| 00:00.26 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-73.apple.com) |
| 00:00.41 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 00:00.42 | drray | so Druken, I'm working on a project you might be interested in |
| 00:00.54 | Druken | oh? |
| 00:00.56 | drray | using a PIC to build a apayphone board |
| 00:01.09 | Druken | a pic? |
| 00:01.19 | gandhijee | microcontroller type device |
| 00:01.37 | drray | this as a starting point |
| 00:01.47 | drray | http://melabs.com/products/labxt.htm |
| 00:02.28 | Druken | ewhh... cexzy |
| 00:02.39 | drray | I'm thinking about |
| 00:02.45 | drray | nothing may come of it |
| 00:03.06 | Druken | thats most projects with me :) |
| 00:03.16 | drray | dtmf, escrow, coin are all you need |
| 00:03.57 | Druken | yup, all the old phones had... |
| 00:05.36 | Zodiacal | can the playback application play .wav's? |
| 00:05.40 | Zodiacal | it doesn't seem to play mine |
| 00:05.42 | drray | I'm thinking about it, I'll let you know how it goes |
| 00:05.59 | Druken | awesome |
| 00:06.03 | drray | you won't need the $349 if I do it |
| 00:06.25 | *** join/#asterisk pjchilds-athome (n=pjchilds@ppp218-87.lns1.adl2.internode.on.net) |
| 00:06.35 | Druken | why do i need $349 again? hehe |
| 00:06.49 | Druken | oh, for the assembled |
| 00:06.52 | Druken | my bad |
| 00:06.52 | drray | well, I'm thinking of getting it to prototype it |
| 00:07.02 | drray | then I'll make a custom board |
| 00:07.10 | drray | which should be below $100 |
| 00:07.19 | Druken | you got etching boards? |
| 00:07.27 | drray | emachineshop.com |
| 00:07.40 | Druken | that works too :) |
| 00:09.30 | drray | I'm getting interested in non-pc PBX's |
| 00:11.37 | Jaxxan | Qwell: you aroundn ? |
| 00:12.01 | Qwell[] | nope |
| 00:12.07 | Qwell[] | I'm a square |
| 00:12.11 | Jaxxan | (= |
| 00:12.22 | Jaxxan | I need pausequeueagent and unpausequeueagent |
| 00:12.34 | Qwell[] | I need $$$ ;/ |
| 00:12.43 | Jaxxan | no problem |
| 00:12.51 | Qwell[] | gonna be around in an hour and a half? |
| 00:12.55 | Jaxxan | yup |
| 00:13.24 | asterboy | I like that http://melabs.com/products/labxt.htm project. |
| 00:13.25 | Qwell[] | can talk about it when I get home. We need to make sure there isn't something else that can do what you need |
| 00:13.36 | Jaxxan | k |
| 00:14.04 | Qwell[] | we already talked about this briefly before, right? |
| 00:14.25 | Druken | sounds farmilliar to me :) |
| 00:14.32 | Qwell[] | Druken: yeah, me too |
| 00:14.32 | Jaxxan | sort of |
| 00:14.42 | Jaxxan | i said: i need this and you said i need $$$ |
| 00:14.46 | Qwell[] | heh |
| 00:14.59 | Qwell[] | Am I that predictable? |
| 00:15.02 | Jaxxan | then you said you were leaving for a week and a half or soemthing |
| 00:15.08 | Qwell[] | ahh, right |
| 00:15.18 | Jaxxan | (= |
| 00:15.31 | Qwell[] | yeah, we can figure it out in a bit here |
| 00:15.42 | Jaxxan | ok |
| 00:18.33 | *** part/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
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| 00:35.22 | Qwell[] | why, if I didn't know any better... |
| 00:35.53 | Qwell[] | AvoidingDeadlock: Your ident is showing :p |
| 00:40.13 | *** join/#asterisk pengyong (n=lala@222.188.131.245) |
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| 01:03.17 | dlynes_ | Anyone have wcfxo segfault on startup? |
| 01:03.28 | dlynes_ | erm i mean ztcfg segfault on startup? |
| 01:03.35 | Jaxxan | let's say i want to handle the voicemail of all calls being passed to asterisk from a dms100. Caller -> Callee -> Voicemail. When I do a " Pri Debug" i can see Caller ID, and the Voicemail number, and i can also see the number they called before being redirected by the dms100 to voicemail. how can i get the Callee number within asterisk when the calls comes down my trunk >? |
| 01:05.22 | Jaxxan | i know CallerID and Voicemail destination number, but in order for me to route them to the right mailbox i'll need to know the original number that they dialed. how can i pull that information into the dialplan ? |
| 01:05.39 | dlynes_ | Jaxxan: ${DNID}? |
| 01:05.48 | Jaxxan | lemme try that |
| 01:06.09 | Jaxxan | what's that do ? |
| 01:06.18 | dlynes_ | Dialed Number ID |
| 01:06.33 | Jaxxan | sounds good |
| 01:06.50 | dlynes_ | If you do a noop on initial answer of the channel |
| 01:06.58 | dlynes_ | you'll see dnid and calleridnum are the same number |
| 01:07.24 | Jaxxan | you have an example ? |
| 01:07.41 | dlynes_ | Voicemail(s${DNID})? |
| 01:08.53 | dlynes_ | oh joy |
| 01:09.03 | dlynes_ | I found a null pointer dereference in zaptel :(( |
| 01:11.35 | Jaxxan | DNID went to Voicemail number and not the original calling number |
| 01:11.57 | Jaxxan | how can i echo the variables to the console ? |
| 01:13.30 | fjean | NoOP |
| 01:14.34 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 01:14.38 | Jaxxan | gotcha |
| 01:16.34 | distortion | dlynes: were you talking about the 250 call limit on 1.2.4? |
| 01:16.42 | distortion | the other day? |
| 01:17.08 | fjean | hehe, is that true.. |
| 01:17.14 | distortion | asterisk, yes :) |
| 01:17.31 | distortion | err |
| 01:17.46 | fjean | what about the subscriptions ? |
| 01:17.54 | fjean | is there a limit too |
| 01:17.59 | distortion | I am reporting im running higher then that on 1.2.4 so I was just curious |
| 01:18.06 | fjean | ok |
| 01:18.23 | fjean | you are saying calls or channels ? |
| 01:18.46 | distortion | 524 active channels 263 active calls |
| 01:18.51 | fjean | ok |
| 01:18.58 | *** join/#asterisk MaddieBoi (n=MaddieBo@210-84-15-248.dyn.iinet.net.au) |
| 01:20.14 | fjean | it must be a good machine... |
| 01:20.20 | fjean | lol |
| 01:21.23 | distortion | dual 2.8 zeon dell. < 3% cpu |
| 01:21.37 | fjean | mmm |
| 01:21.46 | fjean | I am curious now |
| 01:22.01 | fjean | you dial within AGIs ? |
| 01:23.02 | distortion | combination between static dial plans, realtime extensions and a mysql database application, not using any agi. |
| 01:23.14 | fjean | ok |
| 01:23.33 | fjean | cool |
| 01:28.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:28.25 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 01:36.02 | *** join/#asterisk blebleble (n=ble@d149-67-206-171.col.wideopenwest.com) |
| 01:36.19 | blebleble | Can asterisk do voicemail distribution lists? |
| 01:36.59 | dlynes_ | blebleble: you mean where when you leave a voicemail |
| 01:37.09 | dlynes_ | blebleble: it leaves it for ten people all at the same time? |
| 01:37.14 | blebleble | dlynes: yes |
| 01:37.24 | dlynes_ | blebleble: Yes, it handles that, and with ease |
| 01:37.30 | *** join/#asterisk Eggplant (n=none@dsl-434.cascadeaccess.com) |
| 01:37.44 | dlynes_ | ~docs |
| 01:37.48 | jbot | extra, extra, read all about 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 |
| 01:37.51 | blebleble | anyway you can point me at some documentation or poing me in the right location, i havent seen it |
| 01:38.02 | dlynes_ | See the wiki above, and follow the area to the asterisk wiki |
| 01:38.05 | Sedorox | blebleble: hold on a second... I have a line that will probably do it for you |
| 01:38.07 | mtaht3 | hey jbot |
| 01:38.16 | dlynes_ | and then click on 'application commands' and then click on 'voicemail' |
| 01:38.18 | Jaxxan | woot! ${RDNIS} ftw ! |
| 01:38.24 | blebleble | Sedorox: appreciate it |
| 01:38.31 | Sedorox | exten => 240,1,Voicemail(261&201&202&233&231&232) |
| 01:38.31 | Jaxxan | thanks guys |
| 01:38.34 | Sedorox | apparently works for me... |
| 01:38.44 | Sedorox | dial 240.. and it leaves voicemails in those boxes |
| 01:38.58 | dlynes_ | Sedorox: yeah...i was just pointing him at the wiki, so he can see all the other wonderful stuff he can do with voicemail :) |
| 01:39.06 | mjackson | What do (*) represent when used as part of a call regex... does that represent an actual keypress of (*) ? exten => _*NXXNXXXXXX*3429,3,Answer |
| 01:39.07 | Sedorox | well yea |
| 01:39.12 | Sedorox | I would still check out the wiki |
| 01:39.34 | Sedorox | plus.. I dunno if thats right |
| 01:39.37 | Sedorox | thats just what I found worked :p |
| 01:39.44 | dlynes_ | Sedorox: no...that's correct |
| 01:39.56 | dlynes_ | Sedorox: never had a need for it yet, myself |
| 01:40.04 | dlynes_ | Sedorox: but i've gone through that section of the wiki |
| 01:40.12 | Sedorox | Ah... we haven't either.. but a guy asked for it.. so I added it :p |
| 01:40.15 | Sedorox | neat |
| 01:40.21 | Sedorox | I need to catch up on 1.2 badly... |
| 01:40.21 | Sedorox | <PROTECTED> |
| 01:40.28 | dlynes_ | Sedorox: yeah...1.2 kicks ass |
| 01:40.29 | Sedorox | I haven't gotten to work with it since about 1.0.9 |
| 01:40.36 | blebleble | Thanks guys, giving it a shot |
| 01:40.41 | Sedorox | I JUST got to updating a few things... |
| 01:40.44 | dlynes_ | Sedorox: and 1.4 kicks more...it'll have shared line appearances |
| 01:40.51 | Sedorox | blebleble: np.. just remember.. the wiki does help |
| 01:40.55 | Sedorox | ooo nice |
| 01:40.59 | blebleble | noted |
| 01:42.11 | *** join/#asterisk NativePHP (n=ajb@c-69-249-119-114.hsd1.nj.comcast.net) |
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| 02:00.51 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 02:01.05 | Jaxxan | wow |
| 02:01.13 | Jaxxan | guys you have no idea how much i love asterisk right now |
| 02:01.36 | Jaxxan | asterisk just saved me $15,000 dollars upgrading my current voicemail system |
| 02:02.04 | Damin | Jaxxan: Feel free to paypal a little bit of those savings over to me! :) |
| 02:02.05 | Jaxxan | glenayre sucks |
| 02:02.13 | *** join/#asterisk [Outcast] (n=outcast@219-89-206-239.adsl.xtra.co.nz) |
| 02:02.25 | *** join/#asterisk nain (n=nain@202.59.90.182) |
| 02:02.31 | nain | Hi Every body |
| 02:03.22 | Jaxxan | what's the standard voicemail progression? b -> u ? or u -> b ? |
| 02:03.31 | Jaxxan | busy to unavailable sounds logical to me |
| 02:03.47 | nain | Is there any soft dialer which allow us to make change in frame rate or any configration to make change in asterisk frame rate of Codec.... |
| 02:05.21 | nain | Any configration in SIP ? Like in asterisk-oh323 configration we can set the frame rate of codecs like codec=ulaw, frame=2, |
| 02:05.29 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 02:06.57 | fjean | well, with sjlabs you can choose ILBC 20 or 30 ms :-) |
| 02:07.01 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 02:07.41 | fjean | RTP jitter buffer length... |
| 02:07.59 | fjean | driver buffer size.. |
| 02:13.18 | *** join/#asterisk pbx321 (n=pbx321@203.177.234.49) |
| 02:16.06 | nain | fjean: how to change rtp jitter buffer length in asterisk or driver buffer size ? |
| 02:16.14 | nain | I am using eyebeam for dialing.... |
| 02:16.29 | nain | and i don't think so sjlabs provide g729 support |
| 02:18.59 | [Outcast] | any news on the the asterisk locking up while using agentcallback |
| 02:19.15 | [Outcast] | dang i can't type |
| 02:19.51 | docelm0 | [Outcast] thats all you.. I dont have problems with lockup |
| 02:20.07 | [Outcast] | docelm0, what version are you using? |
| 02:20.26 | docelm0 | 1.2.1, 1.2.4, 1.2.7.1 |
| 02:20.41 | [Outcast] | i am refering to this bug? |
| 02:20.42 | gandhijee | anyone here know if its possible to change and IRQ to CPU mapping in the 2.6 kernel??? |
| 02:20.47 | [Outcast] | http://bugs.digium.com/view.php?id=6626 |
| 02:20.59 | Damin | docelm0: Get my shit? |
| 02:22.18 | docelm0 | what shit? |
| 02:22.41 | docelm0 | I have lots ' |
| 02:22.42 | docelm0 | err |
| 02:22.45 | [Outcast] | hehe |
| 02:22.48 | docelm0 | lots 'o shit.. |
| 02:23.02 | docelm0 | Just need to know what your refering with more precicely |
| 02:23.02 | [Outcast] | docelm0, you are using callback agent with 1.2.7.1 |
| 02:23.46 | docelm0 | all of them actually |
| 02:23.57 | [Outcast] | cool |
| 02:24.09 | docelm0 | I run over 15 asterisk boxes in one form or another |
| 02:24.21 | [Outcast] | there has been some major issue with my 1.2.5 install |
| 02:24.27 | docelm0 | Damin Im doing your rate change now.. You will be able to go live in about 10 minutes |
| 02:25.06 | [Outcast] | queues keep locking with call back agents |
| 02:25.27 | [Outcast] | it looks like the same bug is in 1.2.6 according to the bug tracker |
| 02:28.21 | gandhijee | anyone here know if its possible to change and IRQ to CPU mapping in the 2.6 kernel??? |
| 02:29.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-142-12.buff.east.verizon.net) |
| 02:29.12 | *** join/#asterisk Rick_Hunter (n=rhunter@01-143.008.popsite.net) |
| 02:29.16 | dlynes_ | gandhijee: you mean assign an irq to a different card? |
| 02:29.18 | SuPrSluG | hello |
| 02:29.39 | websae | hello |
| 02:29.51 | SuPrSluG | having a voicemail issue |
| 02:30.23 | gandhijee | dlynes: no more like assign an irq to a different CPU |
| 02:30.30 | gandhijee | in a mutli-processor system |
| 02:30.41 | dlynes_ | oh |
| 02:30.56 | gandhijee | yeah |
| 02:31.12 | gandhijee | i want to move the TDM400 from CPU0 to CPU1 |
| 02:31.30 | gandhijee | that what is can generate all the interupts it wants, and hopefully be happy |
| 02:31.44 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
| 02:32.00 | SuPrSluG | when i got to voicemailmain and put in my mailbox : i get ->Incorrect password '' for user '311' (context = default) context is voicemail. where is it getting default from? |
| 02:32.51 | SuPrSluG | my voicemail.conf context is voicemail. this is getting weird |
| 02:33.41 | *** join/#asterisk Souvent22 (n=chatzill@c-69-143-189-36.hsd1.va.comcast.net) |
| 02:33.53 | docelm0 | web its late.. :) |
| 02:34.17 | Souvent22 | Hello. |
| 02:35.05 | Souvent22 | I'm trying to setup outbound dialing with asterisk, and running into a snag...i have kind of a strange setup. |
| 02:35.20 | dlynes_ | Souvent22: spit it out |
| 02:35.33 | Souvent22 | Say you have a PBX that's acting like a SIP proxy, say a Cisco or an inter-tel system.... |
| 02:35.37 | dlynes_ | SuPrSluG: did you restart asterisk? |
| 02:35.41 | pjchilds-athome | gandhijee: CPU affinity... turn off irqbalance then google on /proc/irq/*/smp_affinity |
| 02:35.43 | Souvent22 | you can register a sip phone client against it and make outbound calls..... |
| 02:35.51 | SuPrSluG | yes |
| 02:36.05 | SuPrSluG | upgrade to new svn |
| 02:36.06 | dlynes_ | SuPrSluG: by restart, do you mean reload, or actually restart it? |
| 02:36.09 | gandhijee | thanks |
| 02:36.17 | Souvent22 | so what i'm trying to do is have asterisk register itself to the pbx, and then have a device register to asterisk, and dial out. |
| 02:36.23 | SuPrSluG | restart |
| 02:36.27 | gandhijee | pjchilds: how do i turn off irqbalance? |
| 02:36.47 | Souvent22 | asterisk registers ok, but i'm a little fuzzy with how i setup extentions.conf.... |
| 02:37.01 | Souvent22 | do i just match a dial plan, and then do a SIP dial to asterisk? |
| 02:37.02 | dlynes_ | SuPrSluG: voicemail.conf -----> [default] 311,311,,, |
| 02:37.17 | pjchilds-athome | gandhijee: /etc/init.d/irqbalance stop ... depending on what linux you are running you need to do something to stop it from starting when you start linux... see your vendor linux doco about services control.. |
| 02:37.18 | Souvent22 | e.g. Dial(SIP/ext@localhost) |
| 02:37.27 | dlynes_ | Souvent22: Dial(SIP/exten) |
| 02:37.33 | SuPrSluG | [voicemail] |
| 02:37.43 | dlynes_ | SuPrSluG: use pastebin |
| 02:37.44 | gandhijee | cool |
| 02:37.46 | dlynes_ | ~pb |
| 02:37.50 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:38.11 | Souvent22 | that won't cause it to loop, like its dialing itself since it's registered against the PBX with that extention? |
| 02:38.25 | dlynes_ | Souvent22: why are you trying to call yourself? |
| 02:38.46 | Souvent22 | dlynes: i'm not, trying to dial outbound. |
| 02:39.08 | dlynes_ | Souvent22: Dial(SIP/sipoutboundpeername/${EXTEN}) |
| 02:39.47 | dlynes_ | SuPrSluG: pastebin your voicemail.conf file |
| 02:39.56 | dlynes_ | SuPrSluG: scrub the passwords first though |
| 02:41.24 | *** join/#asterisk Rick_Hunter (n=rhunter@01-143.008.popsite.net) |
| 02:43.28 | SuPrSluG | dlynes_:http://pastebin.ca/54677 |
| 02:43.45 | SuPrSluG | standard issue voicemail.conf |
| 02:44.27 | SuPrSluG | been working fine for ages. any new changes to voiemail w/ svn tree |
| 02:45.05 | Mavvie | brilliant! this Juniper device is blocking the SIP packets from our grandstream phones... |
| 02:45.19 | Mavvie | "SIP parser error SIP-Contact-Url: ":" does not exist" it complains about. |
| 02:46.26 | Mavvie | and yes, this is what it sends: "Contact: *." |
| 02:47.24 | *** part/#asterisk pjchilds-athome (n=pjchilds@ppp218-87.lns1.adl2.internode.on.net) |
| 02:51.50 | SuPrSluG | dlynes_:turns out I had to add @voicemail in extensions.conf to all things voicemail related. Is this new? |
| 02:54.18 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 02:54.21 | dlynes_ | SuPrSluG: no, but '[voicemail]' is not the default context |
| 02:54.25 | SuPrSluG | thanks. many new things like Timeout(Response)=10) and the like been wreaking havoc on me. Old dogs gotta learn new tricks. |
| 02:54.42 | dlynes_ | SuPrSluG: the '@voicemail' has been around since asterisk 1.0.x |
| 02:55.17 | dlynes_ | SuPrSluG: i suspect your voicemail file probably never used anything besides the default context before |
| 02:55.32 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:55.51 | ManxPower | Supaplex, Um, almost nothing in 1.2 is incompatable with 1.0 dialplans. |
| 02:56.20 | SuPrSluG | dlynes_: i've been using voicemail context for over 1 1/2 now. just went south recently. |
| 02:56.24 | ManxPower | I had to make like 2 changes to my 1.0 config files to use them with 1.2 |
| 02:56.35 | dlynes_ | SuPrSluG: you must've changed something to make it go south then |
| 02:56.36 | paolob | hi guys! is there an application to call a subfunction (subroutine) in a dialplan? i.e. something that when it end goes back where it was called |
| 02:56.37 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 02:57.00 | dlynes_ | paolob: DeadAGI |
| 02:57.26 | ManxPower | paolob, it's called....Gosub |
| 02:57.34 | dlynes_ | or that, too :) |
| 02:58.12 | paolob | ManxPower, AsteriskTFOT doesn't document it... |
| 02:58.41 | dlynes_ | anyways...laters peeps |
| 02:58.41 | ManxPower | pauldy, try "show applications" |
| 03:00.23 | paolob | ManxPower, ok, I saw it, how do I end the gosub? |
| 03:00.46 | ManxPower | paolob, "show application return" |
| 03:00.56 | ManxPower | "show application gosub" |
| 03:07.16 | Jaxxan | what would gosub be used for ? |
| 03:07.43 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 03:07.56 | droops | how do i tell what codec im using with sip? |
| 03:08.34 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 03:08.58 | Qwell | droops: show channels |
| 03:09.05 | droops | thanks |
| 03:09.22 | Agrajag- | gday. im trying to setup simple SIP, and it seems to work, but if a user hasn't registered and another tries to call, the softphone (ekiga) says "remote user rejected the call" rather than remote user offline. what've i done wrong? |
| 03:10.17 | Qwell | because the former is a real sip code, would be my guess |
| 03:10.53 | Agrajag- | ok you've lost me :) |
| 03:11.50 | Agrajag- | in my extensions.conf i have exten => wabz,1,Dial(SIP/wabz/20,tr) - so that i can call sip:wabz@foo.com - is that correct? |
| 03:12.02 | Qwell | get rid of the r |
| 03:12.14 | Qwell | unless you can give me a reason why it's there |
| 03:12.17 | Agrajag- | ah yeah.. no i can't |
| 03:12.34 | Jaxxan | if i want to modify vm-*.gsm files that are played... what file is that ? |
| 03:12.44 | Jaxxan | for the voicemail application |
| 03:12.50 | Qwell | Jaxxan: app_voicemail.c |
| 03:12.55 | Jaxxan | sweet |
| 03:13.17 | Jaxxan | can i just modify that with vi ? or do i have to do some special stuff ? |
| 03:17.01 | dlynes | Jaxxan: vi's fine |
| 03:17.09 | dlynes | Jaxxan: then just recompile and reinstall |
| 03:18.33 | Jaxxan | k |
| 03:18.45 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:18.53 | [Outcast] | docelm0, i have duplicated the issue in 1.2.7.1 |
| 03:19.56 | Agrajag- | Qwell: hmm i still get remote user rejected call |
| 03:20.56 | dlynes | Jaxxan: Have you ever programmed in Microsoft tiny basic? |
| 03:21.02 | *** join/#asterisk espn (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 03:21.31 | dlynes | Jaxxan: or gwbasic, or IBM Basic A, for that matter? |
| 03:23.27 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool2-a250.nwlnnh.tds.net) |
| 03:24.19 | pauldy | someone looking for me |
| 03:24.55 | ManxPower | Qwell, "sip show channels" |
| 03:25.06 | Qwell | ManxPower: whichever |
| 03:25.11 | *** part/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool2-a250.nwlnnh.tds.net) |
| 03:25.22 | ManxPower | show channels does not tell you the codec |
| 03:25.44 | Qwell | can never remember which one it is |
| 03:27.56 | *** join/#asterisk pagec (n=cpage@64-252-97-38.adsl.snet.net) |
| 03:28.39 | [TK]D-Fender | dlynes : GW Basic? OMG you cheater! GW = advanced! (Graphics & Windows)! |
| 03:29.05 | [TK]D-Fender | I remember the days or basic in IBM PC bootroms :) |
| 03:29.21 | dlynes | [TK]D-Fender: wtf? graphics and windows? are you on crack? |
| 03:29.33 | dlynes | I don't recall any graphics in GW Basic :) |
| 03:29.41 | [TK]D-Fender | dlynes : Thats what the GW stood for. |
| 03:30.17 | [TK]D-Fender | dlynes : Original basic had no graphics capabilities, and "windows" was a subs scrolling section of TTY within the main screen scope. |
| 03:30.37 | dlynes | ah....figured it just stood for ginormous willies |
| 03:30.38 | [TK]D-Fender | dlynes : I've seen it all :) |
| 03:31.01 | [TK]D-Fender | funny thing is "windows" was actually reffering to a TEXT MODE function :) |
| 03:31.44 | dlynes | [TK]D-Fender: ummm...before Windows 3.0, it was a text mode function in Windows, too :) |
| 03:31.46 | [TK]D-Fender | I remember the rush of writing my first 2D trig calculator program L( |
| 03:32.12 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:32.20 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:33.11 | [TK]D-Fender | dlynes : Not quite, it was a full TTY in a "window". I miss DesqView sometimes. I remember booting off multiple floppies to ram-disk and loading Desqview to be the only multitasking guy in programming class with all the COOL stuff :) |
| 03:33.28 | dlynes | [TK]D-Fender: yeah....Desqview rocked :) |
| 03:33.44 | dlynes | [TK]D-Fender: but Desqview/X was a bit of a disappointment |
| 03:33.45 | [TK]D-Fender | dlynes : And I'm not talking Desqview/X here either... |
| 03:33.55 | dlynes | I thought it would have taken off more than it did |
| 03:34.01 | dlynes | Yeah, i know |
| 03:34.05 | [TK]D-Fender | heh, well X-Windows was in its infancy as well... |
| 03:34.07 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 03:34.08 | dlynes | I never used Desqview/X |
| 03:34.22 | dlynes | I've got a manual for it still sitting on my bookshelf, though :) |
| 03:34.39 | dlynes | no idea what i'll ever do it with it, but... |
| 03:34.45 | [TK]D-Fender | dlynes : windows killed all sorts of things like OS/2, PC-Tools, Norton Utilities, etc.... for my Linux is like the second coming :) |
| 03:34.57 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 03:35.08 | dlynes | windows didn't kill os/2 |
| 03:35.11 | dlynes | ibm killed os/2 |
| 03:35.12 | [TK]D-Fender | I use an old box for LANTastic as a monitor stand, close enough |
| 03:35.15 | iCEBrkr | dlynes: amen |
| 03:35.22 | [TK]D-Fender | dlynes : Same shit really... |
| 03:35.24 | iCEBrkr | IBM couldn't sell water in the desert |
| 03:35.38 | dlynes | ibm has the world's worst marketing team |
| 03:35.45 | [TK]D-Fender | iCEBrkr : Yeah tahts why they're so BROKE hmm? |
| 03:35.51 | iCEBrkr | OS/2 Warp 4 was actually quite superior to Win98 :P |
| 03:35.54 | [TK]D-Fender | iCEBrkr : An hey, long time no see, how goes? |
| 03:35.59 | iCEBrkr | Could be better |
| 03:36.05 | dlynes | Warp 4 kicked win98's ass |
| 03:36.13 | dlynes | but IBM let Microsoft push them around |
| 03:36.14 | iCEBrkr | Jobless. Cuz my fuckass company killed my project and they punted me |
| 03:36.21 | [TK]D-Fender | iCEBrkr : My analog watch is superior to Win98! What's your point? ;) |
| 03:36.31 | [TK]D-Fender | iCEBrkr : OMG! |
| 03:36.37 | iCEBrkr | dlynes: Well,there was this B.S. agreement for OS/2 developers where you had to PAY IBM to write OS/2 apps |
| 03:36.54 | dlynes | iCEBrkr: i never had to pay ibm to write os/2 apps |
| 03:37.00 | dlynes | iCEBrkr: and i wrote a few of them |
| 03:37.15 | iCEBrkr | dlynes: Same here, but if you wanted the OS/2 logo or some shit. |
| 03:37.31 | iCEBrkr | There was some clause that you had to pay them to sell your OS/2 app |
| 03:37.35 | *** join/#asterisk Rick_Hunter (n=rhunter@01-143.008.popsite.net) |
| 03:37.39 | dlynes | iCEBrkr: not quite |
| 03:37.52 | dlynes | iCEBrkr: you could pay them to include your app in their marketing crap |
| 03:38.21 | dlynes | iCEBrkr: you could also pay them to have your app pass os/2 certification |
| 03:38.24 | iCEBrkr | [TK]D-Fender: Yea, so I'm actually across the state job shopping.. Hang'n out in Boca Raton. Interview tomorrow at 11:30 |
| 03:38.29 | asterboy | lantastic...been a long time since I hears that name |
| 03:38.30 | dlynes | iCEBrkr: if it passed, using hte logo was free |
| 03:38.48 | iCEBrkr | dlynes: Yea and to code and sell an app for Winblows was Free! |
| 03:38.56 | asterboy | M$ killed lantastic |
| 03:39.03 | iCEBrkr | No fuss. So where do you think the dev's went? |
| 03:39.06 | dlynes | iCEBrkr: but it wasn't them you were paying...it was some independent company that did os/2 software certification |
| 03:39.11 | iCEBrkr | My OS/2 warp box basically mirrored my Linux box. |
| 03:39.39 | dlynes | iCEBrkr: yeah, but i'm sure you still have to pay someone to get your windows apps certified for Windows |
| 03:39.51 | dlynes | iCEBrkr: just most people don't care if their app's certified or not |
| 03:39.55 | iCEBrkr | Exactly |
| 03:40.18 | dlynes | most peeps didn't care if their apps were certified for os/2 or not, either, unless they were selling it to an ibm shop |
| 03:40.21 | iCEBrkr | I liked OS/2. I still have the CDs |
| 03:40.36 | iCEBrkr | hrrm, maybe that's what I'll do with the old Cyberdyne.org hardware |
| 03:40.38 | dlynes | Same here...still have my Warp 4 cd and warp 4 driver pack cd |
| 03:40.48 | Damin | Hehehe.. I remember the good old days of the OS/2 era! ;) |
| 03:40.51 | dlynes | not to mention my os/2 2.0 3-1/2" floppies |
| 03:40.54 | iCEBrkr | Damin!! |
| 03:41.29 | dlynes | We actually still have a machine running OS/2 4.0 |
| 03:41.37 | Damin | iCEBrkr: Remember when you freaked out when you told me you managed to get all your BBS sessions up and running on the OS/2 box? WTF was Shockwave coded in? Pascal? :) |
| 03:41.52 | iCEBrkr | Turbo Pascal BABY! |
| 03:42.01 | iCEBrkr | Then I ported it to Virtual Pascal for OS/2 |
| 03:42.21 | iCEBrkr | and THEN! and THEN! Ported to whatever that Free/Fast? Whatever Pascal for Linux! |
| 03:42.34 | Damin | You are like the only person in the world who has actually produced something that works using Pascal. :) |
| 03:42.42 | [TK]D-Fender | Damin : RemoteAccess was in TurboPascal, as was another whose name escapes me. |
| 03:42.50 | iCEBrkr | and of course I had to write a telnetd type deal for it for Linux |
| 03:42.51 | Nugget | telnet is eeeeeeevil! |
| 03:43.08 | iCEBrkr | Nugget: It was a BBS! Come'on |
| 03:43.23 | iCEBrkr | T0p s3kr3t c0d3z |
| 03:43.25 | Damin | iCEBrkr: I remember that. That was about the time I passed one of those Socket mirroring programs from Heller.. :) |
| 03:43.39 | iCEBrkr | haha |
| 03:43.43 | [TK]D-Fender | Damin : I was a DOS God thanks to TurboPascal... I've written all kinds of stuff, file managers, etc... comm programs... so on. and I still USE them :) |
| 03:43.52 | iCEBrkr | [TK]D-Fender: I think we all did |
| 03:43.56 | iCEBrkr | COLOR DIR! |
| 03:44.04 | dlynes | [TK]D-Fender: Pro-C was good for that, too :) |
| 03:44.10 | iCEBrkr | Progress meter + bit-compare copy! |
| 03:44.28 | dlynes | [TK]D-Fender: it was sorta the same thing, but it was a C source code generator |
| 03:44.29 | iCEBrkr | 'More' on ANSI-Crack! |
| 03:44.31 | distortion | ok, you guys crossed into uncharted nerd territory |
| 03:44.36 | iCEBrkr | lol |
| 03:44.39 | [TK]D-Fender | iCEBrkr : Yup.. QCD (quick change directory). It'd scan a tree and search based on partial path. immediate jump for single result, pop-up color select box for multiple. |
| 03:44.49 | iCEBrkr | yup yup |
| 03:44.54 | dlynes | yeah...qcd kicked ass |
| 03:44.59 | iCEBrkr | haha Renegade! |
| 03:45.01 | [TK]D-Fender | distortion : Thats the one! |
| 03:45.07 | [TK]D-Fender | distortion : Renegade! |
| 03:45.09 | [TK]D-Fender | th |
| 03:45.10 | distortion | haha |
| 03:45.11 | [TK]D-Fender | x |
| 03:45.22 | iCEBrkr | Renegade, Telegard, pffft, Searchlight.. They were all TP |
| 03:45.41 | dlynes | UltraBBS/ProBBS was pretty cool, too |
| 03:45.46 | Damin | Hahahahhah... |
| 03:45.47 | iCEBrkr | I have the source to Searchlight :X |
| 03:45.47 | *** part/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 03:45.47 | distortion | oblivion i think it was called was sweet |
| 03:45.54 | iCEBrkr | Fuck Obv/2 |
| 03:45.59 | iCEBrkr | and fuck Eric Katz |
| 03:46.04 | [TK]D-Fender | iCEBrkr : I was working on a massively modular one built anround a language I wrote so its like "live-code" your own run-time BBS :) |
| 03:46.05 | iCEBrkr | Sorry, still a bit bitter |
| 03:46.13 | Damin | I have the source to FutureVision and METAL, and the freaking Apple IIGS to RUN IT ON! :) |
| 03:46.20 | iCEBrkr | METAL! |
| 03:46.35 | iCEBrkr | [TK]D-Fender: Yea, I ended up writing my own scripting lanaguage for ShockWavE:PRO |
| 03:46.50 | iCEBrkr | Basically an API to all the internal functions |
| 03:46.52 | Damin | iCEBrkr: Did I ever tell you that I met Joshua Thompson, FINALLY, after like 14 years at Astricon in Atlanta? |
| 03:46.54 | [TK]D-Fender | Damin : I had the good sense to TRASH my gear when the getting was good to get out and upgrade :) |
| 03:47.02 | iCEBrkr | Damin: LOL! Now ay |
| 03:47.04 | iCEBrkr | no way |
| 03:47.08 | Damin | iCEBrkr: Yeah... |
| 03:47.22 | *** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 03:47.23 | iCEBrkr | Damin's BBS was unpredictable. |
| 03:47.31 | Damin | stkn: I'll never, fucking EVER sell or get rid of my Apple IIs... |
| 03:47.33 | iCEBrkr | You'd call it and it did something different every time. |
| 03:47.43 | iCEBrkr | Including the [C]rash the Board |
| 03:48.06 | Damin | iCEBrkr: That is cause I was always testing the boundaries of shit.. |
| 03:48.10 | [TK]D-Fender | iCEBrkr : My language started as a quick string parser for a COMM program, them gained nested functions, then code-flow control and became a whole damn language! The worlds first MID LEVEL language!!!! muahahaha |
| 03:48.18 | [TK]D-Fender | iCEBrkr : Wanna see some code? |
| 03:48.39 | [TK]D-Fender | 2 sec |
| 03:48.40 | iCEBrkr | I just rode my motorcycle clear across Florida.. |
| 03:48.41 | Damin | iCEBrkr: Remember the online Prefix Scanner? :) |
| 03:48.48 | iCEBrkr | Damin: Yea! |
| 03:49.10 | Damin | That rocked.. I still have the code for it somewhere.. |
| 03:49.11 | distortion | all the phreakers all growns up |
| 03:49.21 | iCEBrkr | I installed DosBox on my laptop.. ShockWavE:PRO runs in it! |
| 03:49.30 | Damin | Hahahahah.... |
| 03:49.31 | iCEBrkr | Toneloc! |
| 03:49.45 | iCEBrkr | oh! iCE Breakers War Dialer seems to work too. |
| 03:50.18 | Damin | iCEBrkr: You know how busted you would be if you dropped one of those paging prefix war-dialers on someone these days? :) |
| 03:50.30 | iCEBrkr | lol |
| 03:50.50 | Damin | That had to be the funniest idea of that period that I can remember.. |
| 03:50.55 | iCEBrkr | :D |
| 03:50.59 | Damin | Nothing quite tops it.. |
| 03:51.01 | iCEBrkr | CallerID sucked |
| 03:51.07 | iCEBrkr | No way to block it back then |
| 03:51.24 | iCEBrkr | and besides, *53 or whatever flagged the last call for investigation |
| 03:51.32 | iCEBrkr | So why not let OTHERS do the dialing for ya?! |
| 03:51.48 | Damin | Yep... :) |
| 03:52.25 | Damin | iCEBrkr: You know.. you could have gone all out with the number prefixes as well.. like writing some randomizing functions and such.. |
| 03:52.45 | [TK]D-Fender | iCEBrkr : http://pastebin.com/708828 :D |
| 03:53.02 | iCEBrkr | Damin: it did random dial. I wrote some crazy bitshifting shit to flag numbers as 'dialed' |
| 03:53.08 | iCEBrkr | So it wasn't sequencial |
| 03:53.14 | Damin | iCEBrkr: I.e. to let you define multiple blocks of pager prefixes and then let you randomly bounce calls between... |
| 03:53.22 | iCEBrkr | Oh it did. |
| 03:53.32 | Souvent22 | hm...my sip proxy is giving me an error saying 'Unkown User agent'....it seems it will register though when user=phone |
| 03:53.42 | iCEBrkr | Souvent22: Kick it! |
| 03:53.44 | Damin | iCEBrkr: Heheheh.. |
| 03:53.51 | Souvent22 | is there a way to have asterisk register with sip:1012@host;user=phone |
| 03:54.01 | Damin | iCEBrkr: So you get anywhere on the job front? |
| 03:54.12 | iCEBrkr | Damin: Interview here in Boca Raton tomorro at 11:30 |
| 03:54.13 | Souvent22 | iCEBrkr: I'd love to...unfortuantley, it's safe at the office and i'm working on it remotly. |
| 03:54.21 | Souvent22 | however, i'll be sure to give it a boot in the morning. :) |
| 03:54.53 | iCEBrkr | [TK]D-Fender: That's crazy. Looks pretty familiar even! |
| 03:55.24 | iCEBrkr | Souvent22: What exactly are you trying to do? A phone/extension isn't registering properly? |
| 03:55.24 | Damin | iCEBrkr: Supposed to be a lot of opportunities in florida.. most people are too old down there to understand computers.. :) |
| 03:55.30 | iCEBrkr | lol |
| 03:55.44 | iCEBrkr | Damin: I really hate to move to the EASTSIIIIIIIIDE! But I gotta do what I gotta do |
| 03:55.46 | [TK]D-Fender | iCEBrkr : I wrote the language in TP, and its a hybrid between TP style functions (actually PROCEDURES is a better term), and Assembler level loop / Goto control :) |
| 03:55.53 | iCEBrkr | [TK]D-Fender: Same here. |
| 03:55.59 | iCEBrkr | :Lables |
| 03:56.05 | iCEBrkr | err :Labels even |
| 03:56.12 | Souvent22 | yeah....trying to get asterisk to register to a sip proxy. |
| 03:56.19 | iCEBrkr | and I had @JMP commands instead of GOTO |
| 03:56.33 | iCEBrkr | Souvent22: Oh. I haven't tinkered with SIP proxies. |
| 03:56.41 | Souvent22 | is there a command to see the exact SIP command that asterisk is sending? |
| 03:56.55 | iCEBrkr | Souvent22: tcpdump? :D |
| 03:56.55 | docelm0 | ICE! |
| 03:56.58 | [TK]D-Fender | iCEBrkr : I use JNE etc for conditionals :) |
| 03:56.59 | iCEBrkr | docelm0: werd homie |
