.31 | porche | well I asked 1-2, they said no |
| 08:42.56 | porche | for US mainly |
| 08:42.58 | ectospasm | 'scuse me, warning |
| 08:42.58 | jeffspeff | the calls don't connect, i'm guess due to that error... before that error apeared, calls connected fine. |
| 08:43.08 | porche | for the rest of the world, usually it's known from prefix |
| 08:43.09 | SwK | porche, what kind of volume? |
| 08:43.13 | NMR_1122 | When you say "just set a channel var for the phones that need it" what do you mean? Can I still put both in a dial line, like Dial(SIP/1001&SIP/1002&SIP/VOIPCOMPANY/5555551234) |
| 08:43.13 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:43.23 | porche | for start low swk |
| 08:43.24 | ectospasm | did you do any glibc updates since you compiled Asterisk? |
| 08:43.46 | SwK | porche, define low... what you call low volume and what I call low volume are probably totally different things |
| 08:43.47 | ectospasm | jeffspeff: ^^^ |
| 08:44.00 | porche | max 2 concurrent channles |
| 08:44.19 | jeffspeff | ectospasm, i thought that might have happened with auto updates or something, so i recompiled (twice now) and still have error... i am assuming that when i recompile, it does over write the old files right??? |
| 08:44.21 | SwK | good luck on getting anyone to do ANI-II at that volume |
| 08:44.38 | ectospasm | jeffspeff: not unless you do a make clean or make distclean |
| 08:44.57 | porche | i see |
| 08:44.58 | ectospasm | I'd suggest a make distclean, to be sure everything gets recompiled |
| 08:45.08 | ectospasm | jeffspeff: then be sure to run ./configure |
| 08:45.10 | porche | what must be the volume to do that? |
| 08:45.30 | jeffspeff | ectospasm, where do i run the distclean command from? within the asterisk source dir? |
| 08:45.43 | ectospasm | jeffspeff: Asterisk src root |
| 08:46.39 | jeffspeff | ectospasm, ok, thanks... also, what is the command to stop asterisk? |
| 08:46.48 | ectospasm | jeffspeff: "stop now" in the CLI |
| 08:46.57 | jeffspeff | ok, thanks |
| 08:55.27 | *** part/#asterisk Odd_Bloke (n=oddbloke@libre.fm/user/oddbloke) |
| 08:57.00 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 09:01.35 | *** join/#asterisk Mw3 (i=mw3@2a01:270:dd00:300:0:0:0:42) |
| 09:01.55 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 09:03.30 | kippi | I am getting this error on 1.6.1.0 when voicemail trys to start undefined symbol: ast_smdi_mwi_message_destroy, is this a known bug? |
| 09:04.53 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 09:08.37 | adwerw | Hi! Is there any way to issue some commands on connection of SIP-user to asterisk server? For example - auto join to particular conference? |
| 09:09.16 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:09.34 | jeffspeff | adwerw, what do you mean by "auto join" |
| 09:10.25 | *** join/#asterisk joobie (n=joobie@124-168-19-171.dyn.iinet.net.au) |
| 09:10.59 | adwerw | I mean - as soon some user connected - he immediately join the conferece - wihout to have to enter anything |
| 09:11.21 | joobie | burp |
| 09:11.26 | dandre | hello, |
| 09:11.40 | dandre | I have this error in my sip debug trace: |
| 09:11.50 | dandre | Found no matching peer or user for 'IPADDRESS:8060' |
| 09:12.11 | dandre | My provider has 3 ip address |
| 09:12.18 | dandre | what can I do? |
| 09:12.44 | jeffspeff | adwerw, not that i know of... you'd have to connect to some extension in dialplan and that extension would go to conference |
| 09:13.12 | jeffspeff | dandre, i think that means it's not liking your username when you try to connect |
| 09:13.50 | *** part/#asterisk porche (n=kursad@88.239.77.171) |
| 09:14.34 | dandre | I am correctly registered but the ip address of the incomming call is different from the one I have registered |
| 09:15.30 | adwerw | jeffspeff: too bad... Are you sure? This is very unconvinient, I think |
| 09:15.38 | jeffspeff | have you tried registering with the IP of the incoming call? |
| 09:16.16 | jeffspeff | adwerw, all they have to do is pick up phone, dial 400 (example extension) and then they're in |
| 09:16.29 | SwK | dandre, you can do 2 things.. set up the default incoming user to route any incoming calls and route them into a context public access or create 3 sip peers for each ip of your carrier |
| 09:17.08 | pif | hi, how do priority labels work? if I have n(label) will the dialplan continue to the next 'n' pririty after that? |
| 09:17.15 | adwerw | you know, jeffspeff - some users are very busy and they'll forget everything |
| 09:17.57 | dandre | SwK: the first solution doen't meets my needs necause I must know where the call comes from |
| 09:18.08 | jeffspeff | adwerw, write it down on a post-it or program it to one of the feature buttons on the phone |
| 09:20.11 | jeffspeff | adwerw, charge them everytime they have to call you and ask, and see how quickly they remember. lol |
| 09:20.36 | jeffspeff | adwerw, that's implying that they can remember your number. :p |
| 09:20.46 | adwerw | yes - i could do that. but much more convinient - automatically add all users that i need to config , and then they'll come to office tomorrow - they automatically start breefing |
| 09:21.31 | adwerw | i think, that all user must be happy :) and me too :) |
| 09:22.06 | jeffspeff | adwerw, so are they doing nothing else with these phones but conferencing; and only conferencing with the same people? |
| 09:23.11 | adwerw | no - any phone have several channel =- and they can get incoming calls while they talking |
| 09:24.06 | jeffspeff | so how would they call somebody else if each time the phone is picked up they're connected to a conference? |
| 09:24.17 | adwerw | anyway - it's always good to make something technically, rather than administrativley |
| 09:26.30 | adwerw | and you sure that where is no way to automatically enter some extension on the dialplan, rigat after the user connected? |
| 09:26.50 | adwerw | s/rigat/right/ |
| 09:26.55 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 09:28.20 | jeffspeff | adwerw, something has to be dialed for the 'dialplan' to work... hence the name dialplan |
| 09:30.25 | adwerw | thank you , jeffspeff |
| 09:33.01 | adwerw | Another qu: How can I join to a conference only some SIP-authorized peolpe - and not askig h |
| 09:33.03 | adwerw | im to enter anothrer password? |
| 09:37.20 | *** join/#asterisk Sam2002gs (n=Sam2002g@h-213.61.105.202.host.de.colt.net) |
| 09:37.31 | Sam2002gs | Hello @all. Dose AsteriskNOw work fully on VMWareServers? |
| 09:47.42 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:48.02 | jeffspeff | SwK, is there a way to change the voicemail answer so that it doesn't say "comedian mail" ? |
| 09:55.49 | *** join/#asterisk mattbridges (n=mattbrid@77-98-133-25.cable.ubr07.stav.blueyonder.co.uk) |
| 09:57.37 | SwK | jeffspeff, yes change the soundfile |
| 10:04.54 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 10:09.44 | tompaw | morning |
| 10:10.10 | tompaw | is there any software to put on top of asterisk that will help trade voip? |
| 10:10.27 | tompaw | it would just have to manage dialplan, balance, display stats, usual stuff. |
| 10:12.21 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 10:21.32 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:24.28 | *** part/#asterisk cyd777_wrk (n=jbihari@bihari.eiszk.pte.hu) |
| 10:55.26 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:24.41 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 11:38.43 | ltd | Is there anything special about "*" in the dialplan? It seems to not work. |
| 11:42.11 | *** join/#asterisk FabiOne (n=FabiOne@151.13.190.14) |
| 11:42.15 | FabiOne | hi all |
| 11:42.51 | FabiOne | someone know where to find the mean of PHASESTATUS txfax value? |
| 11:44.46 | freh | ltd, how do you mean? |
| 11:45.02 | freh | what are you trying to do with "*" |
| 11:46.47 | ltd | freh: I've just got a dialplan entry like basically exten => *98,1,VoicemailMain |
| 11:46.56 | ltd | freh: But no matter what I do the bastard won't match... |
| 11:47.31 | ltd | freh: [Apr 30 21:47:16] NOTICE[30788]: chan_sip.c:14035 handle_request_invite: Call from 'linearg' to extension '*98' rejected because extension not found. |
| 11:50.08 | freh | ltd, so it works if you try without the * and just dial 98? |
| 11:50.58 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-3-57.wi.res.rr.com) |
| 11:50.59 | *** join/#asterisk eliel (n=eliels@200.61.172.61) |
| 11:55.55 | *** join/#asterisk Ryushin (i=proxy@71.33.251.73) |
| 11:55.59 | ltd | freh: Yep, works no problem. |
| 11:56.26 | *** join/#asterisk t3hrealadamd (n=t3hreala@65.215.34.250) |
| 11:58.08 | ltd | I recall running into this once before but I can't remember what the answer was. |
| 12:03.58 | freh | ltd, maybe try escaping it? |
| 12:04.14 | freh | like \*98 |
| 12:04.27 | freh | no promises though :-) |
| 12:04.47 | ltd | doesn't work, i've tried most things! |
| 12:05.07 | ltd | well, everything i can think of anyway. |
| 12:06.36 | freh | I see |
| 12:07.31 | beek | ltd: Are you sure that you're actually going into the right context? |
| 12:07.44 | beek | I use those * extensions with no problem. |
| 12:08.22 | ltd | 100%. If i change it to 98 instead of *98, it works no problem. |
| 12:08.36 | beek | Pastebin your dialplan |
| 12:11.17 | ltd | http://pastebin.com/m15465863 |
| 12:11.32 | ltd | that's the relevant details |
| 12:11.49 | *** join/#asterisk l0st-soul (n=lost@scylla.sysif.net) |
| 12:11.50 | ltd | service_15_outdial being the entry point of the sip peer |
| 12:13.02 | ltd | oh dear, I just noticed the problem. |
| 12:13.06 | ltd | the _X. |
| 12:13.09 | Titanous | Has anyone implemented Grandcentral/Google Voices style 'phone switching' in Asterisk? |
| 12:15.58 | beek | ltd: Kinda looks like it. |
| 12:16.39 | ltd | Will _. work fine? |
| 12:17.04 | beek | ltd: It should but I think Asterisk will issue a warning when that dialplan is loaded. |
| 12:17.23 | ltd | Is there a better way? |
| 12:18.20 | beek | ltd: Is all that crap in the service_15_outdial stuff really necessary if they're just going to dump into voicemail? |
| 12:18.35 | beek | If not, why not just add an extension to go directly to voicemailmain? |
| 12:18.57 | *** part/#asterisk AndyML (n=AndyML@pool-173-49-143-205.phlapa.fios.verizon.net) |
| 12:19.12 | beek | ltd: if it is, why not turn all of that into a subroutine or macro. |
| 12:19.45 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:19.56 | beek | morning [TK]D-Fender |
| 12:20.16 | [TK]D-Fender | beek: indeed |
| 12:22.23 | ltd | beek: I could clean it up a bit, i suppose |
| 12:22.40 | beek | ltd: It made my eyeballs bleed. |
| 12:23.35 | ltd | things get drastic in dialplanland when you need to get a job done |
| 12:24.00 | ltd | it's generated, but it could be condensed with a few Gosubs |
| 12:24.43 | beek | ltd: Either that, or add: exten => *98,1,VoicemailMain in the service_15_outdial context |
| 12:25.06 | beek | or include 'other' in that context |
| 12:25.33 | ltd | or _*.,Goto(other) |
| 12:26.01 | *** join/#asterisk macros73 (n=cs_@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 12:26.32 | *** part/#asterisk mattbridges (n=mattbrid@77-98-133-25.cable.ubr07.stav.blueyonder.co.uk) |
| 12:26.35 | *** join/#asterisk Vec (n=Vec@87.74.7.50) |
| 12:27.00 | beek | ltd, which would be yet another level of redirection. |
| 12:27.37 | Vec | Hi, I am trying to execute some code when an agent answers the phone: the call comes in I send them into the Queue(queuename) I then want to execute some code, in the dialplan or wherever once the agent answers that call ? Any ideas ? |
| 12:31.00 | [TK]D-Fender | Vec: How are your agents called? |
| 12:31.52 | Vec | [TK]D-Fender : from the Queue application |
| 12:32.11 | [TK]D-Fender | Vec: lets try another way, Show me your members list |
| 12:32.13 | Vec | they get put in a queue, the queue internally dials their extension (in asterisk source somewhere) |
| 12:33.45 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 12:33.56 | Vec | [TK]D-Fender : members list is > Agent/123, Agent/124, Agent/125 etc |
| 12:34.06 | [TK]D-Fender | Vec: Vec How do they log in? |
| 12:34.19 | Vec | [TK]D-Fender : AgentCallbackLogin() :p |
| 12:34.30 | Vec | which I know will no longer exist soon :O |
| 12:34.45 | [TK]D-Fender | Vec: Ok, then that goes through the dialplan to ring the agent device. Just use M() in your Dial command |
| 12:34.50 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 12:36.13 | Vec | [TK]D-Fender : hmm, ok so queue doesn't dial the SIP/exten directly ?? I thought it did ? if it does I can't use Dial with M() ? |
| 12:36.43 | [TK]D-Fender | Vec: AgentCallBackLogin points to DIALPLAN. How do you not know this? |
| 12:37.07 | [TK]D-Fender | Vec: Go look at an actual queue call |
| 12:37.49 | Vec | [TK]D-Fender : sweet thanks |
| 12:45.15 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.118.128) |
| 12:46.36 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 12:49.52 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:50.08 | pif | are asterisk regex perl-compatible? |
| 12:51.06 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:51.49 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 12:53.32 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:57.31 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:58.49 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 12:59.48 | ltd | pif: just posix afaik |
