| 00:00.10 | Trazz | ok well the person who recorded it said the newer version of asterisk supports more than that now |
| 00:00.13 | Trazz | but i dont see it |
| 00:00.15 | terrapen_ | i need to figure out a way to get a list of parked calls from a Polycom IP501 |
| 00:00.29 | SwK | you know 8K is about the best aphone line can do anyway |
| 00:00.35 | terrapen_ | the microbrowser plan is not going to work |
| 00:00.51 | Trazz | ok here |
| 00:00.53 | Trazz | If you're running the old asterisk system, I'd say we should stick with the old format (16 bit 8,000) but if you hope to upgrade, you'll have to get the prompts redone at some point. |
| 00:00.56 | ManxPower | Asterisk MAY support 16khz WAV, since I believe Asterisk's internal SLN formay is 16Khz |
| 00:01.00 | SwK | so just down sample it to 8K mono with sox (or your other favorite WAV editor) and be done iwth it |
| 00:01.05 | Trazz | that is here email to me stating if you plan to upgrade yo uwill need to redo them |
| 00:01.33 | Trazz | and teh woman who sent me the email is the woman who recorded all the prompts for asterisk to begin with.. but she is not technical |
| 00:01.42 | Trazz | allison |
| 00:01.58 | ManxPower | Trazz, ask on the mailing lists or look on the Wiki also |
| 00:02.33 | SwK | GSM sounds files sound like ass anyway... might as well be using LPC10 for them |
| 00:03.12 | terrapen_ | the "park" soft-key is kind of lame |
| 00:03.23 | terrapen_ | something visual would be best |
| 00:03.32 | terrapen_ | i guess i'll have to wait for microbrowser support on the 501 |
| 00:04.53 | SwK | microbrowser on the 501 hahaha |
| 00:04.57 | SwK | wonthappen |
| 00:05.16 | SwK | thats a 601 selling point |
| 00:05.20 | terrapen_ | hmmm |
| 00:05.26 | terrapen_ | wish i would have known :( |
| 00:05.33 | terrapen_ | how would you handle call parking? |
| 00:05.40 | terrapen_ | i have a bunch of dumbasses using these phones |
| 00:05.51 | SwK | create a key to autotransfer to yor parking extension |
| 00:06.03 | terrapen_ | and they're very busy, so they don't/won't want to sit and listen for a voice speaking the parking space |
| 00:06.09 | SwK | or just tell them to transfer the call to 700 (if you use the default) and listen for the parking spot |
| 00:06.24 | terrapen_ | swk, yeah, i thought about that....but to get to the 'Park' softkey, you have to hit 'More...' key apparently |
| 00:06.33 | SwK | terrapen_: well we do sell a full blown call manager thats windws native |
| 00:06.48 | terrapen_ | i could write a simple web app |
| 00:06.52 | terrapen_ | but that's not ideal |
| 00:06.52 | SwK | drag and drop and all that jazz |
| 00:07.01 | terrapen_ | this is a busy bicycle store |
| 00:07.09 | terrapen_ | not a lot of computers around the far parts of the store |
| 00:07.16 | SwK | then they probably want a key system not an asterisk box |
| 00:07.30 | terrapen_ | no, they already have an asterisk box. it works great |
| 00:07.43 | terrapen_ | except that they are too stupid/busy/whatever to use call parking |
| 00:08.28 | terrapen_ | the old system had 7 lines on every phone. if you called line 3 and someone put you on hold, line 3 would blink on every phoen |
| 00:08.33 | SwK | assign park to a hard key on then |
| 00:08.39 | SwK | but make it a speed dial |
| 00:08.41 | terrapen_ | then, they'd just push the '3' button |
| 00:08.47 | terrapen_ | swk, that might work |
| 00:09.05 | terrapen_ | they'd still have to listen for the parking space but.... |
| 00:09.19 | SwK | terrapen_ thats old system is a key system |
| 00:09.34 | SwK | most small shops like that are all most impossible to convert to a pbx |
| 00:09.38 | SwK | like asterisk |
| 00:09.38 | terrapen_ | old att merlin from the 80s |
| 00:09.44 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 00:09.46 | terrapen_ | well, they like it so far |
| 00:09.57 | SwK | a merlin is a hybrid... but more key then not |
| 00:10.00 | terrapen_ | maybe i can teach them call parking |
| 00:10.31 | SwK | I usually pass up key system upgrades to asterisk unless they really understand what they are getting... |
| 00:10.36 | terrapen_ | they're really not stupid...just insanely busy |
| 00:11.00 | terrapen_ | its a large store w/ lots of employees |
| 00:12.33 | terrapen_ | maybe a nice Mac OS X application that queries parked calls via the manager api and then displays them on the LCD tvs that are everywhere |
| 00:12.44 | terrapen_ | that would be sweet |
| 00:13.01 | terrapen_ | i'd need to compost, in real time, the parked call display and whatever is showing on TV |
| 00:13.12 | SwK | terrapen: thats pretty easy to do w/ something as simple as a script and scroll generator |
| 00:13.49 | terrapen_ | scroll generator? oh, for TV? |
| 00:13.57 | SwK | yeah |
| 00:14.26 | terrapen_ | do you know offhand of any scroll generators? |
| 00:14.48 | ManxPower | I'd like one too for a different project |
| 00:15.02 | SwK | you can check around the net for them |
| 00:15.22 | SwK | several take serial inputs but its been so long since I worked with them regularly I couldnt tell you a make or model to look at |
| 00:15.26 | terrapen_ | http://www.channelstorm.com/Features.html |
| 00:15.40 | terrapen_ | we're all digital/DVI, so it would be pretty expensive |
| 00:16.10 | terrapen_ | digital cable box -> digital composter -> dvi splitter box -> tv1, tv2, tv3 ... |
| 00:20.20 | terrapen_ | maybe a few LED signs |
| 00:23.08 | terrapen_ | ah-ha |
| 00:23.10 | terrapen_ | i have an idea |
| 00:23.22 | terrapen_ | employee presses a quick-dial key on the phone |
| 00:23.33 | terrapen_ | which dials an extension that: |
| 00:23.42 | terrapen_ | gets a list of all parked calls |
| 00:23.50 | terrapen_ | and then speaks them, with location from which they were parked: |
| 00:24.03 | terrapen_ | front counter one: dial 1 |
| 00:24.14 | terrapen_ | service: dial 2, etc. |
| 00:24.24 | terrapen_ | it would have to be an AGI |
| 00:25.10 | terrapen_ | so they don't have to remember long parking space #'s |
| 00:31.42 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:46.30 | Trazz | does anyone know of any good down sampling sofware for my prompts? |
| 00:46.47 | Trazz | sox appears to be a converter but not down sampler |
| 00:52.28 | littleball | when i use spandsp fax out, the recepient machine print nothing. |
| 00:52.31 | littleball | just a plain paper |
| 00:52.41 | littleball | hello, who have experience of using spandsp to fax ? |
| 00:56.32 | Idle | is there some good replacements for the asterisk voicemail |
| 00:56.38 | Idle | as in, the sounds |
| 01:03.34 | InHisName | kruz_ are you still there? Need to get more into your extensions.conf file to finish up. |
| 01:05.50 | MoutaPT | Is there any way to use billing to calls made with dialou.call files? |
| 01:06.10 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 01:07.52 | MoutaPT | i've use it with NVFaxdetec TE110P ast@home |
| 01:09.09 | InHisName | How do I change time on linux box that is running asterisk ? UTC needs to be eastern. |
| 01:09.33 | MoutaPT | date |
| 01:10.12 | InHisName | that shows the date, date /? didn't help that much either. |
| 01:10.33 | InHisName | does date set the time zone and daylight adjustments ? |
| 01:11.08 | InHisName | Will * pay attention to time zone and daylight savings times? |
| 01:11.56 | rob0 | UTC needs to be eastern?? |
| 01:13.13 | rob0 | date(1) is a part of GNU coreutils. All GNU stuff supports "--help" which gives a command syntax summary. |
| 01:13.15 | *** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:13.29 | InHisName | * reports UTC times and I want to read eastern instead, it is so anoying to translate all the time. |
| 01:13.29 | *** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:13.39 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 01:14.20 | rob0 | I'd also suggest you consult your OS documentation about how to maintain the time zone. ntpd is a nice tool for keeping the clock in sync. |
| 01:14.44 | rob0 | ah, perhaps that is as simple as starting * with your TZ set. |
| 01:15.08 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 01:15.09 | Druken | date is a paim in the ass... |
| 01:15.51 | rob0 | Seems like the actual issue is with glibc. |
| 01:16.47 | rob0 | My * keeps times in the local time zone. How? Not sure, glibc takes care of that for me. |
| 01:16.59 | rob0 | My hwclock is set to UTC. |
| 01:17.07 | rob0 | and I run ntpd |
| 01:24.35 | Nugget | rob0 is wise. |
| 01:24.59 | Nugget | system clocks should always be in UTC. |
| 01:26.31 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 01:29.17 | *** join/#asterisk SplasPood (n=jwb@ool-457de215.dyn.optonline.net) |
| 01:33.32 | tehmaze | how do I create a sub-menu that asks for voicemail box# and pin? |
| 01:33.52 | SplasPood | Voicemail(@context) |
| 01:34.09 | SplasPood | oh and pin |
| 01:34.12 | SplasPood | VoicemailMain(@context) |
| 01:34.38 | tehmaze | ah just no box number, thank you |
| 01:34.43 | SplasPood | np |
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| 01:54.09 | *** join/#asterisk [TK]D-Fender (n=joe@CPE0080c6fce706-CM00159a09e31c.cpe.net.cable.rogers.com) |
| 02:09.04 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
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| 02:28.30 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 02:28.53 | mdiehl | Hi all. |
| 02:30.14 | *** join/#asterisk Eggplant (n=none@dsl-72-19-44-253.cascadeaccess.com) |
| 02:32.34 | *** join/#asterisk TeePOG (n=1234@dsl-145-143-190.telkomadsl.co.za) |
| 02:32.39 | TeePOG | morning guys |
| 02:33.54 | mdiehl | Hi. |
| 02:39.07 | TeePOG | anyone successfully running asterisk on mandriva 2006? |
| 02:43.41 | *** join/#asterisk reza_ (i=reza@abort.boom.net) |
| 02:43.49 | reza_ | uck |
| 02:44.05 | reza_ | ok, i can make outgoing calls on this polycom phone but it will not accept incoming calls - any ideas? |
| 02:44.24 | mdiehl | Is it registered with your server? |
| 02:44.35 | reza_ | yeah |
| 02:44.36 | [TK]D-Fender | reza_ : where is it located relative to *? |
| 02:44.48 | reza_ | * -lan- polycom |
| 02:44.49 | *** join/#asterisk SplasPood (n=jwb@ool-457de215.dyn.optonline.net) |
| 02:45.02 | *** join/#asterisk trbldwine (i=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 02:45.11 | mdiehl | Do you get any error message on the Asterisk console? Set verbose=25. |
| 02:45.41 | reza_ | will try |
| 02:46.08 | reza_ | can i set the verbosity from the cli itself, or do i restart with 25 'v's? |
| 02:46.16 | mdiehl | ?From the cli. |
| 02:46.28 | reza_ | ah |
| 02:46.32 | reza_ | no = sign |
| 02:46.41 | reza_ | -- Registered SIP 'NewFrontDesk' at 10.0.0.181 port 5060 expires 300 |
| 02:46.45 | reza_ | it's registering |
| 02:46.56 | mdiehl | Ok, but what happends when you try to call the phone? |
| 02:47.03 | [TK]D-Fender | reza_ : pastebin your CLI output for the call to it |
| 02:47.35 | reza_ | pastebin is slow |
| 02:48.18 | reza_ | ...waiting... |
| 02:48.28 | reza_ | can i just paste here, just 4 lines |
| 02:50.21 | mdiehl | I won't tell anyone.... |
| 02:51.31 | reza_ | <PROTECTED> |
| 02:51.32 | reza_ | <PROTECTED> |
| 02:51.32 | reza_ | <PROTECTED> |
| 02:51.32 | reza_ | <PROTECTED> |
| 02:51.50 | reza_ | that's it -- im still waiting on pastebin |
| 02:52.05 | reza_ | wait, i have a thought.. |
| 02:52.05 | mdiehl | Doesn't indicate that it rang..... HMMm.... |
| 02:52.14 | mdiehl | Oh? |
| 02:52.33 | Druken | pastebin.ca? |
| 02:53.07 | reza_ | no |
| 02:53.07 | reza_ | ok, let me try again.. this will be dumb if it's the case |
| 02:53.32 | reza_ | yeah, that fixed it |
| 02:53.43 | mdiehl | What was it? |
| 02:54.02 | reza_ | <PROTECTED> |
| 02:54.26 | mdiehl | In the phone's config? Bummer. |
| 02:54.34 | reza_ | no |
| 02:54.35 | reza_ | in sip.conf |
| 02:54.38 | reza_ | defaultip=10.0.0.181 |
| 02:54.43 | reza_ | used to be 10.10.10.181 |
| 02:55.39 | reza_ | 10 hours later, got this bastard to work. yay, thanks. |
| 02:55.51 | mdiehl | I only watched. |
| 02:55.55 | reza_ | so this phone has a 'forward' button -- what's that mapped to? |
| 02:56.09 | reza_ | i am so over going through the miles of docs |
| 02:56.12 | reza_ | ugh |
| 02:56.12 | mdiehl | Opposite of reverse? |
| 02:56.15 | reza_ | :P |
| 02:56.58 | Druken | reza_: generally if the phone has a forward, it'll do a sip redirect |
| 02:57.04 | reza_ | didnt work though |
| 02:57.10 | reza_ | phone said '1 new missed call' |
| 02:57.33 | reza_ | ah |
| 02:57.46 | reza_ | <PROTECTED> |
| 02:57.54 | reza_ | 102 isn't defined in the sip context |
| 02:59.22 | reza_ | sweet - it works |
| 02:59.33 | Druken | :) |
| 03:01.16 | reza_ | i wonder how i clear the number of missed calls |
| 03:02.19 | Druken | no idea... it's a what? |
| 03:02.21 | Druken | polycom? |
| 03:02.30 | reza_ | yeah, it went away while i wasn't looking |
| 03:02.36 | reza_ | *shrug* |
| 03:20.52 | reza_ | so i've got a tdm400 w/ 3 fxs's -- how do i know what fxs port is being used to dial a number (so i can set different caller Ids approprietly) |
| 03:21.39 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 03:22.41 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 03:22.59 | Druken | reza_: uhmm... just set the CID in zapata.conf? |
| 03:26.02 | reza_ | how do i define it per channel |
| 03:26.20 | reza_ | i'm ggoing through the wiki |
| 03:26.22 | *** join/#asterisk Splas (n=jwb@ool-457de215.dyn.optonline.net) |
| 03:26.55 | reza_ | ah wait |
| 03:27.04 | reza_ | i define new sections per channel... ugly... |
| 03:29.22 | reza_ | sweet |
| 03:30.33 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 03:37.20 | reza_ | thanks |
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| 04:23.50 | Dimitripietro | Anyone here could help me compiling the sangoma drivers ? |
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| 04:35.36 | CunningPike | Dimitripietro: Sorry - never done it. What's the problem? |
| 04:35.45 | CunningPike | And where is everyone? |
| 04:36.02 | CunningPike | ~seen dlynes_home |