| 03:57.04 | iCEBrkr | [TK]D-Fender: haha |
| 03:57.11 | iCEBrkr | [TK]D-Fender: Damn, ASM! |
| 03:57.15 | Damin | tkdefender: Check this shit out.. :) |
| 03:57.36 | Souvent22 | lol. true. |
| 03:58.09 | docelm0 | nada.. SSDD just being busy |
| 03:58.11 | iCEBrkr | http://www.nacs.net/~icebrkr/shckwave/scripts/reference.html |
| 03:58.16 | iCEBrkr | HOLYSHIT |
| 03:58.21 | iCEBrkr | that's still up there?! |
| 03:58.34 | docelm0 | Damin never cleans house |
| 03:58.41 | docelm0 | except of beer |
| 03:58.42 | docelm0 | :) |
| 03:58.46 | iCEBrkr | lol |
| 03:58.50 | Damin | Hehehe.. http://pastebin.com/708829 |
| 03:59.00 | [TK]D-Fender | iCEBrkr : heres another sample where I was templating how I would invent a "loop" function in a more natural way (commented) http://pastebin.com/708830 |
| 03:59.07 | iCEBrkr | [TK]D-Fender: load that URL |
| 03:59.10 | docelm0 | I mean you did pass out under a piano.. :) |
| 03:59.10 | iCEBrkr | lol |
| 03:59.26 | docelm0 | but whats better is Kristian passing out in my front yard.. |
| 04:00.04 | Damin | Hahahah.. |
| 04:00.06 | iCEBrkr | [TK]D-Fender: That's some crazy shit |
| 04:00.20 | Damin | docelm0: I tried calling his cell phone tonight.. i was going to give him shit about Baltimore last year.. |
| 04:00.41 | iCEBrkr | [TK]D-Fender: http://www.nacs.net/~icebrkr/shckwave/scripts/timebank.html |
| 04:00.43 | iCEBrkr | lol |
| 04:00.43 | Damin | docelm0: But I got some weird guy that answered and was like.. "How yall doin?" |
| 04:01.12 | docelm0 | he is from corn country |
| 04:01.56 | docelm0 | hay D msg me your email.. I will send you kris passed out on my front yard |
| 04:02.11 | iCEBrkr | I swear, I nuked my nacs.net public_html |
| 04:02.27 | iCEBrkr | Damin probably restored it from some old backup when rebuilding a machine or something |
| 04:02.46 | docelm0 | haha |
| 04:02.48 | docelm0 | who knows |
| 04:02.59 | docelm0 | question is.. DOES your pass still work? |
| 04:03.05 | iCEBrkr | I dunno |
| 04:03.12 | iCEBrkr | hax0r it |
| 04:03.55 | Damin | Hahaha.. http://pastebin.com/708833 |
| 04:03.59 | *** join/#asterisk babyju (n=babyju@c-68-39-244-238.hsd1.nj.comcast.net) |
| 04:04.20 | Damin | I remember that when I wrote that code, I was up for like 4 days straight drinking nothing but black coffee... |
| 04:04.56 | [TK]D-Fender | Damin : Cool! |
| 04:05.08 | iCEBrkr | It's funny how all these BBS scripting languages are kinda alike |
| 04:05.19 | Damin | I HAD to be the first one to have some useful code for Wilson's binary mapped modem access code that let you load binary data from the serial port.. |
| 04:05.46 | docelm0 | Damin what BBS did you run? |
| 04:05.52 | babyju | Can anyone recommend an asterisk based pbx provider? |
| 04:05.54 | Damin | iCEBrkr: He had some weird, weird, strange ass character buffering and checksumming crap going on in his serial drivers.. |
| 04:06.06 | Damin | docelm0: Damage Incorporate |
| 04:06.07 | Damin | docelm0: Damage Incorporated |
| 04:06.27 | docelm0 | hmm.. never heard of that one.. I ran Renegade (Telegard) hack |
| 04:06.57 | iCEBrkr | Man. I remember writing the modem drivers for ShockWavE |
| 04:07.18 | iCEBrkr | All that shit was a PITA |
| 04:07.25 | iCEBrkr | Local display, remote display |
| 04:07.28 | Damin | docelm0: Damage Incorporated was FutureVision #1@#6 on the Futurenet, running METAL on an Apple IIGS (that was running an amped up accelerator card at 14mhz!) |
| 04:07.48 | Damin | docelm0: It was also known as "damage.cleve.fnet.org" ;) |
| 04:08.51 | docelm0 | ohh.. I started out with PC.. :) |
| 04:09.09 | docelm0 | hay I got that pic of Kris.. want it? |
| 04:09.11 | Damin | docelm0: Shit.. I was overclocking while you were still messing around in diapers.. ;) |
| 04:09.19 | Damin | docelm0: yeah!!! |
| 04:09.19 | iCEBrkr | haha |
| 04:09.26 | docelm0 | send me your email or IM |
| 04:09.38 | Damin | damin@damage.clevel.fnet.org! ;) |
| 04:09.41 | iCEBrkr | lol |
| 04:09.52 | Damin | Just kidding.. I think Shane let fnet.org lapse.. |
| 04:09.53 | iCEBrkr | Damin: That shit still works? |
| 04:09.59 | babyju | Anyone have a recommendation for buying preinstall asterisk |
| 04:09.59 | iCEBrkr | I was gonna say |
| 04:10.13 | docelm0 | babyju dont use A@H |
| 04:10.16 | Damin | babyju: Asterisk binary edition.. www.digium.com |
| 04:11.47 | babyju | Damin I would like to get asterisk already install out the box for the hardware |
| 04:13.04 | docelm0 | babyju I can build you one |
| 04:13.12 | docelm0 | Where you located? |
| 04:15.27 | Damin | Hahahahahahhaha.. |
| 04:15.34 | Damin | That is freaking hilarious.. |
| 04:16.05 | docelm0 | what? |
| 04:16.12 | Damin | Kielhofner.. |
| 04:16.18 | Damin | Passed out on the lawn.. |
| 04:16.26 | docelm0 | I say we clean it up a bit blow it up and bring it to astricon |
| 04:16.29 | Damin | You have to tell me what led up to that.. |
| 04:16.54 | docelm0 | Vodka/RedBull(8), 5th of some fruity shit |
| 04:17.09 | docelm0 | ohh and he blew chunks in my car.. |
| 04:17.14 | Damin | docelm0: Was it that nasty blue crap called hypnotic? |
| 04:17.19 | docelm0 | yes |
| 04:17.24 | Damin | Dude.. |
| 04:17.33 | iCEBrkr | eeew |
| 04:17.43 | docelm0 | ok maybe good going down.. but damn did it stink coming back up |
| 04:17.49 | Damin | That was what he was drinking that night at the persian restauraunt when we went out to dinner at Astricon w/ Digium.. |
| 04:17.54 | docelm0 | bitch cant hold his liquor |
| 04:18.18 | Damin | Except.. he brought his own bottle in and was filling up his water glass with it.. :) |
| 04:18.48 | Damin | Only w/ the Asterisk crowd have I ever wandered around town w/ full bottles of Liqour... |
| 04:18.48 | iCEBrkr | haha |
| 04:19.29 | Damin | When we were at Etel, we all stopped at some liqour store and bought vodka pints and got crocked riding around San Francisco on Cable Cars.. :) |
| 04:19.49 | Damin | That was my birdthday.. :) |
| 04:20.10 | Damin | Ohh and we were drinking massive amounts of Irish Coffee... |
| 04:20.24 | MikeJ[Laptop] | it was bad... |
| 04:20.30 | MikeJ[Laptop] | and by bad.. I mean good |
| 04:20.43 | iCEBrkr | haha |
| 04:20.47 | Damin | I had a good time.. |
| 04:21.10 | iCEBrkr | I should have a job by the next Astricon.. I gotta go |
| 04:21.13 | iCEBrkr | GOTTTA!! |
| 04:21.26 | Damin | Nothing beats the Astricon - Las Vegas - ISPcon bender from last year though.. |
| 04:21.41 | Damin | I've never been more tired than that experience.. |
| 04:21.54 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 04:22.01 | Damin | It was almost 2 solid weeks of complete partying.. |
| 04:22.28 | distortion | <-- 3 weeks in miami this year for intellicard and the VPF conference |
| 04:22.39 | distortion | i couldnt remember my name until last week |
| 04:22.54 | Damin | distortion: Those Voice Peering Fabric conferences worth attending? :) |
| 04:23.08 | iCEBrkr | After the wedding in Richmond this pas weekend, I don't want to taste another rum and coke until next year! |
| 04:23.15 | distortion | i was hoping to meet technical folk, it was a big sales push |
| 04:23.20 | Damin | iCEBrkr: I'm like that w/ Gin and Tonic this year.. |
| 04:23.36 | Damin | distortion: Ahhh.. |
| 04:28.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:31.11 | gandhijee | how can i see how many interrupts my TDM400 is generating? |
| 04:31.34 | [TK]D-Fender | gandhijee : The magical 1000/sec! |
| 04:32.14 | Damin | cat /proc/interrupts |
| 04:32.29 | gandhijee | 1000/sec?? |
| 04:32.43 | gandhijee | is there away to check it in real-time? |
| 04:32.56 | Damin | Like with a stopwatch? :) |
| 04:33.05 | *** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 04:33.47 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 04:34.25 | iCEBrkr | http://video.google.com/videoplay?docid=6640225987281635407 |
| 04:34.29 | iCEBrkr | Blam! |
| 04:36.23 | Damin | Hahahaha.. |
| 04:36.28 | Damin | Awesome!! :) |
| 04:36.34 | Damin | Wonder what the backstory on THAT one is? |
| 04:37.01 | iCEBrkr | I wasn't able to listen to it |
| 04:37.09 | iCEBrkr | I'm at a friends house and the baby is snoring |
| 04:37.35 | Damin | Hahahaha.. |
| 04:37.43 | Damin | They have remixes of it!! |
| 04:37.51 | Damin | It's like Star Wars Kid! :) |
| 04:37.59 | gandhijee | Damin: no like i am running asterisk and i see the interrupts go up and down |
| 04:40.26 | [TK]D-Fender | Damin : Sorry, but there is only one SWC, and my province is where the poor loser regrettably calls home... |
| 04:42.39 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:42.39 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:42.48 | Damin | [TK]D-Fender: Oh yeah? Well check THIS Star Wars Kid out! :) |
| 04:42.49 | Damin | http://video.google.com/videoplay?docid=2226508347233603434 |
| 04:44.43 | iCEBrkr | well, I suppose I should crashout |
| 04:45.05 | Damin | Hahahahahahaha.. |
| 04:45.12 | Damin | This is freaking AWESOME! http://video.google.com/videoplay?docid=-2139555376132383479 |
| 04:45.25 | iCEBrkr | The starwars kid remixes? |
| 04:45.32 | [TK]D-Fender | Damin : lol |
| 04:46.07 | [TK]D-Fender | iCEBrkr : yeah, that passed around about a week ago... |
| 04:46.16 | [TK]D-Fender | ice ; f'n awesome :) |
| 04:46.27 | iCEBrkr | oh? it's a new one? |
| 04:47.53 | [TK]D-Fender | iCEBrkr : Yours is a week or two old |
| 04:47.58 | [TK]D-Fender | Apr 16 |
| 04:48.03 | iCEBrkr | Yea, I don't follow that shit |
| 04:48.07 | [TK]D-Fender | It spread like wildfire |
| 04:48.16 | [TK]D-Fender | iCEBrkr : Its in the link :) |
| 04:48.21 | iCEBrkr | My friends browser loads up popurls.com or whatever |
| 04:48.28 | iCEBrkr | and I saw it on there |
| 04:48.47 | [TK]D-Fender | "Pause!" |
| 04:48.57 | iCEBrkr | Oh, the Mario Brothers thing? |
| 04:48.58 | Damin | That is pretty funny.. |
| 04:49.34 | Damin | I've never played the game, and I think it is freaking hilarious that someone would go through that elaborate of a setup for a talent show.. :) |
| 04:49.54 | Damin | It makes me want to play that shit on my Xbox 360.. :) |
| 04:50.17 | iCEBrkr | It's unfortunate the kid is about as graceful as a moose on rollerskates |
| 04:50.24 | iCEBrkr | But yea, it's a sweet setup |
| 04:51.33 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:51.52 | iCEBrkr | SplasPood: 'Sup slut |
| 04:51.53 | iCEBrkr | errr |
| 04:52.18 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:52.57 | brookshire | damin: omg.. hahahahahahaahahha |
| 04:53.14 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:53.21 | ManxPower | The chicken eggs hatched |
| 04:53.36 | MikeJ[Laptop] | ? |
| 04:53.47 | brookshire | http://video.google.com/videoplay?docid=-2139555376132383479 |
| 04:53.49 | brookshire | lol |
| 04:54.08 | ManxPower | MikeJ[Laptop], we have a chicken at the campground. (two actually, and a spawn of satan crowing rooster) |
| 04:54.36 | ManxPower | now we have a bunch of baby ones |
| 04:55.06 | MikeJ[Laptop] | ahhh |
| 04:55.11 | MikeJ[Laptop] | not a metaphor |
| 04:56.20 | ManxPower | I'm just glad I sleep with earplugs or the rooster would be dead by now. |
| 04:57.10 | MikeJ[Laptop] | all I have to say is, xml-rpc is keeping me up... which is worse.. |
| 04:58.10 | ManxPower | There is Ruben, the rooster, Henny Penny, and Liza Minelli. (hey, I didn't name them) |
| 04:58.48 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:58.58 | [TK]D-Fender | brookshire : I AVI'd it for antiquity :) |
| 04:59.18 | MikeJ__ | stupid wifi |
| 05:00.08 | brookshire | mikej: boo! |
| 05:00.33 | MikeJ__ | boo wifi, or boo xml-rpc? |
| 05:01.54 | brookshire | both? |
| 05:02.15 | brookshire | your wifi probably runs on xml |
| 05:02.16 | MikeJ__ | actually the xml-rpc stuff rocks... |
| 05:02.19 | brookshire | that's the problem |
| 05:02.20 | brookshire | lol |
| 05:02.50 | MikeJ__ | staying up to finish up somthing on it makes me sleepy |
| 05:03.06 | asterboy | Anyone bridge the Polycom's Handset and Headset ports? |
| 05:03.11 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 05:06.51 | asterboy | Anyone know if an RJ11 splitter exists? |
| 05:07.13 | [TK]D-Fender | asterboy : same thing basically IINM |
| 05:07.30 | [TK]D-Fender | asterboy : and as both are OUTPUTS, what would that accomplish? |
| 05:07.43 | MikeJ__ | nighty all |
| 05:08.00 | asterboy | I want two active outputs |
| 05:08.11 | asterboy | otherwise it is one at a time |
| 05:08.27 | [TK]D-Fender | asterboy : There are however some RJ>2.5mm adapters, but the power output isn't great and not suitable for call-centers |
| 05:08.41 | [TK]D-Fender | asterboy : un amped = forget about it. |
| 05:09.00 | asterboy | ok, so I'm back to coil pickups |
| 05:09.04 | asterboy | dam |
| 05:09.15 | asterboy | wanted something professional looking |
| 05:09.20 | [TK]D-Fender | asterboy : Get a Plantronics amp and a Polaris split cable. |
| 05:09.28 | [TK]D-Fender | asterboy : Thats professional |
| 05:09.37 | asterboy | checking... |
| 05:09.52 | [TK]D-Fender | asterboy : I've got all this gear currently |
| 05:10.13 | asterboy | any URLs? |
| 05:10.25 | [TK]D-Fender | I run M12 amps, and H263 binaural headsets. |
| 05:10.32 | [TK]D-Fender | asterboy : www.plantronics.com ? |
| 05:10.41 | asterboy | http://headsetplus.com/product99/product_info.html |
| 05:10.59 | [TK]D-Fender | P101 = flimsy |
| 05:10.59 | ManxPower | I use the Plantronics M175 But that's 2.5mm |
| 05:11.25 | asterboy | Is 2.5mm RJ14? |
| 05:11.32 | [TK]D-Fender | asterboy : no |
| 05:11.36 | asterboy | RJ11? |
| 05:11.41 | [TK]D-Fender | asterboy : NO |
| 05:11.46 | *** join/#asterisk dlynes (i=1000@S010600c09f9a0fc4.vc.shawcable.net) |
| 05:11.47 | ManxPower | asterboy, 2.5mm is walkman headset size |
| 05:11.48 | asterboy | oh ya |
| 05:11.49 | [TK]D-Fender | RJ anything si a telephone style connector. |
| 05:12.03 | [TK]D-Fender | ManxPower : Style, but smaller. |
| 05:12.09 | ManxPower | or cellphone headset size, these days |
| 05:12.18 | [TK]D-Fender | exactly it |
| 05:12.35 | asterboy | how do you connect it to the Polycom |
| 05:12.49 | gandhijee | actually 2.5 is cell phone size. |
| 05:12.56 | gandhijee | 3.5 is walkman size |
| 05:13.13 | asterboy | I have some converters from the GXPs |
| 05:13.28 | ManxPower | asterboy, I don't connect to the polycom with a headset |
| 05:13.42 | ManxPower | polycom has a headset compat matrix on their site |
| 05:14.04 | asterboy | The problem is that I don't just need a headset. |
| 05:14.23 | asterboy | I need one that will work at the SAME time as the handset. |
| 05:14.31 | asterboy | It's for a crisis center. |
| 05:14.48 | asterboy | A manager listens in without mic, and hand signals |
| 05:15.02 | asterboy | have to be right next to the agent so no zapbarge |
| 05:15.22 | asterboy | sorry to bore you :P |
| 05:15.33 | [TK]D-Fender | asterboy : get a handest-inline style tap then. |
| 05:16.04 | [TK]D-Fender | asterboy : taht yuo can split but will lose some signal |
| 05:16.21 | [TK]D-Fender | ut I'm sure it can be reasonably compensated |
| 05:16.41 | [TK]D-Fender | ok, I'm fried for tonight.. later all. |
| 05:16.58 | asterboy | thanks TK |
| 05:17.44 | litage | the message "manager.c:575 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'phpagi'" is appearing on my asterisk console. is there a way to get more information about these connection attempts? |
| 05:20.04 | Qwell | litage: tcpdump |
| 05:20.46 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 05:20.59 | Damin | Hahahahahaha http://video.google.com/videoplay?docid=5430343841227974645 |
| 05:21.00 | litage | Qwell: so there isn't a way to have asterisk show more info? |
| 05:22.10 | *** join/#asterisk lorinc (n=ang@caracas-5038.adsl.interware.hu) |
| 05:24.07 | brookshire | litage: like??? |
| 05:24.28 | brookshire | set verbose 923180298301928031 |
| 05:24.30 | brookshire | :D |
| 05:25.01 | *** join/#asterisk alerios (n=alerios@201.244.243.43) |
| 05:35.24 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 05:36.19 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 05:38.33 | Druken | uhg... |
| 05:40.37 | dlynes | ...ghu |
| 05:42.44 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 05:42.59 | Druken | dlynes: don't you ever sleep? |
| 05:43.27 | *** join/#asterisk robbie2 (n=rob@howzat.dsl.onthenet.net) |
| 05:44.03 | dlynes | heh...that movie's hilarious :) |
| 05:44.15 | dlynes | Yeah...it's only 10:30pm here |
| 05:44.51 | Druken | oh |
| 05:45.02 | dlynes | lol |
| 05:45.07 | Druken | just seems your here 24/7 :) |
| 05:45.08 | dlynes | grab your dick and double click for porn' |
| 05:45.13 | dlynes | the internet is for porn :) |
| 05:45.20 | Druken | almost 2am here.... |
| 05:45.21 | dlynes | hilarious movie :) |
| 05:45.30 | dlynes | Druken: Well, our company name is 24/7 :) |
| 05:45.59 | dlynes | Damin: That movie is freaking hilarious :) |
| 05:46.53 | *** join/#asterisk stevedl (n=steve_dl@eth87.tas.adsl.internode.on.net) |
| 05:49.06 | *** join/#asterisk Eggplant (n=none@dsl-350.cascadeaccess.com) |
| 05:49.33 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 05:49.56 | Druken | i hate it when i can't sleep |
| 05:50.54 | Druken | overnight tv is almost as bad as daytime tv |
| 05:51.05 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 05:53.25 | asterboy | Get a bowflex! |
| 05:53.35 | asterboy | and a 51 year old grandmother. |
| 05:53.43 | hardwire | ?? |
| 05:53.47 | hardwire | oh |
| 05:53.49 | hardwire | still |
| 05:53.51 | hardwire | ?? |
| 05:54.06 | asterboy | overnight tv "infomercials" |
| 05:54.11 | Druken | hehe gotta love how thats a CANADIAN thing, but the bitch has an ohio licence |
| 05:54.13 | *** part/#asterisk stevedl (n=steve_dl@eth87.tas.adsl.internode.on.net) |
| 05:54.19 | asterboy | lol |
| 05:54.45 | asterboy | Bowflex is an American product to. |
| 05:55.10 | Druken | yeah, but if you are in canada, you see the canadian commercial |
| 05:55.32 | Druken | which states all over the place that it's made in canada, and blah blah blah |
| 05:55.51 | asterboy | Some Canadians who bought them, had serious failures...the elastics broke and smacked em in the face. |
| 05:55.54 | asterboy | hillarious |
| 05:55.55 | dlynes | who cares? |
| 05:56.06 | dlynes | I just wish they'd get that stupid infomercial off the air |
| 05:56.19 | asterboy | yep |
| 05:56.24 | Druken | :) |
| 05:56.27 | asterboy | but they just love to torture us |
| 05:56.51 | asterboy | guess there are enough "Price is Right" watching, Cake Hole stuffing loosers to buy them |
| 05:56.51 | Druken | could be worse, that abflex is worse.... |
| 05:57.07 | asterboy | abflex, lol |
| 05:57.09 | Druken | their "doctor" can't even read the damn teleprompter right :) |
| 05:57.14 | asterboy | lol |
| 05:57.17 | Druken | it's funny as shit |
| 05:57.20 | asterboy | Remeber PunchMan |
| 05:57.23 | dlynes | or Girls Gone Wild! |
| 05:57.27 | dlynes | and Girls Gone Wild 2! |
| 05:57.29 | asterboy | lol |
| 05:57.58 | asterboy | That's like a short Wild On |
| 05:58.05 | Druken | dlynes: that's your personal pron collection, not tv... |
| 05:58.18 | dlynes | lol |
| 05:58.31 | asterboy | too funny |
| 05:59.13 | asterboy | They really do need clean up late night tv...so much garbage. |
| 05:59.26 | asterboy | ah hell...day time is just as bad |
| 05:59.37 | asterboy | Thats why I killed my sat uplink |
| 05:59.42 | Druken | yeah, especially on those damn alberta channels.... they are the worst :) |
| 05:59.49 | asterboy | 100s of channels and NOTHING to watch |
| 06:00.04 | asterboy | drunken red necks! |
| 06:00.04 | dlynes | Bell ExpressVu's not too bad |
| 06:00.10 | dlynes | Lots of movies |
| 06:00.22 | asterboy | they need to come out with some good movies though |
| 06:00.26 | Druken | i have expressvu.. |
| 06:00.36 | dlynes | Not to mention I can watch one of the tv stations from my hometown on bell |
| 06:00.52 | Druken | oh? |
| 06:01.10 | dlynes | not that the station is any good, but at least i can see who's running what business now, and that kinda thing |
| 06:01.50 | asterboy | Only thing I liked on Canadian TV was the short vingets, (however it's spelt) |
| 06:01.59 | dlynes | vignettes |
| 06:02.04 | asterboy | ya thats it |
| 06:02.10 | Druken | doodles :) |
| 06:03.47 | drray | I like DaVinci's INquest |
| 06:03.58 | drray | nice little slow police potboiler |
| 06:04.19 | Druken | don't think i've ever watched that one... |
| 06:04.19 | asterboy | havn't seen that yet...should be good, just cause the Vatican didn't want it published. |
| 06:04.30 | drray | not the code |
| 06:04.33 | dlynes | ya...da vinci's inquest kicks ass |
| 06:04.39 | dlynes | it's filmed here, in vancouver |
| 06:04.44 | asterboy | sweet |
| 06:04.45 | dlynes | based on the life of our last mayor |
| 06:05.02 | dlynes | Before he became mayor, he was the chief coroner for Vancouver |
| 06:05.09 | dlynes | Larry Campbell |
| 06:05.14 | Druken | i like regenesis, but it's a movie network only thing |
| 06:05.17 | drray | before that he was a cop |
| 06:05.19 | *** join/#asterisk parag7732 (n=root@de2-b15916.alshamil.net.ae) |
| 06:05.20 | drray | yeagh |
| 06:05.22 | drray | -g |
| 06:05.28 | dlynes | ah...didn't know that |
| 06:05.28 | parag7732 | Hey In DISA i m only able to dial local extention ...but when i m dialing outside no...its not dialing ?? I mean therw Zap/1-1 |
| 06:05.42 | dlynes | but then again |
| 06:05.49 | dlynes | i've only been living here for about 11 years or so |
| 06:05.54 | asterboy | that is a seriously interesting rap sheet |
| 06:06.06 | dlynes | parag7732: dood....your dialplan is screwed up then |
| 06:06.19 | dlynes | parag7732: check the context you're dialing out in |
| 06:06.44 | parag7732 | yaa...when i m dialing directly from my local extention any outside no....I m able to dial... |
| 06:06.59 | parag7732 | but not therew DISA |
| 06:07.15 | dlynes | parag7732: because they're probably both using different contexts |
| 06:07.32 | dlynes | parag7732: one has a buggy context, the other doesn't |
| 06:08.12 | dlynes | damn |
| 06:08.17 | orlok | what does CDR mean/stand for? |
| 06:08.18 | dlynes | 2200 containers on one ship |
| 06:08.23 | dlynes | Call Detail Record |
| 06:08.46 | orlok | so thats just used for logging call information, it should not actually make the call fuck up, correct? |
| 06:08.56 | dlynes | orlok: correct |
| 06:09.20 | orlok | cool |
| 06:10.05 | dlynes | even if you've got the wrong permissions set on it or some such fucked up shit, asterisk will still plug along, without disconnecting any calls |
| 06:10.22 | asterboy | nice chat..night all. |
| 06:10.26 | dlynes | ever stupid |
| 06:10.29 | dlynes | precooked bacon? |
| 06:10.30 | dlynes | wtf? |
| 06:10.59 | dlynes | half the fun of eating bacon is smelling it cooking on the stove :(( |
| 06:11.37 | Druken | mmmmmmm, bacon |
| 06:11.39 | faberk | goodnight to you folks... goodmorning to us in europe |
| 06:11.48 | faberk | :) |
| 06:12.00 | drray | dlynes - and the other half is belching it up the rest of the day |
| 06:12.14 | dlynes | and afternoon in asia |
| 06:12.35 | dlynes | drray: the other half of asterisk? |
| 06:12.39 | dlynes | heh...yeah |
| 06:12.39 | faberk | :) |
| 06:13.23 | Druken | hmm... maybe i'll give sleep another try |
| 06:13.27 | drray | sleep is for sissies |
| 06:13.31 | dlynes | Druken: slug back a beer |
| 06:13.38 | dlynes | Druken: beer always helps get you to sleep |
| 06:13.39 | orlok | hmm |
| 06:13.45 | orlok | beer oclock in 45 minutes |
| 06:13.48 | Druken | drray: sleep is for the sex deprived... |
| 06:14.14 | faberk | so... try to rest... |
| 06:14.18 | drray | druken - you are suggesting those groups are mutually exclusive? |
| 06:14.20 | faberk | ;) |
| 06:14.46 | Druken | well, i know i need far less sleep when getting on a regular basis.. hehe |
| 06:15.52 | faberk | well... I'm just up, after 4 hours of middle-sleep |
| 06:15.55 | faberk | :D |
| 06:16.09 | Druken | ditto |
| 06:16.10 | dlynes | holy flipping cow |
| 06:16.20 | dlynes | 5 storey tall diesel engine |
| 06:16.30 | dlynes | 93,120 bhp |
| 06:16.33 | Druken | went to bed around 10... it's now 2, but i've been up since 12:30 |
| 06:16.42 | orlok | in a ship? |
| 06:16.48 | dlynes | yeah |
| 06:16.51 | dlynes | largest ship in the world |
| 06:16.57 | faberk | wow |
| 06:17.08 | dlynes | and it can cruise at 25 knots |
| 06:17.15 | Druken | night peoples |
| 06:17.20 | faberk | night |
| 06:17.22 | dlynes | sucks back ten tonnes of diesel an hour, too |
| 06:17.37 | orlok | christ |
| 06:17.38 | dlynes | so if you thought your suv was a fuel pig |
| 06:17.43 | dlynes | it's got nothing on this ship :) |
| 06:18.20 | orlok | heh, no, i dont.. its gearbox is busted |
| 06:18.30 | orlok | it hasnt drunk any fuel in a few years now |
| 06:18.33 | dlynes | lol |
| 06:18.57 | dlynes | there's a vw/audi sale on next saturday |
| 06:19.05 | dlynes | hopefully i can get a good deal on an a3 :) |
| 06:22.15 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:23.52 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:24.23 | *** join/#asterisk PiPiPi (n=suedoh@ppp-69-237-207-23.dsl.irvnca.pacbell.net) |
| 06:25.26 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 06:26.04 | dlynes | well, if it was a standard, it'd be a lot cheaper to fix :) |
| 06:28.03 | PiPiPi | ifeq ($(CROSS_COMPILE),) |
| 06:28.08 | PiPiPi | whats up with that? |
| 06:28.12 | PiPiPi | line 28 on the Makefile |
| 06:28.16 | PiPiPi | is that supposed to be that way? |
| 06:28.43 | dlynes | ? |
| 06:29.16 | PiPiPi | ok gmake fixed that |
| 06:29.17 | PiPiPi | weird |
| 06:29.36 | dlynes | PiPiPi: you weren't using gmake before? |
| 06:29.39 | PiPiPi | no |
| 06:29.40 | PiPiPi | heh |
| 06:29.45 | PiPiPi | using regular make, im on bsd arch |
| 06:29.49 | dlynes | ah |
| 06:30.00 | dlynes | Yeah...i use slackware...gmake is the default |
| 06:30.14 | dlynes | A lot of linuxes use some other make for the default though |
| 06:31.26 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 06:33.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 06:40.00 | *** join/#asterisk Assid (n=assid@203.115.83.213) |
| 06:40.10 | Assid | woopie |
| 06:40.25 | dlynes | ass woopie |
| 06:40.58 | Assid | wassup |
| 06:41.19 | dlynes | not much |
| 06:41.22 | *** part/#asterisk parag7732 (n=root@de2-b15916.alshamil.net.ae) |
| 06:44.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:48.32 | *** part/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 06:52.40 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 06:56.17 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:05.24 | PiPiPi | is there a Web-GUI based Asterisk config/admin available? |
| 07:05.37 | PiPiPi | thats not Asterisk At Home |
| 07:05.58 | orlok | yeah |
| 07:06.06 | orlok | theres one that runs on top of smeserver called "sail" |
| 07:06.17 | PiPiPi | sail ? |
| 07:06.33 | orlok | Selintra Asterisk Integration or soemthing |
| 07:06.33 | PiPiPi | what is the sme server? |
| 07:06.55 | PiPiPi | oic |
| 07:06.55 | orlok | smeserver is based on Centos, its a standard mail server/nat gateway/web/ftp server system |
| 07:06.57 | PiPiPi | how about AMP ? |
| 07:07.03 | orlok | with avvery nice web interface and templating mechanism |
| 07:07.12 | orlok | AMP is A@H essentially isnt it? |
| 07:07.23 | orlok | you can get A@H/AMP for smeserver as well |
| 07:07.30 | dlynes | a@h uses amp |
| 07:07.47 | PiPiPi | i see |
| 07:07.54 | PiPiPi | how about just isntalling AMP and Asterisk |
| 07:07.55 | PiPiPi | is that fine? |
| 07:08.11 | dlynes | a@h is basically centos preloaded with amp/freepbx, apache, php, probably sendmail, and fop |
| 07:08.17 | drray | how about just spending 20 minutes and learning the confs? |
| 07:08.23 | PiPiPi | drray, i am |
| 07:08.27 | drray | amp stinks |
| 07:08.29 | drray | imo |
| 07:08.29 | PiPiPi | i'd like to get all things installed while im at it |
| 07:08.36 | drray | not a poor reflection on you personally |
| 07:08.48 | PiPiPi | i want to get astguiclient and vicidial working also |
| 07:08.53 | PiPiPi | im just getting a feel of asterisk right now |
| 07:08.57 | PiPiPi | and all available tools in front of me |
| 07:08.57 | drray | I don't even use FOP anymore |
| 07:09.01 | PiPiPi | whats' FOP? |
| 07:09.12 | PiPiPi | what about gastman? |
| 07:09.13 | drray | flash operator panel |
| 07:09.40 | PiPiPi | eew gastman |
| 07:09.40 | PiPiPi | heh |
| 07:10.04 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:10.42 | dlynes | gastman's just a command line monitoring tool or something for asterisk, right? |
| 07:10.49 | dlynes | erm nvm |
| 07:10.51 | dlynes | gnome |
| 07:11.00 | dlynes | it's the gui version of astman, i'm guessing |
| 07:13.20 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:15.47 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
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| 07:27.38 | supjigatr | Anyone know how to get G729a working between asterisk and a tnt? I have G711 working but even though I have enable the g729a in the tnt calls fail if * is forcing g729a. |
| 07:28.26 | *** join/#asterisk jsgotangco (n=jerome@ubuntu/member/jsgotangco) |
| 07:29.19 | *** join/#asterisk linstar (n=achu@220.225.191.49) |
| 07:29.24 | linstar | hi |
| 07:29.37 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:29.38 | supjigatr | Hi. |
| 07:30.03 | linstar | How to configure asterisk so that one extension can be used in two phones? |
| 07:30.40 | supjigatr | sip? |
| 07:31.35 | linstar | yes |
| 07:31.49 | qdk | linstar: make the phones have the same context? |
| 07:31.53 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 07:32.13 | linstar | qdk : in sip.conf? |
| 07:32.52 | qdk | linstar: if you have SIP-phones, that would be the place. |
| 07:33.23 | linstar | qdk: everything is looking identical but when I register two phones with same extension the last registered only rings |
| 07:34.25 | linstar | qdk:is any other idea to ring both phones simultaneous with one sip account ? |
| 07:34.29 | qdk | linstar: A device has NOTHING to do with an extention, so i think you lack some understanding of *. |
| 07:35.46 | qdk | linstar: you dont... you need an "account" pr. phone. an then you dial multi SIP IDs with the Dial() command. |
| 07:36.59 | qdk | linstar: you need something like this: sip.conf <- [SIP1] ,,,, [SIP2] and in extentions.conf <- exten => s,1,Dial(SIP/SIP1&SIP/SIP2) |
| 07:39.23 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 07:40.03 | linstar | qdk :my configuration is like this , exten => 2002,1,Macro(exten-vm,2002,2002) |
| 07:40.18 | linstar | exten => ${VM_PREFIX}2002,1,Macro(vm,2002) , exten => 2002,hint,SIP/2002 |
| 07:43.49 | qdk | linstar: yes, and then you only dial 1 divice. |
| 07:43.51 | qdk | device* |
| 07:44.09 | qdk | linstar: google -> asterisk cmd dial |
| 07:46.34 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:47.18 | linstar | qdk:you are telling about call sharing between extensions right? |
| 07:48.16 | qdk | linstar: depends on what you mean by call sharing? |
| 07:48.41 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 07:49.55 | linstar | qdk: Understand my qn pls, I have one sip account say 2002 and want to use it in two lines |
| 07:50.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:50.46 | linstar | qdk : when I try to call from 2003 to 2002 then it rings only in one line and the other line kept silent |
| 07:51.16 | kay2 | linstar: Dial(SIP/phonea&SIP/phoneb) |
| 07:51.17 | linstar | qdk: I want to make active both the phone lines when a call come to 2002 |
| 07:51.42 | qdk | linstar: you seem to be having a hearing disability. |
| 07:51.50 | qdk | linstar: you need something like this: sip.conf <- [SIP1] ,,,, [SIP2] and in extentions.conf <- exten => s,1,Dial(SIP/SIP1&SIP/SIP2) |
| 07:52.17 | kay2 | linstar: otherwise you use Agent |
| 07:52.21 | kay2 | u put them together |
| 07:52.29 | kay2 | and you queue() the user |
| 07:52.38 | kay2 | and in the setting of the queue u put both phone |
| 07:52.51 | kay2 | so both would ring |
| 07:52.55 | qdk | kay2: dont confuse him even more... agents still need 1 unique SIP ID pr. phone. |
| 07:53.11 | qdk | i = one |
| 07:53.24 | qdk | doh... it was a 1. :-) |
| 07:53.25 | kay2 | linstar: otherwise, you rewrite your own app_dial.c |
| 07:53.47 | kay2 | qdk: like that he won't be confused |
| 07:53.50 | kay2 | ;) |
| 07:54.59 | qdk | kay2: well agents, queues, rewrite c-code doesnt help im understand and configure a unique SIP ID pr. phone. |
| 07:55.05 | qdk | im = him |
| 07:55.15 | kay2 | qdk: I was being sarcastic |
| 07:55.41 | qdk | kay2: ok, from which line? |
| 07:55.42 | *** join/#asterisk lately (n=doug@15.5.233.220.exetel.com.au) |
| 07:55.54 | kay2 | [09:56] <kay2> linstar: otherwise, you rewrite your own app_dial.c |
| 07:56.01 | litage | hey guys, i've got a bit of a quandary happening. when i call North America on my GXP-2000 with Asterisk and one particular voip provider, as soon as the callee picks up, i'm disconnected. however, if i remove Asterisk from the equation, the problem disappears. any thoughts?: http://rafb.net/paste/results/YAArbW87.html |
| 07:56.29 | qdk | kay2: ok. |
| 07:56.54 | kay2 | linstar: reinvite=no |
| 07:56.57 | PiPiPi | anybody got AMP working on FreeBSD ? |
| 07:57.00 | kay2 | maybe the reinviting doesnt work |
| 07:57.11 | kay2 | in your sip.conf |
| 07:59.49 | linstar | kay2 : OK |
| 08:02.06 | linstar | kay2 : I am using AMP for most of the configuration |
| 08:02.23 | kay2 | linstar: change it manually then |
| 08:02.32 | kay2 | so we see if it works |
| 08:02.38 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 08:03.01 | linstar | kay2: pls have patience to tell me deeply abt the configuration |
| 08:03.25 | linstar | kay2 : I have extensions.conf and extensions_additional.con |
| 08:03.36 | kay2 | open sip.conf |
| 08:03.42 | kay2 | find reinvite=yes |
| 08:03.47 | kay2 | make it reinvite=no |
| 08:03.54 | PiPiPi | anybody got FreeBBX running on FreeBSD? |
| 08:03.55 | kay2 | if there is a ";" in front of it, remove it |
| 08:03.56 | kay2 | that's it |
| 08:05.11 | kay2 | and if u can't then use ethereal and read rfc of SIP |
| 08:05.12 | qdk | pigpen: isnt it just a webfrontend? which makes the platform as good as irrelevant. |
| 08:10.32 | *** join/#asterisk littlejohn (n=little@host158-76.pool8711.interbusiness.it) |
| 08:11.37 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
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| 08:15.44 | satlan32 | hi |
| 08:15.53 | satlan32 | i need help deisgning something |
| 08:17.10 | satlan32 | i want to call an extension, get a dial tone, dial another number on the keypad, and then dial this number and another number i have (so they will be in conerence) |
| 08:18.01 | satlan32 | what i want to know is how to generate the dial tone again, and how to store the number the user dial |