| 13:02.18 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 13:08.17 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:08.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.25 | jaytee | alot of AGI scripts included as examples and extras in the * tarball are written in perl |
| 13:12.24 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 13:16.35 | VaGoNeTaS | is back from the dead. Gone: 10h 3m 35s |
| 13:22.48 | zamba | is it possible to get timestamps on events in the asterisk console? |
| 13:25.40 | tzafrir_laptop | zamba, yes: use the logs |
| 13:25.54 | tzafrir_laptop | (you could also use the command-line option -t) |
| 13:30.27 | zamba | tzafrir_laptop: asterisk -rt? |
| 13:31.02 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:31.23 | tzafrir_laptop | zamba, no. This has to be added to the command-line of the asterisk server |
| 13:31.36 | zamba | oh |
| 13:31.54 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) |
| 13:32.01 | zamba | like params? -F -g -vvv? |
| 13:36.55 | *** join/#asterisk [gnubie] (i=patintin@119.56.59.7) |
| 13:37.16 | [gnubie] | waves |
| 13:37.49 | [gnubie] | can anyone point me where i can read asterisk 1.4.24.1 vs 1.6.1.0? |
| 13:39.17 | freh | Is there a way for queues to let members with a higher priority be called when no member with a lower priority picks up within x amount of time? |
| 13:44.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:48.16 | VaGoNeTaS | xD |
| 13:50.46 | *** join/#asterisk bgmarete (n=marebri_@196.201.210.130) |
| 13:51.49 | ltd | Is there any way to get rid of the DEBUG app_macro Executed application messages? |
| 13:54.45 | *** join/#asterisk dni (n=dniz0r@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 13:55.17 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 13:55.17 | [TK]D-Fender | [gnubie]: Go read CHANGES.txt in the tarball |
| 13:56.25 | dni | Hello Good Morning room,. I am getting one way audio when my sip peer who is a CCM tries calling out my asterisk server,. I have a pastebin of the sip debug,. Could someone take a quick look at it if you have a chance and see if you see something obvious.. http://pastebin.com/m24817f35 ,.. im able to dial out of their CCM fine but when they try and dial out of my * the one way audio issue happens |
| 13:56.41 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:56.48 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 14:02.28 | mort_gib | dni: Turn off your firewall for starters, or do the rules correctly :-) |
| 14:02.42 | dni | it is off |
| 14:02.44 | *** join/#asterisk peterthesing (n=peter@195-241-39-111.ip.telfort.nl) |
| 14:02.57 | dni | for now because of testing purposes |
| 14:02.57 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 14:03.12 | Stese | Hello all... me again |
| 14:03.12 | Stese | :P |
| 14:03.33 | peterthesing | hi All |
| 14:03.56 | Stese | Has anyone had any experience getting AMV Fritz ISDN cards working with mISDN? |
| 14:04.26 | peterthesing | is there someone who knows anything about the asterisk voipserver application? |
| 14:04.45 | peterthesing | sorry i do not own a fritzcard |
| 14:05.55 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:06.12 | Stese | Sorry, thats an AVM Fritz! Card |
| 14:06.43 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 14:07.10 | *** join/#asterisk Ryushin (i=proxy@71.33.251.73) |
| 14:07.43 | peterthesing | is it possible to connect to asterisk without the use of freepbx, am-portal udp-port 5060 |
| 14:08.05 | [TK]D-Fender | peterthesing: connect how? To do what? |
| 14:08.43 | [TK]D-Fender | peterthesing: FreePBX is a completely separate bolt-on GUI that controls all of your configs to fit its cookie-cutter view of a PBX |
| 14:09.20 | mort_gib | [TK]D-Fender: Well really all GUI's are built from assumptions, that their pitfall |
| 14:09.22 | peterthesing | lets say using a (java ) sipclient to connect from the outside |
| 14:09.32 | [TK]D-Fender | peterthesing: And I don't understand what you mix a question about a GUI interface, the CLI script to launch it, and the UDP port used by SIP in the same question |
| 14:09.58 | [TK]D-Fender | peterthesing: FreePBX only builds * configs for you. |
| 14:10.09 | [TK]D-Fender | peterthesing: It is by no means necessary |
| 14:10.15 | peterthesing | well i am somewhat new at this |
| 14:10.28 | [TK]D-Fender | peterthesing: Here : |
| 14:10.29 | [TK]D-Fender | ~book |
| 14:10.30 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:10.32 | [TK]D-Fender | ^^^^^ |
| 14:10.47 | peterthesing | i can connect directly from one cliet to the server |
| 14:11.08 | [TK]D-Fender | peterthesing: there is no "indirect". |
| 14:11.34 | [TK]D-Fender | peterthesing: a SIP device can register to *, or the reverse, and calls can be placed either way. Configuration is up to YOU. |
| 14:12.57 | peterthesing | i woul like to use a marvel called media proxy so it does not affect my current phone mappings on my modem using port 5060 |
| 14:13.54 | [TK]D-Fender | peterthesing: Phone mappings? |
| 14:14.24 | peterthesing | does book also include the openfire media proxy configuration? |
| 14:14.52 | peterthesing | i am currently using voip from my isp |
| 14:14.54 | [TK]D-Fender | peterthesing: No. |
| 14:15.43 | [TK]D-Fender | peterthesing: I would recommend setting your ISP's SIP account up on *. |
| 14:16.53 | peterthesing | there is some documentation for my zyxel adsl modem setting up 2 accounts |
| 14:17.41 | peterthesing | but i want firs to investigate how a mediaproxy and asterisk can interact (if at all possible?????) |
| 14:18.33 | Stese | peterthesing > I recommend the book that [TK]D-Fender mentioned |
| 14:19.05 | peterthesing | thanx for the advice i will read it asap |
| 14:20.50 | [TK]D-Fender | peterthesing: putting that ADSL VoIP ATA / Router in front of * is asking for trouble. |
| 14:21.53 | anonymouz666 | peterthesing: trust [TK]D-Fender on this one. he's the king of ~sipnat. |
| 14:22.20 | [TK]D-Fender | anonymouz666: You'd almost think there was a reason for that... |
| 14:25.57 | peterthesing | a quick glance tells me to make asterisk available for the outside world i need openser to make asterisk to work? |
| 14:26.59 | [TK]D-Fender | peterthesing: No |
| 14:28.23 | *** join/#asterisk t_ (i=tom@freenode/staff/tomaw) |
| 14:28.44 | peterthesing | then asterisk is capable of proxying? what port do assing an how do i tell i asterisk to listen to this proxy port |
| 14:29.00 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 14:30.24 | [TK]D-Fender | peterthesing: * is NOT a Proxy, it is a B2BUA. Go read the book for a bit |
| 14:30.42 | [TK]D-Fender | VaGoNeTaS: Please turn off your away script in here. |
| 14:31.09 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:33.45 | freh | So anyone an idea on how I can let queue members with I higher priority be called when queue members with a lower priority don't answer a call within x seconds? |
| 14:35.01 | *** join/#asterisk amaache (n=amma@80.249.75.230) |
| 14:35.22 | anonymouz666 | asterisk 1.4? |
| 14:35.24 | [TK]D-Fender | freh: Use multiple queues, the first with a timeout of "X". |
| 14:35.30 | amaache | Plz How to configure Cisco 7905 Help |
| 14:35.39 | anonymouz666 | there's a patch that works called xrrmemory. |
| 14:35.45 | anonymouz666 | do exactly what you want. |
| 14:36.00 | anonymouz666 | it's out of tree, but I use it in production servers. |
| 14:36.42 | amaache | Cisco 7905? |
| 14:39.04 | peterthesing | and now for something completely different >mISDN any ideas on using a analog phone quattrovox a billion isdn card (cologne chipset) calling a sip client? |
| 14:39.14 | *** join/#asterisk cm_5_1_2 (n=cm_5_1_2@58.252.229.215) |
| 14:40.06 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 14:40.24 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 14:40.46 | [TK]D-Fender | peterthesing: And now why are you talking an ANALOG phone in the same sentence as an ISDN card? One is analog, the other is digital. |
| 14:40.52 | *** part/#asterisk cm_5_1_2 (n=cm_5_1_2@58.252.229.215) |
| 14:41.22 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:41.43 | peterthesing | true, as i am told the quattrovox should translate this from analog to digital |
| 14:42.54 | *** part/#asterisk t (i=tom@freenode/staff/tomaw) |
| 14:43.23 | *** part/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) |
| 14:43.29 | [TK]D-Fender | peterthesing: Do you already have these devices? |
| 14:44.17 | peterthesing | i can see on the quattrovox when the phone is off hook but how to determine if asterisk notice this |
| 14:45.01 | peterthesing | using misdn i have set the isdn card to nt-mode |
| 14:45.50 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:47.11 | peterthesing | it is a quite old quattrovox I but i doubt if the isd card pick this up |
| 14:48.22 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 14:49.19 | peterthesing | in the bad old days of dos i could using a terminal to do an echo test to port X |
| 14:49.38 | peterthesing | i do not know how this is done in linux |
| 14:53.13 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 14:53.31 | Stese | peterthesing > on your asterisk CLI have a look at the options for misdn by typing help misdn |
| 14:54.17 | Stese | Anyone had any issues with a * box failing to boot when an AVM Fritz card is installed? |
| 14:55.21 | peterthesing | no such command misdn |
| 14:55.59 | Stese | peterthesing > Looks like mISDN isn't installed/running on your box |
| 14:57.45 | *** part/#asterisk Ng (n=cmsj@nurukipa.tenshu.net) |
| 14:58.56 | peterthesing | misdn is running but asterisk is configured for dahdi |
| 14:59.28 | peterthesing | dahdi is currently a dummy device |
| 14:59.51 | genin | i had to change the kernel of my debian box and it broke the dahdi modules |
| 14:59.52 | peterthesing | any thoughts on dahdi? |
| 14:59.57 | genin | so we had to recompile them |
| 15:00.03 | genin | but now the old modeuls are trying to load |
| 15:00.11 | genin | how do i remove those modules from startup? |
| 15:00.57 | Stese | peterthesing > if mISDN is running correctly, i would expect there to be commands for it in the Asterisk CLI |
| 15:01.29 | peterthesing | sorry there are none |
| 15:02.29 | peterthesing | how do i reconfigure the dahdi driver to use the billion isdn hardware? |
| 15:02.33 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 15:03.45 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 15:04.17 | Stese | No one with Avm Fritz experience around then... :( |
| 15:05.47 | amaache | Cisco 7905 does any one use it with Asterisk? |
| 15:06.37 | Stese | amaache> whats the issue? |
| 15:15.10 | tzafrir_laptop | peterthesing, you can use both dahdi and misdn together (as long as not with the same device...) |
| 15:15.13 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:19.36 | peterthesing | well dahdi is currenty a dummy device so it should logically not address any hardware |
| 15:28.38 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 15:29.02 | peterthesing | sorry i have to go thanks to all for the great help you guys have been wonderfull |
| 15:29.27 | peterthesing | for all those still fighting for answers goodluck |
| 15:30.07 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:33.35 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 15:34.22 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:35.48 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 15:36.12 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-cf6b61a777c11e51) |
| 15:36.27 | ariel_ | Hello folks. |
| 15:36.31 | *** join/#asterisk macros73 (n=cs_@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 15:37.10 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:41.45 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-155-49.telkomadsl.co.za) |
| 15:41.48 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 15:42.05 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:44.34 | *** join/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com) |
| 15:45.00 | genin | yo |
| 15:45.17 | genin | anyone know how i get modules to stop loading on startup with debian |
| 15:45.20 | genin | dahdi_dummy 5992 0 |
| 15:45.26 | genin | specifically dahdi |
| 15:45.46 | genin | it has to be redone because we had to change the kernel for bigmem |
| 15:45.51 | genin | i cant stop the old modules from loading |
| 15:46.06 | [TK]D-Fender | genin: recompile |
| 15:46.18 | genin | recompile dadhi? |
| 15:47.16 | [TK]D-Fender | genin: yes |
| 15:47.59 | genin | my admin did that and he said the old mods are still trying to load |
| 15:48.13 | genin | never says live |
| 15:48.14 | genin | just |
| 15:48.15 | genin | wct4xxp 295386 1 - Loading 0xf8bc2000 |
| 15:48.17 | ariel_ | genin: in debian you can use update RCD_XXXX remove to take it out |
| 15:48.17 | [TK]D-Fender | genin: I'd go prove that it was recompiled and installed |
| 15:48.20 | tryfan | genin: then they are in /lib/modules/<kernel> |
| 15:48.35 | genin | ok ill check |
| 15:48.47 | genin | what i pasted above was from cat /proc/modules |
| 15:48.48 | b14ck | genin /etc/asterisk/modules.conf |
| 15:48.52 | genin | ah okay |
| 15:48.57 | b14ck | add a noload => module_name.so |
| 15:49.02 | b14ck | then restart asterisk |
| 15:49.05 | genin | ill look in all those spots and try |
| 15:49.05 | b14ck | ;) |
| 15:50.25 | dkdkd | hi, can someone take a quick peek at this and tell me if it looks reasonable: http://pastebin.com/m4f5a2377 |
| 15:50.39 | [TK]D-Fender | b14ck: nope. |
