| 00:00.07 | jaxxan | i can do a mass config change, then just reset the POE ports remotely and the phone will just reboot with the new changes. |
| 00:00.26 | JT | yes you can obviously do all that mass deployment stuff with polycoms and other half decent ip phones |
| 00:00.33 | JT | this is not something unique to cisco |
| 00:00.43 | jaxxan | sweet |
| 00:00.51 | wunderkin | you dont have to do a power reset to accomplish that either |
| 00:01.22 | jaxxan | oh yeah ? |
| 00:01.26 | denon | just script a telnet to reboot the phones |
| 00:01.32 | denon | or a wget to the webui or something |
| 00:01.41 | denon | two clicks and the whole company reboots :) |
| 00:02.29 | dlynes_laptop | jaxxan: how do you specify the option 66, anyways? Do you have a sample config line for it? |
| 00:02.40 | dlynes_laptop | jaxxan: I haven't been able to figure out how to put it in there, yet |
| 00:02.47 | jaxxan | option 66 ? |
| 00:02.57 | dlynes_laptop | jaxxan: for the tftp server option in the dhcpd.conf file |
| 00:03.17 | jaxxan | i'm gonna put a target on myself real quick. |
| 00:03.19 | denon | option tftp-server-name "1.2.3.4"; |
| 00:03.28 | dlynes_laptop | denon: ah, thanks |
| 00:03.37 | jaxxan | i have a windows server 2003 dhcp server |
| 00:03.37 | denon | that's for isc dhcpd anyway |
| 00:03.52 | denon | windows lets ya do it too, by option number I think |
| 00:03.55 | denon | I dont recall it's syntax |
| 00:04.00 | denon | or where it's at I mean |
| 00:04.24 | dlynes_laptop | denon: I'm using isc dhcpd anyways :0 |
| 00:04.38 | jaxxan | i just specify 066 Boot Server Host Name with the ip address of my tftp server |
| 00:04.47 | jaxxan | and that's it |
| 00:04.56 | dlynes_laptop | jaxxan: ah...I guess you're not using ISC dhcpd |
| 00:04.56 | JT | windows? |
| 00:05.51 | jaxxan | yeah, i thought it was pointless for me to setup a linux dhcp server for my active directory business network |
| 00:06.29 | JT | heh |
| 00:06.34 | jaxxan | but |
| 00:06.54 | jaxxan | with this new project i'll be doing it all in linux so... i guess i'll have to figure it out |
| 00:08.49 | tzanger | what the fuck |
| 00:08.50 | *** join/#asterisk CyBeR_GeeK (n=CyBeR_Ge@201.89.88.30) |
| 00:09.01 | tzanger | back to this stupid sip shit where it's not recoginizing the peer so it's dumping me into the default context |
| 00:10.00 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:10.58 | Tako-san | Anyone around with channel bank experience with time to help troubleshoot a problem? |
| 00:11.19 | JT | i have a channel bank |
| 00:17.36 | x86 | anyone successfully implement the switch command in a dialplan between two asterisk boxes? |
| 00:17.52 | tzanger | http://pastebin.ca/582525 |
| 00:18.07 | x86 | for me? |
| 00:18.09 | tzanger | very simple... placing a call from Machine A to Machine B via SIP. No Nat. both asterisk 1.4 |
| 00:18.12 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 00:18.12 | tzanger | no |
| 00:18.20 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 00:18.26 | tzanger | machine B is dumping the call into [default] and I have no idea why |
| 00:18.42 | jaxxan | jt: you gotta manufacturer name off the top of your head for that 4port poe 4 port 10/100 ? |
| 00:19.01 | JT | nah, they're just cheapies |
| 00:19.07 | jaxxan | k |
| 00:19.12 | jaxxan | i'll hit up linksys first |
| 00:19.20 | jaxxan | that polycom 320 looks great |
| 00:19.34 | JT | yeah, only downside is no speakerphone |
| 00:19.42 | JT | and silver plastic, i hate silver plastic |
| 00:19.45 | snuffy22 | tzanger, generally if its doing that i make sure i specify a user/pass |
| 00:19.57 | tzanger | snuffy22: with a peer and permit/deny I shouldn't need to |
| 00:20.01 | snuffy22 | that way i know it goes into correct context |
| 00:20.21 | x86 | jaxxan: i like the polycom 430's |
| 00:20.21 | snuffy22 | yes it shouldn't.. but like u can see.. it does wrong |
| 00:20.30 | x86 | jaxxan: actually, anything polycom is great :) |
| 00:20.41 | x86 | i've got three 601's sitting here |
| 00:20.45 | x86 | (at my house) |
| 00:21.06 | x86 | got a 501 at work, which is totally suck compared to the 601's ;) |
| 00:21.22 | tzanger | snuffy22: I specified secret=12345 on both... still no go |
| 00:21.46 | tzanger | x86: I like the 501s and 601s equally |
| 00:21.49 | Tako-san | jt: the problem I have having is that from time to time a call in progress will get "barged" by an incoming call. ever expereinced that? |
| 00:21.59 | JT | no |
| 00:22.22 | Tako-san | Ah, ok then. |
| 00:22.26 | tzanger | oh file |
| 00:22.29 | Tako-san | Well, thanks anyways. |
| 00:22.35 | tzanger | where are you my sip guru |
| 00:23.03 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 00:23.08 | x86 | tzanger: i just like the extra line keys :P |
| 00:23.12 | tzanger | x86: yes |
| 00:23.13 | tzanger | agreed |
| 00:23.22 | x86 | tzanger: other than that, they are the same |
| 00:23.34 | tzanger | speakerphone works the same on them |
| 00:23.40 | tzanger | I like the font in the 501 better actually |
| 00:23.54 | x86 | tzanger: for doing testing things like MoH and load testing, 601's are my friend |
| 00:24.33 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 00:24.54 | Trevor_b | anyone ever connect 2 PRI cards directly to eachother for testing between systems? |
| 00:26.02 | JT | Trevor_b: it can be done |
| 00:26.46 | Trevor_b | mainly wondering if i need an RJ-48C cable to be rollover to connect PRI to PRI, or if the cable can just go between 2 cards. |
| 00:27.31 | coppice | you need a crossover cable, the pinout of which is easy to find with google |
| 00:27.49 | Trevor_b | A standard Ethernet Crossover, or a flat cable crossover? |
| 00:27.53 | JT | Trevor_b: you need a PRI crossover cable |
| 00:27.54 | JT | neither |
| 00:27.59 | JT | not a rollover either |
| 00:28.08 | JT | that's rs-232 serial generally |
| 00:28.24 | JT | Trevor_b: google for t1 crossover cable |
| 00:28.33 | Trevor_b | thanks |
| 00:29.17 | javar | Trevor_b: http://www.t-radtalk.com/howto/how_to_make_a_t1_cross_over.htm |
| 00:29.24 | JT | you use cat5, not flat cable too |
| 00:30.25 | JT | http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1 http://www.voip-info.org/wiki/view/crossover+T1+cable |
| 00:31.23 | Trevor_b | cool, got some boxes and the voip-info link as well. Just trying to prove the Sangoma can ring out and in without setting up my T110P on another system. |
| 00:31.36 | Trevor_b | JT: Thanks |
| 00:32.11 | JT | without setting up another pri card? how will you test it then? multi ports? |
| 00:32.35 | Trevor_b | Yeah A104D I assume set one as master and other as slave, but haven't gotten that far ;) |
| 00:33.17 | JT | one as pri_net one as pri_cpe, the one acting as network provides timing, the other receives |
| 00:34.03 | snuffy22 | mm sucks tzanger.. can't think of anythin else off the top of my head |
| 00:34.25 | Trevor_b | Thanks! Yeah the sangoma util made it sort of obvious, updating my other system kernel with the 110P to match the zaptel module. So if i have enough time before monday ill try to route between them as well, and get familiar with Digium PRI. Sangoma made it EASY even with the extra compiling... |
| 00:36.19 | JT | cool |
| 00:36.20 | Trevor_b | HAHAH, im a tard, i forgot I had a T1 super looper..... |
| 00:38.21 | *** join/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 00:38.32 | jkimball4 | how can I continue in the dialplan after a caller hangsup? |
| 00:38.49 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 00:39.24 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-119-123.columbus.res.rr.com) |
| 00:40.09 | javar | with the h exten |
| 00:41.30 | jkimball4 | How would I put that in the dialplan? |
| 00:41.51 | _VoiceMeUp_COM | exten => h,1, DOSOMETHING |
| 00:42.13 | _VoiceMeUp_COM | also i = invalid and o =0 ? |
| 00:42.14 | Strom_M | just without the space after 1, |
| 00:42.29 | Strom_M | o=0 only if called from within Voicemail() |
| 00:43.49 | jkimball4 | Okay, thanks folks. |
| 00:46.22 | dlynes_laptop | Is there any facility for transfer directly to voicemail, except by creating new extensions for that? |
| 00:46.52 | Strom_M | dlynes_laptop: create new extensions :) |
| 00:47.22 | dlynes_laptop | Strom_M: Yeah...I was hoping for something a little more eloquent....such as what's found on Nortel and Panasonic keysystems |
| 00:47.32 | Strom_M | how so? |
| 00:47.40 | dlynes_laptop | Strom_M: There, you hit one button to transfer, and another button to transfer to voicemail |
| 00:47.40 | Strom_M | you can do a vertical service code, for example |
| 00:48.32 | Strom_M | exten _*95!,1,VoicemailMain(${EXTEN:3}) |
| 00:48.38 | BSD_Tech | exten = *. ,1,Goto(Voicemial,${EXTEN:1} |
| 00:48.38 | Strom_M | er |
| 00:48.41 | dlynes_laptop | Strom_M: So, I would hit Transfer-><button>Strom</button> to transfer to Strom's phone, or Voicemail-><button>Strom</button> to transfer to Strom's voicemail |
| 00:48.42 | Strom_M | exten _*95!,1,Voicemail(${EXTEN:3}) |
| 00:49.06 | dlynes_laptop | Strom_M: Then I don't need to tie up additional buttons on the phone for each user for their voicemail |
| 00:49.19 | Strom_M | dlynes_laptop: see the last line I typed |
| 00:49.47 | dlynes_laptop | Strom_M: That still requires a separate speeddial, correct? |
| 00:50.07 | dlynes_laptop | Strom_M: i.e. how would I get the phone to differentiate between that, and the regular extension? |
| 00:51.17 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 00:51.53 | tzanger | arrrrrrrrrrrrrrruuuuuuuururrrgh |
| 00:52.02 | Strom_M | dlynes_laptop: there may be a way to do it, but i'd have to do some research |
| 00:52.08 | dlynes_laptop | Strom_M: My problem is not so much the extra extensions, but more the extra buttons being used up on the phone |
| 00:52.21 | *** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca) |
| 00:52.27 | Slingky | hi guys! does somebody know how to change custom provider info ? it's not supposed to be in sip.conf ? |
| 00:52.36 | Strom_M | yeah, because god forbid your user should actually have to dial anything, right? |
| 00:52.42 | dlynes_laptop | Strom_M: Using this method (which is what I'm currently using), I need a duplicate button for every extension that I want to send directly to voicemail |
| 00:53.01 | dlynes_laptop | Strom_M: Well, I don't expect them to memorize 40 extensions, either |
| 00:53.30 | dlynes_laptop | Strom_M: Which is why they have a side console for the attendant phone |
| 00:53.36 | tzanger | sip.conf must have changed |
| 00:55.11 | *** part/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 00:57.07 | Slingky | tzanger: i check sip.conf and don't see my custom sip provider: acanac inside |
| 00:57.24 | Slingky | is asterisk now putting files elsewhere ? |
| 00:57.43 | *** join/#asterisk Delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 00:57.49 | Slingky | i was using trixbox and all work find. |
| 00:58.06 | Slingky | but i wanted to try asterisknow, it work but incoming call quality is not good |
| 00:58.28 | JT | asterisk now users users.conf too |
| 00:58.45 | JT | and i don't see why the call quality would change unless your codec settings are different |
| 00:58.49 | Slingky | do you have an idea ? i was trying to look at sip.conf, i usually change "from-trunk" on trixbox but i don't see anything like that |
| 00:59.22 | JT | i don't use trixbox |
| 00:59.28 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:59.28 | Slingky | outgoing call quality is good, but ingoing call is bad, it seems to glitch |
| 00:59.57 | Slingky | i heard the welcome message, but background noise like glitching |
| 01:01.18 | dlynes_laptop | Strom_M: Well, thanks for the attempt, anyways |
| 01:02.04 | JT | Slingky: sip only? |
| 01:02.52 | Slingky | yeah, i only have an sip trunk |
| 01:03.20 | JT | "trunk" ;) |
| 01:03.24 | *** join/#asterisk Un1x_Laptop (i=Sean@CPE0015c5651c2f-CM000039da5c60.cpe.net.cable.rogers.com) |
| 01:03.33 | Un1x_Laptop | Hey, guys needed help on setting a conference |
| 01:03.42 | Un1x_Laptop | how do i go on about doing that is there any literature on it? |
| 01:03.45 | Un1x_Laptop | ~conference |
| 01:03.46 | jbot | hmm... conference is IAX2/asterisk@switch-1.nufone.net/4569 |
| 01:03.51 | Un1x_Laptop | ~meetme |
| 01:03.53 | Strom_M | Un1x_Laptop: show application meetme |
| 01:04.01 | Slingky | i see my config in users.conf ! |
| 01:04.02 | Strom_M | Un1x_Laptop: and calm the hell down |
| 01:04.05 | JT | ~thebook |
| 01:04.06 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:04.08 | Un1x_Laptop | Strom_M not app_conference |
| 01:04.23 | JT | Un1x_Laptop: that is meetme |
| 01:04.27 | Un1x_Laptop | Cool |
| 01:04.38 | JT | why would "show application meetme" show app_conference ? |
| 01:05.13 | *** join/#asterisk illsci (n=illsci@evil.hack3rs.org) |
| 01:06.05 | Slingky | do you know how to edit files manually with asterisknow ? |
| 01:06.15 | Slingky | i have to sudo ? which password ? |
| 01:07.55 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:08.19 | Slingky | ok,. i found |
| 01:10.12 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-18-14-219.nycmny.east.verizon.net) |
| 01:13.47 | Un1x_Laptop | ~meetme.conf |
| 01:13.59 | JT | Un1x_Laptop: ... |
| 01:14.11 | JT | there's plenty of documentation on it |
| 01:14.22 | Un1x_Laptop | JT trying to get some samples of meetme.conf |
| 01:14.32 | Un1x_Laptop | wiki doesn't explain well enough |
| 01:14.37 | Un1x_Laptop | http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 01:14.55 | JT | read the book already |
| 01:15.10 | JT | and show application meetme |
| 01:15.22 | JT | the wiki is a secondary source |
| 01:15.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:18.41 | jaxxan | so i can use asterisk to accept an incoming fax and deliver it to someone via email right ? |
| 01:19.35 | JT | not with asterisk alone |
| 01:19.57 | illsci | hey do any of you do asterisk consulting... |
| 01:20.08 | illsci | or want to make some money helping me set something up |
| 01:20.36 | Slingky | do you know how to allow anonymous sip calls ? |
| 01:20.46 | *** part/#asterisk illsci (n=illsci@evil.hack3rs.org) |
| 01:21.04 | Qwell | good job...wait 40 seconds for an answer before leaving |
| 01:21.16 | javar | heh |
| 01:21.17 | jaxxan | haha |
| 01:22.02 | jaxxan | i use meetme when i'm playing network games @ work with friends haha |
| 01:22.37 | jaxxan | i've not played with any moderator features though |
| 01:24.03 | Un1x_Laptop | So jt one question i dont have to put anything in extensions.conf do i for my conference calling? |
| 01:24.21 | [TK]D-Fender | Un1x_Laptop, Of course you do! |
| 01:24.47 | JT | Un1x_Laptop: the book please |
| 01:24.49 | Un1x_Laptop | well im looking at th ebook right now, i see nothing... |
| 01:24.56 | JT | read everything that comes up in a search for meetme |
| 01:25.00 | JT | there's plenty |
| 01:25.01 | Un1x_Laptop | DUDE, im looking at page 324 |
| 01:25.17 | [TK]D-Fender | Un1x_Laptop, perhaps you missed the most basic concept that devices connected to * (SIP, IAX, Zap, etc) do not inherently do ANYTHING TOGETHER MAGICALLY. |
| 01:25.17 | Un1x_Laptop | right now, and show nothing except a sample meetme.conf |
| 01:25.33 | Un1x_Laptop | No [TK]D-Fender i know that but i cant find anything on it |
| 01:25.40 | Un1x_Laptop | what exactly to put in the dailplan.. |
| 01:25.50 | [TK]D-Fender | Un1x_Laptop, Lets say you have 2 SIP phone configured in sip.conf. Can they DO anything at that point? N*NO* |
| 01:25.51 | JT | MeetMe |
| 01:25.59 | Un1x_Laptop | no the cannot |
| 01:26.28 | [TK]D-Fender | Un1x_Laptop, the DIALPLAN controls everything you can CALL, and every APPLICATION used in that call. |
| 01:26.58 | Un1x_Laptop | [TK]D-Fender i understand where your coming from, but there is nothing in the page i see page 324 right now and meetmeconf example is 6 lines and says nothing about extensions.conf... |
| 01:26.59 | [TK]D-Fender | Un1x_Laptop, so go read the instruction on how to CALL MeetMe (show application meetme) <- in * CLI |
| 01:27.02 | Un1x_Laptop | i looked at wiki http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 01:27.11 | Un1x_Laptop | and it doesn't say what to add or how to do add in extensions.conf |
| 01:27.17 | [TK]D-Fender | Un1x_Laptop, if you don't understand the dialplan yet, SCREW MEETME. |
| 01:27.33 | [TK]D-Fender | Un1x_Laptop, Dialplan is your #1 priority. |
| 01:27.48 | Un1x_Laptop | dude either im saying it wrong or your getting me wrong |
| 01:27.56 | Un1x_Laptop | im saying it shows nothing on how to tell it to go to meetme.. |
| 01:27.57 | [TK]D-Fender | Un1x_Laptop, meetme.conf needs maybe *2* lines in it of so. |
| 01:28.03 | JT | i think he's getting you right |
| 01:28.07 | JT | you don't know what to do |
| 01:28.11 | [TK]D-Fender | Un1x_Laptop, EVERYTHING IS IN EXTENSIONS.CONF |
| 01:28.16 | JT | which means you don't understand the purpose of the dialplan |
| 01:28.29 | JT | how can you call a conference if there is no way of getting there? |
| 01:28.29 | [TK]D-Fender | uni1You don't magically END UP THERE. |
| 01:28.31 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 01:28.51 | [TK]D-Fender | Un1x_Laptop, Go learn how to set up things you can DIAL. THEN worry about what they DO. |
| 01:29.21 | Un1x_Laptop | Ok |
| 01:29.36 | Un1x_Laptop | i know everything is in extensions conf |
| 01:29.45 | Un1x_Laptop | i dont get how to direct it to meetme tho i'll lookup dail see if it helps |
| 01:29.45 | Qapf | Anyone here ever use a cisco 7961? some ebay guy sent me one instead of a 7960 and this thing behaves nothing like the 7960's |
| 01:29.50 | Un1x_Laptop | or run through extensions.conf again |
| 01:30.10 | JT | page 117 of the book |
| 01:30.19 | [TK]D-Fender | Un1x_Laptop, and the instructions for how to call MeetMe acan be viewed by entering "show application meetme" in the * CLI. |
| 01:30.21 | JT | you should really learn to use your pdf viewer's search function |
| 01:31.01 | Un1x_Laptop | ok :) |
| 01:31.08 | Un1x_Laptop | Jt i have the book |
| 01:31.12 | Un1x_Laptop | both in paper, and pdf |
| 01:31.15 | Un1x_Laptop | wanna pic? |
| 01:31.18 | JT | well use it! |
| 01:31.32 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 01:31.32 | Un1x_Laptop | its just im confused tonite, i guess my wording is messed up but i'll look through carefully |
| 01:31.33 | JT | what's not annoying me is that this is a very well documented application |
| 01:31.45 | JT | s/not// |
| 01:32.08 | [TK]D-Fender | JT : hardly needs it... |
| 01:32.23 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:32.29 | [TK]D-Fender | JT : puny ass config file, easy dialplan call..... |
| 01:32.46 | RyanW | Hello, how can i get my snom 360 to display the name of the extension i dial when i press the corresponding speed dial button? |
| 01:34.02 | RyanW | I've got the button configured as a "DEST" but when i press it, it shows dialing 121 instead of 121 - Reception |
| 01:36.23 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
| 01:40.51 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 01:41.17 | e-milio | hello all |
| 01:41.46 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-68-39.pskn.east.verizon.net) |
| 01:41.59 | e-milio | I seem to be having a problem that many people is talking about related to waitexten and timeout |
| 01:42.11 | JT | [TK]D-Fender: very well documented, and easy to use :) |
| 01:43.03 | e-milio | [TK]D-Fender: It just sometimes does to recognize all the digits dialed |
| 01:43.39 | e-milio | does not. If I type 232 a little to fast i wont catch all the numbers |
| 01:43.50 | [TK]D-Fender | JT : plenty easy enough :) I had to seariously beat the concepts of Local Channels into Mercestes' head the other day when he couldn't fathon why his queue calls were considered even though his phone was "ringing".... |
| 01:44.00 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:44.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:44.07 | JT | heh |
| 01:44.34 | [TK]D-Fender | JT : Pulling teeth would have been considerably easier and more rewarding... |
| 01:45.11 | [TK]D-Fender | e-milio, What is the call coming in on? |
| 01:45.22 | e-milio | zaptel |
| 01:45.39 | [TK]D-Fender | e-milio, What hardware exactly. FXS / FXO? other? |
| 01:45.58 | e-milio | [TK]D-Fender: 2 dual t with echo cancell |
| 01:46.16 | [TK]D-Fender | e-milio, Digital card? |
| 01:46.22 | e-milio | [TK]D-Fender: yes |
| 01:46.27 | [TK]D-Fender | e-milio, You playing around with gains at all? |
| 01:46.39 | e-milio | [TK]D-Fender:no |
| 01:47.03 | [TK]D-Fender | e-milio, add "relaxdtmf=yes", restart zaptel/Asterisk and give it a whirl |
| 01:47.14 | e-milio | [TK]D-Fender: will do |
| 01:49.23 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:49.42 | *** join/#asterisk distatica (n=distatic@unaffiliated/distatica) |
| 01:50.37 | *** join/#asterisk atomic484 (n=atomicda@74-206-0-80.static-ip.m.telepacific.net) |
| 01:54.33 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 01:59.33 | e-milio | [TK]D-Fender: Thanks, dont know if it worked yet cause server hung up |
| 01:59.38 | [TK]D-Fender | :) |
| 01:59.41 | [TK]D-Fender | :| |
| 01:59.44 | [TK]D-Fender | :/ |
| 02:00.05 | e-milio | [TK]D-Fender: when doin service zaptel restart it just hungs up |
| 02:00.20 | [TK]D-Fender | e-milio, Oh that WILL nuke * if running... |
| 02:00.29 | [TK]D-Fender | e-milio, like ripping the carpet out from under it |
| 02:00.42 | e-milio | [TK]D-Fender: it was stoped |
| 02:01.22 | e-milio | [TK]D-Fender: if i do rmmod the modules it stop but if i do server zaptel stop it hungs up. I have to reboot everytime |
| 02:02.15 | e-milio | [TK]D-Fender:it is a dell poweredge 2950 and te412p if i am not wrong. i tried the two slots, upgrading zaptel etc |
| 02:03.05 | e-milio | seems that digiumg cards do not share love with dell |
| 02:03.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 02:04.53 | *** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 02:05.01 | *** join/#asterisk atomic484 (n=atomicda@74-206-0-80.static-ip.m.telepacific.net) |
| 02:06.32 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 02:08.17 | *** join/#asterisk bbryant_laptop (n=Brett@user-24-214-124-177.knology.net) |
| 02:12.50 | *** join/#asterisk GothAlice (n=amcgrego@209.161.123.42) |
| 02:13.16 | GothAlice | So, I have an extension rule like: _1NXXNXXXXXX,1,Dial(IAX2/user@host/${EXTEN}) |
| 02:13.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:14.07 | GothAlice | I want to use a similar rule for "local" (non 1-XXX-) numbers, by prefixing the "local" number with its long-distance version: a. la: _NXXXXXX,1,Dial(IAX2/user@host/011507${EXTEN}) |
| 02:14.10 | GothAlice | This does not work. |
| 02:14.30 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 02:16.39 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 02:16.58 | [hC] | anyone here use anything for queue statistics reporting? |
| 02:17.11 | [hC] | i like the one asteriskguru offers but its dependent on postgres, which i dont use. |
| 02:25.04 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 02:26.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:26.20 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 02:28.08 | JT | postgres is nice |
| 02:28.21 | JT | but yeah, people should be learning the whole database independance thing by now |
| 02:28.35 | JT | it is hard though if you use sql properly |
| 02:28.35 | DrukenLPY | postgres is sweet |
| 02:28.45 | JT | as pgsql does a lot of things that mysql cannot do |
| 02:28.46 | *** join/#asterisk evoxy (n=rbw@83.212.57.238) |
| 02:29.08 | evoxy | hi, can i use asterisk to send sms messages to my mobile phone? |
| 02:29.44 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 02:30.30 | *** join/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 02:30.36 | JT | not really |
| 02:32.08 | JT | well you can, but combined with other software, generally |
| 02:32.11 | Qapf | does buggymwi=yes work in asterisk 1.2.18? or do i have to get to 1.4.2 |
| 02:32.17 | evoxy | which other software |
| 02:32.18 | evoxy | ? |
| 02:32.36 | JT | look for sms software |
| 02:32.45 | JT | but asterisk is not an sms gateway platform |
| 02:32.55 | Un1x_Laptop | Hey JT connf is working simple problem tho if you can please help |
| 02:32.56 | Un1x_Laptop | [conf1] |
| 02:32.56 | Un1x_Laptop | exten => s,1,MeetMe(6180,i,3355) |
| 02:33.08 | Un1x_Laptop | when people call in and so forth, it doesn't ask them for password.. |
| 02:33.12 | evoxy | JT: do you know of sms software that will let me do that |
| 02:33.14 | evoxy | ? |
| 02:33.32 | JT | kannel os the only open source sms gateway i know of |
| 02:33.36 | JT | is |
| 02:33.40 | JT | depends what you want to do |
| 02:33.51 | JT | there is plenty of software that will let you just SEND an sms |
| 02:34.01 | _VoiceMeUp_COM | or write one |
| 02:34.04 | _VoiceMeUp_COM | or use php |
| 02:34.11 | evoxy | JT: send free sms to mobile phones and *if* possible reply from my phone to it |
| 02:34.12 | Un1x_Laptop | JT help please? |
| 02:34.14 | _VoiceMeUp_COM | php and clickatel.com |
| 02:34.22 | _VoiceMeUp_COM | with fopen does the trick for me |
| 02:34.40 | JT | evoxy: best off interfacing with an sms provider |
| 02:34.43 | evoxy | but don't you have to pay to the phone company? |
| 02:34.49 | JT | yes |
| 02:34.55 | JT | smses are not free, as a rule |
| 02:35.10 | Strom_M | any interfacing you do to the PSTN isn't free, as a rule |
| 02:35.20 | JT | indeed |
| 02:35.25 | JT | some people have other ideas, however |
| 02:35.42 | _VoiceMeUp_COM | yeah and dont go spamming your cell card with a gateway they will send you a bill |
| 02:35.42 | evoxy | JT: so what can i do with kannel then? |
| 02:35.53 | JT | act as a gateway |
| 02:36.02 | JT | please use this useful tool: google.com |
| 02:36.30 | _VoiceMeUp_COM | hey i heard of it |
| 02:36.36 | evoxy | JT: but i still have to pay if i use kannel right? |
| 02:36.37 | JT | cool! |
| 02:36.40 | _VoiceMeUp_COM | usually comes with the bigger site right ? |
| 02:36.45 | _VoiceMeUp_COM | the one with the www in front ? |
| 02:36.47 | JT | evoxy: the answer is "probably" |
| 02:36.51 | _VoiceMeUp_COM | lots of sites have that too |
| 02:36.57 | JT | evoxy: stop being so demanding and do some of your own research |
| 02:37.04 | JT | unless you feel like paying for consulting |
| 02:37.05 | Un1x_Laptop | Someone help please system aint asking for a password.. |
| 02:37.20 | evoxy | if i knew i wouldn't be asking |
| 02:37.34 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:37.43 | JT | irc is not the only way in the world to find out informaiton |
| 02:37.48 | JT | information |
| 02:37.57 | JT | bah |
| 02:38.24 | aptura | evening. okay now this is odd. Asterisk is haning up the call just as it starts to process it. Another anomoly among many. |
| 02:40.48 | BSD_Tech | ok anyone have a issue with 1.4.5 not seeing chan zap |
| 02:41.10 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:41.19 | BSD_Tech | I just built 1.4.5 but my zaptel went into hiding |
| 02:41.23 | JT | heard a few people with zap problems in 1.4.5, but i wasn't sure if they were all nubs |
| 02:41.25 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 02:41.53 | VJFROMGT | port 10000-20000 is forwarded to trix, sip_nat.conf is configured, still getting no audio |
| 02:41.54 | BSD_Tech | ok I will pull svn and test |
| 02:42.10 | *** part/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 02:42.13 | JT | i'd have though pulling 1.4.4 would be more failsafe, but ok :P |
| 02:42.30 | JT | sip_nat must be trix specific |
| 02:42.36 | [TK]D-Fender | VJFROMGT, pastebin it. |
| 02:42.39 | JT | this channel is for asterisk support |
| 02:42.48 | [TK]D-Fender | JT : FreePBX |
| 02:43.03 | JT | right, you know what i mean :) |
| 02:43.08 | [TK]D-Fender | VJFROMGT, AEVERYTHING related to sip.conf |
| 02:43.21 | [TK]D-Fender | JT : But I assasinate on what I see ;) |
| 02:43.54 | Un1x_Laptop | [TK]D-Fender hey quick question when people call in it doesn't ask them for the password.. |