| 08:22.07 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 08:22.44 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
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| 08:25.47 | *** part/#asterisk jsgotangco (n=jerome@ubuntu/member/jsgotangco) |
| 08:29.36 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-247-72.dsl.snantx.swbell.net) |
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| 08:38.35 | tzafrir | satlan32, the current extension is also the vaiable EXTEN (${EXTEN}) . You can use it in the dialplan logic. |
| 08:39.02 | tzafrir | BTW: any other .il users here? check #asterisk-il |
| 08:39.56 | xbit` | my background cmd dont return and continue executing more commands. it is a bug or a feature i can control? |
| 08:42.57 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 08:43.10 | AsteriskAlbania | Hi there |
| 08:43.25 | AsteriskAlbania | I need some help with asterisk to suport video with EyeBeam |
| 08:43.48 | AsteriskAlbania | I am reading about Asterisk HEAD 02/25/2005 but I dont knwo if it is needed or how to use it ? |
| 08:44.22 | AsteriskAlbania | anyone can help ??? |
| 08:46.01 | qdk | asterboy: why are you messing about with a year old CVS HEAD version of Asterisk? |
| 08:46.18 | qdk | AsteriskAlbania: even |
| 08:46.50 | qdk | AsteriskAlbania: and no, i know nothing about Asterisk and Video. |
| 08:47.03 | AsteriskAlbania | Thats what I'm wondering . it 2005 |
| 08:47.19 | AsteriskAlbania | does asterisk need it now or it is already implemented in it |
| 08:47.30 | AsteriskAlbania | I am running the latest version 1.2.7.1 |
| 08:47.52 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 08:48.39 | qdk | AsteriskAlbania: 1.2.7.1 should defently contain all the good features of a year old CVS HEAD. |
| 08:49.32 | qdk | AsteriskAlbania: i guess its just a matter of a common protocol to communicate by. |
| 08:50.00 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-59-94.dsl.tul2ok.sbcglobal.net) |
| 08:54.33 | AsteriskAlbania | qdk: If I not add videosupport=yes I see at cli asterisk complaining about uknown format, if I enable videosuport asterisk do nothing when I send video request |
| 08:54.55 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 08:55.06 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 08:55.11 | hwt | May 10 10:45:46 NOTICE[23007]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. |
| 08:55.18 | hwt | how do i turn that off in xlite (linux)? |
| 08:56.18 | *** part/#asterisk drray (n=drray@c-67-183-123-24.hsd1.wa.comcast.net) |
| 08:57.23 | qdk | AsteriskAlbania: ok, dont know the video-specific stuff, so cant help you with that... I know Ahrimanes knows a lot about it, but he is probably busy. |
| 08:57.32 | Ahrimanes | i am? |
| 08:58.05 | qdk | Ahrimanes: hehe... well you tell you boss that. :-) |
| 08:58.10 | qdk | your* |
| 08:58.52 | *** join/#asterisk Faithful (n=Faithful@202.6.145.116) |
| 08:59.19 | Ahrimanes | qdk: of course i do :) |
| 08:59.50 | Ahrimanes | AsteriskAlbania: what do you need to know about video? getting it to work with eyebeam? |
| 09:00.05 | *** join/#asterisk prog (n=prog@gen005-28kix.kh-net.cz) |
| 09:01.03 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 09:02.11 | AsteriskAlbania | I am trying to send video to the other eyebeam and it does not show |
| 09:02.35 | AsteriskAlbania | Ahrimanes: is it a problem that the other customer dont have video installed on his comp? |
| 09:03.01 | Ahrimanes | AsteriskAlbania: have you allowed codecs h261 and/or 263 in the sip.conf and in the eyebeam setups? |
| 09:03.01 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
| 09:03.25 | AsteriskAlbania | Ahrimanes: yes |
| 09:03.49 | Micetto | hi |
| 09:04.01 | AsteriskAlbania | disallow=all |
| 09:04.01 | AsteriskAlbania | allow=alaw |
| 09:04.01 | AsteriskAlbania | allow=h261 |
| 09:04.01 | AsteriskAlbania | allow=h263 ; H.263 is our video codec |
| 09:04.01 | AsteriskAlbania | allow=h263p ; H.263p is the enhanced video codec |
| 09:04.01 | AsteriskAlbania | dtmfmode=rfc2833 ; |
| 09:04.01 | Micetto | ^_^ |
| 09:04.09 | Micetto | I have a problem with queue |
| 09:04.22 | Micetto | How can I make a circular queue with asterisk ? |
| 09:05.38 | Ahrimanes | Micetto: circular? |
| 09:05.44 | Micetto | yes, circular |
| 09:05.45 | Ahrimanes | AsteriskAlbania: and in the eyebeam setup? |
| 09:05.53 | Micetto | first 201, secondo 202 and so on |
| 09:06.07 | catch23 | just solve it with AGI, you can solve just about anything there and so much easier too :P |
| 09:06.24 | qdk | Micetto: so you want people to be able to skip the line? |
| 09:06.46 | Ahrimanes | Micetto: ah the order it calls the agenst? |
| 09:06.57 | Micetto | Ahrimanes: yes! :D |
| 09:06.58 | qdk | catch23: and you can kill your performance while you are there as well. |
| 09:07.18 | AsteriskAlbania | Ahrimanes: yes , I have them at settings-> video-> advanced -> codecs |
| 09:07.31 | Micetto | i try with penalty field...but doesn't work :( |
| 09:08.09 | catch23 | qdk: bah, i prefer clean implementation over preoptimization. as knuth restated, preoptimization is the root of evil |
| 09:08.16 | Ahrimanes | Micetto: hm i guess round-robin is the closest match available right now then |
| 09:08.44 | AsteriskAlbania | Ahrimanes: should the test works if I call myself from eyebeam ? |
| 09:08.55 | qdk | catch23: and you probabbly ues PHP for this clean implementation? |
| 09:09.00 | Ahrimanes | AsteriskAlbania: hm strange, with videosupport=yes, allow=h263 and h263 setup in eyebeam it works nicely here |
| 09:09.06 | catch23 | qdk: nah probably ruby |
| 09:09.18 | qdk | catch23: ok |
| 09:09.20 | catch23 | i hate php |
| 09:09.33 | Ahrimanes | AsteriskAlbania: hm, try setting up the echo application on an extension and call that, it should loop back the video you send to it |
| 09:09.35 | catch23 | the ugliest thing since it was a macro language |
| 09:09.37 | qdk | catch23: hehe |
| 09:09.49 | Micetto | Ahrimanes: :(( |
| 09:10.14 | SheriF_WorK | i have a strange problem ... x-lite can talk with hardphone using G711u and X-lite to X-lite can't talk to eachother even the codecs are the same and it's allowed in asterisk @ sip.conf .. any idea ? |
| 09:10.15 | Ahrimanes | Micetto: but afaik it should work with penalty, so if that doesnt work, maybe you should report a bug on it |
| 09:10.29 | SheriF_WorK | it rings and everything but the 2 users can't hear eachothers |
| 09:10.29 | Micetto | :'( |
| 09:10.30 | Ahrimanes | Micetto: are your agents defined in queues.conf as Member => ? |
| 09:10.37 | AsteriskAlbania | Ahrimanes: how to do that |
| 09:10.55 | qdk | SheriF_WorK: NAT issues? |
| 09:11.17 | Micetto | ehm...I haven't agents....I have this |
| 09:11.28 | Micetto | member => Local/201@default, 1 |
| 09:11.30 | Micetto | and |
| 09:11.40 | Micetto | member => Local/202@default,2 |
| 09:11.43 | SheriF_WorK | qdk:the hardphones and the computers that host x-lite all in the same private network . |
| 09:11.56 | qdk | SheriF_WorK: ok |
| 09:12.57 | SheriF_WorK | qdk: and the Xlite can talk to hardphone but not to othe softphone.. which drive me nuts .. i foreced xlite to use the same codecs that using in the call with xlite to hardphone and didn't work .. so really i'm lost. |
| 09:13.05 | qdk | SheriF_WorK: if you have allready good through the log of the case im out if ideas. |
| 09:13.45 | qdk | SheriF_WorK: yes, it sounds strange. "should" work just as well. |
| 09:14.21 | SheriF_WorK | qdk: i'll try again later i'm already very tired and my head hurts :D |
| 09:14.35 | SheriF_WorK | qdk: thinking of trying another softphones as well |
| 09:15.56 | satlan32 | how can i dial someone, and make him go directly to a confrence room when he answer, without any intervention from his side. |
| 09:15.57 | satlan32 | ? |
| 09:16.01 | qdk | SheriF_WorK: yes, try sjphone or some other software and see if that works. |
| 09:16.54 | SheriF_WorK | qdk: thx for the advice |
| 09:17.23 | Ahrimanes | Micetto: hm maybe the penalty is specific to agents.. not sure |
| 09:17.37 | Ahrimanes | AsteriskAlbania: exten => 123,1,Echo i guess |
| 09:21.14 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 09:22.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:23.15 | AsteriskAlbania | Ahrimanes: -- Executing Echo("SIP/692073737-20fb", "") in new stack and nothing happens |
| 09:23.38 | santoshr | can i give a custom sound if pickup fails |
| 09:29.18 | Ahrimanes | fenlander: ah very true :P |
| 09:29.44 | Ahrimanes | AsteriskAlbania: there's a start video button, i think it's grey or blue.. at the top of the pane where you see your own picture on the webcam pane |
| 09:30.37 | nettie | Hi guys, I have a couple of soundpoints 501's behind NAT, when the internet link fails they wait minutes before registering again. |
| 09:31.09 | Ahrimanes | nettie: sounds like you have a big problem with your internet connection :) |
| 09:31.16 | nettie | Anyone know if there's a way to enable keepalive or some sort of check to be sure they'll register faster please? |
| 09:31.18 | *** join/#asterisk apardo (n=apardo@87.217.145.14) |
| 09:31.40 | nettie | Ahrimanes the problem is not the internet link.. this might be caused by an IOS upgrade |
| 09:31.45 | Ahrimanes | nettie: i should think there a "register expires" or something like that setting in them that you can lower |
| 09:31.57 | Ahrimanes | nettie: ok, just if it happens often it sounds bad |
| 09:32.00 | nettie | I'm not saying it's happening every second.. |
| 09:32.10 | nettie | yeah |
| 09:32.13 | nettie | definitely |
| 09:32.13 | nettie | eheh |
| 09:32.15 | nettie | anyway |
| 09:32.21 | nettie | I'll look for that setting |
| 09:33.53 | qdk | nettie: the only way is lowering the expire in the unit... you cant make any tricks in asterisk due to NAT issues. |
| 09:38.58 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 09:41.54 | cuco | any echo canceller gurus on the house? |
| 09:42.37 | AsteriskAlbania | Ahrimanes: the button start video is active is blue |
| 09:44.36 | AsteriskAlbania | I tested with echo and is working :) |
| 09:45.25 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 09:45.54 | santoshr | cuco: u having echo issue with some specific devices ? |
| 09:46.17 | AsteriskAlbania | Ahrimanes: But I can not send video to the other eyebeam, I will install another camera on the other phone and let you know |
| 09:46.26 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 09:46.27 | nettie | back |
| 09:46.29 | nettie | sorry |
| 09:46.43 | nettie | qdk where should I check that? |
| 09:47.00 | nettie | are you sure the value can be configued? |
| 09:47.02 | nettie | configured? |
| 09:47.17 | Ahrimanes | AsteriskAlbania: yes camera at both ends should be good, but still you should be able to send video, but you do need to press the button to enable sending video in eyebeam on every call |
| 09:47.21 | qdk | nettie: thats entirely up to your choise of product. |
| 09:47.27 | nettie | well |
| 09:47.31 | nettie | it should :p |
| 09:47.34 | nettie | be there |
| 09:47.34 | nettie | eheh |
| 09:47.34 | cuco | santoshr, i need to fix this problem in a more generic issue... but generally, fxo channels (x100p and asteribank fxo) |
| 09:47.42 | Ahrimanes | nettie: theres something about sessiontimers i could see in a pdf.. |
| 09:47.43 | nettie | I'll figure out in the man |
| 09:47.53 | AsteriskAlbania | Ahrimanes: yes I do press it |
| 09:47.54 | nettie | another question.. |
| 09:47.57 | qdk | nettie: the its probably just a matter of finding something about "expire" in the documentation. |
| 09:47.59 | nettie | do I need to qualify them? |
| 09:48.28 | santoshr | actually i was enquiring because i also face an echo issue but the echo goes once the fxo starts connecting the call |
| 09:48.38 | qdk | nettie: no, qualify just add monitoring features and such. |
| 09:48.46 | nettie | well |
| 09:48.47 | nettie | at lest |
| 09:48.53 | qdk | nettie: the product will work either way. |
| 09:48.56 | nettie | doing sip show peers I can see if they're regged or not |
| 09:48.58 | Ahrimanes | AsteriskAlbania: ok |
| 09:49.08 | nettie | that's where I see: UNREACHABLE |
| 09:49.20 | nettie | I monitor the phones with that trick |
| 09:49.25 | nettie | trick/feature |
| 09:49.28 | qdk | nettie: yes, and with qualify you have latency as well. |
| 09:49.31 | Ahrimanes | nettie: yes, qualify is good for that |
| 09:49.58 | qdk | nettie: but asterisk cant help you solve an UNREACHABLE problem due to NAT. |
| 09:50.33 | nettie | qdk sure I get this :) |
| 09:51.01 | *** join/#asterisk KNK_ (n=KNK@KNK.Varna.Net) |
| 09:51.10 | santoshr | cuco: actually i was enquiring because i also face an echo issue but the echo goes once the fxo starts connecting the call |
| 09:51.16 | qdk | nettie: SIP <-> FW (with NAT) <-> Asterisk... if the FW losses its NAT-states the phones are dead in Asterisk until the register again. |
| 09:51.46 | nettie | qdk exactly, that's the reason I need find a way to control registration process |
| 09:52.10 | nettie | exactly what happens after a reload of the router |
| 09:52.23 | cuco | santoshr, i went a little depper, and tested all EC availale on zaptel 1.2.5, i even modified the code to know that they are beeing called. i know that the code tries to run, but it has no effect on the quality of voice. i am stucked at this point. |
| 09:52.59 | qdk | nettie: you could make som 3-party software through ie. AMI which can connect to the VoIP unit and force it to register again, but that require quite an effort. |
| 09:53.24 | lately | Hi. Are there many single port FXO cards available to work with FreeBSD? |
| 09:53.48 | nettie | qdk, I think I'll doubleckeck the manual to see if there's a way to have the units re-register after a "x" number of seconds |
| 09:54.16 | nettie | qdk unfortunately polycom seems to have the support page down |
| 09:54.18 | nettie | atm |
| 09:54.21 | nettie | so I cant download the manual |
| 09:54.52 | qdk | nettie: a locale copy is always a good think. :-P |
| 09:55.23 | nettie | yeah |
| 09:55.23 | nettie | ehehe |
| 09:55.37 | nettie | my bad |
| 09:55.39 | nettie | :) |
| 09:55.41 | qdk | nettie: but im pretty sure there is such a setting... you can set it in you sip.conf as well, but there is no garanti that the unit will honor that. |
| 09:57.03 | Ahrimanes | nettie: http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,4922,00.pdf <- mentions voIpProt.SIP.keepalive.session- |
| 09:57.06 | Ahrimanes | Timers |
| 09:57.46 | *** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net) |
| 09:59.12 | NewSole | any one alive here... use vs.net |
| 10:02.16 | AsteriskAlbania | Ahrimanes: with camera on both ends works |
| 10:02.34 | Ahrimanes | AsteriskAlbania: good :) |
| 10:02.44 | AsteriskAlbania | Ahrimanes: no problems at all except of bandwidth 56 kbps :) |
| 10:02.55 | Ahrimanes | AsteriskAlbania: that's not much for video ;) |
| 10:03.47 | *** join/#asterisk skeffling (n=chatzill@andrew.1ec.aaisp.net.uk) |
| 10:06.15 | skeffling | Hello, I have a E100P card, and zttool is showing BLU/RED alarms, zaptel loads Ok, but can't place calls... This is an old version of zaptel, be was working fine yesterday! |
| 10:06.22 | nettie | Ahrimanes what happens exactly when the phone will declare time rin suppot header? |
| 10:06.31 | nettie | support |
| 10:07.57 | Ahrimanes | nettie: not sure |
| 10:09.27 | *** join/#asterisk telenieko (n=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
| 10:10.43 | telenieko | hi ppl. I have a TE110P, and if the party I call is busy "pri debug span 1" shows hangups cause 17, but the hangup does not arrive to the Dial() so the extension can know about. That happens with any hangup that occurs before the remote party answers, any clues ?? |
| 10:10.59 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:11.35 | *** join/#asterisk sebasatWork (n=Sebastia@ip-5-112.sn2.eutelia.it) |
| 10:13.41 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 10:13.43 | NewSole | any one here... use vs.net... I am having problems with debugging |
| 10:14.39 | sebasatWork | hi, i need a hand with asterisk.. i already configure it and it's works fine, but when receiving a call i get this message |
| 10:14.44 | sebasatWork | Target address 195.62.225.244 is not local, substituting externip |
| 10:14.48 | sebasatWork | That's odd... Got a response on a call we dont know about |
| 10:15.20 | sebasatWork | i'm behind a nat, already configure the nat=yes and exterinip=xxx.xxx.xxx.xxx |
| 10:15.30 | sebasatWork | what am i missing here? |
| 10:18.47 | sebasatWork | anyone? |
| 10:22.47 | qdk | sebasatWork: have you set the locale net as well? |
| 10:23.07 | sebasatWork | yes, localnet = 192.168.1.0/255.255.255.0 |
| 10:23.15 | *** join/#asterisk parag7732 (n=root@de2-b1976.alshamil.net.ae) |
| 10:23.23 | sebasatWork | i'm using asterisk@home |
| 10:23.49 | parag7732 | I want to configure minifone service provider with asterisk....anybody can help me.... |
| 10:25.25 | sebasatWork | qdk, this means that i receive the call, right? so the nat problem is solved? i have some problem in the inbound conf? |
| 10:25.31 | qdk | sebasatWork: ok, no idea then... it just works here... and i dont use externip=. |
| 10:26.06 | *** join/#asterisk linstar (n=achu@220.225.191.30) |
| 10:26.24 | linstar | hi |
| 10:26.34 | qdk | sebasatWork: your SIPs are behind NAT and Asterisk isnt, right? |
| 10:26.51 | *** part/#asterisk linstar (n=achu@220.225.191.30) |
| 10:27.26 | sebasatWork | asterisk is behind a nat, and sip is voip.eutelia.it, an italian company |
| 10:28.08 | sebasatWork | but we got a public IP |
| 10:29.23 | *** join/#asterisk kiddy (n=achu@220.225.191.30) |
| 10:29.26 | kiddy | hi |
| 10:30.15 | kiddy | is asterisk is configured to use one ext for two lines? |
| 10:33.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:34.27 | *** join/#asterisk oink (n=ziga@al.co.ve) |
| 10:35.12 | *** join/#asterisk saftsack (n=saftsack@p54A7F4CE.dip.t-dialin.net) |
| 10:37.17 | oink | Hey, I'm having some weird NAT issues.. I can get my (outside) NATed phones to register to my asterisk which is not natted, I can also make calls from outside to inside, but I can't get calls from inside to my NATed outside phone.. Actually, asterisk doesn't even try to send packets outside.. Any clues ? |
| 10:37.31 | sebasatWork | qdk any hints? |
| 10:38.00 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:38.16 | qdk | sebasatWork: oh, then you shouldnt use NAT setting at all. |
| 10:38.37 | qdk | kiddy: what are you asking? |
| 10:39.04 | kiddy | one extension for 3 phone connections |
| 10:39.14 | qdk | oink: WTF is outside nat'ed? |
| 10:39.53 | kiddy | qdk : 3 phones are kept in 3 places and anyone can take from anyone extension |
| 10:39.55 | oink | qdk: A phone that is outside the asterisk network. And which is NATed. |
| 10:40.09 | qdk | kiddy: could you use just a few more seconds asking the question, so people know a bit you what you are asking? |
| 10:40.21 | oink | I guess a schema would make the topology easier to understand. |
| 10:42.08 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 10:42.48 | qdk | oink: well i for one is lost in your insideout nat'ed asterisk outside phones. |
| 10:42.49 | *** part/#asterisk kiddy (n=achu@220.225.191.30) |
| 10:44.27 | santoshr | what is the general standard key for transfer. i mean to ask which key do most of u use for transfer ? |
| 10:46.12 | sebasatWork | sorry qdk.. maybe i missunderstood the concept... my clients and asterisk are inside a network with NAT, but the router forward all trafiic 5006-5080 and 10000-20000 to asterisk... we have a pubblic IP... but the only problem comes when receiving calls |
| 10:46.24 | sebasatWork | should i set NAT to my phones and Not to asterisk? |
| 10:48.04 | santoshr | i want to be able to use blind and attended transfer both without confusing the users alot. ? currenly i am using '*' for transfer and am confused as to wht to use for blind transfer |
| 10:50.00 | qdk | sebasatWork: with that setup you kinda "eliminate" all NAT. So look at as a non NAT setup. |
| 10:50.30 | sebasatWork | thanks |
| 10:51.18 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 10:51.19 | qdk | sebasatWork: nat=yes in sip.conf never hurts... the NAT is between your asterisk and your provider, and when you have a public IP to the asterisk, you have no NAT issues. |
| 10:52.42 | sebasatWork | thanks a lot |
| 10:53.15 | oink | qdk: Here you go, http://cold.waycom.net/asterisk-nat-problem.png ;-) |
| 10:53.25 | oink | I hope it's clear enough |
| 10:54.09 | oink | A call made by 403 to 101, 102 .. works |
| 10:54.22 | oink | But not from 1xx to 403 |
| 10:55.18 | oink | nat is set to yes everywhere, and asterisk has the correct ip and port in the "sip show peers" table. |
| 10:55.19 | qdk | oink: ok, so no ringing or anything at all? |
| 10:55.25 | oink | qdk: Busy tone |
| 10:55.42 | oink | qdk: asterisk sends back a 404 not found and nothing in verbose or debug logs |
| 10:55.49 | qdk | oink: what does the CLI log say when you dial out? |
| 10:55.49 | *** join/#asterisk adorah (n=FreePBX0@87.68.148.177.cable.012.net.il) |
| 10:55.52 | oink | It doesn't even send a packet to the phone ! :( |
| 10:55.55 | oink | qdk: Nothing |
| 10:56.22 | oink | verbosity and debug level set to maximum |
| 10:56.37 | qdk | oink: and sip show peer 403 shows the phone as "alive"? |
| 10:57.37 | AsteriskAlbania | what TCP and UDP ports should be opened in the server running asterisk ? |
| 10:57.51 | oink | Hrm, I don't see any occurences of alive in it if it's what you asking ;-) |
| 10:57.59 | adorah | Hi I have a serious problem using DISA anyone can help? |
| 10:58.00 | *** join/#asterisk gmaruzz (n=Miranda@myskin.iet.unipi.it) |
| 10:58.01 | oink | let me pastbin the output |
| 10:58.38 | oink | Here you go |
| 10:58.41 | oink | http://pastebin.ca/54718 |
| 10:59.15 | qdk | AsteriskAlbania: feel free to pick anyone you like, just remember you tell your confs it as well. :-D |
| 10:59.56 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-97.buckeyecom.net) |
| 11:00.29 | qdk | oink: Status : Unmonitored <- you dont have qualify=yes on that device, do you? |
| 11:00.58 | adorah | When I call one pstn insert a passwotd get access to the other pstn and call. Then when I disconnect the mobile that initiated the DISA both PSTN still occupied. The pbx doesn"t recognize hangup.. |
| 11:01.42 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 11:01.48 | lme | hi guys |
| 11:02.20 | gambolputty | Hi. Can someone call me via SIP? |
| 11:03.14 | lme | anyone having experience with E1 in israel ? |
| 11:03.19 | lme | & bezeq |
| 11:05.50 | *** join/#asterisk RaYmAn-Bx (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 11:06.05 | santoshr | what is the general standard key for transfer. i mean to ask which key do most of u use for transfer ?currenly i am using '*' for transfer and am confused as to wht to use for blind transfer |
| 11:08.20 | tzanger | transfer should be transfer, blind or attended, not this current nonsense we have |
| 11:08.41 | tzanger | if you transfer and hang up before the called party answers, it's blind. If you don't, you can speak to them and it's therefore attended |
| 11:08.49 | tzanger | (doesn't help you, santoshr, I'm just ranting) |
| 11:09.00 | santoshr | yeah i understand |
| 11:09.46 | santoshr | tht is something which happens in tradtional epbax's. has anyone been able to replicat similar behaviour ? |
| 11:10.05 | tzanger | not yet I don't think |
| 11:10.08 | tzanger | I haven't really tried though |
| 11:10.14 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 11:10.36 | oink | qdk: I don't |
| 11:10.53 | oink | Should I ? |
| 11:12.23 | qdk | oink: yes, try that... then you can see the latency between the SIP unit and asterisk. |
| 11:13.29 | *** join/#asterisk Rick_Hunter (n=rhunter@05-160.008.popsite.net) |
| 11:13.42 | oink | ok |
| 11:14.22 | oink | oh |
| 11:14.48 | oink | I now have a status unreachable |
| 11:15.00 | oink | But I know why. |
| 11:15.04 | *** join/#asterisk axscode (n=axscode@210.213.106.190) |
| 11:15.04 | qdk | oink: yes, thats bad... and the reason. :-D |
| 11:15.07 | oink | ;-) |
| 11:15.18 | oink | let's see.. |
| 11:15.51 | qdk | try "sip show peers" if you dont know that command. |
| 11:16.20 | qdk | and add qualify=yes to you template in sip.conf, so you get latency listed as well.. |
| 11:16.27 | qdk | <- launch |
| 11:16.42 | oink | lunch you mean ? :p |
| 11:16.43 | oink | Ok |
| 11:16.44 | oink | Thanks :) |
| 11:17.14 | qdk | yes, but only if you spell it correctly. :-D |
| 11:18.49 | oink | <PROTECTED> |
| 11:18.54 | oink | Still busy though :o |
| 11:22.19 | santoshr | tzanger: but yes i am intersted in knowing whether others had this same issue and what did they do abt it |
| 11:24.17 | tzanger | santoshr: personally I don't have any issues with it since I don't use it. |
| 11:25.12 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 11:25.15 | santoshr | i googled for some sample features.conf file but to no avail |
| 11:26.25 | *** part/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 11:28.50 | codebreaker | is a latency from 200ms fast enough for voip? |
| 11:29.18 | satlan32 | hi |
| 11:29.20 | satlan32 | question |
| 11:29.24 | hwt | codebreaker: yeah. over 250ms each way makes it hard, though. |
| 11:29.28 | darkskiez | codebreaker: i'd say it'd be irritating |
| 11:30.02 | satlan32 | when i'm setting an extension to record all calls, what is the file name it record the conversation? |
| 11:30.07 | darkskiez | usable but irritating |
| 11:30.10 | codebreaker | okay i will hope it works :) |
| 11:30.21 | *** join/#asterisk littlejohn (n=little@host62-81.pool877.interbusiness.it) |
| 11:31.48 | Assid | actualy i get lags of 350 odd .. |
| 11:31.51 | Assid | does fine even still |
| 11:31.55 | SheriF_WorK | May 10 14:39:50 NOTICE[12479]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'default' <---- where should i look ? i don't have context called default anyway in sip or extension.conf |
| 11:32.12 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 11:34.15 | hwt | SheriF_WorK: you need to have a [default]. |
| 11:34.19 | hwt | SheriF_WorK: but you can leave it empty. |
| 11:34.23 | tzanger | SheriF_WorK: that's exactly what it's saying |
| 11:34.24 | tzanger | hwt: NO YOU DO NOT |
| 11:34.33 | hwt | tzanger: you had to, at least. :) |
| 11:34.45 | tzanger | in fact I congradulate SheriF_WorK for not having [default] and throwing everything into it. |
| 11:34.52 | *** join/#asterisk Modcuts (n=bob@82.133.98.155) |
| 11:35.00 | hwt | tzanger: i just have an empty [default], wfm. |
| 11:35.20 | tzanger | SheriF_WorK: somewhere in your configuration you have something dropping into a default context (either explicitly, or through not having a context => line) |
| 11:35.29 | *** join/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p143.telkom-ipnet.co.za) |
| 11:35.50 | Supercross | hello everybody |
| 11:36.00 | Supercross | just quick question |
| 11:36.09 | Modcuts | Morning all, could anyone recommend a good iax2 softphone? |
| 11:36.16 | SheriF_WorK | tzanger: it's not my installation anyway :P i'm still new into asterisk stuff :D |
| 11:36.16 | Supercross | the tdm2400p cards |
| 11:36.21 | SheriF_WorK | but i have to mantain this server |
| 11:36.26 | Supercross | anybody got any wiring diagrams for the RJ21 connectors? |
| 11:36.52 | tzanger | hwt: yes that'll work, I generally have a matchall _.X that does nothing but Hangup() |
| 11:37.05 | *** part/#asterisk parag7732 (n=root@de2-b1976.alshamil.net.ae) |
| 11:37.08 | tzanger | well and a Nop() so I can see it in the dialplan :-) |
| 11:38.47 | hwt | tzanger: :) |
| 11:47.38 | *** part/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p143.telkom-ipnet.co.za) |
| 11:49.33 | oink | Anyone has a clue regarding my problem ? |
| 11:50.25 | oink | (http://cold.waycom.net/asterisk-nat-problem.png a call from 403 to 101, 102, .. works but not from 101, 102 .. to 403) |
| 11:52.24 | oink | 101 -> 403 = Busy (asterisk answer 404 Not found) |
| 11:54.37 | SheriF_WorK | oink: i think u need to open the ports in the nat router before 403 |
| 11:54.49 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-37-dsl.end.estpak.ee) |
| 11:55.02 | SheriF_WorK | oink: for the inbond interface |
| 11:55.04 | oink | SheriF_WorK: It's done |
| 11:55.10 | oink | SheriF_WorK: and it works: |
| 11:55.12 | oink | 13:19:04 < oink> Status : OK (160 ms) |
| 11:55.24 | SheriF_WorK | oink: ur using SIP ? |
| 11:55.27 | oink | Yes |
| 11:55.33 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 11:55.47 | oink | And asterisk doesn't even send a packet to 403 when I make call from 101 |
| 11:56.19 | SheriF_WorK | oink: him then it's may be the extension.conf |
| 11:56.27 | SheriF_WorK | check the extn => 403 section |
| 11:56.40 | oink | Right |
| 11:56.48 | oink | it's probably there.. |
| 11:58.26 | oink | Thanks! |
| 11:58.28 | oink | It works ;-) |
| 12:00.20 | *** join/#asterisk tamp4x (n=Lab@64.201.13.172) |
| 12:00.39 | SheriF_WorK | oink: glad to help;-) |
| 12:01.45 | SheriF_WorK | oink: what application u used to draw this digram ? |
| 12:02.20 | oink | SheriF_WorK: Visio |
| 12:05.39 | tamp4x | anyone here hhave some pbx that supports t38 pass through that i can temporarly register to? |
| 12:06.15 | *** join/#asterisk chuvak[work] (n=chuvak@ss13.lb4.ltk.com.ua) |
| 12:06.29 | chuvak[work] | |
| 12:06.52 | qdk | tamp4x: asterisk 1.4 will be able to do that, but doesnt help you right now. |
| 12:08.37 | *** join/#asterisk pdunkel (n=pdunkel@213.235.192.21) |
| 12:08.54 | pdunkel | Hi everyone! |
| 12:09.10 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:10.14 | pdunkel | Anyone here know how IAX2 channel names are picked? |
| 12:13.55 | tamp4x | asterisk 1.4 will be able to serve me a cup of coffee in the morning too |
| 12:14.23 | tamp4x | pbxnsip 1.5 supports it but i cant make calls to/from the machine |
| 12:14.26 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:16.35 | AsteriskAlbania | asterisk does not accept 00 as prefix if I dial with 00 it fails instead if I dial withow 00 it calls |
| 12:16.53 | tzanger | AsteriskAlbania: check your dialplan |
| 12:17.05 | AsteriskAlbania | i have at extension _X. and _00. |
| 12:18.42 | AsteriskAlbania | exten => _X.,1,mcc2(${EXTEN})|direct|Dial,Zap/g1/${EXTEN} |
| 12:18.42 | AsteriskAlbania | exten => _00X.,1,mcc2(${EXTEN})|direct|Dial,Zap/g1/${EXTEN} |
| 12:18.46 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 12:18.48 | AsteriskAlbania | is it OK ? |
| 12:19.31 | AsteriskAlbania | I need to cut 00 in the Dial command ? |
| 12:20.49 | tzanger | AsteriskAlbania: you do realize that _X. will match 00 right? |
| 12:22.36 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 12:23.42 | qdk | _00X. is still more specific |
| 12:24.11 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 12:29.56 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:30.27 | *** join/#asterisk eastwolf (i=eastwolf@222.130.187.44) |
| 12:30.47 | eastwolf | hello |
| 12:32.17 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:33.37 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:35.01 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 12:36.07 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 12:36.09 | file | or rather, raises! |
| 12:36.35 | *** join/#asterisk lorinc (n=ang@caracas-2081.adsl.interware.hu) |
| 12:37.17 | file | cuco: have you already done any of the recommended tuning? |
| 12:37.47 | tzanger | heh |
| 12:38.27 | file | (I just read his email) it didn't mention if you mucked with gains at all |
| 12:39.21 | tzanger | you can't call back |
| 12:39.38 | file | darn |
| 12:39.53 | file | why not? |
| 12:40.29 | azeteg | anyone here with experience of using asterisk with h323 channel driver? |
| 12:40.29 | file | azeteg: which one? |
| 12:40.46 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 12:40.55 | file | ack! ctooley! |
| 12:41.07 | azeteg | inaccess |
| 12:41.18 | ctooley | I'm looking at the areski CDR Stats tool, which seems nice, but it doesn't like my php installation. Are there any others out there that people would recommend? |
| 12:41.27 | wasim | didn't their singer die? |
| 12:41.33 | tzanger | file: hahahaha |
| 12:41.48 | ctooley | hey file. How are things on your side of the world? |
| 12:41.48 | AsteriskAlbania | tzanger: Yes I do , I just added it once when I was not sure, now I have deleted :) |
| 12:41.54 | file | great |
| 12:42.06 | file | just got back from lunch, been to this same place for 3 days in a row now |
| 12:42.08 | file | food is delicious |
| 12:42.19 | azeteg | mmm |
| 12:42.22 | azeteg | what did you get? |
| 12:42.24 | ctooley | file, where are you? |
| 12:42.30 | tzanger | file: lunch? |
| 12:42.34 | file | Pisa, Italy |
| 12:42.41 | coppice | in italy? nah. try vietnam :-) |
| 12:42.46 | ctooley | Ah. Must be nice |
| 12:42.51 | tzanger | oh you are in italy? |
| 12:42.53 | file | yes |
| 12:43.14 | tzanger | awesome. how is it? |
| 12:43.16 | file | for astridevcon europe |
| 12:43.22 | file | beautiful! |
| 12:43.27 | ctooley | file, you based out of Huntsville now or are you telecommuting? |
| 12:43.33 | file | telecommuting |
| 12:43.45 | wasim | leaning file of telecommuting |
| 12:43.45 | file | the entire software development team is here though :D |
| 12:44.16 | coppice | will they eat their way through the leaning tower of pizza? :-) |
| 12:44.18 | file | I'd take a picture of the view but I don't know of any way to get it off my camera without grabbing a cable from my hotel room |
| 12:44.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 12:44.47 | ctooley | file, you could pay the 8 bucks to MMS it to me and I could email it from my phone to the mailing list. :) |
| 12:44.54 | file | ha |
| 12:45.22 | file | I don't have MMS here :P |
| 12:45.26 | file | just calling/sms |
| 12:45.30 | coppice | MMS == Much too Much to Send |
| 12:45.48 | file | lemme look in my backpack to see if I have the cable... |
| 12:45.52 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 12:46.45 | file | oej just showed up |
| 12:46.56 | file | I borrowed a PCMCIA SD adapter :D so pics here we come |
| 12:47.12 | *** join/#asterisk eastwolf (i=eastwolf@222.130.187.44) |
| 12:47.38 | ctooley | :) Yay |
| 12:48.03 | ctooley | file, your phone got Mini SD or are you using a real camera? |
| 12:48.12 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.50.Dial1.SanJose1.Level3.net) |
| 12:48.35 | file | real camera |
| 12:49.13 | *** join/#asterisk eastwolf (i=eastwolf@222.130.187.44) |
| 12:49.36 | ctooley | Oh, I meant to ask the channel... Anyone front ended Asterisk with Netrake SBCs? I'm about to order some hardware and wanted to check on that. |