| 15:50.48 | [TK]D-Fender | b14ck: DAHDI != * |
| 15:50.51 | dkdkd | i am not getting a response from SIP OPTIONS or SIP REGISTER, and i'm trying to figure out why |
| 15:51.03 | _brent_ | does anyone know, when asterisk receives a 302 (call forward) from a UA, how it decides if it's local or if it should spawn a call directly to the Contact: uri? |
| 15:51.10 | b14ck | whoops, i just heard 'dont load' and 'modules' |
| 15:51.18 | b14ck | ignore what i previously said, in that case |
| 15:51.44 | [TK]D-Fender | dkdkd: What have your forwarded to * exactly? |
| 15:52.05 | genin | heh |
| 15:52.19 | dkdkd | i am just trying to get REGISTER to succeed |
| 15:52.37 | [TK]D-Fender | _brent_: its always local direct to the dialplan. * is not a proxy, it is a B2BUA |
| 15:52.43 | [TK]D-Fender | dkdkd: What have your forwarded to * exactly? <--------- |
| 15:53.05 | dkdkd | i haven't forwarded anything to *. Asterisk is the UA sending a REGISTER to voicepulse |
| 15:53.08 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 15:53.41 | b14ck | anyone here use snom phones? |
| 15:53.44 | _brent_ | [TK]D-Fender: in the logs, i see asterisk spitting out: No such host: foo.bar.com |
| 15:53.47 | b14ck | i kinda wanna buy one, i wonder if they are any good |
| 15:53.51 | b14ck | they look cool ^^ |
| 15:54.04 | _brent_ | app_dial.c:524 in do_forward: Unable to create local channel for call forward to SIP/1001::::UDP@foo.bar.com |
| 15:54.17 | _brent_ | b14ck: i just got an 820 |
| 15:54.20 | _brent_ | it's pretty sweet |
| 15:54.32 | _brent_ | best screen i've seen on a phone yet |
| 15:54.41 | b14ck | ya? |
| 15:54.47 | b14ck | lemme look at that model real quick |
| 15:54.51 | _brent_ | yeah, sound quality is great, too |
| 15:54.51 | b14ck | they have blf support right? |
| 15:55.01 | mort_gib | b14ck: We use loads of Snoms |
| 15:55.07 | _brent_ | you mean presence monitoring? |
| 15:55.14 | mort_gib | b14ck: I think TK is starting to like them too |
| 15:55.23 | b14ck | neat! |
| 15:55.30 | b14ck | i always hear mixed things about them |
| 15:55.33 | b14ck | but they look awesome |
| 15:55.47 | b14ck | i havent owned any voip phones yet. but i work for a telephony company, ehh |
| 15:55.48 | b14ck | *heh |
| 15:55.55 | b14ck | i just bought my first phone aastra 57i ct yesterday :) |
| 15:56.25 | _brent_ | [TK]D-Fender: anyway, in the logs, it appears that * is trying to spawn an outgoing call, rather than keeping it local |
| 15:56.33 | ariel_ | b14ck: I have a few snom's 320,360 for a few years now. There great phones. |
| 15:57.07 | _brent_ | b14ck: i've found the snoms to be flakey if you have dirty power or network |
| 15:57.19 | b14ck | well i've got a good network |
| 15:57.21 | _brent_ | i don't know if the problem persists on the 8XX |
| 15:57.25 | b14ck | its just for my home for playing aorund with really |
| 15:58.24 | _brent_ | b14ck: the 870 (not out yet) will have a touch screen, too |
| 15:58.29 | b14ck | oO |
| 15:58.38 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:59.23 | b14ck | hey _brent_ |
| 15:59.32 | b14ck | i thikn the snom 870 is out, http://www.888voipstore.com/snom-870-pr-19725.html |
| 15:59.33 | mort_gib | b14ck: We use a mix of 300 and 370 handsets, users like them somewhat better than Polycoms because of the website |
| 15:59.46 | b14ck | how is the speakerphone support on the snoms? |
| 16:00.00 | mort_gib | b14ck: Not as good as IP 650 |
| 16:00.01 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 16:00.14 | b14ck | as long as its as good as the 500 im happy |
| 16:00.20 | _brent_ | maybe it is out! |
| 16:00.27 | mort_gib | b14ck: You have to play with Mic |
| 16:00.41 | ariel_ | b14ck: there ok in speaker, But polycom's are better in my view for speaker phones. I like the polycom's as a better biz phone. |
| 16:01.01 | b14ck | im not too worried about speaker. im just setting up a home system for myself to play aorund with |
| 16:01.14 | b14ck | i think im gonna put the aastra 57i in for my g/f |
| 16:01.14 | _brent_ | snom (company) isn't such a pain to deal with |
| 16:01.17 | b14ck | and use the snom for myself |
| 16:02.08 | _brent_ | you're right, though, polycoms sound the very best |
| 16:02.13 | _brent_ | but don't get me started about their UI |
| 16:02.22 | b14ck | at work i mainly use the polycoms |
| 16:02.24 | b14ck | the 501 |
| 16:02.35 | b14ck | but they look dull |
| 16:02.38 | _brent_ | (web UI, phone UI, corp web site, support site all suck) |
| 16:02.39 | b14ck | the aastra 57i looks awesome |
| 16:03.00 | coppice | polycom is the only company to take speakerphones seriously. balancing that, they have a bad track record for bugs :-) |
| 16:03.16 | b14ck | and they boot really slow :( |
| 16:03.16 | _brent_ | they don't even have a bug reporting system |
| 16:03.23 | _brent_ | only "feature requests" |
| 16:03.41 | coppice | they have issues :-) |
| 16:03.59 | coppice | issue, issue, all fall down |
| 16:04.06 | _brent_ | does it bug anyone else that the 'x' button on the 550/650/670 doesn't do anything except delete typed chars? |
| 16:04.16 | [TK]D-Fender | mortSnom? Nope. Too pricy, history of instability, audio doesn't measure up to Polycom, etc |
| 16:04.42 | [TK]D-Fender | b14ck: Boot time doesn't amtter so much when they simply don't crash. |
| 16:04.57 | _brent_ | the newer snom firmwares take just as long to boot |
| 16:05.14 | [TK]D-Fender | mine take a little under 2 min to boot |
| 16:05.47 | _brent_ | yeah, 2 minutes isn't bad if it's once a month |
| 16:05.53 | _brent_ | it's forever if you're developing |
| 16:05.53 | [TK]D-Fender | _brent_: I feel the same way about the 'Backspace" key on my keyboard! At long last a kindred spirit! |
| 16:06.05 | b14ck | man i really wanna get the snom 870 now |
| 16:06.09 | b14ck | i applied for a quote for one, looks badass |
| 16:06.22 | [TK]D-Fender | _brent_: What are developing that requires you to reboot the phone so often? |
| 16:06.29 | _brent_ | central provisioning |
| 16:07.11 | coppice | I hate turning on the TV and having to wait for Linux to boot :-\ |
| 16:07.27 | _brent_ | these guys had the 820 in stock: http://www.abptech.com/ |
| 16:07.39 | [TK]D-Fender | _brent_: For me the boot once out of the box and go right to a fully finished state within 5 minutes |
| 16:07.52 | b14ck | _brent_, ever use 888voip store? they sell phones really cheeap there |
| 16:08.03 | b14ck | unless u guys know of some better resellers ^^ |
| 16:08.14 | _brent_ | i get the bulk of my phones from ingram micro, but they don't carry snom or aastra |
| 16:09.11 | _brent_ | [TK]D-Fender: yeah, the boot time it fine for end users. it's just a royal pain when you're doing something that requires tweak -> reboot iterations |
| 16:09.25 | ariel_ | I use the polycom for work as it's very easy to deploy via the ftp and dhcp settings. Just edit files save them and plug phones in. |
| 16:09.58 | b14ck | cant you deploy snoms like that too though? |
| 16:10.02 | b14ck | just put the firmware onto a tftp server |
| 16:10.05 | b14ck | and let it go? oo |
| 16:10.08 | _brent_ | yeah, you can do http, too |
| 16:10.25 | _brent_ | better in many regards than tftp |
| 16:10.28 | [TK]D-Fender | _brent_: Tweaking is for people who don't get it right the first time ;) |
| 16:10.48 | _brent_ | :-) |
| 16:11.01 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 16:11.26 | _brent_ | i'm in awe that you could get all of polycom's thousands of cascading parameters right the first time ;-) |
| 16:11.31 | _brent_ | you're smarter than me |
| 16:11.59 | [TK]D-Fender | _brent_: specialized in them long ago. |
| 16:12.10 | [TK]D-Fender | _brent_: Everything comes with a price. |
| 16:12.26 | _brent_ | yeah, i've paid that price and now i have a central provisioning system |
| 16:12.37 | _brent_ | but i cursed polycom's slow boot for a couple of months |
| 16:12.41 | b14ck | central provisioning system? oo |
| 16:12.53 | b14ck | you mean for all your clients they pull configs from one server? |
| 16:12.56 | _brent_ | yeah |
| 16:13.06 | b14ck | thats a nice way to do version control |
| 16:13.35 | _brent_ | yeah, and it's not polycom specific. all the phones grab configs & firmware from the same system |
| 16:14.04 | b14ck | ye |
| 16:18.33 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 16:19.13 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:19.45 | _brent_ | so ... does anyone have any ideas about this call forwarding? [TK]D-Fender? |
| 16:19.58 | b14ck | what call forwarding? |
| 16:20.07 | mesfet | Is there anybody who can help me setting PSTN parameters for Bulgaria POTS ? I don't know how to set busypattern and signalling. |
| 16:20.12 | [TK]D-Fender | _brent_: pastebin a complete call with SIP debug, etc. |
| 16:20.16 | [TK]D-Fender | ~pb |
| 16:20.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 16:20.17 | [TK]D-Fender | ^^^^6 |
| 16:20.41 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 16:21.08 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 16:21.48 | *** join/#asterisk skanker (n=gonbanan@202.128.43.187) |
| 16:23.04 | _brent_ | http://pastebin.com/d1ca460b3 |
| 16:23.34 | [TK]D-Fender | _brent_: what part of "COMPLETE call with SIP DEBUG" did you miss? |
| 16:23.34 | _brent_ | i see both "no such host" and "unable to create local channel" |
| 16:23.59 | _brent_ | ah, the SIP DEBUG part ;-) |
| 16:24.04 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 16:24.52 | *** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net) |
| 16:24.54 | voxter | cesar_CR: hey man, hows it going |
| 16:25.08 | hardwire | I have an asterisk system hitting a high load every so often.. can't seem to get enough logs to tell me whats going wrong. |
| 16:25.23 | hardwire | how should I execute asterisk to get a good debug dump of the running processes? |
| 16:26.10 | cesar_CR | voxter, ? fine :) |
| 16:26.40 | voxter | cesar_CR: im not sure if you remember, we talked before. i might have been using my other name, [hC] then. |
| 16:27.31 | cesar_CR | [hC] yes now I remember!!! hi man!!! |
| 16:27.49 | cesar_CR | voxter, everithing OK ? |
| 16:28.37 | voxter | cesar_CR: yeah its great. infact i am flying to CR tonight, and im going to be spending most of my time there again |
| 16:28.43 | voxter | cesar_CR: how about you? |
| 16:29.58 | cesar_CR | voxter, well I am here like allways : ) |
| 16:31.13 | cesar_CR | voxter, working, let me know if you need some help here, you have my email right ? how long are you staying ? |
| 16:31.19 | voxter | cesar_CR: we should get together some time, i will probably be getting into some different business ideas there and what not |
| 16:31.46 | *** join/#asterisk skanker (n=gonbanan@202.128.43.187) |
| 16:31.48 | voxter | cesar_CR: private message me your email again just incase. Ill be there for 3 weeks first, then probably come back to canada for 4-5 days then back to CR again |
| 16:33.37 | _brent_ | http://pastebin.com/m6ee087eb |
| 16:33.57 | _brent_ | it's a tcpdump of the sip traffic, not an * CLI sip debug |
| 16:34.05 | *** join/#asterisk skanker (n=gonbanan@202.128.43.187) |
| 16:34.15 | *** join/#asterisk abchirk (n=rapunzel@cl-2502.ham-01.de.sixxs.net) |
| 16:34.39 | SparFux | It is said that the sporadic DTMF tones problem is due to mISDN, it is a well known phenomenom. But the point is, it only occurs with my ATA and the analog phone, not with other lines and softphones. This is really strange. |
| 16:35.23 | coppice | sporadic and mISDN go together like crackers and cheese |
| 16:36.02 | *** join/#asterisk Globettrotter (n=eric@ool-457a1c8a.dyn.optonline.net) |
| 16:37.26 | [TK]D-Fender | _brent_>it's a tcpdump of the sip traffic, not an * CLI sip debug <- I care what * tinks, not jsut a failure packet. |
| 16:38.24 | _brent_ | ok, one more time |
| 16:41.27 | PinkFreud | [TK]D-Fender: I see that asterisk gui 2.0 requires at least asterisk 1.6.0. Do you know the minimum release version for gui 1.0? |
| 16:41.49 | [TK]D-Fender | PinkFreud: got a problem with 1.6.0+? |
| 16:42.21 | PinkFreud | not in particular. I'm still waffling on whether I want to go with official distro-supported asterisk or 1.6.0.9 |
| 16:42.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 16:42.30 | PinkFreud | just figuring out my options right now :) |
| 16:42.39 | [TK]D-Fender | PinkFreud: What distro? |
| 16:43.01 | PinkFreud | currently, centos. considering pushing for a change to debian lenny. |
| 16:43.56 | PinkFreud | I see CDRs are still listed as beta in gui 2.0, alas. |
| 16:49.32 | PinkFreud | hmm |
| 16:54.07 | [TK]D-Fender | PinkFreud: Upgrade and compile it yourself |
| 16:55.00 | _brent_ | [TK]D-Fender: http://pastebin.com/d53c70ad |
| 16:57.08 | abchirk | Hi, I am trying to get a asterisk working on a LAN only between my and my friend, where to start at sip? |
| 16:58.35 | PinkFreud | [TK]D-Fender: indeed. however, I'm still curious about the status of gui 1.0 - does that work with 1.4.21? |
| 16:58.37 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 16:59.04 | PinkFreud | if so, it *may* be worthwhile to use distribution-provided binaries with 1.0 |
| 16:59.18 | [TK]D-Fender | PinkFreud: Yes, but I wouldn't recommend it |
| 16:59.20 | PinkFreud | again, I'm looking at all of my options here. |
| 16:59.24 | PinkFreud | [TK]D-Fender: ahhh . why not? |
| 17:00.01 | [TK]D-Fender | PinkFreud: there's a reason for the big update. I was a giant flaming turd before. Now they've put out the open flames . |