| 02:44.12 | Un1x_Laptop | i read over and over again at the examples the book has did them exactly same even still dont ask for password |
| 02:44.12 | [TK]D-Fender | Un1x_Laptop, Could that phrase tell me any LESS? |
| 02:44.24 | Un1x_Laptop | no im going to past the conf one sec |
| 02:44.37 | [TK]D-Fender | Un1x_Laptop, easy lesson (and most valuable here) : PASTEBIN! www.pastebin.ca |
| 02:44.40 | Un1x_Laptop | exten => s,1,MeetMe(6180,i,3355) |
| 02:44.45 | Un1x_Laptop | its one line.. |
| 02:44.52 | Un1x_Laptop | its not asking for a password.. when people call |
| 02:44.57 | [TK]D-Fender | Un1x_Laptop, I want to see the CLI output as well. |
| 02:45.06 | Un1x_Laptop | ok |
| 02:45.11 | [TK]D-Fender | ALL OF IT |
| 02:45.36 | DrukenLPY | evening [TK]D-Fender |
| 02:45.49 | _VoiceMeUp_COM | sip_nat and sip_additional are trixbox |
| 02:47.23 | [TK]D-Fender | _VoiceMeUp_COM, So its not part of the stock FreePBX setup? |
| 02:47.49 | _VoiceMeUp_COM | ah i dont know |
| 02:47.56 | _VoiceMeUp_COM | i knwknow its there ;) |
| 02:48.00 | _VoiceMeUp_COM | and its messed up |
| 02:48.00 | Un1x_Laptop | [TK]D-Fender |
| 02:48.01 | Un1x_Laptop | http://www.pastebin.ca/582797 |
| 02:49.33 | [TK]D-Fender | Un1x_Laptop, and your meetme.conf please (all of it) |
| 02:49.42 | BSD_Tech | ok svn of branch fixed it so there is a bug in 1.4.6 |
| 02:49.44 | Un1x_Laptop | ok |
| 02:49.47 | BSD_Tech | 1.4.5 |
| 02:50.05 | BSD_Tech | grrr |
| 02:50.09 | _VoiceMeUp_COM | ouf |
| 02:50.18 | Un1x_Laptop | [TK]D-Fender http://pastebin.ca/582810 |
| 02:52.33 | [TK]D-Fender | neato... |
| 02:52.34 | *** join/#asterisk ivanfm (n=ivanfm@c934f322.virtua.com.br) |
| 02:52.35 | [TK]D-Fender | hrm |
| 02:53.35 | JT | ~question |
| 02:53.36 | jbot | What do you get if you multiply six by nine? |
| 02:53.42 | JT | ... |
| 02:53.45 | JT | ~question |
| 02:53.45 | jbot | What do you get if you multiply six by nine? |
| 02:53.49 | Un1x_Laptop | 54 |
| 02:53.55 | JT | query jbot |
| 02:53.58 | JT | oops |
| 02:54.10 | _VoiceMeUp_COM | ~question |
| 02:54.10 | jbot | What do you get if you multiply six by nine? |
| 02:54.13 | Strom_M | 42! |
| 02:54.22 | _VoiceMeUp_COM | what do you get when you mix asterisk and crap stuff ? |
| 02:54.28 | Un1x_Laptop | reepbs |
| 02:54.30 | Un1x_Laptop | *freepbx |
| 02:54.30 | Strom_M | bonerphone |
| 02:54.33 | _VoiceMeUp_COM | Jbot: trix |
| 02:54.35 | _VoiceMeUp_COM | ;) |
| 02:55.06 | JT | ~question |
| 02:55.06 | jbot | it has been said that question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 02:55.09 | JT | fixed |
| 02:55.11 | JT | :) |
| 02:55.40 | Un1x_Laptop | jot |
| 02:55.47 | Un1x_Laptop | [TK]D-Fender so any idea? |
| 02:57.17 | jkimball4 | How can I have someone enter his/her social security number or press 0 for the operator without having a long delay while asterisk waits to see if another digit is pressed? |
| 02:57.28 | Un1x_Laptop | lol |
| 02:57.28 | jkimball4 | Especially since SSNs can start with 0s |
| 02:57.41 | Qwell | mine starts with a 0 |
| 02:57.46 | jkimball4 | Exactly. |
| 02:57.50 | Qwell | oh, you said can, heh |
| 02:57.55 | Qwell | and I lied, it doesn't :p |
| 02:57.59 | Un1x_Laptop | why would you need peoples ssn's anyway |
| 02:58.17 | Qwell | maybe he works for the IRS |
| 02:58.19 | _VoiceMeUp_COM | aint it illegal to ask for that without being gov or bank or medical ? |
| 02:58.25 | Qwell | _VoiceMeUp_COM: no |
| 02:58.29 | jkimball4 | I work for a call center which does federal student loan processing |
| 02:58.37 | _VoiceMeUp_COM | ahahah the day irds will use asterisk ill start coding on it |
| 02:58.41 | _VoiceMeUp_COM | and put a few backdoors |
| 02:58.50 | _VoiceMeUp_COM | s/irds/irs/ |
| 02:59.00 | Qwell | canada revenue uses asterisk |
| 02:59.04 | JT | jkimball4: do social security numbers start with * or # ? |
| 02:59.05 | _VoiceMeUp_COM | wtf ? |
| 02:59.11 | _VoiceMeUp_COM | nah |
| 02:59.16 | aptura | Qwell where did you get that info from? |
| 02:59.26 | Qwell | Juggie works there.. |
| 02:59.27 | aptura | Qwell its called Revenue canada. |
| 02:59.28 | jkimball4 | No, but the execs don't approve of */# being used to reach the operator |
| 02:59.30 | Qwell | whatever |
| 02:59.45 | _VoiceMeUp_COM | hurry..patch pbx.. when calleird is 666-666-6666 and you press ***** on the queue.. it gets your dues to lower... |
| 02:59.54 | aptura | I worked at the Revenue canada building in Vancouver and only saw Mitel PBxs |
| 02:59.55 | JT | jkimball4: then the execs have to live with the delay |
| 03:00.08 | jkimball4 | That's what I thought. Just wanted to double check |
| 03:00.14 | jkimball4 | Thanks everyone who helped. |
| 03:00.17 | JT | silly execs ;) |
| 03:00.32 | Qwell | aptura: *shrug* ask Juggie about it sometime |
| 03:00.35 | Qwell | they use it for something, heh |
| 03:00.37 | aptura | I will :) |
| 03:01.09 | _VoiceMeUp_COM | its still mafia wheter called revenue cananda or canada revenue.. |
| 03:01.44 | _VoiceMeUp_COM | see this guy in usa they tried to shoot him coz he didnt pay is taxes |
| 03:01.49 | aptura | Qwell if you came across a industrial wireless business phone that can enatrate 12 stories or 325 thousand aceres what price would you put on it? The company that makes them say its been used as far as 1.2 km's |
| 03:02.04 | Qwell | $20 |
| 03:02.13 | *** part/#asterisk jkimball4 (n=jerrid@pc006629.mbsc.unomaha.edu) |
| 03:02.23 | Qwell | pretty much all wifi phones suck, in general |
| 03:02.27 | aptura | This is not a service but a real phone and companies are buying them at 950 dollars for the base and handset. |
| 03:02.27 | _VoiceMeUp_COM | http://www.quebedia.com/index.php?module=blog&page=article&id=660 |
| 03:02.32 | JT | 1.2km is easy if it doesn't use wifi |
| 03:02.39 | Qwell | dect? |
| 03:03.05 | aptura | These units sell mostly to farms and ranches |
| 03:03.08 | _VoiceMeUp_COM | the guy is offernig 1 million (value of is estate) if you can show him the law a personna has to pay taxes.. |
| 03:03.12 | aptura | The company is also the mfg. |
| 03:03.14 | _VoiceMeUp_COM | and nobody could as of now |
| 03:03.32 | _VoiceMeUp_COM | so if any of you wiseguys wanna try |
| 03:03.47 | Strom_M | Qwell: i've played briefly with a preproduction UTStarcom wifi/gsm phone here at our distributor in melbourne...it's actually quite nice |
| 03:04.01 | aptura | so it covers up to 3,000 aceres i was way over stating it :) |
| 03:04.19 | _VoiceMeUp_COM | yeah ustar is hot |
| 03:04.27 | _VoiceMeUp_COM | they do sherif vehicles with voip phones now |
| 03:04.32 | _VoiceMeUp_COM | i worked with one of the guys on that |
| 03:05.08 | aptura | what do you mean worked with |
| 03:05.18 | _VoiceMeUp_COM | only thing we hadnt wworking at the end was 911 lol |
| 03:05.22 | _VoiceMeUp_COM | and they where the sherif dept |
| 03:05.33 | aptura | what phones were thay using? |
| 03:05.34 | _VoiceMeUp_COM | worked on the install etc |
| 03:05.43 | _VoiceMeUp_COM | no idea was a gsm gw |
| 03:05.59 | _VoiceMeUp_COM | let me check |
| 03:06.09 | JT | dodgy |
| 03:06.21 | JT | surprised that passed public safety regulations |
| 03:06.29 | JT | i guess they still have their radio system... right |
| 03:06.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:06.50 | aptura | I wonder if there is any wifi phones that will seek out unsecured connection with a dhcp adress and will send a audible beep that it just registered with the pbx. |
| 03:06.58 | _VoiceMeUp_COM | not sure |
| 03:07.11 | _VoiceMeUp_COM | yes |
| 03:07.26 | aptura | Voice which one have you tested? |
| 03:07.27 | _VoiceMeUp_COM | its like.. intercom or something dont remmebr was a quick thing |
| 03:07.33 | riddlebox | does anyone have an asterisk server, with an auto attendant, that I can call to test my asterisk server? |
| 03:07.41 | _VoiceMeUp_COM | they had a few bugs.. i ssh'ed and fixed the peer |
| 03:07.45 | Strom_M | riddlebox: digium does |
| 03:07.58 | Strom_M | IAX2/guest@misery.digium.com/s |
| 03:08.00 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 03:08.07 | _VoiceMeUp_COM | just rmemeber he said was sat based.. or gsm based in car.. |
| 03:08.13 | aptura | _VoiceMeUp_COM but you do not recall the model? |
| 03:08.47 | riddlebox | Strom_M, I want to test my broadvoice account, to do it, I am trying to call another asterisk server that has an auto attendant but when I press anything nothing happens |
| 03:08.57 | _VoiceMeUp_COM | nah |
| 03:09.04 | _VoiceMeUp_COM | and site looks different |
| 03:09.14 | Strom_M | why must it be an asterisk server? why not call something on the PSTN? |
| 03:09.17 | aptura | riddle just use your softphone to call back into your public ip. |
| 03:10.11 | riddlebox | Strom_M, I have called many ivr's and auto attendants on the pstn and it works, but when I call this one in particular, it doesnt work, and the person who runs it says they have dtmfmode=auto in sip.conf |
| 03:10.34 | _VoiceMeUp_COM | no transcoding ? |
| 03:10.36 | Strom_M | that sounds like it's their problem then |
| 03:10.50 | Strom_M | what do you have your dtmf mode set to? |
| 03:10.59 | riddlebox | auto |
| 03:12.25 | riddlebox | I did get it to work if I used dtmfmode=rfc-2833 but I could not hear anyone in the conference room |
| 03:12.46 | Strom_M | riddlebox: if your trunk to broadvoice works for calls to the PSTN, then it's not your problem |
| 03:12.57 | Strom_M | i'd likely blame broadvoice doing something stupid on their end |
| 03:13.13 | riddlebox | yeah, they have broadvoice as well |
| 03:14.14 | _VoiceMeUp_COM | hmm wait i had a guy come with same problem 2 days ago with b v |
| 03:14.15 | *** join/#asterisk johann8384_home (n=johann83@71-81-221-188.dhcp.stls.mo.charter.com) |
| 03:17.15 | riddlebox | _VoiceMeUp_COM, did you resolv it? |
| 03:17.28 | _VoiceMeUp_COM | well he got an accotn with us |
| 03:17.36 | _VoiceMeUp_COM | so yeah he solved it |
| 03:17.40 | _VoiceMeUp_COM | but B v no idea |
| 03:17.52 | Strom_M | flip a coin. if it's heads, BV is fucked up. |
| 03:18.07 | _VoiceMeUp_COM | if its tails.. reflip |
| 03:18.08 | _VoiceMeUp_COM | ;) |
| 03:18.14 | Strom_M | HAHAHAH |
| 03:18.18 | _VoiceMeUp_COM | hehe |
| 03:18.22 | _VoiceMeUp_COM | that a good math joke |
| 03:19.44 | riddlebox | _VoiceMeUp_COM, so you sell sip service then? |
| 03:19.54 | _VoiceMeUp_COM | and iax2 |
| 03:20.05 | riddlebox | what is the website? |
| 03:20.15 | dlynes_laptop | riddlebox: probably voicemeup.com :) |
| 03:20.26 | Un1x_Laptop | hello dlynes_laptop how have you been |
| 03:20.29 | _VoiceMeUp_COM | dlynes_laptop how you knew ? |
| 03:20.31 | Qwell | not .voicemeup.com? |
| 03:20.32 | riddlebox | ahh didnt even look at that... |
| 03:20.35 | dlynes_laptop | Un1x_Laptop: Good...and you? |
| 03:20.36 | Qwell | clearly _ = . |
| 03:20.52 | Un1x_Laptop | havent been better dlynes_laptop i have a simple problem meetme is not asking for pin to join conference |
| 03:20.53 | dlynes_laptop | _VoiceMeUp_COM: Just a WAG |
| 03:21.00 | _VoiceMeUp_COM | hehe |
| 03:21.03 | _VoiceMeUp_COM | good hunch |
| 03:21.04 | Qwell | unless he's blatantly advertising in here, and he wouldn't be doing that |
| 03:21.04 | Un1x_Laptop | i can pastebin both cli output and meetme.conf |
| 03:21.11 | Un1x_Laptop | http://www.pastebin.ca/582797 |
| 03:21.13 | Qwell | *cough* |
| 03:22.08 | Qwell | you know most clients only show the last few nicks in the list? |
| 03:22.13 | riddlebox | _VoiceMeUp_COM, you can port a number right? |
| 03:22.15 | Qwell | so v would be near the end... |
| 03:22.38 | _VoiceMeUp_COM | heehe |
| 03:23.07 | Qwell | wtf is that company twisted works for? |
| 03:23.08 | _VoiceMeUp_COM | has more search terms |
| 03:23.15 | _VoiceMeUp_COM | twisted minds inc |
| 03:23.21 | _voipyvoip_com | Qwell: asteria? |
| 03:23.23 | Qwell | yes |
| 03:23.24 | _VoiceMeUp_COM | subsidiary of shadow corp |
| 03:23.29 | rob0 | huh? |
| 03:23.34 | Qwell | on the consultants page, it's sorted by state and company name |
| 03:23.37 | rob0 | subsidiary? |
| 03:23.45 | _VoiceMeUp_COM | whatever hte english term is |
| 03:23.57 | _VoiceMeUp_COM | had a guy who calle dis biz WTFRC |
| 03:23.59 | _VoiceMeUp_COM | google it |
| 03:23.59 | Qwell | There are a few comments on there, from pissed off people, about them being on top, heh |
| 03:24.05 | rob0 | SwK and probably others work at asteria |
| 03:24.08 | _VoiceMeUp_COM | Who the fuck really cares as acronym |
| 03:24.11 | _VoiceMeUp_COM | nice for banking |
| 03:24.23 | rob0 | I almost did :) |
| 03:24.34 | Qwell | but? |
| 03:24.55 | rob0 | Another opportunity came along. |
| 03:24.59 | Qwell | ahh |
| 03:25.17 | rob0 | I did do a little consulting for them (set up a VPN) |
| 03:25.27 | _VoiceMeUp_COM | which page |
| 03:25.40 | Qwell | ~asterisk consultants |
| 03:25.40 | jbot | a list of Asterisk consultants can be found at http://voip-info.org/wiki/view/Asterisk+consultants |
| 03:25.48 | Qwell | I think |
| 03:26.05 | Qwell | ahahaha |
| 03:26.08 | Qwell | somebody added "AAAVEER" |
| 03:26.25 | Qwell | I call shenanigans! |
| 03:26.43 | _VoiceMeUp_COM | yeah |
| 03:26.44 | [hC] | Im going to rename my company AAAAAAAAaaaaaaaaaVoxterRules. |
| 03:26.44 | [hC] | :) |
| 03:26.47 | _VoiceMeUp_COM | email is bankai |
| 03:26.54 | _VoiceMeUp_COM | should be Relisted in b's |
| 03:26.55 | _VoiceMeUp_COM | ;) |
| 03:27.18 | _VoiceMeUp_COM | like listings on the web.. _________!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!000000000000000000000NAME |
| 03:27.27 | _VoiceMeUp_COM | since its alphanumeric.. the _ then ! then 0 |
| 03:27.45 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 03:31.01 | BSD_Tech | ok so are they going to put out 1.4.6 to fix the chan_zap issue |
| 03:31.40 | Qwell | what chan_zap issue? |
| 03:31.47 | *** join/#asterisk lmoreira (n=lmoreira@201009104198.user.veloxzone.com.br) |
| 03:31.48 | BSD_Tech | I h ave built 1.4.5 on 5 machines and on 5 machines |
| 03:31.58 | JT | a number of people have built 1.4.5 and chan_zap doesn't build |
| 03:32.03 | BSD_Tech | it sees zaptel when you run.configure |
| 03:32.09 | JT | they could all be making the same mistake |
| 03:32.11 | JT | but i dunno |
| 03:32.20 | BSD_Tech | but when you do a make menuselect it does not see chan_zap |
| 03:32.21 | Qwell | did they install the latest LINUX VERSION of zaptel? |
| 03:32.28 | JT | i'd hope so |
| 03:32.36 | Qwell | $20 says no |
| 03:32.36 | JT | otherwise they'd all be nubs |
| 03:32.37 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 03:32.53 | lmoreira | JT: Thanks for help. No more call droping after IRQ issue fixed. |
| 03:33.03 | BSD_Tech | libpri 1.4.0 |
| 03:33.07 | RyanW | can i specify more then one announcement in queues.conf for the announcement thankyou= ? |
| 03:33.09 | BSD_Tech | zaptel 1.4.3 |
| 03:33.10 | JT | lmoreira: nice, were you the one with 4 devices sharing? |
| 03:33.52 | Qwell | BSD_Tech: On Linux? |
| 03:33.54 | lmoreira | yes. I switch the CPU, now one IRQ for each device. |
| 03:33.55 | _VoiceMeUp_COM | try anon1&anon2 |
| 03:33.56 | BSD_Tech | yes |
| 03:33.58 | _VoiceMeUp_COM | maybe |
| 03:34.02 | Qwell | works on my box |
| 03:34.05 | BSD_Tech | all 5 machines are linux |
| 03:34.14 | Qwell | haven't seen a bug report |
| 03:34.15 | RyanW | VoiceMeUp, i tried that in queues.conf but it plays neither. |
| 03:34.25 | lmoreira | 3000 Calls and no drops. |
| 03:34.53 | Qwell | I haven't even seen anything on any of the mailing lists |
| 03:35.05 | Qwell | So no, we haven't fixed it |
| 03:35.39 | JT | lmoreira: how many concurrent calls are you doing on average? |
| 03:35.40 | BSD_Tech | ok |
| 03:35.58 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:36.02 | BSD_Tech | hmmm |
| 03:37.12 | lmoreira | agv 26 concorrent calls |
| 03:37.41 | JT | nice |
| 03:37.42 | lmoreira | 25 agents |
| 03:37.44 | JT | e1 isn't it? |
| 03:38.08 | lmoreira | yes. IRV + 2 queues + 25 agents. |
| 03:39.21 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 03:39.51 | lmoreira | actually, agv is 15 concorrents, and max 25. |
| 03:40.02 | BSD_Tech | well I just built from branches and it works |
| 03:40.17 | BSD_Tech | so I am back in the fast lane |
| 03:40.21 | *** part/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 03:42.12 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:42.41 | _VoiceMeUp_COM | using 1.4 on prod is not too good of a choice mate |
| 03:42.50 | _VoiceMeUp_COM | unless you building test farm |
| 03:43.10 | _VoiceMeUp_COM | coz youll redo your work every other day |
| 03:43.20 | _VoiceMeUp_COM | maybe you paid on installs; ) |
| 03:43.40 | *** join/#asterisk n0skillz (n=kruptos@cpe-72-179-157-195.satx.res.rr.com) |
| 03:44.21 | n0skillz | hey all, very new to asterisk, set up a server and trying to connect with a client, getting an error, can anyone give me a clue what i may be doing wrong :-( |
| 03:44.42 | n0skillz | Registration from '"blah" <sip:blah@sip>' failed for '192.168.1.69' - Device does not match ACL |
| 03:45.08 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:45.11 | _VoiceMeUp_COM | http://i72.photobucket.com/albums/i190/SpankMyMonkey2/spankmymonkey.gif |
| 03:45.12 | _VoiceMeUp_COM | lol |
| 03:45.16 | _VoiceMeUp_COM | good one |
| 03:45.39 | _VoiceMeUp_COM | hmm ACL is usally auth by ip |
| 03:45.47 | _VoiceMeUp_COM | but that for a registration ? |
| 03:45.52 | n0skillz | yeap |
| 03:45.55 | _VoiceMeUp_COM | is that a 1.4.X new hting ? |
| 03:46.04 | _VoiceMeUp_COM | dont auth if its by ip |
| 03:46.04 | n0skillz | first client soft phone im trying to connect |
| 03:46.11 | _VoiceMeUp_COM | then xlite / |
| 03:46.12 | _VoiceMeUp_COM | ? |
| 03:46.25 | _VoiceMeUp_COM | sip accounts -> mod -> dont register to server |
| 03:46.41 | _VoiceMeUp_COM | or put host=dynamic in the sip.conf |
| 03:47.19 | n0skillz | yeap, i have the dynamic in conf already |
| 03:47.37 | n0skillz | but im still learning here...lol so i maay just be sonfiguring soft phone wrong..lol |
| 03:47.57 | n0skillz | using Twinkle for softphone on linux, |
| 03:48.08 | _VoiceMeUp_COM | would be nice .. i know that if you reg box 1 user 1 by ip ..to box A and box 1 user2 to box A .. then once user2 connect you cant recieve calls to user1 |
| 03:48.22 | _VoiceMeUp_COM | never twinkled here.. yet ;) |
| 03:48.28 | n0skillz | haha |
| 03:48.32 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:49.54 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 03:49.57 | kn0x | hey guys |
| 03:49.59 | kn0x | quick question |
| 03:50.14 | kn0x | im trying to pass arguments to VoiceMail()... |
| 03:50.36 | kn0x | im doing VoiceMail(s2525) in my dialplan |
| 03:50.46 | kn0x | the s is to skip the instructions |
| 03:51.01 | kn0x | voicemail without the s workss |
| 03:51.08 | Qwell | show application voicemail |
| 03:51.11 | Qwell | The syntax changed |
| 03:51.19 | kn0x | but when I add the s, i get the following on the cli |
| 03:51.37 | kn0x | <PROTECTED> |
| 03:51.48 | kn0x | show application voicemail no such command |
| 03:51.53 | kn0x | im on 1.4-trunkj |
| 03:51.56 | Qwell | Why are you using trunk? |
| 03:52.12 | kn0x | because thats what I've allways used |
| 03:52.17 | kn0x | whats the new syntax? |
| 03:52.30 | Qwell | well...then you need to read the deprecation warnings when they're given to you |
| 03:52.45 | kn0x | I have looked on voip-info |
| 03:52.47 | Qwell | When you're using trunk, it's possible that there won't be any - ever |
| 03:52.50 | kn0x | it still says to do that |
| 03:52.52 | Qwell | don't use trunk |
| 03:52.57 | *** join/#asterisk bmg505 (n=leon@196.209.181.241) |
| 03:53.01 | Qwell | and update the wiki when you figure it out |
| 03:53.04 | [TK]D-Fender | "core show application voicemail" |
| 03:53.24 | kn0x | thanks TK |
| 03:53.27 | kn0x | i see | |
| 03:53.38 | kn0x | Ill update the wiki with this depreciation |
| 03:53.48 | Qwell | ~deprecation |
| 03:53.52 | Qwell | ~deprecate |
| 03:53.55 | Qwell | stupid bot |
| 03:54.01 | JT | ~hi |
| 03:54.02 | jbot | hello, jt |
| 03:54.05 | JT | haha |
| 03:54.16 | Qwell | I expected a witty response from him on that |
| 03:54.27 | _VoiceMeUp_COM | jbot: who's your daddy ? |
| 03:54.28 | jbot | YOU are, Mr Sexy Pants! |
| 03:54.31 | _VoiceMeUp_COM | yep |
| 03:54.58 | _VoiceMeUp_COM | mod him to asnwer BKW is ;) |
| 03:55.17 | *** join/#asterisk saftsack (n=oliver@p54A7F366.dip.t-dialin.net) |
| 03:55.36 | kn0x | haha |
| 03:56.39 | n0skillz | what is a good or common sip client softphone for linux? |
| 03:57.00 | JT | ekiga |
| 03:58.00 | n0skillz | thanks :-) |
| 03:58.37 | _VoiceMeUp_COM | NOVACOM : Belgian company specialized in Open Source software and VoIP. They are also commercializing a business grade Asterisk IPBX. They are actually my employer and I am managing the VoIP division. |
| 03:58.40 | _VoiceMeUp_COM | on ekiga |
| 03:58.52 | _VoiceMeUp_COM | hmm you can commercialize asterisk business ? |
| 03:59.00 | _VoiceMeUp_COM | tougth that was digiums work |
| 03:59.17 | _VoiceMeUp_COM | Our Products: the NOVACOM IPBX and the NOVACOM IPBX Manager in particular |
| 03:59.21 | _VoiceMeUp_COM | they call it novacom |
| 03:59.34 | _VoiceMeUp_COM | im pretty sure you cantjust use asterisk and rebrand your name in there and sell that for 5k |
| 03:59.36 | JT | it wouldn't come from the business edition i'd think |
| 03:59.39 | Qwell | Do they give out the source? |
| 03:59.46 | JT | unless they paid for a special licence |
| 04:00.21 | _VoiceMeUp_COM | http://www.novacom.be/article.php3?id_article=30 |
| 04:00.22 | Strom_M | you can sell GPL software |
| 04:00.33 | Strom_M | just as long as you comply with the terms of the license |
| 04:00.35 | _VoiceMeUp_COM | no source here |
| 04:01.04 | Qwell | Well, buy it, and ask for the source |
| 04:01.05 | _VoiceMeUp_COM | basically asterisk BE on a rack case.. with a logo .. and puished a gui to manage |
| 04:01.13 | _VoiceMeUp_COM | ahah i don't have time.. |
| 04:01.26 | JT | do you have to buy it to ask for source? |
| 04:01.28 | _VoiceMeUp_COM | start policing.. but just wanted to know. |
| 04:01.28 | Qwell | if they don't give it to you, then yes, they're probably violating the GPL - unless they've licensed the source from Digium |
| 04:01.29 | Qwell | JT: yes |
| 04:01.35 | JT | hrm |
| 04:01.39 | Qwell | well |
| 04:01.40 | Qwell | no |
| 04:01.43 | *** join/#asterisk waKKu (n=wakku@unaffiliated/wakku) |
| 04:01.45 | Qwell | I mean, you can ask |
| 04:01.49 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 04:02.02 | PioneerVM2 | Hey anyone know how to check voice mail when voice mail is triggered |
| 04:02.15 | Strom_M | PioneerVM2: VoicemailMain() |
| 04:02.21 | PioneerVM2 | when it says "leave a msg after the tone", is ther ea way to hit a key to go to it |
| 04:02.36 | PioneerVM2 | rather than having to have a different extension for VoicemailMain |
| 04:02.41 | Strom_M | you can use the a extension |
| 04:02.51 | PioneerVM2 | how does that work |
| 04:03.07 | PioneerVM2 | tried to search on how to do this but was having trouble finding it |
| 04:03.15 | Strom_M | when a user presses * while listening to the greeting, voicemail() will jump to the a extension in the current context |
| 04:03.20 | _VoiceMeUp_COM | i emailed novacom to see |
| 04:03.49 | PioneerVM2 | ahh... |
| 04:04.05 | PioneerVM2 | hmmm ok, so how would i handle multipel mailboxes |
| 04:04.24 | PioneerVM2 | for example, person calls in -- says press 1 for joe and 2 for mary -- 1 goes to joes vm and 2 goes to marys |
| 04:04.39 | PioneerVM2 | if the person presses 1 for joe, and i hit *, how will the a extension know whos box it came from |
| 04:04.42 | _VoiceMeUp_COM | hmm make a dialplan and |
| 04:04.50 | Qwell | _VoiceMeUp_COM: see section 3 of the GPL |
| 04:04.52 | PioneerVM2 | i would have to change contexts i guess |
| 04:04.53 | _VoiceMeUp_COM | exten => 1,Mailbox(100) |
| 04:05.00 | _VoiceMeUp_COM | etc |
| 04:05.09 | Strom_M | PioneerVM2: you could always set a channel variable before jumping to voicemail |
| 04:05.12 | PioneerVM2 | so when i hit 1, have it "goto" a different subgroup |
| 04:05.16 | PioneerVM2 | ahhh yea ok good idea |
| 04:05.22 | kn0x | wow, I feel cool |
| 04:05.27 | Strom_M | welcome to asterisk - don't forget to think |
| 04:05.31 | kn0x | I updated the wiki with something useful :) |
| 04:05.49 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 04:07.05 | Aces1Up | this is maybe an odd question, but i am trying to get demographics for different areas, like calling patterns, for instance I would like to know the prime calling time in a cerain area like 4:00pm - 9:00pm, does anyone know where i can go to find these figures? |
| 04:08.15 | Strom_M | Aces1Up: generally your call traffic will ramp up at 8:30 AM, peak at 10:00 AM, dip down during lunch, peak again at 2:00, then settle in at the same level until about 5:00 PM |
| 04:08.42 | Strom_M | with the 10:00 peak being more severe than the 2:00 |
| 04:09.15 | Aces1Up | strom thanks. |
| 04:10.10 | Strom_M | assuming an office PBX and 8:00-5:00 operating hours |
| 04:10.13 | *** join/#asterisk KeltusEx (n=Keltus@c-76-21-119-166.hsd1.ca.comcast.net) |
| 04:19.36 | PioneerVM2 | what happens if you specify two mailboxes in teh VoiceMail command |
| 04:19.36 | PioneerVM2 | does it save msg in both but use the first one for the announcements? |
| 04:21.08 | BSD_Tech | yes |
| 04:21.11 | BSD_Tech | it should |
| 04:21.20 | BSD_Tech | we call that voicemail blasting |
| 04:21.25 | *** join/#asterisk n0skillz (n=kruptos@cpe-72-179-157-195.satx.res.rr.com) |
| 04:22.03 | PioneerVM2 | cool |
| 04:22.07 | n0skillz | mm, anyone have a sip.conf set up for a softphone? if so can you pastie it :-) think im missing something |
| 04:22.27 | PioneerVM2 | im trying to setup a phone box for home where u can press 1 for me, 2 for wife, 0 for both |
| 04:22.27 | Strom_M | n0skillz: better idea: pastie yours |
| 04:22.37 | PioneerVM2 | so 0 would send to all of us but use mailbox 0 for msgs |
| 04:22.43 | PioneerVM2 | i mean outgoing announcement |
| 04:22.53 | n0skillz | Strom_M: okies :-) |
| 04:23.56 | PioneerVM2 | ok, now heres something where u need to be creative |
| 04:24.10 | PioneerVM2 | I have a phone # where I want to have home msgs like 1 for john, 2 for mary |
| 04:24.19 | *** join/#asterisk saftsack (n=oliver@p54A7F4ED.dip.t-dialin.net) |
| 04:24.24 | Strom_M | PioneerVM2: we call that an IVR menu |
| 04:24.37 | PioneerVM2 | and i also want to forward my cell phone to that voice mail -- my cell phone offers *71(number) where i can forward voice mail from my cell to another number |
| 04:24.55 | PioneerVM2 | now i want my cell to go directly to my mailbox, rather than the general "press 1 for john 2 for mary" |
| 04:25.03 | Strom_M | PioneerVM2: have a separate DID for that |
| 04:25.05 | PioneerVM2 | the only way i can think of doing that is to have a separate incoming DID |
| 04:25.10 | PioneerVM2 | yea ok, thats what i thought |
| 04:25.23 | PioneerVM2 | so i need 3 DIDs, one for general, one for me, one for my wife |
| 04:25.37 | JT | unless you discriminate in callerid |
| 04:25.39 | JT | on |
| 04:25.47 | PioneerVM2 | i already tested that the caller ID of the original caller is passed from the cell |
| 04:26.00 | PioneerVM2 | if my cells caller ID was passed (which would partially suck) then i could do it based upon that |
| 04:26.12 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 04:26.13 | PioneerVM2 | or yes, i can have a dbase of caller IDs to forward but that will be a mess |
| 04:26.20 | Strom_M | PioneerVM2: there's a forwarded flag, but you have to be able to read the SS7 messages to get at it :) |
| 04:26.27 | PioneerVM2 | whats an SS7 msg |
| 04:26.35 | PioneerVM2 | i was going to ask that next |
| 04:26.36 | Strom_M | don't worry about it |
| 04:26.43 | PioneerVM2 | i was curious if there was extra data buried in the forward |
| 04:26.47 | Strom_M | SS7 is the phone network's signaling protocol |
| 04:26.48 | PioneerVM2 | so i knew where it came from |
| 04:26.56 | Strom_M | but you as an end user are never going to get at it |