| 04:36.13 | jbot | dlynes_home is currently on #asterisk. Has said a total of 4 messages. Is idling for 23h 49m 16s, last said: 'Idle: probably works well if you make sure echocancelwhenbridged=no'. |
| 04:36.22 | CunningPike | Slacker |
| 04:39.35 | *** join/#asterisk mrdigital (n=wild@pool-72-81-28-149.phlapa.east.verizon.net) |
| 04:39.50 | mrdigital | can any FXO/FXS Device work with Asterisk? |
| 04:40.34 | CunningPike | mrdigital: As long as, if it needs drivers, drivers that Asterisk can talk to are available (usually zaptel) |
| 04:40.47 | mrdigital | Will the Zoom 5102 work? |
| 04:40.59 | CunningPike | mrdigital: Never heard of it...... who makes it? |
| 04:41.07 | mrdigital | Zoom |
| 04:41.10 | CunningPike | lol |
| 04:41.12 | CunningPike | Figures |
| 04:41.19 | mrdigital | its a box simliar to Sipura 3000/3102 |
| 04:41.31 | mrdigital | has FXO port |
| 04:41.32 | mrdigital | FXS |
| 04:41.47 | CunningPike | mrdigital: Oh, an external box? If it has an ethernet port and is SIP compliant, it will work |
| 04:41.48 | mrdigital | does the power thing where if the box cant connect it switchs the line |
| 04:41.55 | mrdigital | uhm |
| 04:42.16 | mrdigital | CunningPike: cam i pm you the specs? |
| 04:42.35 | mrdigital | or i can just paste the key sepcs |
| 04:42.40 | CunningPike | mrdigital: No need - if it's SIP, it will work |
| 04:42.48 | mrdigital | Supports SIPv2 |
| 04:43.18 | CunningPike | mrdigital: Perfect |
| 04:43.20 | mrdigital | will that work? |
| 04:43.23 | mrdigital | Perfect? |
| 04:43.43 | CunningPike | mrdigital: Yup - set it up just like an SPA-3000 |
| 04:43.53 | mrdigital | Awesomeness |
| 04:44.07 | mrdigital | now anyone wanna help me write a menu? |
| 04:45.18 | mrdigital | question.. so the call comes from the pstn line.. then goes to the FXO port... to asterisk then sent back to the box to the fxs port then to the phone? |
| 04:45.41 | CunningPike | mrdigital: It's quite simple - you need sound recording that you BackGround() and which reads the options. Then, in the same context, you need single-digit extensions (exten => 1,1,DoSomethingDQ()) that do what ever you want each menu option to do |
| 04:45.59 | mrdigital | wait what CunningPike: LOL |
| 04:46.26 | CunningPike | mrdigital: If you have an analog phone, yes, it should be able to do that |
| 04:46.57 | mrdigital | you know those Cordless phones that come with 2 headsets or 4 ? |
| 04:46.59 | mrdigital | will those work? |
| 04:50.27 | mrdigital | whats the wiki url for asterisk? |
| 04:55.21 | mrdigital | anyone here? |
| 04:57.47 | Dimitripietro | <CunningPike> : I,m getting an error while compiling the sangoma drivers concerning udev |
| 04:58.05 | Dimitripietro | search for voip wiki on google |
| 04:58.19 | CunningPike | Dimitripietro: 2.6 kernel, I assume. Have you modified your rules and permissions files? |
| 04:58.23 | mrdigital | i found it |
| 04:58.29 | CunningPike | ~thewiki |
| 04:58.32 | jbot | it has been said that thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 04:58.37 | Dimitripietro | I done nothing |
| 04:59.02 | CunningPike | Dimitripietro: You have installed zaptel, right? I think that's needed with Sangoma cards |
| 04:59.05 | CunningPike | ~udev |
| 04:59.06 | jbot | it has been said that udev is at http://www.kernel.org/pub/linux/utils/kernel/hotplug/udev-FAQ , or broken and shit |
| 04:59.08 | mrdigital | CunningPike: is there a howto on making menus? |
| 04:59.14 | Dimitripietro | YesI did |
| 04:59.29 | Dimitripietro | I was running my system with a TDM400 ten min ago everything was fine |
| 04:59.40 | CunningPike | mrdigital: Yes - there is a good one - search for IVR on the wiki (even though it's not an IVR) |
| 04:59.43 | Dimitripietro | The a decided to give a try to the sangoma |
| 04:59.45 | mrdigital | alright |
| 05:00.19 | CunningPike | Dimitripietro: OK - your problem is probably beyond my knowledge then |
| 05:00.34 | mrdigital | CunningPike: Is this the right url? http://www.voip-info.org/wiki/view/IVR |
| 05:00.48 | Dimitripietro | Thx for trying :-) |
| 05:01.34 | CunningPike | mrdigital: http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu |
| 05:01.58 | CunningPike | Dimitripietro: Ya - sorry. If it's a Sangoma specific thing, I have no idea.... |
| 05:02.26 | Dimitripietro | First time i'm trying a sangoma |
| 05:03.02 | Dimitripietro | Anyway |
| 05:03.10 | Dimitripietro | Going to bed |
| 05:03.23 | Dimitripietro | Waiting for a replys from Sangoma |
| 05:03.26 | Dimitripietro | + |
| 05:03.32 | CunningPike | OK - next time dlynes_home is on, he might be able to help you |
| 05:04.03 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 05:07.52 | DrkShdw | <PROTECTED> |
| 05:07.56 | DrkShdw | mt |
| 05:10.07 | *** part/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 05:14.10 | Splas | Hrm... is there any way I can let users modify the PINs for MeetMe rooms via the dialplan, or is the ONLY way to modify meetme.conf by hand? |
| 05:15.08 | CunningPike | SplasPood: I haven't found a dialplan app to do it - what we're doing is writing a shell script to modify the file and calling that from the dialplan |
| 05:16.06 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:23.08 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 05:23.16 | SplasPood | CunningPike: yea that was the only thing I had come up with... Might I be able to get a copy of the script you wrote? |
| 05:23.35 | CunningPike | SplasPood: Got a time machine? :D |
| 05:23.45 | SplasPood | oh, *is writing* |
| 05:23.48 | SplasPood | I read that as |
| 05:23.50 | SplasPood | we wrote |
| 05:23.56 | CunningPike | SplasPood: Sorry |
| 05:24.14 | SplasPood | np, if I end up writing one of the next couple days, I'll drop you a msg |
| 05:24.24 | SplasPood | god |
| 05:24.30 | SplasPood | I can't read *OR* write tonight |
| 05:24.58 | CunningPike | SplasPood: That would be great - I'm writing a whole bunch of scripts for our help desk folks for making basic changes and I plan to release those to the list over the summer |
| 05:25.43 | SplasPood | I've been working on a full blown frontend, but its mostly backend stuff and db schema right now.. Gotta get one of our PHP devs to create a UI for me.. (I'm not good at that, nor do I enjoy it) |
| 05:26.05 | SplasPood | maybe its worth checking out that app_conference or whatever |
| 05:26.08 | SplasPood | I'd heard good things |
| 05:27.08 | CunningPike | SplasPood: Ya - we're pretty happy with the basic stuff - I just want to offload some of the day-to-day MAC stuff to our Tier 1 folks |
| 05:28.42 | TeePOG | anyone available to help a newbie?' |
| 05:30.02 | TeePOG | i need to set up SIP in our internal network to have everyone call through the same phone line |
| 05:30.23 | TeePOG | getting the digium card tomorrow, just need to set up the SIP in the meantime |
| 05:31.46 | TeePOG | sorry for the conversation killer hmm? |
| 05:36.07 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 05:37.23 | SplasPood | TeePOG: Your question is about 100x too vague |
| 05:40.28 | TeePOG | ok, that's because that's about as much of an idea i have of how to configure this thing. I have installed Mandriva and installed asterisk; Asterisk is running as a service, and I do not know where to go from here |
| 05:41.09 | TeePOG | i would appreciate any guidance SplasPood |
| 05:43.22 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:45.45 | Qwell | How well would g729 fare over dialup? |
| 05:45.54 | Qwell | yes, I understand the consequences of doing so |
| 05:46.35 | SplasPood | TeePOG: Well you need a specific question like... "I've setup a sip user and I cannot get my client to register, I get this message..." |
| 05:47.03 | CunningPike | TeePOG: What type of card are you getting, and what are you connecting it to? |
| 05:47.18 | SplasPood | the "I need a PBX that handles SIP, how do I make it work?" question isn't gonna garner a lot of answers other than .. RTFM |
| 05:48.08 | TeePOG | OK SplasPood -- I just thought that, as expert users, you would have an idea of where to start, a beginners' guide as it were |
| 05:48.27 | SplasPood | TeePOG: Well there's the asterisk book... |
| 05:48.32 | SplasPood | anyone recall the ~ code for that? |
| 05:48.36 | CunningPike | ~thebook |
| 05:48.38 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:48.46 | SplasPood | TeePOG: www.voip-info.org is a generally good source of reference |
| 05:48.52 | TeePOG | CunningPike: cannot recall the model number, but it's the one that replaced the x100p ... it's a 1fx0 card connecting to an analog line |
| 05:48.54 | CunningPike | ~thewiki |
| 05:48.55 | jbot | thewiki is, like, at http://www.voip-info.org/wiki-Asterisk |
| 05:49.23 | CunningPike | TeePOG: You realize you will only be able to make OR receive one call at a time, right? |
| 05:49.54 | SplasPood | CunningPike: good point |
| 05:49.57 | TeePOG | yes CunningPike, it's just 3 extensions... it's just to centralise the 3 offices' lines into 1. |
| 05:50.11 | TeePOG | and we will be getting an isdn line in near future |
| 05:50.33 | DrkShdw | isdn? does that still exist? :P |
| 05:50.35 | TeePOG | thanks guys, will rtfm and get back with meaningful questions |
| 05:50.51 | CunningPike | TeePOG: OK - good luck! |
| 05:50.57 | SplasPood | I always wanted an ISDN line |
| 05:51.00 | CunningPike | DrkShdw: Depends what you call ISDN |
| 05:51.17 | CunningPike | DrkShdw: Our 24-channel PRI is called ISDN by some folks |
| 05:51.18 | TeePOG | in this country, analog and gprs are considered normal ways to connect to internet |
| 05:51.28 | CunningPike | TeePOG: Where are you? |
| 05:51.32 | TeePOG | yes, but where you have pri we have BRI |
| 05:51.39 | TeePOG | CunningPike: South Africa |
| 05:51.50 | CunningPike | TeePOG: Ah - OK |
| 05:51.56 | DrkShdw | CunningPike: I'm well aware :) I was joking ;) |
| 05:52.10 | CunningPike | DrkShdw: Too late at night for subtle humor ;) |
| 05:52.37 | CunningPike | Anyway, I'm off to bed - probably just as well :D |
| 05:52.42 | TeePOG | too late at night? it's almost 8AM here, and that means i've been up for 25hours |
| 05:52.55 | TeePOG | probably good for ~12hours yet |
| 05:53.31 | ManxPower | I'm too old for that sort of thing |
| 05:54.08 | brookshire | bah! you're never too old for that ;) |
| 05:54.57 | ManxPower | it's 1am and I'm back from the party already. |
| 05:55.31 | brookshire | hah.. you know good and well the parties don't start in alabama until 2am |
| 05:56.03 | ManxPower | brookshire, You don't know the kind of parties I go to. 8-) |
| 05:56.18 | brookshire | i like it that way |
| 05:56.19 | brookshire | :D |
| 05:56.27 | TeePOG | ...when dad and the cousins comes home to the family? *duck & run* |
| 05:58.02 | brookshire | depends on who you call daddy, i guess |
| 05:58.03 | brookshire | :/ |
| 05:58.20 | TeePOG | soooo, who's *your* daddy? |
| 05:58.29 | Qwell | I'm so not going there |
| 05:59.01 | brookshire | but qwell, |
| 05:59.09 | Assid | sup Qwell |
| 06:04.23 | *** part/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
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| 06:05.26 | *** part/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 06:14.09 | *** join/#asterisk RoyK[de] (n=roy@static.148.bras.breisnet.com) |
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| 06:15.12 | *** part/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 06:16.02 | *** join/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 06:16.18 | *** part/#asterisk joe_acme (i=HydraIRC@mar92-9-82-237-75-54.fbx.proxad.net) |
| 06:21.35 | Qwell | way to kill a channel, btw :P |
| 06:21.45 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 06:27.58 | SplasPood | hrm, it'd also be nice if you could somehow Authenticate() people using their voicemail pins |
| 06:29.50 | RoyK[de] | morgen |
| 06:30.05 | RoyK[de] | SplasPood: agi :) |
| 06:30.25 | Qwell | SplasPood: like, say...vmauthenticate? |
| 06:30.30 | SplasPood | RoyK[de]: parsing... wair .. |
| 06:30.32 | SplasPood | qwell |
| 06:30.33 | SplasPood | wha?! |
| 06:30.33 | Qwell | nah, too easy |
| 06:30.46 | SplasPood | omg. |
| 06:30.46 | SplasPood | hah |
| 06:30.49 | SplasPood | perfect |
| 06:30.52 | SplasPood | Qwell you da man. |
| 06:33.47 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:37.29 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 06:48.50 | SheriF_WorK | i have a question about mpg123 and muslic on hold .. it's normal that mpg123 keep running even after the musiconhold call has been disconnected ? |
| 07:00.18 | SplasPood | yes |
| 07:00.27 | SplasPood | they're generally spawned at asterisk startup |
| 07:03.28 | *** join/#asterisk evilmnky (n=evilmnky@adsl-153-142-75.mia.bellsouth.net) |
| 07:09.57 | *** part/#asterisk my007ms (n=noor@217.139.224.194) |
| 07:17.04 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
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| 07:44.07 | FuriousGeorge | anyone not a newbie finding that there is no real advanced support for asterisk users |
| 07:49.15 | dlynes_home | FuriousGeorge: nope |
| 07:49.26 | dlynes_home | FuriousGeorge: I've found enough help on here, myself |
| 07:49.43 | dlynes_home | SheriF_WorK: mpg123 is buggy as all hell |
| 07:49.52 | dlynes_home | SheriF_WorK: I wouldn't use it on a production system |
| 07:49.59 | dlynes_home | SheriF_WorK: try madplay, or native mode |
| 07:51.37 | SheriF_WorK | dlynes_home: will use mpg321 |
| 07:51.49 | SheriF_WorK | madplay were geting the sound out at the pc speaker :) |
| 07:52.03 | SheriF_WorK | now 1st i have to fix a dtmf problem issue :-s |
| 07:52.21 | dlynes_home | mpg321 sucks, too |
| 07:53.38 | FuriousGeorge | dlynes_home: ive been at this for almost two years now. starting about one year in, just about any problem i take here, to the forum, or to the mailing list, i know before hitting send/submit that i'm basically gonna have to figure it out on my own. i cant put my finger on why. i dunno if its just a matter of catching people with the right level of experience, or if * is just quirky as it stands now with 1.2 |
| 07:54.22 | FuriousGeorge | for instance, my server will take the last entry for the CID for zapata,conf, even if its an fxs, and put that as the cid |