| 17:00.04 | *** part/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 17:00.15 | [TK]D-Fender | PinkFreud: 1.0 is not being maintained at all AFAICT |
| 17:00.45 | [TK]D-Fender | PinkFreud: Better option : use Digium's RPM repo |
| 17:03.44 | _brent_ | [TK]D-Fender: was that last paste more useful? anything look abnormal? |
| 17:04.23 | [TK]D-Fender | _brent_: well you masked ip/hosts, and I'm grey on the "::::" parsing. what does the phone say to forward to? |
| 17:04.36 | _brent_ | 1001 |
| 17:04.45 | _brent_ | the :::: looked curious to me, too |
| 17:05.03 | [TK]D-Fender | _brent_: got a 1001 in the target context? Because I use these just fine for that here |
| 17:05.18 | PinkFreud | [TK]D-Fender: does digium provide a .deb repo? |
| 17:05.38 | _brent_ | i have a catchall that hits AGI in the target context, so yes, 1001 is covered there |
| 17:05.45 | [TK]D-Fender | PinkFreud: Not AFAIK |
| 17:05.57 | [TK]D-Fender | PinkFreud: I still recommend CentOS for this |
| 17:06.05 | Qwell | seanbright: :( |
| 17:06.08 | PinkFreud | [TK]D-Fender: do you mind if I ask why? |
| 17:06.09 | [TK]D-Fender | _brent_: :/ |
| 17:06.49 | _brent_ | [TK]D-Fender: thanks for your help. i'll keep picking away at this |
| 17:06.58 | [TK]D-Fender | _brent_: Alrighty |
| 17:07.13 | [TK]D-Fender | abchirk: ... |
| 17:07.15 | [TK]D-Fender | ~book |
| 17:07.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:07.16 | [TK]D-Fender | ^^^^^^^ |
| 17:07.18 | PinkFreud | I'm comformtable with compling software. That being said, though, I usually prefer to stick with distro-provided software where possible, both for manageability purposes, and for the support provided by the distribution. |
| 17:08.17 | [TK]D-Fender | abchirk: Need to learn how to configure SIP peers and a basic dialplan. use this for some "inspiration" in your learning process : |
| 17:08.20 | [TK]D-Fender | ~jerjerguide |
| 17:08.20 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:08.37 | [TK]D-Fender | PinkFreud: then use centOS + Digium's repo |
| 17:08.39 | PinkFreud | I've used CentOS and Debian fairly extensively, but tend to prefer Debian, due to a larger software repository. We're also in the middle of switching our infrastructure to Debian, so I prefer to keep all of our systems, including the eventual phone system, on a single platform, if possible. |
| 17:08.55 | abchirk | [TK]D-Fender thank you... but do I have to use a real phone? I just want to use my mic from my laptop |
| 17:08.59 | SparFux | I want to use asterisk dtmf detection stuff, not mISDN. how can I turn the mISDN part off? |
| 17:09.09 | [TK]D-Fender | PinkFreud: I'v seen glaciers move faster than Debian's dev team... |
| 17:09.19 | [TK]D-Fender | abchirk: No. |
| 17:09.23 | [TK]D-Fender | ~softphone |
| 17:09.23 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 17:09.24 | [TK]D-Fender | ^^^^ |
| 17:09.47 | abchirk | ah ok thank you. :) |
| 17:10.20 | PinkFreud | [TK]D-Fender: debian releases can be few and far between - yet, CentOS (and RH, of course) tend to hang on to older package releases |
| 17:10.54 | PinkFreud | not to mention that etch was only a year and a half old when lenny finally made release. that was pretty quick turnaround for debian. :P |
| 17:11.07 | [TK]D-Fender | PinkFreud: yes, and I have specifically directed you to use Digium's repo's to maintain yourself of "complete from vendor" |
| 17:11.12 | abchirk | eh but I haven't a VoIP-provider.. so I have to "simulate" on with asterisk? |
| 17:11.32 | PinkFreud | [TK]D-Fender: indeed you have. |
| 17:11.39 | [TK]D-Fender | PinkFreud: Do you buy Ford coffee just because you use the cup-holder in your car? |
| 17:12.03 | PinkFreud | first off, I don't drive Ford. Secondly, - can you buy Ford coffee? :P |
| 17:12.13 | [TK]D-Fender | abchirk: * is a complete PBX & telephony toolkit. It can easily do what you want |
| 17:12.19 | [TK]D-Fender | abchirk: Adding more is just... more. |
| 17:12.22 | PinkFreud | thirdly, I'm not sure I'd expect - or want! - support from Ford for my coffee. |
| 17:12.32 | [TK]D-Fender | abchirk: Your dialplan may only have a dozen lines in it if even. |
| 17:12.48 | [TK]D-Fender | PinkFreud: double-double :) |
| 17:13.08 | PinkFreud | I don't call Dunkin Donuts for support for my coffee, either. |
| 17:13.12 | PinkFreud | :) |
| 17:13.24 | PinkFreud | (uh, hi, how do I drink this?) |
| 17:14.26 | [TK]D-Fender | PinkFreud: Careful of you'll soon find there are complete instructions on the cup alongside the "YES dumbass, this is full of hot liquid that will burn you if you're an idiot" warning. |
| 17:14.37 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:14.52 | PinkFreud | [TK]D-Fender: sadly, that's already the case. |
| 17:14.59 | [TK]D-Fender | kicks McD's lawyers for failing to grow a pair. |
| 17:15.28 | [TK]D-Fender | wonders where he should kick now... the quick & easy target is gone... |
| 17:15.30 | abchirk | hm well, this all sounds like a big elefant...... :P isn't there a easier way to connect me and my friend through voice over lan in a console? :P |
| 17:15.50 | [TK]D-Fender | abchirk: Load up Ekiga on both and jsut dial the IP |
| 17:16.22 | [TK]D-Fender | abchirk: ~ekiga |
| 17:16.27 | [TK]D-Fender | ~ekiga |
| 17:16.27 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 17:16.37 | abchirk | yeah found it on aptitude :) |
| 17:16.51 | _brent_ | "GA release [for the SNOM 870] is scheduled for Q3, 2009." :-( |
| 17:16.55 | [TK]D-Fender | abchirk: Excellent |
| 17:17.04 | abchirk | ehe uff |
| 17:17.58 | [TK]D-Fender | _brent_: Never seen a point to buy Snom in the US |
| 17:18.31 | _brent_ | i just want to touch the screen on that one... |
| 17:20.22 | b14ck | okok im gonna get the snom 870 |
| 17:20.25 | b14ck | it is too cool |
| 17:20.31 | b14ck | big color LCD with touch screen! |
| 17:20.32 | b14ck | <3 |
| 17:20.44 | SparFux | Why is mISDN so crappy? |
| 17:21.05 | _brent_ | cuz it sounds too much like MSDN |
| 17:23.10 | coppice | mISDN keeps changing its name. there is a reason for this :-) |
| 17:24.32 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 17:26.14 | mmlj4 | how do I jump contexts? specifically, I want to be able to call * and authenticate via a PIN perhaps, then be allowed to dial further |
| 17:27.15 | [TK]D-Fender | mmlj4: "core show application authenticate", and any boring little IVR context. |
| 17:27.37 | [TK]D-Fender | mmlj4: could use DISA as well |
| 17:27.48 | [TK]D-Fender | mmlj4: "core show application disa" |
| 17:27.54 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 17:28.27 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:32.24 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 17:32.24 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.0 (2009/04/28), Asterisk 1.6.0.9 (2009/04/06), 1.4.24.1 (2009/04/02), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.1 (2008/12/02), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 17:34.47 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 17:38.09 | *** join/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) |
| 17:39.23 | jplank | before I reinvent the wheel, does anyone know of a tapi program that with integrate with outlook, and allow a "click to call" option. Basically bridging the end-users phone with the number they are trying to call, I'm thinking someone has had to do it before |
| 17:50.42 | *** join/#asterisk tyfon (n=tyfon@master.robotmafia.org) |
| 17:50.43 | *** join/#asterisk wilsonj (n=jeremy@unaffiliated/dethstar) |
| 17:53.00 | SuPrSluG | jplank:http://www.voip-info.org/wiki/view/Asterisk+TAPI |
| 17:54.19 | mocker | God damn lightening strikes taking out my PRI. |
| 17:55.05 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 17:55.19 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:55.23 | Corydon76-dig | mocker: add more lightning rods |
| 17:55.45 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 17:55.52 | Corydon76-dig | You can't repel lightning, but you can conduct it down a safer path |
| 17:57.16 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 17:58.05 | jaytee | jplank, check this out. I've used it with Outlook and * and it works well http://www.ipcom.at/index.php?id=561 |
| 17:58.06 | grandpapadot | Hey guys, on Polycom phones, is there a way (prior to the 3.1 Enhanced Feature Keys licensed feature), to change the function of one of the hard buttons? They come with "blanks" but I can't find a way to re-assign button actions/dialing strings/etc. |
| 17:59.34 | mocker | Corydon76-dig: Tell that to AT&T. :) |
| 18:00.10 | pmhaddad-work | so i have a question for those of you who have the dCAP: what are some good study strategies? I know what areas I need to study for the most (I think), but what are some good ways to study for it? Should I be working on implementing dialplans etc, or should I focus more on conceptual information, or both? |
| 18:02.03 | pmhaddad-work | grandpapadot, depeding on the model it should be under the Lines section from the Admin panel webpage that you get when browsing to the phone's IP |
| 18:02.16 | pmhaddad-work | somewhere near the bottom IIRC |
| 18:02.20 | [TK]D-Fender | grandpapadot: Yes, you've been able to remap those for a long time prior |
| 18:02.23 | jplank | jaytee: that looks perfect, question though, because it uses a refer message instead of connecting to the AMI, how do the CDR's look? |
| 18:02.46 | [TK]D-Fender | grandpapadot: And that is done in provisioning. Well documented in the admin guide. |
| 18:03.11 | grandpapadot | Thanks, tk. |
| 18:07.15 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 18:07.47 | *** join/#asterisk t_corr (n=Richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:08.08 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-45292b9daf9c0872) |
| 18:11.40 | SuPrSluG | is there a way to lower the default time out for the polcom phones so they don't take forever to boot when the can't upload their log files |
| 18:15.57 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 18:16.01 | mocker | pmhaddad-work: Make sure you can build a working PBX from scratch in less than 30 minutes with a variety of hardware. |
| 18:16.18 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:16.25 | *** join/#asterisk areay (n=areay@93-97-161-123.zone5.bethere.co.uk) |
| 18:16.58 | pmhaddad-work | mocker, ok, that's a good start |
| 18:17.07 | pmhaddad-work | i've done that a lot with a lot of hardware |
| 18:17.18 | pmhaddad-work | i assume they have linux already installed right? |
| 18:17.26 | mocker | pmhaddad-work: by different hardware I mean T1 card, FXO/FXS |
| 18:17.40 | mocker | Yes, I they shouldn't test you on a linux install. :) |
| 18:17.44 | pmhaddad-work | :) |
| 18:18.05 | pmhaddad-work | ok, i've done several asterisk installs with digium and rhino hardware |
| 18:18.17 | pmhaddad-work | mostly fxo cards though |
| 18:18.29 | *** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 18:18.33 | DavidR2008 | jaytee: do you have any information on how to configure SIP TAPI for use with outlook? I'm playing around with it and I'm not really sure what it's trying to do, but it's not working correctly, that much I do know :-) |
| 18:19.09 | grandpapadot | Hey TK, I got a key reassigned, no problem, easy. Do you know of a way to set a key to "transfer" to a specific extension (keycode), I'm trying to map a park feature. |
| 18:19.09 | mocker | pmhaddad-work: I'm only certified against 1.2, so there may have been changes. |
| 18:19.26 | mocker | But if you can build a fully functioning (make calls) pbx in 30 minutes that's a good start. :) |
| 18:19.42 | pmhaddad-work | mocker, ok man, that's a huge help |
| 18:19.52 | pmhaddad-work | has his exam a week from monday |
| 18:20.05 | pmhaddad-work | i haven't actually tried timing myself yet |
| 18:20.05 | mocker | Also make sure you know the codec stuff.. |
| 18:20.41 | jeffspeff | how can i set the caller id for individual extensions? http://pastebin.ca/1408709 <-- extensions.conf |
| 18:20.42 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 18:20.51 | pmhaddad-work | mocker, like g711 etc? |
| 18:20.56 | jaytee | jplank, sorry I had to take a call from Time Warner Telecom. As far as the CDR goes, if I use my Outlook Contacts to click to call it just looks like my actual Polycom phone called the number in the contact file |
| 18:21.05 | mocker | pmhaddad-work: Yup. |
| 18:21.07 | jaytee | DavidR2008, hang on a second for your TAPI issue |
| 18:21.16 | pmhaddad-work | mocker, ok sweet |
| 18:21.22 | [TK]D-Fender | grandpapadot: No way to do this |
| 18:21.24 | mocker | pmhaddad-work: Good luck man. |
| 18:21.26 | pmhaddad-work | mocker, is there a lot of PSTN stuff discussed on there? |
| 18:21.32 | pmhaddad-work | i'm a bit weak on all of that |
| 18:21.40 | grandpapadot | tk, that's what I thought, worth a shot. Back to inline with it then. Thanks! ;p |
| 18:21.47 | [TK]D-Fender | mocker: 30 MINUTEs? Geez |
| 18:21.58 | mocker | pmhaddad-work: Eh, not that I remember. |
| 18:22.13 | pmhaddad-work | yeah, i probably can't do it in 30 right now |
| 18:22.21 | pmhaddad-work | i'm probably sitting around an hour |
| 18:22.35 | pmhaddad-work | depending on the hardware maybe 45 minutes |
| 18:22.51 | mocker | I don't remember how long they actually gave for the practical and I can't find anything on the web. |
| 18:23.06 | pmhaddad-work | mocker, yeah they never told me |
| 18:23.20 | pmhaddad-work | i know i start at 9:30 and they expect me to finish around 2pm |
| 18:23.21 | mocker | Ohh, looks like 90 minutes. |