| 04:26.58 | PioneerVM2 | any way for me to see extra data in the forward? |
| 04:27.00 | PioneerVM2 | ohh ok |
| 04:27.12 | PioneerVM2 | would be cool if i could see it. |
| 04:28.00 | PioneerVM2 | ok next trick, is there a way to have two different voice mailboxes, for example one for me personally and one for me at my business where i sound more formal |
| 04:28.06 | PioneerVM2 | but have the msgs go to the same mailbox |
| 04:28.31 | PioneerVM2 | so one says "Hi this is john... " and one says "You have reached John Smith at Acme Corp" |
| 04:28.39 | PioneerVM2 | but so i dont have to dial into two different voice mails to check |
| 04:28.40 | Strom_M | not that i'm aware of |
| 04:28.49 | PioneerVM2 | i could voice mail blast i guess? |
| 04:29.03 | PioneerVM2 | so VoiceMail (1@home&1@work) or vice versa? |
| 04:29.36 | *** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net) |
| 04:29.38 | kn0x | is there any jitterbuffer support for SIP? |
| 04:29.53 | n0skillz | http://www.pastebin.ca/582935 << any help please :-) |
| 04:30.45 | PioneerVM2 | is there an automated way to empty a mailbox then -- if i voice mail blast, i could automated always delete from one mailbox so it doesnt fill up |
| 04:31.24 | PioneerVM2 | like if i do DID WORK -> VoiceMail (1@work&1@home) and DID HOME -> VoiceMail (1@home&1@work) -- and i just want to check mail at home but have mail going into box at work just get wiped out so it doesnt fill up |
| 04:31.28 | Corydon76-home | delete=yes flag |
| 04:31.35 | PioneerVM2 | so automatically deletes? |
| 04:31.50 | Corydon76-home | It automatically deletes the message after notification |
| 04:32.07 | PioneerVM2 | what is "notification", meaning email or text msg? |
| 04:32.21 | Corydon76-home | It's intended for people who only want to receive voicemail via email notification |
| 04:32.21 | n0skillz | disreguard my pastie above, user error :-) |
| 04:32.28 | PioneerVM2 | ahh nice |
| 04:32.38 | PioneerVM2 | so i can just not notify at all (empty strings) and delete |
| 04:32.41 | PioneerVM2 | cool i think this will work |
| 04:32.41 | nephfl | i am using 8 ananolg extensions on a tdm2400 and im going to switch to 8 polycom 601's... can anyone tell me what that is going to do as far as system resources? will it take alot more cpu/memory? |
| 04:32.55 | JT | less |
| 04:33.31 | nephfl | less? since it doesnt have to do a/d ? |
| 04:33.50 | JT | no zaptel |
| 04:33.58 | JT | i'm not sure if a/d is cpu based |
| 04:34.09 | nephfl | well...i would still be running 4 analog trunks on the tdm |
| 04:34.35 | JT | but no echo cancelling if you were doing any in software |
| 04:34.38 | JT | it's less work anyway |
| 04:35.12 | kn0x | is anyone using jbforce=yes in sip.conf? |
| 04:35.21 | kn0x | I'm not sure if I notice any difference |
| 04:35.55 | nephfl | cool...we'll see how that works...also my memory usage has been slowly increasing, i am up in the 70% from 50 after 4 days..should i be concerned? (im going to throw on some more anyway) |
| 04:36.19 | JT | err |
| 04:36.29 | JT | how are you measuring "memory use"? |
| 04:36.56 | nephfl | just looking at the simple report on trixbox |
| 04:37.15 | JT | memory use of the whole system, or just asteerisk? |
| 04:37.17 | JT | eww trix |
| 04:37.38 | nephfl | havent been watching it close enough to tell what is increasing in size |
| 04:37.55 | PioneerVM2 | hmmm anyone know why i dont need to enter my password now on my mailbox |
| 04:38.04 | PioneerVM2 | it was asking before when i got VoiceMailMain but now it stopped |
| 04:38.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:38.25 | JT | nephfl: pastebin.ca the output of free -m |
| 04:38.47 | Strom_M | trixbox is what happens when you take a perfectly good piece of software and bury it under sixteen metric tons of animal dung |
| 04:39.26 | PioneerVM2 | ahh nevermind, i accidently put |su option on VoiceMailMain lines |
| 04:39.28 | nephfl | yeah...i was thinking i might have done the same with freepbx/asterisk... but im pretty new to asterisk |
| 04:39.57 | Strom_M | freepbx makes up about 10 of those metric tons :) |
| 04:40.05 | nephfl | lol |
| 04:40.12 | nephfl | what would you recommend for GUI? |
| 04:40.16 | Strom_M | none |
| 04:40.19 | Strom_M | use vim |
| 04:40.23 | Strom_M | or emacs or nano or something :) |
| 04:40.49 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:40.52 | Strom_M | the digium gui looks promising but I haven't played with it enough to be able to recommend it yet |
| 04:41.04 | nephfl | if i were admin or on site with this system i could do that... but i wanted something that someone who knows very little about computers could change basic phone system settings |
| 04:41.30 | Strom_M | nephfl: from my experience, if you think it's idiotproof, they'll still manage to fuck it up |
| 04:41.32 | JT | nephfl: so where are those memory usage details? |
| 04:41.39 | kn0x | Strom_M, I am allways getting jitter when playing back this recording- I'm experimenting with jbforce=yes, but that really hasnt helped |
| 04:41.40 | nephfl | i tried asterisknow... that gui is very beta the gui broke the config in the first few minutes |
| 04:42.01 | kn0x | i've put it in slin format |
| 04:42.08 | kn0x | and its going to ulaw |
| 04:42.12 | JT | Strom_M: usually it's more likely to fuck up, because managers aren't as scared as they should be, playing with the pbx settings |
| 04:42.34 | kn0x | anything short of putting it in native, what can I do to get a better transcode? |
| 04:42.37 | Strom_M | kn0x: what is your sample rate? how many bits per sample? |
| 04:42.45 | Strom_M | what else are you doing on the system? |
| 04:42.52 | kn0x | nothing |
| 04:43.04 | kn0x | 400+MB of free memory |
| 04:43.08 | Strom_M | kn0x: what is your sample rate? how many bits per sample? |
| 04:43.08 | kn0x | quad xeon |
| 04:43.14 | kn0x | hang on let me see |
| 04:43.45 | *** join/#asterisk saftsack (n=oliver@p54A7F563.dip.t-dialin.net) |
| 04:44.07 | nephfl | http://pastebin.ca/582963 |
| 04:44.12 | kn0x | Strom, I converted from a ADPCM wav w/ sox foo-in.wav -t raw -r 8000 -s -w -c 1 foo-out.sln |
| 04:44.37 | Strom_M | kn0x: that doesnt tell me how many bits per sample |
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| 04:45.43 | kn0x | 32 |
| 04:45.48 | kn0x | strom 32-bit |
| 04:45.53 | Strom_M | that's the problem |
| 04:45.57 | kn0x | ohh? |
| 04:46.00 | kn0x | 16? |
| 04:46.00 | Strom_M | it should be 16 bits per sample |
| 04:46.03 | kn0x | i see |
| 04:46.05 | kn0x | thank you :) |
| 04:46.08 | JT | nephfl: you're using 157MB of memory and no swap, that's hardly anything |
| 04:46.24 | Strom_M | kn0x: see if that fixes it and let me know |
| 04:46.32 | nephfl | yeah..i know..im not worried about the about the amount of usage...just that it has slowely been increasing |
| 04:46.45 | Strom_M | nephfl: i blame trixbox |
| 04:46.53 | JT | nephfl: 153MB sorry |
| 04:47.05 | JT | nephfl: do you understand how to read memory usage in *nix? |
| 04:47.41 | nephfl | somewhat... i can hack my way through with man pages and google... |
| 04:48.05 | JT | it is normal for linux to use as much ram as possible |
| 04:48.09 | JT | unused ram is wasted ram |
| 04:48.31 | JT | you actual program/system usage is used - buffers - cached |
| 04:48.47 | JT | it buffers and caches files to speed up operations |
| 04:49.00 | nephfl | yeah...i was just noticing that i started at 50% usage idling and have gone to 7x%...if it is cache that is fine...just so long as it levels off |
| 04:49.10 | Aces1Up | what are the measurements called that measure how many calls were completed, etc ASR, ACD?? what are they i forgot. |
| 04:49.28 | JT | it will level off at close to 100% eventually |
| 04:49.34 | JT | if enough files are open |
| 04:49.58 | nephfl | i see... would more memory help the system at all? |
| 04:50.11 | JT | no. |
| 04:50.18 | JT | you aren't even touching swap space |
| 04:50.26 | JT | and there's heaps of memory free |
| 04:50.33 | nephfl | ok, then i wont waste the money... |
| 04:50.54 | Strom_M | Aces1Up: call detail records |
| 04:50.55 | nephfl | im getting into a microsoft/vista mindset about memory...it doesnt have gigs so im worried...lol |
| 04:51.20 | JT | heh |
| 04:53.09 | nephfl | my boss asked me what mb/cpu to buy for this system...i told him i have had best success with linux on something i find in my closet...he bought a new asus mb/amd cpu... after a day wrestling with it i dumped it for a celeron i had left over from last year |
| 04:53.33 | JT | nice |
| 04:53.35 | Strom_M | news flash: all hardware sucks |
| 04:53.51 | Aces1Up | strom_m looking for something like this. if i have 5 outgoing lines, how many users can that support. |
| 04:54.02 | Strom_M | Aces1Up: depends on the needs of your users |
| 04:54.12 | Strom_M | traffic engineering is not a lookup table |
| 04:54.21 | Aces1Up | strom_m hrmm. |
| 04:54.30 | nephfl | 1 person on 5 lines at a time or 1 person on 5 lines right? |
| 04:54.51 | nephfl | lol i meant 5 people on 1 line |
| 04:55.08 | kn0x | Strom_M, its in 16-bit but its still not sounding too great |
| 04:55.19 | Strom_M | are they salespeople? are they managers? do they spend a lot of time writing documents? |
| 04:55.24 | Aces1Up | nephfl, well i don't want people to get busy signals but i want to get the most people sharing those lines. |
| 04:55.39 | Strom_M | kn0x: is it just the one file? |
| 04:55.50 | Aces1Up | strom_m these are home users. |
| 04:56.08 | Strom_M | you've got five analog lines in a single house? |
| 04:56.43 | Aces1Up | strom_m these lines are for a calling card. |
| 04:56.53 | Aces1Up | channels for 1 did,. |
| 04:56.59 | Strom_M | ...you're running a calling card service on analog lines? |
| 04:57.06 | kn0x | Strom_M, more so than others, but I get jitter on some of the asterisk-base sounds |
| 04:57.07 | nephfl | calling card? |
| 04:57.10 | Aces1Up | so if i have 1 did that has 5 channels, wondering how many users that could support. |
| 04:57.27 | Strom_M | Aces1Up: what kind of entrance facilities are you using? |
| 04:57.33 | Strom_M | i think i didn't catch that detail |
| 04:57.35 | JT | analogue lines are shit for anything important |
| 04:57.45 | Aces1Up | strom_m no not using analog at all. |
| 04:57.59 | Strom_M | Aces1Up: so what kind of entrance facilities /are/ you using? |
| 04:58.01 | Aces1Up | home users = non corporate system. |
| 04:58.17 | Aces1Up | just a did forwarding service. |
| 04:58.22 | Aces1Up | to an asterisk box. |
| 04:58.24 | nephfl | ? |
| 04:58.32 | Strom_M | Aces1Up: you're not answering my question |
| 04:58.35 | Strom_M | I can ask again... |
| 04:58.42 | Strom_M | what kind of entrance facilities are you using? |
| 04:58.57 | nephfl | >aces i dont think your scenario was clear |
| 04:58.59 | Aces1Up | strom_m forgive me, i do not know what entrance facilities are. |
| 04:59.07 | JT | think about it |
| 04:59.09 | JT | calls enter |
| 04:59.11 | JT | somehow |
| 04:59.25 | Aces1Up | enter via a DID i have purchased with 5 channels. |
| 04:59.38 | Aces1Up | lol |
| 04:59.44 | Aces1Up | heh sorry. |
| 04:59.46 | Strom_M | is it T1? is it SIP? is it ISDN BRI? what the fuck is it, man? |
| 05:00.02 | Aces1Up | lol heh ohhhh yeh.. its sip ... |
| 05:00.04 | JT | how to be obtuse 101 |
| 05:00.12 | Aces1Up | lol |
| 05:00.22 | Aces1Up | man i that gave me a great laugh thanks. |
| 05:00.24 | nephfl | sip? can you purchase a number of channels on sip? |
| 05:00.33 | Strom_M | nephfl: depending on the provider |
| 05:00.49 | Strom_M | Aces1Up: a better question to ask is this |
| 05:00.59 | Strom_M | how many users are you going to be supporting |
| 05:01.00 | nephfl | ive only dealt with vtwhite for wholesale sip..they will allow whatever you can handle |
| 05:01.13 | JT | nephfl: same, i only get unlimited |
| 05:01.35 | Aces1Up | nephfl how are they, i have been looking at them. |
| 05:02.10 | Strom_M | Aces1Up: ............how many users are you going to be supporting? |
| 05:02.13 | Aces1Up | strom_m i would like to support as many users as possible. I am trying to figure the capacity 5 channels can handle, |
| 05:02.27 | Strom_M | Aces1Up: you're going at it backwards |
| 05:02.30 | JT | 2/5ths of not much |
| 05:02.32 | nephfl | easy to set up...no big maintenance or monthly fees..really low rates. but ive gotten a bit of delay...but for my use at the time that was fine |
| 05:02.40 | Aces1Up | so if i have say 30 users will what percentage of people will be getting busy signals using those 5 channels. |
| 05:03.04 | Strom_M | first you're supposed to figure out how many users you want to handle, then you're supposed to figure out their calling patterns, then you run your erlag formula |
| 05:03.10 | Strom_M | s/erlag/erlang/ |
| 05:03.23 | Strom_M | and only THEN do you sit down and say "ok, now what service do I need to buy?" |
| 05:03.47 | *** join/#asterisk littleball (n=littleba@bb219-74-144-83.singnet.com.sg) |
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| 05:03.52 | Aces1Up | hrmm. |
| 05:03.56 | Aces1Up | ok. |
| 05:04.18 | kn0x | Strom_M, it seems I'm getting these transcode problems on ALL audio files |
| 05:04.43 | Strom_M | kn0x: ok - what hardware do you have and what are you running on the box? |
| 05:04.55 | kn0x | im on a vps |
| 05:05.00 | kn0x | unlimited cpu |
| 05:05.03 | Strom_M | well there's your first mistake |
| 05:05.09 | kn0x | on quad xeons |
| 05:05.12 | kn0x | 512mb ram |
| 05:05.15 | JT | ... |
| 05:05.16 | JT | vps |
| 05:05.20 | Strom_M | yeah, but it's a VIRTUAL SERVER |
| 05:05.23 | JT | of course it will be stuffed |
| 05:05.26 | Strom_M | duh |
| 05:05.33 | Strom_M | they can't guarantee unlimited cpu |
| 05:05.34 | JT | asterisk does not play nicely with a VPS |
| 05:05.40 | littleball | hi, i want to have a tool to monitor asterisk. WHen the service process is stopped (due to bug etc reason), the monitoring tool should auto restart the service. What tool can do this? |
| 05:05.46 | kn0x | well they say they don't overload the vpses |
| 05:05.47 | JT | also i/o contention is a major issue |
| 05:05.59 | JT | and some VPSes are not optimal for running asterisk at all |
| 05:06.02 | JT | like UML |
| 05:06.03 | kn0x | they're allways at < 50% load |
| 05:06.07 | JT | useless for running asterisk |
| 05:06.08 | Strom_M | kn0x: it'll work fine for apache, but asterisk is way more sensitive |
| 05:06.09 | nephfl | >aces so, what is it you were working on? |
| 05:06.15 | JT | you clearly aren't understanding the issues |
| 05:06.16 | kn0x | im on OpenVZ |
| 05:06.20 | JT | nothing to do with load |
| 05:06.23 | Strom_M | kn0x: you're not listening |
| 05:06.31 | Strom_M | asterisk does not go on a VPS |
| 05:06.32 | Strom_M | period |
| 05:06.34 | Strom_M | end of story |
| 05:06.38 | kn0x | Strom, well I don't have an alternative with a budget of $20/month |
| 05:06.43 | *** join/#asterisk saftsack (n=oliver@p54A7E9B9.dip.t-dialin.net) |
| 05:06.47 | JT | if you want asterisk to work properly, try dedicated server or co-lo |
| 05:06.54 | JT | xen you might get some success |
| 05:07.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:07.07 | Strom_M | you can't run a pbx on $20 a month :) |
| 05:07.16 | Strom_M | at least not in a colo facility |
| 05:07.29 | Strom_M | maybe in your house |
| 05:07.31 | JT | i'm running one for double that |
| 05:07.32 | Aces1Up | nephfl can i PM ya? |
| 05:07.33 | JT | at co-lo |
| 05:07.35 | JT | :) |
| 05:07.42 | kn0x | JT, where do you get colo? |
| 05:07.56 | JT | kn0x: in sydney, australia |
| 05:08.00 | nephfl | if you like...but im not talking dirty to you...im not that kinda guy |
| 05:08.11 | Aces1Up | lol |
| 05:08.14 | Strom_M | <nephfl> oh oh yes, yes, dial me harder |
| 05:08.25 | kn0x | eww strom |
| 05:08.30 | Strom_M | oh come on |
| 05:08.33 | Strom_M | it's a phone joke, silly |
| 05:08.37 | nephfl | lol |
| 05:09.58 | kn0x | please no sexually-oriented conversations, minority orientations included... there might be women around |
| 05:10.40 | JT | are you serious? |
| 05:10.42 | Strom_M | what was so sexual about pressing touch-tone buttons increasingly harder? |
| 05:10.56 | Strom_M | you must have a far dirtier mind than I |
| 05:12.12 | kn0x | no offense, but its WHO says it :| |
| 05:12.36 | JT | eh? |
| 05:12.42 | Strom_M | you get offended too easily then :) |
| 05:12.52 | kn0x | im not offended |
| 05:13.05 | Strom_M | this is the Internet, land of catsex, dogballs, dead hookers, and bonerpills |
| 05:13.07 | kn0x | im just happy all the time |
| 05:13.09 | kn0x | weeeee............ |
| 05:13.19 | kn0x | eww kittycats? |
| 05:13.29 | JT | kittens are made somehow |
| 05:13.42 | Strom_M | it's the catcycle |
| 05:13.55 | Strom_M | cat -> catsex -> kitten -> cat |
| 05:14.01 | JT | so yeah, do you want co-lo in sydney, kn0x? :P |
| 05:14.03 | kn0x | alright alright... |
| 05:14.10 | kn0x | JT, yeah I might |
| 05:14.19 | kn0x | depend how much shipping is from Chi-town |
| 05:14.24 | kn0x | what specs you get? |
| 05:14.36 | JT | where abouts are you? |
| 05:14.44 | JT | space in a rack, 100Mbit/s switch port |
| 05:14.44 | kn0x | chicago, usa |
| 05:14.53 | kn0x | JT, transfer? |
| 05:14.56 | JT | surely you could find something over there |
| 05:14.58 | Strom_M | kn0x: you know, Australians /do/ know where Illinois is :) |
| 05:15.03 | JT | pay per GB |
| 05:15.08 | JT | your data would be cheaper |
| 05:15.43 | kn0x | ohh i see |
| 05:15.51 | kn0x | i dont know anything about AU tho |
| 05:15.51 | JT | but it's 160ms from CA |
| 05:16.01 | JT | 10ms from me |
| 05:16.03 | JT | :) |
| 05:16.18 | Strom_M | kn0x: the best advice I can offer you with regard to phones is this |
| 05:16.23 | Strom_M | DON'T BE A CHEAPASS |
| 05:19.07 | Strom_M | sure. you might save twenty dollars, but you'll spend more than that on headache pills, chiropractors, jacuzzis, fancy chairs, back supports, etc etc etc |
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| 05:53.36 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 05:55.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 05:56.43 | Strom_M | #asterisk - so quiet you can hear a pin drop |
| 05:57.09 | SplasPood | #asterisk - so quiet you can hear a pin drop, if not for all that echo |
| 05:57.38 | wunderkin | #asterisk - the new Sprint |
| 05:58.33 | wunderkin | #asterisk - now AT&T |
| 05:58.34 | wunderkin | ha ha |
| 06:00.29 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:03.36 | flenders | wow, guys, I just have to say polycom's speakerphones are awesome! |
| 06:03.57 | Strom_M | i think people have been saying that in #asterisk for years no |
| 06:04.01 | Strom_M | s/no/now/ |
| 06:04.29 | JT | flenders: i will be sending you the cleaning bill if my IP501 comes back with white stains on it |
| 06:04.32 | flenders | yeah, but until you test it... |
| 06:04.51 | flenders | JT: HAHAHAHA I'll keep my boss away from it |
| 06:04.53 | Strom_M | JT: haha |
| 06:05.41 | ai-a | s/a/b/g |
| 06:05.51 | wunderkin | l/m/n/o/p |
| 06:06.01 | Strom_M | b/o/n/e/r/s |
| 06:06.11 | wunderkin | c/a/t/s/e/x//? |
| 06:06.18 | Strom_M | d/o/g/b/a/l/l/s |
| 06:06.24 | JT | flenders: so is the linksys speakerphone quite different? |
| 06:07.08 | flenders | JT: quite? |
| 06:07.14 | flenders | mate, you can't compare them |
| 06:07.26 | JT | hrm fair enough |
| 06:07.26 | flenders | crystal clear with the 501 |
| 06:07.45 | flenders | JT: can I keep it for another week? |
| 06:07.56 | flenders | we'll be ordering one today or monday |
| 06:08.03 | JT | yeah i suppose, just don't keep too addicted to it :P |
| 06:08.08 | JT | buying from the us? |
| 06:08.11 | flenders | and as soon as we get it, I'll bring yours back |
| 06:10.05 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:11.03 | *** join/#asterisk saftsack (n=oliver@p54A7EF3D.dip.t-dialin.net) |
| 06:11.30 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:12.40 | flenders | JT: yeah, we'll probably get it delivered on our SF office |
| 06:12.53 | JT | good idea |
| 06:13.07 | JT | telephonydepot.com is one of the cheaper places |
| 06:13.14 | JT | and reputed to be not shit like voipsupply |
| 06:13.15 | flenders | thanks mate |
| 06:13.21 | wunderkin | heh heh |
| 06:14.17 | flenders | what's the difference between the PoE bundle and the other one? |
| 06:14.30 | flenders | JT: is yours PoE bundled? |
| 06:15.37 | JT | flenders: the 501 is an older model, and you need the correct cable to use an external DC supply, or a PoE cable, to connect to a PoE switch |
| 06:15.38 | flenders | and also, speaker quality on the 430 should be the same, no? |
| 06:15.49 | JT | flenders: it's the dc model, standard |
| 06:15.58 | JT | won't do PoE without the other cable |
| 06:16.12 | JT | the 430 is newer, but i assume the speakerphone is as good |
| 06:16.13 | flenders | ok, so yours is the new one? |
| 06:16.21 | JT | 501 is higher end |
| 06:16.55 | JT | 430 is smaller screen, only 2 line appearances |
| 06:17.03 | JT | no greyscale |
| 06:17.27 | flenders | I don't think we need greyscale |
| 06:17.30 | flenders | :) |
| 06:17.47 | flenders | phone will be used 2 or 3 times a day only on conference calls |
| 06:19.04 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 06:19.06 | zeeesh | hi all |
| 06:20.22 | flenders | so I guess 430 would be alright |
| 06:20.25 | JT | flenders: not buying a conference phone? |
| 06:20.32 | flenders | JT: no need mate |
| 06:20.43 | JT | hmm ok |
| 06:20.51 | flenders | it works just perfect |
| 06:21.00 | JT | the 501 is not that different from the 430 from the us |
| 06:21.09 | JT | otherwise it's almost the same to buy the 430 from ay |
| 06:21.10 | JT | au |
| 06:21.35 | flenders | wouldn't it be cheaper there? |
| 06:21.42 | flenders | I'll call westan now |
| 06:22.13 | JT | the 430 is similar after shipping from memory |
| 06:22.17 | JT | but by all means do the maths |
| 06:30.48 | flenders | damn they're slow at westan |
| 06:30.54 | flenders | been on hold for 8 minutes |
| 06:31.07 | zeeesh | <JT>: want little help . .i was using simple dialplan for call out .. exten => _X.,1,Answer ,,, _X.,2,WaitExten _X.,3,Dial(SIP/${EXTEN}@ibasis) .. |
| 06:31.07 | flenders | all I wanted was a simple quote |
| 06:31.23 | zeeesh | it was working well .. |
| 06:32.27 | zeeesh | <JT>: now i try to change extension _X., with extension => s,1,Answer .. .there is no even any error msg .. |
| 06:32.34 | zeeesh | will u pls guide |
| 06:33.33 | flenders | zeeesh: no, wrong |
| 06:33.51 | JT | flenders: weird, i didn't have to wait |
| 06:33.56 | flenders | zeeesh: when you do '_.X' you're maching a template |
| 06:34.04 | JT | zeeesh: why are you doing "<JT> ? |
| 06:34.08 | flenders | JT: just asked them ring me back when they find the price list |
| 06:34.36 | zeeesh | sorry |
| 06:34.46 | flenders | JT: he's a secret admirer mate |
| 06:35.19 | zeeesh | how got u |
| 06:36.12 | JT | flenders: i have the price from a couple months ago here |
| 06:36.19 | flenders | zeeesh: so, basically you need to say 'if you dial this number, do this', like '_.1800XXXXXX', youre matching a 10 digit number that starts with 1800 |
| 06:36.41 | flenders | I think you sent me that before, but I didn't save it anywhere |
| 06:36.49 | JT | $240ex in single quantities |
| 06:37.01 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 06:37.04 | flenders | so, ~280 |
| 06:37.25 | flenders | or 270 |
| 06:37.43 | flenders | might be around 200 from the US |
| 06:38.11 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:38.34 | flenders | zeeesh: got it? |
| 06:40.11 | wunderkin | silly flanders, the dot does not go at the beginning |
| 06:40.17 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:40.30 | flenders | JT: seen a 550 yet? |
| 06:40.40 | JT | flenders: $264 |
| 06:40.43 | JT | inc tax |
| 06:40.48 | flenders | and delivery? |
| 06:40.52 | JT | dunno about that |
| 06:41.04 | flenders | that's why I was guessing 280 |
| 06:41.09 | JT | hmm |
| 06:41.16 | JT | 550, isn't very economical |
| 06:41.23 | JT | better to go 601 or 650 |
| 06:41.59 | JT | flenders: if you get it through a service other than USPS, you might have to pay gst and possibly duty |
| 06:45.06 | flenders | JT: 240 ex |
| 06:45.21 | flenders | and 501 is 281 ex |
| 06:46.26 | JT | so no changes in price whatsoever :) |
| 06:46.37 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:46.50 | flenders | :) |
| 06:49.21 | zeeesh | <flenders>: i really don't concern with 's' |
| 06:50.03 | zeeesh | by using _X., my this line is not working how .. to adjust this exten => 5557,1,Meetme,54321? |
| 06:50.36 | zeeesh | exten => 5557,4,Meetme,54321 i used it at 4th .. |
| 06:51.53 | zeeesh | at line exten => _X.,3,Dial(SIP/${EXTEN}@ibasis) ,,, it working but at exten => 5557,4,Meetme is not working |
| 06:52.14 | flenders | JT: the sales rep from westan tried to sell me these ones: |
| 06:52.16 | flenders | http://www.zultys.com/index.jsp?tab=productdetail&product=zip4x4l&detail=datasheet-zip4x4l&type=phones |
| 06:53.26 | JT | hehe, don't let the sales guy nubify you ;) |
| 06:53.35 | flenders | I know |
| 06:53.37 | flenders | hahah |
| 06:53.47 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 06:54.09 | flenders | he started saying, 'well, we have this other phones which everyone is talking about now' |
| 06:54.12 | flenders | hahaha |
| 06:56.00 | JT | what a tool |
| 06:56.16 | JT | they're the nsw distributors for polycom, they shouldn't do shit like that |
| 07:00.09 | flenders | damn, on hold again |
| 07:00.12 | flenders | :) |
| 07:00.23 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 07:04.48 | JT | you still talking to them? |
| 07:04.49 | flenders | have you ordered from them before? |
| 07:05.04 | flenders | they'll send me a fax |
| 07:05.22 | JT | nup, i haven't |
| 07:19.33 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 07:20.02 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:39.34 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 07:58.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:00.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 08:02.04 | walhala | io |
| 08:09.38 | Pagautas | hi |
| 08:10.05 | Pagautas | is it able to run chan_h323 under jail in freebsd? |
| 08:13.09 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 08:14.37 | s0ck | morning :) |
| 08:14.58 | *** part/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 08:15.02 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 08:15.07 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 08:19.38 | *** join/#asterisk Strom_M (n=strom@dsl-202-173-183-69.vic.westnet.com.au) |
| 08:31.32 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 08:32.39 | *** join/#asterisk lesouvage (n=lesouvag@213.197.251.110) |
| 08:35.24 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 08:36.23 | FreezeS | how can I kill a sip channel ? |
| 08:36.48 | *** join/#asterisk swiftkick (i=x@c-24-12-180-247.hsd1.il.comcast.net) |
| 08:37.04 | Strom_M | soft hangup SIP/whatever |
| 08:37.04 | swiftkick | hi |
| 08:37.09 | Strom_M | boner |
| 08:37.20 | Strom_M | the hi/boner protocol (RFC 6882) |
| 08:37.42 | swiftkick | does sccp still exist for 1.4.x asterisk? |
| 08:37.48 | Strom_M | yes |
| 08:37.51 | Strom_M | chan_skinny |
| 08:38.51 | swiftkick | everything i see has dates in the 2005 range |
| 08:38.52 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:38.52 | FreezeS | Strom_M: thanks |
| 08:39.07 | Strom_M | swiftkick: chan_skinny |
| 08:39.14 | Strom_M | or is it chan_sccp |
| 08:39.20 | Strom_M | whichever qwell rewrote :) |
| 08:39.21 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 08:39.25 | swiftkick | chan_sccp = appears to have stopped development back in 2005 |
| 08:40.03 | Strom_M | well then it's chan_skinny |
| 08:40.14 | swiftkick | how about this question then: anyone know any utils to help manage the button/URL/directory for Cisco 79xx phones in SIP mode? |
| 08:40.31 | Strom_M | swiftkick: are you even listening to me, man? |