| 07:54.34 | dlynes_home | probably just a matter of catching the attention of people that have the knowledge |
| 07:54.59 | FuriousGeorge | what sense does that make? for the longest time i couldnt figure out why half my calls seem to hace the cid of one of my internal extensions |
| 07:55.21 | dlynes_home | FuriousGeorge: it's going out on a pri? |
| 07:55.39 | FuriousGeorge | no pri here. then there was the issue where at random, setting a voicemail message fails to delete the temp.gsm/.wav message |
| 07:56.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:56.21 | dlynes_home | FuriousGeorge: the failure to delete the temp gsm/wav file was probably the well-known race condition with voicemail |
| 07:56.39 | FuriousGeorge | so that mailbox will only play the temp message. my solution was to rename unavail.* to temp.* till after a few months of asking in here, qwell told me what was going on |
| 07:56.46 | dlynes_home | FuriousGeorge: if you don't have a pri, how do you set the callerid of outbound calls? |
| 07:57.14 | FuriousGeorge | im talking about inbound calls. when * doesnt get CID info it just pulls the CID from the last entry of zapata.conf |
| 07:57.14 | dlynes_home | FuriousGeorge: well, i'm in the process of rewriting voicemail |
| 07:57.24 | FuriousGeorge | thank you |
| 07:57.34 | dlynes_home | FuriousGeorge: so hopefully if you can read C code, it'll be a little clearer soon what's going on in app_voicemail.so |
| 07:57.48 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 07:58.06 | dlynes_home | FuriousGeorge: ah...for me, it's not setting it to anything |
| 07:58.14 | dlynes_home | FuriousGeorge: so i've been forcing it to 'Unavailable' |
| 07:58.46 | FuriousGeorge | i can read some. then there is the retrieve button on my snoms. on some phones, if there is no mail in the mailbox, it does nothing, which is fine. on others hitting that button calls extension 109 when there is no voicemail |
| 07:58.48 | dlynes_home | FuriousGeorge: but that could be that the callerid='' is getting applied to whatever channel it's most closely associated with in the zapata.conf file |
| 07:59.14 | dlynes_home | what's special about extension 109 on your system? |
| 07:59.16 | FuriousGeorge | nothing to really make you feel uneasy about your line of work than that kind of unpredictable behavior. and thats just some of it |
| 07:59.44 | FuriousGeorge | i dont understand why set(__CALLERID=MM-${CALLERID}) doesnt do what i expect it to for incoming calls |
| 08:00.40 | dlynes_home | FuriousGeorge: that would set a local variable '__CALLERID' to 'MM-${CALLERID}', right? |
| 08:00.53 | dlynes_home | except that won't actually work quite the way you think it will |
| 08:00.56 | dlynes_home | you need to rephrase it |
| 08:01.11 | dlynes_home | set(__CALLERID=MM-${CALLERID(num)}) |
| 08:01.20 | FuriousGeorge | i think this last one is my favorite: parking doesnt work on one of my servers. i can blind x-fer, and not know where the call went (forcing me to install metermaid patch on 1.2.9.1, which works ok), or i can not use pariking |
| 08:01.26 | FuriousGeorge | im trying to change the callerid name |
| 08:01.37 | dlynes_home | ah |
| 08:01.44 | FuriousGeorge | so if its John Smit it will become MM-John Smith |
| 08:02.05 | dlynes_home | set(CALLERID(name)=MM-${CALLERID(name)}) |
| 08:02.35 | FuriousGeorge | ill give that a whirl |
| 08:02.47 | FuriousGeorge | but its just all this shit together is starting to frustrate me |
| 08:03.11 | dlynes_home | FuriousGeorge: it sounds like for your parking issue that you might not have parkedcalls(?) included properly in the necessary context in your dialplan |
| 08:03.29 | FuriousGeorge | dlynes_home: i sure do, remember i can blind-xfer to it |
| 08:03.41 | FuriousGeorge | but the server literally does NOTHING if i try an atxfer to actually hear the spot |
| 08:04.14 | FuriousGeorge | if i try it from a zap channel, i just get dead air, and the CLI says --Starting Simple Switch on Zap1/1 --Hungup Zap1/1 |
| 08:04.37 | dlynes_home | ah |
| 08:04.46 | FuriousGeorge | from a zip channel, the call is just instantly destroyed |
| 08:04.51 | dlynes_home | yeah...i Just installed my first client for call parking on Friday |
| 08:04.51 | FuriousGeorge | again no response from CLI |
| 08:04.56 | dlynes_home | before that, I never used call parking |
| 08:06.10 | FuriousGeorge | so i find myself doing now what i was doing before *v1.2 came out. i log into the chat every day, pick a problem to work on, float the question out there, knowing im not gonna get a response that acutally help 99% of the time, but really I'm in here to see if the topic switches to "1.4 beta released" |
| 08:06.52 | FuriousGeorge | because i think (probably falsely) that the server behavior will be more predictable with version 1.4 |
| 08:06.52 | dlynes_home | I don't think 1.4 will get released until August, myself |
| 08:06.57 | dlynes_home | I was thinking July before |
| 08:06.58 | FuriousGeorge | i believe that |
| 08:07.00 | dlynes_home | but looks like august now |
| 08:07.05 | FuriousGeorge | for the beta? |
| 08:07.11 | dlynes_home | for the final |
| 08:07.32 | FuriousGeorge | yeah, im expecting beta in July and final some time in August. |
| 08:07.59 | dlynes_home | just bought a laptop on friday |
| 08:08.10 | dlynes_home | so hopefully i'll have more time to work on the voicemail code now |
| 08:08.51 | FuriousGeorge | its to the point that when i install a system i tell people that ill upgrade them to the next version later this summer for free, because i know there gonna come at me with something like "why does the retrieve button on my snom call sandy when i have no mail" and im gonna have no answer for it but "lets see what happens with the new version which ill install for free" |
| 08:09.30 | dlynes_home | I take it you don't know how to code in C, eh? |
| 08:09.32 | FuriousGeorge | i mean, i usually dont complain this much, but its not good for my nerves this shit |
| 08:09.38 | FuriousGeorge | i know a little |
| 08:10.01 | dlynes_home | but not enough to fix your problems with the code? |
| 08:10.19 | FuriousGeorge | but what does that have to do with anything. when i got into this no one said "its only gonna work right if your an experienced coder" |
| 08:10.53 | dlynes_home | no, but I have found the odd block I run into, I need to modify some code to change the behaviour |
| 08:11.26 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 08:12.23 | *** part/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 08:12.34 | SheriF_WorK | seems that asterisk at me don't pass the dtmf |
| 08:12.36 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 08:12.37 | SheriF_WorK | any idea how to debug this ? |
| 08:12.57 | FuriousGeorge | im hoping that this asterisk thing will really take off when they get SLA working, and there will be real presure on Digium to tie the lose ends together. i mean, fine, maybe if i were a more experienced programmer or user i could get this working right, but at some point you gotta stop blaming your users |
| 08:13.04 | FuriousGeorge | SheriF_WorK: sip channel? |
| 08:13.33 | SheriF_WorK | yes |
| 08:14.09 | FuriousGeorge | SheriF_WorK: what dtmfmode in sip.conf, what codec you using? |
| 08:14.29 | SheriF_WorK | FuriousGeorge: i'm using RFC2833 |
| 08:14.35 | SheriF_WorK | and g723 |
| 08:14.44 | SheriF_WorK | amd tried info too and didn't work |
| 08:14.52 | FuriousGeorge | afaik, that should work |
| 08:14.57 | SheriF_WorK | <PROTECTED> |
| 08:15.03 | SheriF_WorK | FuriousGeorge: yes but it's not working :-s |
| 08:15.20 | FuriousGeorge | SheriF_WorK: i know the feeling |
| 08:15.32 | FuriousGeorge | try ulaw for sanity's sake and make sure it works |
| 08:15.38 | FuriousGeorge | what client are you using? |
| 08:16.58 | SheriF_WorK | FuriousGeorge: it's complecated setup 1 min i'll try to use ulaw in the other end. |
| 08:18.10 | FuriousGeorge | SheriF_WorK: try a different client see if you have better luck |
| 08:18.29 | FuriousGeorge | call some 800 numbers with IVRs, see if you can get the dtmf to act reliable |
| 08:20.16 | SheriF_WorK | FuriousGeorge: same with u-law |
| 08:20.27 | FuriousGeorge | try ulaw with inband dtmf |
| 08:20.31 | FuriousGeorge | dtmfmode=inband |
| 08:21.38 | SheriF_WorK | FuriousGeorge: same |
| 08:21.40 | SheriF_WorK | hum |
| 08:22.11 | FuriousGeorge | you are doing a sip reload between changing settings in sip.conf, right |
| 08:22.24 | dlynes_home | SheriF_WorK: have you tried using rtp debug, to make sure it's passing rtp? |
| 08:22.33 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 08:27.55 | *** join/#asterisk X-Rob (n=rob@CPE-60-231-85-52.qld.bigpond.net.au) |
| 08:31.48 | *** join/#asterisk SparFux (n=player@e182020196.adsl.alicedsl.de) |
| 08:32.34 | FuriousGeorge | correct me if im wrong, but if he can hear the other side, then he is passing enough rtp for inband dtmf |
| 08:32.37 | SparFux | Before Answer() I do a wait() on my bri line. When I do Wait(9) or everything above 8, the line won't be picked up. How can this be? |
| 08:33.03 | X-Rob | SparFux, do a 'Ringing' first. |
| 08:33.03 | FuriousGeorge | most people answer then wait(3) |
| 08:33.13 | X-Rob | or a Progress |
| 08:33.24 | SparFux | X-Rob: What Progress? |
| 08:33.25 | X-Rob | you need to tell the exchange you're doing something with the call, otherwise it times out |
| 08:33.35 | X-Rob | SparFux, 'show application profress' |
| 08:33.35 | SparFux | X-Rob: ah... |
| 08:33.46 | X-Rob | progress even |
| 08:33.59 | TeePOG | a question: how do I create a username on asterisk for my xten-lite sip client? the client keeps saying "403 Forbidden" but in the asterisk console, the output of "sip show peers" shows the name of the program, but (Unspecified) under Host |
| 08:35.06 | FuriousGeorge | [xten1] username=frank |
| 08:35.15 | FuriousGeorge | secret= |
| 08:35.33 | FuriousGeorge | throw in a host=dynamic or soemthing |
| 08:35.43 | TeePOG | well, the IP is static |
| 08:35.54 | FuriousGeorge | or something |
| 08:36.04 | TeePOG | is secret the "Password" field in the client? |
| 08:36.13 | TeePOG | or blank? |
| 08:36.17 | FuriousGeorge | host=123.456.789.102 |
| 08:36.23 | FuriousGeorge | yes |
| 08:36.34 | FuriousGeorge | and it can be left blank |
| 08:37.08 | SparFux | X-Rob: Hell, it works! I had this ringing issue sometime already, but I keep forgetting about it :-( |
| 08:37.29 | FuriousGeorge | then in the dialplan you would "dial(sip/xlite1" because that is the peername defined in the [] in sip.conf |
| 08:37.38 | X-Rob | SparFux, http://www.aussievoip.com and click on the 'donate now' button 8) |
| 08:37.45 | FuriousGeorge | lol |
| 08:37.57 | TeePOG | FuriousGeorge: which field does the "number" then go? from another sip client on the lan |
| 08:38.17 | FuriousGeorge | i guess i owe you for those public domain classical pieces |
| 08:38.52 | FuriousGeorge | exten => 101,1,dial(sip/xlite1) |
| 08:38.59 | FuriousGeorge | so when someone dials 101 it calls xlite |
| 08:39.17 | FuriousGeorge | since most clients cant dial "xlite1 directly |
| 08:39.24 | TeePOG | does that last line go in sip.conf? |
| 08:39.51 | TeePOG | this is my first day trying to get this working, be gentle |
| 08:40.09 | FuriousGeorge | teepog, that line was a "dialplan code". it should be pretty recognizeable |
| 08:40.12 | FuriousGeorge | im always gentle |
| 08:40.20 | FuriousGeorge | but im allergic to the underinformed |
| 08:40.32 | FuriousGeorge | ~docs |
| 08:40.34 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:40.34 | FuriousGeorge | !!!!!!!!! |
| 08:42.28 | FuriousGeorge | TeePOG: i started off with a copy of xlite (for linux i think) and i got an account on sipphone.conf which is now gizmoproject.com |
| 08:42.38 | TeePOG | oh ffs, i've been awake for 27 hours and counting, redoing entire network for 3 shops. If you do not want to help, then don't, OK? No need to be nasty; I try to help where I can as well |
| 08:42.49 | TeePOG | this is to centralise the phones for 3 shops |
| 08:42.54 | FuriousGeorge | i wasnt being nasty |
| 08:42.55 | FuriousGeorge | ? |
| 08:43.10 | TeePOG | i have been reading since forever on those docs |
| 08:43.18 | TeePOG | i just need some assistance |
| 08:43.21 | TeePOG | but never mind ok? |
| 08:43.43 | FuriousGeorge | how did you miss the part where everything that starts with exten=> is part of the dialplan |
| 08:43.59 | TeePOG | tml |
| 08:44.02 | FuriousGeorge | ok, that was a little nasty, but i thought i was being helpful all along |
| 08:44.03 | TeePOG | stml even |
| 08:44.16 | TeePOG | short term memory loss,literally |
| 08:44.21 | TeePOG | i have serious adhd |
| 08:44.36 | TeePOG | and just need some telling and translating |
| 08:44.37 | FuriousGeorge | anyway as i was saying before you started spouting off at me |
| 08:44.43 | TeePOG | sorry |
| 08:44.52 | SparFux | X-Rob: How about electronic cash? Not possible? |
| 08:44.54 | TeePOG | please continue |
| 08:45.25 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 08:45.44 | FuriousGeorge | get a client and first try to log it in to asterisk, which is i think what you were doing when you came in here. with the info i gave you at the start you should be able to register a peer with asterisk (use nat=yes if not local) |
| 08:45.59 | TeePOG | it's all local |
| 08:46.41 | FuriousGeorge | so? |
| 08:46.47 | FuriousGeorge | sip reload at cli |
| 08:47.16 | TeePOG | what "type" is xten1? |
| 08:47.24 | TeePOG | it's asking for that |
| 08:47.32 | FuriousGeorge | a friend |
| 08:47.35 | TeePOG | ok |
| 08:47.53 | FuriousGeorge | peers call out, users get calls, firends are peers + users |
| 08:48.55 | TeePOG | ok |
| 08:48.56 | TeePOG | now |
| 08:49.10 | TeePOG | xten still says "403 - Forbidden" |
| 08:49.28 | FuriousGeorge | did you sip reload at cli |
| 08:49.32 | TeePOG | yup |
| 08:49.36 | TeePOG | and sip show peers |
| 08:49.50 | FuriousGeorge | pastebin your [xlite] section in sip.conf |
| 08:50.01 | FuriousGeorge | ~pb |