| 12:50.00 | *** join/#asterisk eastwolf (i=eastwolf@222.130.187.44) |
| 12:50.07 | debaser | with my motorola phone from verizon and an office connect kit (which was a $40 usb cable) |
| 12:50.41 | debaser | i can file photos from anywhere with a laptop and a signal |
| 12:51.28 | file | downloaded! |
| 12:51.58 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-252-202.telkomadsl.co.za) |
| 12:52.01 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 12:52.01 | ctooley | The LG phone I've got from Verizon has a 1.3Megapixel camera and MiniSD to store them on and it has bluetooth so I can file them wherever. But, 1.3M on a camera phone is... how should I say it.... not so superb |
| 12:52.32 | coppice | I think shitty is the word you are looking for |
| 12:52.50 | debaser | yeah, images from cameraphones are unacceptable (unless its random people who were right in the middle of a terrorist attack, like in london.) |
| 12:53.20 | coppice | you can get 5M pixel cameras in phones now, but with a nasty mishapen blob of plastic for a lens, the quality lacks something |
| 12:53.24 | file | please be holding |
| 12:53.43 | tzanger | I would like a cell phone without a fucking camera in it |
| 12:53.54 | tzanger | I like my treo, but drop the phone, add a better BT stack and add wifi. |
| 12:53.56 | file | uploading pics of the window from the meeting room where we are |
| 12:54.11 | debaser | camera phones have serious noise problems.. it's not really a megapixel problem. oh, and they don't have anything even close to resembling an acceptable lens. |
| 12:54.16 | tzanger | and make the vibrate stronger, and fix up the software :-) |
| 12:54.31 | file | http://quark.file-radio.com/josh/italy/ |
| 12:54.37 | tzanger | debaser: I'm waiting for someone to put a standard SLR lens mount on a cameraphone :-) |
| 12:54.43 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 12:54.50 | Ahrimanes | just need a windows mobile 5.0 sip client |
| 12:55.12 | coppice | even the couple of phones that have put serious lenses in have big problems. when used as a phone the lens gets gunged in human grease |
| 12:55.19 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:55.52 | tzanger | yep |
| 12:55.57 | debaser | heh, exactly. if you want to take pictures, use a camera. if you want to call someone, use a phone. |
| 12:56.10 | coppice | HTC, who make those qtek phones, are growing at an amazing rate. i think they are probably the only company to ever make money from wince |
| 12:56.17 | file | yeah |
| 12:56.30 | file | those phones are sold by so many carriers |
| 12:56.46 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 12:56.46 | ctooley | My phone was the cheapest one I could buy retail that had bluetooth and ev-do. My dogs at my Nokia that didn't have a camera at all. |
| 12:57.05 | ctooley | s/at/ate/ |
| 12:57.27 | ctooley | :) should have been more specific in my find and replae |
| 12:57.30 | tamp4x | anyone here have some pbx that supports t38 pass through that i can temporarly register to? |
| 12:58.12 | ctooley | file, I'm glad I'm not there. Looking out the window at that would only make me frustrated that I was inside. |
| 12:58.55 | file | so... we sort of created a network spanning a switch and a hub here at this meeting table |
| 12:59.10 | file | you can feel the geekness |
| 12:59.24 | debaser | next event you guys should hire a photographer to document it for you |
| 12:59.44 | file | hehe |
| 13:00.05 | debaser | i could recomend someone, in act. :P |
| 13:00.10 | debaser | in fact rather |
| 13:00.36 | *** join/#asterisk eastwolf (i=eastwolf@222.130.187.44) |
| 13:00.44 | ctooley | Me too... I have this amazing camera that even take calls... |
| 13:01.04 | file | no way! |
| 13:01.05 | ctooley | My photography is unlike anything you've ever wanted to see. |
| 13:01.16 | file | and the bells ring in the background... literally |
| 13:01.58 | *** join/#asterisk k0r3 (n=k0r3@c1-143-8.rrba.isadsl.co.za) |
| 13:02.06 | eastwolf | hello |
| 13:02.13 | eastwolf | yes |
| 13:02.31 | ctooley | Man, PHP/GD sucks. I've never had so many problems making something work |
| 13:02.47 | eastwolf | 你好 |
| 13:03.32 | *** join/#asterisk joerg (n=joerg@p548892A8.dip0.t-ipconnect.de) |
| 13:03.34 | joerg | hi... |
| 13:03.41 | k0r3 | hi, i was wondering if anyone could help me. We have an asterisk box and i need to do call tagging |
| 13:03.45 | joerg | anyone using the latest cvs snapshot of asterisk? |
| 13:04.03 | joerg | I'm experiencing problems with the firmware... |
| 13:04.10 | joerg | chan_iax2.c:1254 try_firmware: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum |
| 13:04.12 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 13:04.13 | eastwolf | 大家好，我新来的，多多照顾。谢了！ |
| 13:05.04 | Nugget | wakirimasen |
| 13:05.52 | Katty | a Nugget |
| 13:06.06 | key2 | Katty: ? |
| 13:06.15 | Katty | key2: yes? |
| 13:06.19 | trelane_ | eastwolf, english please? |
| 13:06.52 | *** join/#asterisk eastwolf (i=eastwolf@222.130.187.44) |
| 13:07.14 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 13:07.41 | eastwolf | ok. I want to hello everyone with chinese way.hehe. |
| 13:08.14 | Katty | i like it when people say hello with chinese food. |
| 13:08.17 | key2 | eastwolf: why don't u do it in #asterisk-china ? |
| 13:08.42 | Katty | key2: chill. |
| 13:08.52 | Katty | key2: a little chinese hello isn't going to ruin your day, is it? |
| 13:09.10 | key2 | Katty: yes it is |
| 13:09.27 | eastwolf | haha. |
| 13:09.28 | Katty | key2: well then maybe /you/ outta do something about it yourself (= |
| 13:10.26 | key2 | Katty: ok just kidding |
| 13:11.04 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 13:11.15 | key2 | mog_home: what's goin on in italy? |
| 13:11.40 | darkskiez | #asterisk-china is very unfriendly |
| 13:11.52 | Katty | seems like #asteris is pretty unfriendly this morning too. |
| 13:12.03 | darkskiez | key2 set a ban on *!*n=darksk*@*.ukonline.co.uk |
| 13:12.13 | iDunno | what? where? |
| 13:12.16 | Katty | iDunno: YOU |
| 13:12.28 | Katty | iDunno: you've been selected for a hug quality assurance test. |
| 13:12.39 | iDunno | ohh - rocking :) |
| 13:13.03 | darkskiez | that hug failed, it needs retested |
| 13:13.04 | Katty | please return this, with cookies, to my address. kthx. |
| 13:13.07 | coppice | Katty: are there severe penalties for failing the test? |
| 13:13.16 | iDunno | heh :) |
| 13:13.17 | Katty | coppice: oh, yes. |
| 13:13.32 | Katty | coppice: you'd become a Test Subect for Hug Improvements. |
| 13:14.30 | Katty | coppice: including a new idea called McHuggles, where you're asked: Would you like a hug with that? |
| 13:14.47 | coppice | Katty: knowing the US, there's probably a uni somewhere that runs degrees in fondling |
| 13:15.05 | Katty | coppice: i'm sure it's an elective somewhere. |
| 13:15.25 | coppice | an erective somewhere? |
| 13:15.31 | Katty | no, that's japan. |
| 13:15.35 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 13:16.09 | coppice | "its really hard to pass this test" :-) |
| 13:16.51 | cuco | file cuco: have you already done any of the recommended tuning? |
| 13:16.55 | cuco | file, what? |
| 13:17.03 | key2 | Katty: why ? where u from? |
| 13:17.23 | Katty | key2: why what? |
| 13:17.33 | file | cuco: tuning gains |
| 13:17.41 | Katty | file: mew. |
| 13:18.22 | Katty | file: waving? |
| 13:18.24 | cuco | file, no, i want to make the echo canceller work, not to fix the echo on MY system. |
| 13:18.25 | Katty | file: what is this?! |
| 13:18.26 | key2 | Katty: saying "knowing the us..." |
| 13:18.34 | file | cuco: it does work. |
| 13:18.36 | cuco | file, want to fix this for all my clients |
| 13:18.42 | key2 | Katty: that's why I ask u where u from |
| 13:18.49 | Katty | key2: i believe /coppice/ typed that. |
| 13:18.50 | key2 | Katty: but I could figureout myself |
| 13:18.52 | cuco | file, it does not work for me |
| 13:18.57 | file | it just won't work in all situations, and some may require tuning gains |
| 13:18.59 | Ahrimanes | man, cant find any working sip or iax clients for windows mobile 5 smartphone.. anyone seen any? |
| 13:19.06 | key2 | Katty: oh ok |
| 13:19.16 | key2 | Katty: my irc client sucks :) |
| 13:19.17 | file | like on FXO ports... |
| 13:19.32 | cuco | wait... fxo ports what? |
| 13:19.37 | coppice | there is a crude iax client for Win CE. There are several SIP ones |
| 13:19.41 | *** join/#asterisk Cresl1n (n=matt@myskin.iet.unipi.it) |
| 13:19.46 | cuco | not me? |
| 13:20.06 | Katty | Cresl1n: well hello to you too! |
| 13:20.11 | Ahrimanes | coppice: problem is win ce apps wont seem to install on windows mobile 5 |
| 13:20.15 | Cresl1n | Katty: !!!!! |
| 13:20.16 | Cresl1n | hey |
| 13:20.20 | Katty | Cresl1n: oh my. |
| 13:20.25 | Katty | Cresl1n: what's got you all excited? |
| 13:20.40 | Katty | Cresl1n: wait! let me guess! |
| 13:20.46 | Katty | Cresl1n: SER is not giving you a headache? |
| 13:20.47 | Cresl1n | I think it's my body's response to jet lag |
| 13:20.49 | Cresl1n | no :-) |
| 13:20.52 | Katty | oh. |
| 13:20.57 | Cresl1n | fortunately it isn't |
| 13:21.06 | Katty | Jet lag? did you go to Dallas? |
| 13:21.12 | Katty | or, come back from Dallas, I should say. |
| 13:21.15 | coppice | windows mobile 5 seems very broken. dell offer a downgrade to wince 2003 for those who bought the upgrade to mobile 5 |
| 13:21.15 | key2 | Katty: big river tel ? |
| 13:21.21 | Cresl1n | Katty: what's in dallas? |
| 13:21.26 | darkskiez | cuco: tried enabling a different echo canceller in zconfig.h? |
| 13:21.28 | Katty | Cresl1n: i believe there was a conference. |
| 13:21.30 | file | cuco: you need to at least try to tweak your gains... and make sure they are set right - if not, the echo canceller won't work or will work poorly |
| 13:21.43 | Katty | Cresl1n: that may not be entirely true though, all i know is that twisted was driving there last weekend. |
| 13:21.58 | Katty | key2: pardon? |
| 13:22.14 | Cresl1n | Katty: oh, no, no dallas for me |
| 13:22.15 | Cresl1n | :-) |
| 13:22.25 | Katty | Cresl1n: i see, i see. |
| 13:22.37 | Ahrimanes | coppice: heh.. well not sure this qtek 8310 will actually run 2003.. havent found any roms anyway.. guess i'll just have to wait |
| 13:22.45 | cuco | darkskiez, tried them all, tried enabling agressive as well |
| 13:23.01 | key2 | katty: bigrivertelephone is where u work or your isp ? |
| 13:23.15 | darkskiez | cuco: tried reducing the gain then? |
| 13:23.24 | darkskiez | cuco: do u know where the echo source is ? |
| 13:23.39 | coppice | i think maybe someone ought to write a proper echo canceller :-) |
| 13:23.41 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 13:23.44 | Katty | key2: it's pretty obvious where i work. |
| 13:23.55 | [TK]D-Fender | Katty : belated "mew" |
| 13:23.58 | Katty | key2: if you can't put one and two together, then i'm not going to bother replying to that question. |
| 13:23.59 | cuco | darkskiez, as i said, i am supposed to make the echo canceller usable, and not fix my private situtation. and no, i am not sure where the echo is comming from. :( |
| 13:24.04 | Katty | [TK]D-Fender: mew. |
| 13:24.11 | Katty | Hmmhesays: mornin |
| 13:24.39 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-37-dsl.end.estpak.ee) |
| 13:24.53 | Hmmhesays | Hello Katty |
| 13:25.27 | kmilitzer | Hi everyone ... I have the following situation: A few gotoifs, that distribute a call following some rules. If nothing matches I send the call into a context with only a i extension |
| 13:25.35 | Cresl1n | cuco: adjust your gains, dangit! |
| 13:25.49 | kmilitzer | Now I want the hangupreason to be 1, so that the resulting SIP-Code is a 404 |
| 13:25.59 | kmilitzer | But it does not work. Any ideas? |
| 13:26.12 | *** join/#asterisk gmaruzz (n=Miranda@myskin.iet.unipi.it) |
| 13:27.41 | darkskiez | kmilitzer: the i extension does not match in that case |
| 13:27.59 | darkskiez | kmilitzer: the i exten only kicks in from a WaitExten call or similar. |
| 13:28.00 | cuco | Cresl1n, i am working on xorcom. what i am trying to do is not fixing our own echo, but making our products work with the echo canceller. this is done for end users. agjusting the gains for me will not help out customers. |
| 13:28.01 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:28.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:28.36 | kmilitzer | darkskiez: So what would be the appropriate way to do it? |
| 13:28.40 | darkskiez | kmilitzer: do a catch all _X. => blah, extension in the last context instead |
| 13:28.56 | Hmmhesays | Katty did you get that software? |
| 13:29.05 | Katty | Hmmhesays: mew? |
| 13:29.09 | Hmmhesays | gp5 |
| 13:29.11 | Katty | Hmmhesays: oh, that music thing? |
| 13:29.14 | Hmmhesays | yeah |
| 13:29.16 | Katty | Hmmhesays: no. |
| 13:29.24 | Katty | Hmmhesays: url? |
| 13:29.33 | kmilitzer | darkskiez: But how do I get the Hangupcause set correctly on a SIP channel? |
| 13:29.34 | Katty | Hmmhesays: pretty please (= |
| 13:29.49 | Hmmhesays | pm'd |
| 13:30.20 | coppice | cuco: you'd better start work on a decent echo canceller, if you want one that just does its job reliably in a hassle free way |
| 13:31.40 | cuco | coppice, so none of the EC on asterisk really work? |
| 13:31.50 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.50.Dial1.SanJose1.Level3.net) |
| 13:31.50 | darkskiez | kmilitzer: in that case, you dont even bother putting a match all line in, as it wont find any exten to connect you with. |
| 13:32.10 | tzanger | cuco: they work, for specific cases of 'work' :-) |
| 13:32.44 | coppice | cuco: they kinda sorta work some of the time. none of them would get through more than one or two of the tests in G.168 |
| 13:32.50 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 13:32.59 | Katty | morning tzanger (= |
| 13:34.07 | tzanger | good morning, Katty |
| 13:34.25 | mjackson | too early |
| 13:34.40 | kmilitzer | darkskiez: It does not work that way. :( My last line if nothing matches for my catch-all distribution is: exten => _[*0-9].,13,Goto(error,BYEXTENSION,1) and error is an empty context |
| 13:35.07 | kmilitzer | But on the SIP side there always is's alway 100 trying and then 603 declined :( |
| 13:36.54 | darkskiez | kmilitzer: oh i see. Um, not sure at this po |
| 13:36.56 | darkskiez | int |
| 13:38.46 | [TK]D-Fender | kmilitzer : Step 13 doesn't look like such a good idea there :) |
| 13:39.11 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 13:39.17 | [TK]D-Fender | kmilitzer : pastebin both contexts pleas |
| 13:39.18 | kmilitzer | [TK]D-Fender: Guess that is the problem ;) |
| 13:39.19 | [TK]D-Fender | ~pb |
| 13:39.22 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 13:39.42 | paryl | is it possible to log only messages/errors to the console related to a particular extension? or specifically, IAX |
| 13:39.54 | tzafrir | How different is MG2 in zaptel trunk from MG2 in 1.2? |
| 13:40.08 | tzafrir | (the echo canceller) |
| 13:40.25 | paryl | i'm having issues where a remote server is forwarding calls to a queue in my local server, and when they pick up the line they hear operator tones |
| 13:40.40 | paryl | but i can't track down why it's happening, or where the call originates |
| 13:41.01 | coppice | if anyone bothered to look closely, they would see that all these ECs are basically the same |
| 13:41.09 | *** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 13:41.24 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:41.32 | kmilitzer | [TK]D-Fender: Here: http://pastebin.com/709357 |
| 13:41.48 | watchy | i need some help with echo |
| 13:42.01 | tzafrir | coppice, are you ommiting a certain word in the end of that sentence? |
| 13:42.04 | watchy | whenever a call comes in or goes out theres echo for like 20 seconds then it goes away |
| 13:42.15 | RoyK | watchy: blame digium |
| 13:42.32 | RoyK | watchy: and say you want your money back |
| 13:42.36 | watchy | royk: anyway to fix it? |
| 13:42.41 | Cresl1n | RoyK: you troll! |
| 13:42.43 | coppice | tzafrir: what did you expect? point of sale? |
| 13:42.45 | RoyK | lol |
| 13:42.54 | RoyK | Cresl1n: i'm being ironic, sir |
| 13:43.04 | Cresl1n | it was very ironic |
| 13:43.08 | RoyK | you get what you pay for, and it's free :) |
| 13:43.10 | watchy | roy: you have any idea what i should look into to fix it |
| 13:43.14 | *** join/#asterisk MMaster23 (n=no@ip229-116-173-82.adsl2.versatel.nl) |
| 13:43.19 | watchy | its strange how the echo actually goes away |
| 13:43.33 | [TK]D-Fender | kmilitzer : Well so far th "13" isn't the problem as you do take 12 other steps leading to it (assume they worked), but you have no exten called "BYEXTENSION" in there and even if you did, it'd be a fixed target so you might as well just keep adding step to your catch-all. |
| 13:43.43 | Cresl1n | watchy what kind of card is it on? |
| 13:43.45 | MMaster23 | hey guys |
| 13:43.48 | coppice | the ECs really are very similar. there are just minor tweaks separating them. they should have been merged into one logn ago. at least they would be a meaningful starting point for improvement. |
| 13:43.53 | watchy | one of the 4port analogs |
| 13:43.59 | MMaster23 | i'm new to asterisk ... having some questions about queues |
| 13:44.04 | watchy | lemme look the model up i cant remember off the top of my head |
| 13:44.07 | Cresl1n | watchy: what version of zaptel? |
| 13:44.13 | watchy | hmm |
| 13:44.13 | RoyK | MMaster23: just ask :) |
| 13:44.22 | watchy | 1.2.4 |
| 13:44.31 | MMaster23 | when i make an agent ... and i log in with X-lite .. the line stays open |
| 13:44.43 | MMaster23 | when i hang up, asterisk shows i'm as an agent offline |
| 13:44.47 | Cresl1n | watchy: what's your echo training option set to? |
| 13:44.55 | watchy | hrm hold |
| 13:45.22 | watchy | echotraining=yes |
| 13:45.46 | Cresl1n | watchy: oh wait |
| 13:45.53 | Cresl1n | I totally forgot the most important thing |
| 13:45.59 | Cresl1n | watchy: have you run fxotune yet? |
| 13:46.08 | watchy | nope. |
| 13:46.18 | watchy | wheres it come with |
| 13:46.18 | Cresl1n | that's the first thing you do with an analogue card |
| 13:46.23 | Cresl1n | watchy: zaptel |
| 13:47.02 | watchy | /dev/zap/1 absent: Device or resource busy |
| 13:47.13 | Cresl1n | watchy: turn off asterisk |
| 13:47.29 | watchy | as soon as i run this it modifys them to sound better? |
| 13:47.34 | kmilitzer | [TK]D-Fender: BYEXTENSION takes the same extension as in the current context to the context you jump to. This is not the problem |
| 13:47.38 | Cresl1n | watchy: did you read the manual? |
| 13:47.51 | watchy | a bit but never seen fxotune |
| 13:48.31 | russellb | a bit ... :( |
| 13:49.54 | kmilitzer | Whatever I do, I only get a 603 declined error ... there seems to be a bug somewhere ... |
| 13:50.20 | Cresl1n | watchy: it tunes the card to your analogue lines |
| 13:50.38 | Cresl1n | watchy: it's like the first thing you do when you have echo problems on analogue cards |
| 13:50.58 | *** join/#asterisk exonic (n=exonic@sig.triton.net) |
| 13:51.00 | watchy | if i use fxotune do i use echotraining=yes? |
| 13:51.08 | Cresl1n | watchy: just don't change it yet |
| 13:51.16 | Cresl1n | watchy: run fxotune first |
| 13:51.20 | *** join/#asterisk bkw__ (n=brian@m543236d0.tmodns.net) |
| 13:51.24 | watchy | hrm i wonder how long it takes |
| 13:51.29 | watchy | cuz the company is using the phones now |
| 13:51.39 | Cresl1n | watchy: it takes a couple minutes per line |
| 13:52.21 | [TK]D-Fender | kmilitzer : Where do you get that idea? Thats not the format of Goto.... |
| 13:52.46 | [TK]D-Fender | kmilitzer : Pastebin CLI output of a failed call. |
| 13:53.00 | mut | mornin tk |
| 13:53.18 | [TK]D-Fender | mut : y0 |
| 13:53.25 | Cresl1n | f !!!!!!! |
| 13:53.36 | Cresl1n | bkw_: !!!!!!! |
| 13:53.41 | watchy | ok sweet. im gonna run it after work |
| 13:53.44 | f | ?!?!?1 |
| 13:53.48 | watchy | i put in like 28 digiums for them yesterday |
| 13:53.50 | kmilitzer | [TK]D-Fender: I don't know where I found it, guess in the wiki somehwere, but it does exactley what it needs to do ... |
| 13:53.51 | watchy | oops |
| 13:53.53 | watchy | i mean polys |
| 13:54.19 | watchy | it seems to be working pretty good other then the echo |
| 13:54.19 | kmilitzer | The Problem is that I cannot set a hangupcause by hand |
| 13:54.41 | watchy | Cresl1n: thanks for youe hlp |
| 13:54.56 | watchy | ill try fxotune tonight and come back and let you know if it worked |
| 13:55.15 | Cresl1n | watchy: np :-) |
| 13:55.32 | *** join/#asterisk j3g (n=rafael@200.130.8.1) |
| 13:55.35 | [TK]D-Fender | kmilitzer : Well SetVar is deprecated as well... |
| 13:55.59 | tamp4x | anyone here have some pbx that supports t38 pass through that i can temporarly register to? |
| 13:56.01 | MMaster23 | guys |
| 13:56.10 | j3g | what would be the best way to interlink (allowing both parties to dial freely between the branches) 2 offices (that currently have only analog phone lines.. one asterisk at each? |
| 13:56.19 | MMaster23 | when i log in as an agent .. the line stays open |
| 13:56.26 | kmilitzer | [TK]D-Fender: Take a look at pbx.c near the end ... |
| 13:56.28 | MMaster23 | when i hang up ... i'm logged out ?? |
| 13:56.32 | MMaster23 | how to fix this ? |
| 13:56.57 | [TK]D-Fender | j3g : VoIP gateway on both sides (FXO or FXO depending on analog channel availability) and 1 * box |
| 13:57.15 | [TK]D-Fender | kmilitzer : What of it? |
| 13:57.33 | [TK]D-Fender | kmilitzer : I'd still like to see a call pas the whole process. |
| 13:58.01 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 13:59.24 | MMaster23 | when i log in as an agent .. the line stays open |
| 13:59.26 | MMaster23 | when i hang up ... i'm logged out ?? |
| 13:59.28 | MMaster23 | how to fix this ? |
| 13:59.51 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:00.40 | C4T3l | hola mundo |
| 14:00.49 | j3g | [TK]D-Fender: what is a 1 * box? |
| 14:00.59 | watchy | anyone here every configure side cars on a 601? |
| 14:01.08 | j3g | oh asterisk box hehe |
| 14:01.14 | j3g | that was stupid of me hehe |
| 14:01.31 | j3g | [TK]D-Fender: but then, not all extensions (analog) would be mapped, right? |
| 14:02.01 | j3g | [TK]D-Fender: people on office1 won't be able to dial extensions on office 2 |
| 14:02.09 | kmilitzer | [TK]D-Fender: it's added to the pastebin |
| 14:05.31 | shiznatix | whats something small but fun i could do with asterisk to kill time? |
| 14:05.50 | mercestes | watchy: yes. |
| 14:06.01 | exonic | shiznatix, write an AGI application |
| 14:06.04 | azeteg | make an analog-digital conterter |
| 14:06.07 | mercestes | shiznatix: play with festival...... |
| 14:06.12 | azeteg | transfer calls over shoestring |
| 14:06.31 | mercestes | shiznatiix: Pick through /var/lib/asterisk/sounds.. "This is not hte extension you are looking for." |
| 14:06.42 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 14:07.44 | shiznatix | hummmm lets see |
| 14:07.45 | [TK]D-Fender | j3g : You're talking abot bridging 2 analog normal PBX's right? Not * PBX's? |
| 14:08.06 | [TK]D-Fender | kmilitzer : If you add to a pastebin, the number changes.... copy it on |
| 14:08.15 | [TK]D-Fender | watchy : I have |
| 14:08.32 | kmilitzer | <PROTECTED> |
| 14:08.44 | C4T3l | shiznatix: write a dialplan that will auto dial someone you dont like repeatedly every time their line is free for 24 hours |
| 14:09.14 | j3g | [TK]D-Fender: yes... 2 analog pbxs... using voip (not necessarily with asterisk) |
| 14:09.15 | watchy | haha |
| 14:09.46 | watchy | anyone here use polycom 501s/601s? |
| 14:10.12 | stoffell | watchy, using 501's |
| 14:10.13 | [TK]D-Fender | kmilitzer : See this error? Your Goto = no good. # |
| 14:10.14 | [TK]D-Fender | <PROTECTED> |
| 14:10.14 | [TK]D-Fender | # |
| 14:10.14 | [TK]D-Fender | May 10 16:01:13 WARNING[21341]: pbx.c:2354 __ast_pbx_run: Channel 'SIP/81.173.194.190-0822d8b0' sent into invalid extension '02419908000' in context 'error', but no invalid handler |
| 14:10.29 | watchy | soff: how are you setting the time? |
| 14:10.46 | [TK]D-Fender | watchy : I use all of their desk models |
| 14:11.20 | [TK]D-Fender | kmilitzer : I don't see the attempt to set HANGUPCAUSE or anything in your call detail |
| 14:11.24 | stoffell | watchy, use ftp provisioning and create good configs (time: NTP) |
| 14:11.58 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:12.11 | watchy | can you set the time offset in the .cfg? |
| 14:12.20 | asterboy | watchy, the easiest method is to use http:/ipOFphone |
| 14:12.43 | mercestes | yes....the offset is in seconds. |
| 14:12.57 | [TK]D-Fender | asterboy : Don't go spreading around bad habits for conguring Polycom's now :) |
| 14:13.09 | mercestes | -6 is not -6 hours....-21600 is -6 hours. |
| 14:13.21 | [TK]D-Fender | BLARG. I am completely losing my ability to type coherently. |
| 14:13.25 | asterboy | well yes, it's *not* the best method |
| 14:13.29 | jpabuyer | asterboy, what's the good way? |
| 14:13.54 | asterboy | I use ntps1-1.uni-erlangen.de |
| 14:14.05 | watchy | <PROTECTED> |
| 14:14.06 | asterboy | use a good ntp server |
| 14:14.06 | watchy | that it? |
| 14:14.13 | mercestes | ntp.pool.org time.apple.com |
| 14:14.20 | [TK]D-Fender | asterboy : Far safer = pool.ntp.org rather than a specifi peer which may go down. |
| 14:14.47 | mercestes | TK is right, it's pool.ntp.org...dang it..stupid lysdexia. |
| 14:14.54 | asterboy | Not sure why, but the Germans have the best time keeping I've seen so far. |
| 14:15.04 | [TK]D-Fender | mercestes :iedned! |
| 14:15.05 | asterboy | always works and is fast |
| 14:15.10 | watchy | i think my offset is like -26100 or something |
| 14:15.26 | jpabuyer | aaahh you were talking about time :) of course.. ntpd |
| 14:15.32 | watchy | 21600 then |
| 14:16.01 | brif8 | Is there any difference between (1) Stop now and then asterisk -vvvvgc AND (2) CLI> restart taking into account memory leaks etc.. ? |
| 14:17.33 | asterboy | tcpIpApp.sntp.address="" |
| 14:18.23 | kmilitzer | [TK]D-Fender: It's the same if I have a i extension ... I'll pastebin it ... takes a second |
| 14:18.43 | [TK]D-Fender | kmilitzer : you DON'T have one! They're all commented out! |
| 14:18.53 | [TK]D-Fender | kmilitzer : The error speaks for itself |
| 14:19.29 | kmilitzer | [TK]D-Fender: I am not dumb ... I know what I am doing, I commented it out in between and it made no difference! |
| 14:19.37 | kmilitzer | That's what I will pastebin now ... |
| 14:20.27 | [TK]D-Fender | kmilitzer : well your Set's aren't even being called, so there is nothing to "fail there. |
| 14:22.45 | kmilitzer | [TK]D-Fender: Look: http://pastebin.com/709416 |
| 14:23.03 | ManxPower | I assume everyone has seen this: http://news.zdnet.co.uk/communications/3ggprs/0,39020339,39267682,00.htm |
| 14:24.54 | mercestes | ok.... |
| 14:24.57 | [TK]D-Fender | kmilitzer : well your noop isn't getting called because you manually kill the call with HANGUP in there... so you don't get to see if it worked or not |
| 14:24.59 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:25.12 | MMaster23 | when i log in as an agent .. the line stays open |
| 14:25.14 | MMaster23 | when i hang up ... i'm logged out ?? |
| 14:25.15 | MMaster23 | how to fix this ? |
| 14:25.38 | mercestes | Yea, it's still calling every other line in invalid. |
| 14:26.30 | mercestes | because....your goto goes to an invalid context/extension. |
| 14:26.38 | mercestes | hence...the designator...invalid.... |
| 14:28.06 | kmilitzer | [TK]D-Fender: It will not work, because HANGUPCAUSE IMHO cannot be set, only read ... |
| 14:28.23 | mut | oh tk |
| 14:28.35 | mut | nenah didn't know why hdlc wasn't working either |
| 14:28.58 | mut | hardware hdlc8 |
| 14:29.52 | mercestes | So, this guy calls tech support...his power supply is smoking.. Tech support says buy a new power supply. |
| 14:30.11 | mercestes | Guy argues with tech support for 6 hours, because he thinks is a setting in his config.sys. |
| 14:30.39 | mercestes | finally tech support gives up and tells him to type "load nosmoke.com." Of course, it doesn't work... |
| 14:30.54 | mercestes | Tech support sends him to microsoft after a newer version of DOS that comes with "nosmoke.com." |
| 14:31.21 | mercestes | 3 days later...the guy comes back wanting a new power supply. MIcrosoft told him his power supply is incompatible with nosmoke. |
| 14:31.46 | *** join/#asterisk Cresl1n (n=matt@myskin.iet.unipi.it) |
| 14:32.08 | mjackson | mercestes: does nosmoke run under wine? it sounds cool |
| 14:32.12 | j3g | [TK]D-Fender: how would you link 2 analog pbxs? |
| 14:32.31 | ManxPower | j3g, Usually with a wire. |
| 14:32.54 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 14:34.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:34.05 | brif8 | [TK]D-Fender: Is there any difference between (1) Stop now and then asterisk -vvvvgc AND (2) CLI> restart taking into account memory leaks etc.. ? that you are aware of ? |
| 14:35.25 | *** join/#asterisk Essobi (i=kstone@216.26.137.75) |
| 14:35.59 | Essobi | Anyone from the Louisville area looking for work? |
| 14:36.25 | Cresl1n | Essobi: what are you talking about? |
| 14:36.40 | r_evolution | he wants someone who is looking to start in softcore porn :) |
| 14:36.45 | r_evolution | star* |
| 14:37.39 | Essobi | lol |
| 14:37.53 | Essobi | Need 3 programmers from the Louisville Kentucky area. |
| 14:37.56 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:38.17 | azeteg | what do you want programmed? |
| 14:38.46 | mercestes | In what languages? |
| 14:38.55 | mercestes | and haven't you ever heard of "telecommuting?" |
| 14:39.01 | Cresl1n | espanol, russki, i polski |
| 14:39.01 | Essobi | I'm not looking for contact hires. |
| 14:39.02 | Cresl1n | :-) |
| 14:39.11 | Essobi | 3 full-timers. |
| 14:39.16 | azeteg | I refuse to do anything with porn,web,carrots or peanuts |
| 14:39.22 | russellb | I usually charge $100,000 ... but for you, I'll do it for $90,000 |
| 14:39.30 | azeteg | per hour? |
| 14:39.32 | azeteg | thats cheap |
| 14:39.56 | azeteg | tell what you want to do, Essobi |
| 14:40.06 | r_evolution | but you're down for watermelons azeteg? |
| 14:40.13 | r_evolution | scary. |
| 14:40.19 | Cresl1n | I have a $20,000 minimum on a job |
| 14:40.20 | Cresl1n | :-D |
| 14:40.39 | mercestes | so why does telecommuting != fulltime? |
| 14:40.43 | GerbilWrk | and how often do you get jobs? |
| 14:40.50 | r_evolution | so i guess he's looking for full-time people... Cresl1n = 20K a year? |
| 14:40.52 | mercestes | If you want locals, man, try a greensheet dude..not the internet. |
| 14:40.55 | j3g | ManxPower: a 200km wire? hehe the offices are not on the same location |
| 14:41.08 | azeteg | I usually charge around $8-10k per month |
| 14:41.11 | azeteg | for c++ work |
| 14:41.11 | Cresl1n | GerbilWrk: enough to eat :-) |
| 14:41.18 | PiPiPi | azeteg, nice price |
| 14:41.21 | PiPiPi | thats fair |
| 14:41.35 | PiPiPi | i get paid $2k/month for everything |
| 14:41.37 | PiPiPi | cause im a slave |
| 14:41.42 | azeteg | oh |
| 14:41.42 | watchy | i like puppies |
| 14:41.50 | Katty | puppies? |
| 14:41.52 | r_evolution | Sweet.. slave. |
| 14:41.52 | azeteg | well, I have to pay taxes etc on that later |
| 14:41.57 | watchy | yea puppies are awesome |
| 14:42.01 | azeteg | which cuts like 50% off |
| 14:42.08 | mercestes | I like slaves. |
| 14:42.08 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 14:42.15 | PiPiPi | then hire me |
| 14:42.17 | azeteg | and I have an office rent to pay... and so forth |
| 14:42.18 | Cresl1n | oej: !!!!! |
| 14:42.20 | PiPiPi | you slave peddlers! |
| 14:42.28 | oej | Back! |
| 14:42.28 | r_evolution | yeah... slaves are definitely fun. depending on which use you're using which slave for |
| 14:42.35 | PiPiPi | azeteg, why do you have an office if you are an employee? |
| 14:42.40 | azeteg | I am not |
| 14:42.44 | azeteg | I run a company |
| 14:42.48 | PiPiPi | oh i see |
| 14:43.01 | r_evolution | azeteg is a free-lance rectal-cleaner :) |
| 14:43.02 | PiPiPi | consultant company? |
| 14:43.05 | azeteg | not a big office, about 60m^2 |
| 14:43.15 | azeteg | I do coding work mosltly |
| 14:43.19 | watchy | can i move in azeteg |
| 14:43.20 | PiPiPi | nice |
| 14:43.23 | azeteg | Qt applications |
| 14:43.34 | azeteg | osx/linux/win32 |
| 14:43.38 | mercestes | I write VB.Net (stop laughing at me) |
| 14:43.47 | Essobi | lol |
| 14:43.49 | Essobi | No. |
| 14:43.54 | azeteg | VB.Net is nice .. eh |
| 14:44.00 | PiPiPi | how much do network administrators / systems administrators usually make now-adays? |
| 14:44.04 | r_evolution | holy crap... i thought essobi was gone. |
| 14:44.05 | azeteg | depends on the country |
| 14:44.07 | mercestes | my six year old son writes in vb.net. |
| 14:44.15 | PiPiPi | United Stats, California |
| 14:44.16 | r_evolution | hey mercestes... at least you're not writing in straight up vb :) |
| 14:44.31 | asterboy | [TK]D-Fender, figured out the solution to Agent-Manager use |
| 14:44.37 | [TK]D-Fender | brif8 : no clue |
| 14:44.38 | MicioMacio | ufff..... |
| 14:44.45 | [TK]D-Fender | j3g : I already told you how. |
| 14:44.48 | watchy | i guess im gonna get dressed and goto work |
| 14:44.53 | MicioMacio | why my queue end at 1 minute..... :( ? |
| 14:44.53 | watchy | im tired of sitting here naked |
| 14:44.57 | Essobi | r_evolution Nah, I just hang out in ... channels with a lower noise to signal ratio. :) |
| 14:45.02 | asterboy | 2 H51 Headsets, 1 M12 Amp, and 1 Y adaptor |
| 14:45.25 | [TK]D-Fender | kmilitzer : You may beleive that but you prevented your test from PROVING it, so its a sign of poor (or more like blind) faith |
| 14:45.29 | r_evolution | i'm tired of hearing that you're sitting there naked watchy... |
| 14:45.39 | Essobi | lol.. |
| 14:45.54 | [TK]D-Fender | asterboy : sounds like a chead headset model. |
| 14:45.57 | r_evolution | and essobi... this channel is a *LOT* of noise ;x |
| 14:45.57 | watchy | r_evolution: it is very tiring |
| 14:46.04 | asterboy | chead? |
| 14:46.04 | r_evolution | yes... yes it is. |
| 14:46.11 | Essobi | Actually.. A few months ago I remember someone from Louisville used to hang out here.. |
| 14:46.15 | [TK]D-Fender | asterboy : cheap |
| 14:46.24 | r_evolution | and someone has their f*cking phone turned up so load that it makes me jump every time it rings |
| 14:46.30 | Essobi | And EVERYONE needs a job working with *. :) |
| 14:46.32 | asterboy | well, it's for a crisis center and many people will be using it. |
| 14:46.36 | Essobi | r_evolution HAHAHA. |
| 14:46.37 | asterboy | so no ear plugs |
| 14:46.54 | Hmmhesays | finally got my vonage account working again |
| 14:46.55 | asterboy | has a nice voice tube |