| 08:40.32 | Strom_M | CHAN |
| 08:40.34 | Strom_M | SKINNY |
| 08:40.38 | Strom_M | ^^^^ |
| 08:41.14 | swiftkick | Strom_M, you seem not to be reading the screen, I asked a second question, but I understand the asterisk community is full of people of varying degrees of politeness and aptitude. |
| 08:42.04 | swiftkick | neither of which you seem to posess to any appreciable degree. |
| 08:42.14 | Strom_M | well, fuck you too |
| 08:42.35 | Strom_M | (less eloquent but certainly more direct) |
| 08:49.46 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:54.31 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:55.49 | *** part/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 08:55.54 | Uatec | calm down guys |
| 08:56.26 | Uatec | hey, does anybody know how i can dump to configuration of my SPA922 in to a file? |
| 08:56.37 | Strom_M | apparently he didn't appreciate the hi/boners protocol |
| 08:56.51 | Strom_M | Uatec: as far as I'm aware, that's not possible |
| 08:57.09 | Uatec | yes it is, a guy in my office did it yesterday |
| 08:57.16 | Strom_M | oh |
| 08:57.23 | Uatec | but he's in a meeting now |
| 08:57.24 | Uatec | :( |
| 08:57.30 | Uatec | :) |
| 08:57.31 | Uatec | lol |
| 08:57.54 | Uatec | well, what i really want to know, is how to change the configuration file so i can provision the phone to turn OFF call waiting... |
| 09:01.18 | Strom_M | well, you got me...i only have a 942 and it's on the other side of the planet |
| 09:01.24 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 09:01.56 | Uatec | lol |
| 09:02.21 | Uatec | did you provisin it? |
| 09:02.22 | s0ck | anyone had problems compiling zaptel on a >2.6.18 kernel? |
| 09:02.37 | Strom_M | Uatec: no, i just used the browser |
| 09:02.49 | s0ck | just found this: http://www.linuxquestions.org/questions/showthread.php?t=506363 |
| 09:02.52 | Strom_M | s0ck: which version - the latest 1.4 release? |
| 09:02.55 | s0ck | no |
| 09:02.58 | s0ck | 1.2 |
| 09:02.59 | Uatec | ah |
| 09:03.02 | Strom_M | hm |
| 09:03.08 | Uatec | thing is, strom, i need to provision 16 phones today |
| 09:03.13 | Uatec | and possibly 70 more next month |
| 09:03.17 | Strom_M | ouch. |
| 09:03.20 | Uatec | i'm not doing THAT by the web browser |
| 09:03.25 | Uatec | i can write some software to do it |
| 09:03.29 | Strom_M | haha, yeah, that's a pain |
| 09:03.29 | Uatec | using a barcode scanner i have here |
| 09:03.35 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 09:04.01 | Uatec | but i need to know how to configure the config file for call waiting, and find the correct dial tone |
| 09:04.56 | s0ck | Strom_M: you're using 1.4 | k >18 fine? |
| 09:06.17 | Uatec | thing is, i have a copy of the config file |
| 09:06.27 | Uatec | but i can't find the string to disable call waiting |
| 09:06.54 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 09:07.40 | Strom_M | s0ck: i forget my specific kernel version |
| 09:07.45 | Strom_M | IIRC it's 1.2.16 or something |
| 09:08.03 | Strom_M | Uatec: I'm looking for a guide online... |
| 09:09.37 | Uatec | even finding a list of the british call progress tones |
| 09:10.43 | penguinFunk | s0ck: get latest zaptel |
| 09:10.54 | s0ck | i have... |
| 09:11.08 | Strom_M | is there a call progress tone that means "Irritatingly, we've closed at 5:30 PM just like every other store in London"? |
| 09:11.12 | penguinFunk | [10:02:37] <s0ck> 1.2 |
| 09:11.18 | s0ck | ah |
| 09:11.20 | s0ck | you mean go to 1.4? |
| 09:11.36 | penguinFunk | get zaptel 1.4.2 |
| 09:11.49 | Strom_M | 1.4.2 is not the latest zaptel |
| 09:11.50 | s0ck | i assumed that went hand in hand with asterisk 1.4* |
| 09:11.52 | s0ck | works fine with 1.2? |
| 09:12.07 | Strom_M | s0ck: which 1.2 release do you have? |
| 09:12.23 | penguinFunk | oh i see, your using old version of asterisk? |
| 09:12.24 | s0ck | asterisk 1.2.19 / zaptel 1.2.18 afaik |
| 09:12.45 | s0ck | i guess i could move to 1.4 |
| 09:12.50 | s0ck | my perception was that it's devel |
| 09:12.52 | *** join/#asterisk jzweng (n=john@85-125-14-210.static.xdsl-line.inode.at) |
| 09:12.54 | s0ck | and 1.2 is stable |
| 09:12.59 | Strom_M | s0ck: try upgrading to 1.2 svn branch and see if that fixes the problem |
| 09:13.02 | s0ck | cept 1.2 doesn't appear to work very well :P |
| 09:13.12 | Strom_M | 1.4 is stable-ish; 1.2 has worked fine for a long time now |
| 09:13.30 | penguinFunk | we have used asterisk 1.4 for nearly a year works perfect for us |
| 09:13.49 | Strom_M | 1.4 hasn't even been out nearly a year |
| 09:13.53 | penguinFunk | i would recommend trying it |
| 09:14.06 | penguinFunk | well since it came out |
| 09:14.10 | penguinFunk | 6months+ |
| 09:14.52 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 09:14.53 | Strom_M | Uatec: try this |
| 09:14.54 | Strom_M | http://www.netvoice.ca/sipura/SipuraSPAAdminGuidev2.0.9.pdf |
| 09:16.15 | s0ck | btw |
| 09:16.24 | s0ck | it's not zaptel that doesn't compile... it does |
| 09:16.33 | s0ck | but i need to make b410p from the same makefile |
| 09:16.40 | s0ck | which now fails with the latest kernel |
| 09:16.48 | Strom_M | so really, it's an misdn problem |
| 09:16.55 | s0ck | i guess |
| 09:16.59 | *** join/#asterisk feardotnet (n=feardotn@58.186.249.101) |
| 09:17.29 | s0ck | linux/config.h deprecated as of 2.6.19 it seems |
| 09:17.48 | penguinFunk | cant you got back to an older kernel to fix it? |
| 09:18.10 | s0ck | this is just a royal pain in the ass tbh |
| 09:18.12 | penguinFunk | or try newer packages to go with the newer kernel |
| 09:18.14 | feardotnet | hi every body |
| 09:18.34 | Strom_M | no, clearly every linux system must run the absolute latest hemorrhaging-edge kernel, lest everyone point at you and laugh |
| 09:18.48 | penguinFunk | lol |
| 09:18.48 | Strom_M | hi dr nick |
| 09:19.20 | s0ck | Strom_M: appears to be a known issue with receiving calls on misdn with anything less than 2.6.15 |
| 09:19.27 | s0ck | i thought, what the hell, i'll grab the latest |
| 09:19.40 | Strom_M | well, how about....2.6.16! |
| 09:19.45 | Strom_M | :) |
| 09:19.46 | s0ck | got 18 to try now :P |
| 09:20.07 | s0ck | i should join the circus |
| 09:20.11 | s0ck | kernel juggler |
| 09:21.30 | Strom_M | clearly i shouldn't be allowed to be this silly in #asterisk |
| 09:21.52 | s0ck | what time is it in .au then |
| 09:22.02 | Strom_M | 7:21 PM in melbourne |
| 09:22.23 | s0ck | 10:22 in sunny wales |
| 09:22.27 | s0ck | am |
| 09:23.01 | Strom_M | i've got one night left here before I go back to los angeles...just waiting for freeway traffic to die off before I head into town |
| 09:23.21 | s0ck | jet set! |
| 09:23.33 | Strom_M | economy class |
| 09:25.18 | *** join/#asterisk VietDuc (n=VietDuc@58.186.249.101) |
| 09:25.42 | feardotnet | hello VietDuc |
| 09:25.47 | VietDuc | hello ku |
| 09:25.51 | VietDuc | ;)) |
| 09:25.58 | Strom_M | hello boner |
| 09:26.43 | feardotnet | I have problem with DTMF detect pick_up |
| 09:26.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 09:28.51 | penguinFunk | Digium has released a stable, production ready version of Asterisk 1.4, making the earlier posts about Asterisk 1.4 still being beta quality now obsolete. |
| 09:28.51 | penguinFunk | Digium STRONGLY ENCOURAGES EVERYONE to migrate to Asterisk 1.4 |
| 09:29.41 | penguinFunk | http://www.asterisk.org/node/48359 |
| 09:30.48 | Strom_M | that's not in the link you sent |
| 09:30.54 | s0ck | aye |
| 09:31.00 | s0ck | im sure they'd love more guinea pigs :P |
| 09:31.10 | Strom_M | i run 1.4 at home and i like it |
| 09:31.18 | penguinFunk | i run 1.4 and works perfect |
| 09:31.25 | Strom_M | but hey, don't migrate until you're ready :) |
| 09:31.38 | penguinFunk | we started with 1.4 |
| 09:31.44 | s0ck | im not opposed to it |
| 09:31.52 | penguinFunk | Strom_M: ready for what exactly? |
| 09:31.54 | s0ck | but i have developers working on 1.2 codebase |
| 09:32.09 | s0ck | so i can't just up and change things as and when i feel like it ;/ |
| 09:32.13 | penguinFunk | i see |
| 09:32.48 | VietDuc | yes, the question is whether asterisk version 1.4 is stable enough to bring it into production system? |
| 09:33.09 | s0ck | who knows... lots of positive comments in here |
| 09:33.11 | VietDuc | it's sure that version 1.4 has a lot of wonderful features |
| 09:33.11 | Strom_M | VietDuc: i see no problem in doing that |
| 09:33.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:34.11 | penguinFunk | VietDuc: we use it in production, 1 office 4 * analogue lines (tdm400p) another office 6 * analogue lines (tdm2400p) |
| 09:34.27 | VietDuc | ok... anyway, Digium slowly to release new version for asterisk 1.4 these days and it means that 1.4 is more and more stable now |
| 09:35.19 | penguinFunk | <PROTECTED> |
| 09:36.18 | s0ck | what kernel you running btw |
| 09:36.26 | penguinFunk | 2.6.20 |
| 09:36.37 | penguinFunk | zaptel 1.4.2 |
| 09:37.11 | penguinFunk | in one office... |
| 09:37.49 | penguinFunk | 2.6.18 / zaptel 1.4.0 in the other office with the good uptime |
| 09:37.52 | s0ck | you got those two boxes hooked up to each other? |
| 09:37.56 | s0ck | iax trunk or something |
| 09:38.05 | penguinFunk | nah, we use sip over ipsec |
| 09:38.30 | s0ck | keep hearing a lot about iax, haven't looked at it yet |
| 09:39.17 | *** join/#asterisk badcfe (i=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 09:39.32 | penguinFunk | yeah iax2 would be a better solution do to the multiplexing channels |
| 09:39.33 | VietDuc | <PROTECTED> |
| 09:39.39 | penguinFunk | i did look at it briefly before |
| 09:40.23 | VietDuc | I'm trying ways to get the best quality for calls between internal extensions |
| 09:40.31 | VietDuc | Could someone show me one? |
| 09:40.51 | Strom_M | VietDuc: use ulaw / alaw |
| 09:41.05 | badcfe | i have defined a sip peer with host=a.b.c.d and when he sends me invites i get the message "Found no matching peer or user for 'a.b.c.d:50148'" |
| 09:41.21 | Strom_M | badcfe: did you set username=whatever |
| 09:41.29 | badcfe | why is the sending port number appended to the name so that asterisk dont recongnise it as the peer i have defined? |
| 09:42.10 | badcfe | Strom_M: yes, i have username=something |
| 09:43.13 | badcfe | Strom_M: the session invite is coming from a cisco. its e.g. From: "anonymous" <sip:493434287@a.b.c.d> |
| 09:44.05 | badcfe | Strom_M: maybe ill try with username=anonymous then huh? |
| 09:45.31 | VietDuc | Strom_M: yes, i know that. but if 2 servers not in LAN, ulaw is not very suitable because it takes a lot of bandwidth |
| 09:45.36 | badcfe | Strom_M: nope, still get Found no matching peer or user for '217.13.51.140:59107' |
| 09:46.17 | VietDuc | by the way, did anyone try to sniff SIP package... It's plain and can be heard by any audio player |
| 09:46.36 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:46.50 | Strom_M | VietDuc: only takes a lot of bandwidth if you've got crap bandwidth to begin with |
| 09:47.18 | badcfe | Strom_M: any other suggestions about what to check, any hinst? |
| 09:51.45 | badcfe | Strom_M: the invite looks like this here |
| 09:54.36 | badcfe | Strom_M: Strom_M .. http://pastebin.ca/583345 |
| 09:55.13 | Strom_M | badcfe: I'm not a SIP god |
| 09:55.19 | Strom_M | it's black magic to me |
| 09:55.31 | badcfe | Strom_M: thats the INVITE. so why dont * recongnize it as the peer having host=xx.b.c.d ? |
| 09:55.49 | badcfe | Strom_M: where is the sip god? |
| 09:56.52 | badcfe | i dont manage to get * recognize the INVITES like this here http://pastebin.ca/583345 |
| 09:57.29 | badcfe | on my sip peer configured with host=xx.b.c.d (and yes, i know about "sip reload" after a modification in the config) |
| 09:58.38 | badcfe | anyone has a hint about what i could try or check in my sip config of the peer? |
| 10:00.51 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 10:02.11 | cy303 | yo |
| 10:03.16 | penguinFunk | badcfe: tried host=dynamic? |
| 10:03.37 | penguinFunk | and make sure the user=blah matches what is in the phone |
| 10:04.18 | cy303 | gehh |
| 10:04.21 | cy303 | callwithus down |
| 10:04.30 | badcfe | penguinFunk: so if i see 'From: "anonymous" <sip:493434287@' then i should set user=anonymous ?? |
| 10:05.05 | penguinFunk | the username set in the phone has got to match the user= in sip.conf |
| 10:05.40 | badcfe | penguinFunk: ok. its not a phone but a cisco gw .. |
| 10:05.52 | penguinFunk | ah i see |
| 10:06.07 | penguinFunk | whatever device really |
| 10:06.13 | penguinFunk | should be the same principle |
| 10:06.30 | badcfe | penguinFunk: so whats the difference between username= and user= ? |
| 10:07.51 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 10:08.26 | penguinFunk | i use (user=) not username= |
| 10:08.32 | penguinFunk | it think they might be the same |
| 10:09.31 | penguinFunk | http://forums.digium.com/viewtopic.php?p=49075&sid=3863310475b4fd280b7f0014541931d0 |
| 10:09.39 | penguinFunk | other people having the same confusion |
| 10:09.45 | badcfe | penguinFunk: yes. i just said its not a phone to tell that i dont configure it but just have to align me with this here http://pastebin.ca/583345 |
| 10:10.45 | badcfe | penguinFunk: and i cant set host=dynamic because i have many peers and they have to be distinguished |
| 10:11.52 | penguinFunk | host=dynamic means a device from any ip can connect as that user |
| 10:11.56 | penguinFunk | could just test it |
| 10:12.13 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 10:15.58 | penguinFunk | <badcfe> penguinFunk: so if i see 'From: "anonymous" <sip:493434287@' then i should set user=anonymous ?? |
| 10:15.59 | penguinFunk | yes |
| 10:16.00 | penguinFunk | try it |
| 10:20.10 | badcfe | Found no matching peer or user for '217.13.51.140:53495' |
| 10:20.26 | badcfe | even with host=dynamic and user=anonynous |
| 10:20.31 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.16.107.revip2.asianet.co.th) |
| 10:20.37 | *** part/#asterisk BugKhaM (n=LAMER@ppp-58.8.16.107.revip2.asianet.co.th) |
| 10:20.37 | JT | badcfe: i don't think you understand what host=dynamic does |
| 10:20.45 | badcfe | for INVITE having From: "anonymous" <sip:493434287@217.13.51.140> |
| 10:21.21 | badcfe | JT: okay |
| 10:21.32 | JT | badcfe: umm, how about pastebinning something useful |
| 10:21.34 | badcfe | JT: what does it do? |
| 10:21.38 | JT | like configs |
| 10:21.49 | JT | and a whole call's sip dialogue and verbose console |
| 10:21.53 | JT | not one message |
| 10:21.56 | badcfe | JT: http://pastebin.ca/583345 |
| 10:22.02 | JT | yes i've seen it |
| 10:22.03 | JT | useless |
| 10:22.07 | JT | 1 part of 1 call |
| 10:22.23 | badcfe | JT: i have ten configs that doesnt work. i pastebin? |
| 10:22.38 | penguinFunk | this could take some time... |
| 10:22.58 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 10:22.58 | JT | badcfe: pastebin the current config, and pastebin an ENTIRE call (whole sip debug dialogue as well as asterisk console with verbose set to 10) |
| 10:23.03 | badcfe | given the INVITE, it should be possible to tell what the peer config should look like |
| 10:23.08 | JT | no |
| 10:23.12 | JT | stop assuming |
| 10:23.24 | JT | if you want your problem solved, give us something to work with |
| 10:24.04 | badcfe | JT: i thought i had |
| 10:24.23 | JT | i'm letting you know we can't do much with just that |
| 10:26.58 | Uatec | mwuahah |
| 10:27.00 | Uatec | 000E08BC5F30 |
| 10:27.06 | Uatec | :D |
| 10:27.09 | JT | uhoh |
| 10:27.16 | badcfe | JT: my question was, what should my sip peer config look like given this incoming invite message |
| 10:27.42 | JT | do you want to send and receive calls? |
| 10:27.49 | badcfe | JT: only receive |
| 10:28.13 | JT | the cisco has a static ip? |
| 10:28.13 | badcfe | JT: no authentification, just receive calls |
| 10:28.19 | badcfe | JT: yes |
| 10:28.19 | JT | i see |
| 10:28.28 | JT | set host= to the hostname/ip |
| 10:28.36 | badcfe | JT: done |
| 10:29.06 | badcfe | however, for some reason i think * catenates the sending portnum and thus doesnt recognize on that ip... |
| 10:29.23 | *** join/#asterisk Fulk (n=fulk@87-194-176-39.bethere.co.uk) |
| 10:29.25 | JT | what makes you think that |
| 10:29.34 | JT | and i assume you meant to say concatenates |
| 10:29.53 | badcfe | cause * says Found no matching peer or user for 'aa.bb.cc.dd:53495' (aa.bb.cc.dd is its ip address) |
| 10:30.10 | JT | allowguest=yes |
| 10:30.34 | badcfe | JT: cat yes. then * goes to the default context wich is boring |
| 10:30.36 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:30.50 | badcfe | JT: ill try allowguest=yes |
| 10:32.19 | purserj | win 23 |
| 10:33.12 | badcfe | JT: still Found no matching peer or user for '217.13.51.140:52576' |
| 10:34.23 | badcfe | JT: i guess ill compare line by line this INVITE with INVITEs going to peer configs that work here... |
| 10:34.33 | JT | ok |
| 10:35.01 | badcfe | purserj: win 23 dozens |
| 10:35.10 | purserj | weeee! |
| 10:40.28 | *** join/#asterisk |chodorenko| (n=chodoren@etm005.nl.ded.neolocation.net) |
| 10:40.39 | |chodorenko| | Hello All |
| 10:40.55 | Fulk | hey |
| 10:41.05 | _E-bola | ~blf |
| 10:41.06 | jbot | i guess blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 10:41.55 | *** join/#asterisk dreamind (n=dreamind@p54A7B341.dip0.t-ipconnect.de) |
| 10:42.31 | penguinFunk | nice! |
| 10:42.35 | dreamind | hi folks |
| 10:42.37 | penguinFunk | not heard of that before |
| 10:42.44 | dreamind | can anybody help me with DTMF problems and asterisk? |
| 10:42.54 | penguinFunk | ~dtmf |
| 10:42.55 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
| 10:43.00 | dreamind | I tried using dtmf using WaitExten() and it works fine |
| 10:43.10 | |chodorenko| | I call to myself in office. I lift a tube on secretarial phone, I hear normally. Then I do axfer, the station speaks that switches, thus to the subscriber music is not played, though the station itself writes that has included music, after switching the subscriber hears, that to whom have switched does not hear the subscriber |
| 10:43.14 | dreamind | but only when * call* the asterisk |
| 10:43.20 | |chodorenko| | asterisk 1.4.5 |
| 10:43.26 | dreamind | when the asterisk calls me via a call file, dtmf only works on some phones. |
| 10:43.31 | *** join/#asterisk ivanfm_ (n=ivanfm@c934f322.virtua.com.br) |
| 10:43.31 | dreamind | for some very strange reason. |
| 10:44.00 | |chodorenko| | debug log is http://pastebin.ca/583380, |
| 10:44.07 | penguinFunk | dreamind: turn dtmf debug on and check whether * is receiving correct dtmf tones from the phones |
| 10:44.20 | |chodorenko| | please help |
| 10:44.38 | dreamind | penguinFunk: hm, ok will try that |
| 10:44.57 | dreamind | hm, ok well dtmf debugging doesn't exist with asterisk 1.2 |
| 10:45.07 | penguinFunk | lol |
| 10:45.09 | dreamind | I'm running * 1.2.13 |
| 10:45.15 | penguinFunk | why the hell isn't anyone using asterisk 1.4 |
| 10:45.26 | penguinFunk | am i like the only one or something |
| 10:45.45 | dreamind | well no, but there are several things which are not yet ported to 1.4 |
| 10:45.52 | penguinFunk | http://forums.digium.com/viewtopic.php?p=49075&sid=3863310475b4fd280b7f0014541931d0 |
| 10:45.59 | penguinFunk | i see |
| 10:46.04 | penguinFunk | such as? |
| 10:46.05 | dreamind | and the quite compley dialplan is not yet tested with * 1.4 |
| 10:46.28 | dreamind | for example bristuff is only experimental for 1.4 |
| 10:47.09 | _E-bola | Where can you find information about how BLF works on sip lines especially snom phones? |
| 10:47.16 | _E-bola | Do you need to setup anything in asterisk? |
| 10:48.26 | dreamind | penguinFunk: the really strange point is, when I bridge 2 phones via that asterisk and press any key on them, I hear normal dtmf sounds |
| 10:48.47 | dreamind | penguinFunk: but when I try to use WaitExten() on outbound calls, it fails. |
| 10:48.49 | Uatec | Grrr |
| 10:49.00 | Uatec | why does the SPA-1001 have it's model number written on it, but not the SPA922? |
| 10:49.01 | Uatec | grr |
| 10:49.05 | Uatec | in barcode i mean |
| 10:50.22 | |chodorenko| | lf in office. I lift a tube on secretarial phone, I hear normally. Then I do axfer, the station speaks that switches, thus to the subscriber music is not played, though the station itself writes that has included music, after switching the subscriber hears, that to whom have switched does not hear the subscriber |
| 10:50.22 | |chodorenko| | asterisk 1.4.5 |
| 10:50.22 | |chodorenko| | debug log is http://pastebin.ca/583380,Please help me |
| 10:52.09 | *** join/#asterisk L8n (n=lejudd@96.83.233.220.exetel.com.au) |
| 10:52.32 | badcfe | penguinFunk: i see that the cisco does not have ";user=phone" -- does that make a difference to * ? the ones that work has ";user=phone" in their invites... |
| 10:53.52 | penguinFunk | badcfe: read the book |
| 10:53.55 | penguinFunk | ~thebook |
| 10:53.55 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:55.04 | badcfe | penguinFunk: irc and book are two different medias. i dont have time to read the book right now. |
| 10:55.20 | badcfe | penguinFunk: but ill do a grep user=phone on it |
| 10:55.29 | L8n | badcfe: a ";" before a line in the config means that line is commented out. |
| 10:56.10 | badcfe | L8n: the "user=phone" i mentioned is in the sip message, not the config. |
| 10:57.08 | L8n | badcfe: Ah, ok... |
| 10:58.27 | badcfe | L8n, penguinFunk: thanks for help. i have downloaded the book now. |
| 11:13.47 | *** join/#asterisk Corydon76-home (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 11:13.47 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:16.01 | awk | excuse me? |
| 11:17.41 | e-ddie | not really |
| 11:19.58 | ai-a | We have just installed Asterisk and installed IdeFisk to test it.. created an IAX account, registers fine.. when i dial a conference.. or any number the asterisk console says "Rejected connect account from %ip%, request '335@' does not exist.. 335 is the conference ext. |
| 11:20.15 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 11:21.34 | DrukenLPY | sounds like a context issue to me.... |
| 11:21.56 | L8n | does idefisk register? |
| 11:22.27 | DrukenLPY | he says it does |
| 11:24.53 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 11:25.12 | ai-a | yes. |
| 11:25.15 | ai-a | sorry, was playing with it. |
| 11:25.29 | ai-a | ive had this on both asterisk and asterisknow systems. |
| 11:25.51 | L8n | I think, DrukenLPY may be right there... |
| 11:26.15 | ai-a | whats a context issue ? |
| 11:26.23 | L8n | The console is fairly helpful in debugging problems... |
| 11:26.56 | L8n | Sounds like you might have a bit of reading to do? :P |
| 11:27.19 | ai-a | but asterisknow should work out of the box ? |
| 11:28.17 | penguinFunk | yes but it still needs correct configuration |
| 11:28.26 | L8n | bump... haven't really played with asterisknow... I usually edit the dialplan by hand... |
| 11:34.02 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:34.06 | *** join/#asterisk lwh (n=lwh192@66.212.165.24.tor.pathcom.com) |
| 11:34.10 | Dandre | hello, |
| 11:36.43 | Dandre | I am trying to get all agents from manager apr. If I send an Agents command to the manager I get "Success" "Agents will follow". How could I get those Agents? |
| 11:37.06 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 11:41.39 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:43.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:44.37 | *** join/#asterisk achu (n=achu@122.167.63.163) |
| 11:45.21 | achu | can anybody tell me how to configure jivetel extension on Asterisk ? |
| 11:46.20 | *** join/#asterisk ru_wing|wrk (n=wing@newoffice-5.tvcom.ru) |
| 11:46.52 | shido6 | jivetel ? |
| 11:47.11 | achu | yes www.jivetel.com |
| 11:47.56 | Dandre | I am trying to get all agents from manager apr. If I send an Agents command to the manager I get "Success" "Agents will follow". How could I get those Agents? |
| 11:49.03 | L8n | Is this a bug...? When in the remote console, if I issue "sip show registry" the first character of the last entry is removed... |
| 11:52.17 | L8n | This seems to happen when I run "asterisk -rc" in a remote terminal, however not if asterisk is run in a terminal screen... |
| 11:53.34 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
| 11:55.38 | *** join/#asterisk jwh (i=jwh@knickertron.com) |
| 11:56.09 | tzafrir_laptop | L8n, 'asterisk -rc' ? why the -c? |
| 11:59.28 | penguinFunk | yeh just use 'asterisk -rvvvvvvvv' |
| 12:05.38 | shido6 | ZZzZ |
| 12:05.49 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
| 12:09.16 | *** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk) |
| 12:12.09 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:12.14 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 12:12.16 | zdrulio | hello all |
| 12:12.30 | zdrulio | for what is agent ? |
| 12:13.15 | *** join/#asterisk Corydon76-home (i=four@pdpc/supporter/sustaining/Corydon76-home) |
| 12:13.15 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 12:14.13 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 12:14.49 | *** join/#asterisk DirtyD (n=DigiD@ool-18bddad8.dyn.optonline.net) |
| 12:15.16 | DirtyD | Why is my volume to low when doing native bridging on TDM2400p |
| 12:15.29 | DirtyD | I played with rx and tx.. |
| 12:15.37 | penguinFunk | turn your tx/rx gains up |
| 12:15.40 | penguinFunk | oh |
| 12:15.47 | DirtyD | I had to set rx to -2 to get the thing to dial. |
| 12:16.08 | DirtyD | I cranked TX all the way up, no effect. |
| 12:16.28 | penguinFunk | download fxotune |
| 12:16.44 | DirtyD | really? |
| 12:16.44 | penguinFunk | it can help with this sort of thing |
| 12:16.44 | penguinFunk | yeh works for us |
| 12:16.46 | DirtyD | nice I'll try. I didnt even know it existed. |
| 12:17.18 | penguinFunk | are you having echo issues? |
| 12:17.53 | DirtyD | no echos |
| 12:17.54 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:18.03 | penguinFunk | hardware echo canellation? |
| 12:18.22 | DirtyD | I pulled the hardware cancellation module off the board, same deal. |
| 12:19.03 | DirtyD | it's just very low volume when I'm bridging zap to zap |
| 12:19.15 | DirtyD | sip to zap sounds great |
| 12:19.21 | penguinFunk | ahh i see |
| 12:19.30 | penguinFunk | we only ever bridge zap to sip and vica versa |
| 12:19.47 | penguinFunk | tx/rx gain makes a difference then |
| 12:19.54 | DirtyD | where do I find fxotune? |
| 12:20.23 | penguinFunk | there is one in the zaptel source |
| 12:20.32 | penguinFunk | but it needs to be patched |
| 12:20.50 | DirtyD | got it. |
| 12:21.03 | DirtyD | why is it broke? |
| 12:21.17 | DirtyD | I was playing with ztmonitor yesterday and it locked up my box. |
| 12:22.04 | penguinFunk | that doesn't sound right |
| 12:22.19 | penguinFunk | http://svn.digium.com/view/zaptel/trunk/fxotune.c?view=markup&pathrev=1348 |
| 12:22.39 | penguinFunk | if echo's aren't the issue then fxotune might not help you _that_ much |
| 12:22.44 | *** join/#asterisk YonahW (n=kvirc@IGLD-83-130-71-223.inter.net.il) |
| 12:22.52 | zdrulio | tell me plz. for what is Agent ? |
| 12:23.10 | zdrulio | what is Agent. i don`t understand meaning and for what is use in asterisk ? |
| 12:23.27 | penguinFunk | zdrulio: They are sentient intelligent agents within the computer-generated world of the Matrix, guarding it from anyone or anything that could reveal it as a false reality. |
| 12:23.53 | [TK]D-Fender | zdrulio: Go look up "queues" and "agents" on the WIKI |
| 12:23.55 | jwh | lol |
| 12:24.17 | [TK]D-Fender | Mr. Anderson, how good to see you again... |
| 12:24.22 | penguinFunk | hehehe |
| 12:24.31 | penguinFunk | my name... is NEO! |
| 12:25.40 | penguinFunk | DirtyD: it would help you a lot if you could get ztmonitor working to measure the rx/tx volumes |
| 12:28.15 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 12:29.20 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:29.29 | Uatec | hey, has anybody ever provisioned a firmware upgrade to any sipura/linksys phones? |