| 18:23.22 | *** join/#asterisk jtodd (n=jtodd@131.sub-75-218-106.myvzw.com) |
| 18:23.23 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:23.25 | mocker | for the practical. |
| 18:23.29 | mocker | http://ircarchive.info/asterisk/2007/3/28/50.html |
| 18:23.34 | mocker | That should be easy. |
| 18:23.39 | pmhaddad-work | mocker, that's just one pbx build right? |
| 18:23.43 | pmhaddad-work | i don't have to do like 4 |
| 18:24.24 | jaytee | DavidR2008, for configuring the SIPTAPI provider in Windows Control Panel under Phone and Modem applet just find it in the list and click the configure button. put your * server's FQDN or IP address in the SIP Domain field, leave the outbound proxy blank, add your SIP account and password and extension in the appropriate fields. |
| 18:24.26 | mocker | pmhaddad-work: Hah, no. |
| 18:24.34 | pmhaddad-work | mocker, whewie |
| 18:24.41 | pmhaddad-work | i'm just a bit nervous about it |
| 18:24.47 | pmhaddad-work | goin all the way down there and such |
| 18:24.55 | pmhaddad-work | it would be a real downer if i failed it |
| 18:25.06 | pmhaddad-work | mocker, is there a lot of dialplan stuff covered? |
| 18:25.21 | mocker | Just make sure you can do the hardware stuff w/o googling. |
| 18:25.33 | [TK]D-Fender | pmhaddad-work: There'd have to be. Dialplan = * |
| 18:25.45 | mocker | But how hard really is the dialplan? |
| 18:25.59 | DavidR2008 | jaytee: sorry, I know I hijacked you're thread, and if you don't have time that's fine, but it looked interesting. I got the SIPTAPI configured, saw how to do that on the website, I guess my question is more on the * side: it ring's my phone but as soon as I answer it hangs up. Trying to figure out what to troubleshoot |
| 18:26.08 | pmhaddad-work | [TK]D-Fender, i should have said "advanced" dialplan stuff... like AGI and such |
| 18:26.09 | mocker | . o O ( It's easy until it doesn't work! ) |
| 18:26.24 | jaytee | pmhaddad-work, use the WIKI search tool on voip-info.org to search for Asterisk+sample+configuration and also look at the stuff on the WIKI for a simple IVR with Day/Night mode using GotoIfTime |
| 18:26.41 | [TK]D-Fender | pmhaddad-work: In the practical side, I'd bet on "no" |
| 18:26.47 | areay | i set up my asterisk server about a month ago, but didn't implement it because i wasn't ready... i was doing some testing today, and the incoming trunk works fine, but for some reason I can't get any sip client on my local network to connect as a phone... i have set sip debug on, and every time i try to register i see NOTHING at all. i don't understand why it's stopped working because i haven't changed the configuration files for asteri |
| 18:26.47 | areay | sk at all since they worked |
| 18:26.56 | pmhaddad-work | jaytee, awesome thanks |
| 18:27.02 | Qwell | seanbright: fix it! :( |
| 18:27.10 | [TK]D-Fender | pmhaddad-work: They shouldn't test you on the assumption of knowing another programming language |
| 18:27.17 | jaytee | DavidR2008, what kind of phone? |
| 18:27.28 | pmhaddad-work | [TK]D-Fender, that makes sense ya |
| 18:27.32 | DavidR2008 | Grandstream gxp 2000 |
| 18:27.52 | [TK]D-Fender | areay: Seeing noting either means the phones aren't pointed towards the server, or you have a firewall/networking probelm |
| 18:28.07 | jaytee | DavidR2008, in your sip.conf file for that account make sure you allow reinvites with canreinvite=yes |
| 18:28.17 | mocker | wonders if it's dahdi in the test or zaptel nowadays. |
| 18:29.00 | pmhaddad-work | i've used both |
| 18:29.18 | pmhaddad-work | mocker, when did you take it? |
| 18:29.32 | mocker | God, awhile ago. |
| 18:29.32 | jaytee | I took the class and the test in November in Huntsville and they use Dahdi and 1.6 in the class and in the test |
| 18:29.42 | mocker | Back when sokol and associates still existed and gave boot camps. |
| 18:29.49 | pmhaddad-work | mocker, i hear its become a bit easier |
| 18:29.57 | mocker | pmhaddad-work: Rock! |
| 18:30.01 | mocker | Gives mine more creedence. :) |
| 18:30.07 | pmhaddad-work | lol |
| 18:30.34 | areay | [TK]D-Fender, i know... which i don't get... it's a local network, so there's no NAT, and i've uninstalled the firewalls on the server and client as a precaution... i installed a dns server (bind9) the other day but that shouldn't have any effect on it, should it? |
| 18:31.57 | *** join/#asterisk EUSEricDCAP (n=chatzill@ip67-152-18-226.z18-152-67.customer.algx.net) |
| 18:32.28 | EUSEricDCAP | Hey guys. |
| 18:32.30 | EUSEricDCAP | What's up |
| 18:33.03 | mocker | pmhaddad-work: Looks like EUSE is a dcap too. :) |
| 18:33.16 | mocker | maybe more recent |
| 18:33.20 | EUSEricDCAP | Yeah, last june |
| 18:33.45 | EUS-Eric-DCAP | Might clarify it |
| 18:33.47 | EUS-Eric-DCAP | heh. |
| 18:33.57 | [TK]D-Fender | areay: Go prove there is no FW in the way and that * is running and listening at all. |
| 18:34.08 | mocker | EUS-Eric-DCAP: You work at EUS? |
| 18:34.10 | EUS-Eric-DCAP | yeah |
| 18:34.24 | EUS-Eric-DCAP | I'm there right now. |
| 18:34.27 | mocker | I met Jeronimo a couple years ago at Astricon. |
| 18:34.30 | EUS-Eric-DCAP | Nice. |
| 18:34.32 | mocker | Nice guy. |
| 18:34.41 | EUS-Eric-DCAP | Yeah, he's cool. |
| 18:34.47 | Qwell | what's an EUS? |
| 18:34.50 | EUS-Eric-DCAP | I'm baing my head against this insane problem. |
| 18:35.05 | EUS-Eric-DCAP | Only the best Asterisk shop in the world. |
| 18:35.06 | Qwell | oh |
| 18:35.10 | Qwell | EUS...right. |
| 18:35.19 | *** part/#asterisk Thiago_Lima (n=chatzill@200.159.31.7) |
| 18:36.01 | EUS-Eric-DCAP | anyone want to take a crack at this insane BLF/hint problem I'm ready to hang myself over? |
| 18:36.25 | [TK]D-Fender | EUS-Eric-DCAP: don't ask to ask, just spit it out :) |
| 18:36.30 | [TK]D-Fender | EUS-Eric-DCAP: pastebin is your friend... |
| 18:36.32 | [TK]D-Fender | ~pb |
| 18:36.33 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 18:36.47 | EUS-Eric-DCAP | heheh |
| 18:36.49 | EUS-Eric-DCAP | ok. |
| 18:36.53 | EUS-Eric-DCAP | So this client has a very small network. |
| 18:37.06 | EUS-Eric-DCAP | Seven phones, * pbx, one switch and firewall. |
| 18:37.08 | EUS-Eric-DCAP | Nothing fancy. |
| 18:37.14 | EUS-Eric-DCAP | C.1.8.1 |
| 18:37.26 | EUS-Eric-DCAP | Now, when they get a call to their main number, it's set to ring all seven phones. |
| 18:37.33 | EUS-Eric-DCAP | the phones are aastra 35i's. |
| 18:37.47 | EUS-Eric-DCAP | Each phone has a BLFs for the other 6 phones |
| 18:38.05 | EUS-Eric-DCAP | When a call rings all seven phones, they all go "state ringing" on both the console and the phones |
| 18:38.20 | eppigy | DONDE ESTA |
| 18:38.25 | EUS-Eric-DCAP | When the call is answered, and six of the phones go back idle, the go idle on the console, but not on the phones |
| 18:38.30 | EUS-Eric-DCAP | they stay off hook. |
| 18:38.40 | EUS-Eric-DCAP | Now, I see this on the console, which I've never seen before. |
| 18:38.42 | EUS-Eric-DCAP | : |
| 18:39.03 | EUS-Eric-DCAP | Extension Changed 16[local-extensions] new state Ringing for Notify User 11 (queued) |
| 18:39.20 | EUS-Eric-DCAP | It seems the BLF updates are being queued or something. |
| 18:39.31 | EUS-Eric-DCAP | I know BLFs and DNS are related, so maybe the PBX has a DNS problem? |
| 18:39.44 | EUS-Eric-DCAP | The PBX runs BIND and all the phones point at it for DNS. |
| 18:41.00 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 18:41.34 | EUS-Eric-DCAP | if I call a phone directly from another phone, the hints work find and the blf indicators update properly |
| 18:41.36 | EUS-Eric-DCAP | any ideas? |
| 18:42.58 | [TK]D-Fender | EUS-Eric-DCAP: What ver of *? |
| 18:43.06 | EUS-Eric-DCAP | C.1.8.1 |
| 18:43.15 | [TK]D-Fender | oh.. ABE... |
| 18:43.17 | EUS-Eric-DCAP | BE |
| 18:43.19 | EUS-Eric-DCAP | yeah. |
| 18:43.28 | EUS-Eric-DCAP | Is this the wrong place for talk of BE? |
| 18:43.43 | [TK]D-Fender | EUS-Eric-DCAP: sometimes.. not sure on yourse. |
| 18:44.07 | [TK]D-Fender | EUS-Eric-DCAP: You're doing a basic Dial() with multiple people? |
| 18:44.39 | EUS-Eric-DCAP | it's running thirdlane, but it's pretty much doing a basic dial. |
| 18:44.56 | [TK]D-Fender | EUS-Eric-DCAP: and 7 phones being called should be 6x6 updates = 36 per state change |
| 18:45.11 | [TK]D-Fender | (max) |
| 18:45.50 | EUS-Eric-DCAP | Yeah, that makes sense |
| 18:45.54 | DavidR2008 | jaytee: turns out outlook was only sending phone without area code and * only accepts numbers with the area code, problem solved. Thanks for mentioning this, it's pretty cool! |
| 18:46.19 | EUS-Eric-DCAP | You can change your dialplan to allow dialing with out area code |
| 18:46.21 | *** join/#asterisk hi365 (n=hi365@94.159.178.139) |
| 18:46.26 | [TK]D-Fender | DavidR2008: * can accept any # you tell it to :) |
| 18:46.48 | DavidR2008 | I should have said: I configured * to .... |
| 18:47.00 | DavidR2008 | appologies ;-) |
| 18:48.45 | EUS-Eric-DCAP | no worries |
| 18:48.50 | EUS-Eric-DCAP | any ideas on my question, Fender? |
| 18:53.45 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 18:54.04 | pmhaddad-work | EUS-Eric-DCAP, ah yeah, i plan on taking the dCAP in about a week |
| 18:54.15 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:54.19 | pmhaddad-work | was just trying to get a feel for how hard it was and what i need to study up on |
| 18:54.22 | EUS-Eric-DCAP | good luck |
| 18:54.28 | EUS-Eric-DCAP | the practical is not hard, just very short |
| 18:54.28 | pmhaddad-work | thanks :) |
| 18:54.40 | pmhaddad-work | that's what i've heard |
| 18:54.43 | EUS-Eric-DCAP | you have an hour and a half to get a basic phone system up from a base linux install. |
| 18:54.43 | pmhaddad-work | hows the written? |
| 18:54.51 | EUS-Eric-DCAP | ridiculous |
| 18:55.01 | EUS-Eric-DCAP | lots of questions on specific asterisk apps |
| 18:55.09 | EUS-Eric-DCAP | lots of ones that I don't use too often |
| 18:55.15 | pmhaddad-work | such as? |
| 18:55.24 | pmhaddad-work | doesn't use too many either |
| 18:55.25 | EUS-Eric-DCAP | like how to properly write a command for pretty much every app |
| 18:55.34 | pmhaddad-work | yeesh |
| 18:56.04 | pmhaddad-work | like I can do meetme and IVR |
| 18:56.11 | pmhaddad-work | what other apps are there even? |
| 18:56.50 | EUS-Eric-DCAP | database apps, setvar apps, |
| 18:56.53 | EUS-Eric-DCAP | lots of database questions |
| 18:57.01 | pmhaddad-work | does a show applications |
| 18:58.25 | pmhaddad-work | wow |
| 18:58.36 | pmhaddad-work | i've only used a handlful of those |
| 18:59.39 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:01.14 | EUS-Eric-DCAP | yeah, it's not very fun |
| 19:01.23 | EUS-Eric-DCAP | I had a year of installing asterisk systems before I took it. |
| 19:01.31 | EUS-Eric-DCAP | And I'm a genius |
| 19:02.01 | pmhaddad-work | EUS-Eric-DCAP, i've been doing it for almost 2 |
| 19:02.09 | pmhaddad-work | i am not a genius |
| 19:02.11 | EUS-Eric-DCAP | you'll be ok. |
| 19:02.40 | pmhaddad-work | most of my experience is with freepbx and trixbox too, the by hand stuff has really only been the last 6 months or so |
| 19:02.57 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 19:03.58 | *** join/#asterisk manipura (n=Mike@S01060022b0d49327.cg.shawcable.net) |
| 19:08.24 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:11.00 | *** join/#asterisk Neil_UK (n=X@78.143.203.116) |
| 19:11.16 | Neil_UK | Hi, im looking for a asterisk expert in the Dallas, TX area? |
| 19:12.35 | MaliutaLap | Dallas? you mean it's not just a tv series? |
| 19:12.45 | Neil_UK | :) |
| 19:14.22 | awk_r | MaliutaLap, actuallys he's looking for an asterisk expert for the tv series |
| 19:14.32 | [TK]D-Fender | hides the revolver... |
| 19:14.41 | adwerw | How could I use already authorized SIP-users for access to their Voicemail / Conferences and so on? |
| 19:14.45 | Neil_UK | Im looking for an asterisk guru that can come out to Dallas with me and set up a new PBX |
| 19:15.01 | *** join/#asterisk hi365 (n=hi365@94.159.178.139) |
| 19:15.04 | [TK]D-Fender | adwerw: its your dialplan, go make some extensions. |
| 19:17.01 | adwerw | if I place VoicemailMain - it will ask me a password |
| 19:18.19 | [TK]D-Fender | adwerw: "core show application voicemailmain" <- go read its instructions |
| 19:19.33 | adwerw | I'm reading it already. I do not fully understand - is it mean tha if i supply mailbox as an ARG1 it wont ask me a password? |
| 19:21.24 | adwerw | "s - Skip checking the passcode for the mailbox" - sorry - thats it i think |
| 19:22.25 | EUS-Eric-DCAP | We have an office in Texas |
| 19:22.43 | *** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
| 19:23.27 | iEatChildren | whats the command to see line voltage? |
| 19:28.52 | pmhaddad-work | dahdi_monitor |
| 19:29.45 | iEatChildren | what about for zaptel? |
| 19:30.05 | pmhaddad-work | iEatChildren, i think its just zap_monitor or zaptel_monitor |
| 19:30.29 | iEatChildren | dont have those commands |
| 19:31.05 | pmhaddad-work | hrm, do you have the zaptel drivers and such installed? |
| 19:31.14 | pmhaddad-work | logs into a system that uses zap |
| 19:31.39 | pmhaddad-work | ew |
| 19:31.54 | iEatChildren | yes, i have zaptel drivers installed |
| 19:32.04 | pmhaddad-work | ok |
| 19:32.07 | pmhaddad-work | asterisk -rvv |
| 19:32.11 | pmhaddad-work | zap show cadences |
| 19:32.14 | pmhaddad-work | i think is what you want |
| 19:32.24 | iEatChildren | okay, ill try it |
| 19:32.40 | pmhaddad-work | its been a long time since i used zap though |