| 08:50.03 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 08:50.06 | TeePOG | shows xten1 as registered, but Unmonitored |
| 08:50.21 | FuriousGeorge | you get 403 when you dial or when you log in |
| 08:50.23 | FuriousGeorge | sip show peers |
| 08:50.37 | TeePOG | sip show peers |
| 08:50.40 | TeePOG | shows xten1 as registered, but Unmonitored |
| 08:50.41 | SparFux | Is there some kind of CAPI app_senddigit? So that I can send digits over capi device? |
| 08:50.54 | FuriousGeorge | so it seems that you have logged in |
| 08:51.23 | TeePOG | so how do i test whether that client can receive calls? |
| 08:51.52 | FuriousGeorge | now you need to a> have someone to call, and b> define an extension in the [context] that belongs to "outbound calls from softphones" or something |
| 08:52.07 | FuriousGeorge | ok you wanna get called? |
| 08:52.13 | FuriousGeorge | you need a> someone to call you |
| 08:52.23 | TeePOG | i can install xten on another pc |
| 08:52.28 | TeePOG | it's a cybercafe |
| 08:52.29 | SheriF_WorK | strange not working at all ! |
| 08:52.32 | SheriF_WorK | :-s |
| 08:52.33 | FuriousGeorge | now were cooking with gas |
| 08:52.37 | TeePOG | brb ok |
| 08:53.43 | TeePOG | we have this voip standalone phone here, but i have no idea how to configure it for asterisk |
| 08:54.17 | TeePOG | it's supposed to use iax2, but it also just sits there and says "Login Failed" |
| 08:54.26 | FuriousGeorge | can you set a username and a password in the phone? can you tell the phone where to look for a server? when you plug the phone is, does it have a dhcp client that gets an ip? |
| 08:54.41 | FuriousGeorge | so now we gotta add a friend to iax.conf |
| 08:55.08 | TeePOG | ok |
| 08:55.23 | FuriousGeorge | first is the answer to all three of the above questions yes? |
| 08:55.27 | FuriousGeorge | "yes" |
| 08:55.38 | TeePOG | yes |
| 08:55.46 | TeePOG | and |
| 08:55.47 | TeePOG | yes |
| 08:55.48 | FuriousGeorge | point the phone at asterisk |
| 08:56.00 | FuriousGeorge | then point iax.conf at the phone, then iax2 reload |
| 08:56.05 | FuriousGeorge | and get back to me |
| 08:57.09 | TeePOG | ok |
| 08:57.18 | FuriousGeorge | be more verbose |
| 08:57.43 | FuriousGeorge | oh, nm, i thought you finished. |
| 08:57.46 | TeePOG | all right, i shall return in short time |
| 08:57.52 | TeePOG | momentarily, even |
| 08:57.54 | TeePOG | ;-] |
| 08:58.05 | FuriousGeorge | i wait in anticipation |
| 09:01.38 | TeePOG | this is strange... the phone now shows up under iax2 show peers, but with subnet mask 255.255.255.255 (on the phone it's 255.255.255.0) and the phone still says "Login failed". |
| 09:02.35 | FuriousGeorge | dont worry about the mask mine does the same thing |
| 09:03.26 | FuriousGeorge | in iax.conf your username is the thing in [] |
| 09:03.29 | FuriousGeorge | the top line |
| 09:03.31 | FuriousGeorge | the peername |
| 09:03.44 | TeePOG | ok |
| 09:03.48 | FuriousGeorge | make type friend, secret can be blank i believe |
| 09:04.35 | FuriousGeorge | X-Rob: ill give aussievoip money if you can correctly tell me why i can blindxfer to my parking extension but not atxfer |
| 09:04.43 | FuriousGeorge | on one of my servers :) |
| 09:04.46 | TeePOG | ok, i made secret blank on phone and in iax.conf |
| 09:05.03 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 09:05.06 | FuriousGeorge | username in phone = [peername] in iax.conf |
| 09:05.09 | X-Rob | FuriousGeorge, ummm |
| 09:05.10 | FuriousGeorge | iax2 reload |
| 09:05.18 | X-Rob | atxfer with the phone, or with the feature? |
| 09:05.23 | X-Rob | featurecode |
| 09:05.36 | TeePOG | just did, rebooting phone |
| 09:05.51 | FuriousGeorge | X-Rob: i mean by dialing the parking extension |
| 09:05.52 | TeePOG | no joy |
| 09:06.02 | TeePOG | phone still says login failed |
| 09:06.03 | FuriousGeorge | TeePOG: loginfauked |
| 09:06.09 | TeePOG | indeed |
| 09:06.15 | FuriousGeorge | you tried with a secret befire? |
| 09:06.19 | TeePOG | yup |
| 09:06.19 | FuriousGeorge | before* |
| 09:06.47 | X-Rob | FuriousGeorge, you can't atxfer to the parkxtn |
| 09:06.53 | X-Rob | you have to use blindxfer |
| 09:07.01 | TeePOG | what exactly should the value of context= be? |
| 09:07.21 | X-Rob | (well, you can, but it's insanely stupid to do so, and it won't work as you expect) |
| 09:07.25 | FuriousGeorge | X-Rob: sure and i have metermaid patch applied so i get blinkey lights, but you dont get audio feedback with blindxfer |
| 09:07.38 | X-Rob | FuriousGeorge, you should. |
| 09:07.44 | X-Rob | All my customers do |
| 09:07.44 | FuriousGeorge | so if two people park at the same time who went where |
| 09:08.00 | X-Rob | they're all trained to 'park is ##70' |
| 09:08.01 | FuriousGeorge | TeePOG: is there anything special about that cybercafe's network |
| 09:08.12 | X-Rob | ## = blindxfer, and 70 =parkxtn |
| 09:08.58 | TeePOG | not really... mostly doze machines, a ipcop firewall and a PCLinuxOS mediaplayer machine besides the Mandriva box running asterisk |
| 09:09.04 | FuriousGeorge | whe i blind xfer, the cli says its plaingback 7, the user never hears it, and the call is destroyed before the next digit is played by the server |
| 09:09.31 | FuriousGeorge | TeePOG: then heres what i suggest, you seemed to have no problem getting the last client installed, do it again on another box |
| 09:09.50 | X-Rob | FuriousGeorge, how exactly are you doing a blindxfer? |
| 09:09.55 | X-Rob | are you using a phone feature? |
| 09:10.18 | X-Rob | because that sounds like your phone is doing the blind xfer, not * |
| 09:10.35 | FuriousGeorge | X-Rob: on my snoms i simply hit the xfer button, enter extension, hit send (check button) |
| 09:10.39 | TeePOG | i did, and was putting in its details into sip.conf just now |
| 09:10.46 | X-Rob | FuriousGeorge, bingo, that's your problem |
| 09:10.59 | X-Rob | but I see what you want to do now |
| 09:11.10 | X-Rob | set the blind xfer button to be '##70' |
| 09:11.13 | FuriousGeorge | X-Rob: not so fast :) |
| 09:11.20 | X-Rob | _not_ xtn 70 |
| 09:11.20 | FuriousGeorge | from zap channels i cant call the parking extension |
| 09:11.35 | FuriousGeorge | though i suspected my phones were at fault before |
| 09:11.53 | FuriousGeorge | not that any mention is made on the wiki about that, despite mounds of documentation |
| 09:12.03 | X-Rob | are you including parkexten or whatever the magic context is? |
| 09:12.17 | FuriousGeorge | but the cli doesnt even react, same thing when i call the extension directly for an at-xfer |
| 09:12.24 | FuriousGeorge | X-Rob: yeah |
| 09:12.48 | X-Rob | 100% definately works for me, my home system is a TDM4 |
| 09:12.52 | X-Rob | so I'm guessing it's a config option |
| 09:13.00 | TeePOG | just to be sure FuriousGeorge: in extensions.conf, i would now put exten => 102,1,dial(sip/xten2)\ |
| 09:13.04 | FuriousGeorge | i got three other serers it works on with both zap and other sip clients |
| 09:13.07 | TeePOG | delete last \ |
| 09:13.25 | FuriousGeorge | TeePOG: im impressed, BUT what [context] do you put that in and why |
| 09:13.51 | TeePOG | i'm still unsure of what you mean by context, sorry, i must have missed something |
| 09:14.04 | TeePOG | [context] i mean |
| 09:14.26 | FuriousGeorge | extensions.conf takes extensions and contexts. contexts are what they litterally sound like. they start a section and live inside [] |
| 09:14.37 | FuriousGeorge | you only have two extensions right now |
| 09:14.39 | X-Rob | TeePOG, do you know any programming languages, and if so, which? |
| 09:14.43 | FuriousGeorge | you can call your context whatever |
| 09:14.47 | FuriousGeorge | [softphones] |
| 09:14.52 | FuriousGeorge | one word though |
| 09:14.58 | FuriousGeorge | [internal] |
| 09:15.18 | FuriousGeorge | [phones] probably makes most sense at this point |
| 09:15.19 | TeePOG | X-Rob, to be frank, not much... i come from a background of journalism and only recently getting into linux etc |
| 09:15.59 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 09:16.04 | TeePOG | ok FuriousGeorge... so all my exten => lines i created should be inside a context like [softphones] ? |
| 09:16.10 | X-Rob | TeePOG, that makes it a bit harder. A context is a 'group' of things. In the extensions.conf file, it's defined as a word enclosed in square brackets [foo] |
| 09:16.14 | TeePOG | i mean, under that heading |
| 09:16.22 | salviadud | is anyone connected to FWD right now? |
| 09:16.32 | salviadud | i'm getting registration refused thingies. |
| 09:16.36 | *** join/#asterisk vlt (n=daniel@dslb-088-073-215-164.pools.arcor-ip.net) |
| 09:16.40 | TeePOG | i'm teaching myself python, X-Rob, if that helps? |
| 09:17.11 | FuriousGeorge | TeePOG: so far so good, but how do you tell the server, that calls FROM this phone belong to a certain context of calls |
| 09:17.17 | X-Rob | TeePOG, uh, I doin't know python. But it's like a function, or subroutine. |
| 09:17.34 | X-Rob | (.. a context is like a ...) |
| 09:17.52 | TeePOG | i see X-Rob, functions i know a little about by now |
| 09:18.08 | X-Rob | TeePOG, cool. So you're aware that they're totally seperate things. That's what a context is. |
| 09:18.19 | TeePOG | in iax.conf and sip.conf FuriousGeorge? |
| 09:18.32 | FuriousGeorge | right |
| 09:18.35 | FuriousGeorge | context= |
| 09:19.12 | TeePOG | so in the function [softphones] i would define my sip clients' usernames as subroutines? |
| 09:19.13 | TeePOG | ok |
| 09:19.27 | FuriousGeorge | a context is even simpler than a function. a macro is more like a function. a context is just that. a context where the dialing of this extension makes sense. everything belongs to a context. inbound calls, local calls, outbound calls. |
| 09:19.40 | TeePOG | under each username, then, i put context=softphones ? |
| 09:19.50 | FuriousGeorge | so your peer needs to belong to a context with extensions |
| 09:19.55 | FuriousGeorge | TeePOG: right |
| 09:20.03 | TeePOG | so if it has no context, that call can't go anywhere ...? |
| 09:20.13 | X-Rob | exactly! |
| 09:20.20 | X-Rob | (well, it does default to [default]) |
| 09:20.24 | FuriousGeorge | right, because there is no context in which this phone dialing a number does anything. litterally |
| 09:20.32 | FuriousGeorge | X-Rob: true |
| 09:20.33 | X-Rob | so we, in freepbx, put [default] exten => _.,1,Playback(ss-noservice) |
| 09:21.02 | salviadud | is somebody not trying to register to FWD? please, raise your hand. |
| 09:21.13 | FuriousGeorge | X-Rob: freepbx configuration very similar to asterisk configuration |
| 09:21.20 | TeePOG | ok... literally a dead-line mp3 X-Rob? |
| 09:21.49 | X-Rob | FuriousGeorge, freePBX is a web gui for configuring asterisk |
| 09:21.58 | X-Rob | so, it's _Exactly_ the same! 8) |
| 09:22.16 | salviadud | forgot it about it. i got registered. |
| 09:22.31 | FuriousGeorge | TeePOG: hes saying that if you dial an extension that doesnt exist in your context (because for instance you dont have any context) the server will put it in the default context (if you have one) |
| 09:22.51 | TeePOG | which you then define an event for the default context |
| 09:22.52 | FuriousGeorge | and if it cant find that context in default it will go to extension s in context default |
| 09:23.09 | FuriousGeorge | TeePOG: there you go |
| 09:23.23 | FuriousGeorge | X-Rob: im confisuing it with openpbx |
| 09:23.33 | FuriousGeorge | X-Rob: so a@h is now freepbx |
| 09:23.35 | FuriousGeorge | ? |
| 09:23.42 | FuriousGeorge | or are they separate entities? |
| 09:24.20 | X-Rob | A@H is a bunch of stuff that includes freepbx |
| 09:24.35 | FuriousGeorge | gotcha |
| 09:24.53 | TeePOG | is there a package like that for mandriva, i wonder? |
| 09:25.25 | TeePOG | ok, all this is awesome, but my clients and phone still don't want to connect |
| 09:25.51 | FuriousGeorge | so TeePOG think of it this way: ideally you want to have a bunch of contexts that have similar extensions. like a context of calls for [internal_calls}, and maybe one for {outbound_calls] and [inbound_calls] |
| 09:26.33 | TeePOG | ok... |
| 09:26.43 | FuriousGeorge | then you have other contexts which your peers belong to. in these contexts you include the ones with extensions, thereby having different levels of users. you can have certain peers belong to [super_users], and in that context you include everything |
| 09:26.49 | X-Rob | TeePOG, uh. Realisticly, if you're new to linux and asterisk and stuff, download trixbox. |
| 09:26.52 | FuriousGeorge | then you have a contect called [mailroom] |
| 09:26.55 | X-Rob | it's an ISO that installs and sets up everything for you |
| 09:26.59 | X-Rob | HOWEVER |
| 09:27.02 | FuriousGeorge | and there you include=> internal_calls only |
| 09:27.15 | X-Rob | You're 99% of the way there now, by the sound of it. Keep perservering 8) |
| 09:27.42 | TeePOG | ok... i will look into trixbox for another site X-Rob, thanks! |
| 09:28.03 | TeePOG | but this is all really interesting and not too difficult once you can talk to someone about it |
| 09:28.19 | X-Rob | TeePOG, when/if you do, join #freepbx, and they'll tell you how to get the latest version and remove all the extra cruft it sticks in there |
| 09:28.33 | TeePOG | ok X-Rob... *log* |
| 09:29.04 | X-Rob | [ext-test] |
| 09:29.04 | X-Rob | include => ext-test-custom |
| 09:29.04 | X-Rob | exten => 7777,1,Goto(from-pstn,s,1) |
| 09:29.04 | X-Rob | exten => 666,1,Goto(ext-fax,in_fax,1) |
| 09:29.04 | X-Rob | exten => h,1,Macro(hangupcall,) |
| 09:29.11 | X-Rob | There's a nice simple context there, TeePOG |
| 09:29.59 | X-Rob | Do you understand what those 5 lines do? |
| 09:30.25 | TeePOG | i wonder FuriousGeorge... should I perhaps set up my asterisk machine as the dhcp machine? but it shouldn't make a difference since all my phones are static IPs |
| 09:30.32 | TeePOG | erm, no... i don't know |
| 09:30.42 | X-Rob | OK, line 1 defines the context |
| 09:30.56 | X-Rob | line 2 includes another context called [ext-test-custom] |
| 09:31.14 | X-Rob | line 3 says that if you dial 7777, go to the context [from-pstn] exten => s,1,Dosomething |