| 12:30.22 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 12:33.29 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:36.00 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:36.11 | Zeeek | hey now |
| 12:36.24 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 12:36.28 | penguinFunk | hey now brown cow |
| 12:36.32 | mattfletcher | does asterisk have any particular modules/functions/whatever to integrate with x10 hardware? |
| 12:37.25 | *** join/#asterisk SuD (n=Ask@89.140.32.2.static.user.ono.com) |
| 12:37.49 | Zeeek | penguinFunk sorry, that's how now brown cow |
| 12:38.00 | SuD | phone hangup detection is driving me crazy (spain) |
| 12:39.38 | [TK]D-Fender | Zeeek: MOO. |
| 12:39.50 | Zeeek | hey now TKD |
| 12:40.18 | *** join/#asterisk penguinFunk_ (n=penguinF@unaffiliated/penguinfunk) |
| 12:40.35 | hackeron | can someone recommend a softphone that works on OSX to use with Asterisk? -- I tried JackenIAX and LoudHush, but both have terrible echo using built in speakers and mic (unlike something like skype) |
| 12:40.51 | Zeeek | oh by the way folks I'm running live video feed right now http://www.blogtv.com/Shows/787 |
| 12:41.19 | Zeeek | hackeron X-Lite and Gizmoproject both work great on OSX |
| 12:41.47 | Zeeek | but nothing will do with speakersz and mic - need a headset |
| 12:44.53 | penguinFunk | . |
| 12:44.56 | Zeeek | .. |
| 12:44.58 | ccesario | hi |
| 12:45.33 | Zeeek | _XXX5551212 |
| 12:45.57 | ccesario | I'm testing app_rxfax, and when I send fax the app_rxfax generate a blank tif file.... somebody have idea about this ? |
| 12:46.29 | Zeeek | ccesario make sure you try this with several different fax machines. Many won't work, some will |
| 12:47.42 | hackeron | Zeeek: Hmm, is there nothing I can change on the asterisk or client side to use built in speakers and mic? -- What echo cancellation is Skype using, it even works when I plug in my 5.1 surround system to this macbook! |
| 12:47.43 | ccesario | Zeeek, hmmmmmmmmmm can maybe this... I'm testing with hp officejet 4300 |
| 12:48.05 | Zeeek | hackeron try the clients I mentioned then come back and ask |
| 12:48.16 | Zeeek | they both have boxes to check for echo cancel |
| 12:48.48 | Zeeek | ccesario I've "used" rx_fax for three years. It actually gets about 3 faxes out of 5 |
| 12:49.24 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:49.28 | hackeron | Zeeek: gizmo is definitely unusable -- trying X-Lite now |
| 12:49.35 | ccesario | Zeeek, hmmmmmm |
| 12:49.49 | [TK]D-Fender | Zeeek: What are the latest versions of * you have gotten it working under? |
| 12:49.53 | Zeeek | gizlmo works great for me |
| 12:49.55 | s0ck | update: it compiled |
| 12:49.55 | ccesario | Zeeek, U try other solution ? exists other solution to solve this ? |
| 12:50.05 | [TK]D-Fender | Zeeek: (SpanDSP that is) |
| 12:50.17 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:50.18 | Zeeek | [TK]D-Fender if you call 3/5 working, I'd hate to purchase software from you :) |
| 12:50.19 | s0ck | anyone got a snom360? |
| 12:50.24 | s0ck | just heard it ring for the first time... wtf lol |
| 12:50.42 | [TK]D-Fender | Zeeek: I consider that a considerable upgrade from automatic segfult ;) |
| 12:50.49 | Zeeek | [TK]D-Fender last installed on 1.2.16 - works occasionally, always spam faxes |
| 12:51.10 | Zeeek | Oh it doesn't break anything, it just doesn't get faxes. |
| 12:51.26 | [TK]D-Fender | Zeeek: You could in turn realize that perhaps ALL people send you is junk, and are lucky enough to get SOME! |
| 12:51.34 | Zeeek | heh |
| 12:51.45 | Zeeek | I tell people to join the 21st century, screw fax |
| 12:52.05 | Zeeek | it's so trivial to forge a fax I see no legal value to them either |
| 12:52.33 | Zeeek | I always thought it was because it was free, there's nothing youi can do |
| 12:52.50 | Zeeek | I have a $29 windoze shareware that has never ever missed a fax |
| 12:52.53 | coppice | Zeeek: very forgable, but they do have legal value. before FAX Telex had an even better legal standing, and those were trivial to forge |
| 12:53.14 | Zeeek | coppice I know and that is what's so stupid about the whole fax thing |
| 12:53.18 | coppice | Zeeek: thousands of people have no problems at all with rxfax |
| 12:53.31 | Zeeek | coppice no offense, it's a wonderful effort |
| 12:53.40 | Zeeek | but it doesn'twork for me and a lot of others |
| 12:53.49 | Zeeek | I hate fax anyway so no matter :) |
| 12:54.29 | coppice | because your setup is broken. have you tried iaxmodem? the timing in * seems more relaxed passing audio through, and few people seem to have trouble with that |
| 12:54.30 | ccesario | O_o |
| 12:55.04 | coppice | there are people take 2 or 3 E1s full of fax all day with rxfax |
| 12:56.02 | Zeeek | I'm using ZAP which is the base technology for faxes in my world |
| 12:57.13 | coppice | someone has recently been testing 500 concurrent FAXes with a new 4 core Intel box |
| 12:57.46 | ccesario | coppice, using rxfax ? |
| 12:57.54 | coppice | yes |
| 12:58.37 | ccesario | O_o |
| 12:59.16 | ccesario | coppice, do you from ? |
| 12:59.58 | ccesario | the pstn operator can cause problem in rxfax ? |
| 13:01.25 | Zeeek | ccesario I would say that depends on whether you receive faxes on anything else, hardware or software on the same line |
| 13:01.26 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:01.33 | coppice | its usually either a misconfigured E1/T1, or crazy gain settings, or data loss within * that screws it up |
| 13:01.51 | Zeeek | coppice is it very CPU or RAM intensive? |
| 13:01.56 | ccesario | hmmmmm |
| 13:02.18 | coppice | you can do quite a few concurrent channels on most boxes |
| 13:03.30 | Zeeek | It's a timing issue I think. I can send a fax on a laptop and it's received perfectly. OTH, most of my customers' fax machines don't work with asterisk/spandsp. They do work with whatever else I hang on the same phone line though |
| 13:03.38 | waKKu | folks.. have some brazilians here ? i'm thinking about wakeup asterisk-br... |
| 13:04.09 | Zeeek | In the end, the only faxes I need to receive don't come through at all |
| 13:04.19 | Zeeek | so we ledture them to send PDF files :) |
| 13:04.27 | Zeeek | s/ledture/lecture |
| 13:05.31 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:05.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:07.10 | Zeeek | . |
| 13:07.12 | hackeron | Zeeek: no echo, yay to X-Lite! |
| 13:07.26 | Zeeek | great, glas it worked. It's my favorite SIP client |
| 13:07.33 | Zeeek | s/glas/glad/ |
| 13:07.50 | Zeeek | dans la pluie? |
| 13:07.55 | tzafrir_laptop | coppice, have you managed to build rxfax with asterisk 1.4? I hve some headers conflicts between spansdsp and asterisk |
| 13:08.18 | tzafrir_laptop | yeah, I know what's the reason for those originally, but still |
| 13:08.45 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.14 | coppice | I don't play with asterisk any more, so I haven't sorted out those issues. there shouldn't be much to change, though. various people seem to have done it for themselves |
| 13:09.38 | ru_wing|wrk | can anyone give me example AGI script on python, working with mysql? |
| 13:10.03 | Zeeek | coppice so no future for spandsp with asterisk? |
| 13:10.31 | coppice | spandsp is moving ahead very nicely |
| 13:10.56 | Zeeek | do you talk about it somewhere we can read? |
| 13:11.15 | coppice | it does have a web site |
| 13:12.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:12.21 | Zeeek | really? |
| 13:14.06 | *** join/#asterisk allen__s (n=chatzill@72.242.225.99) |
| 13:14.06 | tzafrir_laptop | soft-switch.org ? |
| 13:14.20 | coppice | yes |
| 13:14.35 | Zeeek | well, there was an update in April... |
| 13:15.15 | coppice | if you look at the downloads there was an update this week |
| 13:15.56 | Zeeek | [15:10] <Zeeek> do you talk about it somewhere we can read? |
| 13:16.45 | Zeeek | why don't you come and tell the world about what's new on the Asterisk Users COnf today |
| 13:16.53 | Zeeek | prolly too late in the day for you though |
| 13:17.20 | Zeeek | Maybe I could call you for a recorded interview? |
| 13:17.43 | coppice | if people want it, they can find it. i really don't care |
| 13:17.57 | tzafrir_laptop | that user conf, seems to require some windows program or a call to the US |
| 13:18.11 | tzafrir_laptop | is there any way to connect to it with voip? |
| 13:18.13 | Zeeek | tzafrir any SIP client can call it |
| 13:18.20 | tzafrir_laptop | to where? |
| 13:18.22 | Zeeek | http://x2z.eu |
| 13:18.55 | Zeeek | http://x2z.eu/asterisktalkshoecallinsetup.htm |
| 13:20.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 13:20.18 | Zeeek | tzafrir_laptop were you at Astricon Paris last year? |
| 13:20.26 | tzafrir_laptop | hmm... no |
| 13:20.31 | Zeeek | ok |
| 13:20.43 | Zeeek | I thought you worked for a company that was there |
| 13:30.23 | *** join/#asterisk _DAW_ (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:35.03 | *** join/#asterisk badcfe (i=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 13:35.23 | badcfe | somebody know how to configure a sip peer to receive calls from an Cisco-SIPGateway/IOS-12.x |
| 13:36.36 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 13:39.36 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 13:39.44 | badcfe | ? |
| 13:39.56 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:39.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:40.36 | badcfe | anybody could point me to info about how configure a sip peer in order to receive calls into * coming from a Cisco-SIPGateway/IOS-12.x ? |
| 13:41.22 | *** join/#asterisk af_ (n=getsmart@81-174-44-78.dynamic.ngi.it) |
| 13:45.29 | DarKnesS_WolF | <PROTECTED> |
| 13:45.57 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-175-183.buff.east.verizon.net) |
| 13:46.16 | SuPrSluG | ~rhino |
| 13:46.34 | [TK]D-Fender | ~hippo |
| 13:46.35 | jbot | it has been said that hippo is hegemon |
| 13:46.46 | badcfe | DarKnesS_WolF: that theres no reply to your msg |
| 13:46.54 | badcfe | DarKnesS_WolF: maybe a nat or fw problem |
| 13:46.57 | DarKnesS_WolF | badcfe: why ? how to fix it ? |
| 13:47.27 | SuPrSluG | looking for the bots say on rhino cards. |
| 13:47.38 | badcfe | DarKnesS_WolF: how to fix? well, this is actually a question about your overall network config, not asterisk |
| 13:47.55 | badcfe | DarKnesS_WolF: maybe you have a NAT router between * and the phone? |
| 13:48.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:48.37 | badcfe | DarKnesS_WolF: and make shure that port 5060 is not going to the toilet somewhere |
| 13:49.03 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:49.36 | badcfe | DarKnesS_WolF: and configure * to nat=yes or nat=no accordingly. setup the NAT IP Address on your phone |
| 13:49.39 | DarKnesS_WolF | badcfe: this phone can call anotehr sip phones " sipphone ---> nat ----< * <----sipphone " works |
| 13:49.51 | DarKnesS_WolF | this happen when this phone trying to call sip provider |
| 13:49.59 | DarKnesS_WolF | at hte same network a softphone works :-s |
| 13:50.54 | file | [TK]D-Fender: how goes? |
| 13:52.08 | [TK]D-Fender | file: Meh. Its friday, got Monday off, mounted my center channel to my surround setup last night (ceiling), so now I jsut need to wire up the rear surrounds (easy), and GET MY FRIGGEN MOUNT (Stupid promo shipping error). |
| 13:52.26 | [TK]D-Fender | file: AndrewVision 2.0 has been awaiting completion for almost a month now. |
| 13:52.50 | *** part/#asterisk |chodorenko| (n=chodoren@etm005.nl.ded.neolocation.net) |
| 13:53.45 | *** join/#asterisk syamajala (n=syamajal@c-75-68-225-205.hsd1.ma.comcast.net) |
| 13:53.57 | file | [TK]D-Fender: get to it! |
| 13:54.08 | syamajala | anyone use trixbox? |
| 13:54.12 | syamajala | their channel is dead |
| 13:54.41 | DarKnesS_WolF | file: anyidea how can i use teh option m in authenticate cmd ? what is teh password file synatx in that case ? |
| 13:54.42 | s0ck | #freepbx is usually a bit busier |
| 13:54.53 | coppice | i expect bringing it back to life is one of their trix |
| 13:55.09 | file | DarKnesS_WolF: it tells you. |
| 13:55.23 | file | <PROTECTED> |
| 13:55.23 | file | <PROTECTED> |
| 13:55.23 | file | <PROTECTED> |
| 13:55.24 | file | <PROTECTED> |
| 13:56.37 | DarKnesS_WolF | yes exactly i didn't understand :-s |
| 13:56.55 | DarKnesS_WolF | i did like this exten => XXX,1,Authenticate(/etc/asterisk/password|am) |
| 13:57.11 | DarKnesS_WolF | and password inside it 108:1234 |
| 13:57.14 | DarKnesS_WolF | 110:4321 |
| 14:11.51 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:14.58 | *** join/#asterisk achu (n=achu@122.167.63.163) |
| 14:16.18 | achu | when I make outgoing calls it shows the error : "Got SIP response 603 "Declined" back from x.x.x.x" |
| 14:16.29 | achu | I am getting incoming calls |
| 14:16.45 | achu | can anybody tell me why this happens ? |
| 14:17.07 | [TK]D-Fender | achu: pastebin the full CLI output of a failed call attempt at verbose 10 w/ SIP debug enabled. |
| 14:17.09 | [TK]D-Fender | ~pb |
| 14:17.10 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 14:17.11 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 14:17.47 | badcfe | i see that my /var/log/asterisk/cdr.db is growing. i dont want that. how do i disable? |
| 14:18.15 | achu | k |
| 14:19.02 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:19.22 | badcfe | apparently its an sqlite db file. i dont see where in the asterisk conf i can stop this db logging of cdr |
| 14:20.02 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 14:22.29 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:23.55 | *** join/#asterisk putnopvu1 (i=putnopvu@nat/digium/x-6b3bc4357a088bd7) |
| 14:24.45 | vader-- | did an uptime on my asterisk PBX today |
| 14:24.51 | vader-- | 221 Days running strong |
| 14:24.56 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 14:25.09 | Qwell[] | on asterisk, or the server? |
| 14:25.14 | vader-- | both |
| 14:25.22 | *** join/#asterisk cpehouston (n=administ@63.247.107.130) |
| 14:25.24 | cpehouston | ahoy |
| 14:25.32 | Qwell[] | You should consider upgrading if it's publicly accessible |
| 14:25.36 | cpehouston | I cannot check my voicemail, it just goes silent |
| 14:25.36 | vader-- | it's not |
| 14:25.38 | cpehouston | any ideas? |
| 14:25.40 | vader-- | internal use only |
| 14:25.42 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 14:25.46 | cpehouston | if i put in the wrong password it says incorrect |
| 14:25.50 | cpehouston | but the right password just sits there |
| 14:27.56 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:28.19 | kippi | is there some type of HA within asterisk? |
| 14:28.53 | kombi_ | by HA you mean high availability? |
| 14:28.54 | *** join/#asterisk dijungal (n=chatzill@209.59.110.36) |
| 14:29.02 | [TK]D-Fender | kippi: No. |
| 14:29.13 | dijungal | me again... got the cisco router to connect to the asteriskbox |
| 14:29.23 | dijungal | the problem was firewalls on the cisco side.. lol |
| 14:29.31 | [TK]D-Fender | kippi: It takes untold amounts of duct-tape to holdup * under strain. |
| 14:29.43 | dijungal | can anyone tell me what's causing this and how to fix it: "Internal RTCP NTP clock skew detected:" |
| 14:29.46 | [TK]D-Fender | dijungal: Never would have guessed... |
| 14:30.11 | dijungal | TKD-Fender: the cisco sends the call, the asterisk picks it up, but no audio |
| 14:30.22 | UCFmethod | quick question... I know 1.4 has been out awhile now... I was wondering if there have been updated wiki's and updated books on all the differences/deprecated stuff... |
| 14:31.02 | UCFmethod | It doesn't appear that voip-info.org has been updated by anyone regarding a few of the small issues I am running into..... any pointers would be awesome |
| 14:31.33 | kippi | [TK]D-Fender: is this coming to asterisk |
| 14:32.09 | _VoiceMeUp_COM | would stop when convenient stopp accepting new calls ? |
| 14:32.51 | [TK]D-Fender | kippi: very far away if ever. Remember * is a telephony tool-kit and as such isn't made up of seperably operable modules, so it in itself cannot survive failure. the SIP stack is broken, etc, which is when everyone needing HA runs SER in front, etc. |
| 14:33.05 | [TK]D-Fender | _VoiceMeUp_COM: When no calls are in progress |
| 14:33.20 | _VoiceMeUp_COM | yeah |
| 14:33.22 | [TK]D-Fender | _VoiceMeUp_COM: Actually is does NOT refuse new calls |
| 14:33.26 | _VoiceMeUp_COM | hmm |
| 14:33.29 | _VoiceMeUp_COM | ok |
| 14:33.31 | _VoiceMeUp_COM | tk |
| 14:33.40 | _VoiceMeUp_COM | call 418 948 3125 |
| 14:33.45 | _VoiceMeUp_COM | youll see what i hear |
| 14:33.48 | _VoiceMeUp_COM | its only on span2 |
| 14:33.51 | _VoiceMeUp_COM | span1 g2 |
| 14:33.53 | _VoiceMeUp_COM | i mean |
| 14:34.01 | _VoiceMeUp_COM | g1 is fine and g1 is 514 |
| 14:34.06 | cpehouston | so where should I start looking when voicemail stops working |
| 14:34.07 | _VoiceMeUp_COM | so i guess gtis nuked |
| 14:34.08 | [TK]D-Fender | ICK! |
| 14:35.09 | dijungal | what is this and how do i fix it? "Internal RTCP NTP clock skew detected:" |
| 14:35.34 | _VoiceMeUp_COM | 0 tx 0 rx in zapat |
| 14:36.33 | _VoiceMeUp_COM | group = 1 |
| 14:36.33 | _VoiceMeUp_COM | channel => 1-21 |
| 14:36.33 | _VoiceMeUp_COM | group = 2 |
| 14:36.33 | _VoiceMeUp_COM | channel => 22-23 |
| 14:36.40 | _E-bola | If a client wanted a userfirnely GUI to control a few things, would u guys recomend trixbox or asteriskNOW? |
| 14:36.45 | _VoiceMeUp_COM | sorry for paste |
| 14:37.26 | *** join/#asterisk negativeduck (n=ducky@donald.darkwing.net) |
| 14:37.40 | *** join/#asterisk breanna_ (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 14:37.43 | _VoiceMeUp_COM | but it started this morning |
| 14:37.44 | _VoiceMeUp_COM | ;) |
| 14:38.08 | _VoiceMeUp_COM | i hear it too sometiem |
| 14:39.13 | breanna_ | voices? |
| 14:39.29 | _VoiceMeUp_COM | ehheeh |
| 14:39.35 | _VoiceMeUp_COM | no a group thats nuked |
| 14:39.39 | [TK]D-Fender | _E-bola: ... |
| 14:39.43 | [TK]D-Fender | ~toywr |
| 14:39.43 | _VoiceMeUp_COM | like g2 of span 1 |
| 14:40.05 | _E-bola | [TK]D-Fender: mmm bot should say something? |
| 14:40.12 | _VoiceMeUp_COM | breanna_ you going to hunt bears ? |
| 14:40.13 | _VoiceMeUp_COM | ;) |
| 14:40.28 | [TK]D-Fender | _E-bola: The One You Write Yourself |
| 14:40.32 | breanna_ | You have the bears, not I |
| 14:40.53 | _E-bola | Why on earth would i write one myself? When there are several public options? |
| 14:41.22 | [TK]D-Fender | _E-bola: Because all of the free ones are flaming piles of shit </bile> |
| 14:41.30 | _E-bola | haha |
| 14:41.38 | Nugget | [TK]D-Fender speaks the truth. |
| 14:41.39 | _VoiceMeUp_COM | hehe |
| 14:41.42 | breanna_ | Is flaming good or bad? |
| 14:41.44 | _E-bola | i admit i found trixbox hardcore confusing |
| 14:41.54 | [TK]D-Fender | breanna_: Helps the smell travel <- |
| 14:42.29 | exoxe | hello |
| 14:42.40 | [TK]D-Fender | And Trixbox isn't a GUI... its more like a manure box so you can keep all your shit together! |
| 14:42.50 | Zeeek | jeeeze |
| 14:43.00 | kombi_ | does the type of codec have any implication on latency? |
| 14:43.35 | _E-bola | [TK]D-Fender: How about connecting windows mobile devices to asterisk then, via wifi? |
| 14:43.58 | [TK]D-Fender | kombi_: in itself I don't believe so. Depends on packetization rate. |
| 14:44.28 | kombi_ | ok thanks! it all comes down to ping rates really.. |
| 14:44.31 | [TK]D-Fender | _E-bola: Been done. Works fine ins as much as... |
| 14:44.32 | [TK]D-Fender | ~wifisip |
| 14:44.33 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 14:44.37 | *** join/#asterisk ccesario (n=ccesario@201-0-124-46.dsl.telesp.net.br) |
| 14:44.45 | Zeeek | HARD i good |
| 14:44.57 | kombi_ | and that's what the bot says.. |
| 14:45.27 | cpehouston | this is pretty odd |
| 14:47.21 | _VoiceMeUp_COM | one module id think would be nice.. GPS_monitor or soemthing.. you poll a gps.. hand have coords grid you need to stay in.. i fgps reports out of grid it plays a "your chidlren has stepped out of range " message to the mom's cell |
| 14:47.27 | exoxe | is there a way to handle a non-bridge when executing a macro within Dial()? I have a macro that allows the caller to accept or reject the call based on a voice prompt, and that works, but if the called party just hangs up instead of choosing an option, it ends the call instead of sending them to voice mail (as I have configured in the macro)... the CLI shows the following data: User disconnected (next line) == Spawn extension (macro-answercall |
| 14:48.33 | [TK]D-Fender | exoxe: "g" <- |
| 14:48.51 | exoxe | okay let me look at that option |
| 14:48.53 | exoxe | thank you |
| 14:48.56 | Juggie | you cant transfer to voicemail within the macro |
| 14:49.25 | cpehouston | are there any other asterisk support channels around |
| 14:49.30 | Juggie | not reliably anyways, you should run your macro on the dial, and then based on the return from the dial, go to vm, etc. |
| 14:49.41 | [TK]D-Fender | Juggie: No, He's doing that right after dial, except dial terminates because he didn't put "g" |
| 14:50.03 | Juggie | oh, i misread what he said, i thought he was trying to push the user to vm within the macro. |
| 14:50.11 | [TK]D-Fender | Juggie: I'd be betting that it counts "like" answered, and needs to check that var that gets set on the reject. |
| 14:50.38 | Juggie | [TK]D-Fender, i've done this before (a macro which allows agent to accept/reject call) but i dont remmeber what app_dial does |
| 14:50.51 | Juggie | not off the top of my head anyways, regardless it was managable to get out the information i needed. |
| 14:51.24 | UCFmethod | anyone know a CLI tool to turn mp3 into ulaw for MOH |
| 14:51.52 | exoxe | [TK]D-Fender: works! |
| 14:51.55 | exoxe | I owe you one |
| 14:52.10 | exoxe | I forget that Dial() has like 300 options, heh |
| 14:52.44 | [TK]D-Fender | UCFmethod: You can use MPG123 to go to wav, and SOX to convert to ULAW easy |
| 14:53.07 | Zeeek | what is needed is not so much another book or document about asterisk, but a radical new approach. I nominate TAGS |
| 14:53.29 | [TK]D-Fender | Zeeek: Approach to what? |
| 14:53.35 | DirtyD | Digium has such wonderful tech support. I've been on hold for 15 minutes.. |
| 14:53.38 | Zeeek | Tough Assed Girl Scouts |
| 14:53.48 | UCFmethod | well going from mp3 to wav I already have memorized using lame, but from wav to ulaw I am clueless.. I will search google for the sox CLI options |
| 14:53.55 | Zeeek | DirtyD they're so good no one wants to let them hangup |
| 14:53.58 | cpehouston | safe_asterisk startup succeeded |
| 14:53.58 | UCFmethod | [TK]D-Fender: Thanks for the heads up |
| 14:54.12 | [TK]D-Fender | Zeeek: Do you know what ritual a Cub goes through to become a Boy Scout? |
| 14:54.28 | Zeeek | no |
| 14:54.29 | [TK]D-Fender | Zeeek: He has to eat a Brownie ;) |
| 14:54.35 | Zeeek | but I know I have no beer |
| 14:54.50 | Zeeek | that is both titilating and sick |
| 14:54.59 | exoxe | heh, you said tit |
| 14:55.15 | waKKu | good afternoon folks.. i'm looking for somebody from Digium, i want talk about an irc-channel here on freenode and need some permissons from them.... someone here ? : ) |
| 14:55.19 | exoxe | you know what I like about high school girls? |
| 14:55.24 | DirtyD | a screw calling tech support, I'm want a brownie. |
| 14:55.32 | Zeeek | waKKu just start the channel |
| 14:55.44 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 14:56.04 | cpehouston | i can't find anything about this on the forum, but basically when I check any voicemail on the network with the correct password asterisk doesnt respond, it just goes silent |
| 14:56.16 | waKKu | Zeeek #asterisk-br is already registered.. i need somebody from digium to do a redirect or unregister them |
| 14:56.22 | cpehouston | i've got about ten users who need to check their vmail this morning |
| 14:56.35 | [TK]D-Fender | waKKu: Does it have people in there? |
| 14:56.42 | waKKu | [TK]D-Fender no |
| 14:56.52 | Zeeek | digium doesn't control freenode AFAIK |
| 14:57.01 | [TK]D-Fender | waKKu: Ask FreeNODE admins |
| 14:57.09 | waKKu | i already talk with staff.. this is right procedure. |
| 14:57.23 | Zeeek | who is asterisk-br registered to? |
| 14:57.36 | waKKu | /chanserv info #asterisk-br |
| 14:57.36 | DirtyD | damn this zit hurts man. |
| 14:58.06 | exoxe | pop it! |
| 14:58.27 | DirtyD | I can't yet |
| 14:58.32 | Zeeek | last seen 1 year and 1 day ago. Give him some time |
| 14:58.33 | exoxe | ahhh... one of those :( |
| 14:58.33 | DirtyD | it's not to a head. |
| 14:58.41 | exoxe | I wouldn't mind some head |
| 14:58.44 | exoxe | ... |
| 14:58.47 | Zeeek | girls scouts |
| 14:58.59 | DirtyD | girls scouts and head? |
| 14:59.08 | Juggie | UCFmethod, try, 'file convert' |
| 14:59.12 | exoxe | that's gettin a little touchy.. |
| 14:59.17 | exoxe | 18yo girlscout? |
| 14:59.23 | DirtyD | Girl scout moms |
| 14:59.31 | Juggie | this is all OFF TOPIC! |
| 14:59.39 | ccesario | any solution to empty fax file (rxfax) :P ? |
| 14:59.41 | exoxe | er.. uh.. okay |
| 14:59.56 | Juggie | <- #asterisk <NOT> #girlscouts |
| 14:59.58 | [TK]D-Fender | Juggie: Actually if its all we're talking about.. it IS the topic ;) |
| 15:00.07 | Zeeek | indeed |
| 15:00.20 | exoxe | so, is it possible to addon to my macro that accepts/rejects a caller and ALSO monitor their call if I send them to voice mail? |
| 15:00.30 | exoxe | our Televantage system does it, so I'd like to mimic it |
| 15:00.40 | Juggie | monitor their voicemail call |
| 15:00.41 | Juggie | ? |
| 15:00.41 | exoxe | basically, you press 3 and it sends them to voice mail but you can also listen to what they're saying |
| 15:00.45 | exoxe | yeah |
| 15:00.51 | Juggie | but you have the recordings |
| 15:01.05 | exoxe | well yeah, but you can listen in before the file is created |
| 15:01.08 | exoxe | you know, evil stuff |
| 15:01.09 | [TK]D-Fender | exoxe: in your macro I suspect you could do an AMI originate / call-file that will ring your back to call ChanSpy. |
| 15:01.28 | Juggie | [TK]D-Fender, would doing just a Monitor() work? |
| 15:01.45 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 15:01.46 | [TK]D-Fender | Juggie: I'm quite certain he wants to listen LIVE. |
| 15:01.56 | [TK]D-Fender | Juggie: like a dumb answering machine. |
| 15:01.57 | exoxe | It's not a big deal... just wanted to see what you guys thought, and to change the subject |
| 15:02.03 | exoxe | yes |
| 15:02.05 | exoxe | TK is right |
| 15:02.06 | Juggie | ah |
| 15:02.23 | [TK]D-Fender | <- Intuitive bastard |
| 15:02.30 | [TK]D-Fender | I know I kow.... |
| 15:02.34 | Zeeek | what subject? |
| 15:02.36 | [TK]D-Fender | I had you ... at "bastard" ;) |
| 15:02.44 | exoxe | off of the girlscouts subjects |
| 15:02.46 | Juggie | [TK]D-Fender, chanspy says only bridged calls though |
| 15:02.53 | [TK]D-Fender | Juggie: Does it? |
| 15:02.57 | Juggie | i'm not sure voicemail would be considered a bridge |
| 15:03.10 | [TK]D-Fender | Juggie: I wouldn't think so |
| 15:03.26 | exoxe | it's not |
| 15:03.31 | DirtyD | whats the difference between asterisk 1.2.9 and 1.4.5 |
| 15:03.32 | [TK]D-Fender | Juggie: in a bridged calls there are 2 channels and TX/RX is inverted, but you don't need BOTH to spy... |
| 15:03.40 | Juggie | [TK]D-Fender, well, maybe it was changed, it says bridged on the wiki |
| 15:03.45 | exoxe | about 1.6 |
| 15:03.48 | Juggie | but in 'show application chanspy' it does not |
| 15:03.50 | [TK]D-Fender | Juggie: As long as you hear BOTH sides. |
| 15:04.05 | [TK]D-Fender | Juggie: And the WIKI is *never* wrong ;) |
| 15:04.13 | *** join/#asterisk mrdigital (n=mrdigita@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 15:04.13 | tzanger | :-) |
| 15:04.13 | exoxe | hah |
| 15:04.19 | mrdigital | is there a RSS Rwader for Asterisk? |
| 15:04.21 | mrdigital | Reader |
| 15:04.25 | _E-bola | cool |
| 15:04.33 | [TK]D-Fender | Juggie: It could very well work EITHER way, and I don't have any experience to corroborate, but I wouldn't imagine this requirement. |