| 19:32.46 | EUS-Eric-DCAP | ztmonitor is what you want. |
| 19:33.32 | adwerw | How could I force some users to join a particular conference right on their login? |
| 19:33.40 | pmhaddad-work | EUS-Eric-DCAP, hm, i dont have that command on my zap box |
| 19:33.40 | *** join/#asterisk mclugh (n=mpearson@67.214.244.42) |
| 19:33.50 | pmhaddad-work | i thought it was something like that, but it wasnt there :( |
| 19:34.26 | pmhaddad-work | doh there it is |
| 19:34.35 | EUS-Eric-DCAP | which was it |
| 19:34.47 | iEatChildren | i have that command....just have to figure out how to see voltages now |
| 19:34.50 | pmhaddad-work | EUS-Eric-DCAP, i wasn't logged in as root so it wasn't in my path |
| 19:34.59 | pmhaddad-work | xD |
| 19:35.18 | iEatChildren | i can see the audio level.... |
| 19:35.21 | mclugh | I have some questions about a potential asterisk setup. Is it appropriate to ask those questions in this irc? |
| 19:35.50 | rob0 | Potentially. |
| 19:36.04 | rob0 | Potentially questionable, as well. |
| 19:36.06 | adwerw | I'm reading it already. I do not fully understand - is it mean tha if i supply mailbox as an ARG1 it wont ask me a password? |
| 19:36.34 | [TK]D-Fender | adwerw: No, there was that obviously stated parameter that will tell it not to as you a password |
| 19:38.01 | mclugh | My question is in regards to astericks ability to create conference calls and send the the output to a dedicated sound card on the machine? |
| 19:39.11 | seb- | [TK]D-Fender: can i ask you a question? |
| 19:39.29 | seb- | [TK]D-Fender: i can't register from work either even though you can...i'm getting this error... |
| 19:39.37 | [TK]D-Fender | seb-: You just did. that'll be $4.95 for the next one :) |
| 19:39.55 | seb- | [TK]D-Fender: :) cha-ching! |
| 19:40.03 | seb- | [TK]D-Fender: [Apr 30 12:39:26] WARNING[19948]: chan_sip.c:1783 __sip_xmit: sip_xmit of 0xb6ff0350 (len 561) to 199.106.103.254:57888 returned -1: Operation not permitted |
| 19:40.52 | [TK]D-Fender | mclugh: * doesn't creat conference calls, but it can certinaly host them, and yes you can arrange to have it go out the soundcard by having a Local channel use OSS or a local softphone. |
| 19:41.08 | [TK]D-Fender | seb-: PB the full attempt up |
| 19:41.27 | seb- | [TK]D-Fender: what do you mean? |
| 19:41.34 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 19:41.36 | [TK]D-Fender | ~pb |
| 19:41.37 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:41.38 | [TK]D-Fender | ^^^^^^ |
| 19:41.44 | [TK]D-Fender | seb-: I want to see the complete SIP debug |
| 19:41.53 | mclugh | Cool thats what I need to know |
| 19:41.56 | mclugh | thank you |
| 19:42.03 | MaliutaLap | [TK]D-Fender: they never want to give the details, do they? |
| 19:42.19 | [TK]D-Fender | grabs his ClueBat (tm) |
| 19:42.26 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 19:42.38 | MaliutaLap | has Mr Stabby at the ready |
| 19:42.53 | [TK]D-Fender | MaliutaLap: Oh? What is it? |
| 19:43.05 | EUS-Eric-DCAP | Mclugh, with asterisk you can do a console call which will output audio to the sound card of the PC |
| 19:43.09 | MaliutaLap | [TK]D-Fender: 5" lock knife |
| 19:43.20 | MaliutaLap | [TK]D-Fender: they'll think death is a career |
| 19:43.36 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 19:43.36 | *** join/#asterisk hi365 (n=hi365@94.159.178.139) |
| 19:43.40 | [TK]D-Fender | MaliutaLap: My baby : http://www.roninswords.com/custom_kiku_in_tea.htm |
| 19:43.40 | seb- | [TK]D-Fender: http://pastebin.com/m10a1c79b <--starts a little down after some blank lines |
| 19:44.21 | MaliutaLap | [TK]D-Fender: nice, I prefer close and quite for most things |
| 19:44.37 | jaytee | NINJA!!!!! |
| 19:44.41 | watchy | i want that sword |
| 19:44.54 | MaliutaLap | [TK]D-Fender: I carry a Maxim 5" most of the time |
| 19:44.54 | *** join/#asterisk ITguru (n=ITGuru@5ad2ca70.bb.sky.com) |
| 19:44.57 | iEatChildren | [TK]D-Fender: do you know how to go about viewing the voltage for each channel? i tried ztmoniter per another suggestion but im only getting audio levels...maybe im doing somethign wrong here |
| 19:45.18 | seb- | [TK]D-Fender: n/m i fixed it |
| 19:45.25 | seb- | [TK]D-Fender: f*** me |
| 19:45.37 | jaytee | that's a sweet lookin sword |
| 19:45.59 | [TK]D-Fender | iEatChildren: Lick it and count how many hairs raise :) |
| 19:46.05 | iEatChildren | hahaha |
| 19:46.10 | [TK]D-Fender | seb-: .... rather not personally ;) |
| 19:46.37 | *** join/#asterisk CrazyTux1 (n=brandon@216-110-94-230.static.twtelecom.net) |
| 19:46.42 | [TK]D-Fender | Me tests for his 3rd kyu this sunday |
| 19:46.43 | MaliutaLap | seb-: no, but I'll fuck you up |
| 19:46.45 | [TK]D-Fender | tests for his 3rd kyu this sunday |
| 19:46.48 | [TK]D-Fender | dangit :) |
| 19:46.57 | seb- | [TK]D-Fender: i have 2 IP addresses on my server....i just had to point to right one |
| 19:47.06 | CrazyTux1 | Hey guys -- I'm playing with DISA however -- randomly upon after entering in the digits the DISA seems to go to Fast Busy without any "real output" as to why? |
| 19:47.14 | iEatChildren | [TK]D-Fender: you do martial arts? |
| 19:47.24 | *** part/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com) |
| 19:47.31 | MaliutaLap | mmm Disa, she is hot ... but marrying someone else |
| 19:47.33 | CrazyTux1 | Half the time it does that, and half the time it does what it should -- any thoughts? I've tried this on 1.6.0.1-rc2 and 1.60.1-stable, and 1.4.24.1 |
| 19:47.35 | MaliutaLap | :( |
| 19:47.58 | MaliutaLap | kyu is a Go ranking |
| 19:48.05 | MaliutaLap | in my workd anyhow |
| 19:48.05 | seb- | [TK]D-Fender: i'm sorry learning this stuff is so messy... |
| 19:48.11 | [TK]D-Fender | iEatChildren: http://en.wikipedia.org/wiki/Tenshin_Shoden_Katori_Shinto-ryu |
| 19:48.34 | iEatChildren | thats sweet |
| 19:48.36 | iEatChildren | i love martial arts |
| 19:49.02 | iEatChildren | im a level 2 in JKD and the filipino arts...and a blue belt in brazilian jiu jitsu |
| 19:49.17 | MaliutaLap | I prefer them not to see or hear me coming, they don't have a chance to fight back then |
| 19:49.32 | MaliutaLap | unless I want them to follow me into a killing field |
| 19:49.53 | [TK]D-Fender | MaliutaLap: http://www.youtube.com/watch?v=2REG3-Wb5gM |
| 19:49.57 | MaliutaLap | CCCP army new how to set a nice fire sack |
| 19:50.19 | seb- | i don't have time to do the whole black belt thing...i'm going to try Krav Maga |
| 19:51.10 | *** part/#asterisk ITguru (n=ITGuru@5ad2ca70.bb.sky.com) |
| 19:51.16 | [TK]D-Fender | iEatChildren: JKD is more of a treatise... Gave it a good read and summarized it as "Ball-Fu" given BL's testicle-centric approach |
| 19:51.49 | [TK]D-Fender | seb-: flash in the pan stuff its hard to prove cert on, etc. Just like every "ninjutsu" school out there. |
| 19:51.50 | iEatChildren | JKD is a no art system...there is no "ball-fu" about it |
| 19:52.33 | [TK]D-Fender | iEatChildren: In a way, but BL's writings tend to be the "If a guy tries to kick you, kick him first." Well WTF... why didn't *I* think of that! ME SO STUPID! |
| 19:53.11 | iEatChildren | the dude was amazing fast...he would spend 8 hours a day doing 1 punch on a wooden dummy |
| 19:53.21 | [TK]D-Fender | iEatChildren: Ask yourself how many "schools" take the "no art" and end up with "no form", "no history", and "no proven track-record" |
| 19:53.39 | [TK]D-Fender | iEatChildren: this is separating the art from the artist. |
| 19:53.50 | iEatChildren | umm....brock lesnar and sean sherk train under inosoanto certified JKD instructors |
| 19:54.12 | iEatChildren | not that they are good examples of martial artist...but there is a track record |
| 19:54.23 | [TK]D-Fender | iEatChildren: Yup, those names I know... |
| 19:54.24 | iEatChildren | try paul vunak...he trains the most elite warriors there are |
| 19:54.46 | jaytee | "If someone tries to kill you, you kill em right back!" |
| 19:54.56 | [TK]D-Fender | iEatChildren: I'm talking about the other 90% of those claiming to teach XYZ |
| 19:55.02 | iEatChildren | that applies to any art |
| 19:55.07 | [TK]D-Fender | jaytee: Except try to finish yours FIRST ! |
| 19:55.21 | [TK]D-Fender | iEatChildren: Only the common and popularizes stuff :) |
| 19:55.27 | [TK]D-Fender | d* |
| 19:55.45 | iEatChildren | you mean only the arts that you hear about? |
| 19:55.54 | [TK]D-Fender | does like these new & peasant arts. Kickin' it old-school y0! |
| 19:56.42 | [TK]D-Fender | iEatChildren: I mean those who generalize Karate, TKD, JKD, BJJ (gee, thanks Gracie's), etc |
| 19:57.04 | iEatChildren | first off... TKD is an art only...its hardly affective ina fight |
| 19:57.15 | [TK]D-Fender | iEatChildren: Kyokushin = "Americal" full-contact karate. Bastard offshoot. |
| 19:57.28 | iEatChildren | BJJ....is probably the most effective fighting style out there |
| 19:57.39 | iEatChildren | gracies proved that, but then it turned in to a marketing system |
| 19:57.50 | iEatChildren | now you can get a blackbelt online through them |
| 19:58.05 | iEatChildren | i rolled with a couple gracie students at a tourny....i wasnt that impressed |
| 19:58.07 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 19:58.19 | pmhaddad-work | a black belt online?!??! |
| 19:58.21 | pmhaddad-work | wtf! |
| 19:58.25 | [TK]D-Fender | iEatChildren: Actually the Gracie's proved something entirely different. It's MARKETING, pure and simple |
| 19:58.30 | iEatChildren | yeah, they have "gracie university" now |
| 19:58.31 | pmhaddad-work | is a third degree black belt in TKD.. |
| 19:58.36 | pmhaddad-work | i did NOT get it online lol |
| 19:58.43 | iEatChildren | [TK]D-Fender: they prove bjj is very effective |
| 19:59.00 | [TK]D-Fender | iEatChildren: BJJ goes right out the door when the rules aren't steered in their favor. the first UFC's and its very creation were RIGGED and set to promote gracie BJJ |
| 19:59.05 | iEatChildren | im not saying all blackbelts are obtained online, but there are a lot of mcdojo's iout there |
| 19:59.13 | pmhaddad-work | and iEatChildren i totally disagree thats its not useful in a fight |
| 19:59.27 | iEatChildren | bjj does not go out the window...i dont see where you get that |
| 19:59.29 | *** join/#asterisk adwerw (n=max@80-240-220-48.dnat.migtel.ru) |
| 19:59.35 | iEatChildren | they teach punching and kicking in bjj...its not JUST grappling |
| 19:59.55 | [TK]D-Fender | iEatChildren: Very effective in UFC. Lets see those fuckers try flying around like that on CONCRETE and where the enemy can gouge their eyes out |
| 20:00.16 | iEatChildren | you cant do all the same moves...i agree there |
| 20:00.23 | iEatChildren | but there is a LOT you can still do |
| 20:00.23 | [TK]D-Fender | iEatChildren: UFC & its rules were engineers to favour BJJ |
| 20:00.36 | iEatChildren | so go with pride rules |
| 20:00.45 | iEatChildren | you could knee in the head when they are grounded |
| 20:00.58 | iEatChildren | try rio heros...just about anything goes except eye gouging and hair pulling and biting |
| 20:01.05 | iEatChildren | bjj works in them all |
| 20:01.05 | Brixius | Hello, I have a question about app_addon_sql_mysql. Not so much in it's use, but in that the connid will not recycle and so runs out after a while. |
| 20:01.22 | [TK]D-Fender | iEatChildren: And the first few put up supposed "martial artists" in a match to look like "Karate vs Kun-Fu, who's better?", only to get owned by grapplers who have NO clue what to do when the close is made |
| 20:01.44 | iEatChildren | thats back when it was style vs style.....now that everything is more rounded you dont see that |
| 20:02.15 | iEatChildren | grapplers are typically afraid to strike or get hit, and strikers hate the ground and panic when they are on their backs |
| 20:02.18 | iEatChildren | there is give and take to it all |
| 20:02.33 | areay | what does "SIP/2.0 401 Unauthorized" mean? |
| 20:02.42 | [TK]D-Fender | iEatChildren: And their qualifications were actually BS. Shamrock won out like he did because he did have some grappling experience. However I've watched a bunch of lower-level UFC fights to see a bunch of show-boating ass-clown running around like a circus side-show attraction. |
| 20:02.50 | [TK]D-Fender | areay: "GTFO" :) |
| 20:03.17 | areay | [TK]D-Fender, lol |
| 20:03.52 | iEatChildren | im not saying a ground game is all you need, but you NEED to have a ground game |
| 20:04.00 | iEatChildren | if you dont know bjj you are screwed when you are put on your back |
| 20:04.02 | [TK]D-Fender | iEatChildren: having started on the jiujustu component of my training I have a real appreciation for the old warrior way. |
| 20:04.09 | iEatChildren | and if you dont know a striking game you are screwed on yoru feeet |
| 20:04.10 | [TK]D-Fender | iEatChildren: In UFC? Hell yeah |
| 20:04.18 | areay | [TK]D-Fender, is it because of network settings, or the wrong login information in my sip client (or sip.conf for that matter)? |
| 20:04.18 | iEatChildren | in real life too |
| 20:04.19 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:04.24 | *** join/#asterisk stack_ (n=sgerstac@mail.edpaymentsystems.com) |
| 20:04.31 | [TK]D-Fender | iEatChildren: thats why its almost more "sport" than "fight" in a way |
| 20:04.36 | iEatChildren | watch felony fights and tell me how many times they go to the ground |
| 20:04.38 | iEatChildren | it happens all the time |
| 20:04.57 | [TK]D-Fender | areay: typically bad user&pass |