| 14:47.13 | [TK]D-Fender | asterboy : Voice tube is loud but too omni-directional. |
| 14:47.19 | asterboy | which can be interchanged with rainbow colors! (appeals to pot heads like me) |
| 14:47.23 | [TK]D-Fender | asterboy : Avoid. Get the EC one |
| 14:47.27 | hwt | oej: you there? |
| 14:47.42 | [TK]D-Fender | asterboy : My guys used to use those... |
| 14:47.42 | asterboy | more EC? |
| 14:47.59 | [TK]D-Fender | asterboy : its so powerful it even picks up the EAR-PIECE, |
| 14:48.04 | asterboy | * EC, Polycom EC and Headset EC? |
| 14:48.08 | *** join/#asterisk adker (n=adker@67-136-210-63.dsl1.glv.ny.frontiernet.net) |
| 14:48.09 | [TK]D-Fender | asterboy : change your order if possible |
| 14:48.16 | [TK]D-Fender | asterboy Headset. |
| 14:48.18 | asterboy | havn't made it yet |
| 14:48.33 | oej | well, kind of |
| 14:48.37 | j3g | [TK]D-Fender: i know... but is there a way to have both offices map all extensions? using 2 voip gateways + an asterisk box won't allow 1-to-1 mapping on the analog extensions |
| 14:48.40 | [TK]D-Fender | asterboy : Good.... seriously, voice-tube = not so great an idea. |
| 14:49.02 | asterboy | H51N then? |
| 14:49.18 | oink | H5N1 ? |
| 14:49.22 | oink | :o |
| 14:49.28 | asterboy | http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880043/prod440076 |
| 14:50.05 | [TK]D-Fender | j3j : yes. with a VoIP gateway on both sides, you'd dial an access # to call the analog extension on YOUR PBX. that would ring the gatweay on yourside which would pick up and give you a 2nd dial-tone. from there you dial the exten on the other side. * then takes that call and pumps it out the gateway on the OTHER PBX |
| 14:50.24 | watchy | ok if i dont leave now i never will |
| 14:50.30 | watchy | you guys are to addicting |
| 14:50.39 | [TK]D-Fender | j3j : So you'd need 2 x SPA-3000 gateways (1FXO/1FXS) and an * server. Total cost < 200$USD |
| 14:50.47 | *** join/#asterisk gmaruz1 (n=Miranda@myskin.iet.unipi.it) |
| 14:50.49 | hwt | oej: the usb-conference-device you brought to meetasterisk in oslo last week. what was that? |
| 14:50.51 | j3g | Oh I see |
| 14:50.58 | hwt | oej: the name of it. |
| 14:51.15 | kmilitzer | [TK]D-Fender: I know it as a fact. I had problems with a similar thing a while ago ... |
| 14:51.28 | oej | xorcom astribank |
| 14:51.57 | kmilitzer | [TK]D-Fender: If you cannot help and/or don't understand what the problem is, than just say so ... |
| 14:52.16 | mercestes | DUDE>....what is your problem? |
| 14:52.18 | mercestes | your wrong... |
| 14:52.21 | [TK]D-Fender | asterboy : Better |
| 14:53.04 | [TK]D-Fender | kmilitzer : Well your test never made it to the point of proving that you can't set it because you never NoOp it. Doesn't offer much to see now does it? |
| 14:53.59 | mercestes | kmilitzer: Your powersupply isn't compatible with nosmoke...sorry. |
| 14:54.05 | PiPiPi | anyone know if its possible to get FreePBX (amp) running on FreeBSD ? |
| 14:54.20 | jpabuyer | read the topic |
| 14:54.35 | mercestes | The channel #freepbx would be glad to answer that for you. |
| 14:55.44 | kmilitzer | mercestes: I guess that is the problem ... |
| 14:56.38 | I-MOD | ~centos |
| 14:56.41 | jbot | hmm... centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 14:56.47 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:56.50 | I-MOD | ~centosbug |
| 14:56.51 | jbot | rumour has it, centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 14:56.53 | azeteg | anyone here setup an asterisk with h323 channel to gatekeeper? |
| 14:57.07 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 14:57.12 | PiPiPi | well #freepbx is quiet :-( |
| 14:57.50 | mut | can i set chan vars in a context? |
| 14:57.54 | mut | like |
| 14:58.02 | mut | [level1] |
| 14:58.07 | jpabuyer | ~centosbug |
| 14:58.08 | jbot | from memory, centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 14:58.12 | mut | defaultroute=sip/1 |
| 14:58.14 | hwt | oej: no, i was thinking of the usb conferencing-device. mic/speaker. |
| 14:58.24 | hwt | oej: attached with a retractable usb-chord. |
| 14:58.27 | mut | exten => 1234,1,Dial($defaultroute}) |
| 14:58.31 | hwt | anyone else? |
| 14:58.32 | asterboy | great fun, call Platronics and ask the sales agent what kind of headset they are using, then tell them the sound horrible. :P |
| 14:58.41 | oej | hwt: Well, you can't get it unless you travel to Japan |
| 14:58.55 | r_evolution | you know... |
| 14:58.58 | oej | Buffalo something |
| 14:59.06 | r_evolution | this is probably an asshole thing to do... |
| 14:59.06 | hwt | oej: oh, well. that's a shame. :/ |
| 14:59.16 | hwt | oej: maybe there are similar devices out there. |
| 14:59.29 | r_evolution | but im thinking about making the dial-plan AEL style... |
| 14:59.52 | PiPiPi | ~freebsd |
| 14:59.53 | jbot | A stable secure open source operating system.. URL: http://www.freebsd.org/ FreeBSD: Nothing runs like a daemon with a pitch fork. |
| 15:00.03 | PiPiPi | ~debian |
| 15:00.09 | PiPiPi | ~asterisk |
| 15:00.11 | jbot | asterisk is the best free PBX in the world. |
| 15:00.26 | Cresl1n | jbot: roxors! |
| 15:00.28 | jbot | extra, extra, read all about it, roxors is something that doesn't suxors |
| 15:00.36 | azeteg | I need to setup asterisk to forward calls to an external h323 gatekeeper |
| 15:00.42 | azeteg | anyone care to help me? |
| 15:02.09 | Hmmhesays | not really |
| 15:02.24 | *** join/#asterisk Navire (n=navire@200.217.77.227) |
| 15:02.25 | Hmmhesays | compile ooh323 and you should be fine |
| 15:02.32 | azeteg | I did taht |
| 15:02.40 | Navire | Hi |
| 15:02.41 | azeteg | its the registration process I'm not sure about |
| 15:02.43 | Hmmhesays | ok then what is your problem |
| 15:03.07 | Hmmhesays | you said "send calls" not register |
| 15:03.23 | ManxPower | I didn't know H323 supported registration |
| 15:03.41 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:04.13 | azeteg | well - say like this then - I have no idea how to get asterisk to talk to that h323 gatekeeper |
| 15:04.41 | azeteg | what do I put in extensions.conf? |
| 15:05.12 | Navire | Is possible iax2 p2p? Asterisk setup call for cliente and audio stream client to client. |
| 15:05.28 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 15:06.14 | azeteg | I put in my context something like : exten => 00,xxxxxxxxxxx |
| 15:07.01 | Navire | Audio stream dont through Asterisk. |
| 15:07.15 | *** join/#asterisk alphaque (n=alphaque@myskin.iet.unipi.it) |
| 15:07.43 | azeteg | http://www.voip-info.org/wiki/view/Asterisk+talking+to+Ericsson+PBX+using+H323 could help me? |
| 15:08.30 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 15:11.37 | *** join/#asterisk __flag__ (n=__flag__@59.163.66.98) |
| 15:12.11 | mut | is there a way to set a variable based on the entering context |
| 15:12.13 | mut | other than |
| 15:12.22 | *** join/#asterisk marlowe (n=mmarlowe@pdpc/supporter/active/marlowe) |
| 15:12.24 | mut | exten => _.,1,set(var=value) |
| 15:13.49 | __flag__ | I am facing a problem "488 Not acceptable here"...can anybody help me |
| 15:13.53 | __flag__ | I face this codec issue only when a call is received from pstn.... |
| 15:13.56 | __flag__ | i have disallow=all, allow=ulaw in config... |
| 15:14.02 | *** part/#asterisk Bentley (n=Bentley@S010600301baf55dd.cg.shawcable.net) |
| 15:14.21 | __flag__ | can anyone know what may be going wrong? |
| 15:14.37 | docelmo | do a SIP trace |
| 15:14.42 | __flag__ | i did.... |
| 15:14.48 | docelmo | ~pb |
| 15:14.54 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:14.58 | docelmo | send it there |
| 15:15.20 | __flag__ | it says us=0x0, peer=<codecs>....effective=0x0 |
| 15:15.23 | __flag__ | ok |
| 15:16.44 | __flag__ | http://pastebin.com/709507 |
| 15:17.27 | __flag__ | any comments, anyone..... |
| 15:17.36 | *** join/#asterisk littlejohn (n=little@host32-82.pool877.interbusiness.it) |
| 15:17.46 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 15:18.23 | docelmo | ya.. |
| 15:18.31 | docelmo | you need to force a codec on both sides.. |
| 15:18.51 | docelmo | thats your problem.. One side is presenting everything it has and asterisk is like BS I dont like them |
| 15:19.02 | docelmo | check your order make g711u priority |
| 15:19.04 | docelmo | or g729 |
| 15:19.16 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 15:19.20 | __flag__ | one party is pstn which might be forcing codec |
| 15:19.28 | __flag__ | other party does not seem to like it |
| 15:19.45 | docelmo | when you say PSTN.. how are you connecting to the PSTN? |
| 15:19.52 | Navire | Asterisk is client/server? Dont possible work p2p with iax2. |
| 15:20.05 | docelmo | Asterisk is neither |
| 15:20.12 | docelmo | Its a PBX |
| 15:20.14 | Navire | Client/server using iax |
| 15:20.22 | docelmo | Well its more Server than anything |
| 15:20.23 | __flag__ | one asterisk (client) registers with other (main) asterisk... |
| 15:20.41 | Navire | With SIP I work clients p2p. |
| 15:20.58 | __flag__ | client receives a call from pstn line connected to main asterisk.... |
| 15:21.16 | __flag__ | and client does not like the codec whatever is the forced by pstn.... |
| 15:21.51 | __flag__ | so.... |
| 15:22.00 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 15:22.38 | mercestes | pstn =ringing=> main.asterisk =ringing=> client.asterisk? |
| 15:22.56 | __flag__ | yes... |
| 15:23.29 | __flag__ | main.asterisk => client.asterisk is sip communication..... |
| 15:23.30 | kaz0358 | if i was looking at having a machine with 4fxo and 4 fxs (1 TDM04B and 1 TDM40B), would you recommend as the minimum hardware? and assuming that it might have up to 4 concurrent SIP calls |
| 15:23.31 | mercestes | main.asterisk should be able to do a codec conversino between main.asterisk and client.asterisk. |
| 15:23.45 | mercestes | are you iaxing across? |
| 15:23.46 | kaz0358 | s/would/what/ |
| 15:23.53 | __flag__ | yes, ideally.... |
| 15:23.58 | __flag__ | no, i am not using iax |
| 15:24.05 | __flag__ | it's simple sip.... |
| 15:24.16 | mercestes | then your codec coversion would be pstn => iax. |
| 15:24.43 | mercestes | simple sip in which codec? |
| 15:24.56 | *** join/#asterisk holaaa (n=holaa@85.137.83.71) |
| 15:25.05 | __flag__ | i do not know... i am not forcing any codec... |
| 15:25.34 | mercestes | then force g711u on main and client. |
| 15:25.40 | holaaa | I place rtptimeout=20 rtpholdtimeout=40 in zapata.conf, between "group" and "channel" but my TDM does hang up always 1 minutos and 2 seconds after a phone call ends with no response... any ideas why? |
| 15:25.41 | mercestes | it might be picking one it doesn't like. |
| 15:25.52 | __flag__ | ok.... |
| 15:26.28 | __flag__ | i thought main asterisk was forcing a codec as call comes from pstn... is that g711u...? |
| 15:26.40 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:26.41 | MMaster23 | when i log in as an agent .. the line stays open |
| 15:26.42 | Assid | err.. whose a good carrier for toll free numbers? |
| 15:26.43 | MMaster23 | when i hang up ... i'm logged out ?? |
| 15:26.44 | MMaster23 | how to fix this ? |
| 15:26.54 | Assid | hey tkd : iyou there |
| 15:27.25 | kaz0358 | assid, what kind of rate are you hoping to get? |
| 15:27.27 | wunderkin | MMaster23, use agentcallbacklogin |
| 15:27.31 | Assid | 3c/min ? |
| 15:27.33 | Assid | or better |
| 15:27.38 | Assid | with decent quality |
| 15:27.52 | kaz0358 | assid, do you need to support fax/data? |
| 15:28.01 | Assid | nah |
| 15:28.20 | kaz0358 | assid, what kind of call volume are you expecting? |
| 15:28.24 | *** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net) |
| 15:28.28 | Assid | dunno |
| 15:28.38 | kaz0358 | assid, an order of magnitude? :) |
| 15:28.44 | asterboy | Anyone use the Plantronics CS50 wireless headset? Testimonials? |
| 15:28.51 | MMaster23 | wunderkin ... tnx! |
| 15:29.15 | kaz0358 | assid, how many different toll free numbers do you need? |
| 15:29.35 | Assid | gotta port a number |
| 15:29.44 | ohad | hi, i am looking for an elegent way to push caller data from my asterisk box to the browser of my users. any suggestion for api's / other classes that already exist that can handle the push of data from the server to the user's webclient ? |
| 15:30.18 | exonic | ohad, "push of data" .. is that just a database? |
| 15:30.27 | kaz0358 | assid, ahh just a single number.. well i don't know too many. teliax only charges 2.9/min and i had good luck with them when i was personally using them. there might be better rates. i don't know |
| 15:30.51 | Assid | i wouldnt mind 3c |
| 15:31.00 | Assid | but.. i need quality |
| 15:31.04 | Assid | and simultanous calls |
| 15:31.07 | kaz0358 | assid, https://www.teliax.com/newaccount/ ... |
| 15:31.32 | Assid | no no,.. not teliax |
| 15:31.35 | kaz0358 | assid, with their pay as you go plan i don't think there is a limit. and i never had any problems with quality with them |
| 15:31.38 | Assid | they kinda going nutty |
| 15:31.55 | kaz0358 | assid, really? that's a shame. i haven't used them for about 6 months or so |
| 15:32.22 | *** join/#asterisk alerios (n=alerios@201.244.243.43) |
| 15:32.26 | mut | is there a way to set a variable based on the entering context? |
| 15:33.24 | ohad | exonic, i have a custom built crm system and i want to add a module to it where the caller's data will be pushed to the crm client's browser - instead of pulling that data from the asterisk server |
| 15:34.44 | ohad | exonic, i am not sure if i'm making any sense |
| 15:35.37 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:36.24 | Assid | err.. anyone here using junction networks |
| 15:37.09 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:37.14 | *** join/#asterisk AsteriskAddict (n=speedy@r172h230.dixie-net.com) |
| 15:37.22 | *** join/#asterisk p1p (i=pip@64.200.16.25) |
| 15:37.33 | mjackson | can somebody point me to an example on how to register an asterisk to asterisk IAX2 channel? The example i've found in the wiki doesn't work. I get 'Registration Refused' on both ends. |
| 15:37.54 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:37.57 | holaaa | I place rtptimeout=20 rtpholdtimeout=40 in zapata.conf, between "group" and "channel" but my TDM does hang up always 1 minutos and 2 seconds after a phone call ends with no response... any ideas why? |
| 15:38.10 | holaaa | In other words. When I make a phone call from pstn to a sip phone, and no one picks it up (just an Answer from my dialplan but no voicemail or anything) when bouth hangs up because the time is over, asterisk realizes there is no activity and hangs up 1 minutes 2 sencods after... is there any way to change this time? |
| 15:38.21 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 15:38.40 | Assid | kaz0358: know any other providers? |
| 15:39.12 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:39.17 | *** join/#asterisk ivanfm (n=ivanfm@201-27-67-81.dsl.telesp.net.br) |
| 15:40.10 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 15:40.43 | exonic | ohad, aye, well I guess it depends on what you want to relay to your client. Using the asterisk manager events interface, you can easly pipe that information anywhere you want from the asterisk server. |
| 15:41.36 | exonic | ohad, it makes it easy to react in realtime |
| 15:43.13 | alerios | has anybody used cdr_custom successfully here? I'm getting lots of 'Function CDR not registered' messages on cli |
| 15:43.56 | mjackson | You talking about the UserfieldAppend() stuff? |
| 15:44.01 | file | means func_cdr is not loaded |
| 15:44.14 | mut | http://pastebin.ca/54777 |
| 15:44.19 | mut | ^^ my question is that |
| 15:44.32 | p1p | Can anyone give me a hand or link me to docs for setting up a paging extension that pages out the soundcard on the * server? |
| 15:45.04 | jsharp | exten => 1234,1,Dial(OSS) |
| 15:45.14 | jsharp | That makes extension 1234 call the sound card. |
| 15:45.17 | alerios | mjackson, no, I mean the custon cdr backend, I want to fill a new custom cdr field |
| 15:45.33 | alerios | file, let me check... |
| 15:46.30 | mjackson | alerios: I haven't tried that... I use the custom userfield and define delimited keypairs in a sigle field. |
| 15:48.07 | alerios | file, that's it! thank you. |
| 15:48.14 | file | aye aye |
| 15:48.30 | alerios | file, seems like the asterisk debian package doesn't have it compiled by default |
| 15:50.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 15:50.33 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:50.46 | Assid | okay .. anyone know any good carriers for toll free? |
| 15:51.15 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 15:51.16 | noky | hi |
| 15:51.46 | ohad | <PROTECTED> |
| 15:58.10 | vader-- | anyone see the article where Microsoft Exchange 12 will have SIP and voice support and stuff? |
| 16:00.42 | exonic | ohad, definitely possible, I react on realtime events for various reasons |
| 16:01.01 | exonic | ohad, the manager interface can always stay connected |
| 16:01.17 | ohad | exonic, do i just change the length of the session handler on the asterik event manager? |
| 16:01.18 | *** join/#asterisk Cresl1n (n=matt@myskin.iet.unipi.it) |
| 16:02.00 | Assid | okay i need a good carrier for toll free |
| 16:02.03 | Assid | anyone know any |
| 16:02.19 | exonic | ohad, i'm not sure I follow.. There is no "session length" |
| 16:02.20 | ohad | exonic, how do you then push it to a browser? is it possible to change that feed to run on http headers? |
| 16:02.26 | tzafrir | ohad, how about an IM client? or a client to the FOP (flash operator panel) server? |
| 16:03.12 | tzafrir | ohad, you need some sort of proxy on the asterisk side and a client. |
| 16:03.38 | ohad | tzafrir, i'm not into flash. and an IM client makes it more complicated. i want this solution to be intergrated into my browser/my crm app |
| 16:03.57 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 16:04.55 | tzafrir | ohad, how do you intend to push the data (technically)? |
| 16:04.56 | ohad | tzafrir, i would not want to deploy IMs or any addtional app to run on the client web browser. I need this solution to work with a browser alone |
| 16:05.24 | ohad | tzafrir, (hi) xml data feeds if i can trasnlate them to run on http headers |
| 16:06.00 | tzafrir | but this means polling. Think of some 100 clients polling periodically |
| 16:06.08 | ohad | tzafrir, kindda like multipart xml data feeds |
| 16:06.41 | ohad | tzafrir, exactly. i am interested in pushing rather then pulling. i don't want to have 100 users hitting the server every second |
| 16:06.43 | tzafrir | And it has to be every second or so to look "real-time" |
| 16:06.53 | ohad | tzafrir, yep |
| 16:07.03 | tzafrir | So basically you have to have a connection open |
| 16:07.13 | ohad | tzafrir, exactly! |
| 16:07.51 | ohad | tzafrir, but it seems that i can only use java applet or some flash module to have that work |
| 16:08.31 | ohad | i am not interested in using java-applets (they are so 90s), and as for flash, well i rather skip that technology as well |
| 16:08.39 | ohad | (sorry to be a bugger) |
| 16:09.19 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
| 16:09.59 | *** join/#asterisk tld (n=tld@343006.ds.nac.net) |
| 16:10.23 | sangee | how do i make extension busy if that extension are already inuse? |
| 16:10.33 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-188.z143-154-67.customer.algx.net) |
| 16:10.34 | tld | Does anyone know if when a call is hung up, the AGI script gets killed, or will it run until completion? |
| 16:10.43 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:11.15 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
| 16:12.09 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:13.23 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:13.39 | exonic | sangee, what kind of Channel ? |
| 16:13.47 | sangee | Sip channel |
| 16:14.09 | exonic | sangee, just disable call waiting |
| 16:14.20 | sangee | how do i do that? |
| 16:14.51 | exonic | callwaiting=no |
| 16:15.26 | sangee | can i do it on run time (using AGI script)? |
| 16:15.55 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 16:17.20 | exonic | sangee, i'm mistaken. I'm not aware of the parameter to do it. It's up to the SIP client to report busy/available. |
| 16:17.38 | exonic | sangee, I was looking at a custom db parameter by mistake |
| 16:18.37 | sangee | i think sip does not detect busy? am i right? |
| 16:19.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:19.58 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-37-dsl.end.estpak.ee) |
| 16:21.08 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 16:21.47 | stack_ | I see there is a 'w' & 'W' option for the dial command... is there one for Queues as well? |
| 16:22.32 | p1p | Anyone know if polycom has released spip1.6.6 yet? |
| 16:29.04 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 16:29.07 | PakiPenguin | hello |
| 16:29.40 | stack_ | quiet in here today |
| 16:30.00 | PakiPenguin | yeah |
| 16:30.32 | stack_ | Looks like everyone is actually working and not hanging out in IRC :) |
| 16:31.54 | [TK]D-Fender | p1p : Yes |
| 16:32.23 | *** join/#asterisk Lino` (n=Lino@i577BF0D0.versanet.de) |
| 16:32.36 | *** join/#asterisk elvisthedj (n=elvisthe@th20.montanavision.com) |
| 16:33.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:35.37 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 16:35.53 | *** join/#asterisk ckwall (n=ckwall@63.149.122.94) |
| 16:36.54 | ckwall | I am using dtmfmode=rfc2833. But I called a phone system ivr yesterday and it would not recognise my key presses. I had to call back on my cell. It worked fine from my cell. |
| 16:37.07 | ckwall | is there a more standard dtmfmode that I should be using? |
| 16:37.27 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 16:37.39 | snitt | depends on your itsp |
| 16:37.58 | ckwall | where should i go from here to find out? |
| 16:38.08 | ckwall | I guess I am not sure what questions i need to ask |
| 16:39.40 | Faithful | ckwall you could try inbanc |
| 16:39.44 | Faithful | ckwall you could try inband |
| 16:41.09 | [TK]D-Fender | ckwall : there are multiple points of DTMFMODE. There is between your phone and *, then between * and the PSTN. Can you give some details as to everything in the path between the caller and to place you are calling... |
| 16:42.08 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 16:46.22 | *** join/#asterisk extremis (i=extremis@unon.net) |
| 16:46.31 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 16:46.48 | extremis | How do I speed up the boot process on a cisoc 7960? Recent versions of the firmware take up to 5 minutes to "configuring VLAN" |
| 16:47.08 | Zodiacal | anyone know how to make the Playback app play wav files? i read somewhere that i have to tell it which codecs to use? can someone point me in the right direction? |
| 16:47.43 | ManxPower | Zodiacal, it will play .wav files just fine by default. |
| 16:47.45 | luke-jr_ | Can anyone get ahold of SellVoIP? |
| 16:47.56 | Zodiacal | manxpower mine doesn't seem too :/ |
| 16:48.03 | ManxPower | what is this "ask about how to do things that happen by default week"? |
| 16:48.14 | ManxPower | Zodiacal, Playback(/path/to/audio/blah) |
| 16:48.25 | ManxPower | YOU NEVER PUT AN EXTENSION ON THE FILE NAME IN PLAYBACK |
| 16:48.58 | Zodiacal | im acctualy trying to play it from a .call file.. it works with gsm, just not my .wav file |
| 16:49.05 | Zodiacal | are there specific specs that the wav has to be in? |
| 16:49.33 | luke-jr_ | probably |
| 16:49.38 | luke-jr_ | seeing that "WAV" isn't a codec at all |
| 16:49.47 | luke-jr_ | it's just a wrapper, like AVI |
| 16:50.13 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 16:50.15 | Zodiacal | heres my .call file |
| 16:50.16 | Zodiacal | http://pastebin.com/709672 |
| 16:50.16 | luke-jr_ | or like tar/ZIP |
| 16:50.24 | Zodiacal | it works fine when i change the data to: beep |
| 16:50.28 | Zodiacal | cuz thats a .gsm |
| 16:50.33 | Zodiacal | that come with * |
| 16:50.40 | ManxPower | Zodiacal, the default, if you don't specify a path, is /var/lib/asterisk/sounds |
| 16:50.41 | [TK]D-Fender | Zodiacal : Use SOX to convert it then. |
| 16:50.48 | ctooley | What's the extension that gets jumped to after a call is hung up? it's like context,h,1 right? |
| 16:50.49 | mut | http://pastebin.ca/54777 |
| 16:50.52 | mut | [TK]D-Fender |
| 16:50.54 | mut | any idea |
| 16:50.55 | mut | ? |
| 16:50.57 | *** join/#asterisk qu4k3 (n=tommy_bo@20150027237.user.veloxzone.com.br) |
| 16:51.01 | Zodiacal | manxpower ya i put it in that dir, and i have tried putting the full path too |
| 16:51.08 | ManxPower | Zodiacal, put your .call file on pastebin |
| 16:51.20 | Zodiacal | http://pastebin.com/709672 |
| 16:51.25 | Zodiacal | manxpower |
| 16:51.30 | [TK]D-Fender | mut : fugly over-gloabalised.... |
| 16:51.39 | *** join/#asterisk ToTo (n=ToTo@host90-162.pool875.interbusiness.it) |
| 16:52.09 | [TK]D-Fender | mut : And no you can't do variable stuff like that. |
| 16:52.17 | ManxPower | Zodiacal, pastebin the CLI messages |
| 16:52.23 | mut | well i can do it the way i showed at the bottom |
| 16:52.35 | mut | i'de just have to reset priorities all over up 1 notch |
| 16:52.48 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 16:53.17 | [TK]D-Fender | mut : you can't define constants within a context like that. |
| 16:53.25 | mut | i know |
| 16:53.26 | [TK]D-Fender | mut : only in [globals] |
| 16:53.31 | mut | did you not see the example at bottom tho |
| 16:53.31 | Zodiacal | manxpower http://pastebin.com/709677 |
| 16:53.39 | mut | i can do a _.,1, set extension |
| 16:53.43 | [TK]D-Fender | mut : make 2 dialout contexts. |
| 16:53.45 | mut | before it hits anything else in any other context |
| 16:54.08 | mut | how do i know which context to go to though? |
| 16:54.10 | ManxPower | Zodiacal, too complicated. do it outside of a .call file first |
| 16:54.26 | ckwall | <[TK]D-Fender>sorry... got pulled away from the computer. |
| 16:54.32 | mut | my 2 entrance context sends the call to the same dial contexts and macros |
| 16:54.35 | ManxPower | If you use the _. pattern you will go to hell. Don't do that. |
| 16:54.49 | Hmmhesays | some dipshit is trying to guilt me into fixing his A@H box because it was "something I destroyed" |
| 16:54.57 | Zodiacal | manxpower know how i can run it from the cli? |
| 16:54.59 | Zodiacal | playback that is? |
| 16:55.01 | mut | the only way to tell where the call should go is when i enter whichever context the person is in |
| 16:55.04 | Zodiacal | or should i create a ext and try it |
| 16:55.11 | mut | when it enters that context i can set it exit point |
| 16:55.15 | ManxPower | Zodiacal, you don't. Call an extension that runs playback |
| 16:55.17 | mut | then process the call and exit on that point |
| 16:55.18 | ckwall | I am going from a polycom soundpoint ip 501 to asterisk. going out through TE410P t1 card to an adtran mx2800 to a DS3 then off to the cloud |
| 16:55.22 | Zodiacal | manxpower okie 1 sec |
| 16:55.35 | stack_ | Looks like everyone is actually working and not hanging out in IRC :) |
| 16:55.40 | stack_ | I see there is a 'w' & 'W' option for the dial command... is there one for Queues as well? |
| 16:55.48 | stack_ | sorry, wrong one :) |
| 16:56.38 | [TK]D-Fender | mut : 1 sec |
| 16:56.43 | CoffeeIV_ | I have two * connected by IAX2. When I Dial() one from the other it doesn't pass the original caller id through, however, and I want it to. How can I do that ? |
| 16:56.56 | ManxPower | CoffeeIV_, it will do so by default |
| 16:56.57 | Hmmhesays | some dipshit is trying to guilt me into fixing his a@h box, because it must have been something I did to that broke it, even though it was working fine. |
| 16:56.58 | mut | would s work for that? |
| 16:57.01 | mut | never thought of that |
| 16:57.10 | mut | exten s,1,set(defaultroute |
| 16:57.11 | ManxPower | Hmmhesays, did you ever connect into the box. |
| 16:57.16 | Hmmhesays | of course |
| 16:57.16 | mut | then continue with the dialplan |
| 16:57.28 | mut | or does s only work for macros |
| 16:57.29 | Hmmhesays | I set up everything he needed for 1 carrier, he added a 2nd and broke it |
| 16:57.33 | Hmmhesays | mut no |
| 16:57.34 | ManxPower | mut, Extensions "s" is ONLY EVER RUN if Asterisk does not know what extension to send the call to. |
| 16:57.50 | mut | hm |
| 16:58.00 | ManxPower | CoffeeIV_, well 1.2 will do that by default, at least |
| 16:58.14 | mut | argh, do you know of any other way to set variables based on context execution then? |
| 16:58.16 | ManxPower | sorry, too many noobs here todat |
| 16:58.30 | ManxPower | mut, don't don't do that. You do it based on extension. |
| 16:58.48 | mut | the extension is an outgoing call |
| 16:58.53 | mut | to NXXNXXXXXX |
| 16:59.03 | Zodiacal | manxpower i just did Answer, playback(schoolbell), hangup. i didn't hear it play into the phone either: http://pastebin.com/709683 |
| 16:59.07 | mut | and can possibly go out on 2 different trunks |
| 16:59.11 | ManxPower | mut, well, there's an exten => line in there SOMEWHERE to run the Dial |
| 16:59.14 | Zodiacal | manxpower the call file trys to play it thru my soundcard.. |
| 16:59.26 | mut | so i have to double my entire dial plan to dial out a seperate trunk? |
| 16:59.38 | [TK]D-Fender | mut :http://pastebin.ca/54806 |
| 16:59.47 | CoffeeIV_ | ManxPower, it didn't do so by default, and both asterisks are 1.2 -- one is older than the other though -- is there a way to pass that info explicitly ? |
| 16:59.49 | ManxPower | Zodiacal, Asterisk thnks it played the file. |
| 16:59.59 | mut | right |
| 17:00.06 | mut | i'm not going to double my entire dial plan |
| 17:00.11 | Zodiacal | manxpower it didn't :/ |
| 17:00.19 | mut | i'll just do it with the _.,1,set method |
| 17:00.24 | mut | and renumber priorities |
| 17:00.24 | ManxPower | CoffeeIV_, If it doesn't do it by default then you have some OTHER problem. |
| 17:00.35 | ManxPower | mut, Best of luck with that. |
| 17:00.35 | [TK]D-Fender | mut : I just gave you the final answer |
| 17:00.40 | Hmmhesays | I'm going to fix this guys stuff and tell him to go fsck himself |
| 17:00.50 | mut | [TK]D-Fender: right, that dial plan i pasted ya yesterday |
| 17:00.52 | CoffeeIV_ | ManxPower: thanks, I'll investigate further and do a pastebin if necessary |
| 17:00.55 | mut | i'de have to double the entire thing |
| 17:00.58 | [TK]D-Fender | ckwall : Your SIP phone should be RFC2833, your T1 should be INBAND |
| 17:01.03 | mut | just to send it out trunk 1 or trunk 2 |
| 17:01.12 | mut | or instead i could renumber the priorities +1 |
| 17:01.17 | mut | and do it the other way |
| 17:01.23 | [TK]D-Fender | mut : thats 2 lines! Good grief! You non-functional idea took MORE lines! |
| 17:01.23 | ManxPower | mut, what not use the 1.2 "n" priorities |
| 17:01.36 | *** part/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 17:01.47 | mut | tk |
| 17:01.56 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 17:01.57 | mut | what happens when i sent it to a context before my dialout? |
| 17:02.04 | mut | i have to double that context as well don't i? |
| 17:02.10 | [TK]D-Fender | mut : thats what contexts are for.. to choose how to process things, unfortunately the SCOPE of a CONSTANT is GLOBAL. thats it, thats all. |
| 17:02.41 | [TK]D-Fender | mut : You're working too hard for nothing... scale it up for me to see how bad you think it'll get... |
| 17:03.10 | mut | ok |
| 17:03.26 | elvisthedj | somebody better check this out http://pastebin.com/709690 |
| 17:03.38 | Zodiacal | manxpower any ideaS? |
| 17:03.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.79.93) |
| 17:03.42 | Zodiacal | S = s |
| 17:05.18 | CoffeeIV_ | Is it possible that Callerid not passed though when connecting between two * servers doing Dial(IAX2/server/whatever) but it IS passed through when connecting via "swtich => IAX2/server" ? |
| 17:06.01 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 17:07.25 | [TK]D-Fender | CoffeeIV_ : It'd get overridden with Dial(IAX2/server/whatever) if your peer entry SETS callerid. |
| 17:11.17 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-143-38-79.dsl.tul2ok.sbcglobal.net) |
| 17:12.18 | Zodiacal | manxpower nm, i converted the .wav to gsm, seems ok quality |
| 17:12.19 | Zodiacal | thanks! |
| 17:13.09 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:19.24 | [Airwolf] | Can anyone tell me if it's correct that regular expressions aren't possible in If statements ? |
| 17:19.35 | [Airwolf] | Set(var=${IF($[${EXTEN:4}=65[1-9]]?s-conf:s-call)}) |
| 17:19.42 | [Airwolf] | Because that doesn,t work. |
| 17:20.07 | [Airwolf] | And if I remove the expression and just replace it with "1" it does work |
| 17:21.40 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 17:22.00 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 17:23.58 | dlynes | [Airwolf]: it was my understanding that regex was only valid in extension patterns |
| 17:24.04 | dlynes | [Airwolf]: but, i could be wrong |
| 17:25.59 | dlynes | Has anyone that's on atm ever worked with H323 and Asterisk? |
| 17:28.09 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 17:29.03 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:29.32 | [Airwolf] | dlynes, hmm, that would suck if it's true :P |
| 17:29.47 | fjean | hey guys, how are you, anybody using CentOS ? I need to setup the symbolic link to the kernel source in /usr/src |
| 17:30.07 | TripleFFFFFFFFFF | anyone know if somethign wrong with 1.2.7.1 AGI vars ?.. im setting callerid via agi and its not workingright |
| 17:30.09 | [Airwolf] | fjean, the symbolink link to what ? |
| 17:30.24 | fjean | to the kernel sources, sorry |
| 17:31.49 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 17:32.02 | *** join/#asterisk triple-e (n=piaergnj@adsl-70-128-78-22.dsl.stlsmo.swbell.net) |
| 17:32.52 | triple-e | commence hair pulling |
| 17:33.28 | TripleFFFFFFFFFF | lol |
| 17:33.35 | Hmmhesays | how does asterisk match an incoming peer when insecure=very? |
| 17:33.47 | TripleFFFFFFFFFF | its now.. very,port |
| 17:33.51 | Hmmhesays | is it based on username ip addy er what? |
| 17:33.52 | [Airwolf] | fjean, what do you mean |
| 17:33.53 | TripleFFFFFFFFFF | i mean invite,,port |
| 17:34.06 | [Airwolf] | You want to create a symlink from your kernel sources to /usr/src/linux ? |
| 17:34.18 | triple-e | why doesn't this work |
| 17:34.19 | triple-e | exten => 205,1,Goto(outbound-allroutes,13143693609,1) |
| 17:34.33 | fjean | airwolf, yes, what should be the link name ? linux ? |
| 17:34.52 | [Airwolf] | yes, thats the normale name |
| 17:34.57 | [TK]D-Fender | Hmmhesays : the "from" field (before the @). then checks the MD5 on the user/pass |
| 17:34.59 | jake1932 | triple-e: what's the error? |