| 15:04.33 | Zeeek | when someone calls and goes to vmail I can listen to them using zapscan |
| 15:04.37 | _E-bola | the builtin voip software on the hp ipaq 512 works with asterisk |
| 15:04.42 | Zeeek | wouldn't that be the same? |
| 15:04.46 | [TK]D-Fender | mrdigital: Go check the WIKI |
| 15:04.56 | Juggie | [TK]D-Fender, he should be good to go i agree |
| 15:05.51 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:08.09 | Uatec | when i call a sip hphone, i get XXX is Cirtuit busy |
| 15:08.12 | Uatec | what does this mean? |
| 15:10.34 | Mercestes | Uatec, it means their Cirtuit has already reached it's maximum utilization. |
| 15:11.15 | Uatec | it can't have though |
| 15:11.21 | Uatec | there are no other calls being made |
| 15:12.01 | Mercestes | cirtuits are very fickle like that. Sometimes their maximum utilization is 0. |
| 15:12.12 | Mercestes | are you sure the phone is online? |
| 15:12.19 | Uatec | yes |
| 15:12.24 | Uatec | it's ringing for a fraction of a second |
| 15:12.28 | Uatec | and then asterisk reports that error |
| 15:14.02 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:14.08 | [TK]D-Fender | Uatec: Never ask questions like that without providing the full CLI output of the failed call at verbose 10 & SIP debug enabled. |
| 15:16.12 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 15:16.13 | mxmasster | hi all |
| 15:16.28 | mxmasster | i am trying to find information on implementing BLF with asterisk and polycom phones |
| 15:16.58 | mxmasster | in addition to displaying the line as busy/inuse is it possible to pick up and bridge the phones? |
| 15:17.15 | mxmasster | executives at our company want this functionality with their assistants |
| 15:18.00 | _E-bola | Hmm i cant get hints working properly |
| 15:18.12 | _E-bola | state is always unavaileble: anybody got any sugestions to what can be wrong? |
| 15:18.24 | [TK]D-Fender | mxmasster: See lines/phones in use = yes, hijack held calls etc, no. |
| 15:18.26 | Mercestes | _E-bola, what type of phone? |
| 15:18.32 | _E-bola | Mercestes: snom 320's |
| 15:18.43 | Mercestes | _E-bola, what does show hints say? |
| 15:18.54 | _E-bola | 200 : SIP/200 State:Unavailable Watchers 0 |
| 15:19.05 | mxmasster | [TK]D-Fender: what would happen if a person were to select the line that was already in use? |
| 15:19.08 | [TK]D-Fender | _E-bola: Your phone isn't even LOOKING at it |
| 15:19.29 | _E-bola | ahh doh |
| 15:19.38 | _E-bola | i used the extension instaid of the sip name in the hint line |
| 15:19.40 | _E-bola | it says idle now |
| 15:19.41 | [TK]D-Fender | mxmasster: it isn't a button to take that "line or anything. you need to completely FORGEET key-system functionality now. |
| 15:19.54 | _E-bola | Mercestes: do u have experience with those snom's? |
| 15:20.08 | Mercestes | _E-bola, well, that's part of it, but your phone still isn't looking at it. "Watchers 0" means "no one is watching my phone." |
| 15:20.10 | _E-bola | i dont know if i should mark the button/LED as extension, line or shared line |
| 15:20.13 | mxmasster | [TK]D-Fender: so it will show the line in use, but if the user selects the line nothing will happen |
| 15:20.13 | Mercestes | _E-bola, no, just polycoms |
| 15:20.57 | [TK]D-Fender | mxmasster: there is no such thing as "selects that line". What you SEE on your Polycom for presences is not a LINE, it is a speed-dial entry with presence support. |
| 15:21.21 | robl^ | _E-bola: on the snom.. set the buttons you want to use to minitor other phones as extenstions. |
| 15:22.16 | _E-bola | ok, ile try, can it be right that u have to restart the phone for it to work? |
| 15:24.25 | _E-bola | hmm it doesnt want to register as a watcher |
| 15:24.35 | _E-bola | i had it working a few hours ago, but then the button just was on constantly |
| 15:25.01 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:25.18 | _E-bola | ahh there we go |
| 15:25.36 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 15:26.27 | lee_is_me | Is zapata.conf/CallProgress still experimental? I had it turned on and it caused one way audio.... |
| 15:27.28 | *** join/#asterisk af_ (n=getsmart@81-174-44-245.dynamic.ngi.it) |
| 15:27.29 | [TK]D-Fender | lee_is_me: It is the new name for "dropcallsrandomly=yes" and "ifconnectedspoardicglitches=yes" |
| 15:27.35 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 15:27.44 | lee_is_me | lol |
| 15:28.00 | lee_is_me | Sangoma tech told me to enable it, wrong answer. |
| 15:28.04 | mxmasster | is there a web configurator for polycom phones available that will work with asterisk? |
| 15:28.20 | [TK]D-Fender | mxmasster: the 2 have NOTHING to do with each other. |
| 15:28.21 | mxmasster | i need something that will create the config files automatically |
| 15:28.35 | Uatec | Mercestes, it was something to do with the firmware |
| 15:28.39 | Uatec | it's on the latest firmware now |
| 15:28.40 | [TK]D-Fender | mxmasster: Trixbox has one in there somewhere. Go see if you can slice it off. |
| 15:28.52 | lee_is_me | mxmasster: I started on a software based system, but got caught up with another project |
| 15:29.01 | mxmasster | last i looked at trixbox it only supported cisco |
| 15:29.19 | [TK]D-Fender | mxmasster: Look again. SIPX also has one IIRC |
| 15:31.21 | aptura | well looks like CID spoofing abuse is hitting the congressional floor and cnn.com |
| 15:32.04 | [TK]D-Fender | aptura: Fuck gov't, its only fair :) |
| 15:32.36 | aptura | Yea I know. Take a few CID abusers/resellers and it spoils it all for us lagit companies. |
| 15:33.16 | aptura | TK have you tested any wifi phones that just auto logon when in the presents of a dhcp/wifi connection? |
| 15:33.24 | aptura | presence |
| 15:33.25 | aptura | :) |
| 15:33.29 | Strom_M | legiy |
| 15:33.31 | Strom_M | er |
| 15:33.33 | Strom_M | legit |
| 15:33.46 | Strom_M | there's no "a" in it :) |
| 15:33.54 | aptura | okay :) |
| 15:34.13 | aptura | I did not even know it was a Dictionary based word :) |
| 15:34.19 | Strom_M | legitimate -> legit |
| 15:34.24 | aptura | I know |
| 15:34.25 | aptura | :) |
| 15:34.30 | Strom_M | doofus |
| 15:36.20 | tzanger | haha |
| 15:36.23 | tzanger | I haven't heard that word in ages |
| 15:36.24 | tzanger | doofus |
| 15:36.49 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 15:40.33 | anonymouz666 | anyone in here knows what defines a xferfailsound = beeperr ; to indicate a failed transfer ? |
| 15:44.33 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 15:49.29 | Uatec | i'm using Snap to dial my desk phone |
| 15:49.34 | Uatec | but for some reason, i just get: |
| 15:49.42 | Uatec | <PROTECTED> |
| 15:49.42 | Uatec | <PROTECTED> |
| 15:49.46 | Uatec | in my CLI |
| 15:49.54 | Uatec | where ntsols is the manager username |
| 15:49.59 | Uatec | it's just not dialing |
| 15:50.08 | Uatec | anybody know why? |
| 15:50.09 | _VoiceMeUp_COM | error soemwhere |
| 15:50.14 | _VoiceMeUp_COM | check the logs at full debug |
| 15:52.58 | anonymouz666 | Qwell |
| 15:53.18 | anonymouz666 | I installed svn trunk and I am trying to compile the chan_mobile |
| 15:53.29 | Uatec | how can i turn up debugging |
| 15:53.41 | anonymouz666 | I got everything but I am getting error on HCI_BLAH_BLAH constant |
| 15:53.44 | anonymouz666 | do you have any idea? |
| 15:53.55 | Mercestes | Uatec, core debug 1 and add "debug" to the console line in "logger.conf" and then pull a logger reload |
| 15:54.48 | Uatec | i just did that |
| 15:54.55 | Uatec | i'm still getting no more information |
| 15:56.30 | *** join/#asterisk seele_ (n=seele@webserver.datawareltda.com) |
| 15:56.39 | seele_ | Hi |
| 15:56.45 | seele_ | <PROTECTED> |
| 15:56.53 | *** join/#asterisk af_ (n=getsmart@81-174-9-236.dynamic.ngi.it) |
| 15:56.57 | *** join/#asterisk ManxPower (n=manxpowe@192.sub-70-220-233.myvzw.com) |
| 15:56.59 | seele_ | in this moment my caller ID is *LOC*XXXXXXX#->*LOC*XXXXXXX# |
| 15:57.15 | af_ | asterisk supports video as well? |
| 15:57.16 | seele_ | I need to change this for use a phone book application |
| 15:57.23 | seele_ | af_, yes |
| 15:57.40 | kombi_ | can one ban callerids? |
| 15:57.58 | af_ | h323 and sip? * can act as mcu ? any doc somewhere on this? |
| 15:59.17 | *** part/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:00.01 | Zeeek | Join us in exactly 30 minutes for the AUC: http://x2z.eu and IRC #asterisk-users-conference |
| 16:02.07 | *** join/#asterisk Defraz (n=t0tal@64-13-40-162.boi.clearwire-dns.net) |
| 16:03.04 | Zeeek | our first guest: http://pagead2.googlesyndication.com/pagead/imgad?id=CO-XrMO02bW_VBB4GMIEMggOrSt2VHD7Rg |
| 16:03.19 | *** part/#asterisk SuD (n=Ask@89.140.32.2.static.user.ono.com) |
| 16:05.13 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 16:05.15 | holiday_42 | kombi_: sure, it's even document in THE book |
| 16:05.20 | holiday_42 | ~thebook |
| 16:05.20 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:05.44 | holiday_42 | look for "ex-girlfriend" |
| 16:12.56 | Zeeek | asterisk is too male-centric |
| 16:14.15 | kombi_ | holiday_42: thanks! |
| 16:14.33 | kombi_ | Zeeek's ex-girlfriend from that picture will be on it.. |
| 16:14.40 | *** join/#asterisk stoffell_h (n=stoffell@d51A58109.access.telenet.be) |
| 16:15.00 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 16:15.00 | Zeeek | I would not use asterisk or anything else to prevent that woman from calling me! |
| 16:15.23 | Zeeek | or showing me how to remain calm |
| 16:15.25 | *** join/#asterisk DirtyD (n=DigiD@ool-18bddad8.dyn.optonline.net) |
| 16:15.29 | DirtyD | wah |
| 16:15.32 | kombi_ | because you don't know she suffers from mild shankar.. |
| 16:15.40 | Zeeek | Ravi's cousin? |
| 16:15.40 | seele_ | LOL |
| 16:16.05 | Zeeek | I jammed with Mild Shankar once, he really knows his ragas |
| 16:16.14 | Zeeek | but we gigress |
| 16:16.23 | kombi_ | playing what? |
| 16:16.26 | Zeeek | gigress is to digress a billions times |
| 16:16.38 | Zeeek | that beer really helps, eh? |
| 16:16.57 | kombi_ | if only there was some already.. |
| 16:17.13 | Zeeek | some what? |
| 16:17.21 | kombi_ | stout.. |
| 16:17.33 | Zeeek | there were only two beers and I drank one on the way home |
| 16:17.44 | kombi_ | you're sober then! |
| 16:17.57 | Zeeek | good to go |
| 16:18.09 | kombi_ | now, what's with that strange phone in podcast? |
| 16:18.19 | Zeeek | what phone? |
| 16:18.47 | Zeeek | oops time to start the pre-conf warmup |
| 16:18.47 | kombi_ | exactly that i meant |
| 16:19.25 | kombi_ | do i just dial 123@thatip and we all speak (the whole male world of * hackers)? |
| 16:19.48 | Zeeek | I want females to join |
| 16:19.52 | Zeeek | but they have lives |
| 16:20.08 | kombi_ | as a matter of fact they do, (some at least) |
| 16:21.15 | robl^ | life is overrated. ;-) ( woulda said that females were overrated, but I value by continued health) |
| 16:21.47 | kombi_ | does dial command take != for a condition? |
| 16:22.04 | kombi_ | robl: is she in the room? |
| 16:22.41 | kombi_ | sorry, I mean GoToIf of cource. |
| 16:22.57 | k31th | ISDN qsiq wat is that ? linking a isdn pbx to a asterisk box ? |
| 16:23.03 | k31th | is that hard work? |
| 16:23.51 | kombi_ | exten => 123,1,GotoIf($[${CALLERIDNUM} != 123456789]?20:10) <- possible? |
| 16:28.08 | *** join/#asterisk Vec (n=Vec@dsl-241-210-159.telkomadsl.co.za) |
| 16:29.10 | Zeeek | PINs for the conference 8775594637 6665551212 |
| 16:32.04 | Vec | I have a serious, but wierd problem, with my asterisk server, after a while a few months, when a call is made it does not ring on either end, just goes dead, however on the CLI it ends with "-- Called 2116" but does not go to Ringing, if I restart asterisk it will fix it, but I am trying to find the source of the problem so it does not happen again ? |
| 16:33.52 | Zeeek | Join us now for the AUC: http://x2z.eu and IRC #asterisk-users-conference |
| 16:36.37 | *** join/#asterisk tekor (n=will@adsl-147-85-232.bhm.bellsouth.net) |
| 16:37.14 | BSD_Tech | ok this sucks |
| 16:37.24 | BSD_Tech | asterisk 1.4.5 crashes on bsd |
| 16:37.43 | BSD_Tech | everytime you load it up it crashes |
| 16:38.19 | *** join/#asterisk peanutb_ (n=paulb@c-24-16-243-186.hsd1.wa.comcast.net) |
| 16:38.36 | kombi_ | Zeeek: ask him about cisco phones |
| 16:38.39 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:39.58 | headasplode | why would voicemail not do anything after putting in my password? |
| 16:40.00 | headasplode | any idas |
| 16:40.02 | headasplode | ideas* |
| 16:40.14 | headasplode | i am turning into one of those ppl on irc that i hate, asking the same questions every 10 minutes |
| 16:40.22 | headasplode | heh |
| 16:41.17 | stoffell_h | headasplode,get your verbosity up and look at the cli when you put in your password.. |
| 16:42.54 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 16:42.55 | BSD_Tech | wow audio sounds horriable |
| 16:43.25 | kombi_ | not quite endorsing ip telephony, is it.. |
| 16:43.32 | Zeeek | heh |
| 16:43.39 | Zeeek | yes, a little rough... |
| 16:43.43 | coppice | BSD_Tech: listening to Britney Spears again? |
| 16:43.43 | Zeeek | 1 bit audio |
| 16:43.49 | *** part/#asterisk evoxy (n=rbw@83.212.57.238) |
| 16:43.57 | BSD_Tech | no to the astconf |
| 16:44.14 | anonymouz666 | app that never will be used in asterisk? app_morsecode |
| 16:44.18 | Zeeek | IRC #asterisk-users-conference |
| 16:44.24 | kombi_ | can you make the guy say something about cisco 79x1 phones? (like trash them or not) |
| 16:44.37 | kombi_ | morsecode, kewl! |
| 16:44.53 | BSD_Tech | zeek what firmware |
| 16:49.20 | BSD_Tech | ok svn works |
| 16:49.22 | BSD_Tech | yes |
| 16:49.30 | BSD_Tech | no crash but no zaptel still |
| 16:51.15 | *** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 16:52.19 | *** join/#asterisk ToyMan (n=Stuart@fw.hvs.bsdwebsolutions.com) |
| 16:52.36 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 16:52.55 | *** join/#asterisk Gtwy (i=nobody@65.111.166.56) |
| 16:52.57 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:53.30 | Gtwy | other than the D-Link DVG-2001S, what is a good suggestion for an Analog Phone Converter? |
| 16:53.30 | *** join/#asterisk Borbosha (n=Borbosha@72.16.139.65) |
| 16:53.49 | Gtwy | newegg had many complaints about the D- |
| 16:53.51 | Gtwy | D-Link |
| 16:54.28 | stoffell_h | Gtwy, the iaxy from digium? |
| 16:54.33 | Borbosha | I have a problem with my Trixbox install. I have the system setup and our SIP softphones can connect from inside the network and everything works correctly. However, when we try to connect our softphones from our work network into our Asterisk box (on a DMZ at home) we can't get any audio at all. Any ideas? |
| 16:55.21 | stoffell_h | Borbosha, network issue, sip and nat .. have a look at voip-info.org, coz sip and nat is not really the best marriage.. |
| 16:55.56 | Borbosha | So, because our softphones are behind a NAT router at work we are having problems? |
| 16:56.59 | stoffell_h | if you use nat.. you should be very careful 9;any pitfalls0, so that's my first idea, yes |
| 16:57.14 | stoffell_h | ouch.. that should read: many pitfalls ... :p |
| 16:58.00 | Borbosha | Hmm, alright. It seems a little odd that with the server on a DMZ that it would have that many problems. I will continue my research and see if I can figure it out |
| 16:58.33 | stoffell_h | Borbosha, yep. have a look at sip and nat on voip-info.org ... g'luck |
| 16:59.58 | *** join/#asterisk AtomicDawg (n=atomicda@74-206-0-80.static-ip.m.telepacific.net) |
| 17:00.55 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 17:01.17 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 17:02.56 | Gtwy | what do you guys think of this ? http://reviews.cnet.com/networking-and-wi-fi/sipura-spa-1001-voip/4507-3243_7-31656779.html?tag=nav |
| 17:04.38 | Hmmhesays | what up guys |
| 17:06.53 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 17:07.43 | *** part/#asterisk jzweng (n=john@85-125-14-210.static.xdsl-line.inode.at) |
| 17:12.34 | *** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com) |
| 17:18.45 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:19.01 | BSD_Tech | ok svn branches head works this is nice I have my tester box back |
| 17:21.46 | murr4y | http://isitfriday.net/ |
| 17:29.17 | _VoiceMeUp_COM | anyway to debug DTMF |
| 17:30.33 | stoffell_h | murr4y, tgif ! |
| 17:30.47 | *** join/#asterisk snook3r (n=root@bzq-219-46-202.isdn.bezeqint.net) |
| 17:31.05 | murr4y | tgif? |
| 17:31.18 | Mercestes | <PROTECTED> |
| 17:31.18 | murr4y | thank god it's friday |
| 17:31.19 | murr4y | :) |
| 17:31.23 | murr4y | indeed |
| 17:31.26 | stoffell_h | ;) |
| 17:31.31 | Mercestes | 404 |
| 17:31.52 | murr4y | yea it didn't work here either, didn't test it first |
| 17:31.57 | murr4y | http://isitsaturday.net/ worked |
| 17:32.02 | murr4y | let's save that for tomorrow ;) |
| 17:32.15 | stoffell_h | lol |
| 17:32.31 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 17:38.02 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 17:41.29 | *** join/#asterisk el_4_jinete (n=kvirc@mail.pulxar.com.co) |
| 17:41.51 | el_4_jinete | Hi everybody! :) |
| 17:42.10 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:42.44 | el_4_jinete | Someone has installed a te110p on a ML150 G3? Any problems? |
| 17:43.28 | stoffell_h | el_4_jinete,does it have iLO ? |
| 17:43.51 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:43.52 | el_4_jinete | yes |
| 17:43.58 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:45.00 | stoffell_h | el_4_jinete, well, in case you have problems, just disable iLO .. i had some 'serious' issues with an ML350 G5 and iLO when using an E1 |
| 17:45.18 | stoffell_h | solution: disable iLO and go crazy at HP .. |
| 17:45.40 | stoffell_h | but if you make sure it doesn't drop any IRQ's and doesn't share to many IRQ's, you should be fine |
| 17:47.00 | el_4_jinete | That's the reason to change the server, but the only way is a ML150 G3 |
| 17:47.11 | stoffell_h | what server you use now ? |
| 17:47.29 | el_4_jinete | No way... it's a clone with intel 945 |
| 17:47.45 | stoffell_h | do you share IRQ's right now, on that server ? |
| 17:48.34 | el_4_jinete | I don't see shared irqs in the card. and I don't know whats wrong |
| 17:49.02 | stoffell_h | what is the error exactly ? |
| 17:49.19 | el_4_jinete | ok just a min.. |
| 17:49.38 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 17:50.10 | el_4_jinete | PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 17:50.35 | el_4_jinete | FCS error and some other related |
| 17:51.17 | stoffell_h | does the E1 turn red or orange/green after that ? |
| 17:51.27 | ramindia_ | can some one clarify this, if the user using g729 and server side kept g729, and provider accepting g729, do i need g729 license at my asterisk server ? |
| 17:51.34 | *** part/#asterisk Borbosha (n=Borbosha@72.16.139.65) |
| 17:52.58 | el_4_jinete | in the asterisk log I see yellow |
| 17:53.11 | Hmmhesays | so has anyone tried these grandstream video phones? |
| 17:53.16 | el_4_jinete | sorry can you explain it? |
| 17:53.58 | xkev | ramindia_, you don't need g729 license for passthrough |
| 17:54.06 | stoffell_h | el_4_jinete,okay, if you ' head /proc/zaptel/1' does it show IRQ misses ? |
| 17:54.10 | xkev | but if you expect to play any sounds you'll need one |
| 17:54.34 | ramindia_ | ok |
| 17:54.36 | ramindia_ | thanks |
| 17:54.39 | xkev | ..if something fails to negotiate 729, you may run into a need for transcode in the path |
| 17:54.47 | xkev | it's a sketchy situation imo |
| 17:55.11 | el_4_jinete | Yes, but a few |
| 17:55.13 | ramindia_ | that will take care by Voip Provider right |
| 17:55.43 | stoffell_h | el_4_jinete,how many ? and more importantly, does the number add up? |
| 17:55.47 | xkev | well, if you make sure you offer only 729 to provider/phone then it will be consistent |
| 17:55.57 | [TK]D-Fender | el_4_jinete: pastebin "cat /proc/interrupts" , "dmesg" |
| 17:55.59 | [TK]D-Fender | ~pb |
| 17:56.00 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:56.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^66 |
| 17:56.19 | xkev | but if you expect to have voicemail on * or something, then you have to offer non-729 to phone and you can end up negotiating the wrong codec |
| 17:56.50 | el_4_jinete | Sorry, I don't have the server here. I've decided to change the server to ML150g3 |
| 17:57.27 | ramindia_ | no iam just using as a registrar to forward a call to provider |
| 17:57.34 | ramindia_ | i dont run any services |
| 17:57.45 | [TK]D-Fender | el_4_jinete: thats what SSH is for... |
| 17:59.05 | el_4_jinete | Yes, I know but the customer doesn't allow me ssh |
| 18:00.14 | el_4_jinete | Ok, well. If someone has installed on a ML150G3 pls tell me your experience |
| 18:00.22 | [TK]D-Fender | el_4_jinete: Let me guess you also have to work on it blindfolded with 1 arm tied behind your back too! ;) |
| 18:00.44 | [TK]D-Fender | el_4_jinete: With that kind of working environment, don't expect much help... |
| 18:00.49 | *** join/#asterisk syamajala (n=syamajal@c-75-68-225-205.hsd1.ma.comcast.net) |
| 18:00.53 | *** join/#asterisk limpc (i=m-00kie@ip67-155-169-170.z169-155-67.customer.algx.net) |
| 18:01.01 | el_4_jinete | Thank, you. I'm going lunch |
| 18:01.08 | syamajala | anyone have any experience with the linksys spa942 |
| 18:01.24 | [TK]D-Fender | el_Sorry, just the truth.... we can't help much if yuo can't even get to the box... |
| 18:01.41 | [TK]D-Fender | syamajala: Yes. Its a phone. |
| 18:01.45 | syamajala | yeah |
| 18:01.46 | [TK]D-Fender | NEXT!@!@!@!@ (c) BKW |
| 18:01.56 | stoffell_h | lol |
| 18:02.02 | syamajala | i'm planning an asterisk system for my parents |
| 18:02.10 | Gtwy | can anyone suggest a good US VOIP company |
| 18:02.11 | limpc | ew. |
| 18:02.38 | syamajala | i just did a basic setup at home with trixbox so i could figure out how it works |
| 18:02.40 | limpc | syamajala - why would you do that? :P |
| 18:02.46 | [TK]D-Fender | Gtwy: Cisco. |
| 18:02.49 | [TK]D-Fender | NEXT!@!@!@!@ (c) BKW |
| 18:02.57 | syamajala | limpc: they have a partner system |
| 18:03.26 | limpc | but now they have an excuse to get you over to their house - "our phone crashed!" |
| 18:03.27 | limpc | :P |
| 18:03.37 | exoxe | Are you Being Served? |
| 18:03.37 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.145.12) |
| 18:03.50 | [TK]D-Fender | syamajala: figure out how "what" works? |
| 18:04.02 | syamajala | limpc: we have 2 lines at home i sit up asterisk on the second line |
| 18:04.23 | limpc | yeah i figured as much |
| 18:04.26 | exoxe | :-\ |
| 18:04.28 | syamajala | [TK]D-Fender: asterisk |
| 18:04.29 | limpc | but again.. WHY? heh |
| 18:04.50 | limpc | asterisk isnt of much use in the home |
| 18:04.51 | [TK]D-Fender | syamajala: No, Asterisk DOESN'T work, we're all here talking about GIRL SCOUTS. |
| 18:05.01 | syamajala | well i wouldn't want to go their office and just try to setup asterisk on the spot |
| 18:05.14 | limpc | [TK]D-Fender is licking stamps laced with pcp. |
| 18:05.28 | syamajala | this is just a test setup |
| 18:05.42 | [TK]D-Fender | syamajala: If you're talking TRIXBOX then don't say "Asterisk", because thats only 1 piece and TRXBOX won't help you learn ANYTHING about *. |
| 18:05.46 | limpc | you have a digium card? |
| 18:05.56 | syamajala | limpc: yes |
| 18:06.04 | [TK]D-Fender | limpc: You missed our Girl Scout conversation earlier..... |
| 18:06.26 | limpc | what do you plan to do with the box? |
| 18:06.44 | [TK]D-Fender | limpc: SOOO many ways I could trist that ;) |
| 18:06.48 | [TK]D-Fender | twist* |
| 18:07.05 | limpc | well just as long as you stay away from my box :P |
| 18:07.12 | syamajala | it should replace their partner system and get them in a good position to eventually migrate to voip |
| 18:07.15 | [TK]D-Fender | homonym Freudian slip ;) |
| 18:07.44 | limpc | why didnt you just plug in sipuras? :) |
| 18:07.45 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 18:07.55 | [TK]D-Fender | syamajala: * does not necessarily mean VoIP. |
| 18:08.20 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 18:08.31 | syamajala | [TK]D-Fender: i mean get rid of their analog phone service and get their phone service from a voip provider |
| 18:08.50 | [TK]D-Fender | syamajala: Don't need * to do that.... nor would I SUGGEST it. |
| 18:09.26 | syamajala | this isn't for the home |
| 18:09.28 | syamajala | its for their office |
| 18:10.12 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 18:10.31 | stoffell_h | now, how to avoid music-on-hold giving noise (distortion) when the music is a bit silent ? |
| 18:11.13 | limpc | that's probably jitter |
| 18:11.24 | syamajala | their office has 5 lines |
| 18:12.18 | limpc | syamajala - voip isnt too cheap yet with less than 10 lines.. are you going to upgrade them? do they max out 5 lines? |
| 18:12.18 | stoffell_h | limpc, oh, is there something i can check? (i have the issue when calling in from isdn (bristuff, zaptel) to a pbx.. the music files are already in pcm format |
| 18:12.51 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 18:13.02 | limpc | its been a while since ive tweaked asterisk configs - i dont think i can be much help right now |
| 18:13.07 | *** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net) |
| 18:13.14 | syamajala | limpc: well my original plan was to stick with the analog lines for right now and then move to voip somewhere down the road |
| 18:13.32 | *** part/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 18:13.34 | stoffell_h | okay, thanks anyway ;) |
| 18:13.58 | limpc | syamajala - if this is WAY down the road, like 1-2 years, then theres not much point in this :P Asterisk will likely have had a facelift or changed its config format by then.. heh |
| 18:14.32 | limpc | why not just sign em up with something like Skype and install Sipura ATA boxes :) |
| 18:14.39 | sandorp | what's the command to "train" the zaptel hardware so it is "optimized" for working with my hardware? |
| 18:14.50 | Trevor_b | fxotune? |
| 18:15.07 | sandorp | Trevor_b: aha, that's it |
| 18:15.09 | sandorp | thanks |
| 18:15.22 | Trevor_b | i think that optimizes the card for impedance and such, not the hardware its installed in. |
| 18:15.34 | [TK]D-Fender | syamajala: I highly recommend AGAINST VoIP termination FOR companies. |
| 18:15.56 | syamajala | hmm |
| 18:16.09 | sandorp | Trevor_b: I think you're right .. I knew it optimized something or other |
| 18:16.24 | limpc | it optimizes my nuts for optimal cooling distance |
| 18:16.35 | limpc | once i found them on the floor :( |
| 18:16.42 | BSD_Tech | Internal RTCP NTP clock skew detected: lsr=1936982016, now=2219555784, dlsr=2115119343 (32274:159ms), diff=1832545575 |
| 18:16.49 | BSD_Tech | what is going on with asterisk |
| 18:17.05 | BSD_Tech | my screen is fullof these in the cli |
| 18:17.30 | syamajala | so basically, the 3 solutions would be the sipura boxes, asterisk connecting to the analog lines, or leaving everything as it is |
| 18:17.49 | sandorp | BSD_Tech: maybe your PC clock is inaccurate so NTP is trying to get your time back in sync? |
| 18:17.58 | [hC] | Anyone seen polycom sip firmware 2.1.2 around? One of their support people told me i should grab it to fix some problems, yet i cant find it on their site. |
| 18:17.58 | limpc | yeah. you could pull in a T1 trunk and split the lines between voice and data :D |
| 18:18.03 | BSD_Tech | but the system clock is right |
| 18:18.08 | BSD_Tech | its set by ntp |
| 18:18.17 | limpc | what about hwclock? |
| 18:18.25 | [TK]D-Fender | syamajala: What do you expect for * to do for you that the previous system DOESN'T? |
| 18:18.26 | BSD_Tech | this only started happening with 1.4.5 |
| 18:18.43 | limpc | [TK]D-Fender - cook, of course |
| 18:18.53 | limpc | asterisk makes an awesome lead soup |
| 18:19.10 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 18:19.21 | limpc | sometimes it even throws in some dielectric and burnt carbon.. my favorite |