| 20:05.00 | iEatChildren | thats what JKD is about, being well rounded. which is why i practise it |
| 20:05.09 | iEatChildren | even dana white admits bruce lee is the father of MMA |
| 20:05.18 | iEatChildren | fedor said the 1 person he wouldnt want to fight is bruce lee |
| 20:05.20 | stack_ | I upgraded my Asterisk box's OS from an old version of Ubuntu to Ubuntu Hardy, which no longer has soxmix. Is there an alternative for joining monitor outputs? |
| 20:05.23 | mmlj4 | MMA? |
| 20:05.29 | iEatChildren | mixed martial arts |
| 20:05.38 | mmlj4 | ah. |
| 20:06.01 | [TK]D-Fender | iEatChildren: And Bruce Lee was scared shitless at the proposed fight with Mohammed Ali :) |
| 20:06.08 | areay | [TK]D-Fender, cool, thanks |
| 20:06.11 | iEatChildren | bruce lee....i doubt that |
| 20:06.17 | [TK]D-Fender | iEatChildren: "OMG, he hits me I'll crumple!" |
| 20:06.23 | iEatChildren | he seemed like a dick to be honest |
| 20:06.26 | iEatChildren | bruce knew he was good |
| 20:06.26 | [TK]D-Fender | iEatChildren: No, he's quoted for it :) |
| 20:06.48 | mmlj4 | "float like a butterfly and sting bruce lee" |
| 20:06.59 | [TK]D-Fender | iEatChildren: thats one thing boxers really have going for them. they take hits. So few martial arts really do. |
| 20:07.00 | iEatChildren | i can see that if they are talking about straight boxing |
| 20:07.01 | tzafrir_home | stack_, sox |
| 20:07.02 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 20:07.05 | iEatChildren | but in an all out fight ali has nothing on lee |
| 20:07.14 | iEatChildren | [TK]D-Fender: and boxers have HUGE gloves |
| 20:07.21 | iEatChildren | how many boxers have lasted in mma? |
| 20:07.23 | iEatChildren | not many |
| 20:07.32 | iEatChildren | watched fedor knock a couple flat on their behind too |
| 20:07.34 | stack_ | tzafrir_home, right, but how do I tell asterisk to behave differently? |
| 20:07.38 | [TK]D-Fender | iEatChildren: technique matters, but so does raw power, timing and the ability to take a hit. Strategy, etc |
| 20:07.40 | iEatChildren | and he has a very bizare striking style |
| 20:07.55 | iEatChildren | anderson silva is talking about going in to boxing....i cant wait to see it |
| 20:07.59 | tzafrir_home | stack_, IIRC you can write your own soxmix wrapper script |
| 20:08.17 | [TK]D-Fender | iEatChildren: How many were real "boxers"? And again... it isn't aboxing match. Someone with no kicking & grappling experience is likely to get owned |
| 20:08.25 | [TK]D-Fender | iEatChildren: the rules make the game. |
| 20:08.35 | *** join/#asterisk afink (n=andrew@asa.redglaze.com) |
| 20:08.50 | iEatChildren | what if there are no rules? i dont see a boxer getting very far unless hes fighting the average joe |
| 20:08.54 | iEatChildren | or that good a boxer |
| 20:09.11 | iEatChildren | i hate talking fighting math though.....its been proven incorrect so many times |
| 20:09.19 | [TK]D-Fender | iEatChildren: take that power, then take the gloves off :) |
| 20:09.57 | iEatChildren | depends on the boxing style too though, if they are an inside, outside, or brawler style boxer |
| 20:09.57 | iEatChildren | each have their strengths and weakness |
| 20:10.01 | [TK]D-Fender | iEatChildren: Yup, the volume punchers don't necessarily last eather |
| 20:10.18 | stack_ | tzafrir_home: is there an example somewhere... I can't seem to find one |
| 20:10.37 | [TK]D-Fender | iEatChildren: Thats where the endurance ones fail on the street and the power-punchers like Tyson would reign. |
| 20:10.59 | iEatChildren | they can if they do it right, like brawlers usually overwhelm in side punchers, outside strikers can use speed and footwork to overcome bralwers...and inside strikers usually overwelm outside boxers |
| 20:11.14 | iEatChildren | tyson was a freak |
| 20:11.19 | iEatChildren | he could box inside or brawl just as easy |
| 20:11.25 | iEatChildren | and had the power for it all |
| 20:11.31 | iEatChildren | like his leaping left hook i love so much |
| 20:11.42 | iEatChildren | he like hops and throws this thing....and man does it have some power |
| 20:13.15 | iEatChildren | easily the best boxer ever if his trainer wouldnt have died causing tyson to show how crazy he really is |
| 20:13.50 | [TK]D-Fender | iEatChildren: Yes, but look at what happens if you play it out a few rounds and Tyson slows down. thats the trick against him |
| 20:14.04 | iEatChildren | yeah, thats what happens to brawlers |
| 20:14.06 | iEatChildren | they lose their wind |
| 20:14.25 | iEatChildren | they have to get those early knockouts otherwise the outside striker will jsut wait and wait till he gets his chance |
| 20:14.49 | iEatChildren | most people dont realize how tired you get throwing a flurry of punches |
| 20:15.03 | [TK]D-Fender | iEatChildren: Just for the love og god don't them land :) |
| 20:15.07 | iEatChildren | lol |
| 20:15.08 | iEatChildren | yup |
| 20:15.13 | iEatChildren | only takes 1 hit |
| 20:17.06 | iEatChildren | and earlier...i did say that TKD was useless in a fight...i know thats not true...but i think TKD teaches you some piss poor things to do while in a real fight |
| 20:17.29 | iEatChildren | but then again....a blackbelt in TKD probably knows better |
| 20:17.34 | iEatChildren | i would hope so anyways |
| 20:18.29 | iEatChildren | how long have you been in the arts now? |
| 20:19.46 | *** join/#asterisk DarthWar (i=user@69-92-91-117.cpe.cableone.net) |
| 20:20.00 | areay | [TK]D-Fender, could one-way audio be a codec problem? |
| 20:20.27 | DarthWar | iEatChildren to many peeps in here... |
| 20:20.38 | iEatChildren | lol |
| 20:21.00 | iEatChildren | its all good DarthWar....i think the martial arts talk just ended.... |
| 20:21.10 | iEatChildren | if anyone wants to talk martial arts join ##mma |
| 20:21.14 | iEatChildren | sorry for that spam |
| 20:21.22 | iEatChildren | i realize how off topic i got everyone there for a minute |
| 20:21.31 | DarthWar | well cuz your ebul |
| 20:21.36 | *** part/#asterisk DarthWar (i=user@69-92-91-117.cpe.cableone.net) |
| 20:22.45 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 20:23.52 | [TK]D-Fender | areay: Nope |
| 20:23.58 | [TK]D-Fender | areay: NETWORKING again |
| 20:24.35 | areay | [TK]D-Fender, kk |
| 20:24.42 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 20:25.08 | [TK]D-Fender | iEatChildren: a little over 3 years of TSKSR, and interest started about 20 years ago in Wing Chun Kung-Fu & Aikido |
| 20:25.56 | [TK]D-Fender | iEatChildren: true kobudo takes the light and fluffy of Aikido and laughs at it :) |
| 20:26.25 | iEatChildren | lol good |
| 20:26.33 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 20:26.49 | [TK]D-Fender | iEatChildren: The old school stuff wasn't just "self defense", it was "this is WAR and you're DEAD". |
| 20:26.55 | [TK]D-Fender | has no time for "do" |
| 20:28.53 | iEatChildren | i started about 3 years ago with muay thai because its a brutal art, then started including weapons training (jkd and the filipino arts) and doing bjj. now my main focus is bjj. |
| 20:29.09 | iEatChildren | simply because i enter a bjj tourney about every 3 months |
| 20:30.02 | iEatChildren | i really enjoy the filipino stuff...its some dirty freakin boxing |
| 20:30.11 | iEatChildren | lots of elbows :-) |
| 20:30.24 | [TK]D-Fender | iEatChildren: Add kali then. Very natural extension for you |
| 20:30.28 | iEatChildren | yes |
| 20:30.32 | iEatChildren | i do kali |
| 20:30.40 | iEatChildren | thats included in my JKD arts |
| 20:30.42 | [TK]D-Fender | iEatChildren: A member of my class did that as well |
| 20:30.43 | xuser | my 9mm kill kick both your ass :P |
| 20:30.44 | iEatChildren | dan inosanto is HUGE on kali |
| 20:31.02 | [TK]D-Fender | Alrighty, checkout time, BBIAB |
| 20:31.08 | iEatChildren | later man |
| 20:31.09 | [TK]D-Fender | heads home |
| 20:31.10 | iEatChildren | good talkin to ya |
| 20:32.14 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 20:32.51 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:33.40 | jameswf | I can now generatte queued callbacks with twitter Direct messages :) |
| 20:34.13 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 20:35.56 | *** part/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 20:35.59 | adwerw | How can I add some SIP-users to a particular conference iimediately on their login? |
| 20:36.05 | iEatChildren | i f***ing hate twitter |
| 20:36.34 | DavidR2008 | is bored |
| 20:37.11 | DavidR2008 | adwerw: what do you mean by their "login" |
| 20:37.53 | adwerw | "login: means - register their phone to aasterisk |
| 20:38.30 | DavidR2008 | adwerw: not sure if any event can be triggered by a registration |
| 20:39.35 | adwerw | this is rather strange |
| 20:39.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:40.05 | hardwire | frame.c:216 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 20:40.07 | hardwire | ugh. |
| 20:40.15 | hardwire | how the crap do I get rid of this notice? |
| 20:40.36 | hardwire | interop with a quintam box.. dunno the make or model yet. |
| 20:43.08 | nkohh | cat log | grep -v "dropping extra frame" |
| 20:43.33 | nkohh | also, see http://bugs.digium.com/view.php?id=5539 |
| 20:45.03 | DavidR2008 | adwerw: I looked at the docs (cause I'm bored ;-) ) and I don't see anyway of triggering something based on a registration. What you'd probably have to do is write something using AMI that polls every so often for new registrations then, calls them with an auto-answer that bridges in to the conference. |
| 20:47.46 | *** join/#asterisk telecos (n=sergio@42.166.219.87.dynamic.jazztel.es) |
| 20:48.00 | DavidR2008 | assuming your sip clients support auto-answer |
| 20:50.23 | *** join/#asterisk nauticalthinker (n=mratliff@74.5.197.138) |
| 20:53.02 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 20:54.14 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:56.40 | adwerw | DavidR2008: thank you for your help - but isn't this wierd that no one need to execute some code right after registration on user device?? |
| 20:57.13 | [TK]D-Fender | adwerw: Many people ask for it actually. |
| 20:57.29 | DavidR2008 | so is possible? |
| 20:58.04 | DavidR2008 | is it* |
| 20:58.55 | [TK]D-Fender | DavidR2008: Directly no. Suggestion I made yesterday, poll via AMI for the existance of an exten created by "regexten" for the peer |
| 20:59.18 | adwerw | i think this is "not elegant" - at least |
| 20:59.23 | DavidR2008 | ok, so I was accurate in recommending an AMI poll |
| 21:00.03 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 21:03.11 | DavidR2008 | heading home |
| 21:03.14 | ddickenson | hello there, one of these days I'll get good enough at this to give advice in this room, but for now I'm stuck being the guy with all the questions... What all can cause this error 'Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)' I have run into this before when I forgot to su before issuing commands but I'm logged in as root using the 'su -' command |
| 21:03.44 | EUS-Eric-DCAP | you need to start asterisk |
| 21:03.45 | [TK]D-Fender | ddickenson: * has to be RUNNING, and you have to have rights to it & the PID file |
| 21:03.53 | EUS-Eric-DCAP | service asterisk start |
| 21:04.01 | EUS-Eric-DCAP | or chkconfig asterisk on and then reboot |
| 21:04.03 | EUS-Eric-DCAP | hehe |
| 21:04.16 | [TK]D-Fender | loves seeing distro-specific advice thrown around here... |
| 21:04.23 | hardwire | so when I watch the interupt count for irq 20.. holding the tdm card I'm using (4 port t1) it usually stays at 500 per 2 seconds |
| 21:04.31 | ddickenson | like the centos/redhat etc... |
| 21:04.34 | hardwire | right now it's at 17379 per 2 seconds |
| 21:04.42 | hardwire | which seems.. like a lot. |
| 21:04.56 | hardwire | but it's a lot of small frames going back and forth as well.. for 12 active lines |
| 21:05.01 | ddickenson | yeah it seems that just puts * in a loop trying to start and exiting on signal 11 |
| 21:05.09 | adwerw | if you do "service asterisk restart" for asterisk 1.6 - it wont start in routhly 30% of cases |
| 21:05.28 | ddickenson | exit status 139? |
| 21:05.32 | [TK]D-Fender | adwerw: How wonderfully unqualified! |
| 21:05.56 | [TK]D-Fender | ddickenson: Considered initializing Zaptel/DAHDI first? |
| 21:05.58 | adwerw | [TK]D-Fender: but checket for myself :) |
| 21:06.08 | adwerw | *checked |
| 21:06.13 | EUS-Eric-DCAP | run asterisk -vvvvc |
| 21:06.21 | EUS-Eric-DCAP | and see why it's dying when you try to start it. |
| 21:07.37 | ddickenson | haven't initialized dahdi, how do I do that? |
| 21:07.49 | [TK]D-Fender | ddickenson: dahdi_cfg -vvvv |
| 21:08.01 | ddickenson | EUS-Eric-DCAP: That's the command I was running to try and start asterisk that was giving the original error |
| 21:08.13 | EUS-Eric-DCAP | ok |
| 21:08.38 | [TK]D-Fender | ddickenson: What you tried first wasn't STARTING Asterisk, you were trying to conenct to an assumed alkready started instance |
| 21:08.40 | EUS-Eric-DCAP | Usually when there's an error that prevents asterisk from starting, and it's zaptel related, it's a typo in the zaptel configs |
| 21:09.25 | ddickenson | you're right, I was assuming the thing started with boot like it was set to and used to be doing. I take that step for granted I guess |
| 21:09.31 | [TK]D-Fender | EUS-Eric-DCAP: Hardly. Far more common is the "OMG I just never initialized it and yeah the configs are fine, and I really ought to learn to setup my init scripts in the right order!" |
| 21:09.41 | EUS-Eric-DCAP | hahaha |