| 17:35.07 | Hmmhesays | good to know |
| 17:35.09 | Hmmhesays | danke |
| 17:35.15 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:35.25 | triple-e | CLI shows no error |
| 17:35.25 | Hmmhesays | what if there is nothing in the from field before the @ |
| 17:35.34 | [TK]D-Fender | Hmmhesays : Which can royaal screw you up on unauthed calls matching user extensions... |
| 17:35.44 | [TK]D-Fender | Hmmhesays : dunno |
| 17:35.45 | jake1932 | triple-e: pastebin exensions.conf |
| 17:35.46 | triple-e | softphone has 603 declined "person unavailable" |
| 17:36.11 | mercestes | hey...forwarding a polycom phone to a cellphone via sipconnect....I get no voice path...any ideas? both legs are establishing g711u and I have canreinvite=no. Nat settings are having no effect. |
| 17:36.11 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:36.59 | jake1932 | is sipconnect a ITSP? |
| 17:37.14 | mercestes | yea basically.. |
| 17:37.17 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 17:37.56 | elvisthedj | This is probably dumb, but does anyone know how to split a conferenced call on a 79xx phone (running sip 7.5)? I can only drop both calls.. want it to work like 3-way where i can drop the last. Right now i'm doinga soft hangup on the console to get rid of the 3rd party |
| 17:38.10 | mercestes | I'm coming sip into the phone, getting the Got SIP response 302 "Moved Temporarily" back...and then forwarding back out SIP via provider... |
| 17:38.13 | *** part/#asterisk alerios (n=alerios@201.244.243.43) |
| 17:38.27 | mercestes | g711u on both sides..under sip channels... |
| 17:38.38 | mercestes | no voice path. |
| 17:39.08 | jake1932 | have you done a sip debug yet? |
| 17:39.11 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 17:39.17 | triple-e | pastbin url ? |
| 17:39.22 | jake1932 | ~pb |
| 17:39.24 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:39.26 | mercestes | doing that now. |
| 17:39.39 | Hmmhesays | yeah I just proved this guy wrong, can't wait for his response |
| 17:40.00 | japerry | has anyone had troubles with a TP100 T1 card where someone would call but get nothing-- then a fast busy after about 30secs? |
| 17:40.23 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 17:41.29 | Hmmhesays | I think you are mistaken on that from field [TK]D-Fender |
| 17:41.40 | jake1932 | if they find Keith Richards has brain damage, how can they be sure it didn't happen before the accident? |
| 17:43.04 | TripleFFFFFFFFFF | hey |
| 17:43.08 | noky | how can i change the context of a dialpplan/ |
| 17:43.10 | TripleFFFFFFFFFF | how you copy data from one to anoter |
| 17:43.21 | TripleFFFFFFFFFF | mysql.. like .. insert into blah select * form blue ? |
| 17:43.21 | elvisthedj | Keith's head looks like it's worn out three bodies by now |
| 17:43.46 | jake1932 | ha |
| 17:44.17 | noky | if my call enter in a context 'incoming' and i want to change this context to other... how can i do this ?!... |
| 17:44.21 | jake1932 | noky: use a text editor? can you be more specific |
| 17:44.34 | noky | sorry jake1932 i was writting |
| 17:45.11 | jake1932 | for sip? |
| 17:45.18 | noky | mmmm |
| 17:46.04 | *** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:46.10 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 17:46.30 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 17:46.32 | noky | no... look my h323 calls enter by a context called 'incoming'... when this context match with some extension i want to change the context |
| 17:46.46 | noky | Goto() ? |
| 17:46.46 | jake1932 | noky: just go to the config file whatever technology your using and change the context=blah |
| 17:47.05 | jake1932 | h323.conf |
| 17:47.12 | jake1932 | not sure if that's the file |
| 17:47.47 | elvisthedj | noky If you want the h323 calls to come into incoming and only move some of them to a new context based on the extension, then yes you'd use Goto |
| 17:47.51 | triple-e | http://pastebin.ca/54822 |
| 17:48.09 | noky | ok thanks |
| 17:48.20 | jake1932 | oops - missed the 2nd part |
| 17:48.44 | Hmmhesays | that doesn't make sense, on the initial from field in the invite, you get the number that i'm calling from, It looks more like it matches the to: field |
| 17:49.57 | jake1932 | aarg |
| 17:51.26 | [TK]D-Fender | Hmmhesays : The From field seems to prove right an aweful lot |
| 17:51.36 | Hmmhesays | i'm looking into it more |
| 17:51.50 | jake1932 | triple-e: there is no context [outbound-allroutes] in the file you posted |
| 17:52.01 | [TK]D-Fender | Hmmhesays : I proved it for a number of people using un-authed calls that clash with local SIP phones |
| 17:52.06 | Hmmhesays | k |
| 17:52.32 | triple-e | outbound-allroutes is in extensions_additional.conf |
| 17:52.47 | Hmmhesays | in my vonage incoming the from field is always the number i'm calling from |
| 17:53.13 | b00mer_ | I am having echo problems on calls going out my PRI. Not every call... but it has been common on some calls to people with POTS lines. I turned on echotraining and bumped up my rxgain +2, but still having issues. Any ideas? |
| 17:53.47 | b00mer_ | I can reproduce on certain numbers, again to POTS lines... how can I debug the issues? |
| 17:54.06 | jake1932 | triple-e: does it work in extensions.conf? |
| 17:54.44 | [TK]D-Fender | Hmmhesays : Keep in mind thats why in most ITSPS you use your DID as your exten => line :) |
| 17:54.56 | jake1932 | btw - this really isn't the channel for "Asterisk Management Portal (AMP)" |
| 17:55.00 | triple-e | if i put that exten line in extensions.conf -- just inside the from internal context it goes to from internal but bombs out |
| 17:55.04 | *** join/#asterisk nain (n=nain@202.59.90.182) |
| 17:55.07 | nain | Hi |
| 17:55.15 | [TK]D-Fender | Hmmhesays : errr... TO normally... think I'm mixing myself up |
| 17:55.18 | triple-e | im pasting the extensions_additional |
| 17:55.21 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 17:55.21 | Hmmhesays | when the "from" contains the number i'm calling from the asterisk shouldn't find a match |
| 17:55.49 | triple-e | http://pastebin.ca/54828 |
| 17:57.08 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 17:58.30 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 17:58.50 | jake1932 | triple-e: doesn't look like you have any valid path for that exten you mentioned |
| 17:59.36 | nain | Is there any softphone which allow us to change frame rate or even in asterisk some thing like so we can change the codec frame rate similar to astersik-oh323 drvier settings. codec=gsm, frame=2 ???? |
| 18:00.22 | triple-e | outbound-allroutes line 648 http://pastebin.ca/54828 |
| 18:00.25 | jake1932 | triple-e: you need exten => _1NXXNXXXXXX,n,Command |
| 18:00.51 | jake1932 | at least somewhere |
| 18:01.24 | triple-e | exten => 205,1,Goto(outbound-allroutes,13143693609,1) <== this used to work on 1.2.1 |
| 18:02.13 | jake1932 | i just looked at your dialplan - there is no path for _1NXXNXXXXXX in extensions.conf and your addl file.conf |
| 18:02.40 | jake1932 | therefore there is no match and it fails |
| 18:03.01 | joerg | hi... |
| 18:03.17 | joerg | chan_iax2.c:1254 try_firmware: Firmware file '/var/lib/asterisk/firmware/iax/iaxy.bin' fails checksum |
| 18:03.27 | joerg | is there something broken in the latest cvs version? |
| 18:03.53 | dlynes_ | noky: I see you're using h323 |
| 18:03.59 | dlynes_ | noky: Just curious how stable it is? |
| 18:04.27 | noky | my asterisk unregister with my gatekeeper |
| 18:04.30 | nain | Hmmhesays: could you plz advise me ? Is there any softphone which allow us to change frame rate or even in asterisk some thing like so we can change the codec frame rate similar to astersik-oh323 drvier settings. codec=gsm, frame=2 ???? |
| 18:04.31 | noky | i'm using ooh323 |
| 18:04.45 | Hmmhesays | nain: no idea |
| 18:04.47 | noky | i don't know why was unregistering |
| 18:05.03 | noky | dlynes_: :/ is not good stable... |
| 18:05.16 | dlynes_ | no? |
| 18:05.34 | dlynes_ | That sucks...we were kinda hoping to start using this one carrier |
| 18:05.35 | noky | no... but i'm testing... |
| 18:05.45 | dlynes_ | And they don't support SIP; they only support H.323 |
| 18:06.03 | dlynes_ | Is it chan_h323 that isn't stable, or is it Open H323? |
| 18:06.30 | noky | i think that chan_h323 but i don't test so much |
| 18:06.37 | dlynes_ | ah |
| 18:06.47 | dlynes_ | Ok...just so I know where to start looking |
| 18:08.06 | fjean | oh |
| 18:08.24 | fjean | I got this error with zaptel 1.2.5: |
| 18:08.38 | fjean | CC [M] /usr/src/zaptel-1.2.5/zaptel.o |
| 18:08.47 | fjean | zaptel.c:384: error: syntax error before "zone_lock" |
| 18:08.48 | fjean | ? |
| 18:08.55 | dlynes_ | nice |
| 18:09.01 | dlynes_ | fjean: Which kernel version? |
| 18:09.04 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 18:09.16 | fjean | 2.6.9-22.ELsmp |
| 18:09.20 | noky | i have a question with IVR... if my IVR answer my call and wait for some digit will pressed... the dtmf that i dial.. only is an extension not?!... in my ivr i need differents context like "Customer" and "Technology" with your extension..... |
| 18:09.32 | dlynes_ | fjean: Do you have udev installed, and do you have kernel-dev installed? |
| 18:09.37 | noky | sorry about my english... i suck in englis |
| 18:09.38 | noky | h |
| 18:09.44 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 18:09.46 | triple-e | OMG -- i can't believe it was something so bloody simple -- thanks for the second set of eyes jake1932 |
| 18:09.56 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 18:10.09 | dlynes_ | noky: Background(...), and hten say the user hits '867', it'll jump to extension 867 in the current context |
| 18:10.22 | jake1932 | triple-e: np |
| 18:10.43 | dlynes_ | noky: So then you could do in the current context: exten => 867,1,Goto(Customer,1,1) |
| 18:10.55 | noky | dlynes_: thanks you! |
| 18:11.01 | fjean | dynes_: yes... |
| 18:11.21 | dlynes_ | fjean: ok, and the kernel-dev you've got installed matches the kernel version you're using? |
| 18:11.39 | fjean | dynes_: let me check |
| 18:12.16 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 18:12.35 | b00mer_ | I am having echo problems on calls going out my PRI. Not every call... but it has been common on some calls to people with POTS lines. I turned on echotraining and bumped up my rxgain +2, but still having issues. Any ideas? |
| 18:13.35 | dlynes_ | b00mer_: Some numbers you're not going to be able to clear up the echo, no matter what you do |
| 18:13.52 | dlynes_ | b00mer_: those numbers might just have bad acoustics in their handsets |
| 18:13.53 | jake1932 | yeah - some people have real crappy phones |
| 18:14.10 | luke-jr_ | Can anyone get ahold of SellVoIP? |
| 18:14.15 | b00mer_ | dlynes: but why would I not hear it before? |
| 18:14.24 | dlynes_ | b00mer_: on what? |
| 18:14.32 | b00mer_ | dlynes: non-voip phone |
| 18:14.44 | dlynes_ | b00mer_: analog->analog probalby |
| 18:14.48 | b00mer_ | dlynes: or on a vonage phone calling the same numbers |
| 18:15.14 | dlynes_ | b00mer_: good question...in that case, I would suspect it might be the hardware you're using, or the driver |
| 18:15.24 | dlynes_ | b00mer_: Which particular hardware are you using? |
| 18:15.33 | dlynes_ | b00mer_: If it's the x100p, i'd stop looking |
| 18:15.36 | fjean | dlynes_: udev i386 039 10.10.EL4 / kernel-devel i686 2.6.9 34.EL / kernel i686 2.6.9 22.EL |
| 18:15.43 | b00mer_ | dlynes: cisco 7960 |
| 18:15.44 | dlynes_ | b00mer_: x100p's a piece of crap |
| 18:15.50 | dlynes_ | b00mer_: no...for the pots line |
| 18:15.53 | b00mer_ | dlynes: Sagonma T1 |
| 18:16.10 | b00mer_ | pots line is the number being called |
| 18:16.11 | dlynes_ | b00mer_: are you using a hardware echo canceller? |
| 18:16.23 | b00mer_ | I'm having issues from my PRI to their POTS |
| 18:16.35 | dlynes_ | b00mer_: Are you using hte sangoma hardware echo canceller? |
| 18:16.38 | b00mer_ | don't have a hardware echo cancelleer |
| 18:16.44 | dlynes_ | b00mer_: Then buy one |
| 18:16.47 | b00mer_ | :( |
| 18:17.06 | dlynes_ | b00mer_: the echo cancellers in zaptel are far from being top of the line |
| 18:17.14 | dlynes_ | b00mer_: but they're sufficient for most situations |
| 18:17.26 | b00mer_ | dlynes_: do they even make one on their one port cards? |
| 18:17.29 | [TK]D-Fender | dlynes : They're likely worth exactly what you paid for them ;) |
| 18:17.49 | dlynes_ | nothing! |
| 18:17.50 | dlynes_ | :) |
| 18:18.01 | b00mer_ | so $500 for a PRI card is cheap... I should have bought a more expensive one? |
| 18:18.03 | [TK]D-Fender | dlynes : And maybe more! |
| 18:18.10 | dlynes_ | nah...the echo cancellers in zaptel are ok...but they're still not that great |
| 18:18.23 | dlynes_ | b00mer_: no...just buy the e104d(?) |
| 18:18.30 | [TK]D-Fender | b00mer_ : For some Zaptel works fine, for others crap. HWEC on the card can make all the difference |
| 18:18.35 | b00mer_ | eek 4T1s is overkill |
| 18:18.35 | dlynes_ | b00mer_: it's the echo canceller addon for sangoma's pri cards |
| 18:18.50 | b00mer_ | i'll check in to it... thanks |
| 18:18.52 | dlynes_ | b00mer_: the 104d i think it is, is just an echo canceller...no pri card |
| 18:18.58 | nain | Any Advise on Setting Frame Rate in Asterisk? |
| 18:19.04 | asterboy | When does the /dev/dsp device get created? On * load if compiled to use module? |
| 18:19.24 | dlynes_ | asterboy: when you install a soundcard |
| 18:19.41 | [TK]D-Fender | dlynes : Never seen the EC sold seperately. the A104d is 4-port PRI w/ HWEC |
| 18:19.48 | dlynes_ | [TK]D-Fender: ah |
| 18:19.51 | asterboy | hmm, I have a sound card installed and compiled into kernel...no dsp though |
| 18:20.16 | asterboy | checking dmesg...maybe its disabled in BIOS |
| 18:20.28 | Corydon-w | asterboy: or it's the wrong driver |
| 18:21.34 | dlynes_ | asterboy: The /dev/dsp device usually gets created as part of your makedev package in your distribution for kernel 2.4.x and lower, or as part of the udev package, in 2.6 and higher |
| 18:21.55 | dlynes_ | asterboy: but you have to have the waveform driver loaded for your soundcard for that to be there |
| 18:23.24 | asterboy | ya, dmesg is not showing any sound devices. |
| 18:23.46 | dlynes_ | asterboy: are you using 2.6.x? |
| 18:23.52 | asterboy | make menuconfig shows that it has sound enabled.. |
| 18:24.00 | asterboy | ...yes, using 2.6 |
| 18:24.03 | asterboy | udev |
| 18:24.03 | dlynes_ | asterboy: as a module, or in the kernel? |
| 18:24.08 | asterboy | in kernel |
| 18:24.59 | dlynes_ | asterboy: cat /proc/devices |
| 18:24.59 | asterboy | Not on site, so I will have to check to see if the onboard is disabled in BIOS. |
| 18:25.09 | dlynes_ | asterboy: do you see 'alsa' listed? |
| 18:26.27 | asterboy | no alsa but sound is showing on 14 |
| 18:26.39 | dlynes_ | asterboy: how about oss? |
| 18:26.50 | dlynes_ | although lord only knows why you'd be using oss :) |
| 18:26.57 | asterboy | show modules in cli shows no oss |
| 18:27.03 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 18:27.18 | dlynes_ | who cares about asterisk at this point? if the kernel can't see it, asterisk sure as heck ain't gonna see it |
| 18:27.28 | asterboy | yep |
| 18:27.39 | asterboy | need oss so I can setup paging |
| 18:27.39 | dlynes_ | asterboy: so does cat /proc/devices show oss? |
| 18:27.48 | dlynes_ | asterboy: you can use alsa, too |
| 18:27.56 | brif8 | I have a message that plays with timeout and digit timeout set yet it never goes to i or t why ? |
| 18:28.16 | asterboy | no proc devs has no oss |
| 18:28.23 | asterboy | and no alsa |
| 18:28.30 | dlynes_ | asterboy: but it does have snd, right? |
| 18:28.46 | asterboy | but oss doesn't show normally there anyway does it? Thought is was an * module |
| 18:29.04 | dlynes_ | oss == open sound system |
| 18:29.05 | asterboy | yes there is sound, but if I disabled it in BIOS it sure isn't going to load |
| 18:29.12 | dlynes_ | alsa == advanced linux sound architecture |
| 18:29.18 | asterboy | ah |
| 18:29.19 | dlynes_ | it's two competing sound driver standards for linux |
| 18:29.27 | dlynes_ | nothing to do with asterisk |
| 18:29.28 | tzanger | [TK]D-Fender: around? |
| 18:29.35 | dlynes_ | asterisk just gives you the choice of using either |
| 18:29.36 | *** join/#asterisk MacDome (n=eseidel@A17-255-100-64.apple.com) |
| 18:29.51 | dlynes_ | asterboy: so you can use either chan_alsa, or chan_oss, but not both |
| 18:30.23 | dlynes_ | asterboy: anyways, it sounds like you've got soundcard support loaded, but not the soundcard driver for your particular soundcard |
| 18:30.34 | asterboy | ok, then I may not have compiled in to kernel, cause I just have sound card suppport but no alsa or oss checked off |
| 18:30.58 | dlynes_ | asterboy: yeah, and with 2.6, i don't think oss is even supported in the kernel anymore |
| 18:31.08 | brif8 | I have a message that plays with timeout and digit timeout set yet it never goes to i or t why ? http://pastebin.com/709901 I never see i,1,NoOp or t,1,NoOp why ? |
| 18:31.09 | asterboy | ya, it says (depricated) |
| 18:31.12 | dlynes_ | asterboy: only oss emulation, using alsa for older programs that only support oss |
| 18:31.52 | asterboy | ok, great help...when I go onsite, I'll make sure BIOS is enabled and alsa is compiled into kernel. |
| 18:32.04 | asterboy | then I'll setup oss.conf appropriately |
| 18:32.16 | *** join/#asterisk bmg505 (n=leon@c1-55-7.rndf.isadsl.co.za) |
| 18:32.19 | asterboy | as long as I get /dev/dsp, should be ok |
| 18:32.21 | dlynes_ | asterboy: just compile the alsa soundcard driver as a module |
| 18:32.29 | asterboy | why? |
| 18:32.31 | dlynes_ | asterboy: no point compiling everything straight into the kernel |
| 18:32.53 | dlynes_ | asterboy: well, if something happens that fubars the soundcard |
| 18:33.06 | dlynes_ | asterboy: you might not be able to boot up |
| 18:33.18 | asterboy | then I have to try and remember how to add modules into boot, or does that happen auto with udev |
| 18:33.18 | dlynes_ | asterboy: if it's a module, you can disable the module load, and life will go on |
| 18:33.24 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 18:33.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:33.44 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 18:33.49 | asterboy | modprobe something |
| 18:33.49 | dlynes_ | asterboy: pci hotplug will usually take care of that |
| 18:33.59 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:34.07 | asterboy | ok, now its coming back to me |
| 18:34.19 | dlynes_ | asterboy: i think pci hotplug is new as of 2.6 though |
| 18:34.23 | dlynes_ | asterboy: but not sure |
| 18:34.30 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 18:34.32 | dlynes_ | asterboy: in 2.4 hotplug was used for pcmcia and usb though |
| 18:35.05 | gandhijee | what processor does asterisk optimize for if you don't set them in the Makefile? |
| 18:35.31 | *** join/#asterisk littlejohn (n=little@host133-99.pool8256.interbusiness.it) |
| 18:35.44 | dlynes_ | asterboy: i don't know what distro you're using, but, on slackware if you disable rc.hotplug, you can manually specify all driver loads in /etc/rc.d/rc.modules, using modprobe lines |
| 18:36.21 | asterboy | I'm lfs, no rc.modules |
| 18:36.30 | [Airwolf] | dlynes, do you perhaps know how I can do a logical OR in an if statement ? |
| 18:36.31 | dlynes_ | no idea what lfs is |
| 18:36.40 | gandhijee | Linux from Scratch |
| 18:36.59 | dlynes_ | [Airwolf]: you mean a normal OR, not a logical bitwise OR? |
| 18:37.04 | [Airwolf] | gandhijee, oeh, hoe to make it yourself difficult :P |
| 18:37.10 | [Airwolf] | dlynes, just a normal OR |
| 18:37.16 | dlynes_ | [Airwolf]: using the '|' operator |
| 18:37.38 | [Airwolf] | ah, I was trying the java way, with || |
| 18:37.48 | gandhijee | airwolf: huh? |
| 18:38.03 | dlynes_ | brif8: they've probably gotten to the wait(10) by the time they hit something |
| 18:38.22 | asterboy | how does hotplug look in the precess list? |
| 18:38.42 | brif8 | dlynes_: but surely the timeout should kick in first before the wait(10) ? |
| 18:38.48 | asterboy | I have hotplug in my rc.d/init.d but I don't think it's loading in the rc3.d |
| 18:38.53 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 18:39.33 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 18:39.46 | *** join/#asterisk jef (i=fischer@pdpc/supporter/active/Jef) |
| 18:39.47 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 18:39.48 | stack_ | The Dial command has 'w' & 'W' for recording... does Queue have something similar? |
| 18:39.51 | [Airwolf] | gandhijee, with lfs I mean |
| 18:39.54 | dlynes_ | brif8: As the documentation for the wait command states, wait() silently ignores any dtmf tones; if you want to wait for a response and capture dtmf tones instead, set your response timeout to a good number, and don't define anything after the background() |
| 18:40.08 | dlynes_ | brif8: iow, don't use Wait() :) |
| 18:40.17 | asterboy | Airwolf, using nalfs is a breeze |
| 18:40.30 | asterboy | /nalfs/alfs |
| 18:40.44 | [Airwolf] | asterboy, well, I just like to have some package manager avalible |
| 18:40.49 | dlynes_ | gandhijee: i c...i like my slackware just fine, though :) It's nice and easy to work with, and nice and stable |
| 18:40.52 | [Airwolf] | I use gentoo myself |
| 18:41.34 | gandhijee | as do i |
| 18:41.37 | brif8 | dlynes: what changes do you suggest? set timeout to ??? and then how do I rotate if they don't press anything ? |
| 18:41.52 | gandhijee | emerge is the bomb |
| 18:42.00 | SwK[work] | http://www.funroll-loops.org |
| 18:42.30 | asterboy | Airwolf, use paco |
| 18:42.48 | asterboy | Check out the tiny process list you get with LFS, http://pastebin.ca/54842 |
| 18:43.05 | asterboy | This is running on my 233MHz |
| 18:43.09 | dlynes_ | brif8: Set(TIMEOUT(response)=13) ; Background(office-hours) ; and then nothing else after it |
| 18:43.41 | gandhijee | how long did it take you to setup LFS? |
| 18:43.47 | dlynes_ | brif8: and try playing around with timeout(response)...you might need to put it after the background |
| 18:44.06 | [Airwolf] | asterboy, ok, true |
| 18:44.08 | asterboy | My first install was a few days to get past learning curve, but well worth it. |
| 18:44.10 | [Airwolf] | That is not much |
| 18:44.16 | brif8 | dlynes_: I was going to ask that when does the timout timer start when it it called or after the message plays ? |
| 18:44.21 | dlynes_ | brif8: you can set the digit timeout too...i just forgot to type that in |
| 18:44.25 | [Airwolf] | Maybe in my vacation, I will try it. |
| 18:44.39 | [Airwolf] | Can anyone tell me what a "SIP response 503 "Error switch" means ? |
| 18:44.43 | asterboy | Now I setup LFS in minutes using dd |
| 18:44.44 | dlynes_ | brif8: but do the digit timeout before, just to be on the safe side |
| 18:44.47 | brif8 | dlynes_: I have that http://pastebin.com/709901 |
| 18:45.14 | asterboy | alfs is can be a couple of hours, depending on how fast your machine can compile |
| 18:45.16 | dlynes_ | brif8: I'm not sure...I was trying to use the wait() application myself the same way you are |
| 18:45.32 | dlynes_ | brif8: and then when I just looked up the documentation for it, that when i realized you can't use it for that |
| 18:45.56 | brif8 | dlynes_: I had to. when I first did NOT have it it would play the message and then hangup and not go back and replay the message again |
| 18:46.00 | dlynes_ | brif8: it probably needs to go after the background() |
| 18:46.08 | brif8 | also is there a way to count the number of times it plays the message ? |
| 18:46.27 | dlynes_ | brif8: but you've had the goto() after it the whole time, right? |
| 18:46.46 | brif8 | no added the goto with the wait |
| 18:46.56 | dlynes_ | brif8: yeah...lose the goto |
| 18:47.06 | dlynes_ | brif8: it's already in the t, extension |
| 18:47.18 | brif8 | that's what I |
| 18:47.30 | brif8 | 'm saying the t extension never fired |
| 18:47.51 | dlynes_ | Yeah...lose the wait, put the response timeout after the background call, and make sure nothing is after the response timeout |
| 18:48.00 | dlynes_ | Set the response timeout to 10, instead of 3 |
| 18:48.42 | dlynes_ | brif8: regarding counting the number of times it has played the message, you can do that with variables, or with the asterisk database |
| 18:48.54 | dlynes_ | brif8: variables are the easiest way to do it |
| 18:49.08 | brif8 | ok I'll do that thanks |
| 18:52.01 | dlynes_ | brif8: yeah...just tested it |
| 18:52.21 | extremis | How do I speed up the boot process on a cisoc 7960? Recent versions of the firmware take up to 5 minutes to "configuring VLAN" |
| 18:52.22 | dlynes_ | brif8: Background() or Playback(), and then Set(TIMEOUT(response)=10) |
| 18:52.25 | brif8 | can you pastebin you test so I can check it against mine |
| 18:54.39 | dlynes_ | http://pastebin.com/709954 |
| 18:56.02 | asterboy | I did gentoo, but once I moved to lfs, never looked back |
| 18:56.24 | asterboy | Tried slackware, even bought the T-shirt, but still prefer my lfs |
| 18:57.44 | asterboy | I want a VIM 7.0 T-shirt... |
| 18:57.52 | dlynes_ | vim 7.0's out now? |
| 18:57.58 | asterboy | ...that will pickup the ladies. |
| 18:58.22 | Kyler | I was asked again today about sending faxes through Asterisk. I caught part of the conversation about it here recently and it sounds like it's still not reliable. Will SpanDSP at least work sometimes and fail gracefully otherwise? I just need to be able to send faxes. |
| 18:58.25 | asterboy | "hey baby, I'm on vim 7.0!"..."Wanna head back to my place?" |
| 18:58.37 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-51-77.w86-217.abo.wanadoo.fr) |
| 18:59.05 | dlynes_ | Kyler: i've heard from some people that have absolutely no problems with it, and other people that can never get it to work...ymmv |
| 18:59.32 | asterboy | ftp://ftp.vim.org/pub/vim/unix/vim-7.0.tar.bz2 |
| 18:59.41 | gandhijee | what would you guys recommend for a DHCP server? |
| 18:59.50 | dlynes_ | Kyler: most of the people i've heard from that can get it work 100% of the time are using specialized equipment too, i think, or they're only using app_rxfax/app_txfax |
| 18:59.51 | Kyler | dlynes_: I'm more concerned about something very bad in the middle - like having it report that a fax was successfully sent when it wasn't, or crashing. |
| 18:59.52 | gandhijee | dhcp-agent or dhcp from ISC |
| 18:59.55 | asterboy | lfs on a P100 |
| 19:00.06 | dlynes_ | Kyler: oh...never had anything like that happening |
| 19:00.12 | dlynes_ | Kyler: if it doens't work, you'll know |
| 19:00.22 | dlynes_ | Kyler: and i've never had it crash, nor heard of it crashing |
| 19:00.52 | dlynes_ | gandhijee: isc-dhcpd |
| 19:01.12 | Kyler | I can use app_txfax. I don't need tight Asterisk integration right now. |
| 19:01.40 | dlynes_ | gandhijee: it supports all the latest gadgetry, and can even do custom configuration directives for those devices that need it |
| 19:01.40 | Kyler | dlynes_: What do you mean by "specialized equipment"? I'm running in a Xen guest domain right now. That concerns me. |
| 19:01.40 | asterboy | Hylafax works great for me. |
| 19:01.46 | cybergypsy | my landline dial command is exiting with :- exited non-zero on 'Zap/1-1' , which means it doesn`t go to my [hangup] context , how can i force it to do so ? |
| 19:01.51 | dlynes_ | Kyler: no idea what a xen guest domain is |
| 19:01.55 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:02.02 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 19:02.06 | Kyler | Xen == virtual machine hypervisor |
| 19:02.17 | dlynes_ | ok, whatever a hypervisor is :) |
| 19:02.38 | dlynes_ | but anyways...sounds like a vm |
| 19:02.39 | Kyler | I'd be willing to locate on a dedicated box with "specialized equipment" if that helps. |
| 19:03.02 | dlynes_ | Kyler: if you use app_txfax/app_rxfax, and you're not using freebsd, you should be fine |
| 19:03.21 | Kyler | dlynes_: O.k., time to dive in. Thanks! |
| 19:03.22 | dlynes_ | Kyler: you might get the odd fax that gets an error on the second page, but that's about it |
| 19:03.53 | dlynes_ | Kyler: keep in mind that spandsp only operates at 9600 |
| 19:04.06 | dlynes_ | Kyler: so you won't be able to talk to fax machines that do 14,400 or higher |
| 19:04.40 | Kyler | dlynes_: Oh...that's a problem. I'm looking at handling hundreds of pages a day. |
| 19:05.03 | Kyler | So is there hardware that can handle this (short of going to analog and back)? |
| 19:05.37 | dlynes_ | Kyler: Well, you could use sipura 2000's or something |
| 19:05.46 | dlynes_ | Kyler: and have a t.38 gateway that hooks into a pri |
| 19:06.22 | Kyler | I'd like something that just uses IP (no analog, no PRI, ...). |
| 19:06.33 | dlynes_ | normally what vonage does is configure port 2 on the sipura 2000's for t.38 |
| 19:06.42 | dlynes_ | and then it connects to their t.38 gateway |
| 19:06.52 | dlynes_ | Kyler: you'd need to have a terminator that supports t.38 then |
| 19:07.11 | dlynes_ | Kyler: app_txfax/app_rxfax both work on pri/t1/... |
| 19:07.32 | dlynes_ | Kyler: ulaw is not reliable for faxing, and probably never will be |
| 19:07.32 | Kyler | Hmmm...I need to digest that. I looked into t.38 earlier and got the feeling that it wasn't for me. |
| 19:08.13 | Kyler | I'd like to have some box that can accept pages over IP (IPP? LPD?) and speak ADPCM (or whatever) via SIP. |
| 19:08.20 | dlynes_ | Kyler: anyways...needless to say, if you go the t.38 right for now, I don't think you'll be able to send it through asterisk |
| 19:08.44 | dlynes_ | s/right/route |
| 19:08.56 | brif8 | dlynes_: You used Playback though and not background to play the messages with background and then timeout it hangs ups afterward |
| 19:09.01 | Kyler | I could handle using a dedicated provider/account just for sending faxes. |
| 19:09.13 | dlynes_ | brif8: Background, Playback...doesn't make a different |
| 19:09.14 | dlynes_ | brif8: Background, Playback...doesn't make a difference |
| 19:09.53 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
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| 19:10.00 | brif8 | yes with Playback you can't press anything while the message plays with background it return control right away according to the wiki |
| 19:10.02 | dlynes_ | brif8: and i do not hang up right away...it'll keep looping until the caller hangs up |
| 19:10.33 | dlynes_ | brif8: Yes, but you can put Background() in there instead |
| 19:10.45 | dlynes_ | brif8: I've just got playback in there for mine, because my recording is very short |
| 19:11.15 | brif8 | well with background() then set(timeout)=response=13 after the message plays it hangs up |
| 19:11.57 | dlynes_ | brif8: you might need to have a wait(3) or something in between them, then |
| 19:11.57 | dlynes_ | brif8: one second |
| 19:12.41 | dlynes_ | brif8: ah...it's either response timeout, or background, but not both |
| 19:13.02 | gmfm | put WaitExten(13) or such after your Background() |
| 19:14.05 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 19:14.14 | CoffeeIV_ | is there any documentation on the dialplan "switch => " thing ? I am trying to use it to go between two * servers using IAX2, and all I have found is one example on voip-info |
| 19:15.25 | *** join/#asterisk sixninem (n=bmuita@212.49.87.220) |
| 19:16.22 | Kyler | dlynes_: You suggested T.38. Are there providers who speak T.38 on one side (to me) and POTS fax on the other? |
| 19:16.59 | dlynes_ | Kyler: they all do |
| 19:17.16 | dlynes_ | Kyler: otherwise, there's no possible way they could talk to a normal fax machine |
| 19:17.39 | Kyler | What's an example of a provider? I thought I found one on voip-info awhile ago but I don't see it now. |
| 19:17.41 | brif8 | gmfm: ok |
| 19:18.53 | dlynes_ | Kyler: no idea...but then again, i haven't looked terribly hard, either |
| 19:19.13 | dlynes_ | Kyler: You can try asking on here, or on asterisk-dev...i know some of the people on here do termination |
| 19:21.29 | *** join/#asterisk bufh (n=user@ip216-239-71-75.vif.net) |
| 19:21.30 | gmfm | gafachi does t.38 termination/origination for the US... I can't attest to the reliability or usability, but they certainly claim to do it. |
| 19:21.39 | bufh | hello dudes |
| 19:21.49 | carrar | where's my car |
| 19:21.55 | b00mer_ | sweet |
| 19:21.59 | dlynes_ | i stole it |
| 19:22.52 | Kyler | gmfm: Ah! Yes, of course. That's what I looked into before. (I already have a Gafachi account...somewhere...) Thanks. |
| 19:22.57 | stack_ | I'm getting warnings on the console when some people check their voicemail, they go to listen to a message and the line is hung up. The console reads "app_voicemail.c:3803 play_message: No message attribute file?!!" This has happened 4 times in the past 2 days |
| 19:23.41 | dlynes_ | stack_: the .txt file is missing for it |
| 19:23.49 | dlynes_ | stack_: Which version of asterisk are you using? |
| 19:23.59 | stack_ | dlynes_: 1.2.7.1 |
| 19:24.12 | dlynes_ | stack_: and it's only happening to maybe one user? |
| 19:24.43 | stack_ | dylnes: two of twelve |
| 19:25.14 | dlynes_ | stack_: What's probably causing is that someone was logged into their voicemail, deleting voicemail, when someone called and left a new voicemail |
| 19:25.24 | dlynes_ | s/causing/causing that |
| 19:25.57 | dlynes_ | stack_: It's a known issue, but one that hasn't been resolved yet |
| 19:26.10 | stack_ | dlynes_, I know that wasn't the case the first time it happened: the user had only voicemail from ~9AM and was checking it at ~11AM with no incoming calls |
| 19:26.18 | *** join/#asterisk lzhang (n=rjrae@67.95.13.46) |
| 19:26.37 | dlynes_ | stack_: Then, what I would do, is file a new bug report with as much information as you can possibly give the developers |
| 19:26.48 | lzhang | can I do agents.conf realtime? I am already using a queue_member_table, but I want to store just the agents in the db also |
| 19:26.49 | dlynes_ | stack_: but make sure there isn't already an existing bug report that's the same as yours |
| 19:26.53 | stack_ | dlynes_: k thanks |
| 19:27.07 | dlynes_ | stack_: They're trying to get all those kinds of bugs squashed before the 1.4 release |
| 19:27.22 | dlynes_ | stack_: the total bug count last time I checked was down to 270 |
| 19:27.36 | dlynes_ | stack_: they're attacking the bugs pretty aggressively lately |
| 19:27.38 | [TK]D-Fender | only 269 fatal!~ |
| 19:27.43 | stack_ | dlynes_: yikes... |
| 19:28.13 | dlynes_ | stack_: 270 is not many, considering how much code there is in asterisk, and considering how complicated the code is |
| 19:28.46 | stack_ | dlynes_: how many developers are working on asterisk? |
| 19:28.47 | dlynes_ | I'm sure Windows has exponentially more than that |