| 18:19.50 | BSD_Tech | lol |
| 18:19.52 | anonymouz666 | nice chan_mobile crashes at every single safe_asterisk |
| 18:21.10 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 18:21.37 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 18:21.52 | [TK]D-Fender | limpc: My * used to make me coffee, so I guess thats a viable plan... |
| 18:22.03 | stoffell_h | okay, the music on hold has something to do with the files.. some play better then others.. so i guess it's back to analysing them now.. |
| 18:22.04 | syamajala | [TK]D-Fender: well its a doctor's office, so when my dad is on call he is not always in the office |
| 18:22.11 | [TK]D-Fender | anonymouz666: At least its consistent and isn't just crashing SOME of them ;) |
| 18:22.20 | syamajala | so things need to get routed to his cell phone |
| 18:22.27 | BSD_Tech | chan_mobile |
| 18:22.45 | [TK]D-Fender | BSD_Tech: chan_loadmeforrandomsegfaults.so |
| 18:23.09 | BSD_Tech | lol |
| 18:23.13 | [TK]D-Fender | stoffell : no VBR, no ID3 tags |
| 18:23.31 | limpc | [TK]D-Fender - feasible. |
| 18:23.43 | Trevor_b | does genzaptelconf give any interface when it finds PRI equipment, or does it just guess at the pri_net/pri_cpe and the signalling? |
| 18:23.51 | limpc | chan_win32bsoddemo.so |
| 18:24.16 | BSD_Tech | chan_crashmysystem.soi |
| 18:24.22 | BSD_Tech | chan_crashmysystem.so |
| 18:24.33 | limpc | syamajala - i dont think asterisk will be theb est option for that.. |
| 18:24.45 | limpc | syamajala - it'd tie up both lines forwarding the call |
| 18:24.46 | [TK]D-Fender | limpc: entirely. If I ahd my old microwave it could make me popcorn (wait... I HAVE an old manual popper!), kill the lights and start my movie for me and stop ringing on incoming calls. :) |
| 18:24.57 | stoffell_h | [TK]D-Fender, aye.. but i already converted it to pcm and also tried with gsm, no mp3's here.. but will compare some files with an audio editor now |
| 18:25.10 | syamajala | limpc: i see |
| 18:25.15 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:25.25 | [TK]D-Fender | stoffell : in that case make sure they're 8khz mono |
| 18:25.45 | *** join/#asterisk peanutb_ (n=paulb@c-24-16-243-186.hsd1.wa.comcast.net) |
| 18:25.58 | Gtwy | no, i was looking for something like http://connect.voicepulse.com |
| 18:26.02 | Gtwy | but i dont know if i want to go with them |
| 18:26.02 | limpc | syamajala - you probably want call forwarding as part of your phone service |
| 18:26.17 | limpc | are his 5 lines from an analog trunk? |
| 18:26.39 | sandorp | how do I configure voicemail to be recorded at a higher volume? voicemails sound really quite when I play them back |
| 18:26.42 | syamajala | as far as i know |
| 18:27.16 | syamajala | they are |
| 18:27.56 | limpc | you could install something like a nortel unit to do the forwarding |
| 18:28.18 | BSD_Tech | gawd |
| 18:28.22 | limpc | nortel callpilot |
| 18:31.26 | rob0 | chan_bonkers.so |
| 18:31.52 | mocker | woo, just had to setup an asterisk system for my CEOs house. |
| 18:32.10 | mocker | That's either really good, or really bad. :( |
| 18:32.17 | limpc | heh |
| 18:32.36 | limpc | probably the latter :P why is everyone setting up asterisk on home phones?? geez. |
| 18:32.55 | mocker | limpc: He just got a new house and only wired CAT-6 to the rooms. |
| 18:33.08 | limpc | um did he have it built? |
| 18:33.12 | mocker | I guess so. |
| 18:33.25 | limpc | well he's still dumb :P |
| 18:33.35 | limpc | should have gone with fiber. |
| 18:33.40 | mocker | hah. |
| 18:33.42 | Gtwy | whats wrongn with asterisk in a home seteting |
| 18:33.43 | Gtwy | setting |
| 18:33.50 | denon | nothing, most of us have * boxes at home |
| 18:34.03 | limpc | i dont see the point of an asterisk box in the home.. |
| 18:34.05 | mocker | My wife hates it, but I love it. :) |
| 18:34.18 | mocker | limpc: super cheap phone bill |
| 18:34.27 | denon | limpc: why not? Ive got cross-connects to so my work calls follow me home and such |
| 18:34.29 | Gtwy | i am setting one up now |
| 18:34.29 | limpc | not really |
| 18:34.35 | Jingles | the point? man, I love having an * box at home. |
| 18:34.36 | Gtwy | but i am unsure what company to go with to get service |
| 18:34.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:34.38 | denon | and it lets me call my overseas friends via ip easier |
| 18:34.48 | Jingles | my roommate have a list of 'known #s' that route calls straight to our respective phones |
| 18:35.04 | Jingles | and anyone that's not on the 'callerID list' gets shoved to a voice prompt system. |
| 18:35.27 | Gtwy | mocker: why does your wife hate it |
| 18:35.58 | mocker | Gtwy: Because for awhile I was using a Diguim card to push signal to my phones.. |
| 18:36.05 | limpc | denon - for the serious techie it may have some uses but for the general public, theres just no point.. i dont want to get called to some neighbor's home cuz some dofus convinced them to let him set them up with asterisk, then moved out of state :P |
| 18:36.14 | mocker | That digium card was on the same PCI bus as two PVR-250s |
| 18:36.35 | mocker | And an NVidia card.. |
| 18:36.44 | Gtwy | so it wasnt working correctly? |
| 18:36.47 | mocker | Suffice it to say... things would get a little choppy. |
| 18:36.52 | Gtwy | ah |
| 18:37.09 | limpc | doesnt the first PCI slot have priority? |
| 18:37.21 | limpc | first or last cant remember which |
| 18:37.41 | mocker | You can usually find out where your cards are with lspci -bv or something like that.. |
| 18:37.49 | mocker | But the machine in general was overloaded. |
| 18:37.51 | Gtwy | now, i am new to this, so tell me if i am wrong.. i am just attaching it to my network and using a digital to analog converter, such as something like this http://www.sipura.com/products/spa3000.htm to hook a phone up to the network |
| 18:38.12 | mocker | Now I colo my asterisk server and use an SPA to connect to it and drive my home phones. |
| 18:38.16 | Gtwy | it will be able to connect to the * server despite me not having any type of digium cards or whatever , right? |
| 18:38.23 | _VoiceMeUp_COM | anyone can point me to the debug switch to see dtmf ? |
| 18:38.32 | limpc | mocker - yeah thats what i did :) |
| 18:38.55 | mocker | limpc: That way I don't have to worry about my cable modem freezing up. :) |
| 18:38.58 | BSD_Tech | asterisk 1.4.5 is a pain |
| 18:39.02 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 18:39.04 | limpc | had this awesome colo that colo'd our boxes for $50 a month per 1U, and $50 a month for a full T1/PRI |
| 18:39.13 | mocker | 1.2 fo' life |
| 18:39.14 | mocker | ;) |
| 18:39.23 | BSD_Tech | this is a testbox with the asterisk gui for dev |
| 18:39.28 | mocker | limpc: Free colo! |
| 18:39.28 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 18:39.29 | Gtwy | mocker: did you see my question |
| 18:39.33 | _VoiceMeUp_COM | seems like AST -> PRI -> PRI -> someones pbx not sure make and model not taking dtmf |
| 18:39.40 | _VoiceMeUp_COM | but if i call via a land line all is good |
| 18:39.42 | BSD_Tech | 1.2.x will die off in a year |
| 18:39.57 | _VoiceMeUp_COM | in 12 montsh 1.4 will be stable |
| 18:39.58 | irule | can you recommend me a nice path to getting started with CDR? |
| 18:40.02 | _VoiceMeUp_COM | if not 1.6 |
| 18:40.03 | limpc | unfortunately they changed business models and shut down their pbx service :( |
| 18:40.23 | limpc | now im squatting over at Level3 |
| 18:40.32 | mocker | Gtwy: That seems fine for a basic setup. |
| 18:40.34 | limpc | bit pricey but they have some nice pipes :) |
| 18:40.47 | mocker | So use that to connect to your asterisk box and drive your analog phones.. |
| 18:41.10 | Gtwy | yeah, there is only one set of wireless phones with one plug |
| 18:41.29 | BSD_Tech | Asterisk SVN-branch-1.4-r71123M ok I just updated and the issue goes away |
| 18:42.19 | irule | BSD_Tech what issue goes away? |
| 18:42.34 | BSD_Tech | Internal RTCP NTP clock skew detected: lsr=1936982016, now=2237679233, dlsr=2115119343 (32274:159ms), diff=1814422126 |
| 18:43.03 | BSD_Tech | 1.4.5 was loading my screen with this |
| 18:43.08 | mocker | BSD_Tech: Weird, you sync to an NTP server, right? |
| 18:43.12 | BSD_Tech | I just updated and no more issue |
| 18:43.14 | mocker | Oh, it was an asterisk issue.. |
| 18:43.20 | _VoiceMeUp_COM | well aint it bad to sync when in calls ? |
| 18:43.33 | _VoiceMeUp_COM | would make me think it would mess with timing |
| 18:43.35 | _VoiceMeUp_COM | no? |
| 18:43.53 | BSD_Tech | zaptel/ztdummy |
| 18:44.05 | _VoiceMeUp_COM | nice skew.. you syncing with mars ? |
| 18:44.06 | BSD_Tech | but now with the update the issue is gon |
| 18:44.07 | _VoiceMeUp_COM | ;) |
| 18:44.18 | BSD_Tech | lol |
| 18:44.22 | BSD_Tech | no your anus |
| 18:44.25 | BSD_Tech | lol |
| 18:44.26 | _VoiceMeUp_COM | ah |
| 18:44.39 | _VoiceMeUp_COM | that why |
| 18:48.50 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 18:49.56 | exoxe | anyone know the magic sample rate that asterisk is looking for in wav files? |
| 18:50.01 | exoxe | keeps telling me to go away |
| 18:50.15 | exoxe | frequency that is |
| 18:50.27 | exoxe | [TK]D-Fender: ? |
| 18:50.29 | exoxe | hehe, jk |
| 18:53.10 | [hC] | 8000mhz |
| 18:53.12 | exoxe | let me conver tit to gsm with sox and see what happens |
| 18:53.16 | [hC] | if i recall |
| 18:53.17 | exoxe | er, convert it* |
| 18:53.28 | [hC] | 8k mono |
| 18:53.34 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 18:53.45 | [TK]D-Fender | [hC]: KILHERTZ |
| 18:53.50 | lee_is_me | I'm trying to use the AMI Originate function to call a internal phone and transfer it to a specific context, exten and priority. The phone (Polycom 301) rings, but it won't let me answer it... |
| 18:53.54 | lee_is_me | http://www.pastebin.ca/584389 |
| 18:54.03 | lee_is_me | (sample originate) |
| 18:54.23 | [TK]D-Fender | lee_is_me: pastebin the DIALPLAN please... |
| 18:54.24 | [hC] | KILL-IT-HURTS! |
| 18:54.30 | [hC] | [TK]D-Fender: how ya doin :) |
| 18:54.43 | lee_is_me | [TK]D-Fender: http://www.pastebin.ca/584389 |
| 18:55.06 | [TK]D-Fender | lee_is_me: ************* DIALPLAN ******************* |
| 18:55.58 | exoxe | sox wins again |
| 18:56.15 | lee_is_me | [TK]D-Fender: I edited the existing pastebin. Simple context |
| 18:57.03 | lee_is_me | The phone rings, but doesn't do anything when I hit the Answer soft button or pick up the handset. |
| 18:57.54 | mocker | Gtwy: One other thing if you're just starting out. I would avoid the temptation to try trixbox or the like. |
| 18:59.01 | [TK]D-Fender | lee_is_me: You can't edit an old #, it gives you a NEW one... |
| 18:59.47 | lee_is_me | [TK]D-Fender: Sorry, didn't notice that. Here is the link: http://www.pastebin.ca/584399 |
| 19:00.09 | Un1x_Laptop | guys is there a way i can tell asterisk to stop Avoiding 3-way call when in an external conference |
| 19:00.11 | lee_is_me | [TK]D-Fender: This is the only exten: exten=>s,1,MeetMe(${dropped_conf}) |
| 19:00.30 | [TK]D-Fender | lee_is_me: I want to see the entire context for EVERYTHING dialed. |
| 19:00.44 | [TK]D-Fender | lee_is_me: And CLI output to match. |
| 19:06.10 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 19:06.21 | dijungal | hi guys... |
| 19:06.51 | Daejeo1 | TK]D-Fender |
| 19:06.58 | *** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net) |
| 19:07.15 | Daejeo1 | [TK]D-Fender ping |
| 19:07.25 | dlynes | pong |
| 19:07.39 | dijungal | i have a cisco 3661 router sending H.323 calls to an asterisk 1.4.5 server, the box picks up the call but i get no audio on the calling end.. any ideas? |
| 19:08.01 | [TK]D-Fender | Daejeo1: pong |
| 19:08.05 | *** join/#asterisk kimo_sabe (n=nlopez@int.espri.arizona.edu) |
| 19:08.20 | nephfl | my asterisk just hung |
| 19:08.23 | nephfl | and i dont know why |
| 19:08.35 | dlynes | nephfl: define 'hung' |
| 19:08.36 | nephfl | and i am on site...and people are pissed...lol |
| 19:08.44 | nephfl | asterisk running 98 |
| 19:08.47 | nephfl | % cpu |
| 19:08.50 | [TK]D-Fender | nephfl: push the button labeled "Reset" |
| 19:09.00 | nephfl | calls in dont play back |
| 19:09.08 | nephfl | ssh is seriously delayed |
| 19:09.10 | kimo_sabe | tz=mountain in the [general] section is supposed to set the default voicemail timezone, right? |
| 19:09.24 | nephfl | ok... i killed the process via slow ssh |
| 19:09.51 | Mercestes | nephfl, Try unplugging it and plugging it in really quick so it doesn't really *loose* power but kills the voltage enough to cause it to drop intensive processes |
| 19:09.59 | Mercestes | nephfl, In fact, try that several times in rapid successsino. |
| 19:10.14 | dlynes | Mercestes: snicker |
| 19:10.18 | Mercestes | lol\ |
| 19:10.22 | Daejeo1 | my box is on private network, my friend is trying register from his office. " error 408 time out" |
| 19:10.25 | Mercestes | sorry, been reading BOFH stories. : ( |
| 19:10.28 | kimo_sabe | Mercestes: does that actually work to let the smoke out? I've never tried it |
| 19:11.10 | Daejeo1 | >[TK]D-Fender : my box is on private network, my friend is trying register from his office. " error 408 time out" |
| 19:11.20 | Mercestes | kimo_sabe, not anymore, really. |
| 19:11.23 | [TK]D-Fender | ~sipnat |
| 19:11.24 | jbot | somebody said sipnat was for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:11.38 | Mercestes | kimo_sabe, Unless they did something silly like, "restore previous powerstate on powerloss" in their bios. |
| 19:12.02 | BSD_Tech | nat: If you are behind a NAT you probably need to create an /etc/asterisk/sip_nat.conf file with AT LEAST these two lines: 1) externip=your.external.dotted.IPaddess 2) localnet=192.168.0.0/255.255.255.0 (assuming your local network uses 192.168.0.x addresses). Then sip reload from the CLI. Also make sure that the correct UDP ports are open in your firewall and pointed at your Asterisk server-see ? ports and ? rt |
| 19:12.11 | dlynes | Mercestes: not so silly |
| 19:12.15 | BSD_Tech | rtp: Make sure that /etc/asterisk/rtp.conf shows rtpstart=10001 (NOT 10000 due to potential conflict with Webmin) and rtpend=20000 |
| 19:12.19 | UCFmethod | anyone know what happened to ${CALLERIDNUM} in 1.4? It says its been deprecated, but how else do you allow access to voicemail without passwords? |
| 19:12.29 | dlynes | Mercestes: it's kind of a must in areas where you'll have a power outage that will outlast the UPS |
| 19:12.33 | kimo_sabe | UCFmethod: ${CALLERID(num)} |
| 19:12.41 | [TK]D-Fender | UCFmethod: "show funtion CALLERID" |
| 19:12.42 | dlynes | Mercestes: and that you don't have 24 hr access to |
| 19:12.44 | Mercestes | dlynes, True, if it's a server, on a UPS so power state shoul dnever ben an issue |
| 19:12.52 | UCFmethod | kimo_sabe: YOU ROCK |
| 19:12.58 | UCFmethod | [TK]D-Fender: YOU also ROCK |
| 19:13.03 | *** join/#asterisk peanutb (n=paulb@c-24-16-243-186.hsd1.mn.comcast.net) |
| 19:13.05 | [TK]D-Fender | UCFmethod: And read the damn upgrade.txt and million articles that come out with every major release! |
| 19:13.24 | Mercestes | no.....don't read the articles. |
| 19:13.25 | dlynes | [TK]D-Fender: and ChangeLog.txt!!! |
| 19:13.31 | Mercestes | we dont' want you to see how much ppl say we suck... |
| 19:13.35 | UCFmethod | [TK]D-Fender: I did, it explained those were deprecated, but neversaid what replaced it |
| 19:13.36 | Mercestes | just read the upgrade.txt |
| 19:13.43 | [TK]D-Fender | "It was a sad day when the Lone Ranger learned that Kimosabe was actually native for 'Horse's Ass'"" |
| 19:13.55 | kimo_sabe | [TK]D-Fender: Exactly! |
| 19:14.07 | BSD_Tech | lol |
| 19:14.12 | holiday_42 | erm... i thought it meant "soggy bush" |
| 19:14.28 | dlynes | and bogan was native for shit, as in 'tobogan' |
| 19:14.31 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:14.41 | dlynes | erm toboggan |
| 19:17.25 | nephfl | ok i restarted asterisk and everything is fine...now how do i find the problem that made it hang? |
| 19:18.34 | [TK]D-Fender | nephfl: "load res_fluxcapacitor.so" & "load res_psychic.so" |
| 19:19.09 | nephfl | im checking through logs now.. |
| 19:19.55 | holiday_42 | Don't forget to upgrade the power supply to 3.7gigawatts |
| 19:22.31 | [TK]D-Fender | holiday_42: 2.1! You don't want him to fry! |
| 19:24.20 | lee_is_me | [TK]D-Fender: Sorry, was on the phone. I tried the same operation with both a GrandStream and XLight and it worked fine. The problem is only when the polycom is called |
| 19:24.57 | nephfl | looks like a zap channel was freaking out |
| 19:24.59 | [TK]D-Fender | lee_is_me: I seriously not asking again... |
| 19:25.02 | sandorp | voicemails left using analog line->fxo device are really quiet; how can I turn up the volume? |
| 19:25.08 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:25.21 | [TK]D-Fender | sandorp: increase your gains in zapata |
| 19:26.22 | sandorp | [TK]D-Fender: I did that already but it doesn't seem to make a difference for the voicemail but I can definitely hear the remote caller better |
| 19:26.49 | lee_is_me | [TK]D-Fender: I appreciate your help. I was just wondering if it was still necessary to paste all that data considering that the problem only happens on the polycom. But I will paste it now... |
| 19:27.57 | nephfl | Anybody know this error? WARNING[19633] chan_zap.c: We're Zap/8-1, not "User Name" <221> |
| 19:28.06 | jlewis | anyone around who can help clarify a paid support issue with digium? |
| 19:28.07 | sandorp | grr ... why does my zapata.conf keep reverting to rxgain=0, txgain=0 ? |
| 19:28.19 | *** join/#asterisk seva (i=seva@sevatech.com) |
| 19:28.26 | seva | how do i figure out what codec a client is using? |
| 19:28.34 | [TK]D-Fender | sandorp: because you are running some stupid that is regenerating your configs, why else do you think? |
| 19:28.48 | [TK]D-Fender | seva : "sip show channels" <- for example |
| 19:29.02 | seva | thanks |
| 19:29.18 | [TK]D-Fender | jlewis: Try calling them |
| 19:30.38 | jlewis | [TK]D-Fender: been getting the run around...though I think we just got an answer |
| 19:31.04 | lee_is_me | [TK]D-Fender: Here is the pastebin: http://www.pastebin.ca/584445 |
| 19:33.08 | jlewis | on with client now...they just finally got a call back (been playing tag with digium support for about a month). They've encountered an odd bug that leaves zap channels locked up. Digium support says its a sip bug, but that they won't fix/support it...that they should buy asterisk business edition, and either it'll "just work" or then digium will try fixing the bug. |
| 19:33.27 | *** join/#asterisk sosoriosv (n=salvador@200.31.160.4) |
| 19:34.24 | [TK]D-Fender | lee_is_me: STILL missing something...... |
| 19:34.56 | lee_is_me | [TK]D-Fender: What is that? |
| 19:35.16 | [TK]D-Fender | jlewis: Expecting * support for a SIP bug without running the Business Edition isn't going to get you far. |
| 19:35.25 | [TK]D-Fender | lee_is_me: CLI OUTPUT. |
| 19:37.07 | jlewis | client was perfectly happy to pay for support...actually already send digium $995 |
| 19:37.24 | jlewis | which is coincidentally about what ABE costs :) |
| 19:37.30 | Mercestes | Digium requires you to have asterisk business edition to fix a sip bug? |
| 19:37.30 | Hmmhesays | could have gotten the business edition for that |
| 19:37.42 | Mercestes | Retarded. |
| 19:37.49 | [TK]D-Fender | Could have jsut UPGRADED |
| 19:37.50 | Hmmhesays | why? |
| 19:37.56 | Hmmhesays | thats business |
| 19:37.58 | jlewis | well, as they haven't gotten anything for that $995, I expect they're either getting it back or getting ABE for it |
| 19:38.01 | Mercestes | It's retarded. |
| 19:38.04 | lee_is_me | [TK]D-Fender: http://www.pastebin.ca/584455 not much CLI output I'm afraid. |
| 19:38.17 | Hmmhesays | Digium is a business |
| 19:38.21 | Hmmhesays | it is there job to make money |
| 19:38.30 | Mercestes | Well, if they screw ppl to do it they won't be a business any longer. :P |
| 19:38.32 | [TK]D-Fender | lee_is_me: Funny... no answer there... |
| 19:38.44 | jlewis | reminds me of SCO selling driver bug fixes when you report a bug that crashes the system to them |
| 19:38.49 | anonymouz666 | anyone in here use freebsd with asterisk? |
| 19:39.32 | lee_is_me | [TK]D-Fender: The polycom rings a second or two and the options ANSWER REJECT FORWARD are displayed on the screen, but the phone will not pick up |
| 19:39.33 | sci_05 | anyone know what would cause a call to fad in and out? |
| 19:39.51 | kimo_sabe | sci_05: bad automatic gain control? |
| 19:40.13 | [TK]D-Fender | lee_is_me: Sounds like they set their phone up to reject your call. |
| 19:40.16 | lee_is_me | that is why it went unanswered. Grandstream/XLight/Cellphone will all successful] |
| 19:40.25 | sci_05 | kimo_sabe would that be on the t1 card setting, in the zaptel.conf? |
| 19:40.49 | lee_is_me | I set the phone up, its on my desk. I have both lines configured |
| 19:41.31 | lee_is_me | but have not explicitly made any changes to reject calls. |
| 19:42.31 | Mercestes | maybe your subconciously rejectin gyourself. |
| 19:43.08 | jlewis | is business edition closer to / based on 1.2, 1.4 or just totally its own thing? |
| 19:43.15 | file | 1.2 |
| 19:43.23 | *** part/#asterisk sosoriosv (n=salvador@200.31.160.4) |
| 19:43.25 | VJFROMGT | <PROTECTED> |
| 19:43.58 | Jingles | GotoIfTime |
| 19:45.48 | Un1x_Laptop | Someone please, help i was wondering how come meetme is not asking for PIN's |
| 19:46.02 | bkruse | the gui has a time routing tab now also |
| 19:46.06 | jlewis | or if you want to be really cute, include a file modified via cron |
| 19:46.45 | rob0 | Two eyebrows to me? I fold. :( |
| 19:47.53 | *** join/#asterisk lorinc (n=ang@pool-4172.adsl.interware.hu) |
| 19:49.03 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 19:50.43 | CoffeeIV_ | from perl I am using $AGI->set_priority(10) I would prefer to put a label instead of a number for the priority, will that work (provided I have the label in the dial plan ) ? |
| 19:51.33 | [TK]D-Fender | CoffeeIV : No, the extensions parser interpret all of those on LOAD. |
| 19:51.52 | [TK]D-Fender | CoffeeIV : Same with hints. Can't be dynamic |
| 19:52.10 | CoffeeIV_ | ok, thanks -- I'll find a different way to do what I want to do |
| 19:52.10 | ccesario | Hi, is possible pass one variable to other context using Goto ? |
| 19:52.35 | sandorp | maybe set and ENV variable |
| 19:52.36 | [TK]D-Fender | ccesario: Nothing to PASS. the variable still exists, you never left your CHANNEL |
| 19:53.36 | ccesario | [TK]D-Fender, hmmmm after hangup the variable is set to null ? |
| 19:55.21 | [TK]D-Fender | ccesario: You don't seem to understand the SCOPE of a variable... what is there to DO after a hangup? |
| 19:55.40 | mcab | Can we get captures of the issues that they are seeing |
| 19:55.45 | mcab | gah |
| 19:55.50 | mcab | sorry, wrong window :-p |
| 19:56.33 | dijungal | ahhh the reason for my no audio is rtp.c: RTP transmission error of packet |
| 19:56.46 | dijungal | how can that be fixed |
| 19:57.49 | *** join/#asterisk dasuberdavid (i=root@nat/digium/x-1d8858526200fbd1) |
| 20:00.14 | sci_05 | anyone know where I can find somekind of automatic gain control, would that be on the iax.conf zapata.conf?? |
| 20:01.06 | [TK]D-Fender | sci_05: no such thing in *. Sounds like your card is flakey crap. Buy a better one. |
| 20:02.05 | sci_05 | its a te207p from digium.....what would be better then that? |
| 20:02.37 | _VoiceMeUp_COM | [TK]D-Fender reobting the box fixed the pri |
| 20:02.40 | _VoiceMeUp_COM | aint that weird ? |
| 20:02.50 | _VoiceMeUp_COM | also dtmf was not ebing passed while it did that |
| 20:02.59 | [TK]D-Fender | _VoiceMeUp_COM: Not really... seen before... would ahve figured you'd have tried that already :) |
| 20:03.13 | _VoiceMeUp_COM | nope |
| 20:03.21 | _VoiceMeUp_COM | had to push all calls out of it |
| 20:03.27 | _VoiceMeUp_COM | since theyre where call son group1 |
| 20:03.29 | _VoiceMeUp_COM | and.. well |
| 20:03.30 | [TK]D-Fender | _VoiceMeUp_COM: Sometimes it queues up crap and you need to reboot or fully down the card and flush beffers to clear up. |
| 20:03.40 | _VoiceMeUp_COM | waiting on those.. the pri just dided |
| 20:03.46 | exoxe | is there any way to beam a beautiful girl to me by pressing 4 on a phone? |
| 20:03.47 | [TK]D-Fender | sci_05: Sangoma A102d |
| 20:03.58 | _VoiceMeUp_COM | exoxe tried |
| 20:04.04 | exoxe | no luck? |
| 20:04.09 | _VoiceMeUp_COM | hehe maybe ;) |
| 20:04.25 | [TK]D-Fender | exoxe: Yes, but packet overhead adds 50lbs & 20 years. |
| 20:04.40 | _VoiceMeUp_COM | lol |
| 20:04.46 | _VoiceMeUp_COM | omg i just fell off hte chair |
| 20:04.53 | [TK]D-Fender | <- teh shiznit y0 |
| 20:05.02 | _VoiceMeUp_COM | transcoding will get her as a thai treansvestite too |
| 20:05.09 | [TK]D-Fender | EW! |
| 20:05.10 | exoxe | hmm |
| 20:05.15 | exoxe | yeah, i definitely don't want that |
| 20:05.17 | _VoiceMeUp_COM | yeah.. so better stay ulaw ;) |
| 20:05.22 | _VoiceMeUp_COM | lol |
| 20:05.26 | *** join/#asterisk eatmypiano_ (n=eatmypia@host86-132-181-229.range86-132.btcentralplus.com) |
| 20:05.27 | holiday_42 | lol |
| 20:05.45 | _VoiceMeUp_COM | almost went to thailand once to live there for 1 year.. to learn thai and party |
| 20:05.56 | _VoiceMeUp_COM | then saw a show where 9/10 where all males..shemales |
| 20:05.56 | tzanger | [TK]D-Fender: hahaha |
| 20:06.01 | exoxe | and to have the biggest dick in the country? |
| 20:06.11 | _VoiceMeUp_COM | and found them like attractive .. befor ei knew ;) |
| 20:06.29 | _VoiceMeUp_COM | so i said. never will i got there.. coz if it happened ill end up in jail for murder 1 and spend the rest of my life there |
| 20:06.45 | _VoiceMeUp_COM | scary place |
| 20:06.53 | _VoiceMeUp_COM | prefer quebec,canada , ;) |
| 20:07.08 | _VoiceMeUp_COM | only thing here is they all think theyr princesses like paris style |
| 20:07.26 | [TK]D-Fender | seulement les Montrealaise ;) |
| 20:08.42 | irule | where can I getr asterisk now to install from source? |
| 20:08.50 | seva | what are the /l |
| 20:08.59 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 20:09.03 | seva | er |
| 20:09.04 | *** part/#asterisk seva (i=seva@sevatech.com) |
| 20:09.23 | holiday_42 | irule:asterisk.org |
| 20:09.42 | irule | I can only find iso images in there |
| 20:10.43 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 20:10.48 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 20:10.53 | Jingles | wrong clickie. heh |
| 20:11.18 | holiday_42 | irule: asterisk.org, not asterisknow.org |
| 20:14.10 | irule | all links in http asterisk.org send me to ftp.digium.com but ateriskNOW is not there |
| 20:14.12 | irule | it is it? |
| 20:14.27 | irule | or is it? |
| 20:15.56 | ai-a | irule: asteriskNOW, if you read about it, is an ISO image that makes Astrisk easy (for dumbies) that dont know about source compiling or command line. |
| 20:16.05 | ai-a | so you would not expect to see AstNow source code. |
| 20:16.10 | ai-a | Ast Source IS asterisk |
| 20:16.18 | ai-a | hence,, asterisk.org HAS source code. |
| 20:16.44 | ai-a | the source code is availabile here -> http://www.asterisk.org/developers/get-source |
| 20:16.52 | ai-a | took me all 1 minute to find it. |
| 20:17.18 | irule | ai-a what if I want asterisk-NOW GUI installed on my debian-etch dith my own bells and whistles? |
| 20:17.29 | ai-a | so,, download the GUI then. |
| 20:18.05 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 20:18.10 | irule | ai-a where? |
| 20:18.10 | diclophis-work | whats an uncompressed codec? |
| 20:18.17 | ai-a | iratsu: are you dumb ? |
| 20:18.23 | ai-a | the source code is availabile here -> http://www.asterisk.org/developers/get-source |
| 20:18.30 | irule | even the GUI? |
| 20:18.36 | ai-a | GO read it.. |
| 20:18.36 | ai-a | YES |
| 20:18.46 | irule | oh cool thanks lol |
| 20:18.56 | ai-a | if you take 10 minutes to click and read instead of begging, you might be amazed. |
| 20:19.26 | ai-a | diclophis-work: do you mean "which cocec's are uncompressed", or "what are uncompressed codecs" ? |