| 21:09.42 | ddickenson | so check my chan_dahdi.conf |
| 21:09.43 | EUS-Eric-DCAP | likely |
| 21:09.45 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 21:10.11 | ddickenson | yeah when I tried to initialize it kicked out and error. I guess I have some syntax problems |
| 21:10.32 | [TK]D-Fender | ddickenson: What makes you think that? |
| 21:10.42 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 21:12.25 | EUS-Eric-DCAP | you running dahdi or zaptel? |
| 21:12.42 | *** join/#asterisk hi365 (n=hi365@94.159.178.139) |
| 21:12.52 | ddickenson | because when I ran dahdi_cfg it said configuration file is /etc/dahdi/system.conf; line 0 unable to open master device...etc |
| 21:12.53 | ddickenson | dahdi |
| 21:13.01 | ddickenson | asterisk1.6.0.9 |
| 21:13.11 | *** join/#asterisk TXTrader (n=kvirc@72.183.119.173) |
| 21:13.26 | EUS-Eric-DCAP | what happens when you do dahi_cfg -vvv |
| 21:13.28 | EUS-Eric-DCAP | ? |
| 21:13.34 | EUS-Eric-DCAP | oh |
| 21:13.36 | ddickenson | that was with the verbosity |
| 21:13.42 | ddickenson | just didn't type it |
| 21:13.46 | EUS-Eric-DCAP | I didn't see your message up there. |
| 21:13.50 | EUS-Eric-DCAP | ok. |
| 21:14.31 | ddickenson | I guess I should have typed the rest... line 0: Unable to open master device '/dev/dahdi/ctl' |
| 21:15.03 | EUS-Eric-DCAP | do a modprobe dahdi |
| 21:15.21 | ddickenson | FATAL: Module dahdi not found. |
| 21:15.28 | ddickenson | bummer |
| 21:15.55 | EUS-Eric-DCAP | did you just compile and install dahdi? |
| 21:16.07 | ddickenson | What I don't understand is that I had it working and now doesn't seem to be happy at all. Should I just try recompile dahdi? This isn't a production system yet |
| 21:16.17 | ddickenson | I did yesterday |
| 21:16.24 | EUS-Eric-DCAP | have you rebooted? |
| 21:16.25 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:16.27 | ddickenson | yes |
| 21:16.31 | EUS-Eric-DCAP | hmm.. |
| 21:16.50 | EUS-Eric-DCAP | it can't find the device, there's a way to fix this with modprobe, but I don't remember it. |
| 21:16.54 | EUS-Eric-DCAP | Rebooting does the same thing. |
| 21:17.02 | EUS-Eric-DCAP | Recompiling won't hurt |
| 21:17.14 | [TK]D-Fender | ddickenson: What interfaces are you running? |
| 21:17.47 | ddickenson | as in t1 cards and fxo/s etc? |
| 21:17.53 | [TK]D-Fender | ddickenson: yes |
| 21:18.25 | ddickenson | digium 4 port t1 card right now. Only planning on using 2 ports on it |
| 21:18.42 | ddickenson | I forget the number. |
| 21:19.06 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 21:19.26 | EUS-Eric-DCAP | do this: |
| 21:19.27 | ddickenson | is it possible that if I were to have forgotten in my steps to compile dahdi and then wend and edited a say /etc/dahdi/system.conf or the /etc/asterisk/chan_dahdi.conf that it would start kicking out errors |
| 21:19.31 | EUS-Eric-DCAP | are you on centos? |
| 21:19.34 | [TK]D-Fender | ddickenson: modprobe dahdi ; modprobe wcte4xxp |
| 21:19.56 | [TK]D-Fender | ddickenson: then : dahdi_cfg -vvvv |
| 21:20.16 | [TK]D-Fender | ddickenson: Confirm the kernel module is loaded first |
| 21:20.56 | EUS-Eric-DCAP | I think you have to "service dahdi start" before you modprobe? |
| 21:21.00 | ddickenson | it says it isn't |
| 21:21.04 | ddickenson | on both counts |
| 21:21.19 | *** join/#asterisk TXTrader (n=kvirc@72.183.119.173) |
| 21:23.44 | ddickenson | I'm trying recompile (or possibly initial compile if I'm stupid and somehow forgot) and then reboot and try that stuff again |
| 21:25.14 | ddickenson | well when it starts (or rather tries) to start dahdi in the reboot it all fails because modules are missing in the kernel. |
| 21:28.01 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 21:32.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:32.30 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 21:32.46 | nullable_type | Can someone help me with getting g729 working |
| 21:33.25 | nullable_type | I have in sip.conf allow g729. Also i have the .so file in codecs folder |
| 21:33.27 | nullable_type | still no work |
| 21:34.52 | ddickenson | D-Fender/EUS-Eric: So am I just screwed here? I don't know what could have happened |
| 21:35.03 | *** join/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com) |
| 21:35.32 | _brent_ | when a call gets forwarded with a 302 Moved response accompanied by a Diversion: header, will asterisk include any information about the origininal call when it creates the new call leg to the forwardee? |
| 21:36.34 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 21:47.57 | iEatChildren | [TK]D-Fender: found out how to get the line voltage....mind if i PM it to you? |
| 21:48.09 | nullable_type | Hey guys fir g729 do i need to reregister after reinstalling Asteisk?! |
| 21:48.28 | tzafrir_home | ddickenson, what is the output of: lsmod | grep ^dahdi; cat /proc/dahdi/* |
| 21:48.51 | tzafrir_home | and also: modinfo dahdi |
| 21:49.51 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 21:49.56 | *** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za) |
| 21:50.18 | ddickenson | tzafrir_home: it doesn't see the /proc/dahdi/ directory so no output at all |
| 21:50.31 | ddickenson | and could not find module dahdi |
| 21:50.41 | *** join/#asterisk AJayMN (n=root@24-159-234-93.dhcp.mdsn.wi.charter.com) |
| 21:50.49 | tzafrir_home | find /lib/modules -name dahdi.ko |
| 21:50.55 | [TK]D-Fender | iEatChildren: load res_linelick.so :p |
| 21:51.00 | AJayMN | Anyone know how to add H.264 support into Asterisk 1.2.26 ? |
| 21:51.51 | iEatChildren | [TK]D-Fender: i got these from digium support.....didnt involve load res_linelick.so |
| 21:52.50 | iEatChildren | sent it over...anywho...im taking off. good talking to you [TK]D-Fender |
| 21:52.51 | ddickenson | <PROTECTED> |
| 21:53.04 | AJayMN | I seen someone had a write up about it but the links are broken |
| 21:53.15 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.152) |
| 21:53.16 | ddickenson | the interesting thing about that output is my kernel is the non "xen" version of that same version number |
| 21:57.56 | nullable_type | D-Fender >> There was a g729 licence installed in a box with asterisk 1.4. I had to reinstall and it seems the licence is gone. I just need to test few things, do i need to re-register g729 or contact digium for this? |
| 21:58.15 | tzafrir_home | ddickenson, uanme -r |
| 21:58.47 | ddickenson | same kernel except - the xen on the end |
| 21:59.27 | ddickenson | should I be able to just copy that dahdi directory to the l/lib/modules/{my kernelname} directory and everything be happy |
| 21:59.34 | tzafrir_home | no |
| 21:59.47 | ddickenson | dang... too easy |
| 22:00.24 | tzafrir_home | how did you point dahdi-linux to the kernel source tree? |
| 22:01.25 | ddickenson | didnt, although I think I switched kernels at some point, possibly after the install... which is all clicking to me now and I feel like and idiot |
| 22:02.24 | ddickenson | still don't know how to fix it though... |
| 22:05.22 | tzafrir_home | any chance 'make install' failed? |
| 22:06.17 | ddickenson | no, it gave no such error... |
| 22:06.26 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:06.27 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 22:07.19 | tzafrir_home | If not, what is the otput of: modinfo -F vermagic driver/dahdi/dahdi.ko |
| 22:10.49 | tzafrir_home | well, I'm off now |
| 22:11.31 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 22:11.51 | ddickenson | not found |
| 22:11.58 | ddickenson | thx for help anyway |
| 22:16.02 | rob0 | Module not found, or command not found? |
| 22:16.30 | ddickenson | module not found |
| 22:16.35 | ddickenson | im' sorry |
| 22:16.40 | ddickenson | command not found |
| 22:17.04 | rob0 | modinfo(8) is probably in /sbin |
| 22:17.06 | ddickenson | but that's because i fat fingered it |
| 22:17.48 | ddickenson | real output... "modinfo: could not find module driver /dahdi/dahdi.ko |
| 22:18.34 | rob0 | I think he was telling you to use a path to dahdi.ko relative to the dahdi-linux source directory. |
| 22:18.59 | ddickenson | you think that will work? and how do you do it? |
| 22:19.14 | rob0 | You seem to have used an absolute path, /dahdi/dahdi.ko ... obviously nothing there as there is no /dahdi directory. |
| 22:19.52 | rob0 | Work? Work in what way? He was simply trying to determine if you had the module properly compiled and installed. |
| 22:20.31 | ddickenson | oh |
| 22:24.32 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 22:33.37 | *** join/#asterisk XLA187 (n=jchase@189.154.32.251) |
| 22:39.36 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:39.36 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 22:40.10 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 22:44.34 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:44.37 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) |
| 22:45.10 | *** join/#asterisk XLA187 (n=jchase@189.154.32.251) |
| 22:46.59 | leif[mobile] | i don't want to meet your mom! |
| 22:47.36 | *** part/#asterisk XLA187 (n=jchase@189.154.32.251) |
| 22:47.44 | *** join/#asterisk XLA187 (n=jchase@189.154.32.251) |
| 22:52.27 | *** join/#asterisk XLA187 (n=jchase@189.154.32.251) |
| 22:52.45 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.36) |
| 22:55.29 | hardwire | woot |
| 22:55.39 | hardwire | calls IN from an adtran make asterisk spike and want to kill itself |
| 22:55.49 | hardwire | e&m digital trunk |
| 22:55.58 | hardwire | calls OUT work fine. |
| 22:56.02 | hardwire | octastic echo enabled. |
| 22:56.03 | hardwire | grr. |
| 23:22.36 | hardwire | I retract that.. it's PRI |
| 23:22.54 | hardwire | can funky bytes on a PRI line hose zaptel/asterisk? |
| 23:23.15 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 23:23.42 | exothermc | trying to compile asterisk 1.4.24.1 and I get: [CC] astman.c -> astman.o |
| 23:23.42 | exothermc | astman.c:95: error: expected identifier or '(' before numeric constant |
| 23:24.17 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 23:25.03 | exothermc | what would cause this? |
| 23:26.27 | *** join/#asterisk Sanjoy (n=Sanjoy@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 23:26.41 | *** join/#asterisk DarkRift (n=dark@65.92.169.126) |
| 23:29.00 | *** join/#asterisk BuSyAnToS (n=31749@93-44-77-103.ip96.fastwebnet.it) |
| 23:32.04 | exothermc | mine looks just like this one: http://ja.pastebin.ca/980216 |
| 23:32.11 | exothermc | which is all I can find on google. |
| 23:41.07 | *** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net) |
| 23:43.35 | f0ner00t | Hello. |
| 23:44.00 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 23:45.37 | *** join/#asterisk propellerhead (n=yogurt2u@190.246.190.93) |
| 23:46.38 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 23:54.55 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) [NETSPLIT VICTIM] |
| 23:54.55 | *** join/#asterisk CrazyTux1 (n=brandon@216-110-94-230.static.twtelecom.net) [NETSPLIT VICTIM] |
| 23:54.55 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) [NETSPLIT VICTIM] |
| 23:54.55 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) [NETSPLIT VICTIM] |
| 23:54.55 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) [NETSPLIT VICTIM] |
| 23:54.55 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) [NETSPLIT VICTIM] |
| 23:54.55 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 23:54.56 | *** join/#asterisk xpot (n=james@70.91.210.233) |
| 23:54.56 | *** join/#asterisk wdoekes (n=walter@wjd.osso.nl) [NETSPLIT VICTIM] |
| 23:54.57 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 23:54.57 | *** join/#asterisk yang (i=yang@CAcert/Assurer/freenode.sponsor.yang) |
| 23:54.57 | *** join/#asterisk Arkaos` (n=arkaos@chimera.c0ws.biz) [NETSPLIT VICTIM] |
| 23:54.57 | *** join/#asterisk slaney (n=slaney@67-207-131-109.slicehost.net) [NETSPLIT VICTIM] |
| 23:54.57 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-45292b9daf9c0872) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk infinity1 (n=brendon@web2.artsopolis.com) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk toorima (n=bq@ip68-7-79-241.sd.sd.cox.net) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk plantseeker (n=chatzill@77.240.56.22) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) [NETSPLIT VICTIM] |
| 23:54.58 | *** join/#asterisk hawk (n=hawk@l.qw.se) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 23:55.00 | *** join/#asterisk smash- (n=smash@173.11.0.109) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk chiwawa_42 (n=jerome@can59-3-82-233-175-214.fbx.proxad.net) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk C4colo (n=DJpyro@66.185.111.33) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk watchy (n=watchy@76.196.98.139) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 23:55.00 | *** join/#asterisk mocker (n=kyle@shell.mocker.org) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk clintc (n=clintc@n128-227-53-108.xlate.ufl.edu) [NETSPLIT VICTIM] |
| 23:55.00 | *** join/#asterisk _sasa_ (n=sasa@84-255-205-164.static.t-2.net) [NETSPLIT VICTIM] |
| 23:55.01 | *** join/#asterisk jackal (n=jackal@pool-98-118-240-229.clppva.fios.verizon.net) [NETSPLIT VICTIM] |
| 23:55.01 | *** join/#asterisk brian (n=brian@unaffiliated/brian) [NETSPLIT VICTIM] |
| 23:55.01 | *** join/#asterisk McDarren (n=darren@shrubbery.chris.net.au) [NETSPLIT VICTIM] |
| 23:55.01 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) [NETSPLIT VICTIM] |