| 09:31.27 | TeePOG | right... so ext-test-custom is part of the [ext-test] context |
| 09:31.35 | X-Rob | Exactly |
| 09:31.42 | X-Rob | and _that_ is how you start doing funky things |
| 09:31.53 | X-Rob | so you have [extens] exten => 100,1,Dial(SIP/100) |
| 09:31.58 | X-Rob | exten => 101,1,Dial(SIP/101) |
| 09:32.02 | TeePOG | so, go the the exten => s,1,etc subroutine of from-pstn |
| 09:32.08 | FuriousGeorge | this illustrates my point i was making before http://pastebin.ca/76927 . i add users to those main general contexts, then i define acutaly phonenumbers by their type |
| 09:32.16 | X-Rob | and include => extens |
| 09:32.55 | TeePOG | what do you mean their type? user,peer,friend? |
| 09:33.01 | X-Rob | you can have [external] exten => _9.,1,Dial(SIP/VoIP-Provider/${EXTEN:1}) |
| 09:33.31 | TeePOG | sheesh, the syntax is hard |
| 09:33.38 | TeePOG | but it's not impossible |
| 09:33.39 | X-Rob | nah, it's not |
| 09:33.45 | TeePOG | like learning Russian |
| 09:33.48 | X-Rob | suddenly you'll understand it and it'll all make sense 8) |
| 09:33.54 | TeePOG | [which is easier than it looks] |
| 09:33.55 | X-Rob | (which, I s'pose, yes, is like learning russian 8) |
| 09:34.10 | X-Rob | OK |
| 09:34.13 | X-Rob | _9. |
| 09:34.24 | X-Rob | The _ means 'this is a wildcard match' |
| 09:34.27 | X-Rob | not a specific number |
| 09:34.35 | TeePOG | ok, like ? or * |
| 09:34.38 | X-Rob | No |
| 09:34.45 | TeePOG | "fill in the blank" |
| 09:34.45 | X-Rob | it's the _indication_ that the following stuff is a wildcard |
| 09:34.46 | TeePOG | ? |
| 09:34.54 | TeePOG | ok... |
| 09:34.56 | FuriousGeorge | (because dont forget some clients can just as easily dial letters) |
| 09:35.06 | TeePOG | gotcha |
| 09:35.10 | X-Rob | and '.' is like * |
| 09:35.16 | X-Rob | which is 'nothing, or anything' |
| 09:35.29 | X-Rob | so _9. will match 9, 9123, or 9000000000 |
| 09:35.31 | TeePOG | so, if you dial a 9 with anything after it... |
| 09:35.32 | FuriousGeorge | (and a confusing PBX) |
| 09:35.41 | TeePOG | indeed |
| 09:35.42 | FuriousGeorge | it matches that extensions |
| 09:35.50 | FuriousGeorge | IF |
| 09:36.06 | TeePOG | the number starts with a 9 |
| 09:36.06 | TeePOG | ? |
| 09:36.08 | FuriousGeorge | your phone belings to a context where that extension exists |
| 09:36.11 | X-Rob | (Oh, and it'll also match '9FRED' and '9abc-123' |
| 09:36.12 | X-Rob | ) |
| 09:36.20 | FuriousGeorge | lol |
| 09:36.33 | FuriousGeorge | http://pastebin.ca/76927 |
| 09:36.36 | FuriousGeorge | check that out |
| 09:36.46 | TeePOG | ok |
| 09:36.57 | FuriousGeorge | you see how i would add users to "toll", "notoll" and "internal" |
| 09:37.01 | FuriousGeorge | and in that CONTEXT |
| 09:37.11 | FuriousGeorge | it kinda makes sense what context they are allowed to dial |
| 09:37.31 | FuriousGeorge | "toll" users are like super users xuz they are allowed to rack up the phone buills |
| 09:37.53 | X-Rob | 1-900-HOT-XROB |
| 09:37.54 | X-Rob | call now! |
| 09:38.03 | FuriousGeorge | (i shoulda said "it kinda makes sense what EXTENSION they are allowed to dial" |
| 09:38.09 | FuriousGeorge | ) |
| 09:38.17 | TeePOG | ok... but isn't it easy to cross-reference two users? |
| 09:38.19 | TeePOG | erm |
| 09:39.06 | TeePOG | it's still a bit difficult to parse |
| 09:39.11 | TeePOG | but i'm getting there |
| 09:39.45 | FuriousGeorge | think about it [intenal] callers are the bottom of the barrel, they are allowed to call the extensions defined in emergency (just 911), parked (calls that are on hold), and inside (local extensions) |
| 09:40.08 | FuriousGeorge | notoll can call local extensions too |
| 09:40.37 | FuriousGeorge | and local numbers |
| 09:40.55 | TeePOG | and internal can't even call local |
| 09:41.09 | FuriousGeorge | local in that case means down the street |
| 09:41.17 | FuriousGeorge | not to be confused with internal |
| 09:41.21 | FuriousGeorge | which is in the building |
| 09:41.28 | FuriousGeorge | see its all about the context |
| 09:41.40 | TeePOG | outside, to the same phone company exchange, and back |
| 09:41.42 | TeePOG | ok |
| 09:41.53 | TeePOG | almost like iptables |
| 09:41.58 | TeePOG | which i'm also learning atm |
| 09:42.15 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 09:42.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:42.36 | smackus | the module for cdr logging to mysql is cdr_mysql, right? |
| 09:42.40 | smackus | .so |
| 09:42.58 | FuriousGeorge | calls that the phone comapny calls local, which are sometimes free depending on your plan. it makes sense to have a context for those. peers that belong to [outbound] include the context "toll" and if you look at the bottom, i define one extension in toll |
| 09:43.20 | FuriousGeorge | well i define more, but i show one |
| 09:43.32 | smackus | so if i have installed asterisk-addons, i should have a module named cdr_mysql.so. Is that correct? |
| 09:43.38 | FuriousGeorge | which is 411, becaue that costs money, so you gotta belong to the outbound callers to get that |
| 09:43.47 | TeePOG | so they can only call one other city if you only define only one extension...? |
| 09:44.06 | FuriousGeorge | smackus: sounds reasonable to me, but you can always look at the make output to be sure |
| 09:44.22 | X-Rob | smackus, if you have mysql-devel installed, yes |
| 09:44.30 | FuriousGeorge | no 411 in that case is a specific phone number |
| 09:44.38 | FuriousGeorge | you may not be in the states |
| 09:44.55 | FuriousGeorge | but that is for "directory assistance" or "information" around here. you know, to get other listings |
| 09:44.59 | FuriousGeorge | that costs money |
| 09:45.04 | FuriousGeorge | so its in the context toll |
| 09:45.20 | FuriousGeorge | which only users who are in context=outbound have access to |
| 09:45.20 | smackus | oops, cdr_addon_mysql.so |
| 09:45.27 | X-Rob | smackus, if you have mysql-devel installed, yes |
| 09:45.28 | TeePOG | oh i see |
| 09:45.34 | X-Rob | ^^^ that's your issue, most probably. |
| 09:46.18 | FuriousGeorge | TeePOG: if you really see that is 90% of the battle |
| 09:46.21 | FuriousGeorge | mabe less :) |
| 09:46.26 | FuriousGeorge | but its a big chunk |
| 09:46.47 | TeePOG | i get the concept, but the specifics are still beyond me for the moment |
| 09:46.48 | FuriousGeorge | now you need to understand how extensions work. read up on it |
| 09:46.54 | TeePOG | indeed |
| 09:47.18 | FuriousGeorge | if you can grasp python, it is just a matter of new syntax. its way simpler than pythong. |
| 09:47.18 | TeePOG | this places many of the documentation into perspective |
| 09:47.58 | TeePOG | now it's just a matter of getting my softphones to connect |
| 09:48.14 | TeePOG | so strange that they don't want to |
| 09:48.56 | FuriousGeorge | exten =>101,1,wait(3) |
| 09:48.56 | FuriousGeorge | exten =>101,2,dial(sip/xlite1) |
| 09:48.56 | FuriousGeorge | that does just what you think it would. the 1 and 2 are priorites, commands follow that order |
| 09:49.39 | FuriousGeorge | just sit in the cli and "show application blah" you can tab complete. that will give you the available commands and then the syntax |
| 09:50.13 | TeePOG | ok... trying that now |
| 09:50.25 | FuriousGeorge | and read the book |
| 09:50.28 | FuriousGeorge | ~thebook |
| 09:50.30 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:51.23 | FuriousGeorge | me dreams about contexts, extensions, and priorites... even then it doesnt quite work right |
| 09:51.28 | FuriousGeorge | :) |
| 09:51.34 | FuriousGeorge | i gotta sleep now |
| 09:51.37 | TeePOG | that book hates me, or i it, i don't know... blasphemy, i know... but it seems to concentrate on signing up with a PWD server |
| 09:51.53 | FuriousGeorge | FWD? |
| 09:52.08 | TeePOG | thanks for the help FuriousGeorge |
| 09:52.13 | FuriousGeorge | you should chekc out the other 455 pages |
| 09:52.15 | TeePOG | Free World Dialup |
| 09:52.16 | TeePOG | hmm |
| 09:52.38 | TeePOG | thanks, will soldier on |
| 09:52.50 | FuriousGeorge | one more thing |
| 09:52.54 | TeePOG | yes? |
| 09:52.55 | FuriousGeorge | like i stated in the beginning |
| 09:53.04 | FuriousGeorge | get a free acount with a voip provider |
| 09:53.08 | FuriousGeorge | so you can call 800 numbes |
| 09:53.12 | FuriousGeorge | numbers |
| 09:53.18 | TeePOG | even from SA? |
| 09:54.14 | FuriousGeorge | yeah, i dont see why not. US and Brittish 800#s i beleive. if the book concentrates on using FWD its because eventually you are gonna want to use your PBX to GET a phone call from someone |
| 09:54.14 | TeePOG | damn, didn't know that |
| 09:54.24 | FuriousGeorge | or call outside your network |
| 09:54.35 | TeePOG | actually, we're getting a digium card tomorrow |
| 09:54.40 | TeePOG | for analog phone line |
| 09:54.50 | TeePOG | centralising phone bills etc |
| 09:55.00 | TeePOG | for 3 adjacent shops |
| 09:55.07 | FuriousGeorge | right, but im saying the book needs a free example its users can try that doesnt endorse any voip company |
| 09:55.09 | FuriousGeorge | hence FWD |
| 09:55.18 | TeePOG | i see... clever |
| 09:55.27 | TeePOG | will mosey over there directly |
| 09:55.39 | FuriousGeorge | check out gizmoproject.com |
| 09:55.43 | FuriousGeorge | or sipphone.com |
| 09:55.52 | FuriousGeorge | they let you call the us toll frees with out paying money |
| 09:56.00 | FuriousGeorge | later all |
| 09:56.08 | TeePOG | bye FuriousGeorge, sleep well |
| 10:03.50 | TeePOG | thanks X-Rob |
| 10:03.59 | X-Rob | NP |
| 10:04.11 | *** join/#asterisk littleball (n=littleba@cm52.epsilon174.maxonline.com.sg) |
| 10:04.21 | TeePOG | see you soon, ciao |
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| 10:36.09 | rob0 | gizmoproject.com looks interesting, but I only have a "non-supported" Linux to use. Looks like I have to run their software to sign up, then I can use my own * setup? |
| 10:42.16 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 10:43.32 | rob0 | Anyway, can anyone here suggest which Gizmo download might best be convinced to work in a recent Slackware? I was thinking either Fedora or Debian. |
| 10:51.01 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 10:51.53 | rob0 | aha, it appears that sipphone.com provides the same service without the Gizmo softphone. |
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| 10:59.59 | saftsack | hi |
| 11:00.11 | saftsack | what is the cheapest digium card with hardware echocancellation? |
| 11:10.10 | saftsack | is noone here? |
| 11:13.29 | tehmaze | are there any example configs to allow inbound SIP calls? I have this freenumber.nl account which allows me to configure what sip proxy the call should be redirected to |
| 11:36.49 | *** join/#asterisk DrData (n=michael@pD95FD450.dip.t-dialin.net) |
| 11:38.50 | smackus | is it possible on a t1 (e&m wink) to get the configuration of the zapata or zaptel.conf files off in such a way that only 1 dns digit is provided, even if the carrier is sending 4 digits? |
| 11:39.13 | smackus | == Unknown extension '2' in context 'global_crossing' requested |
| 11:39.18 | smackus | dns is 2098 |
| 11:39.27 | smackus | it came through once, but never again. |
| 11:39.32 | smackus | i do not remember changing anything |
| 11:48.43 | *** join/#asterisk matush (n=edo@edo.sh.cvut.cz) |
| 11:48.50 | matush | hi |
| 11:49.33 | matush | I'm looking for a little help in configuring my first asterisk |
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| 11:50.53 | mrdigital | hey CunningPike? |
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| 12:06.50 | saftsack | hi |
| 12:07.01 | saftsack | is someone of the * developer team here? |
| 12:08.36 | smackus | how do i specify where to send a queued call after the timeout period? |
| 12:08.48 | smackus | exten => 90000,4,Queue(evolution|t|8131||45) |
| 12:08.59 | DrData | Does anybody know how to get the Fritz Box working as ATA with asterisk? |
| 12:10.28 | smackus | is that correct? cuz all it does after 45 seconds is hangs up on the caller |
| 12:11.49 | saftsack | are there any new informations about digiums b410p card? |
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| 13:02.52 | PakiPenguin | hello :D |
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| 13:24.13 | frenzy | hi |
| 13:24.20 | frenzy | need some help |
| 13:24.30 | frenzy | I've got two x100p.com FXO cards |
| 13:24.40 | frenzy | I want to setup both on the same box |
| 13:25.01 | frenzy | how do I configure it on zaptel.conf ? |
| 13:25.23 | frenzy | fxsks=1 work but only one card is registerd |
| 13:25.44 | frenzy | fxsks=1-2 creates an error |
| 13:33.04 | frenzy | ? |
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| 13:47.04 | SheriF_WorK | can't belivie it 4 hours looking for softphone supports G729 G723 and using SIP INFO in DTMF and can't find !! |
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| 13:58.40 | _4d4m_ | frenzy: what is the error created by fxsks=1-2? |
| 13:59.03 | frenzy | ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 13:59.31 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.78.Dial1.SanJose1.Level3.net) |
| 13:59.41 | frenzy | the two are identical cards |
| 13:59.52 | frenzy | is it possible modprobe is not catching the other one? |
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| 14:02.18 | _4d4m_ | frenzy: have you checked your /proc/interrupts and /var/log/messages for any errors? |
| 14:02.55 | frenzy | not interuppts |
| 14:03.05 | frenzy | 193: 2636635 IO-APIC-level wcfxo |
| 14:05.09 | frenzy | Jul 2 16:18:20 inet-gw kernel: Zapata Telephony Interface Registered on major 196 |
| 14:05.10 | frenzy | Jul 2 16:18:20 inet-gw kernel: Zaptel Version: 1.2.6 Echo Canceller: KB1 |
| 14:05.10 | frenzy | Jul 2 16:18:20 inet-gw kernel: ACPI: PCI Interrupt 0000:01:01.0[A] -> GSI 22 (level, low) -> IRQ 193 |
| 14:05.10 | frenzy | Jul 2 16:18:20 inet-gw kernel: wcfxo: DAA mode is 'FCC' |
| 14:05.10 | frenzy | Jul 2 16:18:20 inet-gw kernel: Found a Wildcard FXO: Wildcard X100P |
| 14:05.10 | frenzy | Jul 2 16:18:20 inet-gw kernel: ACPI: PCI Interrupt 0000:00:1d.7[D] -> GSI 23 (level, low) -> IRQ 201 |