| 19:28.54 | dlynes_ | stack_: don't know offhand |
| 19:29.10 | dlynes_ | stack_: but i do know digium has been going gangbusters hiring asterisk developers lately |
| 19:29.51 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 19:30.09 | Kyler | Gafachi says "Upon detecting a fax tone, our network will send a re-INVITE to attempt to switch to T.38." So then I need t38modem and I can teach Asterisk how to speak H.323 to it? |
| 19:30.37 | asterboy | way too complicated. |
| 19:30.47 | asterboy | just setup Hylafax and be done with it |
| 19:31.20 | asterboy | I have used it successfully using a plain old modem and mgcp or ulaw |
| 19:32.07 | stack_ | iaxmodem works great with Hylafax + Asterisk too |
| 19:32.45 | Kyler | stack_: Will iaxmodem reliably handle >9600 baud fax? |
| 19:33.21 | stack_ | Kyler, I didn't go into that much testing, I just hit it with a bunch of faxes from multiple places... worked great |
| 19:33.38 | tzanger | stack_: I gotta get that set up again |
| 19:33.53 | Kyler | stack_: I just need to send. I looked at iaxmodem awhile ago and didn't get to the point of installing it. I'll look again. Thanks. |
| 19:34.11 | *** join/#asterisk adker (n=adker@67-136-210-63.dsl1.glv.ny.frontiernet.net) |
| 19:34.51 | Kyler | Oh...iaxmodem just uses SpanDSP. Same limitations. |
| 19:34.53 | stack_ | I have outbound and inbound working flawlessly |
| 19:35.09 | Kyler | Looks like I'm left to T.38. That seems cleaner anyway. |
| 19:35.58 | asterboy | no, cleaner is just leaving the fax as is and running a voip line to the modem |
| 19:37.25 | CoffeeIV_ | I have two * machines connected by IAX2. I did "iax2 debug" on both, and I can see that a call going between them has the right "calling number" set on the receiving *, but the ${CALLERID} is not set on the receiving side. Any ideas ? |
| 19:37.56 | CoffeeIV_ | I meant ${CALLERIDNUM} in the above |
| 19:38.33 | stack_ | Is there a way to enable the '*1' automon and a call that enters a Queue? |
| 19:38.42 | Kyler | CoffeeIV_: I was having trouble with ${CALLERIDNUM} recently and saw that it's deprecated. Does "${CALLERID(num)}" work? |
| 19:40.42 | blitzrage | stack_: try the patch here -- http://bugs.digium.com/view.php?id=7125 and see if that helps |
| 19:41.14 | stack_ | blitzrage: I'm looking at that right now |
| 19:42.11 | blitzrage | stack_: see if you can reproduce it on a test machine, then we can acknowledge the error -- then try running with that patch for a few days and if it helps resolve the problem |
| 19:42.49 | stack_ | blitzrage: I'm working on setting up some tests right now |
| 19:42.56 | blitzrage | bweschke and vechers have been doing a lot of work on the bug tarcker |
| 19:43.00 | blitzrage | stack_: excellent |
| 19:43.10 | dlynes_ | Kyler: yes, it does |
| 19:43.13 | blitzrage | let me know if you need any help -- I've been dong a lot of testing of rthe devs lately |
| 19:43.15 | nain | Any Suggestion on changing Frame Rate in asterisk ???? |
| 19:43.39 | blitzrage | CoffeeIV_: ${CALLERID(number)} I'm pretty sure |
| 19:44.19 | blitzrage | oh -- docs say num -- but I think its number.... if I remember from my testing |
| 19:44.21 | Kyler | dlynes_: I know it works for me. It's not clear that's the problem CoffeeIV_ is having though - depends on the Asterisk version. |
| 19:44.30 | blitzrage | I should just look in the code and know for sure :) |
| 19:44.34 | CoffeeIV_ | Kyler: I switched to using ${CALLERID(num)} but it's still blank |
| 19:44.43 | CoffeeIV_ | trying ${CALLERID(number)} |
| 19:44.59 | blitzrage | CoffeeIV_: let me know if that works for you |
| 19:45.21 | dlynes_ | What's CALLERID(number)? |
| 19:45.32 | dlynes_ | Never seen that before |
| 19:45.57 | blitzrage | dlynes_: number == num -- but I think only one of them works |
| 19:46.10 | dlynes_ | Yeah...num does |
| 19:46.13 | stack_ | blitzrage: I was just able to reproduce the voicemail problem, ill try applying the patch tonight and go from there |
| 19:46.31 | blitzrage | stack_: coolio -- post a note stating that you could reproduce it |
| 19:47.09 | dlynes_ | blitzrage: so was that related to that other race condition for the voicemail you guys were talking about the other dya, then? |
| 19:47.22 | blitzrage | dlynes_: not sure yet |
| 19:47.27 | dlynes_ | ah |
| 19:47.28 | blitzrage | dlynes_: sounds similar though |
| 19:47.43 | dlynes_ | Yeah, but apparently this wasn't a race condition |
| 19:48.00 | blitzrage | yah -- maybe I forget what race condition you mean |
| 19:48.23 | dlynes_ | The one where someone would leave a voicemail at the same time as the voicemail owner was deleting a voicemail |
| 19:48.32 | dlynes_ | and then the voicemail .txt file would not be there |
| 19:48.41 | blitzrage | dlynes_: that sounds like the bug we were just talking about :) |
| 19:49.03 | dlynes_ | blitzrage: yeah, but he insisted that one of those calls nobody was logged in at the same time |
| 19:49.15 | blitzrage | not sure then :) |
| 19:49.39 | blitzrage | } else if (!strncasecmp("num", data, 3) || !strncasecmp("number", data, 6)) { |
| 19:49.41 | *** join/#asterisk bufh (n=user@ip216-239-71-75.vif.net) |
| 19:49.44 | bufh | back :) |
| 19:49.51 | blitzrage | so either num or number work equally in the CALLERID() function |
| 19:50.02 | dlynes_ | heh |
| 19:50.23 | blitzrage | if your callerID isn't working in sip, make sure you have sendrpid=yes so that Remote-Party ID gets sent |
| 19:50.38 | blitzrage | or you'll more than likely end up with the username as the callerid number |
| 19:50.43 | CoffeeIV_ | the UPGRADE.txt in my asterisk source says that CALLERIDNUM and CALLERIDNAME have "been deprecated in favor of their related dialplan functions". So what is the related dialplan function ? show applications doesn't list anything related |
| 19:51.08 | blitzrage | CoffeeIV_: dialplan functions are not dialplan applications -- show functions |
| 19:51.24 | blitzrage | and the related dialplan function is CALLERID(name) and CALLERID(num) or (number) |
| 19:51.25 | bufh | <PROTECTED> |
| 19:51.38 | blitzrage | example: Set(CALLERID(name)=blitzrage) |
| 19:51.51 | blitzrage | to read: NoOp(${CALLERID(name)}) |
| 19:52.03 | blitzrage | the first example is a write |
| 19:52.19 | esculapio_ | quien habla espanol |
| 19:52.21 | esculapio_ | ? |
| 19:53.41 | b00mer_ | can you send callerid info to Vonage? will they use it? |
| 19:53.56 | stack_ | blitzrage: my initial results are posted to the bug tracker... I'll have more tonight |
| 19:54.48 | [TK]D-Fender | blitzrage: ! ! ! |
| 19:55.04 | mercestes | Bufh: yea, ** should read fine in the extensions.conf. By should, I mean "try it and see." of course. But I'm suggesting yes. |
| 19:56.26 | dlynes_ | b00mer_: why would they? |
| 19:56.27 | blitzrage | b00mer_: I doubt it |
| 19:56.38 | esculapio_ | hola quien habla espanol?\ |
| 19:56.59 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 19:57.04 | dlynes_ | esculapio_: tues speaka da anglais? |
| 19:57.23 | Strom_C | dlynes_: wow, you butchered that |
| 19:57.33 | dlynes_ | Strom_C: I don't know spanish :) |
| 19:57.58 | dlynes_ | Strom_C: so i figured some spanish, some canadianized italian, and some french would work good :) |
| 19:57.58 | Strom_C | dlynes, repeat after me: "hablas ingles?" |
| 19:58.07 | blitzrage | [TK]D-Fender: I don't want meet your mom! |
| 19:58.26 | [TK]D-Fender | esculapio_ has a probably interesting question if he could only overcome the language barrier |
| 19:58.40 | fjean | hey, I speak portuguese too ;-) |
| 19:58.40 | bufh | mercestes < i tried and it didn't work, that's why i was asking my question :P |
| 19:58.56 | dlynes_ | [TK]D-Fender: He usually talks to salviadud, or asterboy i think |
| 19:59.10 | bufh | (don't know if i'm doing something wrong or if it's not possible) |
| 19:59.10 | dlynes_ | but i guess asterboy's afk atm |
| 19:59.16 | Strom_C | esculapio_: hablo espanol, pero mi espanol es tan pobre...hablame en #asterisk-es si quieres |
| 20:00.05 | dlynes_ | damn...spoken like a professional :) |
| 20:00.26 | dlynes_ | I only know french and mandarin, unfortunately |
| 20:00.35 | dlynes_ | Not much call to use spanish in Canada |
| 20:01.06 | fjean | esculapio: yo entendio espanhol...pero lo no hablo, hablo portugues |
| 20:01.12 | fjean | :-) |
| 20:04.28 | dlynes_ | wow..i guess there's quite a few channels that cater to spanish |
| 20:06.57 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:09.38 | dlynes_ | [TK]D-Fender: btw...they've got the ec for 1, 2, and 4 port cards |
| 20:10.06 | dlynes_ | [TK]D-Fender: that's as of october of last year |
| 20:10.26 | dlynes_ | [TK]D-Fender: this june, they'll have an 8 port with ec as well |
| 20:10.32 | stack_ | The Dial command has 'w' & 'W' for recording... does Queue have something similar? |
| 20:10.48 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 20:15.18 | *** join/#asterisk batphone (n=will@69.15.174.114) |
| 20:15.34 | batphone | whats up with zttest |
| 20:15.43 | batphone | is there a mobo that digium recommends???? |
| 20:16.18 | batphone | this is a serious headache, ive tried several mobos.. |
| 20:16.20 | dlynes_ | batphone: make sure your cards aren't sharing interrupts |
| 20:16.23 | [TK]D-Fender | dlynes : Ummm.. no. Link it. |
| 20:16.29 | batphone | http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 20:16.31 | batphone | im reading this |
| 20:16.34 | batphone | they are not |
| 20:16.59 | dlynes_ | [TK]D-Fender: Doug Vilim emailed me that info |
| 20:17.00 | batphone | zttest, after about a minute, drops from a nearly consistent 100% to 99.90.. |
| 20:17.16 | batphone | this is with a PDSMA motherboard |
| 20:17.20 | [TK]D-Fender | dlynes : Show me a product I can BUY <- |
| 20:17.38 | [TK]D-Fender | dlynes : I know they've been developing them, but nothing RELEASED |
| 20:17.44 | dlynes_ | [TK]D-Fender: I don't work at Sangoma :) |
| 20:17.51 | *** join/#asterisk AsteriskAlbania (i=Asterisk@80.91.113.253) |
| 20:18.07 | [TK]D-Fender | dlynes : You don't have to to show me that they offer a product no-one else can corroborate :) |
| 20:18.17 | dlynes_ | [TK]D-Fender: It said on October 2005, they releasED an ec for 1, 2, and 4 port |
| 20:18.18 | russellb | batphone: are you actually having problems? |
| 20:18.30 | dlynes_ | [TK]D-Fender: as in past tense |
| 20:18.47 | [TK]D-Fender | dlynes : What said this? |
| 20:18.56 | dlynes_ | [TK]D-Fender: the email that I got from Doug |
| 20:19.07 | AsteriskAlbania | I am trying asterisk for call back service using MCC , I am able to make it if one channel is SIP and the other Mobile or if both numbers are sip, if I try to do it with to mob phones it fails, any idea ? |
| 20:19.23 | *** join/#asterisk Souvent22 (n=chatzill@151.200.137.138) |
| 20:19.29 | [TK]D-Fender | dlynes : Thats nice... and now for something TANGABLE? |
| 20:19.33 | Kyler | I just called Gafachi and was assured that even though their T.38 support is "beta" it's likely to be around awhile and "progress in a positive way." |
| 20:19.36 | batphone | russellb: yes |
| 20:19.38 | dlynes_ | [TK]D-Fender: lol...so it's just talk then? |
| 20:19.45 | batphone | russellb: like hdlc erros, faxing just wont work, etc |
| 20:19.54 | russellb | batphone: have you contacted Digium technical support? |
| 20:19.54 | dlynes_ | [TK]D-Fender: it seemed like it was a letter he composed, not a form letter |
| 20:20.02 | batphone | running hdparm and zttest at the same time causes big problems |
| 20:20.11 | Kyler | "TANGABLE"? Like astronauts? |
| 20:20.15 | batphone | yes, they have no answers, they tell me to try another motherboard |
| 20:20.22 | batphone | id like to know what mobo they use |
| 20:20.24 | [TK]D-Fender | dlynes : They aren't listed on their site, you can't buy or see them anywhere.... you're talking vaoporware ATM... |
| 20:20.32 | [TK]D-Fender | Kyler : typo. |
| 20:20.40 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 20:21.00 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 20:23.17 | batphone | i have a te110p and 1 fxs card in the same box |
| 20:23.30 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:24.50 | batphone | http://pastebin.com/710176 |
| 20:25.09 | batphone | after the 99.90.. shows up it never returns any hirgher |
| 20:25.30 | *** part/#asterisk sixninem (n=bmuita@212.49.87.220) |
| 20:26.51 | batphone | see how the rtc has no interrupts? |
| 20:26.58 | batphone | and there is 1 ERR listed |
| 20:27.13 | CoffeeIV_ | The CALLERID(num) thing didn't work for me when I used it in parentheses like you guys said, I had to do it is curly brackets like ${CALLERID(num)} -- just and FYI ( and thanks for all the help ! ) |
| 20:28.01 | [TK]D-Fender | batphone : What exactly is the problem? Looks OK to me... |
| 20:28.10 | batphone | the zttest lines |
| 20:28.12 | batphone | 99.90?? |
| 20:28.14 | batphone | this is bad |
| 20:28.20 | batphone | this wont even generate a fax document |
| 20:28.21 | [TK]D-Fender | batphone : is it? |
| 20:28.39 | batphone | http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 20:28.43 | batphone | this page explains alot of it |
| 20:28.54 | batphone | but the end result is a barely functional pbx |
| 20:29.15 | batphone | useless for anything more than a sip proxy |
| 20:29.24 | batphone | ..so to speak |
| 20:29.49 | [TK]D-Fender | batphone : Just not FAX. Basically Digium disavows the EXISTANCE of faxes and will never claim support in any capacity |
| 20:30.24 | batphone | and software HDLC? this is the underlying technology behind digiums carrier card line |
| 20:30.27 | Kyler | Are there any "howto" type documents for configuring t38modem with Asterisk? I'm new to T.38 and I've barely used H.323 (but never with Asterisk). |
| 20:31.07 | russellb | [TK]D-Fender: how about the fact that T.38 passthrough support is getting merged for 1.4? |
| 20:31.27 | [TK]D-Fender | russellb : I'll wait for the press release :) |
| 20:31.43 | russellb | batphone: and actually, the TE4XXP and TE2XXP line can do HDLC on the board |
| 20:31.45 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:32.24 | batphone | i only need the single port card |
| 20:32.29 | russellb | ok. |
| 20:33.01 | [TK]D-Fender | ok, I'm outta here.. later all |
| 20:33.29 | russellb | so you shouldn't be worried about doing framing for a single T1 ... |
| 20:34.20 | *** join/#asterisk boch (n=boch@unirc.com.ar) |
| 20:34.32 | batphone | what is this HPET kernel option in relation to zaptel? |
| 20:34.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:37.22 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 20:38.09 | boch | what is the $var to set the max call duration ? |
| 20:40.10 | sevard | does anyone use the queue-out feature? |
| 20:44.16 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
| 20:49.19 | *** join/#asterisk mog_home (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 20:49.25 | batphone | migraine |
| 20:49.57 | Strom_C | dead hookers |
| 20:50.32 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:50.45 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 20:51.00 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:00.01 | Hmmhesays | indeed |
| 21:00.18 | batphone | what does the Timer Frequency option in the kernel config have to do with all this? |
| 21:00.19 | *** join/#asterisk [Outcast] (n=outcast@219-89-206-239.adsl.xtra.co.nz) |
| 21:00.24 | batphone | i have three options there, 100,200 and 1000 |
| 21:00.52 | dlynes_ | I'm curious |
| 21:01.07 | dlynes_ | Why is it that when I make a call out of a pri, it doesn't show up in zap show channels? |
| 21:01.27 | *** join/#asterisk spinzero (n=spinzero@ip67-88-113-150.z113-88-67.customer.algx.net) |
| 21:01.33 | dlynes_ | Only inbound connections show up on zap show channels; not outbound |
| 21:01.41 | spinzero | Any suggestions on a softphone to use with asterisk? |
| 21:01.49 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 21:01.52 | dlynes_ | spinzero: windows, or linux? |
| 21:02.26 | spinzero | linux |
| 21:03.00 | dlynes_ | sjphone (does iax2), ekiga (does sip and h323, i think)...formerly known as gnophone |
| 21:03.36 | spinzero | ok. have ekiga already installed. |
| 21:04.10 | *** join/#asterisk kavit (n=kavit@203-158-40-84.dyn.iinet.net.au) |
| 21:04.17 | spinzero | dlynes: Thanks. |
| 21:04.53 | spinzero | What are opinions of the x-10 softphone software? Associate here is asking me to ask. |
| 21:06.28 | dlynes_ | spinzero: most peeps i've seen on here using windows are usually using x-ten, or the commercial counterpart |
| 21:06.46 | dlynes_ | spinzero: the snom320 softphone's not too bad for windows, either...it's super easy to set up |
| 21:06.59 | dlynes_ | or was it snom360? I can't remember now |
| 21:07.33 | dlynes_ | www.snom.de, at any rate...they've only got one softphone |
| 21:07.55 | dlynes_ | it's identical in function and user interface, and all that kinda stuff to their hardphone |
| 21:08.19 | dlynes_ | the free version however, doesn't include g729 codec |
| 21:08.26 | Altair256 | spinzero where did you get ekiga for windows? |
| 21:08.37 | dlynes_ | Altair256: he's using it on linux, not windows |
| 21:08.43 | Altair256 | ah, sorry, misunderstood |
| 21:08.58 | dlynes_ | Altair256: his associate wants to use x-ten on windows |
| 21:09.01 | Altair256 | I was gonna say... didn't know that ekiga had any releases for the Windows platform yet |
| 21:09.19 | Altair256 | xten is fine, but the free version only supports 3 lines |
| 21:09.22 | dlynes_ | Altair256: yeah...i think a windows version is in the works, or the planning stages or something like that |
| 21:09.42 | dlynes_ | Altair256: How many mouths do you have attached to your head? |
| 21:09.43 | Altair256 | dlynes, I thought the same thing, but they've been saying that for a while |
| 21:09.58 | boch | what is the $var to set the max call duration ? |
| 21:10.00 | dlynes_ | Altair256: I only have one, so therefore I can only talk on one line at a time :) |
| 21:10.12 | dlynes_ | Altair256: you can use call parking if you have a lot of lines |
| 21:10.21 | Altair256 | lol |
| 21:10.23 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:10.42 | Altair256 | we're deploying some SPA-941's at work... only has 4 lines... |
| 21:10.48 | Altair256 | I used the same argument |
| 21:10.49 | dlynes_ | Altair256: doesn't x-ten support 3 line appearances, not 3 actual lines? |
| 21:11.02 | Altair256 | correct, 3 line appearances |
| 21:11.09 | dlynes_ | and not shared line appearances, right? |
| 21:11.34 | Altair256 | not totally sure, but I believe you are correct |
| 21:11.44 | dlynes_ | so, first call will come in on the first line |
| 21:11.47 | Altair256 | I personally don't believe in multiple registrations at a single device |
| 21:11.50 | dlynes_ | next call will come in on the second line and so forth |
| 21:12.06 | dlynes_ | iow, you can't pick a line for it to call out on, and it can't pick a line to call in on |
| 21:12.14 | dlynes_ | it just takes the first available one |
| 21:12.16 | Altair256 | if that's what you need, then switch to separate registrations for your users and devices |
| 21:12.33 | Altair256 | hrm, I don't think so dlynes ... |
| 21:12.44 | Altair256 | I had one registered with FWD and with an asterisk server... |
| 21:12.48 | dlynes_ | that's why i'm saying more than two lines, and the person must be a super sales person or something |
| 21:12.51 | Altair256 | I think you can pick |
| 21:13.00 | sevard | oh god |
| 21:13.02 | sevard | wouldn't exten = s,3,VoicemailMain(company_x) |
| 21:13.15 | sevard | send me to context company_x in voicemail.conf ? |
| 21:13.33 | Altair256 | exten=> |
| 21:13.37 | Altair256 | not exten= |
| 21:13.43 | sevard | => or =, it doesn't matter |
| 21:13.46 | dlynes_ | sevard: =/=> are same thing |
| 21:13.50 | sevard | that should be in the *@&$ing wiki |
| 21:13.50 | Altair256 | ah, my bad |
| 21:13.51 | dlynes_ | erm altair i mean |
| 21:13.55 | boch | how can i limit all calls duration to 60 minutes? |
| 21:14.06 | sevard | it's sending me to voicemail context default |
| 21:14.07 | dlynes_ | sevard: Voicemailmain(@company_x) |
| 21:14.10 | sevard | instead of company_x |
| 21:14.27 | dlynes_ | sevard: you have to have the '@' in front of it to specify a context |
| 21:14.44 | sevard | BAH |
| 21:14.46 | dlynes_ | sevard: otherwise you're trying to go to the company_x voicemailbox in the the default context |
| 21:14.59 | sevard | thank you dlynes_ |
| 21:15.15 | dlynes_ | sevard: So voicemailmain(mymailbox@contextname) |
| 21:15.29 | dlynes_ | sevard: a word of warning |
| 21:15.31 | sevard | once again i love you |
| 21:15.35 | pigpen | Hi all, I have a customer that thinks that FreePBX is the bomb, but he is going to need to be able to do allot of customizations...I know what I think, could I get your opinion? |
| 21:15.37 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:15.46 | dlynes_ | sevard: in my experience, I haven't found contexts to work reliably in voicemail |
| 21:15.58 | sevard | dlynes_: why do you say this |
| 21:16.00 | pigpen | It is a School with about 100 extensions |
| 21:16.29 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 21:16.29 | sevard | pigpen: tell him that your customizations are only available without freepbx and to shove it |
| 21:16.34 | dlynes_ | sevard: at least not when you want to have it jump into a particular mailbox, and prompot for the passowrd, and not the mailbox for admin, or just drop into leaving a message, and not prompting for the voicemailbox |
| 21:16.52 | pigpen | sevard, thanks...that is what I think. |
| 21:16.57 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 21:17.17 | sevard | dlynes_: have you had the misfortune of looking at the source code |
| 21:17.24 | dlynes_ | sevard: if you don't mind getting callers to specify the mailbox number, it'll work though |
| 21:17.38 | boch | guys, is it possible to limit the call duration? |
| 21:17.44 | markit | hi, I've errors compiling zaptel-1.2, it starts with something like "cc1: error: include/linux/autoconf.h: No such file or directory", any idea? libpri and asterisk-1.2 compile just fine instaed |
| 21:17.46 | dlynes_ | sevard: heh...misfortune is a very good description of app_voicemail.c :) |
| 21:18.07 | sevard | dlynes_: did somebody write that with a crayon :s |
| 21:18.12 | dlynes_ | sevard: not to mention app_dial.c :) |
| 21:18.13 | markit | (using asterisk 1.2 svn, kernel 2.6.15, debian) |
| 21:18.14 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 21:18.34 | sevard | dlynes_: rewrite app_voicemail.c :| |
| 21:18.47 | dlynes_ | sevard: yeah...i've been thinking about it, actually |
| 21:18.52 | dlynes_ | sevard: that and app_dial.c |
| 21:18.58 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 21:18.59 | sevard | i know a lot of people here would support you |
| 21:19.23 | dlynes_ | sevard: break up the functions so that they're not 10 pages long, and get rid of all the probably mostly unnecessary goto's |
| 21:19.56 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 21:20.18 | dlynes_ | zwelch: lol |
| 21:20.33 | Dr-Linux | Hi all |
| 21:20.34 | sevard | dlynes_: my roommate says "have a script system with a listen for keypress and collect into one variable" |
| 21:20.42 | *** join/#asterisk kimosabe (n=kimosabe@dsl-200-78-71-61.prod-infinitum.com.mx) |
| 21:21.03 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 21:21.06 | kimosabe | im using xlite and it says registered but when i dial it says call not aproved can some one help me ? |
| 21:21.09 | *** join/#asterisk kiwi_uk (n=jon@kadus.kiwiuk.net) |
| 21:21.18 | Dr-Linux | my roomate is web designer, not asterisk guru :( |
| 21:21.22 | dlynes_ | With function code that many pages long, you couldn't reasonably expect anyone to debug any problems properly, or within a decent amount of time |
| 21:21.40 | Jaxxan | what program does the voicemail module use to send email ? |
| 21:21.45 | Jaxxan | is it just "mail"? |
| 21:21.47 | dlynes_ | Jaxxan: sendmail |
| 21:21.58 | sevard | probably sendmail |
| 21:22.03 | sevard | another horrible thing about voicemail |
| 21:22.15 | Dr-Linux | Jaxxan: bydefault sendmail, but you can use any other MTA |
| 21:22.22 | dlynes_ | sevard: yeah...should just use its own socket code, instead of calling up sendmail |
| 21:22.30 | Jaxxan | i'm talking about client, not MTA |
| 21:22.33 | Jaxxan | i'm using postfix as MTA |
| 21:22.41 | sevard | dlynes_: hop to it you magic gnome |
| 21:22.45 | dlynes_ | Jaxxan: sendmail is both an mua and an mta |
| 21:22.46 | Jaxxan | i wanna change the return-path alias |
| 21:23.01 | pauldy | well broadvoice has officially hit the shitlist for crappiest US based voip provider |
| 21:23.04 | Dr-Linux | Jaxxan: change it in voicemai.conf |
| 21:23.06 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 21:23.15 | sevard | Jaxxan: you can change it in the conf file |
| 21:23.50 | dlynes_ | pauldy: i thought that title was owned by nufone? |
| 21:23.59 | Jaxxan | problem is i need to change the return-path, not the mime headers |
| 21:24.06 | Dr-Linux | Jaxxan: what's the distro? |
| 21:24.09 | kimosabe | can some one help me x-lite please |
| 21:24.10 | dlynes_ | pauldy: judging by all the bitching and whining and complaining i hear on here regularly about nufone, anyways |
| 21:24.16 | pauldy | dlynes: I guess BV decided they could out do them |
| 21:24.17 | Jaxxan | redhat RHEL4 |
| 21:24.21 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
| 21:24.45 | Dr-Linux | Jaxxan: then just use sendmail, i suggest |
| 21:24.58 | Dr-Linux | Jaxxan: i'm using RHEL3 AS |
| 21:26.02 | *** join/#asterisk mischko (i=Scott@207-118-73-11.dyn.centurytel.net) |
| 21:26.40 | mischko | Anyone here using VOIP over a wireless 5.8Ghz WAN connection? |
| 21:26.49 | pauldy | dlynes: check out broadvoices history on the www.bbb.org |
| 21:27.00 | dlynes_ | pauldy: i'd just be wasting my time |
| 21:27.04 | dlynes_ | pauldy: I don't live in the US |
| 21:27.19 | dlynes_ | pauldy: besides...i buy wholesale |
| 21:28.42 | kimosabe | mischko yes im using voice over wireless |
| 21:29.27 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 21:29.32 | mischko | kimosabe, I'm considering solutions. It appears that I can get 1.5Mb/s for $60/month, business grade with SLA's. |
| 21:30.24 | mischko | kimosabe, are you running asterisk in-house? |
| 21:30.37 | kimosabe | yes |
| 21:30.59 | kimosabe | where can u get 1.5 megabits bandwith for 60 a month |
| 21:31.08 | mischko | Corvallis Oregon |
| 21:31.33 | mikefoo | mischko: where? |
| 21:31.42 | mikefoo | what provider? |
| 21:31.49 | mischko | PEAK (www.peak.org) |
| 21:31.50 | mikefoo | dsl? |
| 21:31.55 | mikefoo | t1? |
| 21:32.30 | mischko | wireless. I don't believe it's "DSL" or "T1", just straight wireless to the ISP from my location. Mount an antenna on the roof. |
| 21:32.43 | mikefoo | heh yeah? |
| 21:32.49 | Jaxxan | i hate sendmail... postfix ftw |
| 21:32.59 | mischko | postfix rocks. |
| 21:33.06 | Hmmhesays | i hate this fucking company |
| 21:33.19 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:33.21 | Jaxxan | how does the voicemail application generate the email ? |
| 21:33.22 | mischko | Wireless costs a few hundred for the hardware or you add about $15/month for a lease. |
| 21:33.30 | brookshire | mischko: make sure it's 802.16 and not 802.11 |
| 21:33.39 | brookshire | if it is wireless |
| 21:33.54 | mikefoo | wireless huh? |
| 21:33.59 | mikefoo | thats interesting.. |
| 21:34.06 | brookshire | wait.. if it is 5.8Ghz it's 802.16 |
| 21:34.12 | brookshire | and it runs just fine |
| 21:34.27 | mikefoo | 802.16 overkill for home use? |
| 21:34.31 | dlynes_ | oh great |
| 21:34.34 | brookshire | 802.16 is point to point |
| 21:34.37 | dlynes_ | pwlib uses x11 |
| 21:34.38 | mikefoo | ahh.. |
| 21:34.39 | brookshire | but.. wireless is unstable |
| 21:34.44 | brookshire | at times |
| 21:35.05 | dlynes_ | hopefully it doesn't need the x11 libs :0 |
| 21:35.18 | brookshire | if you don't mind occasional outages.. wireless is alright |
| 21:35.42 | mischko | These guys claim to provide SLA's for that price. |
| 21:35.58 | brookshire | well.. what does the SLA say? |
| 21:36.04 | *** join/#asterisk Faithful (n=Faithful@202.6.145.116) |
| 21:36.10 | brookshire | because 24hr response SLA isn't worth much |
| 21:36.30 | mischko | I haven't seen the SLA's yet. What should I look for? |
| 21:36.42 | brookshire | uptime.. |
| 21:36.48 | brookshire | a SLA can be anything |
| 21:37.02 | brookshire | like.. if you have a problem we fix it within 24hrs... etc |
| 21:37.06 | Nivex | "We agree not to poop on your servers." |
| 21:37.11 | brookshire | or.. 99.99999% uptime |
| 21:38.58 | brookshire | a SLA basically states the minimum amount of service they have to provide.. if you think about it |
| 21:39.14 | Strom_C | "We will fix your problem within five business days or your next pizza is free" |
| 21:39.50 | brookshire | SLAs are rarely good for the customer, imho |
| 21:40.01 | mischko | Thanks for the tips. I'll review the SLA carefully. |
| 21:42.21 | brookshire | i have some pics of my 802.16 setup :) |
| 21:42.22 | brookshire | http://glowstick.org/radios/ |
| 21:42.30 | Jaxxan | does procmail do outbound email ? |
| 21:42.32 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 21:43.29 | brookshire | jaxxan: i think it can be setup to do that |
| 21:44.12 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 21:45.04 | sevard | dlynes_: |
| 21:45.21 | dlynes_ | that's my name, don't wear it out |
| 21:45.23 | sevard | check this on the street brotha |
| 21:45.39 | sevard | i thought in sip.conf voicemail = extno@context was supposed to work |
| 21:46.04 | dlynes_ | sevard: don't know...haven't tried that...one sec |
| 21:46.37 | Strom_C | maibox=whatever |
| 21:46.40 | Strom_C | not voicemail= |
| 21:46.47 | dlynes_ | heh...missed htat |
| 21:46.54 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:47.23 | Strom_C | http://www.stromcarlson.com/misc/myspace.png |
| 21:47.26 | dlynes_ | Strom_C: btw...is there a way to change the default mailbox extension? |
| 21:47.40 | Strom_C | ? |
| 21:47.43 | Strom_C | what do you mean |
| 21:47.46 | brookshire | dlynes: that's all dialplan |
| 21:47.48 | dlynes_ | Strom_C: i.e. the default mailbox extension is 'asterisk' |
| 21:48.13 | dlynes_ | so when you hit the voicemail button on a phone that has such a thing, it tries to dial the extension 'asterisk' |
| 21:48.28 | Strom_C | dlynes_: uh, that depends on the phone, doesnt it? |
| 21:48.34 | dlynes_ | Strom_C: nope |
| 21:48.42 | dlynes_ | Strom_C: that's an asterisk configuration |
| 21:48.44 | Strom_C | my phones are all configured to dial *95 for voicemail :) |
| 21:48.48 | dlynes_ | Strom_C: one sec, i'll show you where it is |
| 21:49.00 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 21:49.34 | tasat | any app_conference users around? |
| 21:49.42 | b00mer_ | when the console says, " Requested transfer capability: 0x00 - SPEECH" What is SPEECH? other than the obvious meaning |
| 21:50.16 | dlynes_ | Strom_C: sip show settings: Voice Mail Extension: asterisk |
| 21:50.58 | dlynes_ | Strom_C: I've only got one phone that even supports that feature though |
| 21:51.18 | dlynes_ | Strom_C: Aastra and probably Polycom don't support that feature afaik |
| 21:52.19 | markit | any idea about why I get compilation errors with zaptel-1.2, while libpri and asterisk compile just fine? kernel 2.6.15 |
| 21:52.37 | dlynes_ | markit: do you have udev installed? |
| 21:53.01 | markit | dlynes_ mmm seems not |
| 21:53.14 | dlynes_ | markit: it's kinda necessary for 2.6 :) |
| 21:53.20 | markit | dlynes_ could it be a problem for compilation? |
| 21:53.37 | dlynes_ | markit: certainly a problem for make install |
| 21:53.43 | dlynes_ | markit: don't know about make |
| 21:54.02 | dlynes_ | markit: is your kernel-dev the same version as your currently running kernel? |
| 21:54.08 | markit | let me try again then |
| 21:55.10 | markit | ehm... kernel-dev? I've not such a .deb package installed |
| 21:56.35 | dlynes_ | markit: well, i just called it kernel-dev cause most people are using redhat or some derivative, thereof |
| 21:56.50 | dlynes_ | markit: do your kernel headers match your kernel version? |
| 21:57.02 | markit | dlynes_ yes, they do |
| 21:57.03 | DrPete | Anyone give me an idea of what parition scheme is good for a basic asterisk server, just a few ip phones, 1 x100p etc |
| 21:57.11 | markit | (btw, udev did not fixed the make error) |
| 21:57.31 | dlynes_ | markit: well, like i said...it'd probalby just fix a make install error |
| 21:57.53 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:58.36 | [Outcast] | is it better to run SVN head or the stable? |
| 21:58.37 | markit | dlynes_ this is my /usr/src http://pastebin.ca/54869 |
| 21:59.00 | markit | [Outcast]: define "better" ;) |
| 21:59.43 | dlynes_ | [Outcast]: you mean svn trunk? |
| 22:00.18 | [Outcast] | well, there seems to be bug in stable and there are the svn trunk. When depolying a system how do i decide on which to use? |
| 22:00.23 | [Outcast] | dlynes, yes |
| 22:00.39 | dlynes_ | [Outcast]: use stable, unless you have a compelling reason to use trunk |
| 22:01.03 | dlynes_ | [Outcast]: and even then, i'd only use patches against stable, not replace lock, stock and barrel with trunk |
| 22:01.17 | *** join/#asterisk keyhack (n=keyhack@68.236.93.219) |
| 22:01.22 | markit | dlynes_ : I have this kind of errors http://pastebin.ca/54873 |
| 22:01.33 | keyhack | is there a web GUI to manage Asterisk or something? |
| 22:01.38 | dlynes_ | markit: yeah...errors are more useful than a directory listing |
| 22:01.51 | dlynes_ | keyhack: lots...try checking the main asterisk wiki |
| 22:02.06 | dlynes_ | keyhack: most are commercial, but there's a free one called 'FreePBX' |
| 22:02.21 | [Outcast] | markit, do you have the kernel source on the system? |
| 22:02.24 | dlynes_ | keyhack: we don't discuss FreePBX on here, though...you can ask abou tit on #freepbx |
| 22:02.39 | markit | [Outcast]: sure, have a look at http://pastebin.ca/54869 |
| 22:02.56 | keyhack | dlynes_: ok, cool, thanks. I'm looking to maybe set up different companies on my PBX, have a different PBX for each business I sell services to |
| 22:03.10 | [Outcast] | what distro are you running? |
| 22:03.19 | markit | [Outcast]: debian sid |
| 22:03.23 | markit | (unstable) |
| 22:03.56 | dlynes_ | markit: you'll be wanting to run oej's test-this-branch then, too...so it'll complement the version of debian :) |
| 22:04.06 | [Outcast] | markit, can you give me a unam -a |
| 22:04.31 | [Outcast] | uname -a |
| 22:04.56 | markit | [Outcast]: is in the pastebin, in any case: Linux meucci 2.6.15-1-k7 #2 Mon Mar 6 15:42:39 UTC 2006 i686 GNU/Linux |