| 20:19.35 | irule | diclophis-work WAV format is not compressed, so it takes a lot of space, in * it is know by alaw, ulaw or something like that |
| 20:19.37 | diclophis-work | which ones are not compressed |
| 20:19.48 | diclophis-work | mmm the wiki says ulaw is compressed... |
| 20:19.48 | ai-a | ok, wav and raw. |
| 20:20.03 | ai-a | ulaw is encoded. |
| 20:20.15 | diclophis-work | hmm which works over sip? |
| 20:20.20 | diclophis-work | raw doesnt work over sip does it? |
| 20:20.52 | ai-a | diclophis-work: rtfm |
| 20:21.01 | irule | diclophis-work http://www.asterisk.org/glossary |
| 20:21.06 | ai-a | do any of you READ before asking ? |
| 20:21.18 | ai-a | where do i find google.com ? |
| 20:21.20 | irule | G.711 An uncompressed codec that samples a 64kbps channel at 8 bits per sample using pulse code modulation. The Two varients of G.711 are known formally as uLaw and aLaw. See also: CODEC, G.729, GSM. |
| 20:21.22 | ai-a | how do i enter search on google ? |
| 20:21.34 | diclophis-work | well, i fouind an article about setting the dtmfmode for a sip channel, and it said if you use inband with ulaw, it might not work |
| 20:21.36 | ai-a | i got 5 brain cells, can you help me ? |
| 20:21.37 | diclophis-work | cause of compression |
| 20:21.43 | diclophis-work | so, maybe that article is wrongt? |
| 20:21.44 | irule | ai-a open www.google.com :) |
| 20:21.59 | ai-a | diclophis-work: if that is true, then dont use it . |
| 20:22.11 | irule | ai-a see? I am being helpfull! not BSing :P lol |
| 20:22.13 | ai-a | diclophis-work: who wrote the article? what url ? |
| 20:22.33 | diclophis-work | arg |
| 20:22.35 | diclophis-work | i read it wrong |
| 20:22.35 | diclophis-work | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode |
| 20:22.42 | diclophis-work | it actually says if its NOT ulaw |
| 20:22.44 | diclophis-work | damn |
| 20:22.48 | ai-a | right. |
| 20:22.50 | diclophis-work | someone changed it quick |
| 20:22.52 | ai-a | i cant cope,, night |
| 20:23.16 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 20:23.26 | irule | funaral for ai-a lol |
| 20:24.11 | _VoiceMeUp_COM | any possiblity to see a sip prune realtime peer XXX but for sip rleoad ? |
| 20:24.35 | *** join/#asterisk _DAW_ (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:25.08 | irule | we should make a friendsly channel for stupid AND advanced questions where ashhls that say RTFM get kicked out lol |
| 20:25.36 | Qwell[] | nobody will ever get help there |
| 20:25.54 | Qwell[] | If nobody is there to teach people to RTFM, who's gonna RTFM to answer questions? |
| 20:26.35 | [TK]D-Fender | “I would never want to belong to a club that would have me as a member,” - Groucho Marx |
| 20:27.31 | [TK]D-Fender | To everyone who won't do any basic reading on the 2 biggest resources we HAND TO THEM FREE : |
| 20:27.34 | [TK]D-Fender | ~osmosis |
| 20:27.34 | jbot | methinks osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 20:27.39 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 20:27.46 | holiday_42 | lol |
| 20:27.53 | irule | well at least say go to TFM page 345 where it has your answer, not just RTFM and by the way feel the bad service, no wonder morons stay in win where they can call dell and get put on hold for an hour, just to be taken step by step |
| 20:28.24 | [TK]D-Fender | irule: PDFs are SEARCHABLE you know... |
| 20:29.24 | [TK]D-Fender | irule: People don't try because they are lazy quitters. |
| 20:29.27 | irule | so is the web! I am talking about the attitude, of course it is in TFM! but sometimes a little clue is more than enough |
| 20:30.29 | [TK]D-Fender | Qwell[]: Remember that guy I SENT to go DL Trixbox? Di EVERYONE a favour. He gets a solution that fits the effort and ends he desires, and it got US rid of HIM. |
| 20:30.55 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 20:31.06 | [TK]D-Fender | Ok, long-weekend time... I'm off, BBIAB |
| 20:31.12 | irule | I never quite :P ..and yes, am sitting in a big lazy looking sofa in a lazy position lol ...did I tell you I am faxing with iaxmodem, hylafax and *? I had to do that myself with 0 clear documentation |
| 20:31.16 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-175-183.buff.east.verizon.net) |
| 20:31.39 | irule | documentation reads like stereo instructions |
| 20:32.01 | SuPrSluG | is there a command to find out what version of zaptel is loaded? |
| 20:32.06 | irule | and its even worse when your native lang is not en :s |
| 20:32.28 | irule | ztcfg -v |
| 20:32.37 | irule | SuPrSluG ztcfg -v |
| 20:32.42 | irule | as root |
| 20:33.02 | SuPrSluG | no that just show that they loaded ok |
| 20:33.02 | rob0 | # modinfo zaptel |
| 20:33.33 | irule | see? no ugly RTFM, maybe his curiosity will send him searching for ztcfg that came from me, not from 3 days worth of RTFM |
| 20:33.40 | dijungal | ahhhhggghhh!! my audio is not going through on my h.323 channel :( |
| 20:34.24 | irule | part of the response I see in ztcfg -v is a line that says "Zaptel Version: 1.4.2.1" so, WTF are you talking about? |
| 20:34.26 | SuPrSluG | not listing the version |
| 20:34.54 | Mercestes | irule, Yea, being online got me in trouble, IRL, because I thought it was normal to be a hairy, flaming ass to everyone. |
| 20:35.09 | irule | what do you have in /usr/src ? maybe if its been compiled, the sources are still there with the version names |
| 20:35.19 | irule | lol |
| 20:35.19 | Mercestes | irule: then I realized, most of the Ppl in IRC are all "yessir mr. boss, let me kiss your butt some more please" IRL.... |
| 20:35.24 | Mercestes | irule: or...they are unemployed. |
| 20:36.05 | SuPrSluG | apparently not w/ the 1.2.X versions |
| 20:36.35 | SuPrSluG | though 1.4 shows that info |
| 20:37.23 | irule | SuPrSluG ohh, :s then I dont know, I am a late adopter, only been on 1.4 |
| 20:37.29 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 20:37.36 | irule | late I mean |
| 20:37.39 | `Sean | Hey, anyone know why i keep getting this |
| 20:37.39 | `Sean | <PROTECTED> |
| 20:37.47 | `Sean | is there a way to tell asterisk to stop avoiding it.. |
| 20:37.57 | *** join/#asterisk Sweeper (i=sweeper@scriggleit.com) |
| 20:38.09 | Sweeper | why does billing have to be such a PITA? |
| 20:38.40 | ManxPower | Sweeper: It's not a PITA for me at all. |
| 20:38.48 | ManxPower | But that's because I don't bill for calls. |
| 20:38.56 | Sweeper | :P |
| 20:39.07 | ManxPower | I let the telco bill for calls. |
| 20:39.07 | `Sean | ManxPower any insight into my problem |
| 20:39.24 | ManxPower | `Sean: if I had chances are I would have said something. |
| 20:39.33 | `Sean | Cool |
| 20:39.40 | ManxPower | you didn't even bother to mention what asterisk version |
| 20:40.04 | `Sean | 1.2.13 |
| 20:44.05 | irule | Mercestes so are you unemplyed, a hairy, flaming ass or what? |
| 20:44.11 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:44.16 | irule | lol\ |
| 20:44.16 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:44.37 | Mercestes | irule, Ask Fender |
| 20:45.57 | SuPrSluG | anyone figure out zaptel 1.4.3 compile error http://pastebin.ca/584577 |
| 20:45.59 | waKKu | someone there uses pickupgroups with asterisk 1.2 ??? |
| 20:46.05 | Mercestes | he'll be glad to tell you. |
| 20:46.10 | waKKu | i really had tried all settings |
| 20:46.23 | Mercestes | waKKu, pastebin your code. |
| 20:46.29 | waKKu | ok.. |
| 20:47.19 | ManxPower | waKKu: what device are you trying to use call pickup from? |
| 20:48.24 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:48.37 | waKKu | Mercestes ManxPower http://pastebin.ca/584588 |
| 20:48.38 | waKKu | iax |
| 20:49.34 | ManxPower | waKKu: chan_iax2 does NOT support pickup groups |
| 20:49.51 | waKKu | really ?... damn.. |
| 20:50.04 | waKKu | i loose a big time |
| 20:50.27 | ManxPower | chan_sip and chan_zap do. Not chan_iax2. I don't know about chan_sccp, chan_skinny, chan_mgcp, or an of the H323 driver |
| 20:50.47 | waKKu | ManxPower http://voip-info.linuxsys.com/wiki/index75b0c79a6a3ecf7dbe65d728324fa861.html?page=Asterisk%20callgroups%20and%20pickupgroups |
| 20:52.37 | waKKu | is wiki wrong ? |
| 20:53.22 | Sweeper | pickup groups are lame anyways |
| 20:54.15 | waKKu | Sweeper why ? seems a good feature to me |
| 20:56.39 | ManxPower | waKKu: half the info on the wiki is wrong |
| 20:56.51 | Trevor_b | SuPrSluG: we turned of menuselect and it seemd to work. |
| 20:56.52 | Sweeper | also |
| 20:57.12 | Sweeper | it says you can DEFINE them |
| 20:57.20 | SuPrSluG | how do you turn it off |
| 20:57.25 | Sweeper | doesn't say it works :) |
| 20:57.46 | Trevor_b | SuPrSluG: although I didnt even check your pastebin, we had issues and disabled that. I am not actaully sure, Kevan did it to build our RPM today, hes gone right now but will be back within an hour |
| 20:58.19 | SuPrSluG | ill look into that, thanks |
| 20:59.41 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-14a3984638807b59) |
| 20:59.41 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 21:07.10 | irule | asterisk now gui source code is not here: :s http://www.asterisk.org/developers/get-source |
| 21:10.26 | Daejeo1 | my friend unable to register on my box. please have a look on my network architecture http://daejeo1.sytes.net/images/asterisk.jpg |
| 21:10.38 | *** join/#asterisk Tako-san (n=Tako-san@24.68.129.29) |
| 21:13.46 | kiscokid | Daejeo1: What would make the phone route to the Asterisk box accross the bridge? |
| 21:14.28 | diclophis-work | what would cause a sip peer to respond with a 500 error code |
| 21:15.41 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 21:17.52 | Daejeo1 | kiscokid: don't know i just shared the internet |
| 21:18.11 | Daejeo1 | internet is working |
| 21:19.23 | Daejeo1 | kiscokid: how can i route then? i do not have router. |
| 21:19.33 | kiscokid | Daejeo1: can you ping anything on the internet from the Asterisk box? |
| 21:19.48 | Daejeo1 | of course |
| 21:19.55 | Daejeo1 | i can call out |
| 21:21.42 | kiscokid | Daejeo1: I think you need a router to create NAT and then open port 5060 on the router to Asterisk box |
| 21:22.23 | Daejeo1 | I do not have a router |
| 21:22.35 | Daejeo1 | i have just lan cards |
| 21:22.36 | kiscokid | buy one for $30 |
| 21:22.40 | Daejeo1 | no |
| 21:23.17 | Daejeo1 | there is an other way |
| 21:23.25 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 21:23.43 | *** join/#asterisk Pie-rate (n=jschall@cust-69-19-214-183.static.o1.com) |
| 21:23.47 | kiscokid | how? |
| 21:24.38 | Daejeo1 | I can directly connect my asterisk box to ISP modem. then linux internet sharing for my private network |
| 21:25.12 | kiscokid | I guess that would work |
| 21:25.41 | Daejeo1 | I will use two lan cards in asterisk box |
| 21:26.03 | Daejeo1 | but I never shared internet on linx |
| 21:26.08 | Tako-san | daejeo1: before you do that I would make sure you have turned off all unneeded services on the asterisk box. |
| 21:26.18 | Tako-san | otherwise you are asking to be hacked |
| 21:26.36 | Tako-san | unless you can properly configure your firewall |
| 21:26.44 | kiscokid | Better to have a NAT router with most services turned off |
| 21:27.13 | *** join/#asterisk PioneerVM2 (n=IceChat7@24-151-103-018.dhcp.nwtn.ct.charter.com) |
| 21:27.41 | PioneerVM2 | hey anyone know a decent VOIP SIP provider that is easy to setup and has cheap DIDs. |
| 21:28.00 | PioneerVM2 | I setup AGN which is < $3 each but they dont have any CT DIDs :( |
| 21:28.06 | PioneerVM2 | and everyone else is like $10 each |
| 21:29.00 | Daejeo1 | kiscokid: can you lead me . i want to share the internet on linux |
| 21:29.54 | _VoiceMeUp_COM | anyreason asterisk not escaping the quoest in a callerid name from a mediatrix ? |
| 21:30.05 | _VoiceMeUp_COM | i would assume that would be the firt thing to do no ? |
| 21:30.52 | PioneerVM2 | anyone suggest any VOIP providers other than teliax, agnvoip and voicepulse connect |
| 21:32.05 | Daejeo1 | kiscokid: ? |
| 21:33.24 | Daejeo1 | PioneerVM2: VTWHITE |
| 21:34.19 | dlynes | PioneerVM2: Try www.voipproviderslist.com |
| 21:34.28 | Tako-san | Daejeo1: try googling for some how-to's for "centos internet sharing" |
| 21:35.13 | Daejeo1 | BSD_Tech: my friend unable to register on my box. please have a look on my network architecture http://daejeo1.sytes.net/images/asterisk.jpg |
| 21:35.20 | Tako-san | daejeo1: here is one example "http://www.howtoforge.com/forums/showthread.php?t=672" |
| 21:35.39 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 21:35.42 | BSD_Tech | I will in a bit on a call |
| 21:36.36 | BSD_Tech | I said I would look when I get off this call |
| 21:37.16 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:37.21 | dlynes | Daejeo1: install a softphone on your windows box, and make sure it can talk to the asterisk box, first |
| 21:37.40 | dlynes | Daejeo1: If that's working, then it's a nat issue |
| 21:37.52 | Daejeo1 | dlynes: it is working |
| 21:38.31 | Tako-san | daejeo1: using your Windows box as a router/gateway is highly not recommended |
| 21:38.37 | Daejeo1 | even though i can call out thru my voip provider |
| 21:38.45 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 21:38.52 | dlynes | Daejeo1: Is the remote softphone using iax2 or sip? |
| 21:39.07 | Daejeo1 | remote sip |
| 21:39.28 | dlynes | Daejeo1: Are you doing any port forwarding on the windows box? |
| 21:39.42 | Daejeo1 | I added the port |
| 21:40.02 | dlynes | Daejeo1: 'the port'? you mean 'the ports'? |
| 21:40.09 | kiscokid | Daejeo1: the softphone on the windows box works because you are using the 192 address |
| 21:40.29 | Daejeo1 | kiscokid :yes |
| 21:40.47 | Daejeo1 | it does not work when i use 124 |
| 21:40.48 | kiscokid | but your friend on the internet can't use this address |
| 21:40.52 | dlynes | Daejeo1: You need to forward more than just 5060; you also need to forward rtp |
| 21:41.04 | dlynes | kiscokid: check his diagram...his friend is using 124, not 192 |
| 21:41.12 | *** part/#asterisk Pie-rate (n=jschall@cust-69-19-214-183.static.o1.com) |
| 21:41.23 | _VoiceMeUp_COM | wow |
| 21:41.25 | _VoiceMeUp_COM | anyone know |
| 21:41.43 | _VoiceMeUp_COM | un 22 17:41:04 ERROR[13717]: chan_sip.c:11507 sipsock_read: We could NOT get the channel lock for sip/blah -0942ca38 - Call ID 1b438f0405b5d1636404186229@someip |
| 21:42.47 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 21:43.02 | Trevor_b | How do you set pri_net and pri_cpe on digium wildcards? |
| 21:43.23 | Daejeo1 | dlynes: how can I do rtp forwarding on windowsxp |
| 21:45.18 | kiscokid | Daejeo1: you'd be much better off putting the Asterisk box directly on the internet if you're not going to use a router |
| 21:46.34 | _VoiceMeUp_COM | *** glibc detected *** free(): invalid pointer: 0x08049d96 *** |
| 21:46.35 | _VoiceMeUp_COM | wow |
| 21:46.39 | _VoiceMeUp_COM | ok im really pissed now |
| 21:47.06 | Daejeo1 | kiscokid: I am going to that |
| 21:51.17 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:53.53 | PioneerVM2 | dlynes, that list is too huge, over 600! :) |
| 21:54.04 | PioneerVM2 | daejeo1 -- i looked at vtwhite but there is no price info there |
| 21:55.03 | PioneerVM2 | can anyone throw out names of voip providers for asterisk u have been using and are happy with |
| 21:55.28 | kiscokid | voicepulse |
| 21:55.34 | alrs | PioneerVM2: I've been using Gafachi for a year without problems. |
| 21:55.45 | alrs | PioneerVM2: But for myself, not for a business. |
| 21:55.57 | Daejeo1 | PioneerVM2: http://www.vtwhite.com/sip_voip_origination.html |
| 21:56.00 | brad_mssw | _VoiceMeUp_COM: what version of asterisk ? |
| 21:57.14 | NightMonkey | Howdy. I'm just getting started with Asterisk, and have a HandyTone 488 set up with an FXO (PSTN) and FXS (analog touch tone phone). However, I'd like the 488 not to try to ring through to the attache FXS. |
| 21:57.23 | PioneerVM2 | dae -- are these guys any good? how long they been around |
| 21:57.26 | *** join/#asterisk Andy_G (i=pdm@pdm.sh) |
| 21:57.30 | PioneerVM2 | 75c dids sounds shady :) |
| 21:57.42 | PioneerVM2 | alrs -- how are their rates |
| 21:57.53 | PioneerVM2 | im looking to get a bunch of DIDs for incoming for my home voice mail |
| 21:58.10 | PioneerVM2 | im trying to reroute all my voice mail for biz, cell and home and for me and wife to one set of mailboxes and need multiple DIDs to do it properly |
| 21:58.13 | NightMonkey | I'd only like the FXO to ring through to Asterisk's console. Anyone done this? |
| 21:58.59 | NightMonkey | I'm thinking this might be a limitation of the 488, tho. |
| 22:00.00 | Tako-san | nightmonkey: can you clarify what you are trying to do? you want incoming calls to only go to the 488 and not the fxs port? or? |
| 22:03.02 | *** join/#asterisk rodger_nc (n=chatzill@cpe-076-182-073-016.nc.res.rr.com) |
| 22:03.03 | dlynes | PioneerVM2: Well, you were wondering who else there was |
| 22:03.27 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 22:03.38 | NightMonkey | Tako-san: Sure. The HandyTone-488 has two ports: FXO and FXS. The FXO is connected to my PSTN phone line. The FXS is connected to an analog old-school POTS phone. However, incoming calls to the FXO first ring the FXS on the 488, and then, after a set number of rings, are routed to Asterisk. |
| 22:03.47 | PioneerVM2 | anyone know anything about vonage sip trunking |
| 22:03.51 | PioneerVM2 | i heard they do it now but cant find it |
| 22:03.54 | NightMonkey | Tako-san: I want this intermediate step eliminated. |
| 22:04.18 | NightMonkey | (FYI, routing is over SIP, if that matters) |
| 22:05.23 | NightMonkey | Some reading I'm doing seems to indicate that this might just be a "feature" of the HT-488, but I was hoping to see if others found another, less sad answer. ;) |
| 22:05.32 | rodger_nc | Subject: Cell phone to replace land line. I want to get rid of my land based line. I want to use a modem as the entrance to the PBX. I would therfore also like to hook my modem into my POTS lines in the house and use regular phones in the house through the cell phone connected to the computer. Does Asterisk allow this?? |
| 22:05.59 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 22:06.41 | rodger_nc | Subject: Cell phone to replace land line. I want to get rid of my land based line. I want to use a modem/ CELL PHONE as the entrance to the PBX. I would therfore also like to hook my modem into my POTS lines in the house and use regular phones in the house through the cell phone connected to the computer. Does Asterisk allow this?? |
| 22:06.58 | NightMonkey | rodger_nc: Don't double post, plz. |
| 22:07.41 | rodger_nc | sorry made a mistake in the first post |
| 22:07.49 | NightMonkey | np |
| 22:08.50 | *** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca) |
| 22:09.44 | NOT_guru | Rodger_nc lookinto GSM gateways |
| 22:09.52 | prashant_jois | Hi everyone, I'm having trouble loading zaptel, I get the following error on modprobe: "Unable to open master device '/dev/zap/ctl'". I have made sure that /etc/udev/rules.d/zaptel.rules is set up right. Any suggestions? |
| 22:09.55 | NOT_guru | they should be able to do what you need to interface into asterisk |
| 22:13.26 | rodger_nc | I have not found a way to connect my cell phone to my modem so that the modem will answer - nor the command codes to access the cell phone from the computer |
| 22:14.55 | rodger_nc | after that I would need to be able to activate the cell phone from a normal phone connected to the other port in the modem |
| 22:15.31 | *** join/#asterisk el_4_jinete (n=kvirc@mail.pulxar.com.co) |
| 22:15.36 | kiscokid | rodger_nc: I think you need something like this: http://www.phonelabs.com/prd05.asp |
| 22:16.13 | *** join/#asterisk tekor (n=will@adsl-147-85-156.bhm.bellsouth.net) |
| 22:16.24 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 22:17.30 | rodger_nc | YES! Dock and Talk is the ticket |
| 22:21.12 | Tako-san | anyone have some ideas as to why ZAP lines would be dropping randomly? |
| 22:23.11 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:24.52 | _VoiceMeUp_COM | anyone have the mediatrix ? |
| 22:26.12 | bkruse | Hey, when you set the fromuser field for sip, what populates the name? |
| 22:26.34 | _VoiceMeUp_COM | fromuser |
| 22:26.38 | _VoiceMeUp_COM | oh |
| 22:26.47 | bkruse | Like if I have fromuser: woot@fakedomain |
| 22:26.48 | bkruse | I get From: "asterisk" <sip:woot@fakedomain@localip> |
| 22:26.48 | _VoiceMeUp_COM | fromuser replaces tehe user name in from_uri |
| 22:26.55 | _VoiceMeUp_COM | yeah |
| 22:26.58 | _VoiceMeUp_COM | fromuser=bob |
| 22:27.00 | *** part/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 22:27.10 | _VoiceMeUp_COM | <PROTECTED> |
| 22:27.19 | _VoiceMeUp_COM | name wil be the callerid name |
| 22:27.27 | _VoiceMeUp_COM | will raplce asterisk |
| 22:27.28 | bkruse | ahh |
| 22:27.35 | bkruse | Thank you@ |
| 22:27.39 | _VoiceMeUp_COM | but some switches use fromuser as the calleird |
| 22:27.40 | bkruse | can you not set the @? |
| 22:27.44 | bkruse | eg |
| 22:27.48 | _VoiceMeUp_COM | since the industry std is using the did number as the from |
| 22:27.49 | bkruse | fromuser=blah@domain |
| 22:27.51 | _VoiceMeUp_COM | not a bogusname |
| 22:27.53 | _VoiceMeUp_COM | so... |
| 22:28.21 | _VoiceMeUp_COM | while industry std is From: "BOB HOPE" <5554443333> <sip:5554443333@domain> |
| 22:28.28 | _VoiceMeUp_COM | wehmm no |
| 22:28.29 | _VoiceMeUp_COM | i mean |
| 22:28.33 | _VoiceMeUp_COM | anyway you get the idea |
| 22:28.46 | file | O.o |
| 22:29.25 | bkruse | haha, thanks :] |
| 22:29.44 | *** part/#asterisk AvoidingSmelling (i=mishehu@cshells.shavedgoats.net) |
| 22:29.55 | _VoiceMeUp_COM | man mediatrix is the worst in therms of gui out there |
| 22:29.58 | _VoiceMeUp_COM | 1102 at least |
| 22:30.22 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 22:33.32 | _VoiceMeUp_COM | http://www.voip-info.org/wiki/view/Mediatrix+settings+for+Voicemeup |
| 22:33.37 | _VoiceMeUp_COM | seems my own guide aint good |
| 22:34.11 | lee_is_me | Are channels re-created with a Redirect() AMI function? I've tried settings some variable using SetVar() and they don't seem to persist.... |
| 22:34.30 | _VoiceMeUp_COM | hmm |
| 22:34.32 | lee_is_me | ...after the redirect. |
| 22:34.38 | _VoiceMeUp_COM | can you __VAR ? |
| 22:34.59 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 22:35.28 | lee_is_me | I haven't used that. isn't it a type of pseudo inheritance? |
| 22:35.56 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:36.37 | *** join/#asterisk Jabroni (n=asskick@red-corp-201.143.126.173.telnor.net) |
| 22:37.14 | lee_is_me | _VoiceMeUp_COM: ooooh. That's nice. Just looked it up again. Read it before but didn't understand what good it was for until now |
| 22:37.35 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:38.07 | dlynes | Anyone get the following error, 'Unknown symbol request_module', when trying to modprobe zaptel.ko from 1.4.3 in a 2.6.19.3 kernel? |
| 22:38.51 | lee_is_me | _VoiceMeUp_COM: thanks. |
| 22:39.49 | _VoiceMeUp_COM | np |
| 22:39.52 | _VoiceMeUp_COM | mediatrix sucks |
| 22:40.02 | _VoiceMeUp_COM | guys who make it are like 2 hours from my place |
| 22:40.15 | _VoiceMeUp_COM | ill got visit them and teach them a thing or 2 about gui's |
| 22:42.45 | *** join/#asterisk AlexGC (n=does@189.157.0.132) |
| 22:44.02 | *** join/#asterisk mindCrime (n=chatzill@65.190.188.124) |
| 22:44.20 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:46.03 | _VoiceMeUp_COM | or we could give the clients some rt31p2 |
| 22:46.19 | _VoiceMeUp_COM | and take mediatrix in exchange and go give them back to the makers |
| 22:46.23 | _VoiceMeUp_COM | its not registering |
| 22:46.26 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 22:46.35 | _VoiceMeUp_COM | well it is but always not found.. and no hash in the register from it |
| 22:49.53 | *** part/#asterisk Jabroni (n=asskick@red-corp-201.143.126.173.telnor.net) |
| 22:51.02 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 22:54.20 | _VoiceMeUp_COM | <PROTECTED> |
| 22:59.47 | *** part/#asterisk AlexGC (n=does@189.157.0.132) |
| 23:01.15 | Gtwy | what are pri lines |
| 23:03.49 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 23:03.52 | Vorondil | Nearly every question of the form, "what are blah", googling "blah" will answer your question. |
| 23:04.09 | Vorondil | Try it. The wisdom and power will flow through you. |
| 23:04.31 | Deeewayne | Gtwy: http://en.wikipedia.org/wiki/Primary_rate_interface |
| 23:04.56 | Vorondil | ...or somebody can hold your hand. |
| 23:05.03 | *** join/#asterisk Nix (n=Nix@opensuse/developer/nix) |
| 23:05.05 | Gtwy | i know, i am just playing with iptables and abscent mindedly typed that question since i am out of screen real estate |
| 23:05.09 | Gtwy | sorry |
| 23:06.20 | Vorondil | You should get a better window manager. |
| 23:06.50 | Gtwy | Vorondil: what do you use? i am at work so i lack options |
| 23:07.14 | Vorondil | Fvwm or gtfo, I say. |
| 23:08.24 | Gtwy | looks neat |
| 23:09.49 | Qwell[] | fvwm2-themes FTW |
| 23:09.52 | CoffeeIV_ | My asterisk receives a call, and then looks some stuff up in an AGI script before routing the call. I do an Answer(), Playtones(ring), then AGI(). When I go into the AGI it seems to stop ringing and there is silence. Is AGI supposed to stop the ringtones ? |
| 23:16.03 | *** join/#asterisk Marshall- (n=Marshall@cpe-76-181-165-37.columbus.res.rr.com) |
| 23:20.55 | dlynes | Anyone get the following error, 'Unknown symbol request_module', when trying to modprobe zaptel.ko from 1.4.3 in a 2.6.19.3 kernel? |
| 23:22.37 | *** part/#asterisk Nix (n=Nix@opensuse/developer/nix) |
| 23:23.59 | dlynes | Is Zaptel 1.4.3 compatible with 2.6.19.3? |
| 23:25.54 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:26.32 | CoffeeIV_ | Ringing() works for me when Playtones(ring) doesn't |
| 23:41.56 | *** join/#asterisk Izzy84075 (n=izzy8407@pool-71-115-244-117.spknwa.dsl-w.verizon.net) |
| 23:42.08 | Trevor_b | zaptel and zapata mention using pri_net and pri_cpe for PRI equipment (says its the only options) however i have an ITSP engineer telling me he is using fxs and fxo for signalling?? |
| 23:43.44 | Izzy84075 | Hi... I have a few questions about Asterisk that I can't find an answer for(At least, not in my quick scan through the site/forums...)... |
| 23:44.48 | Izzy84075 | What hardware is required for it to work with basic phone service? |
| 23:44.59 | Trevor_b | Izzy84075: TDM |
| 23:45.10 | Izzy84075 | Meaning..? |
| 23:45.11 | Trevor_b | Izzy84075: or X100P cards. |
| 23:45.31 | Trevor_b | TDM400 TDM800 TDM2400, these are all models of cards from digium to run POTS lines. |
| 23:45.58 | Izzy84075 | I looked at the digium stuff, but I was hoping there might be something a little cheaper. |
| 23:46.16 | Trevor_b | x100p.com |
| 23:46.23 | Trevor_b | single line interface around $29 |
| 23:46.59 | Izzy84075 | Awesome. Thanks! |
| 23:47.52 | Izzy84075 | Next question: What's the difference between FXO and FXS, and how can I tell which one I have? |
| 23:49.32 | *** join/#asterisk harlequin516 (n=sham@styk.net) |
| 23:49.53 | Izzy84075 | Nevermind... Wikipedia answered my question, I think... |
| 23:49.54 | harlequin516 | Okay I need to know what native bridging is. I have two conflicting definitions... |
| 23:50.53 | harlequin516 | Some people say canreinvite=no set native bridgin off, I have read emails saying that reinvite has nothing to do with native bridging |
| 23:55.43 | NightMonkey | OK, super newbie question. I have a SIP remote FXO that is *not* a zaptel card (obviously). How do I tell Asterisk to answer the calls from that FXO with the console, rather than giving a dialtone (!) to the caller? |
| 23:56.12 | NightMonkey | In other words, where do "channels" live when you don't have a zaptel.conf? |
| 23:57.19 | NightMonkey | I tried to have the HT-488's FXO set up as a SIP proxy, but the device returns "Not Implemented" when Asterisk tries to REGISTER with it. |