| 14:08.29 | _4d4m_ | frenzy: certainly sounds like either a system or hardware issue, but nothing stands out to me as an obvious source of the problem, sorry |
| 14:08.46 | _4d4m_ | have you had 1 card working |
| 14:08.54 | _4d4m_ | before then adding the second card? |
| 14:08.55 | frenzy | yep |
| 14:09.03 | frenzy | smooth.. |
| 14:09.42 | _4d4m_ | have you swapped the cards over to make sure the second card is ok? |
| 14:09.51 | _4d4m_ | using a single card config i mean |
| 14:10.39 | frenzy | nop |
| 14:12.12 | _4d4m_ | frenzy: would be a good way to rule out hardware problems with the card |
| 14:12.21 | *** join/#asterisk skraelings001 (n=skraelin@201.230.239.67) |
| 14:12.37 | skraelings001 | hello |
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| 14:28.56 | rob0 | freebsd_fan: try genzaptelconf |
| 14:29.00 | rob0 | sorry |
| 14:29.18 | rob0 | frenzy: try genzaptelconf |
| 14:30.01 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 14:31.34 | *** join/#asterisk ionix (i=ionix@p5008-ipbfp04miyazaki.miyazaki.ocn.ne.jp) |
| 14:31.55 | ionix | What is the nice way to do TTS beside festival (Let's be honest, festival is horrible) |
| 14:32.20 | TheCops | salut ionix |
| 14:35.52 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 14:35.55 | Assid | heya |
| 14:36.12 | Assid | okay fxo cards are the ones you use for outgoing/incoming calls over POTS right ? |
| 14:36.20 | TheCops | yup |
| 14:36.41 | Assid | fxs is if you wanna give your own dialtone.. |
| 14:36.43 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 14:36.50 | Assid | i keep getting mixed up .. dunno why |
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| 14:37.58 | Assid | gonna ask them to pickup the tdm400p for me to playwith |
| 14:38.35 | rob0 | fxO= connect line from phone Office ; fxS= connect line from Station (if you think of the word "Station" in terms of a client device, like a computer "workstation".) |
| 14:38.53 | acehunky | anyone knows of any fix for callerid detection for india ? |
| 14:39.06 | acehunky | i tried with bugs.digium.com for pstn - callerid .. but doesnt work :( |
| 14:39.57 | Assid | rob0: got a few pstn lines .. they are just sitting idle.. so they wanna use them as well to reduce net traffic |
| 14:40.39 | Assid | how much is the tdm400p with 4fxo anwyays? |
| 14:40.47 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 14:40.48 | TheCops | 299$US |
| 14:40.51 | TheCops | if I remember |
| 14:41.03 | Assid | doable.. |
| 14:41.16 | ionix | TheCops hehe lo |
| 14:42.02 | Assid | whcih one is which ? http://www.digium.com/en/wheretobuy/digiumdirect/products.php?category_id=17 ? |
| 14:42.38 | Dimitripietro | Anyone could help me compiling the sangoma drivers ? |
| 14:43.04 | TheCops | God this is 421$ ouch. |
| 14:43.10 | Assid | yeah |
| 14:43.17 | Assid | why the hell is it soo expensive? |
| 14:44.22 | TheCops | I pay it 280$US |
| 14:44.24 | TheCops | for TDM04B |
| 14:44.26 | TheCops | 4 FXO |
| 14:44.48 | TheCops | Assid, do you have your own compagny ? |
| 14:44.48 | Assid | where from? |
| 14:45.00 | TheCops | www.netxusa.com |
| 14:45.07 | TheCops | need to be reseller |
| 14:45.10 | Dovid | Who dosent have thier own company today ;) |
| 14:45.31 | Dovid | TheCops: what do u want to resell ? |
| 14:45.35 | acehunky | assid where r u located ? |
| 14:45.40 | Dovid | Most of the these "companies" are resellers |
| 14:45.49 | Assid | me personally .. india |
| 14:46.14 | acehunky | aaha .. so i can provide you with the card .. we are official digium resellers here |
| 14:46.16 | Dovid | Assid: what are you tryin to sell ? |
| 14:46.16 | TheCops | Dovid, yeah, but they have good price, not many difference if I'm going directly to the manufacturer |
| 14:46.26 | Dovid | oh |
| 14:46.35 | Dovid | For cards, thought u meant for services |
| 14:46.39 | TheCops | no |
| 14:46.43 | acehunky | TheCops Digium doesnt sell OEM cards directly ! |
| 14:46.50 | acehunky | its sold only via resellers |
| 14:46.56 | TheCops | acehunky, I've verified for Polycom, sorry hehe |
| 14:47.06 | Assid | acehunky: entrux? |
| 14:47.09 | Dovid | Do they do drop shipping ? Seems most companies do drop shipping |
| 14:47.10 | acehunky | thats right |
| 14:47.21 | Assid | i acutally need it in the US |
| 14:47.23 | *** join/#asterisk ktsaou (n=kvirc@ppp.tsaousis.gr) |
| 14:47.34 | Assid | so they gonna pick it up |
| 14:47.42 | Assid | would like to know how much are the polycoms here tho |
| 14:47.59 | *** join/#asterisk ktsaou (n=kvirc@ppp.tsaousis.gr) |
| 14:47.59 | acehunky | Polycom guy is gonna get back to me tommorow .. |
| 14:48.15 | acehunky | they dont have too many obvious resellers here .. atleast not seen from their site |
| 14:48.28 | acehunky | and not any of them would be interested to sell one or two pieces : |
| 14:48.29 | acehunky | :( |
| 14:48.48 | acehunky | minimum 50+ |
| 14:49.14 | Assid | whats the deal with the law for voip now? im still not clear if your IVR is allowed to be on voip |
| 14:49.24 | Assid | even though its connected to a pstn line |
| 14:51.00 | NotJohnDavid | I have a grandstream gxp-2000 that i just upgraded the firmware on. now it hangs when i try to reboot it and the speakerphone doesn't work :( |
| 14:51.09 | Assid | acehunky: did you ever get the caller id to work on indian pstn ? |
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| 14:52.48 | Assid | ace ? |
| 14:52.51 | Assid | oh well |
| 14:53.08 | acehunky | naa .. some guy named abhi did the bug report and patch .. but hasnt done any further check |
| 14:53.18 | acehunky | on the posts that i did .. after incorporating his patch |
| 14:54.13 | Assid | damn.. |
| 14:54.19 | Assid | let me know if you get anything |
| 14:54.36 | acehunky | he he he .. none of digium cards are going to work in that case :P |
| 14:54.53 | Assid | well.. that card.. i need in the US .. not here |
| 14:54.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:55.02 | Assid | sup dennis |
| 14:55.10 | Assid | wonder wheres tkd today |
| 15:05.48 | Dimitripietro | Anyone could help me compiling the sangoma drivers ? |
| 15:06.41 | *** join/#asterisk saftsack (n=saftsack@p54A7EC91.dip.t-dialin.net) |
| 15:07.01 | saftsack | hi |
| 15:07.07 | saftsack | are there some news from digiums b410p card? |
| 15:10.16 | acehunky | Dimitripietro what help you need with sangoma ? |
| 15:12.18 | Dimitripietro | <acehunky> I'm unable to compil the driver |
| 15:12.49 | Dimitripietro | Getting an error concerning udev |
| 15:12.54 | acehunky | what kernel version ? |
| 15:13.05 | Dimitripietro | 2.6.17 |
| 15:13.29 | acehunky | wow :) |
| 15:13.36 | acehunky | try little older :P |
| 15:13.46 | acehunky | 2.6.16 |
| 15:13.52 | acehunky | even numbers are usually stable .. |
| 15:14.13 | Dimitripietro | I did, I upgraded because it wasn't compiling with 2.6.16 |
| 15:15.05 | acehunky | hmm . |
| 15:15.06 | saftsack | is there a huge difference beetwen dsp based echocancelcards and software echocancelled cards? |
| 15:15.14 | acehunky | which card is it ? |
| 15:15.18 | Dimitripietro | i'm getting an error something like udev permission |
| 15:15.21 | Dimitripietro | A200 |
| 15:16.44 | acehunky | yeah .. you would need to check with the changes which zaptel does under /etc/ directory |
| 15:16.49 | acehunky | for udev |
| 15:17.07 | acehunky | i dont have a redhat machine here to tell u the exact directory though |
| 15:19.08 | acehunky | wat distro u using ? |
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| 15:21.16 | PakiPenguin | hello everyone |
| 15:24.15 | saftsack | is nobody from digium here? |
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| 15:30.53 | Assid | is there a way to link to meetme rooms on 2 different servers and mux as 1 call |
| 15:33.26 | Dimitripietro | <acehunky> FC4 |
| 15:33.45 | Dimitripietro | "/etc/udev" |
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| 16:19.40 | frenzy | Is this syntax correct ? exten => _X.,1,SendDTMF(Zap/1/102#) |
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| 16:26.44 | Trazz | where can i find some example custom apps that are written for * ? |
| 16:29.27 | anonymouz666 | app_skel.c? |
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| 16:32.55 | frenzy | is is possible to have a wait on the same line? |
| 16:38.31 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:40.33 | frenzy | any other way to sendDTMF ? |
| 16:40.48 | frenzy | to ZAP interface? |
| 16:41.28 | frenzy | the sendDTMF doesnt work for me.. |
| 16:41.43 | frenzy | for some reason it wont go through |
| 16:41.57 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 16:42.44 | *** join/#asterisk coppice (n=chatzill@61.197.17.210.dyn.pacific.net.hk) |
| 16:48.35 | backblue | frenzy: if sendDTMF dont work, why whould any other way work? |
| 16:48.55 | backblue | try to solve the problem insted of avoiding it. |
| 16:49.14 | frenzy | as-in I have to authenticate with my phone provider by entering digits |
| 16:49.36 | frenzy | exten => _X.,1,SendDTMF(Zap/1/102#) |
| 16:49.36 | frenzy | exten => _X.,2,wait,4 |
| 16:49.37 | frenzy | exten => _X.,3,Dial(Zap/1/1234#) |
| 16:49.37 | frenzy | exten => _X.,4,wait,4 |
| 16:49.37 | frenzy | exten => _X.,5,Dial(Zap/1/${EXTEN}#) |
| 16:50.05 | frenzy | However the provider doesnt acknowlege the digits or something |
| 16:50.22 | frenzy | If is use Dial and punch in things manuall it works |
| 16:51.14 | *** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 16:51.22 | *** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 16:52.44 | frenzy | how does SendDTMF work?... Tone or Pulse? |
| 16:53.34 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 16:56.14 | *** join/#asterisk babyju___ (n=babyju@c-68-39-244-238.hsd1.nj.comcast.net) |
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| 17:02.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:03.16 | znoG | <PROTECTED> |
| 17:03.16 | znoG | <PROTECTED> |
| 17:03.16 | znoG | [Synopsis] |
| 17:03.19 | znoG | Sends arbitrary DTMF digits |
| 17:03.26 | znoG | <PROTECTED> |
| 17:03.27 | znoG | <PROTECTED> |
| 17:03.50 | znoG | one would think you use it like this: SendDTMF(1234) |
| 17:03.57 | *** join/#asterisk Tier_1 (n=tier@c-67-176-28-65.hsd1.co.comcast.net) |
| 17:04.27 | Tier_1 | anyone know of a good clock exten that loops using unixtime |
| 17:04.39 | Tier_1 | I know there was one out there |
| 17:04.43 | Tier_1 | but cant find it |
| 17:05.38 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 17:06.56 | frenzy | ZnoG: what would be the correct way of using it? for Zap out? |
| 17:07.50 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.78.Dial1.SanJose1.Level3.net) |
| 17:09.35 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:14.38 | _4d4m_ | frenzy: are you using Dial() before then trying to use SendDTMF()? |
| 17:14.52 | frenzy | Nop |
| 17:15.55 | _4d4m_ | so when you connect, you want to first send DTMF, then dial a number? |
| 17:16.19 | frenzy | yap |
| 17:16.48 | frenzy | ok I think I found the solution |
| 17:16.54 | frenzy | using w |
| 17:16.54 | znoG | frenzy: you smoke what exactly? |
| 17:16.56 | frenzy | to space out |
| 17:17.45 | frenzy | Zap/1/102#wwwwww1234#wwwwww#{EXTEN}#} |
| 17:17.58 | frenzy | seem to do the trick |
| 17:21.47 | babyju___ | Is there an analog to digital converter compatible with asterisk....Plug a regular pots line into the converter...the converter connects to asterisk via standard rj45/IP. This would be a good alternative to purchasing line cards. |
| 17:22.39 | saftsack | yes there are |
| 17:22.59 | Qwell | babyju___: fxo ata |
| 17:23.09 | saftsack | http://www.digium.com/en/products/hardware/s101i.php |
| 17:23.12 | babyju___ | saftsack can you point me to a url |
| 17:23.16 | babyju___ | thanks |
| 17:23.20 | Qwell | saftsack: That isn't an fxo |
| 17:23.25 | saftsack | yes taht is an fxs |
| 17:23.32 | Qwell | he doesn't want an fxs |
| 17:23.36 | saftsack | ooh you wrote pots line. didnt read that sry |
| 17:23.54 | Qwell | babyju___: look at the sipura spa3000 |
| 17:24.16 | anonymouz666 | anyone know if TDM cards support CallerID being sent DTMF? Or I need to convert to FSK? |
| 17:28.33 | TheCops | Someone know how to register polycom soundpoint 501/601 on the polycom extranet ? I'm entering serial, all the time I've got invalid serial. |
| 17:29.37 | SwK | TheCops: are you trying to do the reseller thing? |
| 17:29.54 | TheCops | no, just trying to register a new product |
| 17:29.58 | TheCops | IP601 and 501 dont work |
| 17:30.17 | SwK | thats par |
| 17:30.22 | SwK | i've seen that before |
| 17:30.41 | SwK | a while back they had a broken process on extranet for that and I had to call the, |
| 17:30.43 | SwK | them |
| 17:31.18 | TheCops | I just applied too for the reseller voip :) |
| 17:31.20 | TheCops | hehe |
| 17:34.45 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
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| 18:31.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
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| 18:50.53 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 18:51.04 | Assid | heya |
| 18:51.40 | Assid | okay.. anyone got the sip 1.6.6 for polycom? |
| 18:55.14 | *** join/#asterisk RoyK[de] (n=roy@83-65-165-114.residenzplatz.xdsl-line.inode.at) |
| 18:56.13 | _problem_ | Does anybody knows IRC channel for SIP Express Router? |
| 18:56.24 | Assid | i forgot to save my files before they reclaimed that box :( |
| 18:58.09 | brookshire | #ser maybe? |
| 18:58.45 | RoyK[at] | ~ser |
| 18:58.46 | jbot | i heard ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 18:59.14 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 18:59.19 | _problem_ | brookshire:thanks |
| 18:59.24 | RoyK[at] | jbot: ser is also an old secret method of obtaining a havoc of NAT problems |
| 18:59.26 | jbot | okay, RoyK[at] |
| 18:59.51 | RoyK[at] | jbot: ser is also at #ser |
| 18:59.53 | jbot | okay, RoyK[at] |
| 18:59.57 | RoyK[at] | ~ser |
| 18:59.58 | jbot | ser is, like, Sip Express Router - see http://www.iptel.org/ser/, or an old secret method of obtaining a havoc of NAT problems, or at #ser |