| 22:05.09 | markit | dlynes_ what do you mean? |
| 22:05.12 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:05.23 | dlynes_ | oej's test-this-branch is unstable, too :) |
| 22:05.38 | markit | dlynes_ ah, I see, lol |
| 22:05.52 | dlynes_ | extremely unstable ;) |
| 22:07.23 | markit | mm the other debina sid pc compiles just fine, but has 2.6.16 kernel |
| 22:07.35 | markit | this one has 2.6.15 due to the necessity to compile visnd |
| 22:07.42 | dlynes_ | visdn? |
| 22:07.44 | markit | but does not compile zaptel... |
| 22:07.55 | markit | yuep, the new isdn layer |
| 22:08.02 | nextime | hi markit! |
| 22:08.03 | *** join/#asterisk adorah (n=FreePBX4@87.68.158.11.cable.012.net.il) |
| 22:08.08 | markit | http://www.visdn.org/ |
| 22:08.11 | markit | hi nextime !!! |
| 22:08.26 | markit | dlynes_ if you want to try it, use the daily build |
| 22:08.26 | dlynes_ | yeah...not in europe...isdn here is too expensive |
| 22:08.58 | nextime | dlynes_: isdn expansive? where? |
| 22:09.01 | dlynes_ | and looks like pastebin.ca is having problems |
| 22:09.07 | dlynes_ | nextime: Canada |
| 22:10.17 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 22:10.53 | dlynes_ | anyone able to get pwlib for openh323 compiling on slackware? |
| 22:18.47 | jlewis | is there a good way to debug a sip peer that keeps getting 401 Unauthorized to get a better idea of exactly what the problem is? |
| 22:21.07 | x86 | jlewis: sip debug in CLI? |
| 22:21.18 | jlewis | done that |
| 22:26.25 | CoffeeIV_ | Any ideas why this is always going to no-callerid ? GotoIf( $["foo${CALLERIDNUM}" = "foo"]?no-callerid,s,1:5) I checked with NoOP that CALLERIDNUM is set correctly, in fact I set it just before this from ${CALLERID(num)} |
| 22:26.58 | dlynes_ | if the only person that's hearing echo is hearing themselves echo, and the call is pure voip, except for cell phone going into pri, would the problem be the handset? |
| 22:27.16 | dlynes_ | the problem only manifests itself once in a while, and all six phones exhibit this problem |
| 22:28.24 | dlynes_ | CoffeeIV_: got a log you can pastebin? |
| 22:28.46 | dlynes_ | CoffeeIV_: don't bother with all the debug crap...just a normal full verboseness log |
| 22:29.16 | dlynes_ | CoffeeIV_: and why are you still using CALLERIDNUM? Use CALLERID(num) |
| 22:29.41 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 22:30.00 | dlynes_ | CoffeeIV_: also, pastebin that section of your dialplan |
| 22:31.37 | watchy | sup g dogs |
| 22:31.38 | gbodemantv | does anyone know how to give a timelimit for how long a person is in a queue |
| 22:32.07 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 22:32.16 | gbodemantv | I have created a queue for my Corporate inbound lines |
| 22:32.29 | gbodemantv | but I need it to roll to a IVR if no one asnwers |
| 22:32.37 | watchy | anyone here bet on american idol |
| 22:32.55 | nahirean | i bet that it sucks |
| 22:33.02 | watchy | me to |
| 22:33.08 | watchy | but i make a living betting on that show |
| 22:33.35 | RoyK | <PROTECTED> |
| 22:33.52 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 22:33.57 | CoffeeIV_ | dlynes: I tried to switch to CALLERID(num), but that didn't work in that GotoIf, so I set $CALLERIDNUM from the other, and that also doesn't work -- somethings wrong eith the GotoIf (I copied it from an older dialplan) |
| 22:35.40 | watchy | hmm |
| 22:35.52 | watchy | i wonder why betonsports wont let me play my american idol bets |
| 22:37.24 | watchy | anyone here score on the delayhoya fight |
| 22:38.26 | watchy | Error: Cannot place a bet on a line that is in progress. |
| 22:38.31 | watchy | how is that shit in progress |
| 22:38.40 | gandhijee | lol |
| 22:38.56 | watchy | Which Idol will be eliminated on May 10th 2006? (Katharine McPhee) +350 |
| 22:38.56 | watchy | Risk: 25 / Win: 87.5 |
| 22:38.58 | watchy | haha |
| 22:39.18 | Dr-Linux | watchy: are you drunk? |
| 22:39.18 | watchy | im gonna go broke gambling on stupid stuff |
| 22:39.22 | watchy | dr: you watch american idol? |
| 22:39.49 | Dr-Linux | watchy: i'm from pakistan, so don't know american games |
| 22:40.13 | watchy | oh. its a stupid tv show where people sing |
| 22:40.43 | Dr-Linux | watchy: Barselona will looser .. and Arsenol will win |
| 22:40.59 | gandhijee | one of those guys is from Richmond VA |
| 22:41.05 | watchy | what guys |
| 22:41.08 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
| 22:41.16 | gandhijee | from american idol |
| 22:41.23 | watchy | i never seen this seasons |
| 22:41.29 | Dr-Linux | wtf is idol? |
| 22:41.29 | watchy | im just going from inside info on who to bet on |
| 22:41.42 | gandhijee | i don;t watch the mess, i just saw it in the paper |
| 22:41.47 | watchy | i cant stand the show personally |
| 22:41.48 | dlynes_ | Dr-Linux: You've seen Indian Idol on Sony TV, right? |
| 22:42.13 | gandhijee | or Saragama |
| 22:42.14 | Dr-Linux | dlynes_: maybe but don't know what's meaning of idol? |
| 22:42.20 | gandhijee | i know i spelled it wrong |
| 22:42.22 | watchy | idol is something you look up to |
| 22:42.25 | Dr-Linux | is it a game ? |
| 22:42.36 | dlynes_ | Dr-Linux: someone you idolize/someone you worship/... |
| 22:42.38 | gandhijee | kinda |
| 22:43.01 | watchy | An image used as an object of worship. |
| 22:43.01 | watchy | A false god. |
| 22:43.01 | watchy | One that is adored, often blindly or excessively. |
| 22:43.01 | watchy | Something visible but without substance. |
| 22:43.17 | Dr-Linux | :S |
| 22:43.20 | dlynes_ | watchy: sounds like the brains of half the hollywood actors :) |
| 22:43.21 | nahirean | the last part is exactly correct about those douchebags on AI |
| 22:43.27 | watchy | dlynes: haha |
| 22:43.30 | dlynes_ | visible, but without substance :) |
| 22:43.42 | watchy | i agree |
| 22:44.01 | watchy | anyone here bet sports? |
| 22:44.05 | dlynes_ | like that one airhead |
| 22:44.11 | dlynes_ | the one from married with children |
| 22:44.17 | watchy | christina applegate |
| 22:44.22 | watchy | shes cute. |
| 22:44.23 | dlynes_ | yeah...that's the ho |
| 22:44.31 | Dr-Linux | watchy: go to betfair.com :) |
| 22:44.41 | gandhijee | i'd still bang that ho |
| 22:44.43 | dlynes_ | i thought she looked slutty, not cute :) |
| 22:44.58 | gandhijee | she's cute in a slutty way |
| 22:45.00 | watchy | never tried betfair.com |
| 22:45.06 | watchy | i used sportsbook.com and betonsports.com |
| 22:45.19 | watchy | how longs it take to get paid by betfair? |
| 22:45.41 | TedC | Anyone seen an RTP codec 95 before? I'm getting it from a tekelec soft switch over SIP. |
| 22:45.45 | Dr-Linux | watchy: it's best and quick one. |
| 22:45.54 | CoffeeIV_ | dlynes: the problem with that Goto is the leading space on the condition -- it makes the condition be " 0" in the false case instead of just "0" |
| 22:46.01 | dlynes_ | TedC: yeah...lotsa times |
| 22:46.08 | TedC | dlynes: what is it? |
| 22:46.08 | watchy | dr: what do you bet on |
| 22:46.13 | dlynes_ | TedC: i get it from a sansay vsx |
| 22:46.21 | dlynes_ | TedC: i just ignore it |
| 22:46.48 | dlynes_ | CoffeeIV_: ah...never used spaces in there...wouldn't know |
| 22:46.49 | Dr-Linux | watchy: 3 year before i tried, Cricket and Soccar |
| 22:46.54 | TedC | dlynes_: it seems to be conincedent with audio drops, although I'm not 100% sure. I also don't get it with G.729. |
| 22:46.59 | watchy | ive never even seen a cricket game |
| 22:47.14 | TedC | Oh well. otherwise, the call seems to work ok. |
| 22:47.16 | Dr-Linux | watchy: really? |
| 22:47.17 | Dr-Linux | woww |
| 22:47.20 | dlynes_ | TedC: not really...it's just a codec that they're sending, and it's a codec that doesn't have a reserved number |
| 22:47.35 | TedC | dlynes: but you don't know what codec it is? |
| 22:47.35 | dlynes_ | TedC: and they're not sending the name along wiht it, liek they're supposed to be |
| 22:47.40 | watchy | dr: its not a very popular sport here. |
| 22:47.42 | Dr-Linux | Cricket is one of papular game |
| 22:47.47 | dlynes_ | TedC: otherwsie you'd know what codec it is |
| 22:47.58 | TedC | dlynes: right. Thanks. |
| 22:48.14 | dlynes_ | TedC: only way to really find out is to debug the sip transmission, and then dig down into the data and analyze it |
| 22:48.22 | dlynes_ | TedC: it's not worth it for me |
| 22:48.27 | Dr-Linux | watchy: yeah, maybe but according to Pakistan/Uk/India that's best side |
| 22:48.32 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 22:48.34 | TedC | makes sense. Yeah. Low level RTP debugging doesn't sound like fun. |
| 22:48.38 | Dr-Linux | American supports are different |
| 22:48.50 | dlynes_ | TedC: besides...i don't know enough about the codec data to know what looks like g729, what looks like g723, and so on and so froth |
| 22:48.57 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 22:49.32 | dlynes_ | I guess gandhijee thinks pretty highly of himself :) |
| 22:50.02 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 22:50.11 | ManxPower | Is it safe yet? |
| 22:50.18 | Dr-Linux | what's the difference between Digium TExxp card and Sangoma a20/t1 cards? |
| 22:50.24 | dlynes_ | There were some crackers after you, ManxPower? |
| 22:50.38 | ManxPower | dlynes, noobs |
| 22:50.41 | dlynes_ | ah |
| 22:50.43 | dlynes_ | hahaha |
| 22:51.08 | dlynes_ | Dr-Linux: price, reliability, number of interrupts generated, all kindsa things different |
| 22:51.33 | dlynes_ | Dr-Linux: also country of origin is different, too |
| 22:51.35 | Dr-Linux | dlynes_: which one is the best? |
| 22:52.03 | dlynes_ | Dr-Linux: I don't know...I haven't had a chance to try sangoma yet...I've got a sangoma 4 port fxo on order |
| 22:52.19 | Dr-Linux | Digium dual port card cost is 891 USD and Sangoma 775 USD |
| 22:52.23 | dlynes_ | Dr-Linux: but judging from everyone else that talks about them, they're the total shizzit |
| 22:52.52 | Dr-Linux | shizzit? |
| 22:53.01 | dlynes_ | da bomb |
| 22:53.22 | Dr-Linux | :S |
| 22:53.39 | dlynes_ | dood |
| 22:53.51 | Dr-Linux | dlynes_: simply Digium hardware is widly prefeable or sangoma? |
| 22:53.52 | dlynes_ | we've gotta get you an american piece of tail so you can improve your English :) |
| 22:54.16 | dlynes_ | Dr-Linux: there's advantages to both |
| 22:54.35 | Dr-Linux | lolz yeah |
| 22:54.52 | Dr-Linux | dlynes_: but if you need one dual card, which one you will buy? |
| 22:55.00 | Dr-Linux | and money is no problem for you? |
| 22:55.26 | dlynes_ | Dr-Linux: probably the sangoma 2 port with hardware echo canceller card |
| 22:55.29 | *** part/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 22:55.38 | dlynes_ | Dr-Linux: but like i said...i havne't tried sangoma yet, so i don't know |
| 22:55.45 | *** join/#asterisk heka (n=heka@82.114.68.123) |
| 22:56.03 | dlynes_ | Dr-Linux: the nice thing about digium is they'll fall hand over foot giving you support for their pri cards |
| 22:56.12 | sevard | dlynes_: sorry, i asked you a question and then my isp went down |
| 22:56.15 | sevard | they had a router issue |
| 22:56.15 | dlynes_ | Dr-Linux: you won't get that level of support from anyone else |
| 22:56.30 | dlynes_ | sevard: i don't remember what the question was....life goes on |
| 22:56.36 | sevard | :) |
| 22:56.45 | sevard | dlynes_: i'm looking forward to learning PRI pretty soon |
| 22:57.11 | Dr-Linux | dlynes_: do you think digium support is good? |
| 22:57.23 | dlynes_ | Dr-Linux: no...i think it's excellent |
| 22:57.24 | Dr-Linux | dlynes_: i'm asking bcoz i'm going to buy cards today |
| 22:57.43 | sevard | Dr-Linux: I'd have to agree with dlynes_ |
| 22:58.06 | sevard | infact, I've never even had to go to their tech support about a card. not yet anyway. the sales guy answered all the questions I had. |
| 22:58.14 | Dr-Linux | dlynes_: what kind of support digium provides for their hardware users ? |
| 22:58.26 | dlynes_ | sangoma might be better quality than digium (from what i've heard), but their level of support couldn't possibly approach digium's |
| 22:58.59 | dlynes_ | i don't know how digium can afford to offer cards that are somewhat price competitive with sangoma, yet have the level of support they have |
| 22:59.30 | dlynes_ | Dr-Linux: they'll ssh into your box if you're having troubles and set up the card for you |
| 22:59.31 | sevard | dlynes_: have you heard about this * buisness addition, it looks like they give you 5 hours of support and a handbook for 900 bucks |
| 22:59.47 | Dr-Linux | dlynes_: only support differnece from digium side? |
| 22:59.59 | dlynes_ | sevard: yeah, i've heard about it...there's even one or two people i've seen on here that are using it |
| 23:00.14 | dlynes_ | Dr-Linux: only difference between the two is quality and support? |
| 23:00.24 | sevard | dlynes_: do you know anything about it? am i right on the assumption that it's a fricken waste of money even for a buisness? |
| 23:00.27 | dlynes_ | Dr-Linux: well, and price? is that what you're asking? |
| 23:00.34 | dlynes_ | sevard: no idea |
| 23:01.06 | Dr-Linux | Sangoma gives 5 year gaurantee, but not sure about digium :S |
| 23:01.10 | Dr-Linux | dlynes_: yes |
| 23:01.11 | dlynes_ | sevard: but i thought it included a single port pri card or something too? |
| 23:01.26 | Dr-Linux | i'm not sure where should i go to buy :S |
| 23:01.27 | dlynes_ | Dr-Linux: digium is lifetime, afaik |
| 23:01.41 | Dr-Linux | dlynes_: are you sure? |
| 23:01.52 | Dr-Linux | i don't think so |
| 23:01.55 | dlynes_ | Dr-Linux: they'll just replace it |
| 23:02.11 | sevard | dlynes_: i didn't see anything about that |
| 23:02.28 | dlynes_ | Dr-Linux: if you want to know what the policy is for sure |
| 23:02.35 | sevard | dlynes_: i saw "tested and proven" and "5 hours of support" and "handbook" |
| 23:02.40 | dlynes_ | Dr-Linux: try asking mog...he's in here and asterisk-dev regularly |
| 23:02.47 | *** join/#asterisk bblack (n=mbarb@38-118-35-20.rapiddsl.net) |
| 23:02.55 | Dr-Linux | dlynes_: as i saw on voipsupply.com, if you buy digium card with with 1 year gaurantee you will pay 89 extra $$$ |
| 23:02.57 | dlynes_ | Dr-Linux: He used to be in tech support...he's a programmer for digium now |
| 23:03.04 | *** join/#asterisk blebleble (n=ble@d149-67-206-171.col.wideopenwest.com) |
| 23:03.05 | kimosabe | how is any one here doing qos |
| 23:03.15 | sevard | with qos |
| 23:03.17 | Dr-Linux | who is mog ? :S |
| 23:03.34 | kimosabe | sevard what u doing qos with |
| 23:03.36 | dlynes_ | Dr-Linux: mog_work and mog_work |
| 23:03.37 | blebleble | Can anyone suggest a really good faxing sollution, it seems every provider that i have its always hit and miss, can't ever get any real good reliability out of it |
| 23:03.52 | sevard | kimosabe: with qos |
| 23:03.59 | jlewis | I'm seeing this cisco phone keep trying to register with a From: of <sip:username@serverip;user=phone> |
| 23:04.08 | Dr-Linux | mog_home: active? |
| 23:04.14 | mog_home | nope |
| 23:04.15 | bblack | hi all. anybody can explain me what's the functionality of SetCallerID(${CALLERID}) ? |
| 23:04.15 | Dr-Linux | mog_work: active? |
| 23:04.18 | dlynes_ | lol |
| 23:04.21 | jlewis | if I search for user=phone, it seems to confuse google and other search engines |
| 23:04.23 | kimosabe | severad what are u doing qos with i have choppy voice isuses |
| 23:04.25 | Dr-Linux | ok sorry |
| 23:04.38 | distortion | blebleble: theres no such thing w/voip |
| 23:04.51 | blebleble | distortion: yah thats what i'm comming to realize? |
| 23:04.53 | sevard | kimosabe: with qos |
| 23:04.53 | jlewis | could that have anything to do with why this peer can't authenticate even though they have the username and secret right? |
| 23:04.55 | blebleble | is there a best alternative? |
| 23:04.57 | russellb | i believe you get 2 years of support, officially |
| 23:05.02 | distortion | blebleble: 80% is about the best you can get on g711 passthrough |
| 23:05.07 | mog_home | but we help anyone we can usually |
| 23:05.12 | blebleble | sevard: this feels like 'who's on first' |
| 23:05.13 | kimosabe | i got choppy voice isues |
| 23:05.18 | distortion | blebleble: if you can get t38 to work that will be about 90% |
| 23:05.22 | russellb | mog_home: word |
| 23:05.22 | Dr-Linux | i see |
| 23:05.30 | sevard | blebleble: what's on second. |
| 23:05.44 | dlynes_ | russellb: i take it employees of digium get shares? |
| 23:05.52 | mog_home | i mean i have never asked someone when they bought their card in the 8 months i worked in support, except for rma sillyness |
| 23:06.11 | Dr-Linux | russellb: and what's cards gaurantee i'm going to buy TE210P two cards |
| 23:06.25 | sevard | why not get a TE410P |
| 23:06.28 | blebleble | I'm not asking you who's on second |
| 23:06.28 | russellb | Dr-Linux: I just said, the official time is 2 years |
| 23:06.32 | sevard | instead of two TE210Ps |
| 23:06.52 | sevard | blebleble: no, what's on second. |
| 23:06.55 | Dr-Linux | but you said that for support |
| 23:07.32 | blebleble | I'm not asking you who's on second |
| 23:07.34 | russellb | well, they generally help whoever they can |
| 23:07.35 | Dr-Linux | sevard: actually we need only two ports, 1 from each card |
| 23:07.42 | russellb | for RMA, 2 years, i believe |
| 23:07.49 | russellb | but don't take this is as an official statement :) |
| 23:08.13 | *** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 23:08.13 | Dr-Linux | other will be for radundancy |
| 23:08.33 | dlynes_ | Dr-Linux: try emailing sales@digium.com to get an official statement of warranty coverage |
| 23:08.34 | niter3 | hey guys, is there a way i can set it so it will tell me the date and time messages were left in my voicemail? |
| 23:08.56 | dlynes_ | niter3: it does that automatically |
| 23:08.56 | *** join/#asterisk Cresl1n (n=matt@host56-92.pool8838.interbusiness.it) |
| 23:08.57 | niter3 | dlynes_: It's not though |
| 23:09.06 | sevard | radundancy |
| 23:09.08 | sevard | lulz |
| 23:09.45 | dlynes_ | niter3: not even when you put in saycid=yes? |
| 23:09.59 | niter3 | dlynes_: which file do I put that in? |
| 23:10.05 | dlynes_ | niter3: voicemail.conf? |
| 23:10.26 | Dr-Linux | sevard: we will use them for IVRs |
| 23:10.58 | russellb | i've been coding all week, and not sleeping enough |
| 23:11.03 | niter3 | dlynes_: hrm, i set it now |
| 23:11.08 | niter3 | do i need to restart asterisk ? |
| 23:11.18 | dlynes_ | niter3: a simple reload might be enouhg |
| 23:11.29 | dlynes_ | i.e. type 'reload' at the asterisk cli |
| 23:11.31 | russellb | we've been having an Asterisk developer conference in Italy this week :) -- super productive |
| 23:11.42 | dlynes_ | cool |
| 23:11.43 | Dr-Linux | niter3: just restart specific module rather than entire Asterisk service |
| 23:11.43 | sevard | Dr-Linux: I just thought 'radundancy' was the coolest thing since Rad Cakes |
| 23:11.54 | dlynes_ | Dr-Linux: voicemail doesn't have a reload |
| 23:12.10 | niter3 | dlynes_: ok done. |
| 23:12.13 | niter3 | i'm using asterisk@home |
| 23:12.14 | Dr-Linux | dlynes_: it have |
| 23:12.18 | Kyler | Is there a known problem with having a bunch (15) of virtual ethernet interfaces (eth0:x) in the dom0? I've been stable for weeks until I added them and it croaked. |
| 23:12.20 | niter3 | so it had a file called asterisk.reload |
| 23:12.25 | dlynes_ | ah |
| 23:12.31 | Dr-Linux | niter3: just do it >> reload app_voicemail.so |
| 23:12.34 | Kyler | Oops! I just sent that to the wrong channel. Sorry! |
| 23:12.59 | Dr-Linux | niter3: join #freepbx |
| 23:13.23 | dlynes_ | Dr-Linux: he's not asking about freepbx specific help, though |
| 23:13.32 | niter3 | Dr-Linux: thanks |
| 23:13.33 | *** part/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 23:13.42 | bblack | is SetCallerID(${CALLERID}) doing anything? my impression is that is setting the caller id on the channel from the caller id on the same channel... am I wrong? |
| 23:14.15 | dlynes_ | bblack: Set(${CALLERID(num)}="84902384932") ; |
| 23:14.21 | Dr-Linux | sevard : you were laughing due to my bad english? |
| 23:14.26 | Dr-Linux | or bad spell? |
| 23:14.44 | bblack | dlynes: of course, but I saw places where they suggest doing SetCallerID(${CALLERID}), like the Voxee support page |
| 23:14.49 | Dr-Linux | and what's Rad Cakes? |
| 23:14.54 | russellb | dlynes_: that's not correct |
| 23:15.07 | russellb | Set(CALLERID(num)=whatever) |
| 23:15.18 | dlynes_ | oh yeah...forgot...no quotes...no variables |
| 23:15.38 | dlynes_ | sorry...thanks for correcting me, russellb |
| 23:15.45 | sevard | Dr-Linux: apparently both |
| 23:15.55 | russellb | just trying to help avoid confusion |
| 23:15.59 | bblack | right... my problem is that I cannot make voxee caller id work reliable and their suggestion seems meaningless to me |
| 23:16.03 | dlynes_ | bblack: the setcallerid is the old asterisk 1.0.x method of doing things |
| 23:16.21 | CoffeeIV_ | is there a php agi for PHP 5 ? |
| 23:16.27 | russellb | bblack: yes, that doesn't make any sense |
| 23:16.27 | distortion | cute, new asterisk install has ascii art |
| 23:16.38 | russellb | distortion: lol, yeah, i did that |
| 23:16.42 | distortion | haha |
| 23:16.44 | dlynes_ | distortion: the trunk, eh? |
| 23:16.53 | Dr-Linux | sevard: i'm here to learning asterisk stuff, but donot care about english. |
| 23:16.54 | distortion | trying out the new t38 version, ya |
| 23:17.01 | sevard | Dr-Linux: english first. |
| 23:17.03 | dlynes_ | distortion: yeah...i thought it was a cannabis leaf coming down the screen the first time i seen it :) |
| 23:17.24 | *** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 23:17.29 | Dr-Linux | sevard: not in my country |
| 23:17.37 | niter3 | ok saycid=yes just told me the # that called. I need date and time |
| 23:17.43 | bblack | russleb: thanks |
| 23:17.44 | bblack | anybody have voxee caller id working reliable? |
| 23:17.45 | Dr-Linux | it's not my native language |
| 23:18.20 | sevard | Dr-Linux: english is the language of programmers and computer professionals. Even linus agrees. |
| 23:18.21 | Dr-Linux | but i learnt enough english to talk to you guys :) |
| 23:19.56 | Dr-Linux | sevard: i know english is neccessary, i wish i know well. however you should not laugh on someone :) |
| 23:20.38 | Dr-Linux | sevard: i'm sure you won't know even a word on my native language. |
| 23:20.58 | *** join/#asterisk danilom (n=danilom@201.209.43.241) |
| 23:21.33 | blebleble | what country Dr-Linux? |
| 23:21.42 | danilom | good evenig |
| 23:21.56 | Dr-Linux | dlynes_: wft is clones T1 cards? |
| 23:22.05 | Dr-Linux | blebleble: Pk |
| 23:22.05 | danilom | X100p works like a analog line? |
| 23:22.18 | dlynes_ | danilom: x100p is an analog line card |
| 23:22.58 | dlynes_ | Dr-Linux: hanjee guddad |
| 23:23.15 | Dr-Linux | dlynes: Lasi :P |
| 23:23.44 | danilom | dlynes: can i connect the asterisk with a PBX? |
| 23:24.07 | dlynes_ | Dr-Linux: the clone cards are probably crap...when i'm spending that much money on a card, i want to be guaranteed it's going to work |
| 23:24.27 | dlynes_ | danilom: you mean to a conventional pbx, or a conventional keysystem? |
| 23:24.29 | Dr-Linux | dlynes_: hehe yeah, |
| 23:24.53 | dlynes_ | Dr-Linux: personally, i wouldn't want to bother trying anything but digium or sangoma |
| 23:24.56 | danilom | dlynes: a conventional PBX! |
| 23:25.31 | dlynes_ | danilom: yeah...you can use either a single, double, or quad span t1/e1 line card |
| 23:25.43 | dlynes_ | danilom: or you can use multiport fxs atas |
| 23:25.53 | Dr-Linux | dlynes_: my manager just said, your time is most vauleable, let's not waist it to think about $200 and let's buy Digium :) |
| 23:26.15 | dlynes_ | well, he's right |
| 23:26.26 | dlynes_ | Why spend three hours trying to troubleshoot a clone |
| 23:26.40 | danilom | dlynes: I have X100P diginet, can i use it for that? |
| 23:26.42 | dlynes_ | when you can spend an extra $100 or $200 and get a card that's guaranteed to work? |
| 23:26.48 | Dr-Linux | dlynes_: for Dialogic cards digium force pplz to buy driver from them. why not same for Sangoma? |
| 23:27.02 | dlynes_ | danilom: x100p is an fxo card, not an fxs card |
| 23:27.22 | dlynes_ | Dr-Linux: huh? |
| 23:27.37 | dlynes_ | Dr-Linux: can you reword that? I didn't understand what you were trying to say |
| 23:27.41 | Dr-Linux | dlynes_: he was talking about sangoma/digium |
| 23:27.41 | sevard | Dr-Linux: I don't need to. |
| 23:27.49 | russellb | Dr-Linux: no, Digium does not force people to buy the dialogic drivers |
| 23:27.56 | russellb | Dr-Linux: that is entirely an Intel decision |
| 23:27.57 | Dr-Linux | dlynes_: do you know about Dialogic card/board ? |
| 23:28.06 | dlynes_ | Dr-Linux: yes...it's made by intel |
| 23:28.40 | dlynes_ | Dr-Linux: dialogic is probably most well known for their conventional pbx compatability cards |
| 23:29.10 | danilom | dlynes_ i need to connect two offices using asterisk, each office had a standar panasonic PBX |
| 23:29.10 | dlynes_ | Dr-Linux: the problem with those cards is that they're so expensive, it makes using that route not cost-viable |
| 23:29.10 | Dr-Linux | dlynes_: currently our all callcenters/IVR/setups are on TV, now we are moving everything on Asterisk. |
| 23:29.25 | dlynes_ | danilom: Panasonic what? |
| 23:29.27 | danilom | dlynes_ what i need a fxo or fxs? |
| 23:29.36 | dlynes_ | danilom: TDA-30? |
| 23:30.08 | dlynes_ | danilom: tda-500? |
| 23:30.28 | dlynes_ | or is it much older? |
| 23:30.38 | Dr-Linux | dlynes_: yeah the are very expensive we are using them with TV . And TV license is also very expensive |
| 23:30.42 | dlynes_ | Dr-Linux: no idea what 'TV' is |
| 23:30.57 | *** join/#asterisk opus_ (n=opus@68.216.187.60) |
| 23:31.02 | Dr-Linux | dlynes_: TeleVantage |
| 23:31.05 | dlynes_ | oh |
| 23:31.12 | dlynes_ | we've got one in our closet collecting dust :) |
| 23:31.13 | opus_ | sorry i just asked the wrong channel, but does anyone know if there is a 'sorry the system is currently down for maintance' recording? |
| 23:31.30 | danilom | dlynes_ panasonic 616 |
| 23:31.31 | Dr-Linux | dlynes_: you mean TV? |
| 23:31.42 | dlynes_ | Dr-Linux: yeah...you already answered my question |
| 23:31.50 | dlynes_ | danilom: yeah...so it's one of the older ones |
| 23:31.58 | dlynes_ | danilom: that's actually a keysystem, not a pbx |
| 23:32.34 | dlynes_ | danilom: anyways...How many incoming lines at each premise? 1, or 2? |
| 23:32.41 | Dr-Linux | dlynes_: actually our only problem was/is operator consol |
| 23:33.00 | dlynes_ | Dr-Linux: televantage is also a voip system |
| 23:33.04 | danilom | dlynes_ one line for each PBX to call each other |
| 23:33.17 | dlynes_ | danilom: and that's voip, or analog? |
| 23:33.24 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 23:33.29 | kink0 | hello, good night |
| 23:33.41 | Dr-Linux | dlynes_: yeah currently we are using TV live. but it will out soon |
| 23:33.58 | Dr-Linux | kink0: hey my spanish friend, where're you going? |
| 23:34.04 | danilom | dlynes_ i want to connect that system to VOIP |
| 23:34.17 | Dr-Linux | Opss you are coming :P |
| 23:34.20 | dlynes_ | danilom: so you're going to do iax2 trunking between the two then? |
| 23:34.33 | kink0 | I get FAILED instead BUSY at CDR when Zap/2-1 is circuit-busy, that happens apparentely after changed nothing from last two days |
| 23:34.46 | kink0 | hi Dr-Linux !! a new mistake with asterisk and cdr |
| 23:35.10 | dlynes_ | kink0: failed is the correct log entry for that |
| 23:35.21 | Dr-Linux | kink0: aww you did or it's broken by itself? |
| 23:35.26 | kink0 | dlynes_, but two days ago I got "BUSY" |
| 23:35.33 | dlynes_ | kink0: it's failed because there's no available lines |
| 23:35.49 | danilom | dlynes_ I only connect both PBX with a cheaper card |
| 23:36.02 | kink0 | dlynes_ hmmm , no no available lines, the busy is the called part |
| 23:36.04 | dlynes_ | kink0: it should only be busy if you call somewhere and get a busy signal |
| 23:36.14 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 23:36.31 | kink0 | dlynes_ that is what I did, I call to any site with my mobile, and then call my mobile from Asterisk |
| 23:36.44 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 23:36.45 | dlynes_ | danilom: yeah, what i would do, is do iax2 trunking between the two asterisk boxes |
| 23:36.55 | kink0 | so i was expecting to get "BUSY", but no... since two days there no any BUSY, but FAILED |
| 23:37.22 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:37.30 | dlynes_ | danilom: then use up to 4 sipura 2000/2002/pap2-na's to provide line service to the panasonic keysystems |
| 23:37.31 | Dr-Linux | kink0: friend i understand the problem, but can't explain in english :) |
| 23:37.33 | kink0 | dlynes_ I have gotten no one BUSY and over 50 failed this afternoon, when two days ago I got BUSY and very little FAILED for other reasons |
| 23:37.46 | Dr-Linux | justinu|laptop wb :) |
| 23:37.53 | kink0 | Dr-Linux, try in spanish :) |
| 23:38.00 | dlynes_ | kink0: there's probably an issue with your zaptel card |
| 23:38.12 | dlynes_ | kink0: for why it's getting circuit busy |
| 23:38.44 | kink0 | dlynes_ may be, in what ISDN CAUSE is getting. I will remove/reload the module and try again. |
| 23:38.45 | dlynes_ | kink0: circuit busy to me would suggest a reorder tone or a yellow alarm |
| 23:38.57 | dlynes_ | kink0: probably a yellow alarm |
| 23:39.01 | Dr-Linux | kink0: heh i know Spanish, then there is no need we have setup our spanish call center in damn Dominicon Republic :) |
| 23:39.04 | kink0 | dlynes_ no... no yellow alarm, is ok the PRI |
| 23:39.16 | dlynes_ | kink0: which zaptel version are you using? |
| 23:39.24 | dlynes_ | kink0: when you dial into the pri, do you get a busy signal? |
| 23:39.30 | kink0 | dlynes_ I am ussing a Digium TE405 , zaptel 1.2.5 |
| 23:39.30 | dlynes_ | kink0: or a reorder tone? |
| 23:39.54 | dlynes_ | kink0: when you try to dial out do you get the same problem as dialing in? |
| 23:39.59 | kink0 | dlynes_ I get busy only if really the called party is busy, but Asterisk reports in CDR FAILED instead BUSY |
| 23:40.17 | danilom | dlynes_ ok what type of card i need to connect the asterisk with the PBX? |
| 23:40.20 | kink0 | dlynes_ I have not dialed in, but dialing out is going fine if the other party is ready |
| 23:40.29 | dlynes_ | kink0: check your indications.conf file then |
| 23:40.44 | dlynes_ | kink0: compare it against the indications.conf file from the distribution |
| 23:40.52 | dlynes_ | kink0: you might find that it's changed |
| 23:40.58 | kink0 | dlynes_ ok, I will check, anywise I checkec all config files changed from last may 8 |
| 23:41.13 | kink0 | no, I did not touch indications.conf |
| 23:41.22 | dlynes_ | kink0: indications.conf is where the tones are defined for the different tone patterns on the line |
| 23:42.02 | kink0 | yes, I know, but since problem appears from 2 days, I did a check what files have been changed , and I have not found any related to this problem |
| 23:42.18 | dlynes_ | danilom: Like I said...try the linksys pap2-na's |
| 23:42.34 | dlynes_ | danilom: they're apparently more reliable than fxs ports on tdm400p cards |
| 23:44.08 | danilom | dlynes_ the linksys pap2-naś is the sipura? |
| 23:44.52 | dlynes_ | danilom: first generation: sipura SPA-2000, second generation: Sipura SPA-2002, third generation: Linksys PAP2-NA |
| 23:45.07 | dlynes_ | it's all the same unit |
| 23:45.17 | dlynes_ | just different packaging, and different default firmware |
| 23:45.24 | Dr-Linux | what about Sipura 3000 ? |
| 23:45.43 | dlynes_ | the 2002 and pap2-na's i think might have double the memory of the old sipura 2000 |
| 23:45.52 | Dr-Linux | i have spa-2100 |
| 23:46.04 | dlynes_ | sipura 3000 is an fxo gateway; it's not an ata |
| 23:46.16 | dlynes_ | spa-2100 is the spa2002 built into a router |
| 23:46.18 | Dr-Linux | yep |
| 23:48.42 | kink0 | dlynes_ peer Asterisk reports BUSY, but gateway Asterisk reports FAILED |
| 23:49.07 | danilom | dlynes_ ok and the FXO card it doesnt work? |
| 23:49.14 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 23:50.07 | kink0 | I got : Zap/1-1 is circuit-busy in CLI, but CDR reports always FAILED |
| 23:51.06 | brookshire | i dunno of any sipura that is actually reliable :( |
| 23:51.57 | *** join/#asterisk mog_home (n=mogorman@host56-92.pool8838.interbusiness.it) |
| 23:52.25 | opus_ | kink0 yeah its pretty bad |
| 23:53.34 | dlynes_ | danilom: the x100p works...not terribly well, but it does work...ymmv...but it's an fxo...it won't help you for interfacing to the panasonic...only for interfacing to the pots line |
| 23:53.47 | dlynes_ | danilom: you need fxs to interface to the panasonic |
| 23:54.00 | dlynes_ | anyways...i've gotta run...gotta be somewhere in 40 minutes |
| 23:54.43 | kink0 | opus_ that happens since two days, when I have modify nothing relative to this issue, just modified files was sip.conf and h323.conf extensions.conf |
| 23:54.46 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 23:55.16 | kink0 | even I have stoped, unloaded zaptel, reload again and restarted, even reseted the other end PRI port |
| 23:55.27 | brookshire | did you reboot? |
| 23:55.27 | kink0 | but no sense. |
| 23:55.43 | opus_ | kink0 learn gdb :) |
| 23:56.14 | kink0 | heyyyy !! debug asterisk is a lot of debuging !! |
| 23:56.22 | brookshire | unloading zaptel and then reloading the modules doesn't make it actually reload! |
| 23:56.34 | danilom | dlynes_ ok thanks |
| 23:57.03 | kink0 | brookshire, hmmmmmmmm, i checked was not present ussing lsmod. |
| 23:57.20 | kink0 | brookshire, do you suggest this issue requires a hard reset ? |
| 23:57.32 | brookshire | kink0: it may, i wouldn't be surprised |
| 23:58.01 | brookshire | if that doesn't fix it, i would contact support for the card |
| 23:58.07 | kink0 | brookshire, anywise Asterisk at CLI reports as busy, but when cdr, then reports as failed |
| 23:58.33 | brookshire | what version are you running? |
| 23:58.50 | FuriousGeorge | so a guy hijacked and disconnected a line that the alrm unit shares. i didnt know that the line actually came in from his side of the building, so when i didnt hear anything on the NID, i called the telco, who went and hooked it up on our side. now the alarm on his side the building has no line. i put a tone to check on the pair, and i picked it up on the other end, but i guess it must still be going through the dead NID on the ot |
| 23:58.50 | kink0 | Asterisk 1.2.27 and zaptel 1.2.25 |
| 23:58.54 | FuriousGeorge | does that sound right? |
| 23:59.00 | brookshire | no |
| 23:59.10 | brookshire | 1.2.27 doesn't exist |
| 23:59.50 | kink0 | sorry.. mistyping... 1.2.7 and 1.2.5 |
| 23:59.57 | opus_ | kink0 try downgrading to something before 1.2.7.1, somebody changed the api in the 1.2.7.1 code that might affect your code |