| 19:00.32 | _problem_ | but is there any channel for openser ? |
| 19:00.55 | Assid | hey royk: you wouldnt happen to have polycom's sip 1.6.6 would you |
| 19:00.56 | brookshire | #openser maybe? |
| 19:01.21 | _problem_ | brookshire: no it is not? |
| 19:01.38 | brookshire | i said maybe! |
| 19:01.54 | _problem_ | brookshire:ok thanks |
| 19:03.24 | Bert- | I really don't understand what 'is wrong with SIP and NAT, as there only some ports to open ... ? |
| 19:03.33 | RoyK[at] | _problem_: don't think so, but ser and openser is more or less the same. |
| 19:03.44 | Bert- | I never be annoyed with nat ... |
| 19:03.53 | RoyK[at] | i'm starting another project to drop SER, sitsotd.org |
| 19:06.03 | _problem_ | RoyK[at]: they were same ..but ser developers stoped its furthur development 2 years ago...now voice-system romania is started developing it with the name of openser...which can work..ser is impossible to work with. |
| 19:06.36 | Assid | bah this sucks.. i cant find the damn 1.6.6 sip software for polycom anywhere |
| 19:07.42 | RoyK[at] | _problem_: i'll try to come up with something useful with sitsotd quite soon. right now i'm travelling across europe and don't do too much coding :) |
| 19:08.50 | _problem_ | RoyK[at]: may i ask u where r u in europe,, i m also in europe? what is sitsotd? |
| 19:09.18 | RoyK[at] | _problem_: sitsotd.org |
| 19:09.42 | RoyK[at] | hm. wtf is .pt? |
| 19:10.45 | RoyK[at] | ah. portugal.... |
| 19:11.02 | _problem_ | yes |
| 19:12.04 | RoyK[at] | congrats, btw :) |
| 19:12.25 | _problem_ | RoyK[at]:for what ? |
| 19:12.49 | RoyK[at] | football :) |
| 19:13.40 | _problem_ | RoyK[at]: :-) ya ...it win over England..it was fantasitic game |
| 19:14.15 | RoyK[at] | it was indeed. and England played really well with one man down |
| 19:15.42 | Bert- | come on portugesh ... :) |
| 19:15.49 | _problem_ | RoyK[at]: ya England Played well...also in previous game Holland played well then Portugal...but ultimately portugal win the game... |
| 19:16.35 | RoyK[at] | _problem_: who's .pt going to meet now? |
| 19:16.37 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 19:17.04 | Bert- | Us |
| 19:18.05 | Bert- | but don't worry we'll win :) |
| 19:18.20 | _problem_ | RoyK[at]: pt with france |
| 19:18.21 | RoyK[at] | Bert-: who's "us"? |
| 19:18.24 | RoyK[at] | ah |
| 19:18.32 | Bert- | rance |
| 19:18.36 | Bert- | France |
| 19:18.40 | _problem_ | and germany with italy |
| 19:18.44 | RoyK[at] | Bert-: where in france do you live? |
| 19:18.55 | Bert- | City of Wine :) |
| 19:18.57 | RoyK[at] | wednesday, right? |
| 19:19.03 | Bert- | Bordeaux |
| 19:19.06 | RoyK[at] | Bert-: there are a few of them....... |
| 19:19.07 | RoyK[at] | ah |
| 19:19.07 | RoyK[at] | ok |
| 19:19.35 | Bert- | wednesday, 21h (Paris's hour) |
| 19:19.50 | Bert- | bourgone really good too :) |
| 19:20.07 | Bert- | more strong |
| 19:20.12 | RoyK[at] | Hautes Cotes de Baune, red, delicious |
| 19:20.18 | Bert- | yep |
| 19:20.47 | RoyK[at] | hm. perhaps I should go there soon :) |
| 19:20.49 | Bert- | try Clairet de Quinsac |
| 19:20.56 | Bert- | a pink one |
| 19:21.08 | Bert- | made just where I live :) |
| 19:21.14 | RoyK[at] | Rosé? |
| 19:21.22 | Bert- | exactly :) |
| 19:21.43 | RoyK[at] | *noting* |
| 19:22.15 | *** join/#asterisk N3GLV (n=n3glv@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 19:22.15 | Bert- | a good one is "Rosé de Madame" (made by Friend). But it is local wine, I suppose it 's difficult to find it out of France |
| 19:22.31 | RoyK[at] | I have an interrail ticket that lasts about three more weeks. perhaps I should come by :) |
| 19:22.55 | Bert- | yep |
| 19:23.06 | Bert- | come take some sun in South West of France |
| 19:23.07 | Bert- | : |
| 19:23.08 | Bert- | :) |
| 19:23.16 | N3GLV | there u go |
| 19:23.36 | N3GLV | I fart in your general direction |
| 19:23.42 | RoyK[at] | hehe |
| 19:23.43 | N3GLV | <wink> |
| 19:23.45 | Bert- | :) |
| 19:23.50 | RoyK[at] | ~fart N3GLV |
| 19:23.55 | jbot | ACTION farts in N3GLV's general direction |
| 19:24.03 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 19:24.17 | N3GLV | ---',-@ rose for Roy |
| 19:24.22 | RoyK[at] | Bert-: I don't think I've been to Bordoeaux... |
| 19:24.37 | RoyK[at] | spelling problems |
| 19:24.38 | RoyK[at] | whatever |
| 19:24.52 | Bert- | hmm |
| 19:24.54 | N3GLV | Bordeaux I think |
| 19:25.00 | RoyK[at] | yes |
| 19:25.02 | RoyK[at] | i know |
| 19:25.02 | Bert- | some french speaks english don't worry :) |
| 19:25.03 | N3GLV | me either |
| 19:25.07 | RoyK[at] | typing problems |
| 19:25.24 | N3GLV | u know why the main drag of Paris has trees on both sides? |
| 19:25.27 | RoyK[at] | Bert-: i know, that's why I don't go there too often. "some" <= all |
| 19:25.44 | Bert- | lol |
| 19:25.45 | pdunkel | Beaucoup des francais parlent anglais, mais il ne plaisent pas :) |
| 19:25.46 | Bert- | not all |
| 19:25.56 | RoyK[at] | quite a few of them iirc |
| 19:26.04 | RoyK[at] | germany is quite bad as well. |
| 19:26.10 | RoyK[at] | austria is better, though |
| 19:26.11 | Bert- | and we speak english with our famous and unique french accent :) |
| 19:26.15 | pdunkel | (Many french speak english, but they don't like it) |
| 19:26.43 | pdunkel | Why is germany bad and austria better? (Not that I disagree being austrian :) ) |
| 19:26.44 | Bert- | N3GLV, no I don't know |
| 19:27.03 | RoyK[at] | pdunkel: more people really speaks English here afaics |
| 19:27.08 | N3GLV | us americans are almost as notorious about ppl who don't speak american as the french/frrancais |
| 19:27.21 | N3GLV | Ahn young ha seyo |
| 19:27.32 | pdunkel | N3GLV: Even worse! |
| 19:27.34 | N3GLV | aka Konihi-wa but in Korean |
| 19:27.37 | RoyK[at] | Bert-: are you from austria? |
| 19:27.39 | N3GLV | ;-) |
| 19:27.48 | pdunkel | RoyK: Where are you at? |
| 19:27.56 | N3GLV | or Nehi (Mandarain) |
| 19:28.00 | Bert- | RoyK[at], no |
| 19:28.13 | RoyK[at] | salzburg |
| 19:28.17 | pdunkel | RoyK: I'm at Café Blaustern in Vienna right now. |
| 19:28.29 | Bert- | but I'was born near Germany ;) |
| 19:28.32 | pdunkel | What do you do for business. Anything asterisk related? |
| 19:28.45 | RoyK[at] | nothing. vacation. just found a bar with wlan |
| 19:28.52 | RoyK[at] | http://at.freewlan.org/ |
| 19:28.56 | RoyK[at] | fill in, please :) |
| 19:29.12 | pdunkel | RoyK: If yes mybe we should meet up some time (If you live in SBG) |
| 19:29.14 | RoyK[at] | prolly going to Graz tomorrow |
| 19:29.25 | RoyK[at] | I live in norway - vacation :) |
| 19:29.33 | pdunkel | Stop by Vienna, I'll invite you for Lunch |
| 19:29.48 | pdunkel | (Vienna is on the way) |
| 19:30.05 | RoyK[at] | erm... not from salzburg.... |
| 19:30.13 | RoyK[at] | look at the map :) |
| 19:30.27 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 19:30.27 | pdunkel | Anyhow. Where in Norway, Got some friends in Flateby (near Oslo) |
| 19:30.28 | RoyK[at] | but I'm probably going to Wien after Graz anyway |
| 19:30.31 | RoyK[at] | Oslo |
| 19:30.34 | RoyK[at] | central oslo |
| 19:30.43 | RoyK[at] | hehe; |
| 19:30.50 | RoyK[at] | Flateby is such a hole :) |
| 19:30.56 | pdunkel | Coolness. (Well if you stop by Vienna give me a ring. |
| 19:31.06 | RoyK[at] | msg me your number |
| 19:31.21 | pdunkel | All I saw of Flateby was their house before going on sailing down the coast. |
| 19:31.33 | anonymouz666 | love is in the air :) |
| 19:31.33 | anonymouz666 | lol |
| 19:31.35 | anonymouz666 | :P |
| 19:33.21 | Bert- | :( |
| 19:33.26 | Bert- | Paris is too cool :( |
| 19:33.35 | Bert- | really too much :( |
| 19:34.30 | anonymouz666 | I don't like France because they always win against Brazil |
| 19:34.48 | Bert- | héhé |
| 19:35.08 | Bert- | I was at one of my best friends's wedding yesterday |
| 19:35.27 | Bert- | goal was a big present for us ;) |
| 19:35.44 | Bert- | and you now why we win ? |
| 19:35.51 | RoyK[at] | it would be so nice to see germany and france in the final :) |
| 19:35.51 | RoyK[at] | hehe |
| 19:35.59 | RoyK[at] | total battle :D |
| 19:36.03 | Bert- | I'm sure France will in fianl with Germany |
| 19:36.06 | Bert- | but I thnik |
| 19:36.12 | Bert- | Germany will win :) |
| 19:36.13 | anonymouz666 | Germany have to win! |
| 19:36.16 | *** join/#asterisk rene- (n=rene@201.152.34.198) |
| 19:36.29 | Bert- | sorry for typo :p |
| 19:36.50 | RoyK[at] | Bert-: see msg |
| 19:36.51 | pdunkel | That would be really bad. The country hosting an event should never win. Although not proovable, it always smacks of cheating. |
| 19:36.58 | anonymouz666 | Henry/Zidane is our nightmare |
| 19:37.18 | pdunkel | So I am hoping for a final w/o Germany |
| 19:37.41 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 19:37.47 | nextime | france - italy |
| 19:37.53 | skraelings001 | i bout some soyo G668, but i don't found it to have call transfer capability, does anyone know how to trick over it? |
| 19:38.25 | *** join/#asterisk jwest- (n=jwest@83.110.124.122) |
| 19:38.30 | muppetmaster | Hello |
| 19:38.33 | *** part/#asterisk jwest- (n=jwest@83.110.124.122) |
| 19:38.47 | _problem_ | portugal is also having some chances people |
| 19:38.49 | muppetmaster | Recently bought a Nokia E61, works well for inbound calls with Asterisk over GPRS and WLAN/WiFi. |
| 19:38.53 | Bert- | pdunkel, you speak about 1998 ?? ;) |
| 19:39.03 | muppetmaster | But, having a problem with incoming calls and not a NAT issue. Getting this error: Jul 2 21:36:06 WARNING[5346]: chan_sip.c:3563 process_sdp: Error in codec string 'eo 0/1 RTP/AVP 103' |
| 19:39.03 | N3GLV | skraelings, u can transfer a call in progress wirh a # |
| 19:39.05 | Bert- | muppetmaster, |
| 19:39.08 | muppetmaster | Any ideas? |
| 19:39.11 | SpaceBass | muppetmaster, nice! i hear they are good devices |
| 19:39.17 | SpaceBass | muppetmaster, do they use STUN? |
| 19:39.19 | N3GLV | u need to have it turned on, of course |
| 19:39.21 | muppetmaster | SpaceBass They are, if only the inbound could work! |
| 19:39.24 | Bert- | I have a N60 or N61 I don't know |
| 19:39.31 | pdunkel | Bert-: For example. I don't really think the French cheated, but it when the hosting country wins it always leaves a bad taste. |
| 19:39.39 | muppetmaster | SpaceBass, as of now they have no NAT traversal capabilities (as best I know) but will soon from what Nokia says. |
| 19:39.39 | skraelings001 | N3GLV: how is that? |
| 19:39.47 | Bert- | I have to test it with asterisk, as we tried with a SIP proxy and sound was not really good |
| 19:40.00 | SpaceBass | muppetmaster, I have a linksys wip330 thats the same way...basically makes the device USELESS |
| 19:40.09 | skraelings001 | N3GLV: you mean in features.conf? |
| 19:40.18 | muppetmaster | SpaceBass Indeed. If this is the case, I am taking it back. |
| 19:40.25 | muppetmaster | If it really is a support/codec problem |
| 19:40.47 | Bert- | I not agree |
| 19:41.05 | Bert- | no bad taste for us :) |
| 19:41.43 | *** part/#asterisk N3GLV (n=n3glv@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 19:41.47 | Bert- | I don't thing Korea cheated |
| 19:41.49 | muppetmaster | Well, posted this at Nokia... http://discussion.forum.nokia.com/forum/showthread.php?p=219147#post219147 |
| 19:41.51 | Bert- | think |
| 19:41.54 | skraelings001 | N3GLV : [featuremap] , i tryed with #1 but didn't work |
| 19:42.19 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 19:42.35 | pdunkel | Bert-: Mais c'est seulement parceque tu es francais. Non? |
| 19:42.44 | Bert- | c'est possible :) |
| 19:43.06 | Bert- | pdunkel, where are you from .? |
| 19:44.31 | Bert- | hmm I should be still connected ... |
| 19:44.51 | *** join/#asterisk RoyK[at] (n=roy@83-65-165-114.residenzplatz.xdsl-line.inode.at) |
| 19:44.59 | pdunkel | Bert-: Je suis d'Autriche. But as any educated person should I do speak a couple of languages (Among them a little french) :) |
| 19:45.03 | RoyK[at] | <PROTECTED> |
| 19:45.25 | Bert- | okay |
| 19:45.38 | Bert- | a lot of ppl from Austria hereso ? |
| 19:47.07 | Bert- | I have a really stupid Idea : does someone ever tried to stream Tv throught Asterisk ? |
| 19:47.19 | RoyK[at] | dunno. usually, i don't see too much of them |
| 19:47.29 | RoyK[at] | Bert-: are you out of your mind? :) |
| 19:47.53 | RoyK[at] | Bert-: try using ffmpeg to pipe stuff into asterisk - might work |
| 19:49.11 | Bert- | hmm I could call our desk 's Asterisk from my personnal one, through internet and then use meetme to let some employees see matches :p |
| 19:49.31 | Assid | err.. whats the changelog between 2.6.1 and 2.6.2 bootrom for polycoms? |
| 19:50.29 | RoyK[at] | 2.6.x, is that a linux kernel? |
| 19:50.32 | Bert- | "real time" encoding with vlc ... |
| 19:50.53 | Assid | Bert- ? |
| 19:51.01 | *** join/#asterisk x6541a (n=SimmerMo@www3.datarack.nl) |
| 19:51.14 | Bert- | gnni ? |
| 19:51.16 | Assid | RoyK[at]: that would be the bootrom firmware |
| 19:52.21 | RoyK[at] | Bert-: don't use vlc. use ffmpeg instead. they both uses libavcodec |
| 19:52.31 | RoyK[at] | and ffmpeg is easier to tune |
| 19:53.02 | pdunkel | Anyone here have a clue for when the * 1.4 release is planned? |
| 19:53.21 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 19:53.30 | RoyK[at] | bkw_:hi |
| 19:53.35 | Bert- | the way is, I receive Stream Tv both from Internet and my tuner, and then with vlc I can encode it then stream it on the internet :) |
| 19:53.39 | Bert- | this part works really fin |
| 19:53.41 | Bert- | fine |
| 19:53.48 | x6541a | hello all, can someone help me with the following problem; |
| 19:53.53 | Bert- | I don't now if I can do that with ffmpeg |
| 19:54.01 | x6541a | I have one asterisk server |
| 19:54.02 | RoyK[at] | pdunkel: i guess 1.4 will come in a few months, 'few' being > 10 and perhaps add a few zeros |
| 19:54.07 | x6541a | with 2 cisco voice routers |
| 19:54.19 | x6541a | 1 on the old line, 1 on the new line |
| 19:54.25 | RoyK[at] | Bert-: ffmpeg can do that iirc |
| 19:54.43 | x6541a | when people dial into the old line, I play a .gsm message about the company having a new number |
| 19:54.59 | pdunkel | RoyK: That long, there are also son many features in bugs.di... marked [post-1.4] so I thought it was around the corner? |
| 19:55.08 | x6541a | after which a virtual extension is being dialed |
| 19:55.13 | RoyK[at] | pdunkel: rotfl |
| 19:55.14 | x6541a | working fine |
| 19:55.21 | RoyK[at] | pdunkel: post 1.4 is at _least_ a year away |
| 19:55.25 | RoyK[at] | pdunkel: backport it |
| 19:55.30 | RoyK[at] | pdunkel: asterisk-backports.org |
| 19:55.44 | x6541a | but the the tone when the phone is ringing, is US style or something. |
| 19:55.58 | pdunkel | RoyK: Since a few of them are mine, I guess I'll just have to do that. |
| 19:55.59 | x6541a | Can it be changed within extensions.conf? |
| 19:56.06 | RoyK[at] | x6541a: indications.conf? |
| 19:56.27 | RoyK[at] | pdunkel: please post them there |
| 19:57.01 | x6541a | RoyK: I'll have to check, never worked with that file. I am using a mgt GUI with a custom script. |
| 19:57.02 | pdunkel | RoyK: Will do as soon as I start putting them through extensive tests. |
| 19:57.26 | x6541a | The script only contains "exten => s,Playback(newnummerbla) |
| 19:57.34 | RoyK[at] | x6541a: there's a 'default' line in the top |
| 19:58.02 | RoyK[at] | pdunkel: post it anyway. it won't hurt. the worst that can happen is that someone finds abug |
| 19:58.09 | RoyK[at] | s/abug/a bug/ |
| 19:58.12 | x6541a | I'll checl |
| 19:58.13 | x6541a | k |
| 19:58.15 | znoG | Bert-: how well does streaming over the internet work? i have a tuner card (i use mythtv) but i thought it would be pointless streaming over the internet due to latency |
| 19:58.34 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 19:58.37 | pdunkel | RoyK: When I said "start extensive tests" that means within 1-2 weeks. |
| 19:58.55 | RoyK[at] | pdunkel: ok, never mind, then :) |
| 19:59.05 | l-fy | hi RoyK[at] |
| 19:59.13 | l-fy | where are you now? |
| 19:59.19 | pdunkel | RoyK: There are actually 2 items. |
| 19:59.26 | RoyK[at] | l-fy: .at |
| 19:59.37 | pdunkel | 1. Adding a passwd change notification vial external Script |
| 19:59.39 | l-fy | cool, where? |
| 19:59.48 | RoyK[at] | Salzburg |
| 19:59.50 | pdunkel | 1. (for VoicemailMail) |
| 19:59.58 | l-fy | RoyK[at] > you are 2 counties away from me |
| 20:00.00 | l-fy | come to visit me |
| 20:00.07 | RoyK[at] | l-fy: I was considering it :) |
| 20:00.21 | pdunkel | 2. Sequential dialing of Numbers (1, 1+2, 1+2+3, 1+2+3+4, ...) |
| 20:00.30 | l-fy | RoyK[at] > cool, come |
| 20:00.34 | l-fy | we have a couch |
| 20:01.12 | skraelings001 | does directed call pickup works with PRI ? |
| 20:03.51 | x6541a | RoyK: can the ring tone also be set in an extension configuration in extensions.conf? |
| 20:04.04 | x6541a | the default line is missing in my indications.conf |
| 20:05.18 | *** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:06.17 | *** part/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
| 20:07.42 | skraelings001 | i can't make directed call pickup works with pri? do i need to set transfer=yes in zapata.conf?? |
| 20:11.44 | x6541a | I have a general section though and the country selected is ok. |
| 20:12.05 | x6541a | I'm not sure that ring tone is the correct definition. |
| 20:12.30 | x6541a | Ring indication tone seems more appropriate |
| 20:14.41 | RoyK[at] | x6541a: sorry. it's country=xx |
| 20:15.11 | RoyK[at] | x6541a: set that to nl |
| 20:15.15 | RoyK[at] | country=nl |
| 20:15.51 | *** join/#asterisk negativecreep (n=xaeem@202.166.160.98) |
| 20:16.04 | x6541a | RoyK[at]: that is set ok, but more people have the same problem: |
| 20:16.11 | x6541a | http://www.asteriskguru.com/tutorials/playtones.html |
| 20:16.26 | x6541a | Malcolm seems to have the same problem |
| 20:17.24 | RoyK[at] | x6541a: well, i've set mine to =no, and it works :) |
| 20:18.49 | xheliox | If I wanted hundreds of FXS ports from Asterisk, are there any devices that would let me do that? Like output over a T1 card and break it down with a like a Rhino channel bank? Though they only do 24 ports.. as far as I know. |
| 20:19.33 | NotJohnDavid | xheliox: why not do IP phones? |
| 20:20.08 | x6541a | RoyK[at]: no as in not, or no as in norway? :-) country=no? |
| 20:20.12 | l-fy | NotJohnDavid > freaking expensive |
| 20:20.27 | xheliox | NotJohnDavid: I'm currently using EdgeGate SIP Gateways, they're fine -- I'd just like to consolidate everything a bit. Just pondering. |
| 20:21.01 | RoyK[at] | x6541a: norway :) |
| 20:24.39 | pdunkel | RoyK: I just added the first patch to asterisk-backports.org |
| 20:24.49 | pdunkel | The second still needs some work :) |
| 20:24.54 | NotJohnDavid | I'm having some noise problems on my SPA3k. background hum/hiss on the FXS port. even if I turn the gain way down it's still there |
| 20:29.39 | RoyK[at] | pdunkel: thanks, mate |
| 20:31.27 | RoyK[at] | x6541a: any more luck? |
| 20:31.45 | pdunkel | RoyK: Seems you are the roy managing the platform. So thanks for putting in that effort!!! |
| 20:31.48 | *** join/#asterisk starlein (n=star@fo0bar.de) |
| 20:32.01 | pdunkel | I am hereby formally requesting a round of applause for RoyK :) |
| 20:32.11 | pdunkel | *clapping* |
| 20:36.24 | x6541a | RoyK[at]: yes. I had to add an 'r' to the dial command in my config ('r': generate a ring indication for the calling party) |
| 20:36.49 | x6541a | Now the tone is dutch style, as it should be. |
| 20:37.19 | RoyK[at] | x6541a: if you don't, the telco you're calling will generate the tones |
| 20:39.40 | Assid | damn sipphone!!!!! |
| 20:40.45 | Assid | it used to work |
| 20:41.05 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 20:51.50 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 20:52.14 | hwt | does spandsp rxfax() work OK over voip networks now? |
| 20:54.18 | pdunkel | Got to go. Bye everyone! |
| 20:56.22 | *** join/#asterisk Un1x (i=seann@CPE00016c29e15a-CM00080d40ee4c.cpe.net.cable.rogers.com) |
| 20:56.32 | Un1x | has anyone used a GSM gateway before? |
| 20:56.57 | _problem_ | Un1x: yes i did ..what u wants to know ? |
| 20:57.08 | Un1x | i was wondering if the FX-300 have you ever heard of it? |
| 20:57.11 | Un1x | http://www.voip-info.org/wiki/view/VOIP+GSM+Gateways |
| 20:57.34 | Un1x | scroll down you will see it was wondering if it will work with asterisk or should i get the one from 2n voicblue |
| 20:58.21 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
| 20:59.36 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 20:59.36 | _problem_ | Un1x: can u see vierling ecotel on that page...well that works fine..i dont know about FX-300...but it can work..i believe |
| 20:59.48 | x6541a | RoyK[at]: thanks for your help. |
| 20:59.53 | Un1x | hmm know where i can get the, Voiceblue one? |
| 20:59.59 | l-fy | Un1x > use the 2n.cz boxes |
| 21:00.12 | Un1x | i-fy where can i get one?? |
| 21:00.13 | _problem_ | no sorry i dont know about that |
| 21:00.51 | Un1x | I-fy? |
| 21:01.36 | l-fy | Un1x > just look for 2n |
| 21:01.55 | l-fy | and is l-fy |
| 21:02.27 | Un1x | well im on 2n's site but i cant see a list of retailers or resellers :S |
| 21:04.27 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:04.42 | *** part/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 21:05.09 | Nivex | Un1x: I had the same problem. |
| 21:05.36 | Un1x | so howd u get by it/ |
| 21:05.43 | Nivex | I haven't. |
| 21:05.47 | Un1x | lol |
| 21:06.04 | Nivex | Guess they don't want to sell these things. |
| 21:06.27 | Nivex | oh here we go http://www.2n.cz/distributors/distributor_list.html |
| 21:07.15 | Assid | anyone here on sipphone? |
| 21:07.20 | Assid | or gizmo? |
| 21:13.41 | muppetmaster | Yes |
| 21:13.44 | muppetmaster | On Gizmo |
| 21:14.41 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.24) |
| 21:16.45 | Assid | can you call me? |
| 21:18.29 | Assid | ? |
| 21:19.14 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 21:22.55 | *** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net) |
| 21:26.44 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 21:47.12 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 21:47.29 | Manipura | hello Everyone |
| 21:47.35 | Manipura | Quick Question.... |
| 21:48.00 | Assid | geez.. how long doews it take to do the bootrom update? |
| 21:48.07 | Assid | its stuck on formatting file system |
| 21:48.24 | Assid | for almost 2 minutes now |
| 21:49.26 | Assid | okay nvm |
| 21:49.47 | Manipura | I'm going to have a single DID, but many Asterisk servers to run all traffic. How do I get the did to go to the next computer once server #1 is getting overloaded |
| 21:53.46 | hwt | Manipura: asterisk cluster? |
| 21:54.00 | hwt | Manipura: or simply DNS round robin. |
| 21:54.15 | Manipura | hwt is a cluster easy to create? |
| 21:54.50 | hwt | Manipura: read http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions |
| 21:55.04 | Manipura | Thank you |
| 21:58.18 | Assid | umm..i think i need some help with my polycom |
| 21:58.32 | Assid | i updated my bootrom.. and now it refuses to pass the "checking application..." |
| 21:58.33 | *** join/#asterisk Tier_1 (n=tier@c-67-176-28-65.hsd1.co.comcast.net) |
| 21:58.43 | Tier_1 | hey all I need some ideas for extensions |
| 21:59.02 | Tier_1 | good ideas not play extensions |
| 22:02.39 | Assid | err.. i think my phones gone nutty |
| 22:02.44 | Assid | it refuses to work |
| 22:05.55 | Tier_1 | did you put it in a straight jacket |
| 22:06.15 | Tier_1 | beat it till it works |
| 22:06.54 | Tier_1 | refuse to feed it dialtone |
| 22:06.58 | Tier_1 | joking |
| 22:07.28 | Assid | stuck on "checking application..." |
| 22:07.35 | Assid | :( |
| 22:07.54 | Assid | me and my brright ideas |
| 22:10.16 | Assid | okay what do i do? |
| 22:10.20 | Assid | anyone know? |
| 22:11.15 | Qwell | RMA |
| 22:11.30 | Tier_1 | reflash |
| 22:11.39 | [TK]D-Fender | Assid : change your firmware |
| 22:12.59 | NotJohnDavid | tkd: finally figured out my SPA3000 problems with PSTN ring thru |
| 22:13.11 | Assid | i just downgraded it back to 2.6.1 .. im still waiting for checking application |
| 22:13.23 | Assid | i touched the sip.* files.. so i think its downloading that now |
| 22:13.49 | [TK]D-Fender | Assid : I'm not talking about boot rom... |
| 22:13.56 | [TK]D-Fender | Assid : I'm talking about SIP |
| 22:14.01 | Assid | oh |
| 22:14.20 | Assid | okay well i just touched sip.* so i think its downloading now |
| 22:14.58 | Assid | "error. application is not present" |
| 22:15.02 | Assid | wtf?!?!?!? |
| 22:16.06 | Assid | please please just work.. |
| 22:17.10 | Assid | worse part is .... i am doing a remote update.. so it takes forever to update the file too |
| 22:18.42 | X-Rob | ~centosbug |
| 22:18.44 | jbot | somebody said centosbug was a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:19.16 | Assid | [TK]D-Fender: you thinkk its recoverable? |
| 22:21.32 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 22:23.48 | SwK | [TK]D-fender downgraded from what to 2.6.1? |
| 22:23.56 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 22:24.39 | SwK | [tk]d-fender are you having the upgrade firmware and the 30X wont boot now |
| 22:24.58 | Assid | SwK: me.. |
| 22:25.11 | Assid | and i loaded up 2.6.2 then my phone died |
| 22:25.16 | Assid | so i touched sip.* |
| 22:25.23 | Assid | then i got that error. |
| 22:25.27 | Assid | now i think it might work |
| 22:25.36 | Assid | its loading application |
| 22:25.59 | Assid | im back to 2.6.1 tho |
| 22:26.21 | Assid | but its alrite.. according to the docs its only a dhcp packet fix |
| 22:27.05 | *** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 22:27.13 | Assid | wooohoooo worked! |
| 22:27.53 | Assid | thanks swk, [TK]D-Fender |
| 22:28.24 | [TK]D-Fender | np |
| 22:28.39 | Assid | im not upgrading bootrom again |
| 22:28.58 | [TK]D-Fender | Assid : Make sure you check your boot/app compatability before you upgrade next time. I might suggest you replace you setup with 1.6.6. now however. |
| 22:29.10 | [TK]D-Fender | Assid : if its an X01 phone |
| 22:29.45 | Assid | yep.. i got 1.6.6 |
| 22:32.32 | Assid | so 300 and 301 uses different bootrom |
| 22:33.40 | [TK]D-Fender | Assid : No its taht once you upgrade an older phone there may be no going back. |
| 22:33.42 | *** part/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 22:34.26 | Assid | okay im not upgrading till i get an all clear from people in this room |
| 22:36.58 | rob0 | What makes you think there are people in this room? Maybe only dogs? |
| 22:37.30 | rob0 | no ... monkeys. We recorded tt-monkeys. |
| 22:37.47 | Assid | that would explain why sometimes people come and ask me the same question |
| 22:38.23 | Assid | oh yeah |
| 22:38.31 | Assid | did i tell ya.. my local linux box died |
| 22:38.44 | Assid | i think the lighning did some damage to it |
| 22:39.30 | rob0 | That's very sad. |
| 22:39.36 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 22:39.57 | Assid | i know |
| 22:40.07 | Assid | it was bad enough it blew up my router.. and my lan card |
| 22:40.13 | Assid | but to take out my linux box :( |
| 22:40.40 | rob0 | Uncalled for. |
| 22:40.51 | Assid | oh well.. im outta here |
| 22:40.56 | Assid | ciao |
| 22:40.57 | rob0 | ok |
| 22:41.34 | Tier_1 | ok I need polycom 1.6x firmware and my polycom contract expired lastweek |
| 22:41.44 | Tier_1 | this sucks |
| 22:42.39 | Tier_1 | lol |
| 22:46.32 | *** join/#asterisk hmodes (i=hmodes@c-71-230-56-196.hsd1.pa.comcast.net) |
| 22:47.53 | *** join/#asterisk RF_MIA (n=mark@68-235-157-31.miamfl.adelphia.net) |
| 22:52.03 | [TK]D-Fender | Tier_1 : Ask you reseller. Should jsut give it to you |
| 22:52.35 | Tier_1 | that means waiting till wednesday |
| 22:52.38 | Tier_1 | grr |
| 22:52.49 | Tier_1 | 4 day weekend |
| 22:53.20 | SwK | i've gotta few 300s that refuse to boot with the newerbootroms |
| 22:53.35 | SwK | doenst matter which 1.6.x firmware you use with them |
| 23:03.21 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 23:03.49 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 23:04.33 | Tier_1 | brb |
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